update to webrtc revision 1349
Updated audio processing modules from revision 180 to 1349.
Main changes are:
- code clean up and reformating
- source path reorganization
- improved performance
Also imported test code that was not included in initial
drop from webrtc.
Change-Id: Ie4eb0e29990052e5f2d7f0b271b42eead40dbb6a
diff --git a/Android.mk b/Android.mk
index aa90dd3..dc214e9 100644
--- a/Android.mk
+++ b/Android.mk
@@ -8,23 +8,60 @@
MY_WEBRTC_ROOT_PATH := $(call my-dir)
-
+# voice
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
+#include $(MY_WEBRTC_ROOT_PATH)/src/modules/utility/source/Android.mk
include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
-# audio processing
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing_library/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
-
-
# build .so
-include $(MY_WEBRTC_ROOT_PATH)/android-webrtc.mk
+LOCAL_PATH := $(call my-dir)
-# build test apps
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/process_test/Android.mk
+include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../external/webrtc/android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_audio_preprocessing
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libwebrtc_spl \
+ libwebrtc_resampler \
+ libwebrtc_apm \
+ libwebrtc_apm_utility \
+ libwebrtc_vad \
+ libwebrtc_ns \
+ libwebrtc_agc \
+ libwebrtc_aec \
+ libwebrtc_aecm \
+ libwebrtc_system_wrappers
+
+# Add Neon libraries.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_WHOLE_STATIC_LIBRARIES += \
+ libwebrtc_aecm_neon \
+ libwebrtc_ns_neon
+endif
+
+LOCAL_STATIC_LIBRARIES := \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/NOTICE b/NOTICE
index 1c39e5b..f96164a 100644
--- a/NOTICE
+++ b/NOTICE
@@ -108,4 +108,59 @@
Scott McMurray
*/
+===============================================================================
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+//
+// Redistribution and use in source and binary forms, with or without
+// modification, are permitted provided that the following conditions are
+// met:
+//
+// * Redistributions of source code must retain the above copyright
+// notice, this list of conditions and the following disclaimer.
+// * Redistributions in binary form must reproduce the above
+// copyright notice, this list of conditions and the following disclaimer
+// in the documentation and/or other materials provided with the
+// distribution.
+// * Neither the name of Google Inc. nor the names of its
+// contributors may be used to endorse or promote products derived from
+// this software without specific prior written permission.
+//
+// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+===============================================================================
+
+// (C) Copyright Greg Colvin and Beman Dawes 1998, 1999.
+// Copyright (c) 2001, 2002 Peter Dimov
+//
+// Permission to copy, use, modify, sell and distribute this software
+// is granted provided this copyright notice appears in all copies.
+// This software is provided "as is" without express or implied
+// warranty, and with no claim as to its suitability for any purpose.
+//
+// See http://www.boost.org/libs/smart_ptr/scoped_ptr.htm for documentation.
+//
+
+// scoped_ptr mimics a built-in pointer except that it guarantees deletion
+// of the object pointed to, either on destruction of the scoped_ptr or via
+// an explicit reset(). scoped_ptr is a simple solution for simple needs;
+// use shared_ptr or std::auto_ptr if your needs are more complex.
+
+// scoped_ptr_malloc added in by Google. When one of
+// these goes out of scope, instead of doing a delete or delete[], it
+// calls free(). scoped_ptr_malloc<char> is likely to see much more
+// use than any other specializations.
+
+// release() added in by Google. Use this to conditionally
+// transfer ownership of a heap-allocated object to the caller, usually on
+// method success.
diff --git a/android-webrtc.mk b/android-webrtc.mk
index 01d6a9e..dc92aeb 100644
--- a/android-webrtc.mk
+++ b/android-webrtc.mk
@@ -6,91 +6,44 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
-MY_APM_WHOLE_STATIC_LIBRARIES := \
- libwebrtc_spl \
- libwebrtc_resampler \
- libwebrtc_apm \
- libwebrtc_apm_utility \
- libwebrtc_vad \
- libwebrtc_ns \
- libwebrtc_agc \
- libwebrtc_aec \
- libwebrtc_aecm
+# These defines will apply to all source files
+# Think again before changing it
+MY_WEBRTC_COMMON_DEFS := \
+ '-DWEBRTC_TARGET_PC' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
+ '-DWEBRTC_ANDROID'
+# The following macros are used by modules,
+# we might need to re-organize them
+# '-DWEBRTC_ANDROID_OPENSLES' [module audio_device]
+# '-DNETEQ_VOICEENGINE_CODECS' [module audio_coding neteq]
+# '-DWEBRTC_MODULE_UTILITY_VIDEO' [module media_file] [module utility]
+ifeq ($(TARGET_ARCH),arm)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM'
+# '-DWEBRTC_DETECT_ARM_NEON' # only used in a build configuration without Neon
+# TODO(kma): figure out if the above define could be moved to NDK build only.
-LOCAL_PATH := $(call my-dir)
+# TODO(kma): test if the code under next two macros works with generic GCC compilers
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM_NEON'
+MY_ARM_CFLAGS_NEON := \
+ -flax-vector-conversions
+endif
-include $(CLEAR_VARS)
+ifneq (,$(filter '-DWEBRTC_DETECT_ARM_NEON' '-DWEBRTC_ARCH_ARM_NEON', \
+ $(MY_WEBRTC_COMMON_DEFS)))
+WEBRTC_BUILD_NEON_LIBS := true
+endif
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_audio_preprocessing
-LOCAL_MODULE_TAGS := optional
-LOCAL_LDFLAGS :=
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_ARCH_ARM_V7A'
+endif
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- $(MY_APM_WHOLE_STATIC_LIBRARIES) \
- libwebrtc_system_wrappers \
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_SHARED_LIBRARY)
-
-###
-
-#LOCAL_PATH := $(call my-dir)
-#
-#include $(CLEAR_VARS)
-#
-#LOCAL_ARM_MODE := arm
-#LOCAL_MODULE := libwebrtc
-#LOCAL_MODULE_TAGS := optional
-#LOCAL_LDFLAGS :=
-#
-#LOCAL_WHOLE_STATIC_LIBRARIES := \
-# libwebrtc_system_wrappers \
-# libwebrtc_audio_device \
-# libwebrtc_pcm16b \
-# libwebrtc_cng \
-# libwebrtc_audio_coding \
-# libwebrtc_rtp_rtcp \
-# libwebrtc_media_file \
-# libwebrtc_udp_transport \
-# libwebrtc_utility \
-# libwebrtc_neteq \
-# libwebrtc_audio_conference_mixer \
-# libwebrtc_isac \
-# libwebrtc_ilbc \
-# libwebrtc_isacfix \
-# libwebrtc_g722 \
-# libwebrtc_g711 \
-# libwebrtc_voe_core \
-# libwebrtc_video_render \
-# libwebrtc_video_capture \
-# libwebrtc_i420 \
-# libwebrtc_video_coding \
-# libwebrtc_video_processing \
-# libwebrtc_vp8 \
-# libwebrtc_vie_core \
-# libwebrtc_vplib \
-# libwebrtc_jpeg \
-# libwebrtc_vpx
-#
-#LOCAL_STATIC_LIBRARIES :=
-#LOCAL_SHARED_LIBRARIES := \
-# libcutils \
-# libdl \
-# libstlport \
-# libjpeg \
-# libGLESv2 \
-# libOpenSLES \
-# libwebrtc_audio_preprocessing
-#
-#LOCAL_ADDITIONAL_DEPENDENCIES :=
-#
-#include external/stlport/libstlport.mk
-#include $(BUILD_SHARED_LIBRARY)
+else ifeq ($(TARGET_ARCH),x86)
+MY_WEBRTC_COMMON_DEFS += \
+ '-DWEBRTC_USE_SSE2'
+endif
diff --git a/src/LICENSE b/src/LICENSE
new file mode 100644
index 0000000..4c41b7b
--- /dev/null
+++ b/src/LICENSE
@@ -0,0 +1,29 @@
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+
+ * Neither the name of Google nor the names of its contributors may
+ be used to endorse or promote products derived from this software
+ without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/src/PATENTS b/src/PATENTS
new file mode 100644
index 0000000..190607a
--- /dev/null
+++ b/src/PATENTS
@@ -0,0 +1,24 @@
+Additional IP Rights Grant (Patents)
+
+"This implementation" means the copyrightable works distributed by
+Google as part of the WebRTC code package.
+
+Google hereby grants to you a perpetual, worldwide, non-exclusive,
+no-charge, irrevocable (except as stated in this section) patent
+license to make, have made, use, offer to sell, sell, import,
+transfer, and otherwise run, modify and propagate the contents of this
+implementation of the WebRTC code package, where such license applies
+only to those patent claims, both currently owned by Google and
+acquired in the future, licensable by Google that are necessarily
+infringed by this implementation of the WebRTC code package. This
+grant does not include claims that would be infringed only as a
+consequence of further modification of this implementation. If you or
+your agent or exclusive licensee institute or order or agree to the
+institution of patent litigation against any entity (including a
+cross-claim or counterclaim in a lawsuit) alleging that this
+implementation of the WebRTC code package or any code incorporated
+within this implementation of the WebRTC code package constitutes
+direct or contributory patent infringement, or inducement of patent
+infringement, then any patent rights granted to you under this License
+for this implementation of the WebRTC code package shall terminate as
+of the date such litigation is filed.
diff --git a/src/common_audio/OWNERS b/src/common_audio/OWNERS
index 0eb967b..84582f2 100644
--- a/src/common_audio/OWNERS
+++ b/src/common_audio/OWNERS
@@ -1 +1,4 @@
-bjornv@google.com
+bjornv@webrtc.org
+tina.legrand@webrtc.org
+jan.skoglund@webrtc.org
+andrew@webrtc.org
diff --git a/src/common_audio/common_audio.gyp b/src/common_audio/common_audio.gyp
new file mode 100644
index 0000000..3d3da3f
--- /dev/null
+++ b/src/common_audio/common_audio.gyp
@@ -0,0 +1,16 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ 'signal_processing/signal_processing.gypi',
+ 'resampler/resampler.gypi',
+ 'vad/vad.gypi',
+ ],
+}
diff --git a/src/common_audio/resampler/Android.mk b/src/common_audio/resampler/Android.mk
new file mode 100644
index 0000000..b1d630a
--- /dev/null
+++ b/src/common_audio/resampler/Android.mk
@@ -0,0 +1,47 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_resampler
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := resampler.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/resampler/OWNERS b/src/common_audio/resampler/OWNERS
deleted file mode 100644
index cf595df..0000000
--- a/src/common_audio/resampler/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-bjornv@google.com
-tlegrand@google.com
-jks@google.com
diff --git a/src/common_audio/resampler/main/interface/resampler.h b/src/common_audio/resampler/include/resampler.h
similarity index 90%
rename from src/common_audio/resampler/main/interface/resampler.h
rename to src/common_audio/resampler/include/resampler.h
index a03ff18..38e6bd3 100644
--- a/src/common_audio/resampler/main/interface/resampler.h
+++ b/src/common_audio/resampler/include/resampler.h
@@ -21,6 +21,8 @@
namespace webrtc
{
+// TODO(andrew): the implementation depends on the exact values of this enum.
+// It should be rewritten in a less fragile way.
enum ResamplerType
{
// 4 MSB = Number of channels
@@ -33,6 +35,7 @@
kResamplerInvalid = 0xff
};
+// TODO(andrew): doesn't need to be part of the interface.
enum ResamplerMode
{
kResamplerMode1To1,
@@ -40,6 +43,7 @@
kResamplerMode1To3,
kResamplerMode1To4,
kResamplerMode1To6,
+ kResamplerMode1To12,
kResamplerMode2To3,
kResamplerMode2To11,
kResamplerMode4To11,
@@ -50,6 +54,7 @@
kResamplerMode3To1,
kResamplerMode4To1,
kResamplerMode6To1,
+ kResamplerMode12To1,
kResamplerMode3To2,
kResamplerMode11To2,
kResamplerMode11To4,
@@ -61,6 +66,7 @@
public:
Resampler();
+ // TODO(andrew): use an init function instead.
Resampler(int inFreq, int outFreq, ResamplerType type);
~Resampler();
diff --git a/src/common_audio/resampler/main/source/Android.mk b/src/common_audio/resampler/main/source/Android.mk
deleted file mode 100644
index b78e3af..0000000
--- a/src/common_audio/resampler/main/source/Android.mk
+++ /dev/null
@@ -1,58 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_resampler
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := resampler.cc
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/resampler/main/source/resampler.gyp b/src/common_audio/resampler/main/source/resampler.gyp
deleted file mode 100644
index 8baf870..0000000
--- a/src/common_audio/resampler/main/source/resampler.gyp
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../../../../common_settings.gypi', # Common settings
- ],
- 'targets': [
- {
- 'target_name': 'resampler',
- 'type': '<(library)',
- 'dependencies': [
- '../../../signal_processing_library/main/source/spl.gyp:spl',
- ],
- 'include_dirs': [
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
- 'sources': [
- '../interface/resampler.h',
- 'resampler.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/common_audio/resampler/main/source/resampler.cc b/src/common_audio/resampler/resampler.cc
similarity index 87%
rename from src/common_audio/resampler/main/source/resampler.cc
rename to src/common_audio/resampler/resampler.cc
index f866739..2db27b1 100644
--- a/src/common_audio/resampler/main/source/resampler.cc
+++ b/src/common_audio/resampler/resampler.cc
@@ -62,8 +62,7 @@
slave_left_ = NULL;
slave_right_ = NULL;
- int res = Reset(inFreq, outFreq, type);
-
+ Reset(inFreq, outFreq, type);
}
Resampler::~Resampler()
@@ -185,7 +184,8 @@
if ((my_type_ & 0xf0) == 0x20)
{
// Change type to mono
- type = (ResamplerType)((int)type & 0x0f + 0x10);
+ type = static_cast<ResamplerType>(
+ ((static_cast<int>(type) & 0x0f) + 0x10));
slave_left_ = new Resampler(inFreq, outFreq, type);
slave_right_ = new Resampler(inFreq, outFreq, type);
}
@@ -209,9 +209,12 @@
case 6:
my_mode_ = kResamplerMode1To6;
break;
+ case 12:
+ my_mode_ = kResamplerMode1To12;
+ break;
default:
my_type_ = kResamplerInvalid;
- break;
+ return -1;
}
} else if (outFreq == 1)
{
@@ -229,9 +232,12 @@
case 6:
my_mode_ = kResamplerMode6To1;
break;
+ case 12:
+ my_mode_ = kResamplerMode12To1;
+ break;
default:
my_type_ = kResamplerInvalid;
- break;
+ return -1;
}
} else if ((inFreq == 2) && (outFreq == 3))
{
@@ -299,6 +305,18 @@
state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
break;
+ case kResamplerMode1To12:
+ // 1:2
+ state1_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:4
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ // 4:12
+ state3_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+ WebRtcSpl_ResetResample16khzTo48khz(
+ (WebRtcSpl_State16khzTo48khz*) state3_);
+ break;
case kResamplerMode2To3:
// 2:6
state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
@@ -367,6 +385,18 @@
state2_ = malloc(8 * sizeof(WebRtc_Word32));
memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
break;
+ case kResamplerMode12To1:
+ // 12:4
+ state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+ WebRtcSpl_ResetResample48khzTo16khz(
+ (WebRtcSpl_State48khzTo16khz*) state1_);
+ // 4:2
+ state2_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+ // 2:1
+ state3_ = malloc(8 * sizeof(WebRtc_Word32));
+ memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
+ break;
case kResamplerMode3To2:
// 3:6
state1_ = malloc(8 * sizeof(WebRtc_Word32));
@@ -458,8 +488,9 @@
return 0;
}
- // Container for temp samples
+ // Containers for temp samples
WebRtc_Word16* tmp;
+ WebRtc_Word16* tmp_2;
// tmp data for resampling routines
WebRtc_Word32* tmp_mem;
@@ -545,6 +576,41 @@
free(tmp);
return 0;
+ case kResamplerMode1To12:
+ // We can only handle blocks of 40 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 40) != 0) {
+ return -1;
+ }
+ if (maxLen < (lengthIn * 12)) {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*) malloc(336 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*) malloc(sizeof(WebRtc_Word16) * 4 * lengthIn);
+ //1:2
+ WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut,
+ (WebRtc_Word32*) state1_);
+ outLen = lengthIn * 2;
+ //2:4
+ WebRtcSpl_UpsampleBy2(samplesOut, outLen, tmp, (WebRtc_Word32*) state2_);
+ outLen = outLen * 2;
+ // 4:12
+ for (int i = 0; i < outLen; i += 160) {
+ // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
+ // as input and outputs a resampled block of 480 samples. The
+ // data is now actually in 32 kHz sampling rate, despite the
+ // function name, and with a resampling factor of three becomes
+ // 96 kHz.
+ WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
+ (WebRtcSpl_State16khzTo48khz*) state3_,
+ tmp_mem);
+ }
+ outLen = outLen * 3;
+ free(tmp_mem);
+ free(tmp);
+
+ return 0;
case kResamplerMode2To3:
if (maxLen < (lengthIn * 3 / 2))
{
@@ -783,6 +849,43 @@
free(tmp);
outLen = outLen / 2;
return 0;
+ case kResamplerMode12To1:
+ // We can only handle blocks of 480 samples
+ // Can be fixed, but I don't think it's needed
+ if ((lengthIn % 480) != 0) {
+ return -1;
+ }
+ if (maxLen < (lengthIn / 12)) {
+ return -1;
+ }
+
+ tmp_mem = (WebRtc_Word32*) malloc(496 * sizeof(WebRtc_Word32));
+ tmp = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
+ tmp_2 = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 6);
+ // 12:4
+ for (int i = 0; i < lengthIn; i += 480) {
+ // WebRtcSpl_Resample48khzTo16khz() takes a block of 480 samples
+ // as input and outputs a resampled block of 160 samples. The
+ // data is now actually in 96 kHz sampling rate, despite the
+ // function name, and with a resampling factor of 1/3 becomes
+ // 32 kHz.
+ WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
+ (WebRtcSpl_State48khzTo16khz*) state1_,
+ tmp_mem);
+ }
+ outLen = lengthIn / 3;
+ free(tmp_mem);
+ // 4:2
+ WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2,
+ (WebRtc_Word32*) state2_);
+ outLen = outLen / 2;
+ free(tmp);
+ // 2:1
+ WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
+ (WebRtc_Word32*) state3_);
+ free(tmp_2);
+ outLen = outLen / 2;
+ return 0;
case kResamplerMode3To2:
if (maxLen < (lengthIn * 2 / 3))
{
diff --git a/src/common_audio/resampler/resampler.gypi b/src/common_audio/resampler/resampler.gypi
new file mode 100644
index 0000000..69f9b0e
--- /dev/null
+++ b/src/common_audio/resampler/resampler.gypi
@@ -0,0 +1,55 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'resampler',
+ 'type': '<(library)',
+ 'dependencies': [
+ 'signal_processing',
+ ],
+ 'include_dirs': [
+ 'include',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'include/resampler.h',
+ 'resampler.cc',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets' : [
+ {
+ 'target_name': 'resampler_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'resampler',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'resampler_unittest.cc',
+ ],
+ }, # resampler_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/common_audio/resampler/resampler_unittest.cc b/src/common_audio/resampler/resampler_unittest.cc
new file mode 100644
index 0000000..9b1061a
--- /dev/null
+++ b/src/common_audio/resampler/resampler_unittest.cc
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+#include "common_audio/resampler/include/resampler.h"
+
+// TODO(andrew): this is a work-in-progress. Many more tests are needed.
+
+namespace webrtc {
+namespace {
+const ResamplerType kTypes[] = {
+ kResamplerSynchronous,
+ kResamplerAsynchronous,
+ kResamplerSynchronousStereo,
+ kResamplerAsynchronousStereo
+ // kResamplerInvalid excluded
+};
+const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
+
+// Rates we must support.
+const int kMaxRate = 96000;
+const int kRates[] = {
+ 8000,
+ 16000,
+ 32000,
+ 44000,
+ 48000,
+ kMaxRate
+};
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+const int kMaxChannels = 2;
+const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
+
+// TODO(andrew): should we be supporting these combinations?
+bool ValidRates(int in_rate, int out_rate) {
+ // Not the most compact notation, for clarity.
+ if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
+ (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
+ return false;
+ }
+
+ return true;
+}
+
+class ResamplerTest : public testing::Test {
+ protected:
+ ResamplerTest();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ Resampler rs_;
+ int16_t data_in_[kDataSize];
+ int16_t data_out_[kDataSize];
+};
+
+ResamplerTest::ResamplerTest() {}
+
+void ResamplerTest::SetUp() {
+ // Initialize input data with anything. The tests are content independent.
+ memset(data_in_, 1, sizeof(data_in_));
+}
+
+void ResamplerTest::TearDown() {}
+
+TEST_F(ResamplerTest, Reset) {
+ // The only failure mode for the constructor is if Reset() fails. For the
+ // time being then (until an Init function is added), we rely on Reset()
+ // to test the constructor.
+
+ // Check that all required combinations are supported.
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ for (size_t k = 0; k < kTypesSize; ++k) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
+ << ", type: " << kTypes[k];
+ SCOPED_TRACE(ss.str());
+ if (ValidRates(kRates[i], kRates[j]))
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+ else
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+ }
+ }
+ }
+}
+
+// TODO(tlegrand): Replace code inside the two tests below with a function
+// with number of channels and ResamplerType as input.
+TEST_F(ResamplerTest, Synchronous) {
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+ SCOPED_TRACE(ss.str());
+
+ if (ValidRates(kRates[i], kRates[j])) {
+ int in_length = kRates[i] / 100;
+ int out_length = 0;
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+ EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+ out_length));
+ EXPECT_EQ(kRates[j] / 100, out_length);
+ } else {
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+ }
+ }
+ }
+}
+
+TEST_F(ResamplerTest, SynchronousStereo) {
+ // Number of channels is 2, stereo mode.
+ const int kChannels = 2;
+ for (size_t i = 0; i < kRatesSize; ++i) {
+ for (size_t j = 0; j < kRatesSize; ++j) {
+ std::ostringstream ss;
+ ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+ SCOPED_TRACE(ss.str());
+
+ if (ValidRates(kRates[i], kRates[j])) {
+ int in_length = kChannels * kRates[i] / 100;
+ int out_length = 0;
+ EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
+ kResamplerSynchronousStereo));
+ EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+ out_length));
+ EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
+ } else {
+ EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
+ kResamplerSynchronousStereo));
+ }
+ }
+ }
+}
+} // namespace
+} // namespace webrtc
diff --git a/src/common_audio/signal_processing_library/main/source/Android.mk b/src/common_audio/signal_processing/Android.mk
similarity index 66%
rename from src/common_audio/signal_processing_library/main/source/Android.mk
rename to src/common_audio/signal_processing/Android.mk
index 401a7f6..787e5c1 100644
--- a/src/common_audio/signal_processing_library/main/source/Android.mk
+++ b/src/common_audio/signal_processing/Android.mk
@@ -10,20 +10,18 @@
include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
LOCAL_ARM_MODE := arm
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
LOCAL_MODULE := libwebrtc_spl
LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := add_sat_w16.c \
- add_sat_w32.c \
+LOCAL_SRC_FILES := \
auto_corr_to_refl_coef.c \
auto_correlation.c \
complex_fft.c \
- complex_ifft.c \
complex_bit_reverse.c \
copy_set_operations.c \
- cos_table.c \
cross_correlation.c \
division_operations.c \
dot_product_with_scale.c \
@@ -34,16 +32,10 @@
filter_ma_fast_q12.c \
get_hanning_window.c \
get_scaling_square.c \
- get_size_in_bits.c \
- hanning_table.c \
ilbc_specific_functions.c \
levinson_durbin.c \
lpc_to_refl_coef.c \
min_max_operations.c \
- norm_u32.c \
- norm_w16.c \
- norm_w32.c \
- randn_table.c \
randomization_functions.c \
refl_coef_to_lpc.c \
resample.c \
@@ -51,45 +43,30 @@
resample_by_2.c \
resample_by_2_internal.c \
resample_fractional.c \
- sin_table.c \
- sin_table_1024.c \
spl_sqrt.c \
+ spl_sqrt_floor.c \
spl_version.c \
splitting_filter.c \
sqrt_of_one_minus_x_squared.c \
- sub_sat_w16.c \
- sub_sat_w32.c \
vector_scaling_operations.c
# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../..
+
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+LOCAL_SRC_FILES += \
+ min_max_operations_neon.c
+LOCAL_CFLAGS += \
+ $(MY_ARM_CFLAGS_NEON)
endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
LOCAL_SHARED_LIBRARIES := libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
LOCAL_LDLIBS += -ldl -lpthread
endif
@@ -98,5 +75,7 @@
LOCAL_SHARED_LIBRARIES += libdl
endif
+ifndef NDK_ROOT
include external/stlport/libstlport.mk
+endif
include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/signal_processing_library/main/source/auto_corr_to_refl_coef.c b/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/auto_corr_to_refl_coef.c
rename to src/common_audio/signal_processing/auto_corr_to_refl_coef.c
diff --git a/src/common_audio/signal_processing_library/main/source/auto_correlation.c b/src/common_audio/signal_processing/auto_correlation.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/auto_correlation.c
rename to src/common_audio/signal_processing/auto_correlation.c
diff --git a/src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c b/src/common_audio/signal_processing/complex_bit_reverse.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c
rename to src/common_audio/signal_processing/complex_bit_reverse.c
diff --git a/src/common_audio/signal_processing/complex_fft.c b/src/common_audio/signal_processing/complex_fft.c
new file mode 100644
index 0000000..1e8503c
--- /dev/null
+++ b/src/common_audio/signal_processing/complex_fft.c
@@ -0,0 +1,425 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_ComplexFFT().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define CFFTSFT 14
+#define CFFTRND 1
+#define CFFTRND2 16384
+
+#define CIFFTSFT 14
+#define CIFFTRND 1
+
+static const WebRtc_Word16 kSinTable1024[] = {
+ 0, 201, 402, 603, 804, 1005, 1206, 1406,
+ 1607, 1808, 2009, 2209, 2410, 2610, 2811, 3011,
+ 3211, 3411, 3611, 3811, 4011, 4210, 4409, 4608,
+ 4807, 5006, 5205, 5403, 5601, 5799, 5997, 6195,
+ 6392, 6589, 6786, 6982, 7179, 7375, 7571, 7766,
+ 7961, 8156, 8351, 8545, 8739, 8932, 9126, 9319,
+ 9511, 9703, 9895, 10087, 10278, 10469, 10659, 10849,
+ 11038, 11227, 11416, 11604, 11792, 11980, 12166, 12353,
+ 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
+ 14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268,
+ 15446, 15623, 15799, 15975, 16150, 16325, 16499, 16672,
+ 16845, 17017, 17189, 17360, 17530, 17699, 17868, 18036,
+ 18204, 18371, 18537, 18702, 18867, 19031, 19194, 19357,
+ 19519, 19680, 19840, 20000, 20159, 20317, 20474, 20631,
+ 20787, 20942, 21096, 21249, 21402, 21554, 21705, 21855,
+ 22004, 22153, 22301, 22448, 22594, 22739, 22883, 23027,
+ 23169, 23311, 23452, 23592, 23731, 23869, 24006, 24143,
+ 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
+ 25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198,
+ 26318, 26437, 26556, 26673, 26789, 26905, 27019, 27132,
+ 27244, 27355, 27466, 27575, 27683, 27790, 27896, 28001,
+ 28105, 28208, 28309, 28410, 28510, 28608, 28706, 28802,
+ 28897, 28992, 29085, 29177, 29268, 29358, 29446, 29534,
+ 29621, 29706, 29790, 29873, 29955, 30036, 30116, 30195,
+ 30272, 30349, 30424, 30498, 30571, 30643, 30713, 30783,
+ 30851, 30918, 30984, 31049,
+ 31113, 31175, 31236, 31297,
+ 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
+ 31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097,
+ 32137, 32176, 32213, 32249, 32284, 32318, 32350, 32382,
+ 32412, 32441, 32468, 32495, 32520, 32544, 32567, 32588,
+ 32609, 32628, 32646, 32662, 32678, 32692, 32705, 32717,
+ 32727, 32736, 32744, 32751, 32757, 32761, 32764, 32766,
+ 32767, 32766, 32764, 32761, 32757, 32751, 32744, 32736,
+ 32727, 32717, 32705, 32692, 32678, 32662, 32646, 32628,
+ 32609, 32588, 32567, 32544, 32520, 32495, 32468, 32441,
+ 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
+ 32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833,
+ 31785, 31735, 31684, 31633, 31580, 31525, 31470, 31413,
+ 31356, 31297, 31236, 31175, 31113, 31049, 30984, 30918,
+ 30851, 30783, 30713, 30643, 30571, 30498, 30424, 30349,
+ 30272, 30195, 30116, 30036, 29955, 29873, 29790, 29706,
+ 29621, 29534, 29446, 29358, 29268, 29177, 29085, 28992,
+ 28897, 28802, 28706, 28608, 28510, 28410, 28309, 28208,
+ 28105, 28001, 27896, 27790, 27683, 27575, 27466, 27355,
+ 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
+ 26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456,
+ 25329, 25201, 25072, 24942, 24811, 24679, 24546, 24413,
+ 24278, 24143, 24006, 23869, 23731, 23592, 23452, 23311,
+ 23169, 23027, 22883, 22739, 22594, 22448, 22301, 22153,
+ 22004, 21855, 21705, 21554, 21402, 21249, 21096, 20942,
+ 20787, 20631, 20474, 20317, 20159, 20000, 19840, 19680,
+ 19519, 19357, 19194, 19031, 18867, 18702, 18537, 18371,
+ 18204, 18036, 17868, 17699, 17530, 17360, 17189, 17017,
+ 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
+ 15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191,
+ 14009, 13827, 13645, 13462, 13278, 13094, 12909, 12724,
+ 12539, 12353, 12166, 11980, 11792, 11604, 11416, 11227,
+ 11038, 10849, 10659, 10469, 10278, 10087, 9895, 9703,
+ 9511, 9319, 9126, 8932, 8739, 8545, 8351, 8156,
+ 7961, 7766, 7571, 7375, 7179, 6982, 6786, 6589,
+ 6392, 6195, 5997, 5799, 5601, 5403, 5205, 5006,
+ 4807, 4608, 4409, 4210, 4011, 3811, 3611, 3411,
+ 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
+ 1607, 1406, 1206, 1005, 804, 603, 402, 201,
+ 0, -201, -402, -603, -804, -1005, -1206, -1406,
+ -1607, -1808, -2009, -2209, -2410, -2610, -2811, -3011,
+ -3211, -3411, -3611, -3811, -4011, -4210, -4409, -4608,
+ -4807, -5006, -5205, -5403, -5601, -5799, -5997, -6195,
+ -6392, -6589, -6786, -6982, -7179, -7375, -7571, -7766,
+ -7961, -8156, -8351, -8545, -8739, -8932, -9126, -9319,
+ -9511, -9703, -9895, -10087, -10278, -10469, -10659, -10849,
+ -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
+ -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
+ -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
+ -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
+ -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
+ -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
+ -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
+ -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
+ -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
+ -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
+ -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
+ -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
+ -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
+ -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
+ -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
+ -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
+ -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
+ -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
+ -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
+ -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
+ -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
+ -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
+ -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
+ -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
+ -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
+ -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
+ -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
+ -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
+ -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
+ -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
+ -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
+ -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
+ -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
+ -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
+ -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
+ -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
+ -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
+ -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
+ -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
+ -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
+ -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
+ -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
+ -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
+ -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
+ -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
+ -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
+ -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
+ -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
+ -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
+ -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
+ -11038, -10849, -10659, -10469, -10278, -10087, -9895, -9703,
+ -9511, -9319, -9126, -8932, -8739, -8545, -8351, -8156,
+ -7961, -7766, -7571, -7375, -7179, -6982, -6786, -6589,
+ -6392, -6195, -5997, -5799, -5601, -5403, -5205, -5006,
+ -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
+ -3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808,
+ -1607, -1406, -1206, -1005, -804, -603, -402, -201
+};
+
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+ int i, j, l, k, istep, n, m;
+ WebRtc_Word16 wr, wi;
+ WebRtc_Word32 tr32, ti32, qr32, qi32;
+
+ /* The 1024-value is a constant given from the size of kSinTable1024[],
+ * and should not be changed depending on the input parameter 'stages'
+ */
+ n = 1 << stages;
+ if (n > 1024)
+ return -1;
+
+ l = 1;
+ k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+ depending on the input parameter 'stages' */
+
+ if (mode == 0)
+ {
+ // mode==0: Low-complexity and Low-accuracy mode
+ while (l < n)
+ {
+ istep = l << 1;
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = -kSinTable1024[j];
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
+
+ ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
+
+ qr32 = (WebRtc_Word32)frfi[2 * i];
+ qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
+ }
+ }
+
+ --k;
+ l = istep;
+
+ }
+
+ } else
+ {
+ // mode==1: High-complexity and High-accuracy mode
+ while (l < n)
+ {
+ istep = l << 1;
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = -kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ WebRtc_Word32 wri;
+ WebRtc_Word32 frfi_r;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+ "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+ "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+ __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+ "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+ __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+ "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+
+#else
+ tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
+
+ ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
+#endif
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CFFTSFT);
+ ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CFFTSFT);
+
+ qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
+ qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
+
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
+ }
+ }
+
+ --k;
+ l = istep;
+ }
+ }
+ return 0;
+}
+
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+ int i, j, l, k, istep, n, m, scale, shift;
+ WebRtc_Word16 wr, wi;
+ WebRtc_Word32 tr32, ti32, qr32, qi32;
+ WebRtc_Word32 tmp32, round2;
+
+ /* The 1024-value is a constant given from the size of kSinTable1024[],
+ * and should not be changed depending on the input parameter 'stages'
+ */
+ n = 1 << stages;
+ if (n > 1024)
+ return -1;
+
+ scale = 0;
+
+ l = 1;
+ k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+ depending on the input parameter 'stages' */
+
+ while (l < n)
+ {
+ // variable scaling, depending upon data
+ shift = 0;
+ round2 = 8192;
+
+ tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
+ if (tmp32 > 13573)
+ {
+ shift++;
+ scale++;
+ round2 <<= 1;
+ }
+ if (tmp32 > 27146)
+ {
+ shift++;
+ scale++;
+ round2 <<= 1;
+ }
+
+ istep = l << 1;
+
+ if (mode == 0)
+ {
+ // mode==0: Low-complexity and Low-accuracy mode
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = kSinTable1024[j];
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+ tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
+ - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
+
+ ti32 = WEBRTC_SPL_RSHIFT_W32(
+ (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
+ + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
+
+ qr32 = (WebRtc_Word32)frfi[2 * i];
+ qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
+ }
+ }
+ } else
+ {
+ // mode==1: High-complexity and High-accuracy mode
+
+ for (m = 0; m < l; ++m)
+ {
+ j = m << k;
+
+ /* The 256-value is a constant given as 1/4 of the size of
+ * kSinTable1024[], and should not be changed depending on the input
+ * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+ */
+ wr = kSinTable1024[j + 256];
+ wi = kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ WebRtc_Word32 wri;
+ WebRtc_Word32 frfi_r;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+ "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+ for (i = m; i < n; i += istep)
+ {
+ j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+ "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+ __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+ "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+ __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+ "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+#else
+
+ tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+ - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
+
+ ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
+#endif
+ tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CIFFTSFT);
+ ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CIFFTSFT);
+
+ qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
+ qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
+
+ frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
+ shift+CIFFTSFT);
+ frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 - ti32 + round2), shift + CIFFTSFT);
+ frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
+ shift + CIFFTSFT);
+ frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+ (qi32 + ti32 + round2), shift + CIFFTSFT);
+ }
+ }
+
+ }
+ --k;
+ l = istep;
+ }
+ return scale;
+}
diff --git a/src/common_audio/signal_processing_library/main/source/copy_set_operations.c b/src/common_audio/signal_processing/copy_set_operations.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/copy_set_operations.c
rename to src/common_audio/signal_processing/copy_set_operations.c
diff --git a/src/common_audio/signal_processing_library/main/source/cross_correlation.c b/src/common_audio/signal_processing/cross_correlation.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/cross_correlation.c
rename to src/common_audio/signal_processing/cross_correlation.c
diff --git a/src/common_audio/signal_processing_library/main/source/division_operations.c b/src/common_audio/signal_processing/division_operations.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/division_operations.c
rename to src/common_audio/signal_processing/division_operations.c
diff --git a/src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c b/src/common_audio/signal_processing/dot_product_with_scale.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c
rename to src/common_audio/signal_processing/dot_product_with_scale.c
diff --git a/src/common_audio/signal_processing_library/main/source/downsample_fast.c b/src/common_audio/signal_processing/downsample_fast.c
similarity index 95%
rename from src/common_audio/signal_processing_library/main/source/downsample_fast.c
rename to src/common_audio/signal_processing/downsample_fast.c
index 9338275..cce463c 100644
--- a/src/common_audio/signal_processing_library/main/source/downsample_fast.c
+++ b/src/common_audio/signal_processing/downsample_fast.c
@@ -52,7 +52,7 @@
// If output is higher than 32768, saturate it. Same with negative side
- *downsampled_ptr++ = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, o, -32768);
+ *downsampled_ptr++ = WebRtcSpl_SatW32ToW16(o);
}
return 0;
diff --git a/src/common_audio/signal_processing_library/main/source/energy.c b/src/common_audio/signal_processing/energy.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/energy.c
rename to src/common_audio/signal_processing/energy.c
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ar.c b/src/common_audio/signal_processing/filter_ar.c
similarity index 96%
rename from src/common_audio/signal_processing_library/main/source/filter_ar.c
rename to src/common_audio/signal_processing/filter_ar.c
index 30a56c1..24e83a6 100644
--- a/src/common_audio/signal_processing_library/main/source/filter_ar.c
+++ b/src/common_audio/signal_processing/filter_ar.c
@@ -36,9 +36,6 @@
WebRtc_Word16* filteredFINAL_ptr = filtered;
WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
- state_low_length = state_low_length;
- filtered_low_length = filtered_low_length;
-
for (i = 0; i < x_length; i++)
{
// Calculate filtered[i] and filtered_low[i]
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c b/src/common_audio/signal_processing/filter_ar_fast_q12.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c
rename to src/common_audio/signal_processing/filter_ar_fast_q12.c
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c b/src/common_audio/signal_processing/filter_ma_fast_q12.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c
rename to src/common_audio/signal_processing/filter_ma_fast_q12.c
diff --git a/src/common_audio/signal_processing_library/main/source/hanning_table.c b/src/common_audio/signal_processing/get_hanning_window.c
similarity index 78%
rename from src/common_audio/signal_processing_library/main/source/hanning_table.c
rename to src/common_audio/signal_processing/get_hanning_window.c
index 112d0e5..6d67e60 100644
--- a/src/common_audio/signal_processing_library/main/source/hanning_table.c
+++ b/src/common_audio/signal_processing/get_hanning_window.c
@@ -10,14 +10,15 @@
/*
- * This file contains the Hanning table with 256 entries.
+ * This file contains the function WebRtcSpl_GetHanningWindow().
+ * The description header can be found in signal_processing_library.h
*
*/
#include "signal_processing_library.h"
// Hanning table with 256 entries
-WebRtc_Word16 WebRtcSpl_kHanningTable[] = {
+static const WebRtc_Word16 kHanningTable[] = {
1, 2, 6, 10, 15, 22, 30, 39,
50, 62, 75, 89, 104, 121, 138, 157,
178, 199, 222, 246, 271, 297, 324, 353,
@@ -51,3 +52,26 @@
16246, 16263, 16280, 16295, 16309, 16322, 16334, 16345,
16354, 16362, 16369, 16374, 16378, 16382, 16383, 16384
};
+
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
+{
+ int jj;
+ WebRtc_Word16 *vptr1;
+
+ WebRtc_Word32 index;
+ WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
+
+ factor = WebRtcSpl_DivW32W16(factor, size);
+ if (size < 513)
+ index = (WebRtc_Word32)-0x200000;
+ else
+ index = (WebRtc_Word32)-0x100000;
+ vptr1 = v;
+
+ for (jj = 0; jj < size; jj++)
+ {
+ index += factor;
+ (*vptr1++) = kHanningTable[index >> 22];
+ }
+
+}
diff --git a/src/common_audio/signal_processing_library/main/source/get_scaling_square.c b/src/common_audio/signal_processing/get_scaling_square.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/get_scaling_square.c
rename to src/common_audio/signal_processing/get_scaling_square.c
diff --git a/src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c b/src/common_audio/signal_processing/ilbc_specific_functions.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c
rename to src/common_audio/signal_processing/ilbc_specific_functions.c
diff --git a/src/common_audio/signal_processing_library/main/interface/signal_processing_library.h b/src/common_audio/signal_processing/include/signal_processing_library.h
similarity index 85%
rename from src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
rename to src/common_audio/signal_processing/include/signal_processing_library.h
index 414e045..651a68c 100644
--- a/src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
+++ b/src/common_audio/signal_processing/include/signal_processing_library.h
@@ -25,11 +25,6 @@
#include <Armintr.h> // intrinsic file for windows mobile
#endif
-#ifdef WEBRTC_ANDROID
-#define WEBRTC_SPL_INLINE_CALLS
-#define SPL_NO_DOUBLE_IMPLEMENTATIONS
-#endif
-
// Macros specific for the fixed point implementation
#define WEBRTC_SPL_WORD16_MAX 32767
#define WEBRTC_SPL_WORD16_MIN -32768
@@ -39,100 +34,97 @@
#define WEBRTC_SPL_MAX_SEED_USED 0x80000000L
#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
-#define WEBRTC_SPL_ABS_W16(a)\
+#define WEBRTC_SPL_ABS_W16(a) \
(((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
-#define WEBRTC_SPL_ABS_W32(a)\
+#define WEBRTC_SPL_ABS_W32(a) \
(((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
#define WEBRTC_SPL_GET_BYTE(a, nr) (((WebRtc_Word8 *)a)[nr])
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- (((WebRtc_Word8 *)d_ptr)[index] = (val))
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ (((WebRtc_Word8 *)d_ptr)[index] = (val))
#elif defined WEBRTC_BIG_ENDIAN
-#define WEBRTC_SPL_GET_BYTE(a, nr)\
+#define WEBRTC_SPL_GET_BYTE(a, nr) \
((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
- ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
- & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
+ ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
+ & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
#else
-#define WEBRTC_SPL_GET_BYTE(a,nr) \
- ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
- ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
- & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
- ((val) << (8 * ((index) & 0x1)))
+#define WEBRTC_SPL_GET_BYTE(a,nr) \
+ ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+ ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
+ ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
+ & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
+ ((val) << (8 * ((index) & 0x1)))
#endif
-#ifndef WEBRTC_ANDROID
-#define WEBRTC_SPL_MUL(a, b) \
- ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
-#endif
-
-#define WEBRTC_SPL_UMUL(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
-#define WEBRTC_SPL_UMUL_RSFT16(a, b)\
+#define WEBRTC_SPL_MUL(a, b) \
+ ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UMUL(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
+#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_16_16(a, b)\
+#define WEBRTC_SPL_UMUL_16_16(a, b) \
((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b)\
+#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
(((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_32_16(a, b)\
+#define WEBRTC_SPL_UMUL_32_16(a, b) \
((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
-#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b)\
+#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_MUL_16_U16(a, b)\
+#define WEBRTC_SPL_MUL_16_U16(a, b) \
((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_DIV(a, b) \
- ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
-#define WEBRTC_SPL_UDIV(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
+#define WEBRTC_SPL_DIV(a, b) \
+ ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UDIV(a, b) \
+ ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
-#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b)\
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
-#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b)\
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
-#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
-
-#ifndef WEBRTC_ANDROID
-#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
- (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
- + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
-#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
- ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
- + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
-#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
- ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
- (WebRtc_Word16)(a32 >> 16)), b32) + \
- (WEBRTC_SPL_MUL_16_32_RSFT16(( \
- (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
+#ifndef WEBRTC_ARCH_ARM_V7A
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+ ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+ (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
+ + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
+ ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
+ + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
+#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
+ ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
+ (WebRtc_Word16)(a32 >> 16)), b32) + \
+ (WEBRTC_SPL_MUL_16_32_RSFT16(( \
+ (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
#endif
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
+ + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
+
#ifdef ARM_WINM
-#define WEBRTC_SPL_MUL_16_16(a, b) \
- _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
-#elif !defined (WEBRTC_ANDROID)
-#define WEBRTC_SPL_MUL_16_16(a, b) \
- ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+ _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
#endif
-#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
- (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
+ (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
- ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+ ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
(((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b)\
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
// C + the 32 most significant bits of A * B
-#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
- (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+ (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
#define WEBRTC_SPL_ADD_SAT_W32(a, b) WebRtcSpl_AddSatW32(a, b)
#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
@@ -146,10 +138,10 @@
#define WEBRTC_SPL_IS_NEG(a) ((a) & 0x80000000)
// Shifting with negative numbers allowed
// Positive means left shift
-#define WEBRTC_SPL_SHIFT_W16(x, c) \
- (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
-#define WEBRTC_SPL_SHIFT_W32(x, c) \
- (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W16(x, c) \
+ (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) \
+ (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
// Shifting with negative numbers not allowed
// We cannot do casting here due to signed/unsigned problem
@@ -166,44 +158,24 @@
#define WEBRTC_SPL_VNEW(t, n) (t *) malloc (sizeof (t) * (n))
#define WEBRTC_SPL_FREE free
-#define WEBRTC_SPL_RAND(a)\
- ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
+#define WEBRTC_SPL_RAND(a) \
+ ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
#ifdef __cplusplus
extern "C"
{
#endif
-#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(char))
-#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
+#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
+ memcpy(v1, v2, (length) * sizeof(char))
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+ memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
-#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
- memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
+#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
+ memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
-// Trigonometric tables used for quick lookup
-// default declarations
-extern WebRtc_Word16 WebRtcSpl_kCosTable[];
-extern WebRtc_Word16 WebRtcSpl_kSinTable[];
-extern WebRtc_Word16 WebRtcSpl_kSinTable1024[];
-// Hanning table
-extern WebRtc_Word16 WebRtcSpl_kHanningTable[];
-// Random table
-extern WebRtc_Word16 WebRtcSpl_kRandNTable[];
-
-#ifndef WEBRTC_SPL_INLINE_CALLS
-WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2);
-WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2);
-WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2);
-WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2);
-WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 value);
-int WebRtcSpl_NormW32(WebRtc_Word32 value);
-int WebRtcSpl_NormW16(WebRtc_Word16 value);
-int WebRtcSpl_NormU32(WebRtc_UWord32 value);
-#else
+// inline functions:
#include "spl_inl.h"
-#endif
// Get SPL Version
WebRtc_Word16 WebRtcSpl_get_version(char* version,
@@ -240,7 +212,7 @@
// Minimum and maximum operations. Implementation in min_max_operations.c.
// Descriptions at bottom of file.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(G_CONST WebRtc_Word16* vector,
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
WebRtc_Word16 length);
WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
WebRtc_Word16 length);
@@ -431,14 +403,6 @@
// FFT operations
int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-int WebRtcSpl_ComplexFFT2(WebRtc_Word16 in_vector[],
- WebRtc_Word16 out_vector[],
- int stages, int mode);
-int WebRtcSpl_ComplexIFFT2(WebRtc_Word16 in_vector[],
- WebRtc_Word16 out_vector[],
- int stages, int mode);
-#endif
void WebRtcSpl_ComplexBitReverse(WebRtc_Word16 vector[], int stages);
// End: FFT operations
@@ -1575,43 +1539,6 @@
// value of -1, indicating error.
//
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-//
-// WebRtcSpl_ComplexIFFT2(...)
-//
-// Complex or Real inverse FFT, for ARM processor only
-//
-// Computes a 2^|stages|-point FFT on the input vector, which can be or not be
-// in bit-reversed order. If it is bit-reversed, the original content of the
-// vector could be overwritten by the output by setting the first two arguments
-// the same. With X as the input complex vector, y as the output complex vector
-// and with M = 2^|stages|, the following is computed:
-//
-// M-1
-// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-// i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Arguments:
-// - in_vector : In pointer to complex vector containing 2^|stages|
-// real elements interleaved with 2^|stages| imaginary
-// elements. [ReImReImReIm....]
-// The elements are in Q(-scale) domain.
-// - out_vector : Output pointer to vector containing 2^|stages| real
-// elements interleaved with 2^|stages| imaginary
-// elements. [ReImReImReIm....]
-// The output is in the Q0 domain.
-// - stages : Number of FFT stages. Must be at least 3 and at most
-// 10.
-// - mode : Dummy input.
-//
-// Return value : The scale parameter is always 0, except if N>1024,
-// which returns a scale value of -1, indicating error.
-//
-#endif
-
//
// WebRtcSpl_ComplexFFT(...)
//
@@ -1657,42 +1584,6 @@
// which returns a scale value of -1, indicating error.
//
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-//
-// WebRtcSpl_ComplexFFT2(...)
-//
-// Complex or Real FFT, for ARM processor only
-//
-// Computes a 2^|stages|-point FFT on the input vector, which can be or not be
-// in bit-reversed order. If it is bit-reversed, the original content of the
-// vector could be overwritten by the output by setting the first two arguments
-// the same. With x as the input complex vector, Y as the output complex vector
-// and with M = 2^|stages|, the following is computed:
-//
-// M-1
-// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-// i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Arguments:
-// - in_vector : In pointer to complex vector containing 2^|stages|
-// real elements interleaved with 2^|stages| imaginary
-// elements. [ReImReImReIm....]
-// - out_vector : Output pointer to vector containing 2^|stages| real
-// elements interleaved with 2^|stages| imaginary
-// elements. [ReImReImReIm....]
-// The output is in the Q0 domain.
-// - stages : Number of FFT stages. Must be at least 3 and at most
-// 10.
-// - mode : Dummy input
-//
-// Return value : The scale parameter is always 0, except if N>1024,
-// which returns a scale value of -1, indicating error.
-//
-#endif
-
//
// WebRtcSpl_ComplexBitReverse(...)
//
@@ -1758,6 +1649,30 @@
// - out_data : Super-wideband speech signal, 0-16 kHz
//
+// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+//
+// Input:
+// - value32 : The value of a 32-bit word.
+//
+// Output:
+// - out16 : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+//
+// Input:
+// - a : The value of the first 16-bit word.
+// - b : The value of the second 16-bit word.
+// - c : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
+
// WebRtc_Word16 WebRtcSpl_get_version(...)
//
// This function gives the version string of the Signal Processing Library.
diff --git a/src/common_audio/signal_processing/include/spl_inl.h b/src/common_audio/signal_processing/include/spl_inl.h
new file mode 100644
index 0000000..23b3209
--- /dev/null
+++ b/src/common_audio/signal_processing/include/spl_inl.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_H_
+#define WEBRTC_SPL_SPL_INL_H_
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+#include "spl_inl_armv7.h"
+#else
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+ WebRtc_Word16 out16 = (WebRtc_Word16) value32;
+
+ if (value32 > 32767)
+ out16 = 32767;
+ else if (value32 < -32768)
+ out16 = -32768;
+
+ return out16;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sum;
+
+ // perform long addition
+ l_sum = l_var1 + l_var2;
+
+ // check for under or overflow
+ if (WEBRTC_SPL_IS_NEG(l_var1)) {
+ if (WEBRTC_SPL_IS_NEG(l_var2) && !WEBRTC_SPL_IS_NEG(l_sum)) {
+ l_sum = (WebRtc_Word32)0x80000000;
+ }
+ } else {
+ if (!WEBRTC_SPL_IS_NEG(l_var2) && WEBRTC_SPL_IS_NEG(l_sum)) {
+ l_sum = (WebRtc_Word32)0x7FFFFFFF;
+ }
+ }
+
+ return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+ WebRtc_Word16 var2) {
+ return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_diff;
+
+ // perform subtraction
+ l_diff = l_var1 - l_var2;
+
+ // check for underflow
+ if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
+ l_diff = (WebRtc_Word32)0x80000000;
+ // check for overflow
+ if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
+ l_diff = (WebRtc_Word32)0x7FFFFFFF;
+
+ return l_diff;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+ int bits;
+
+ if (0xFFFF0000 & n) {
+ bits = 16;
+ } else {
+ bits = 0;
+ }
+ if (0x0000FF00 & (n >> bits)) bits += 8;
+ if (0x000000F0 & (n >> bits)) bits += 4;
+ if (0x0000000C & (n >> bits)) bits += 2;
+ if (0x00000002 & (n >> bits)) bits += 1;
+ if (0x00000001 & (n >> bits)) bits += 1;
+
+ return bits;
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+ int zeros;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ if (!(0xFFFF8000 & a)) {
+ zeros = 16;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xFF800000 & (a << zeros))) zeros += 8;
+ if (!(0xF8000000 & (a << zeros))) zeros += 4;
+ if (!(0xE0000000 & (a << zeros))) zeros += 2;
+ if (!(0xC0000000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+ int zeros;
+
+ if (a == 0) return 0;
+
+ if (!(0xFFFF0000 & a)) {
+ zeros = 16;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xFF000000 & (a << zeros))) zeros += 8;
+ if (!(0xF0000000 & (a << zeros))) zeros += 4;
+ if (!(0xC0000000 & (a << zeros))) zeros += 2;
+ if (!(0x80000000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+ int zeros;
+
+ if (a <= 0) a ^= 0xFFFF;
+
+ if (!(0xFF80 & a)) {
+ zeros = 8;
+ } else {
+ zeros = 0;
+ }
+ if (!(0xF800 & (a << zeros))) zeros += 4;
+ if (!(0xE000 & (a << zeros))) zeros += 2;
+ if (!(0xC000 & (a << zeros))) zeros += 1;
+
+ return zeros;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+ int16_t b,
+ int32_t c) {
+ return (a * b + c);
+}
+
+#endif // WEBRTC_ARCH_ARM_V7A
+
+#endif // WEBRTC_SPL_SPL_INL_H_
diff --git a/src/common_audio/signal_processing/include/spl_inl_armv7.h b/src/common_audio/signal_processing/include/spl_inl_armv7.h
new file mode 100644
index 0000000..689c2ba
--- /dev/null
+++ b/src/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions for ARM processors in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
+#define WEBRTC_SPL_SPL_INL_ARMV7_H_
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
+ WebRtc_Word32 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
+ WebRtc_Word16 b,
+ WebRtc_Word32 c) {
+ WebRtc_Word32 tmp;
+ __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
+ __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(WebRtc_Word32 a,
+ WebRtc_Word32 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+ return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ WebRtc_Word32 tmp;
+ __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+ return tmp;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+ int16_t b,
+ int32_t c) {
+ int32_t tmp = 0;
+ __asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
+ return tmp;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+ WebRtc_Word16 b) {
+ WebRtc_Word32 s_sum;
+
+ __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
+
+ return (WebRtc_Word16) s_sum;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sum;
+
+ __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
+
+ return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+ WebRtc_Word16 var2) {
+ WebRtc_Word32 s_sub;
+
+ __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
+
+ return (WebRtc_Word16)s_sub;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+ WebRtc_Word32 l_var2) {
+ WebRtc_Word32 l_sub;
+
+ __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
+
+ return l_sub;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+ WebRtc_Word32 tmp;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(n));
+
+ return (WebRtc_Word16)(32 - tmp);
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+ WebRtc_Word32 tmp;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp - 1;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+ int tmp;
+
+ if (a == 0) return 0;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+ WebRtc_Word32 tmp;
+
+ if (a <= 0) a ^= 0xFFFFFFFF;
+
+ __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+ return tmp - 17;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+ WebRtc_Word16 out16;
+
+ __asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
+
+ return out16;
+}
+#endif // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/src/common_audio/signal_processing_library/main/source/levinson_durbin.c b/src/common_audio/signal_processing/levinson_durbin.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/levinson_durbin.c
rename to src/common_audio/signal_processing/levinson_durbin.c
diff --git a/src/common_audio/signal_processing_library/main/source/lpc_to_refl_coef.c b/src/common_audio/signal_processing/lpc_to_refl_coef.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/lpc_to_refl_coef.c
rename to src/common_audio/signal_processing/lpc_to_refl_coef.c
diff --git a/src/common_audio/signal_processing_library/main/source/min_max_operations.c b/src/common_audio/signal_processing/min_max_operations.c
similarity index 85%
rename from src/common_audio/signal_processing_library/main/source/min_max_operations.c
rename to src/common_audio/signal_processing/min_max_operations.c
index cf5e9a7..57eaff7 100644
--- a/src/common_audio/signal_processing_library/main/source/min_max_operations.c
+++ b/src/common_audio/signal_processing/min_max_operations.c
@@ -28,8 +28,10 @@
#include "signal_processing_library.h"
+#if !(defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16 *vector, WebRtc_Word16 length)
{
WebRtc_Word32 tempMax = 0;
WebRtc_Word32 absVal;
@@ -37,49 +39,6 @@
int i;
G_CONST WebRtc_Word16 *tmpvector = vector;
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-
- WebRtc_Word16 len4 = (length >> 2) << 2;
-
- for (i = 0; i < len4; i = i + 4)
- {
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- }
-
- for (i = len4; i < len; i++)
- {
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- }
-#else
for (i = 0; i < length; i++)
{
absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
@@ -91,9 +50,10 @@
}
totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
return totMax;
-#endif
}
+#endif
+
// Index of maximum absolute value in a word16 vector.
WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
{
diff --git a/src/common_audio/signal_processing/min_max_operations_neon.c b/src/common_audio/signal_processing/min_max_operations_neon.c
new file mode 100644
index 0000000..158bcc1
--- /dev/null
+++ b/src/common_audio/signal_processing/min_max_operations_neon.c
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
+#include <arm_neon.h>
+
+#include "signal_processing_library.h"
+
+// Maximum absolute value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+ WebRtc_Word16 length) {
+ WebRtc_Word32 temp_max = 0;
+ WebRtc_Word32 abs_val;
+ WebRtc_Word16 tot_max;
+ int i;
+
+ __asm__("vmov.i16 d25, #0" : : : "d25");
+
+ for (i = 0; i < length - 7; i += 8) {
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&vector[i]) : "q13");
+ __asm__("vabs.s16 q13, q13" : : : "q13");
+ __asm__("vpmax.s16 d26, d27" : : : "q13");
+ __asm__("vpmax.s16 d25, d26" : : : "d25", "d26");
+ }
+ __asm__("vpmax.s16 d25, d25" : : : "d25");
+ __asm__("vpmax.s16 d25, d25" : : : "d25");
+ __asm__("vmov.s16 %0, d25[0]" : "=r"(temp_max): : "d25");
+
+ for (; i < length; i++) {
+ abs_val = WEBRTC_SPL_ABS_W32((vector[i]));
+ if (abs_val > temp_max) {
+ temp_max = abs_val;
+ }
+ }
+ tot_max = (WebRtc_Word16)WEBRTC_SPL_MIN(temp_max, WEBRTC_SPL_WORD16_MAX);
+ return tot_max;
+}
+
+#endif
diff --git a/src/common_audio/signal_processing_library/main/source/randn_table.c b/src/common_audio/signal_processing/randomization_functions.c
similarity index 83%
rename from src/common_audio/signal_processing_library/main/source/randn_table.c
rename to src/common_audio/signal_processing/randomization_functions.c
index 734fa79..04271ad 100644
--- a/src/common_audio/signal_processing_library/main/source/randn_table.c
+++ b/src/common_audio/signal_processing/randomization_functions.c
@@ -10,14 +10,19 @@
/*
- * Table with 512 samples from a normal distribution with mean 1 and std 1
- * The values are shifted up 13 steps (multiplied by 8192)
+ * This file contains implementations of the randomization functions
+ * WebRtcSpl_IncreaseSeed()
+ * WebRtcSpl_RandU()
+ * WebRtcSpl_RandN()
+ * WebRtcSpl_RandUArray()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
*/
#include "signal_processing_library.h"
-WebRtc_Word16 WebRtcSpl_kRandNTable[] =
-{
+static const WebRtc_Word16 kRandNTable[] = {
9178, -7260, 40, 10189, 4894, -3531, -13779, 14764,
-4008, -8884, -8990, 1008, 7368, 5184, 3251, -5817,
-9786, 5963, 1770, 8066, -7135, 10772, -2298, 1361,
@@ -83,3 +88,32 @@
2406, 7703, -951, 11196, -564, 3406, 2217, 4806,
2374, -5797, 11839, 8940, -11874, 18213, 2855, 10492
};
+
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
+{
+ seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
+ return seed[0];
+}
+
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
+{
+ return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
+}
+
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
+{
+ return kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
+}
+
+// Creates an array of uniformly distributed variables
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+ WebRtc_Word16 vector_length,
+ WebRtc_UWord32* seed)
+{
+ int i;
+ for (i = 0; i < vector_length; i++)
+ {
+ vector[i] = WebRtcSpl_RandU(seed);
+ }
+ return vector_length;
+}
diff --git a/src/common_audio/signal_processing_library/main/source/refl_coef_to_lpc.c b/src/common_audio/signal_processing/refl_coef_to_lpc.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/refl_coef_to_lpc.c
rename to src/common_audio/signal_processing/refl_coef_to_lpc.c
diff --git a/src/common_audio/signal_processing_library/main/source/resample.c b/src/common_audio/signal_processing/resample.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/resample.c
rename to src/common_audio/signal_processing/resample.c
diff --git a/src/common_audio/signal_processing_library/main/source/resample_48khz.c b/src/common_audio/signal_processing/resample_48khz.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/resample_48khz.c
rename to src/common_audio/signal_processing/resample_48khz.c
diff --git a/src/common_audio/signal_processing/resample_by_2.c b/src/common_audio/signal_processing/resample_by_2.c
new file mode 100644
index 0000000..e239db7
--- /dev/null
+++ b/src/common_audio/signal_processing/resample_by_2.c
@@ -0,0 +1,181 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+
+// allpass filter coefficients.
+static const WebRtc_UWord32 kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
+static const WebRtc_UWord32 kResampleAllpass2[3] =
+ {12199, 37471 << 15, 60255 << 15};
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: state + ((diff * tbl_value) >> 16)
+
+static __inline WebRtc_Word32 MUL_ACCUM_1(WebRtc_Word32 tbl_value,
+ WebRtc_Word32 diff,
+ WebRtc_Word32 state) {
+ WebRtc_Word32 result;
+ __asm__("smlawb %r0, %r1, %r2, %r3": "=r"(result): "r"(diff),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
+//
+// The reason to introduce this function is that, in case we can't use smlawb
+// instruction (in MUL_ACCUM_1) due to input value range, we can still use
+// smmla to save some cycles.
+
+static __inline WebRtc_Word32 MUL_ACCUM_2(WebRtc_Word32 tbl_value,
+ WebRtc_Word32 diff,
+ WebRtc_Word32 state) {
+ WebRtc_Word32 result;
+ __asm__("smmla %r0, %r1, %r2, %r3": "=r"(result): "r"(diff << 1),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+#else
+
+// allpass filter coefficients.
+static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
+static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+#endif // WEBRTC_ARCH_ARM_V7A
+
+
+// decimator
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState) {
+ WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+ WebRtc_Word16 i;
+
+ register WebRtc_Word32 state0 = filtState[0];
+ register WebRtc_Word32 state1 = filtState[1];
+ register WebRtc_Word32 state2 = filtState[2];
+ register WebRtc_Word32 state3 = filtState[3];
+ register WebRtc_Word32 state4 = filtState[4];
+ register WebRtc_Word32 state5 = filtState[5];
+ register WebRtc_Word32 state6 = filtState[6];
+ register WebRtc_Word32 state7 = filtState[7];
+
+ for (i = (len >> 1); i > 0; i--) {
+ // lower allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
+
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState) {
+ WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+ WebRtc_Word16 i;
+
+ register WebRtc_Word32 state0 = filtState[0];
+ register WebRtc_Word32 state1 = filtState[1];
+ register WebRtc_Word32 state2 = filtState[2];
+ register WebRtc_Word32 state3 = filtState[3];
+ register WebRtc_Word32 state4 = filtState[4];
+ register WebRtc_Word32 state5 = filtState[5];
+ register WebRtc_Word32 state6 = filtState[6];
+ register WebRtc_Word32 state7 = filtState[7];
+
+ for (i = len; i > 0; i--) {
+ // lower allpass filter
+ in32 = (WebRtc_Word32)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
+ state2 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state3 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+
+ // upper allpass filter
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
+ state6 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state7 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.c b/src/common_audio/signal_processing/resample_by_2_internal.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/resample_by_2_internal.c
rename to src/common_audio/signal_processing/resample_by_2_internal.c
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h b/src/common_audio/signal_processing/resample_by_2_internal.h
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h
rename to src/common_audio/signal_processing/resample_by_2_internal.h
diff --git a/src/common_audio/signal_processing_library/main/source/resample_fractional.c b/src/common_audio/signal_processing/resample_fractional.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/resample_fractional.c
rename to src/common_audio/signal_processing/resample_fractional.c
diff --git a/src/common_audio/signal_processing_library/main/source/spl.gyp b/src/common_audio/signal_processing/signal_processing.gypi
similarity index 69%
rename from src/common_audio/signal_processing_library/main/source/spl.gyp
rename to src/common_audio/signal_processing/signal_processing.gypi
index 9c052fc..c67bf7c 100644
--- a/src/common_audio/signal_processing_library/main/source/spl.gyp
+++ b/src/common_audio/signal_processing/signal_processing.gypi
@@ -7,33 +7,26 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../common_settings.gypi', # Common settings
- ],
'targets': [
{
- 'target_name': 'spl',
+ 'target_name': 'signal_processing',
'type': '<(library)',
'include_dirs': [
- '../interface',
+ 'include',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'include',
],
},
'sources': [
- '../interface/signal_processing_library.h',
- '../interface/spl_inl.h',
- 'add_sat_w16.c',
- 'add_sat_w32.c',
+ 'include/signal_processing_library.h',
+ 'include/spl_inl.h',
'auto_corr_to_refl_coef.c',
'auto_correlation.c',
'complex_fft.c',
- 'complex_ifft.c',
'complex_bit_reverse.c',
'copy_set_operations.c',
- 'cos_table.c',
'cross_correlation.c',
'division_operations.c',
'dot_product_with_scale.c',
@@ -44,16 +37,10 @@
'filter_ma_fast_q12.c',
'get_hanning_window.c',
'get_scaling_square.c',
- 'get_size_in_bits.c',
- 'hanning_table.c',
'ilbc_specific_functions.c',
'levinson_durbin.c',
'lpc_to_refl_coef.c',
'min_max_operations.c',
- 'norm_u32.c',
- 'norm_w16.c',
- 'norm_w32.c',
- 'randn_table.c',
'randomization_functions.c',
'refl_coef_to_lpc.c',
'resample.c',
@@ -62,19 +49,33 @@
'resample_by_2_internal.c',
'resample_by_2_internal.h',
'resample_fractional.c',
- 'sin_table.c',
- 'sin_table_1024.c',
'spl_sqrt.c',
'spl_sqrt_floor.c',
'spl_version.c',
'splitting_filter.c',
'sqrt_of_one_minus_x_squared.c',
- 'sub_sat_w16.c',
- 'sub_sat_w32.c',
'vector_scaling_operations.c',
],
- },
- ],
+ }, # spl
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'signal_processing_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'signal_processing',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'signal_processing_unittest.cc',
+ ],
+ }, # spl_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
}
# Local Variables:
diff --git a/src/common_audio/signal_processing/signal_processing_unittest.cc b/src/common_audio/signal_processing/signal_processing_unittest.cc
new file mode 100644
index 0000000..b2e8281
--- /dev/null
+++ b/src/common_audio/signal_processing/signal_processing_unittest.cc
@@ -0,0 +1,448 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+#include "gtest/gtest.h"
+
+class SplTest : public testing::Test {
+ protected:
+ virtual ~SplTest() {
+ }
+ void SetUp() {
+ }
+ void TearDown() {
+ }
+};
+
+TEST_F(SplTest, MacroTest) {
+ // Macros with inputs.
+ int A = 10;
+ int B = 21;
+ int a = -3;
+ int b = WEBRTC_SPL_WORD32_MAX;
+ int nr = 2;
+ int d_ptr2 = 0;
+
+ EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
+
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
+ EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
+ WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
+ EXPECT_EQ(65536, d_ptr2);
+
+ EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
+ EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
+ EXPECT_EQ(-2147483645u, WEBRTC_SPL_UMUL(a, b));
+ b = WEBRTC_SPL_WORD16_MAX >> 1;
+ EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_RSFT16(a, b));
+ EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
+ EXPECT_EQ(16382u, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
+ EXPECT_EQ(-49149u, WEBRTC_SPL_UMUL_32_16(a, b));
+ EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
+
+ a = b;
+ b = -3;
+ EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
+ EXPECT_EQ(0u, WEBRTC_SPL_UDIV(a, b));
+
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
+ EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
+ EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
+
+ int a32 = WEBRTC_SPL_WORD32_MAX;
+ int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
+ int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
+ EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
+ EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
+
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
+ EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
+
+ EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
+
+ EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
+
+ EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
+ EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
+ EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
+
+ EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
+
+ // Shifting with negative numbers allowed
+ int shift_amount = 1; // Workaround compiler warning using variable here.
+ // Positive means left shift
+ EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, shift_amount));
+ EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
+
+ // Shifting with negative numbers not allowed
+ // We cannot do casting here due to signed/unsigned problem
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
+
+ EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
+ EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
+ EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1));
+
+ EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
+}
+
+TEST_F(SplTest, InlineTest) {
+ WebRtc_Word16 a = 121;
+ WebRtc_Word16 b = -17;
+ WebRtc_Word32 A = 111121;
+ WebRtc_Word32 B = -1711;
+ char bVersion[8];
+
+ EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
+ EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
+
+ EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
+ EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
+
+ EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
+ EXPECT_EQ(14, WebRtcSpl_NormW32(A));
+ EXPECT_EQ(4, WebRtcSpl_NormW16(B));
+ EXPECT_EQ(15, WebRtcSpl_NormU32(A));
+
+ EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
+}
+
+TEST_F(SplTest, MathOperationsTest) {
+ int A = 117;
+ WebRtc_Word32 num = 117;
+ WebRtc_Word32 den = -5;
+ WebRtc_UWord16 denU = 5;
+ EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
+ EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
+
+
+ EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
+ EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
+ EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
+}
+
+TEST_F(SplTest, BasicArrayOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_UWord8 b8[kVectorSize];
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+
+ WebRtc_UWord8 bTmp8[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+ WebRtc_Word32 bTmp32[kVectorSize];
+
+ WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b16[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b16[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(1, b16[kk]);
+ }
+ WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b32[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b32[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(1, b32[kk]);
+ }
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ bTmp8[kk] = (WebRtc_Word8)kk;
+ bTmp16[kk] = (WebRtc_Word16)kk;
+ bTmp32[kk] = (WebRtc_Word32)kk;
+ }
+ WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b8[kk], bTmp8[kk]);
+ }
+ WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[kk], bTmp16[kk]);
+ }
+// WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(b32[kk], bTmp32[kk]);
+// }
+ EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
+ for (int kk = 0; kk < 2; ++kk) {
+ EXPECT_EQ(kk+2, bTmp16[kk]);
+ }
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b32[kk] = B[kk];
+ b16[kk] = (WebRtc_Word16)B[kk];
+ }
+ WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
+ }
+
+ WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[3-kk], bTmp16[kk]);
+ }
+}
+
+TEST_F(SplTest, MinMaxOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, -1100};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = B[kk];
+ b32[kk] = B[kk];
+ }
+
+ EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, kVectorSize));
+ EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, kVectorSize));
+ EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, kVectorSize));
+ EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, kVectorSize));
+ EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, kVectorSize));
+ EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, kVectorSize));
+
+ EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, kVectorSize));
+ EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, kVectorSize));
+ EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, kVectorSize));
+
+ EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
+}
+
+TEST_F(SplTest, VectorOperationsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_Word16 a16[kVectorSize];
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ a16[kk] = B[kk];
+ b16[kk] = B[kk];
+ }
+
+ WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
+ }
+
+ WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, kVectorSize, 2, 2, 0);
+ for (int kk = 0; kk < 2; ++kk) {
+ EXPECT_EQ(614236, b32[kk]);
+ }
+// EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
+ EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, kVectorSize, 2));
+
+ WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
+ }
+ WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
+ }
+ WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
+ }
+
+ WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
+ for (int kk = 0; kk < kVectorSize - 1; ++kk) {
+ EXPECT_EQ(32767, bTmp16[kk]);
+ }
+ EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
+}
+
+TEST_F(SplTest, EstimatorsTest) {
+ const int kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = B[kk];
+ b32[kk] = B[kk];
+ }
+
+ EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
+}
+
+TEST_F(SplTest, FilterTest) {
+ const int kVectorSize = 4;
+ WebRtc_Word16 A[] = {1, 2, 33, 100};
+ WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
+ WebRtc_Word16 B[] = {4, 12, 133, 110};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word16 bTmp16[kVectorSize];
+ WebRtc_Word16 bTmp16Low[kVectorSize];
+ WebRtc_Word16 bState[kVectorSize];
+ WebRtc_Word16 bStateLow[kVectorSize];
+
+ WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
+ WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+
+ // MA filters
+ WebRtcSpl_FilterMAFastQ12(b16, bTmp16, B, kVectorSize, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ //EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+ }
+ // AR filters
+ WebRtcSpl_FilterARFastQ12(b16, bTmp16, A, kVectorSize, kVectorSize);
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+ }
+ EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
+ 5,
+ b16,
+ kVectorSize,
+ bState,
+ kVectorSize,
+ bStateLow,
+ kVectorSize,
+ bTmp16,
+ bTmp16Low,
+ kVectorSize));
+}
+
+TEST_F(SplTest, RandTest) {
+ const int kVectorSize = 4;
+ WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_UWord32 bSeed = 100000;
+
+ EXPECT_EQ(464449057u, WebRtcSpl_IncreaseSeed(&bSeed));
+ EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
+ EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
+ EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(BU[kk], b16[kk]);
+ }
+}
+
+TEST_F(SplTest, SignalProcessingTest) {
+ const int kVectorSize = 4;
+ int A[] = {1, 2, 33, 100};
+ WebRtc_Word16 b16[kVectorSize];
+ WebRtc_Word32 b32[kVectorSize];
+
+ WebRtc_Word16 bTmp16[kVectorSize];
+ WebRtc_Word32 bTmp32[kVectorSize];
+
+ int bScale = 0;
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ b32[kk] = A[kk];
+ }
+
+ EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, kVectorSize, 1, bTmp32, &bScale));
+ WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+// }
+ WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(a16[kk], b16[kk]);
+// }
+ WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+// }
+ WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
+// for (int kk = 0; kk < kVectorSize; ++kk) {
+// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+// }
+
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+ EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
+ EXPECT_EQ(0, bScale);
+}
+
+TEST_F(SplTest, FFTTest) {
+ WebRtc_Word16 B[] = {1, 2, 33, 100,
+ 2, 3, 34, 101,
+ 3, 4, 35, 102,
+ 4, 5, 36, 103};
+
+ EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
+// for (int kk = 0; kk < 16; ++kk) {
+// EXPECT_EQ(A[kk], B[kk]);
+// }
+ EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
+// for (int kk = 0; kk < 16; ++kk) {
+// EXPECT_EQ(A[kk], B[kk]);
+// }
+ WebRtcSpl_ComplexBitReverse(B, 3);
+ for (int kk = 0; kk < 16; ++kk) {
+ //EXPECT_EQ(A[kk], B[kk]);
+ }
+}
diff --git a/src/common_audio/signal_processing_library/main/source/spl_sqrt.c b/src/common_audio/signal_processing/spl_sqrt.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/spl_sqrt.c
rename to src/common_audio/signal_processing/spl_sqrt.c
diff --git a/src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c b/src/common_audio/signal_processing/spl_sqrt_floor.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c
rename to src/common_audio/signal_processing/spl_sqrt_floor.c
diff --git a/src/common_audio/signal_processing_library/main/source/spl_version.c b/src/common_audio/signal_processing/spl_version.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/spl_version.c
rename to src/common_audio/signal_processing/spl_version.c
diff --git a/src/common_audio/signal_processing_library/main/source/splitting_filter.c b/src/common_audio/signal_processing/splitting_filter.c
similarity index 95%
rename from src/common_audio/signal_processing_library/main/source/splitting_filter.c
rename to src/common_audio/signal_processing/splitting_filter.c
index 98442f4..f1acf67 100644
--- a/src/common_audio/signal_processing_library/main/source/splitting_filter.c
+++ b/src/common_audio/signal_processing/splitting_filter.c
@@ -147,13 +147,11 @@
{
tmp = filter1[i] + filter2[i] + 1024;
tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
- low_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
- tmp, WEBRTC_SPL_WORD16_MIN);
+ low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
tmp = filter1[i] - filter2[i] + 1024;
tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
- high_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
- tmp, WEBRTC_SPL_WORD16_MIN);
+ high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
}
}
@@ -191,10 +189,10 @@
for (i = 0, k = 0; i < kBandFrameLength; i++)
{
tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
- out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
- out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
}
}
diff --git a/src/common_audio/signal_processing_library/main/source/sqrt_of_one_minus_x_squared.c b/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/sqrt_of_one_minus_x_squared.c
rename to src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
diff --git a/src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c b/src/common_audio/signal_processing/vector_scaling_operations.c
similarity index 97%
rename from src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
rename to src/common_audio/signal_processing/vector_scaling_operations.c
index 47362ee..20d239c 100644
--- a/src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
+++ b/src/common_audio/signal_processing/vector_scaling_operations.c
@@ -125,7 +125,7 @@
for (i = 0; i < in_vector_length; i++)
{
tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
- ( *outptr++) = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmpW32, -32768);
+ (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
}
}
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c b/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c
rename to src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c b/src/common_audio/signal_processing/webrtc_fft_t_rad.c
similarity index 100%
rename from src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c
rename to src/common_audio/signal_processing/webrtc_fft_t_rad.c
diff --git a/src/common_audio/signal_processing_library/OWNERS b/src/common_audio/signal_processing_library/OWNERS
deleted file mode 100644
index cf595df..0000000
--- a/src/common_audio/signal_processing_library/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-bjornv@google.com
-tlegrand@google.com
-jks@google.com
diff --git a/src/common_audio/signal_processing_library/main/interface/spl_inl.h b/src/common_audio/signal_processing_library/main/interface/spl_inl.h
deleted file mode 100644
index 8716ca9..0000000
--- a/src/common_audio/signal_processing_library/main/interface/spl_inl.h
+++ /dev/null
@@ -1,293 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_H_
-#define WEBRTC_SPL_SPL_INL_H_
-
-#ifdef WEBRTC_SPL_INLINE_CALLS
-
-#ifdef WEBRTC_ANDROID
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL(WebRtc_Word32 a, WebRtc_Word32 b)
-{
- WebRtc_Word32 tmp;
- __asm__("mul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
- return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
- WebRtc_Word32 b)
-{
- WebRtc_Word32 tmp;
- __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
- return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
- WebRtc_Word16 b,
- WebRtc_Word32 c)
-{
- WebRtc_Word32 tmp;
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
- __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
- return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(
- WebRtc_Word32 a,
- WebRtc_Word32 b)
-{
- WebRtc_Word32 tmp;
- __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
- return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
- WebRtc_Word16 b)
-{
- WebRtc_Word32 tmp;
- __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
- return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
- WebRtc_Word16 b)
-{
- WebRtc_Word32 s_sum;
-
- __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
-
- return (WebRtc_Word16) s_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2)
-{
- WebRtc_Word32 l_sum;
-
- __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
-
- return l_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
- WebRtc_Word16 var2)
-{
- WebRtc_Word32 s_sub;
-
- __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
-
- return (WebRtc_Word16)s_sub;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2)
-{
- WebRtc_Word32 l_sub;
-
- __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
-
- return l_sub;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n)
-{
- WebRtc_Word32 tmp;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(n));
-
- return (WebRtc_Word16)(32 - tmp);
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW32(WebRtc_Word32 a)
-{
- WebRtc_Word32 tmp;
-
- if (a <= 0) a ^= 0xFFFFFFFF;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
- return tmp - 1;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW16(WebRtc_Word16 a)
-{
- int zeros;
-
- if (a <= 0) a ^= 0xFFFF;
-
- if (!(0xFF80 & a)) zeros = 8; else zeros = 0;
- if (!(0xF800 & (a << zeros))) zeros += 4;
- if (!(0xE000 & (a << zeros))) zeros += 2;
- if (!(0xC000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormU32(WebRtc_UWord32 a)
-{
- int tmp;
-
- if (a == 0) return 0;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
- return tmp;
-}
-
-#else
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
- WebRtc_Word16 b)
-{
- WebRtc_Word32 s_sum = (WebRtc_Word32) a + (WebRtc_Word32) b;
-
- if (s_sum > WEBRTC_SPL_WORD16_MAX)
- s_sum = WEBRTC_SPL_WORD16_MAX;
- else if (s_sum < WEBRTC_SPL_WORD16_MIN)
- s_sum = WEBRTC_SPL_WORD16_MIN;
-
- return (WebRtc_Word16)s_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2)
-{
- WebRtc_Word32 l_sum;
-
- // perform long addition
- l_sum = l_var1 + l_var2;
-
- // check for under or overflow
- if (WEBRTC_SPL_IS_NEG (l_var1))
- {
- if (WEBRTC_SPL_IS_NEG (l_var2) && !WEBRTC_SPL_IS_NEG (l_sum))
- {
- l_sum = (WebRtc_Word32)0x80000000;
- }
- }
- else
- {
- if (!WEBRTC_SPL_IS_NEG (l_var2) && WEBRTC_SPL_IS_NEG (l_sum))
- {
- l_sum = (WebRtc_Word32)0x7FFFFFFF;
- }
- }
-
- return l_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_SubSatW16( WebRtc_Word16 var1,
- WebRtc_Word16 var2)
-{
- WebRtc_Word32 l_diff;
- WebRtc_Word16 s_diff;
-
- // perform subtraction
- l_diff = (WebRtc_Word32)var1 - (WebRtc_Word32)var2;
-
- // default setting
- s_diff = (WebRtc_Word16) l_diff;
-
- // check for overflow
- if (l_diff > (WebRtc_Word32)32767)
- s_diff = (WebRtc_Word16)32767;
-
- // check for underflow
- if (l_diff < (WebRtc_Word32)-32768)
- s_diff = (WebRtc_Word16)-32768;
-
- return s_diff;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2)
-{
- WebRtc_Word32 l_diff;
-
- // perform subtraction
- l_diff = l_var1 - l_var2;
-
- // check for underflow
- if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
- l_diff = (WebRtc_Word32)0x80000000;
- // check for overflow
- if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
- l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
- return l_diff;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n)
-{
-
- int bits;
-
- if ((0xFFFF0000 & n)) bits = 16; else bits = 0;
- if ((0x0000FF00 & (n >> bits))) bits += 8;
- if ((0x000000F0 & (n >> bits))) bits += 4;
- if ((0x0000000C & (n >> bits))) bits += 2;
- if ((0x00000002 & (n >> bits))) bits += 1;
- if ((0x00000001 & (n >> bits))) bits += 1;
-
- return bits;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW32(WebRtc_Word32 a)
-{
- int zeros;
-
- if (a <= 0) a ^= 0xFFFFFFFF;
-
- if (!(0xFFFF8000 & a)) zeros = 16; else zeros = 0;
- if (!(0xFF800000 & (a << zeros))) zeros += 8;
- if (!(0xF8000000 & (a << zeros))) zeros += 4;
- if (!(0xE0000000 & (a << zeros))) zeros += 2;
- if (!(0xC0000000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW16(WebRtc_Word16 a)
-{
- int zeros;
-
- if (a <= 0) a ^= 0xFFFF;
-
- if (!(0xFF80 & a)) zeros = 8; else zeros = 0;
- if (!(0xF800 & (a << zeros))) zeros += 4;
- if (!(0xE000 & (a << zeros))) zeros += 2;
- if (!(0xC000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormU32(WebRtc_UWord32 a)
-{
- int zeros;
-
- if (a == 0) return 0;
-
- if (!(0xFFFF0000 & a)) zeros = 16; else zeros = 0;
- if (!(0xFF000000 & (a << zeros))) zeros += 8;
- if (!(0xF0000000 & (a << zeros))) zeros += 4;
- if (!(0xC0000000 & (a << zeros))) zeros += 2;
- if (!(0x80000000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-#endif // WEBRTC_ANDROID
-#endif // WEBRTC_SPL_INLINE_CALLS
-#endif // WEBRTC_SPL_SPL_INL_H_
diff --git a/src/common_audio/signal_processing_library/main/source/add_sat_w16.c b/src/common_audio/signal_processing_library/main/source/add_sat_w16.c
deleted file mode 100644
index d103999..0000000
--- a/src/common_audio/signal_processing_library/main/source/add_sat_w16.c
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AddSatW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2)
-{
- WebRtc_Word32 s_sum = (WebRtc_Word32)var1 + (WebRtc_Word32)var2;
-
- if (s_sum > WEBRTC_SPL_WORD16_MAX)
- s_sum = WEBRTC_SPL_WORD16_MAX;
- else if (s_sum < WEBRTC_SPL_WORD16_MIN)
- s_sum = WEBRTC_SPL_WORD16_MIN;
-
- return (WebRtc_Word16)s_sum;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/add_sat_w32.c b/src/common_audio/signal_processing_library/main/source/add_sat_w32.c
deleted file mode 100644
index 6d83e75..0000000
--- a/src/common_audio/signal_processing_library/main/source/add_sat_w32.c
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AddSatW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2)
-{
- WebRtc_Word32 l_sum;
-
- // perform long addition
- l_sum = var1 + var2;
-
- // check for under or overflow
- if (WEBRTC_SPL_IS_NEG(var1))
- {
- if (WEBRTC_SPL_IS_NEG(var2) && !WEBRTC_SPL_IS_NEG(l_sum))
- {
- l_sum = (WebRtc_Word32)0x80000000;
- }
- } else
- {
- if (!WEBRTC_SPL_IS_NEG(var2) && WEBRTC_SPL_IS_NEG(l_sum))
- {
- l_sum = (WebRtc_Word32)0x7FFFFFFF;
- }
- }
-
- return l_sum;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/complex_fft.c b/src/common_audio/signal_processing_library/main/source/complex_fft.c
deleted file mode 100644
index b6f0c4e..0000000
--- a/src/common_audio/signal_processing_library/main/source/complex_fft.c
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CFFTSFT 14
-#define CFFTRND 1
-#define CFFTRND2 16384
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-extern "C" int FFT_4OFQ14(void *src, void *dest, int NC, int shift);
-
-// For detailed description of the fft functions, check the readme files in fft_ARM9E folder.
-int WebRtcSpl_ComplexFFT2(WebRtc_Word16 frfi[], WebRtc_Word16 frfiOut[], int stages, int mode)
-{
- return FFT_4OFQ14(frfi, frfiOut, 1 << stages, 0);
-}
-#endif
-
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
- int i, j, l, k, istep, n, m;
- WebRtc_Word16 wr, wi;
- WebRtc_Word32 tr32, ti32, qr32, qi32;
-
- /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
- * and should not be changed depending on the input parameter 'stages'
- */
- n = 1 << stages;
- if (n > 1024)
- return -1;
-
- l = 1;
- k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
- depending on the input parameter 'stages' */
-
- if (mode == 0)
- {
- // mode==0: Low-complexity and Low-accuracy mode
- while (l < n)
- {
- istep = l << 1;
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = WebRtcSpl_kSinTable1024[j + 256];
- wi = -WebRtcSpl_kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
- + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
-
- qr32 = (WebRtc_Word32)frfi[2 * i];
- qi32 = (WebRtc_Word32)frfi[2 * i + 1];
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
- }
- }
-
- --k;
- l = istep;
-
- }
-
- } else
- {
- // mode==1: High-complexity and High-accuracy mode
- while (l < n)
- {
- istep = l << 1;
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = WebRtcSpl_kSinTable1024[j + 256];
- wi = -WebRtcSpl_kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND),
- 15 - CFFTSFT);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
- + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND), 15 - CFFTSFT);
-
- qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
- qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
- }
- }
-
- --k;
- l = istep;
- }
- }
- return 0;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/complex_ifft.c b/src/common_audio/signal_processing_library/main/source/complex_ifft.c
deleted file mode 100644
index 184b8de..0000000
--- a/src/common_audio/signal_processing_library/main/source/complex_ifft.c
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexIFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CIFFTSFT 14
-#define CIFFTRND 1
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-extern "C" int FFT_4OIQ14(void *src, void *dest, int NC, int shift);
-
-// For detailed description of the fft functions, check the readme files in fft_ARM9E folder.
-int WebRtcSpl_ComplexIFFT2(WebRtc_Word16 frfi[], WebRtc_Word16 frfiOut[], int stages, int mode)
-{
- FFT_4OIQ14(frfi, frfiOut, 1 << stages, 0);
- return 0;
-}
-#endif
-
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
- int i, j, l, k, istep, n, m, scale, shift;
- WebRtc_Word16 wr, wi;
- WebRtc_Word32 tr32, ti32, qr32, qi32;
- WebRtc_Word32 tmp32, round2;
-
- /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
- * and should not be changed depending on the input parameter 'stages'
- */
- n = 1 << stages;
- if (n > 1024)
- return -1;
-
- scale = 0;
-
- l = 1;
- k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
- depending on the input parameter 'stages' */
-
- while (l < n)
- {
- // variable scaling, depending upon data
- shift = 0;
- round2 = 8192;
-
- tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
- if (tmp32 > 13573)
- {
- shift++;
- scale++;
- round2 <<= 1;
- }
- if (tmp32 > 27146)
- {
- shift++;
- scale++;
- round2 <<= 1;
- }
-
- istep = l << 1;
-
- if (mode == 0)
- {
- // mode==0: Low-complexity and Low-accuracy mode
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = WebRtcSpl_kSinTable1024[j + 256];
- wi = WebRtcSpl_kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
- - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
- + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
-
- qr32 = (WebRtc_Word32)frfi[2 * i];
- qi32 = (WebRtc_Word32)frfi[2 * i + 1];
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
- }
- }
- } else
- {
- // mode==1: High-complexity and High-accuracy mode
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = WebRtcSpl_kSinTable1024[j + 256];
- wi = WebRtcSpl_kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
- - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0) + CIFFTRND),
- 15 - CIFFTSFT);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
- + WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j], 0)
- + CIFFTRND), 15 - CIFFTSFT);
-
- qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
- qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
- shift+CIFFTSFT);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 - ti32 + round2), shift + CIFFTSFT);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
- shift + CIFFTSFT);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 + ti32 + round2), shift + CIFFTSFT);
- }
- }
-
- }
- --k;
- l = istep;
- }
- return scale;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/cos_table.c b/src/common_audio/signal_processing_library/main/source/cos_table.c
deleted file mode 100644
index 7dba4b0..0000000
--- a/src/common_audio/signal_processing_library/main/source/cos_table.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree cos table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kCosTable[] = {
- 8192, 8190, 8187, 8180, 8172, 8160, 8147, 8130, 8112,
- 8091, 8067, 8041, 8012, 7982, 7948, 7912, 7874, 7834,
- 7791, 7745, 7697, 7647, 7595, 7540, 7483, 7424, 7362,
- 7299, 7233, 7164, 7094, 7021, 6947, 6870, 6791, 6710,
- 6627, 6542, 6455, 6366, 6275, 6182, 6087, 5991, 5892,
- 5792, 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
- 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971, 3845,
- 3719, 3591, 3462, 3331, 3200, 3068, 2935, 2801, 2667,
- 2531, 2395, 2258, 2120, 1981, 1842, 1703, 1563, 1422,
- 1281, 1140, 998, 856, 713, 571, 428, 285, 142,
- 0, -142, -285, -428, -571, -713, -856, -998, -1140,
- -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
- -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
- -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
- -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
- -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
- -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
- -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
- -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
- -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
- -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
- -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
- -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
- -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
- -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
- -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
- -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
- -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
- -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
- -1281, -1140, -998, -856, -713, -571, -428, -285, -142,
- 0, 142, 285, 428, 571, 713, 856, 998, 1140,
- 1281, 1422, 1563, 1703, 1842, 1981, 2120, 2258, 2395,
- 2531, 2667, 2801, 2935, 3068, 3200, 3331, 3462, 3591,
- 3719, 3845, 3971, 4095, 4219, 4341, 4461, 4580, 4698,
- 4815, 4930, 5043, 5155, 5265, 5374, 5481, 5586, 5690,
- 5792, 5892, 5991, 6087, 6182, 6275, 6366, 6455, 6542,
- 6627, 6710, 6791, 6870, 6947, 7021, 7094, 7164, 7233,
- 7299, 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
- 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041, 8067,
- 8091, 8112, 8130, 8147, 8160, 8172, 8180, 8187, 8190
-};
diff --git a/src/common_audio/signal_processing_library/main/source/get_hanning_window.c b/src/common_audio/signal_processing_library/main/source/get_hanning_window.c
deleted file mode 100644
index 2845c83..0000000
--- a/src/common_audio/signal_processing_library/main/source/get_hanning_window.c
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetHanningWindow().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
-{
- int jj;
- WebRtc_Word16 *vptr1;
-
- WebRtc_Word32 index;
- WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
-
- factor = WebRtcSpl_DivW32W16(factor, size);
- if (size < 513)
- index = (WebRtc_Word32)-0x200000;
- else
- index = (WebRtc_Word32)-0x100000;
- vptr1 = v;
-
- for (jj = 0; jj < size; jj++)
- {
- index += factor;
- (*vptr1++) = WebRtcSpl_kHanningTable[index >> 22];
- }
-
-}
diff --git a/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c b/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c
deleted file mode 100644
index 53853f0..0000000
--- a/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetSizeInBits().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 value)
-{
-
- int bits = 0;
-
- // Fast binary search to find the number of bits used
- if ((0xFFFF0000 & value))
- bits = 16;
- if ((0x0000FF00 & (value >> bits)))
- bits += 8;
- if ((0x000000F0 & (value >> bits)))
- bits += 4;
- if ((0x0000000C & (value >> bits)))
- bits += 2;
- if ((0x00000002 & (value >> bits)))
- bits += 1;
- if ((0x00000001 & (value >> bits)))
- bits += 1;
-
- return bits;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_u32.c b/src/common_audio/signal_processing_library/main/source/norm_u32.c
deleted file mode 100644
index c903a64..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_u32.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormU32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormU32(WebRtc_UWord32 value)
-{
- int zeros = 0;
-
- if (value == 0)
- return 0;
-
- if (!(0xFFFF0000 & value))
- zeros = 16;
- if (!(0xFF000000 & (value << zeros)))
- zeros += 8;
- if (!(0xF0000000 & (value << zeros)))
- zeros += 4;
- if (!(0xC0000000 & (value << zeros)))
- zeros += 2;
- if (!(0x80000000 & (value << zeros)))
- zeros += 1;
-
- return zeros;
-}
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_w16.c b/src/common_audio/signal_processing_library/main/source/norm_w16.c
deleted file mode 100644
index be6711d..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_w16.c
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormW16(WebRtc_Word16 value)
-{
- int zeros = 0;
-
- if (value <= 0)
- value ^= 0xFFFF;
-
- if ( !(0xFF80 & value))
- zeros = 8;
- if ( !(0xF800 & (value << zeros)))
- zeros += 4;
- if ( !(0xE000 & (value << zeros)))
- zeros += 2;
- if ( !(0xC000 & (value << zeros)))
- zeros += 1;
-
- return zeros;
-}
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_w32.c b/src/common_audio/signal_processing_library/main/source/norm_w32.c
deleted file mode 100644
index d456335..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_w32.c
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormW32(WebRtc_Word32 value)
-{
- int zeros = 0;
-
- if (value <= 0)
- value ^= 0xFFFFFFFF;
-
- // Fast binary search to determine the number of left shifts required to 32-bit normalize
- // the value
- if (!(0xFFFF8000 & value))
- zeros = 16;
- if (!(0xFF800000 & (value << zeros)))
- zeros += 8;
- if (!(0xF8000000 & (value << zeros)))
- zeros += 4;
- if (!(0xE0000000 & (value << zeros)))
- zeros += 2;
- if (!(0xC0000000 & (value << zeros)))
- zeros += 1;
-
- return zeros;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/randomization_functions.c b/src/common_audio/signal_processing_library/main/source/randomization_functions.c
deleted file mode 100644
index 6bc87c7..0000000
--- a/src/common_audio/signal_processing_library/main/source/randomization_functions.c
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the randomization functions
- * WebRtcSpl_IncreaseSeed()
- * WebRtcSpl_RandU()
- * WebRtcSpl_RandN()
- * WebRtcSpl_RandUArray()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
-{
- seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
- return seed[0];
-}
-
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
-{
- return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
-}
-
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
-{
- return WebRtcSpl_kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
-}
-
-// Creates an array of uniformly distributed variables
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
- WebRtc_Word16 vector_length,
- WebRtc_UWord32* seed)
-{
- int i;
- for (i = 0; i < vector_length; i++)
- {
- vector[i] = WebRtcSpl_RandU(seed);
- }
- return vector_length;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2.c b/src/common_audio/signal_processing_library/main/source/resample_by_2.c
deleted file mode 100644
index 7ed4cfd..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_by_2.c
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling by two functions.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-// allpass filter coefficients.
-static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
-static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
-
-// decimator
-void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
- WebRtc_Word16* out, WebRtc_Word32* filtState)
-{
- const WebRtc_Word16 *inptr;
- WebRtc_Word16 *outptr;
- WebRtc_Word32 *state;
- WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
- WebRtc_Word16 i;
-
- // local versions of pointers to input and output arrays
- inptr = in; // input array
- outptr = out; // output array (of length len/2)
- state = filtState; // filter state array; length = 8
-
- for (i = (len >> 1); i > 0; i--)
- {
- // lower allpass filter
- in32 = (WebRtc_Word32)(*inptr++) << 10;
- diff = in32 - state[1];
- tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[0], diff, state[0] );
- state[0] = in32;
- diff = tmp1 - state[2];
- tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[1], diff, state[1] );
- state[1] = tmp1;
- diff = tmp2 - state[3];
- state[3] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[2], diff, state[2] );
- state[2] = tmp2;
-
- // upper allpass filter
- in32 = (WebRtc_Word32)(*inptr++) << 10;
- diff = in32 - state[5];
- tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[0], diff, state[4] );
- state[4] = in32;
- diff = tmp1 - state[6];
- tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[1], diff, state[5] );
- state[5] = tmp1;
- diff = tmp2 - state[7];
- state[7] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[2], diff, state[6] );
- state[6] = tmp2;
-
- // add two allpass outputs, divide by two and round
- out32 = (state[3] + state[7] + 1024) >> 11;
-
- // limit amplitude to prevent wrap-around, and write to output array
- if (out32 > 32767)
- *outptr++ = 32767;
- else if (out32 < -32768)
- *outptr++ = -32768;
- else
- *outptr++ = (WebRtc_Word16)out32;
- }
-}
-
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
- WebRtc_Word32* filtState)
-{
- const WebRtc_Word16 *inptr;
- WebRtc_Word16 *outptr;
- WebRtc_Word32 *state;
- WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
- WebRtc_Word16 i;
-
- // local versions of pointers to input and output arrays
- inptr = in; // input array
- outptr = out; // output array (of length len*2)
- state = filtState; // filter state array; length = 8
-
- for (i = len; i > 0; i--)
- {
- // lower allpass filter
- in32 = (WebRtc_Word32)(*inptr++) << 10;
- diff = in32 - state[1];
- tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[0], diff, state[0] );
- state[0] = in32;
- diff = tmp1 - state[2];
- tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[1], diff, state[1] );
- state[1] = tmp1;
- diff = tmp2 - state[3];
- state[3] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[2], diff, state[2] );
- state[2] = tmp2;
-
- // round; limit amplitude to prevent wrap-around; write to output array
- out32 = (state[3] + 512) >> 10;
- if (out32 > 32767)
- *outptr++ = 32767;
- else if (out32 < -32768)
- *outptr++ = -32768;
- else
- *outptr++ = (WebRtc_Word16)out32;
-
- // upper allpass filter
- diff = in32 - state[5];
- tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[0], diff, state[4] );
- state[4] = in32;
- diff = tmp1 - state[6];
- tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[1], diff, state[5] );
- state[5] = tmp1;
- diff = tmp2 - state[7];
- state[7] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[2], diff, state[6] );
- state[6] = tmp2;
-
- // round; limit amplitude to prevent wrap-around; write to output array
- out32 = (state[7] + 512) >> 10;
- if (out32 > 32767)
- *outptr++ = 32767;
- else if (out32 < -32768)
- *outptr++ = -32768;
- else
- *outptr++ = (WebRtc_Word16)out32;
- }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/sin_table.c b/src/common_audio/signal_processing_library/main/source/sin_table.c
deleted file mode 100644
index ea44666..0000000
--- a/src/common_audio/signal_processing_library/main/source/sin_table.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable[] = {
- 0, 142, 285, 428, 571, 713, 856, 998, 1140,
- 1281, 1422, 1563, 1703, 1842, 1981, 2120, 2258, 2395,
- 2531, 2667, 2801, 2935, 3068, 3200, 3331, 3462, 3591,
- 3719, 3845, 3971, 4095, 4219, 4341, 4461, 4580, 4698,
- 4815, 4930, 5043, 5155, 5265, 5374, 5481, 5586, 5690,
- 5792, 5892, 5991, 6087, 6182, 6275, 6366, 6455, 6542,
- 6627, 6710, 6791, 6870, 6947, 7021, 7094, 7164, 7233,
- 7299, 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
- 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041, 8067,
- 8091, 8112, 8130, 8147, 8160, 8172, 8180, 8187, 8190,
- 8191, 8190, 8187, 8180, 8172, 8160, 8147, 8130, 8112,
- 8091, 8067, 8041, 8012, 7982, 7948, 7912, 7874, 7834,
- 7791, 7745, 7697, 7647, 7595, 7540, 7483, 7424, 7362,
- 7299, 7233, 7164, 7094, 7021, 6947, 6870, 6791, 6710,
- 6627, 6542, 6455, 6366, 6275, 6182, 6087, 5991, 5892,
- 5792, 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
- 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971, 3845,
- 3719, 3591, 3462, 3331, 3200, 3068, 2935, 2801, 2667,
- 2531, 2395, 2258, 2120, 1981, 1842, 1703, 1563, 1422,
- 1281, 1140, 998, 856, 713, 571, 428, 285, 142,
- 0, -142, -285, -428, -571, -713, -856, -998, -1140,
- -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
- -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
- -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
- -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
- -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
- -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
- -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
- -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
- -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
- -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
- -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
- -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
- -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
- -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
- -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
- -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
- -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
- -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
- -1281, -1140, -998, -856, -713, -571, -428, -285, -142
-};
diff --git a/src/common_audio/signal_processing_library/main/source/sin_table_1024.c b/src/common_audio/signal_processing_library/main/source/sin_table_1024.c
deleted file mode 100644
index a2007f9..0000000
--- a/src/common_audio/signal_processing_library/main/source/sin_table_1024.c
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 1024 point sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable1024[] =
-{
- 0, 201, 402, 603, 804, 1005, 1206, 1406,
- 1607, 1808, 2009, 2209, 2410, 2610, 2811, 3011,
- 3211, 3411, 3611, 3811, 4011, 4210, 4409, 4608,
- 4807, 5006, 5205, 5403, 5601, 5799, 5997, 6195,
- 6392, 6589, 6786, 6982, 7179, 7375, 7571, 7766,
- 7961, 8156, 8351, 8545, 8739, 8932, 9126, 9319,
- 9511, 9703, 9895, 10087, 10278, 10469, 10659, 10849,
- 11038, 11227, 11416, 11604, 11792, 11980, 12166, 12353,
- 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
- 14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268,
- 15446, 15623, 15799, 15975, 16150, 16325, 16499, 16672,
- 16845, 17017, 17189, 17360, 17530, 17699, 17868, 18036,
- 18204, 18371, 18537, 18702, 18867, 19031, 19194, 19357,
- 19519, 19680, 19840, 20000, 20159, 20317, 20474, 20631,
- 20787, 20942, 21096, 21249, 21402, 21554, 21705, 21855,
- 22004, 22153, 22301, 22448, 22594, 22739, 22883, 23027,
- 23169, 23311, 23452, 23592, 23731, 23869, 24006, 24143,
- 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
- 25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198,
- 26318, 26437, 26556, 26673, 26789, 26905, 27019, 27132,
- 27244, 27355, 27466, 27575, 27683, 27790, 27896, 28001,
- 28105, 28208, 28309, 28410, 28510, 28608, 28706, 28802,
- 28897, 28992, 29085, 29177, 29268, 29358, 29446, 29534,
- 29621, 29706, 29790, 29873, 29955, 30036, 30116, 30195,
- 30272, 30349, 30424, 30498, 30571, 30643, 30713, 30783,
- 30851, 30918, 30984, 31049,
- 31113, 31175, 31236, 31297,
- 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
- 31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097,
- 32137, 32176, 32213, 32249, 32284, 32318, 32350, 32382,
- 32412, 32441, 32468, 32495, 32520, 32544, 32567, 32588,
- 32609, 32628, 32646, 32662, 32678, 32692, 32705, 32717,
- 32727, 32736, 32744, 32751, 32757, 32761, 32764, 32766,
- 32767, 32766, 32764, 32761, 32757, 32751, 32744, 32736,
- 32727, 32717, 32705, 32692, 32678, 32662, 32646, 32628,
- 32609, 32588, 32567, 32544, 32520, 32495, 32468, 32441,
- 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
- 32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833,
- 31785, 31735, 31684, 31633, 31580, 31525, 31470, 31413,
- 31356, 31297, 31236, 31175, 31113, 31049, 30984, 30918,
- 30851, 30783, 30713, 30643, 30571, 30498, 30424, 30349,
- 30272, 30195, 30116, 30036, 29955, 29873, 29790, 29706,
- 29621, 29534, 29446, 29358, 29268, 29177, 29085, 28992,
- 28897, 28802, 28706, 28608, 28510, 28410, 28309, 28208,
- 28105, 28001, 27896, 27790, 27683, 27575, 27466, 27355,
- 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
- 26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456,
- 25329, 25201, 25072, 24942, 24811, 24679, 24546, 24413,
- 24278, 24143, 24006, 23869, 23731, 23592, 23452, 23311,
- 23169, 23027, 22883, 22739, 22594, 22448, 22301, 22153,
- 22004, 21855, 21705, 21554, 21402, 21249, 21096, 20942,
- 20787, 20631, 20474, 20317, 20159, 20000, 19840, 19680,
- 19519, 19357, 19194, 19031, 18867, 18702, 18537, 18371,
- 18204, 18036, 17868, 17699, 17530, 17360, 17189, 17017,
- 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
- 15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191,
- 14009, 13827, 13645, 13462, 13278, 13094, 12909, 12724,
- 12539, 12353, 12166, 11980, 11792, 11604, 11416, 11227,
- 11038, 10849, 10659, 10469, 10278, 10087, 9895, 9703,
- 9511, 9319, 9126, 8932, 8739, 8545, 8351, 8156,
- 7961, 7766, 7571, 7375, 7179, 6982, 6786, 6589,
- 6392, 6195, 5997, 5799, 5601, 5403, 5205, 5006,
- 4807, 4608, 4409, 4210, 4011, 3811, 3611, 3411,
- 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
- 1607, 1406, 1206, 1005, 804, 603, 402, 201,
- 0, -201, -402, -603, -804, -1005, -1206, -1406,
- -1607, -1808, -2009, -2209, -2410, -2610, -2811, -3011,
- -3211, -3411, -3611, -3811, -4011, -4210, -4409, -4608,
- -4807, -5006, -5205, -5403, -5601, -5799, -5997, -6195,
- -6392, -6589, -6786, -6982, -7179, -7375, -7571, -7766,
- -7961, -8156, -8351, -8545, -8739, -8932, -9126, -9319,
- -9511, -9703, -9895, -10087, -10278, -10469, -10659, -10849,
- -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
- -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
- -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
- -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
- -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
- -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
- -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
- -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
- -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
- -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
- -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
- -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
- -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
- -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
- -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
- -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
- -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
- -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
- -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
- -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
- -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
- -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
- -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
- -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
- -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
- -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
- -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
- -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
- -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
- -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
- -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
- -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
- -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
- -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
- -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
- -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
- -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
- -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
- -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
- -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
- -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
- -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
- -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
- -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
- -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
- -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
- -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
- -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
- -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
- -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
- -11038, -10849, -10659, -10469, -10278, -10087, -9895, -9703,
- -9511, -9319, -9126, -8932, -8739, -8545, -8351, -8156,
- -7961, -7766, -7571, -7375, -7179, -6982, -6786, -6589,
- -6392, -6195, -5997, -5799, -5601, -5403, -5205, -5006,
- -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
- -3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808,
- -1607, -1406, -1206, -1005, -804, -603, -402, -201,
-};
diff --git a/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c b/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c
deleted file mode 100644
index a48c3d5..0000000
--- a/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SubSatW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-#ifndef XSCALE_OPT
-
-WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2)
-{
- WebRtc_Word32 l_diff;
- WebRtc_Word16 s_diff;
-
- // perform subtraction
- l_diff = (WebRtc_Word32)var1 - (WebRtc_Word32)var2;
-
- // default setting
- s_diff = (WebRtc_Word16)l_diff;
-
- // check for overflow
- if (l_diff > (WebRtc_Word32)32767)
- s_diff = (WebRtc_Word16)32767;
-
- // check for underflow
- if (l_diff < (WebRtc_Word32)-32768)
- s_diff = (WebRtc_Word16)-32768;
-
- return s_diff;
-}
-
-#else
-#pragma message(">> WebRtcSpl_SubSatW16.c is excluded from this build")
-#endif // XSCALE_OPT
-#endif // SPL_NO_DOUBLE_IMPLEMENTATIONS
diff --git a/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c b/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c
deleted file mode 100644
index add3675..0000000
--- a/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SubSatW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2)
-{
- WebRtc_Word32 l_diff;
-
- // perform subtraction
- l_diff = var1 - var2;
-
- // check for underflow
- if ((var1 < 0) && (var2 > 0) && (l_diff > 0))
- l_diff = (WebRtc_Word32)0x80000000;
- // check for overflow
- if ((var1 > 0) && (var2 < 0) && (l_diff < 0))
- l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
- return l_diff;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s b/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s
deleted file mode 100644
index c1a893b..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s
+++ /dev/null
@@ -1,227 +0,0 @@
- .globl FFT_4OFQ14
-
-FFT_4OFQ14:
- stmdb sp!, {r4 - r11, lr}
- ldr lr, =s_Q14S_8
- ldr lr, [lr]
- cmp r2, lr
- movgt r0, #1
- ldmgtia sp!, {r4 - r11, pc}
- stmdb sp!, {r1, r2}
- mov r3, #0
- mov r2, r2
-
-LBL1:
- add r12, r0, r3, lsl #2
- add r12, r12, r2, lsr #1
- ldrsh r5, [r12, #2]
- ldrsh r4, [r12], +r2
- ldrsh r9, [r12, #2]
- ldrsh r8, [r12], +r2
- ldrsh r7, [r12, #2]
- ldrsh r6, [r12], +r2
- ldrsh r11, [r12, #2]
- ldrsh r10, [r12], +r2
- add r4, r4, r6
- add r5, r5, r7
- sub r6, r4, r6, lsl #1
- sub r7, r5, r7, lsl #1
- sub r12, r8, r10
- sub lr, r9, r11
- add r10, r8, r10
- add r11, r9, r11
- sub r9, r4, r10
- sub r8, r5, r11
- add r4, r4, r10
- add r5, r5, r11
- sub r10, r6, lr
- add r11, r7, r12
- add r6, r6, lr
- sub r7, r7, r12
- ldr lr, =t_Q14R_rad8
- ldrsh lr, [lr]
- stmdb sp!, {r2}
- add r12, r6, r7
- mul r6, r12, lr
- rsb r12, r12, r7, lsl #1
- mul r7, r12, lr
- sub r12, r11, r10
- mul r10, r12, lr
- sub r12, r12, r11, lsl #1
- mul r11, r12, lr
- ldmia sp!, {r2}
- stmdb sp!, {r4 - r11}
- add r4, r0, r3, lsl #2
- ldrsh r7, [r4, #2]
- ldrsh r6, [r4], +r2
- ldrsh r11, [r4, #2]
- ldrsh r10, [r4], +r2
- ldrsh r9, [r4, #2]
- ldrsh r8, [r4], +r2
- ldrsh lr, [r4, #2]
- ldrsh r12, [r4], +r2
- mov r7, r7, asr #3
- mov r6, r6, asr #3
- add r6, r6, r8, asr #3
- add r7, r7, r9, asr #3
- sub r8, r6, r8, asr #2
- sub r9, r7, r9, asr #2
- sub r4, r10, r12
- sub r5, r11, lr
- add r10, r10, r12
- add r11, r11, lr
- add r6, r6, r10, asr #3
- add r7, r7, r11, asr #3
- sub r10, r6, r10, asr #2
- sub r11, r7, r11, asr #2
- sub r12, r8, r5, asr #3
- add lr, r9, r4, asr #3
- add r8, r8, r5, asr #3
- sub r9, r9, r4, asr #3
- ldmia sp!, {r4, r5}
- add r6, r6, r4, asr #3
- add r7, r7, r5, asr #3
- sub r4, r6, r4, asr #2
- sub r5, r7, r5, asr #2
- strh r7, [r1, #2]
- strh r6, [r1], #4
- ldmia sp!, {r6, r7}
- add r8, r8, r6, asr #17
- add r9, r9, r7, asr #17
- sub r6, r8, r6, asr #16
- sub r7, r9, r7, asr #16
- strh r9, [r1, #2]
- strh r8, [r1], #4
- ldmia sp!, {r8, r9}
- add r10, r10, r8, asr #3
- sub r11, r11, r9, asr #3
- sub r8, r10, r8, asr #2
- add r9, r11, r9, asr #2
- strh r11, [r1, #2]
- strh r10, [r1], #4
- ldmia sp!, {r10, r11}
- add r12, r12, r10, asr #17
- add lr, lr, r11, asr #17
- sub r10, r12, r10, asr #16
- sub r11, lr, r11, asr #16
- strh lr, [r1, #2]
- strh r12, [r1], #4
- strh r5, [r1, #2]
- strh r4, [r1], #4
- strh r7, [r1, #2]
- strh r6, [r1], #4
- strh r9, [r1, #2]
- strh r8, [r1], #4
- strh r11, [r1, #2]
- strh r10, [r1], #4
- eor r3, r3, r2, lsr #4
- tst r3, r2, lsr #4
- bne LBL1
-
- eor r3, r3, r2, lsr #5
- tst r3, r2, lsr #5
- bne LBL1
-
- mov r12, r2, lsr #6
-
-LBL2:
- eor r3, r3, r12
- tst r3, r12
- bne LBL1
-
- movs r12, r12, lsr #1
- bne LBL2
-
- ldmia sp!, {r1, r2}
- mov r3, r2, lsr #3
- mov r2, #0x20
- ldr r0, =t_Q14S_8
- cmp r3, #1
- beq LBL3
-
-LBL6:
- mov r3, r3, lsr #2
- stmdb sp!, {r1, r3}
- add r12, r2, r2, lsl #1
- add r1, r1, r12
- sub r3, r3, #1, 16
-
-LBL5:
- add r3, r3, r2, lsl #14
-
-LBL4:
- ldrsh r6, [r0], #2
- ldrsh r7, [r0], #2
- ldrsh r8, [r0], #2
- ldrsh r9, [r0], #2
- ldrsh r10, [r0], #2
- ldrsh r11, [r0], #2
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r5, r4
- mul r12, lr, r11
- add lr, r10, r11, lsl #1
- mla r11, r5, r10, r12
- mla r10, r4, lr, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r5, r4
- mul r12, lr, r9
- add lr, r8, r9, lsl #1
- mla r9, r5, r8, r12
- mla r8, r4, lr, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r5, r4
- mul r12, lr, r7
- add lr, r6, r7, lsl #1
- mla r7, r5, r6, r12
- mla r6, r4, lr, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1]
- mov r5, r5, asr #2
- mov r4, r4, asr #2
- add r12, r4, r6, asr #16
- add lr, r5, r7, asr #16
- sub r4, r4, r6, asr #16
- sub r5, r5, r7, asr #16
- add r6, r8, r10
- add r7, r9, r11
- sub r8, r8, r10
- sub r9, r9, r11
- add r10, r12, r6, asr #16
- add r11, lr, r7, asr #16
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- add r10, r4, r9, asr #16
- sub r11, r5, r8, asr #16
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- sub r10, r12, r6, asr #16
- sub r11, lr, r7, asr #16
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- sub r10, r4, r9, asr #16
- add r11, r5, r8, asr #16
- strh r11, [r1, #2]
- strh r10, [r1], #4
- subs r3, r3, #1, 16
- bge LBL4
- add r12, r2, r2, lsl #1
- add r1, r1, r12
- sub r0, r0, r12
- sub r3, r3, #1
- movs lr, r3, lsl #16
- bne LBL5
- add r0, r0, r12
- ldmia sp!, {r1, r3}
- mov r2, r2, lsl #2
- cmp r3, #2
- bgt LBL6
-
-LBL3:
- mov r0, #0
- ldmia sp!, {r4 - r11, pc}
- andeq r3, r1, r0, lsr #32
- andeq r10, r1, r12, ror #31
- andeq r3, r1, r8, lsr #32
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s b/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s
deleted file mode 100644
index cc93291..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s
+++ /dev/null
@@ -1,221 +0,0 @@
- .globl FFT_4OIQ14
-
-FFT_4OIQ14:
- stmdb sp!, {r4 - r11, lr}
- ldr lr, =s_Q14S_8
- ldr lr, [lr]
- cmp r2, lr
- movgt r0, #1
- ldmgtia sp!, {r4 - r11, pc}
- stmdb sp!, {r1, r2}
- mov r3, #0
- mov r2, r2
-
-LBL1:
- add r12, r0, r3, lsl #2
- add r12, r12, r2, lsr #1
- ldrsh r5, [r12, #2]
- ldrsh r4, [r12], +r2
- ldrsh r9, [r12, #2]
- ldrsh r8, [r12], +r2
- ldrsh r7, [r12, #2]
- ldrsh r6, [r12], +r2
- ldrsh r11, [r12, #2]
- ldrsh r10, [r12], +r2
- add r4, r4, r6
- add r5, r5, r7
- sub r6, r4, r6, lsl #1
- sub r7, r5, r7, lsl #1
- sub r12, r8, r10
- sub lr, r9, r11
- add r10, r8, r10
- add r11, r9, r11
- sub r9, r4, r10
- sub r8, r5, r11
- add r4, r4, r10
- add r5, r5, r11
- add r10, r6, lr
- sub r11, r7, r12
- sub r6, r6, lr
- add r7, r7, r12
- ldr lr, =t_Q14R_rad8
- ldrsh lr, [lr]
- stmdb sp!, {r2}
- sub r12, r6, r7
- mul r6, r12, lr
- add r12, r12, r7, lsl #1
- mul r7, r12, lr
- sub r12, r10, r11
- mul r11, r12, lr
- sub r12, r12, r10, lsl #1
- mul r10, r12, lr
- ldmia sp!, {r2}
- stmdb sp!, {r4 - r11}
- add r4, r0, r3, lsl #2
- ldrsh r7, [r4, #2]
- ldrsh r6, [r4], +r2
- ldrsh r11, [r4, #2]
- ldrsh r10, [r4], +r2
- ldrsh r9, [r4, #2]
- ldrsh r8, [r4], +r2
- ldrsh lr, [r4, #2]
- ldrsh r12, [r4], +r2
- add r6, r6, r8
- add r7, r7, r9
- sub r8, r6, r8, lsl #1
- sub r9, r7, r9, lsl #1
- sub r4, r10, r12
- sub r5, r11, lr
- add r10, r10, r12
- add r11, r11, lr
- add r6, r6, r10
- add r7, r7, r11
- sub r10, r6, r10, lsl #1
- sub r11, r7, r11, lsl #1
- add r12, r8, r5
- sub lr, r9, r4
- sub r8, r8, r5
- add r9, r9, r4
- ldmia sp!, {r4, r5}
- add r6, r6, r4
- add r7, r7, r5
- sub r4, r6, r4, lsl #1
- sub r5, r7, r5, lsl #1
- strh r7, [r1, #2]
- strh r6, [r1], #4
- ldmia sp!, {r6, r7}
- add r8, r8, r6, asr #14
- add r9, r9, r7, asr #14
- sub r6, r8, r6, asr #13
- sub r7, r9, r7, asr #13
- strh r9, [r1, #2]
- strh r8, [r1], #4
- ldmia sp!, {r8, r9}
- sub r10, r10, r8
- add r11, r11, r9
- add r8, r10, r8, lsl #1
- sub r9, r11, r9, lsl #1
- strh r11, [r1, #2]
- strh r10, [r1], #4
- ldmia sp!, {r10, r11}
- add r12, r12, r10, asr #14
- add lr, lr, r11, asr #14
- sub r10, r12, r10, asr #13
- sub r11, lr, r11, asr #13
- strh lr, [r1, #2]
- strh r12, [r1], #4
- strh r5, [r1, #2]
- strh r4, [r1], #4
- strh r7, [r1, #2]
- strh r6, [r1], #4
- strh r9, [r1, #2]
- strh r8, [r1], #4
- strh r11, [r1, #2]
- strh r10, [r1], #4
- eor r3, r3, r2, lsr #4
- tst r3, r2, lsr #4
- bne LBL1
- eor r3, r3, r2, lsr #5
- tst r3, r2, lsr #5
- bne LBL1
- mov r12, r2, lsr #6
-
-
-LBL2:
- eor r3, r3, r12
- tst r3, r12
- bne LBL1
- movs r12, r12, lsr #1
- bne LBL2
- ldmia sp!, {r1, r2}
- mov r3, r2, lsr #3
- mov r2, #0x20
- ldr r0, =t_Q14S_8
- cmp r3, #1
- beq LBL3
-
-LBL6:
- mov r3, r3, lsr #2
- stmdb sp!, {r1, r3}
- add r12, r2, r2, lsl #1
- add r1, r1, r12
- sub r3, r3, #1, 16
-
-LBL5:
- add r3, r3, r2, lsl #14
-
-LBL4:
- ldrsh r6, [r0], #2
- ldrsh r7, [r0], #2
- ldrsh r8, [r0], #2
- ldrsh r9, [r0], #2
- ldrsh r10, [r0], #2
- ldrsh r11, [r0], #2
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r4, r5
- mul r12, lr, r11
- add r11, r10, r11, lsl #1
- mla r10, r4, r10, r12
- mla r11, r5, r11, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r4, r5
- mul r12, lr, r9
- add r9, r8, r9, lsl #1
- mla r8, r4, r8, r12
- mla r9, r5, r9, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1], -r2
- sub lr, r4, r5
- mul r12, lr, r7
- add r7, r6, r7, lsl #1
- mla r6, r4, r6, r12
- mla r7, r5, r7, r12
- ldrsh r5, [r1, #2]
- ldrsh r4, [r1]
- add r12, r4, r6, asr #14
- add lr, r5, r7, asr #14
- sub r4, r4, r6, asr #14
- sub r5, r5, r7, asr #14
- add r6, r8, r10
- add r7, r9, r11
- sub r8, r8, r10
- sub r9, r9, r11
- add r10, r12, r6, asr #14
- add r11, lr, r7, asr #14
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- sub r10, r4, r9, asr #14
- add r11, r5, r8, asr #14
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- sub r10, r12, r6, asr #14
- sub r11, lr, r7, asr #14
- strh r11, [r1, #2]
- strh r10, [r1], +r2
- add r10, r4, r9, asr #14
- sub r11, r5, r8, asr #14
- strh r11, [r1, #2]
- strh r10, [r1], #4
- subs r3, r3, #1, 16
- bge LBL4
- add r12, r2, r2, lsl #1
- add r1, r1, r12
- sub r0, r0, r12
- sub r3, r3, #1
- movs lr, r3, lsl #16
- bne LBL5
- add r0, r0, r12
- ldmia sp!, {r1, r3}
- mov r2, r2, lsl #2
- cmp r3, #2
- bgt LBL6
-
-LBL3:
- mov r0, #0
- ldmia sp!, {r4 - r11, pc}
- andeq r3, r1, r0, lsr #32
- andeq r10, r1, r12, ror #31
- andeq r3, r1, r8, lsr #32
-
diff --git a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc b/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 5adc339..0000000
--- a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,479 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the SPL unit_test.
- *
- */
-
-#include "unit_test.h"
-#include "signal_processing_library.h"
-
-class SplEnvironment : public ::testing::Environment {
- public:
- virtual void SetUp() {
- }
- virtual void TearDown() {
- }
-};
-
-SplTest::SplTest()
-{
-}
-
-void SplTest::SetUp() {
-}
-
-void SplTest::TearDown() {
-}
-
-TEST_F(SplTest, MacroTest) {
- // Macros with inputs.
- int A = 10;
- int B = 21;
- int a = -3;
- int b = WEBRTC_SPL_WORD32_MAX;
- int nr = 2;
- int d_ptr1 = 0;
- int d_ptr2 = 0;
-
- EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
- EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
-
- EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
- EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
- EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
- WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
- EXPECT_EQ(65536, d_ptr2);
-
- EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
- EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
- EXPECT_EQ(-2147483645, WEBRTC_SPL_UMUL(a, b));
- b = WEBRTC_SPL_WORD16_MAX >> 1;
- EXPECT_EQ(65535, WEBRTC_SPL_UMUL_RSFT16(a, b));
- EXPECT_EQ(1073627139, WEBRTC_SPL_UMUL_16_16(a, b));
- EXPECT_EQ(16382, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
- EXPECT_EQ(-49149, WEBRTC_SPL_UMUL_32_16(a, b));
- EXPECT_EQ(65535, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
-
- a = b;
- b = -3;
- EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
- EXPECT_EQ(0, WEBRTC_SPL_UDIV(a, b));
-
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
- EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
- EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
-
- int a32 = WEBRTC_SPL_WORD32_MAX;
- int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
- int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
- EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
- EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
-
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
- EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
-
- EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
-
- EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
- EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
- EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
-
- EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
- EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
- EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
-
- EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
-
- // Shifting with negative numbers allowed
- // Positive means left shift
- EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, 1));
-
- // Shifting with negative numbers not allowed
- // We cannot do casting here due to signed/unsigned problem
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
-
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U32(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U32(a, 1));
-
- EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
-}
-
-TEST_F(SplTest, InlineTest) {
-
- WebRtc_Word16 a = 121;
- WebRtc_Word16 b = -17;
- WebRtc_Word32 A = 111121;
- WebRtc_Word32 B = -1711;
- char* bVersion = (char*) malloc(8);
-
- EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
- EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
-
- EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
- EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
-
- EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
- EXPECT_EQ(14, WebRtcSpl_NormW32(A));
- EXPECT_EQ(4, WebRtcSpl_NormW16(B));
- EXPECT_EQ(15, WebRtcSpl_NormU32(A));
-
- EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
-}
-
-TEST_F(SplTest, MathOperationsTest) {
-
- int A = 117;
- WebRtc_Word32 num = 117;
- WebRtc_Word32 den = -5;
- WebRtc_UWord16 denU = 5;
- EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
- EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
-
-
- EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
- EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
- EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
- EXPECT_EQ(23, WebRtcSpl_DivU32U16(num, denU));
- EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
-}
-
-TEST_F(SplTest, BasicArrayOperationsTest) {
-
-
- int B[] = {4, 12, 133, 1100};
- int Bs[] = {2, 6, 66, 550};
- WebRtc_UWord8* b8 = (WebRtc_UWord8*) malloc(4);
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
- WebRtc_UWord8* bTmp8 = (WebRtc_UWord8*) malloc(4);
- WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* bTmp32 = (WebRtc_Word32*) malloc(4);
-
- WebRtcSpl_MemSetW16(b16, 3, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(3, b16[kk]);
- }
- EXPECT_EQ(4, WebRtcSpl_ZerosArrayW16(b16, 4));
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(0, b16[kk]);
- }
- EXPECT_EQ(4, WebRtcSpl_OnesArrayW16(b16, 4));
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(1, b16[kk]);
- }
- WebRtcSpl_MemSetW32(b32, 3, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(3, b32[kk]);
- }
- EXPECT_EQ(4, WebRtcSpl_ZerosArrayW32(b32, 4));
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(0, b32[kk]);
- }
- EXPECT_EQ(4, WebRtcSpl_OnesArrayW32(b32, 4));
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(1, b32[kk]);
- }
- for (int kk = 0; kk < 4; ++kk) {
- bTmp8[kk] = (WebRtc_Word8)kk;
- bTmp16[kk] = (WebRtc_Word16)kk;
- bTmp32[kk] = (WebRtc_Word32)kk;
- }
- WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(b8[kk], bTmp8[kk]);
- }
- WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(b16[kk], bTmp16[kk]);
- }
-// WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, 4);
-// for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(b32[kk], bTmp32[kk]);
-// }
- EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, 4, 2, bTmp16));
- for (int kk = 0; kk < 2; ++kk) {
- EXPECT_EQ(kk+2, bTmp16[kk]);
- }
-
- for (int kk = 0; kk < 4; ++kk) {
- b32[kk] = B[kk];
- b16[kk] = (WebRtc_Word16)B[kk];
- }
- WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, 4, b32, 1);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
- }
- WebRtcSpl_VectorBitShiftW16(bTmp16, 4, b16, 1);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
- }
- WebRtcSpl_VectorBitShiftW32(bTmp32, 4, b32, 1);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
- }
-
- WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(b16[3-kk], bTmp16[kk]);
- }
-
-}
-
-TEST_F(SplTest, MinMaxOperationsTest) {
-
-
- int B[] = {4, 12, 133, -1100};
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
- for (int kk = 0; kk < 4; ++kk) {
- b16[kk] = B[kk];
- b32[kk] = B[kk];
- }
-
- EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, 4));
- EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, 4));
- EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, 4));
- EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, 4));
- EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, 4));
- EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, 4));
- EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, 4));
-
- EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, 4));
- EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, 4));
- EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, 4));
- EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, 4));
-
- EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, 4, 1));
-
-}
-
-TEST_F(SplTest, VectorOperationsTest) {
-
-
- int B[] = {4, 12, 133, 1100};
- WebRtc_Word16* a16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
- WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-
- for (int kk = 0; kk < 4; ++kk) {
- a16[kk] = B[kk];
- b16[kk] = B[kk];
- }
-
- WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
- }
-
- WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
- }
-
- WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, 4, 2, 2, 0);
- for (int kk = 0; kk < 2; ++kk) {
- EXPECT_EQ(614236, b32[kk]);
- }
-// EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
- EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, 4, 2));
-
- WebRtcSpl_ScaleVector(b16, bTmp16, 13, 4, 2);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, 4, 2);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, 4);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
- }
-
- WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, 4, 2);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
- }
- WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], 4, 2);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
- }
- WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, 4, 6);
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
- }
-
- WebRtcSpl_SqrtOfOneMinusXSquared(b16, 4, bTmp16);
- for (int kk = 0; kk < 3; ++kk) {
- EXPECT_EQ(32767, bTmp16[kk]);
- }
- EXPECT_EQ(32749, bTmp16[3]);
-}
-
-TEST_F(SplTest, EstimatorsTest) {
-
-
- int B[] = {4, 12, 133, 1100};
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
- WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-
- for (int kk = 0; kk < 4; ++kk) {
- b16[kk] = B[kk];
- b32[kk] = B[kk];
- }
-
- EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
-
-}
-
-TEST_F(SplTest, FilterTest) {
-
-
- WebRtc_Word16 A[] = {1, 2, 33, 100};
- WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
- WebRtc_Word16 B[] = {4, 12, 133, 110};
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word16* bTmp16Low = (WebRtc_Word16*) malloc(4);
- WebRtc_Word16* bState = (WebRtc_Word16*) malloc(4);
- WebRtc_Word16* bStateLow = (WebRtc_Word16*) malloc(4);
-
- WebRtcSpl_ZerosArrayW16(bState, 4);
- WebRtcSpl_ZerosArrayW16(bStateLow, 4);
-
- for (int kk = 0; kk < 4; ++kk) {
- b16[kk] = A[kk];
- }
-
- // MA filters
- WebRtcSpl_FilterMAFastQ12(b16, bTmp16, B, 4, 4);
- for (int kk = 0; kk < 4; ++kk) {
- //EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
- }
- // AR filters
- WebRtcSpl_FilterARFastQ12(b16, bTmp16, A, 4, 4);
- for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
- }
- EXPECT_EQ(4, WebRtcSpl_FilterAR(A5, 5, b16, 4, bState, 4, bStateLow, 4, bTmp16, bTmp16Low, 4));
-
-}
-
-TEST_F(SplTest, RandTest) {
-
-
- WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
- WebRtc_Word16 BN[] = {3459, -11689, -258, -3738};
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_UWord32* bSeed = (WebRtc_UWord32*) malloc(1);
-
- bSeed[0] = 100000;
-
- EXPECT_EQ(464449057, WebRtcSpl_IncreaseSeed(bSeed));
- EXPECT_EQ(31565, WebRtcSpl_RandU(bSeed));
- EXPECT_EQ(-9786, WebRtcSpl_RandN(bSeed));
- EXPECT_EQ(4, WebRtcSpl_RandUArray(b16, 4, bSeed));
- for (int kk = 0; kk < 4; ++kk) {
- EXPECT_EQ(BU[kk], b16[kk]);
- }
-}
-
-TEST_F(SplTest, SignalProcessingTest) {
-
-
- int A[] = {1, 2, 33, 100};
- WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
- WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
- WebRtc_Word32* bTmp32 = (WebRtc_Word32*) malloc(4);
-
- int bScale = 0;
-
- for (int kk = 0; kk < 4; ++kk) {
- b16[kk] = A[kk];
- b32[kk] = A[kk];
- }
-
- EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, 4, 1, bTmp32, &bScale));
- WebRtcSpl_ReflCoefToLpc(b16, 4, bTmp16);
-// for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-// }
- WebRtcSpl_LpcToReflCoef(bTmp16, 4, b16);
-// for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(a16[kk], b16[kk]);
-// }
- WebRtcSpl_AutoCorrToReflCoef(b32, 4, bTmp16);
-// for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-// }
- WebRtcSpl_GetHanningWindow(bTmp16, 4);
-// for (int kk = 0; kk < 4; ++kk) {
-// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-// }
-
- for (int kk = 0; kk < 4; ++kk) {
- b16[kk] = A[kk];
- }
- EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, 4, &bScale));
- EXPECT_EQ(0, bScale);
-}
-
-TEST_F(SplTest, FFTTest) {
-
-
- WebRtc_Word16 B[] = {1, 2, 33, 100,
- 2, 3, 34, 101,
- 3, 4, 35, 102,
- 4, 5, 36, 103};
-
- EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
-// for (int kk = 0; kk < 16; ++kk) {
-// EXPECT_EQ(A[kk], B[kk]);
-// }
- EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
-// for (int kk = 0; kk < 16; ++kk) {
-// EXPECT_EQ(A[kk], B[kk]);
-// }
- WebRtcSpl_ComplexBitReverse(B, 3);
- for (int kk = 0; kk < 16; ++kk) {
- //EXPECT_EQ(A[kk], B[kk]);
- }
-}
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
- SplEnvironment* env = new SplEnvironment;
- ::testing::AddGlobalTestEnvironment(env);
-
- return RUN_ALL_TESTS();
-}
diff --git a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h b/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h
deleted file mode 100644
index d7babe7..0000000
--- a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This header file contains the function WebRtcSpl_CopyFromBeginU8().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#ifndef WEBRTC_SPL_UNIT_TEST_H_
-#define WEBRTC_SPL_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class SplTest: public ::testing::Test
-{
-protected:
- SplTest();
- virtual void SetUp();
- virtual void TearDown();
-};
-
-#endif // WEBRTC_SPL_UNIT_TEST_H_
diff --git a/src/common_audio/vad/Android.mk b/src/common_audio/vad/Android.mk
new file mode 100644
index 0000000..b7be3f0
--- /dev/null
+++ b/src/common_audio/vad/Android.mk
@@ -0,0 +1,50 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_vad
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ webrtc_vad.c \
+ vad_core.c \
+ vad_filterbank.c \
+ vad_gmm.c \
+ vad_sp.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libdl \
+ libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/vad/OWNERS b/src/common_audio/vad/OWNERS
deleted file mode 100644
index 9132851..0000000
--- a/src/common_audio/vad/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-bjornv@google.com
-jks@google.com
diff --git a/src/common_audio/vad/main/interface/webrtc_vad.h b/src/common_audio/vad/include/webrtc_vad.h
similarity index 100%
rename from src/common_audio/vad/main/interface/webrtc_vad.h
rename to src/common_audio/vad/include/webrtc_vad.h
diff --git a/src/common_audio/vad/main/source/Android.mk b/src/common_audio/vad/main/source/Android.mk
deleted file mode 100644
index f52df93..0000000
--- a/src/common_audio/vad/main/source/Android.mk
+++ /dev/null
@@ -1,64 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_vad
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := webrtc_vad.c \
- vad_const.c \
- vad_core.c \
- vad_filterbank.c \
- vad_gmm.c \
- vad_sp.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/vad/main/source/vad_const.c b/src/common_audio/vad/main/source/vad_const.c
deleted file mode 100644
index 47b6a4b..0000000
--- a/src/common_audio/vad/main/source/vad_const.c
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes the constant values used internally in VAD.
- */
-
-#include "vad_const.h"
-
-// Spectrum Weighting
-const WebRtc_Word16 kSpectrumWeight[6] = {6, 8, 10, 12, 14, 16};
-
-const WebRtc_Word16 kCompVar = 22005;
-
-// Constant 160*log10(2) in Q9
-const WebRtc_Word16 kLogConst = 24660;
-
-// Constant log2(exp(1)) in Q12
-const WebRtc_Word16 kLog10Const = 5909;
-
-// Q15
-const WebRtc_Word16 kNoiseUpdateConst = 655;
-const WebRtc_Word16 kSpeechUpdateConst = 6554;
-
-// Q8
-const WebRtc_Word16 kBackEta = 154;
-
-// Coefficients used by WebRtcVad_HpOutput, Q14
-const WebRtc_Word16 kHpZeroCoefs[3] = {6631, -13262, 6631};
-const WebRtc_Word16 kHpPoleCoefs[3] = {16384, -7756, 5620};
-
-// Allpass filter coefficients, upper and lower, in Q15
-// Upper: 0.64, Lower: 0.17
-const WebRtc_Word16 kAllPassCoefsQ15[2] = {20972, 5571};
-const WebRtc_Word16 kAllPassCoefsQ13[2] = {5243, 1392}; // Q13
-
-// Minimum difference between the two models, Q5
-const WebRtc_Word16 kMinimumDifference[6] = {544, 544, 576, 576, 576, 576};
-
-// Upper limit of mean value for speech model, Q7
-const WebRtc_Word16 kMaximumSpeech[6] = {11392, 11392, 11520, 11520, 11520, 11520};
-
-// Minimum value for mean value
-const WebRtc_Word16 kMinimumMean[2] = {640, 768};
-
-// Upper limit of mean value for noise model, Q7
-const WebRtc_Word16 kMaximumNoise[6] = {9216, 9088, 8960, 8832, 8704, 8576};
-
-// Adjustment for division with two in WebRtcVad_SplitFilter
-const WebRtc_Word16 kOffsetVector[6] = {368, 368, 272, 176, 176, 176};
-
-// Start values for the Gaussian models, Q7
-// Weights for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataWeights[12] = {34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103};
-
-// Weights for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataWeights[12] = {48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81};
-
-// Means for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataMeans[12] = {6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863,
- 7820, 7266, 5020, 4362};
-
-// Means for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataMeans[12] = {8306, 10085, 10078, 11823, 11843, 6309, 9473,
- 9571, 10879, 7581, 8180, 7483};
-
-// Stds for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataStds[12] = {378, 1064, 493, 582, 688, 593, 474, 697, 475, 688,
- 421, 455};
-
-// Stds for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataStds[12] = {555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540,
- 1079, 850};
diff --git a/src/common_audio/vad/main/source/vad_const.h b/src/common_audio/vad/main/source/vad_const.h
deleted file mode 100644
index 8980437..0000000
--- a/src/common_audio/vad/main/source/vad_const.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the declarations of the internally used constants.
- */
-
-#ifndef WEBRTC_VAD_CONST_H_
-#define WEBRTC_VAD_CONST_H_
-
-#include "typedefs.h"
-
-// TODO(ajm): give these internal-linkage by moving to the appropriate file
-// where possible, and otherwise tag with WebRtcVad_.
-
-// Spectrum Weighting
-extern const WebRtc_Word16 kSpectrumWeight[];
-extern const WebRtc_Word16 kCompVar;
-// Logarithm constant
-extern const WebRtc_Word16 kLogConst;
-extern const WebRtc_Word16 kLog10Const;
-// Q15
-extern const WebRtc_Word16 kNoiseUpdateConst;
-extern const WebRtc_Word16 kSpeechUpdateConst;
-// Q8
-extern const WebRtc_Word16 kBackEta;
-// Coefficients used by WebRtcVad_HpOutput, Q14
-extern const WebRtc_Word16 kHpZeroCoefs[];
-extern const WebRtc_Word16 kHpPoleCoefs[];
-// Allpass filter coefficients, upper and lower, in Q15 resp. Q13
-extern const WebRtc_Word16 kAllPassCoefsQ15[];
-extern const WebRtc_Word16 kAllPassCoefsQ13[];
-// Minimum difference between the two models, Q5
-extern const WebRtc_Word16 kMinimumDifference[];
-// Maximum value when updating the speech model, Q7
-extern const WebRtc_Word16 kMaximumSpeech[];
-// Minimum value for mean value
-extern const WebRtc_Word16 kMinimumMean[];
-// Upper limit of mean value for noise model, Q7
-extern const WebRtc_Word16 kMaximumNoise[];
-// Adjustment for division with two in WebRtcVad_SplitFilter
-extern const WebRtc_Word16 kOffsetVector[];
-// Start values for the Gaussian models, Q7
-extern const WebRtc_Word16 kNoiseDataWeights[];
-extern const WebRtc_Word16 kSpeechDataWeights[];
-extern const WebRtc_Word16 kNoiseDataMeans[];
-extern const WebRtc_Word16 kSpeechDataMeans[];
-extern const WebRtc_Word16 kNoiseDataStds[];
-extern const WebRtc_Word16 kSpeechDataStds[];
-
-#endif // WEBRTC_VAD_CONST_H_
diff --git a/src/common_audio/vad/main/source/vad_filterbank.c b/src/common_audio/vad/main/source/vad_filterbank.c
deleted file mode 100644
index 11392c9..0000000
--- a/src/common_audio/vad/main/source/vad_filterbank.c
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal filterbank associated functions.
- * For function description, see vad_filterbank.h.
- */
-
-#include "vad_filterbank.h"
-#include "vad_defines.h"
-#include "vad_const.h"
-#include "signal_processing_library.h"
-
-void WebRtcVad_HpOutput(WebRtc_Word16 *in_vector,
- WebRtc_Word16 in_vector_length,
- WebRtc_Word16 *out_vector,
- WebRtc_Word16 *filter_state)
-{
- WebRtc_Word16 i, *pi, *outPtr;
- WebRtc_Word32 tmpW32;
-
- pi = &in_vector[0];
- outPtr = &out_vector[0];
-
- // The sum of the absolute values of the impulse response:
- // The zero/pole-filter has a max amplification of a single sample of: 1.4546
- // Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
- // The all-zero section has a max amplification of a single sample of: 1.6189
- // Impulse response: 0.4047 -0.8094 0.4047 0 0 0
- // The all-pole section has a max amplification of a single sample of: 1.9931
- // Impulse response: 1.0000 0.4734 -0.1189 -0.2187 -0.0627 0.04532
-
- for (i = 0; i < in_vector_length; i++)
- {
- // all-zero section (filter coefficients in Q14)
- tmpW32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], (*pi));
- tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
- tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]); // Q14
- filter_state[1] = filter_state[0];
- filter_state[0] = *pi++;
-
- // all-pole section
- tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]); // Q14
- tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
- filter_state[3] = filter_state[2];
- filter_state[2] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32 (tmpW32, 14);
- *outPtr++ = filter_state[2];
- }
-}
-
-void WebRtcVad_Allpass(WebRtc_Word16 *in_vector,
- WebRtc_Word16 *out_vector,
- WebRtc_Word16 filter_coefficients,
- int vector_length,
- WebRtc_Word16 *filter_state)
-{
- // The filter can only cause overflow (in the w16 output variable)
- // if more than 4 consecutive input numbers are of maximum value and
- // has the the same sign as the impulse responses first taps.
- // First 6 taps of the impulse response: 0.6399 0.5905 -0.3779
- // 0.2418 -0.1547 0.0990
-
- int n;
- WebRtc_Word16 tmp16;
- WebRtc_Word32 tmp32, in32, state32;
-
- state32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*filter_state)), 16); // Q31
-
- for (n = 0; n < vector_length; n++)
- {
-
- tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficients, (*in_vector));
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
- *out_vector++ = tmp16;
- in32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*in_vector)), 14);
- state32 = in32 - WEBRTC_SPL_MUL_16_16(filter_coefficients, tmp16);
- state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1);
- in_vector += 2;
- }
-
- *filter_state = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(state32, 16);
-}
-
-void WebRtcVad_SplitFilter(WebRtc_Word16 *in_vector,
- WebRtc_Word16 *out_vector_hp,
- WebRtc_Word16 *out_vector_lp,
- WebRtc_Word16 *upper_state,
- WebRtc_Word16 *lower_state,
- int in_vector_length)
-{
- WebRtc_Word16 tmpOut;
- int k, halflen;
-
- // Downsampling by 2 and get two branches
- halflen = WEBRTC_SPL_RSHIFT_W16(in_vector_length, 1);
-
- // All-pass filtering upper branch
- WebRtcVad_Allpass(&in_vector[0], out_vector_hp, kAllPassCoefsQ15[0], halflen, upper_state);
-
- // All-pass filtering lower branch
- WebRtcVad_Allpass(&in_vector[1], out_vector_lp, kAllPassCoefsQ15[1], halflen, lower_state);
-
- // Make LP and HP signals
- for (k = 0; k < halflen; k++)
- {
- tmpOut = *out_vector_hp;
- *out_vector_hp++ -= *out_vector_lp;
- *out_vector_lp++ += tmpOut;
- }
-}
-
-WebRtc_Word16 WebRtcVad_get_features(VadInstT *inst,
- WebRtc_Word16 *in_vector,
- int frame_size,
- WebRtc_Word16 *out_vector)
-{
- int curlen, filtno;
- WebRtc_Word16 vecHP1[120], vecLP1[120];
- WebRtc_Word16 vecHP2[60], vecLP2[60];
- WebRtc_Word16 *ptin;
- WebRtc_Word16 *hptout, *lptout;
- WebRtc_Word16 power = 0;
-
- // Split at 2000 Hz and downsample
- filtno = 0;
- ptin = in_vector;
- hptout = vecHP1;
- lptout = vecLP1;
- curlen = frame_size;
- WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
- &inst->lower_state[filtno], curlen);
-
- // Split at 3000 Hz and downsample
- filtno = 1;
- ptin = vecHP1;
- hptout = vecHP2;
- lptout = vecLP2;
- curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
-
- WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
- &inst->lower_state[filtno], curlen);
-
- // Energy in 3000 Hz - 4000 Hz
- curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
- WebRtcVad_LogOfEnergy(vecHP2, &out_vector[5], &power, kOffsetVector[5], curlen);
-
- // Energy in 2000 Hz - 3000 Hz
- WebRtcVad_LogOfEnergy(vecLP2, &out_vector[4], &power, kOffsetVector[4], curlen);
-
- // Split at 1000 Hz and downsample
- filtno = 2;
- ptin = vecLP1;
- hptout = vecHP2;
- lptout = vecLP2;
- curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
- WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
- &inst->lower_state[filtno], curlen);
-
- // Energy in 1000 Hz - 2000 Hz
- curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
- WebRtcVad_LogOfEnergy(vecHP2, &out_vector[3], &power, kOffsetVector[3], curlen);
-
- // Split at 500 Hz
- filtno = 3;
- ptin = vecLP2;
- hptout = vecHP1;
- lptout = vecLP1;
-
- WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
- &inst->lower_state[filtno], curlen);
-
- // Energy in 500 Hz - 1000 Hz
- curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
- WebRtcVad_LogOfEnergy(vecHP1, &out_vector[2], &power, kOffsetVector[2], curlen);
- // Split at 250 Hz
- filtno = 4;
- ptin = vecLP1;
- hptout = vecHP2;
- lptout = vecLP2;
-
- WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
- &inst->lower_state[filtno], curlen);
-
- // Energy in 250 Hz - 500 Hz
- curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
- WebRtcVad_LogOfEnergy(vecHP2, &out_vector[1], &power, kOffsetVector[1], curlen);
-
- // Remove DC and LFs
- WebRtcVad_HpOutput(vecLP2, curlen, vecHP1, inst->hp_filter_state);
-
- // Power in 80 Hz - 250 Hz
- WebRtcVad_LogOfEnergy(vecHP1, &out_vector[0], &power, kOffsetVector[0], curlen);
-
- return power;
-}
-
-void WebRtcVad_LogOfEnergy(WebRtc_Word16 *vector,
- WebRtc_Word16 *enerlogval,
- WebRtc_Word16 *power,
- WebRtc_Word16 offset,
- int vector_length)
-{
- WebRtc_Word16 enerSum = 0;
- WebRtc_Word16 zeros, frac, log2;
- WebRtc_Word32 energy;
-
- int shfts = 0, shfts2;
-
- energy = WebRtcSpl_Energy(vector, vector_length, &shfts);
-
- if (energy > 0)
- {
-
- shfts2 = 16 - WebRtcSpl_NormW32(energy);
- shfts += shfts2;
- // "shfts" is the total number of right shifts that has been done to enerSum.
- enerSum = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(energy, -shfts2);
-
- // Find:
- // 160*log10(enerSum*2^shfts) = 160*log10(2)*log2(enerSum*2^shfts) =
- // 160*log10(2)*(log2(enerSum) + log2(2^shfts)) =
- // 160*log10(2)*(log2(enerSum) + shfts)
-
- zeros = WebRtcSpl_NormU32(enerSum);
- frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_Word32)(enerSum) << zeros)
- & 0x7FFFFFFF) >> 21);
- log2 = (WebRtc_Word16)(((31 - zeros) << 10) + frac);
-
- *enerlogval = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kLogConst, log2, 19)
- + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(shfts, kLogConst, 9);
-
- if (*enerlogval < 0)
- {
- *enerlogval = 0;
- }
- } else
- {
- *enerlogval = 0;
- shfts = -15;
- enerSum = 0;
- }
-
- *enerlogval += offset;
-
- // Total power in frame
- if (*power <= MIN_ENERGY)
- {
- if (shfts > 0)
- {
- *power += MIN_ENERGY + 1;
- } else if (WEBRTC_SPL_SHIFT_W16(enerSum, shfts) > MIN_ENERGY)
- {
- *power += MIN_ENERGY + 1;
- } else
- {
- *power += WEBRTC_SPL_SHIFT_W16(enerSum, shfts);
- }
- }
-}
diff --git a/src/common_audio/vad/main/source/vad_gmm.c b/src/common_audio/vad/main/source/vad_gmm.c
deleted file mode 100644
index 23d12fb..0000000
--- a/src/common_audio/vad/main/source/vad_gmm.c
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal VAD call
- * WebRtcVad_GaussianProbability. For function description, see vad_gmm.h.
- */
-
-#include "vad_gmm.h"
-#include "signal_processing_library.h"
-#include "vad_const.h"
-
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
- WebRtc_Word16 mean,
- WebRtc_Word16 std,
- WebRtc_Word16 *delta)
-{
- WebRtc_Word16 tmp16, tmpDiv, tmpDiv2, expVal, tmp16_1, tmp16_2;
- WebRtc_Word32 tmp32, y32;
-
- // Calculate tmpDiv=1/std, in Q10
- tmp32 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W16(std,1) + (WebRtc_Word32)131072; // 1 in Q17
- tmpDiv = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32, std); // Q17/Q7 = Q10
-
- // Calculate tmpDiv2=1/std^2, in Q14
- tmp16 = WEBRTC_SPL_RSHIFT_W16(tmpDiv, 2); // From Q10 to Q8
- tmpDiv2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); // (Q8 * Q8)>>2 = Q14
-
- tmp16 = WEBRTC_SPL_LSHIFT_W16(in_sample, 3); // Q7
- tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
-
- // To be used later, when updating noise/speech model
- // delta = (x-m)/std^2, in Q11
- *delta = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmpDiv2, tmp16, 10); //(Q14*Q7)>>10 = Q11
-
- // Calculate tmp32=(x-m)^2/(2*std^2), in Q10
- tmp32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9); // One shift for /2
-
- // Calculate expVal ~= exp(-(x-m)^2/(2*std^2)) ~= exp2(-log2(exp(1))*tmp32)
- if (tmp32 < kCompVar)
- {
- // Calculate tmp16 = log2(exp(1))*tmp32 , in Q10
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)tmp32,
- kLog10Const, 12);
- tmp16 = -tmp16;
- tmp16_2 = (WebRtc_Word16)(0x0400 | (tmp16 & 0x03FF));
- tmp16_1 = (WebRtc_Word16)(tmp16 ^ 0xFFFF);
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(tmp16_1, 10);
- tmp16 += 1;
- // Calculate expVal=log2(-tmp32), in Q10
- expVal = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)tmp16_2, tmp16);
-
- } else
- {
- expVal = 0;
- }
-
- // Calculate y32=(1/std)*exp(-(x-m)^2/(2*std^2)), in Q20
- y32 = WEBRTC_SPL_MUL_16_16(tmpDiv, expVal); // Q10 * Q10 = Q20
-
- return y32; // Q20
-}
diff --git a/src/common_audio/vad/main/source/vad_gmm.h b/src/common_audio/vad/main/source/vad_gmm.h
deleted file mode 100644
index e0747fb..0000000
--- a/src/common_audio/vad/main/source/vad_gmm.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the description of the internal VAD call
- * WebRtcVad_GaussianProbability.
- */
-
-#ifndef WEBRTC_VAD_GMM_H_
-#define WEBRTC_VAD_GMM_H_
-
-#include "typedefs.h"
-
-/****************************************************************************
- * WebRtcVad_GaussianProbability(...)
- *
- * This function calculates the probability for the value 'in_sample', given that in_sample
- * comes from a normal distribution with mean 'mean' and standard deviation 'std'.
- *
- * Input:
- * - in_sample : Input sample in Q4
- * - mean : mean value in the statistical model, Q7
- * - std : standard deviation, Q7
- *
- * Output:
- *
- * - delta : Value used when updating the model, Q11
- *
- * Return:
- * - out : out = 1/std * exp(-(x-m)^2/(2*std^2));
- * Probability for x.
- *
- */
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
- WebRtc_Word16 mean,
- WebRtc_Word16 std,
- WebRtc_Word16 *delta);
-
-#endif // WEBRTC_VAD_GMM_H_
diff --git a/src/common_audio/vad/main/source/vad_sp.c b/src/common_audio/vad/main/source/vad_sp.c
deleted file mode 100644
index f347ab5..0000000
--- a/src/common_audio/vad/main/source/vad_sp.c
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD internal calls for Downsampling and
- * FindMinimum.
- * For function call descriptions; See vad_sp.h.
- */
-
-#include "vad_sp.h"
-#include "vad_defines.h"
-#include "vad_const.h"
-#include "signal_processing_library.h"
-
-// Downsampling filter based on the splitting filter and the allpass functions
-// in vad_filterbank.c
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
- WebRtc_Word16* signal_out,
- WebRtc_Word32* filter_state,
- int inlen)
-{
- WebRtc_Word16 tmp16_1, tmp16_2;
- WebRtc_Word32 tmp32_1, tmp32_2;
- int n, halflen;
-
- // Downsampling by 2 and get two branches
- halflen = WEBRTC_SPL_RSHIFT_W16(inlen, 1);
-
- tmp32_1 = filter_state[0];
- tmp32_2 = filter_state[1];
-
- // Filter coefficients in Q13, filter state in Q0
- for (n = 0; n < halflen; n++)
- {
- // All-pass filtering upper branch
- tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_1, 1)
- + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]),
- *signal_in, 14);
- *signal_out = tmp16_1;
- tmp32_1 = (WebRtc_Word32)(*signal_in++)
- - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]), tmp16_1, 12);
-
- // All-pass filtering lower branch
- tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_2, 1)
- + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]),
- *signal_in, 14);
- *signal_out++ += tmp16_2;
- tmp32_2 = (WebRtc_Word32)(*signal_in++)
- - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]), tmp16_2, 12);
- }
- filter_state[0] = tmp32_1;
- filter_state[1] = tmp32_2;
-}
-
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst,
- WebRtc_Word16 x,
- int n)
-{
- int i, j, k, II = -1, offset;
- WebRtc_Word16 meanV, alpha;
- WebRtc_Word32 tmp32, tmp32_1;
- WebRtc_Word16 *valptr, *idxptr, *p1, *p2, *p3;
-
- // Offset to beginning of the 16 minimum values in memory
- offset = WEBRTC_SPL_LSHIFT_W16(n, 4);
-
- // Pointer to memory for the 16 minimum values and the age of each value
- idxptr = &inst->index_vector[offset];
- valptr = &inst->low_value_vector[offset];
-
- // Each value in low_value_vector is getting 1 loop older.
- // Update age of each value in indexVal, and remove old values.
- for (i = 0; i < 16; i++)
- {
- p3 = idxptr + i;
- if (*p3 != 100)
- {
- *p3 += 1;
- } else
- {
- p1 = valptr + i + 1;
- p2 = p3 + 1;
- for (j = i; j < 16; j++)
- {
- *(valptr + j) = *p1++;
- *(idxptr + j) = *p2++;
- }
- *(idxptr + 15) = 101;
- *(valptr + 15) = 10000;
- }
- }
-
- // Check if x smaller than any of the values in low_value_vector.
- // If so, find position.
- if (x < *(valptr + 7))
- {
- if (x < *(valptr + 3))
- {
- if (x < *(valptr + 1))
- {
- if (x < *valptr)
- {
- II = 0;
- } else
- {
- II = 1;
- }
- } else if (x < *(valptr + 2))
- {
- II = 2;
- } else
- {
- II = 3;
- }
- } else if (x < *(valptr + 5))
- {
- if (x < *(valptr + 4))
- {
- II = 4;
- } else
- {
- II = 5;
- }
- } else if (x < *(valptr + 6))
- {
- II = 6;
- } else
- {
- II = 7;
- }
- } else if (x < *(valptr + 15))
- {
- if (x < *(valptr + 11))
- {
- if (x < *(valptr + 9))
- {
- if (x < *(valptr + 8))
- {
- II = 8;
- } else
- {
- II = 9;
- }
- } else if (x < *(valptr + 10))
- {
- II = 10;
- } else
- {
- II = 11;
- }
- } else if (x < *(valptr + 13))
- {
- if (x < *(valptr + 12))
- {
- II = 12;
- } else
- {
- II = 13;
- }
- } else if (x < *(valptr + 14))
- {
- II = 14;
- } else
- {
- II = 15;
- }
- }
-
- // Put new min value on right position and shift bigger values up
- if (II > -1)
- {
- for (i = 15; i > II; i--)
- {
- k = i - 1;
- *(valptr + i) = *(valptr + k);
- *(idxptr + i) = *(idxptr + k);
- }
- *(valptr + II) = x;
- *(idxptr + II) = 1;
- }
-
- meanV = 0;
- if ((inst->frame_counter) > 4)
- {
- j = 5;
- } else
- {
- j = inst->frame_counter;
- }
-
- if (j > 2)
- {
- meanV = *(valptr + 2);
- } else if (j > 0)
- {
- meanV = *valptr;
- } else
- {
- meanV = 1600;
- }
-
- if (inst->frame_counter > 0)
- {
- if (meanV < inst->mean_value[n])
- {
- alpha = (WebRtc_Word16)ALPHA1; // 0.2 in Q15
- } else
- {
- alpha = (WebRtc_Word16)ALPHA2; // 0.99 in Q15
- }
- } else
- {
- alpha = 0;
- }
-
- tmp32 = WEBRTC_SPL_MUL_16_16((alpha+1), inst->mean_value[n]);
- tmp32_1 = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, meanV);
- tmp32 += tmp32_1;
- tmp32 += 16384;
- inst->mean_value[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 15);
-
- return inst->mean_value[n];
-}
diff --git a/src/common_audio/vad/main/source/vad_sp.h b/src/common_audio/vad/main/source/vad_sp.h
deleted file mode 100644
index ae15c11..0000000
--- a/src/common_audio/vad/main/source/vad_sp.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD internal calls for Downsampling and FindMinimum.
- * Specific function calls are given below.
- */
-
-#ifndef WEBRTC_VAD_SP_H_
-#define WEBRTC_VAD_SP_H_
-
-#include "vad_core.h"
-
-/****************************************************************************
- * WebRtcVad_Downsampling(...)
- *
- * Downsamples the signal a factor 2, eg. 32->16 or 16->8
- *
- * Input:
- * - signal_in : Input signal
- * - in_length : Length of input signal in samples
- *
- * Input & Output:
- * - filter_state : Filter state for first all-pass filters
- *
- * Output:
- * - signal_out : Downsampled signal (of length len/2)
- */
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
- WebRtc_Word16* signal_out,
- WebRtc_Word32* filter_state,
- int in_length);
-
-/****************************************************************************
- * WebRtcVad_FindMinimum(...)
- *
- * Find the five lowest values of x in 100 frames long window. Return a mean
- * value of these five values.
- *
- * Input:
- * - feature_value : Feature value
- * - channel : Channel number
- *
- * Input & Output:
- * - inst : State information
- *
- * Output:
- * return value : Weighted minimum value for a moving window.
- */
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst, WebRtc_Word16 feature_value, int channel);
-
-#endif // WEBRTC_VAD_SP_H_
diff --git a/src/common_audio/vad/main/test/unit_test/unit_test.cc b/src/common_audio/vad/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 8ac793e..0000000
--- a/src/common_audio/vad/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD unit tests.
- */
-
-#include <cstring>
-#include "unit_test.h"
-#include "webrtc_vad.h"
-
-
-class VadEnvironment : public ::testing::Environment {
- public:
- virtual void SetUp() {
- }
-
- virtual void TearDown() {
- }
-};
-
-VadTest::VadTest()
-{
-}
-
-void VadTest::SetUp() {
-}
-
-void VadTest::TearDown() {
-}
-
-TEST_F(VadTest, ApiTest) {
- VadInst *vad_inst;
- int i, j, k;
- short zeros[960];
- short speech[960];
- char version[32];
-
- // Valid test cases
- int fs[3] = {8000, 16000, 32000};
- int nMode[4] = {0, 1, 2, 3};
- int framelen[3][3] = {{80, 160, 240},
- {160, 320, 480}, {320, 640, 960}} ;
- int vad_counter = 0;
-
- memset(zeros, 0, sizeof(short) * 960);
- memset(speech, 1, sizeof(short) * 960);
- speech[13] = 1374;
- speech[73] = -3747;
-
-
-
- // WebRtcVad_get_version()
- WebRtcVad_get_version(version);
- //printf("API Test for %s\n", version);
-
- // Null instance tests
- EXPECT_EQ(-1, WebRtcVad_Create(NULL));
- EXPECT_EQ(-1, WebRtcVad_Init(NULL));
- EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
- EXPECT_EQ(-1, WebRtcVad_Free(NULL));
- EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, nMode[0]));
- EXPECT_EQ(-1, WebRtcVad_Process(NULL, fs[0], speech, framelen[0][0]));
-
-
- EXPECT_EQ(WebRtcVad_Create(&vad_inst), 0);
-
- // Not initialized tests
- EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech, framelen[0][0]));
- EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, nMode[0]));
-
- // WebRtcVad_Init() tests
- EXPECT_EQ(WebRtcVad_Init(vad_inst), 0);
-
- // WebRtcVad_set_mode() tests
- EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, -1));
- EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, 4));
-
- for (i = 0; i < sizeof(nMode)/sizeof(nMode[0]); i++) {
- EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[i]), 0);
- }
-
- // WebRtcVad_Process() tests
- EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], NULL, framelen[0][0]));
- EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, 12000, speech, framelen[0][0]));
- EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech, framelen[1][1]));
- EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[0], zeros, framelen[0][0]), 0);
- for (i = 0; i < sizeof(fs)/sizeof(fs[0]); i++) {
- for (j = 0; j < sizeof(framelen[0])/sizeof(framelen[0][0]); j++) {
- for (k = 0; k < sizeof(nMode)/sizeof(nMode[0]); k++) {
- EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[k]), 0);
-// printf("%d\n", WebRtcVad_Process(vad_inst, fs[i], speech, framelen[i][j]));
- if (vad_counter < 9)
- {
- EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech, framelen[i][j]), 1);
- } else
- {
- EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech, framelen[i][j]), 0);
- }
- vad_counter++;
- }
- }
- }
-
- EXPECT_EQ(0, WebRtcVad_Free(vad_inst));
-
-}
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
- VadEnvironment* env = new VadEnvironment;
- ::testing::AddGlobalTestEnvironment(env);
-
- return RUN_ALL_TESTS();
-}
diff --git a/src/common_audio/vad/main/test/unit_test/unit_test.h b/src/common_audio/vad/main/test/unit_test/unit_test.h
deleted file mode 100644
index 62dac11..0000000
--- a/src/common_audio/vad/main/test/unit_test/unit_test.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the declaration of the VAD unit test.
- */
-
-#ifndef WEBRTC_VAD_UNIT_TEST_H_
-#define WEBRTC_VAD_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class VadTest : public ::testing::Test {
- protected:
- VadTest();
- virtual void SetUp();
- virtual void TearDown();
-};
-
-#endif // WEBRTC_VAD_UNIT_TEST_H_
diff --git a/src/common_audio/vad/main/source/vad.gyp b/src/common_audio/vad/vad.gypi
similarity index 61%
rename from src/common_audio/vad/main/source/vad.gyp
rename to src/common_audio/vad/vad.gypi
index 754b684..4b12db0 100644
--- a/src/common_audio/vad/main/source/vad.gyp
+++ b/src/common_audio/vad/vad.gypi
@@ -7,32 +7,27 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../common_settings.gypi', # Common settings
- ],
'targets': [
{
'target_name': 'vad',
'type': '<(library)',
'dependencies': [
- '../../../signal_processing_library/main/source/spl.gyp:spl',
+ 'signal_processing',
],
'include_dirs': [
- '../interface',
+ 'include',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'include',
],
},
'sources': [
- '../interface/webrtc_vad.h',
+ 'include/webrtc_vad.h',
'webrtc_vad.c',
- 'vad_const.c',
- 'vad_const.h',
- 'vad_defines.h',
'vad_core.c',
'vad_core.h',
+ 'vad_defines.h',
'vad_filterbank.c',
'vad_filterbank.h',
'vad_gmm.c',
@@ -41,7 +36,25 @@
'vad_sp.h',
],
},
- ],
+ ], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets' : [
+ {
+ 'target_name': 'vad_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'vad',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'vad_unittest.cc',
+ ],
+ }, # vad_unittests
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
}
# Local Variables:
diff --git a/src/common_audio/vad/main/source/vad_core.c b/src/common_audio/vad/vad_core.c
similarity index 92%
rename from src/common_audio/vad/main/source/vad_core.c
rename to src/common_audio/vad/vad_core.c
index e882999..e05c296 100644
--- a/src/common_audio/vad/main/source/vad_core.c
+++ b/src/common_audio/vad/vad_core.c
@@ -15,12 +15,50 @@
*/
#include "vad_core.h"
-#include "vad_const.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
#include "vad_defines.h"
#include "vad_filterbank.h"
#include "vad_gmm.h"
#include "vad_sp.h"
-#include "signal_processing_library.h"
+
+// Spectrum Weighting
+static const WebRtc_Word16 kSpectrumWeight[6] = { 6, 8, 10, 12, 14, 16 };
+static const WebRtc_Word16 kNoiseUpdateConst = 655; // Q15
+static const WebRtc_Word16 kSpeechUpdateConst = 6554; // Q15
+static const WebRtc_Word16 kBackEta = 154; // Q8
+// Minimum difference between the two models, Q5
+static const WebRtc_Word16 kMinimumDifference[6] = {
+ 544, 544, 576, 576, 576, 576 };
+// Upper limit of mean value for speech model, Q7
+static const WebRtc_Word16 kMaximumSpeech[6] = {
+ 11392, 11392, 11520, 11520, 11520, 11520 };
+// Minimum value for mean value
+static const WebRtc_Word16 kMinimumMean[2] = { 640, 768 };
+// Upper limit of mean value for noise model, Q7
+static const WebRtc_Word16 kMaximumNoise[6] = {
+ 9216, 9088, 8960, 8832, 8704, 8576 };
+// Start values for the Gaussian models, Q7
+// Weights for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataWeights[12] = {
+ 34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
+// Weights for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataWeights[12] = {
+ 48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
+// Means for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataMeans[12] = {
+ 6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
+// Means for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataMeans[12] = {
+ 8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
+};
+// Stds for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataStds[12] = {
+ 378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
+// Stds for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataStds[12] = {
+ 555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
static const int kInitCheck = 42;
@@ -651,10 +689,10 @@
*nmean2ptr -= tmp16_2;
}
- *nmean1ptr++;
- *smean1ptr++;
- *nstd1ptr++;
- *sstd1ptr++;
+ nmean1ptr++;
+ smean1ptr++;
+ nstd1ptr++;
+ sstd1ptr++;
}
inst->frame_counter++;
} else
diff --git a/src/common_audio/vad/main/source/vad_core.h b/src/common_audio/vad/vad_core.h
similarity index 96%
rename from src/common_audio/vad/main/source/vad_core.h
rename to src/common_audio/vad/vad_core.h
index 544caf5a..cad6ca4 100644
--- a/src/common_audio/vad/main/source/vad_core.h
+++ b/src/common_audio/vad/vad_core.h
@@ -28,11 +28,14 @@
WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
+ // TODO(bjornv): Change to |frame_count|.
WebRtc_Word32 frame_counter;
WebRtc_Word16 over_hang; // Over Hang
WebRtc_Word16 num_of_speech;
+ // TODO(bjornv): Change to |age_vector|.
WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
+ // TODO(bjornv): Change to |median|.
WebRtc_Word16 mean_value[NUM_CHANNELS];
WebRtc_Word16 upper_state[5];
WebRtc_Word16 lower_state[5];
diff --git a/src/common_audio/vad/main/source/vad_defines.h b/src/common_audio/vad/vad_defines.h
similarity index 100%
rename from src/common_audio/vad/main/source/vad_defines.h
rename to src/common_audio/vad/vad_defines.h
diff --git a/src/common_audio/vad/vad_filterbank.c b/src/common_audio/vad/vad_filterbank.c
new file mode 100644
index 0000000..63eef5b
--- /dev/null
+++ b/src/common_audio/vad/vad_filterbank.c
@@ -0,0 +1,278 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file includes the implementation of the internal filterbank associated functions.
+ * For function description, see vad_filterbank.h.
+ */
+
+#include "vad_filterbank.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+
+// Constant 160*log10(2) in Q9
+static const int16_t kLogConst = 24660;
+
+// Coefficients used by WebRtcVad_HpOutput, Q14
+static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
+static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
+
+// Allpass filter coefficients, upper and lower, in Q15
+// Upper: 0.64, Lower: 0.17
+static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
+
+// Adjustment for division with two in WebRtcVad_SplitFilter
+static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
+
+void WebRtcVad_HpOutput(int16_t* in_vector,
+ int in_vector_length,
+ int16_t* filter_state,
+ int16_t* out_vector) {
+ int i;
+ int16_t* in_ptr = in_vector;
+ int16_t* out_ptr = out_vector;
+ int32_t tmp32 = 0;
+
+
+ // The sum of the absolute values of the impulse response:
+ // The zero/pole-filter has a max amplification of a single sample of: 1.4546
+ // Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
+ // The all-zero section has a max amplification of a single sample of: 1.6189
+ // Impulse response: 0.4047 -0.8094 0.4047 0 0 0
+ // The all-pole section has a max amplification of a single sample of: 1.9931
+ // Impulse response: 1.0000 0.4734 -0.1189 -0.2187 -0.0627 0.04532
+
+ for (i = 0; i < in_vector_length; i++) {
+ // all-zero section (filter coefficients in Q14)
+ tmp32 = (int32_t) WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], (*in_ptr));
+ tmp32 += (int32_t) WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
+ tmp32 += (int32_t) WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2],
+ filter_state[1]); // Q14
+ filter_state[1] = filter_state[0];
+ filter_state[0] = *in_ptr++;
+
+ // all-pole section
+ tmp32 -= (int32_t) WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1],
+ filter_state[2]); // Q14
+ tmp32 -= (int32_t) WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
+ filter_state[3] = filter_state[2];
+ filter_state[2] = (int16_t) WEBRTC_SPL_RSHIFT_W32 (tmp32, 14);
+ *out_ptr++ = filter_state[2];
+ }
+}
+
+void WebRtcVad_Allpass(int16_t* in_vector,
+ int16_t filter_coefficients,
+ int vector_length,
+ int16_t* filter_state,
+ int16_t* out_vector) {
+ // The filter can only cause overflow (in the w16 output variable)
+ // if more than 4 consecutive input numbers are of maximum value and
+ // has the the same sign as the impulse responses first taps.
+ // First 6 taps of the impulse response: 0.6399 0.5905 -0.3779
+ // 0.2418 -0.1547 0.0990
+
+ int i;
+ int16_t tmp16 = 0;
+ int32_t tmp32 = 0, in32 = 0;
+ int32_t state32 = WEBRTC_SPL_LSHIFT_W32((int32_t) (*filter_state), 16); // Q31
+
+ for (i = 0; i < vector_length; i++) {
+ tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficients, (*in_vector));
+ tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
+ *out_vector++ = tmp16;
+ in32 = WEBRTC_SPL_LSHIFT_W32(((int32_t) (*in_vector)), 14);
+ state32 = in32 - WEBRTC_SPL_MUL_16_16(filter_coefficients, tmp16);
+ state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1);
+ in_vector += 2;
+ }
+
+ *filter_state = (int16_t) WEBRTC_SPL_RSHIFT_W32(state32, 16);
+}
+
+void WebRtcVad_SplitFilter(int16_t* in_vector,
+ int in_vector_length,
+ int16_t* upper_state,
+ int16_t* lower_state,
+ int16_t* out_vector_hp,
+ int16_t* out_vector_lp) {
+ int16_t tmp_out;
+ int i;
+ int half_length = WEBRTC_SPL_RSHIFT_W16(in_vector_length, 1);
+
+ // All-pass filtering upper branch
+ WebRtcVad_Allpass(&in_vector[0], kAllPassCoefsQ15[0], half_length,
+ upper_state, out_vector_hp);
+
+ // All-pass filtering lower branch
+ WebRtcVad_Allpass(&in_vector[1], kAllPassCoefsQ15[1], half_length,
+ lower_state, out_vector_lp);
+
+ // Make LP and HP signals
+ for (i = 0; i < half_length; i++) {
+ tmp_out = *out_vector_hp;
+ *out_vector_hp++ -= *out_vector_lp;
+ *out_vector_lp++ += tmp_out;
+ }
+}
+
+int16_t WebRtcVad_get_features(VadInstT* inst,
+ int16_t* in_vector,
+ int frame_size,
+ int16_t* out_vector) {
+ int16_t power = 0;
+ // We expect |frame_size| to be 80, 160 or 240 samples, which corresponds to
+ // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
+ // have at most 120 samples after the first split and at most 60 samples after
+ // the second split.
+ int16_t hp_120[120], lp_120[120];
+ int16_t hp_60[60], lp_60[60];
+ // Initialize variables for the first SplitFilter().
+ int length = frame_size;
+ int frequency_band = 0;
+ int16_t* in_ptr = in_vector;
+ int16_t* hp_out_ptr = hp_120;
+ int16_t* lp_out_ptr = lp_120;
+
+ // Split at 2000 Hz and downsample
+ WebRtcVad_SplitFilter(in_ptr, length, &inst->upper_state[frequency_band],
+ &inst->lower_state[frequency_band], hp_out_ptr,
+ lp_out_ptr);
+
+ // Split at 3000 Hz and downsample
+ frequency_band = 1;
+ in_ptr = hp_120;
+ hp_out_ptr = hp_60;
+ lp_out_ptr = lp_60;
+ length = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
+
+ WebRtcVad_SplitFilter(in_ptr, length, &inst->upper_state[frequency_band],
+ &inst->lower_state[frequency_band], hp_out_ptr,
+ lp_out_ptr);
+
+ // Energy in 3000 Hz - 4000 Hz
+ length = WEBRTC_SPL_RSHIFT_W16(length, 1);
+ WebRtcVad_LogOfEnergy(hp_60, length, kOffsetVector[5], &power,
+ &out_vector[5]);
+
+ // Energy in 2000 Hz - 3000 Hz
+ WebRtcVad_LogOfEnergy(lp_60, length, kOffsetVector[4], &power,
+ &out_vector[4]);
+
+ // Split at 1000 Hz and downsample
+ frequency_band = 2;
+ in_ptr = lp_120;
+ hp_out_ptr = hp_60;
+ lp_out_ptr = lp_60;
+ length = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
+ WebRtcVad_SplitFilter(in_ptr, length, &inst->upper_state[frequency_band],
+ &inst->lower_state[frequency_band], hp_out_ptr,
+ lp_out_ptr);
+
+ // Energy in 1000 Hz - 2000 Hz
+ length = WEBRTC_SPL_RSHIFT_W16(length, 1);
+ WebRtcVad_LogOfEnergy(hp_60, length, kOffsetVector[3], &power,
+ &out_vector[3]);
+
+ // Split at 500 Hz
+ frequency_band = 3;
+ in_ptr = lp_60;
+ hp_out_ptr = hp_120;
+ lp_out_ptr = lp_120;
+
+ WebRtcVad_SplitFilter(in_ptr, length, &inst->upper_state[frequency_band],
+ &inst->lower_state[frequency_band], hp_out_ptr,
+ lp_out_ptr);
+
+ // Energy in 500 Hz - 1000 Hz
+ length = WEBRTC_SPL_RSHIFT_W16(length, 1);
+ WebRtcVad_LogOfEnergy(hp_120, length, kOffsetVector[2], &power,
+ &out_vector[2]);
+
+ // Split at 250 Hz
+ frequency_band = 4;
+ in_ptr = lp_120;
+ hp_out_ptr = hp_60;
+ lp_out_ptr = lp_60;
+
+ WebRtcVad_SplitFilter(in_ptr, length, &inst->upper_state[frequency_band],
+ &inst->lower_state[frequency_band], hp_out_ptr,
+ lp_out_ptr);
+
+ // Energy in 250 Hz - 500 Hz
+ length = WEBRTC_SPL_RSHIFT_W16(length, 1);
+ WebRtcVad_LogOfEnergy(hp_60, length, kOffsetVector[1], &power,
+ &out_vector[1]);
+
+ // Remove DC and LFs
+ WebRtcVad_HpOutput(lp_60, length, inst->hp_filter_state, hp_120);
+
+ // Power in 80 Hz - 250 Hz
+ WebRtcVad_LogOfEnergy(hp_120, length, kOffsetVector[0], &power,
+ &out_vector[0]);
+
+ return power;
+}
+
+void WebRtcVad_LogOfEnergy(int16_t* vector,
+ int vector_length,
+ int16_t offset,
+ int16_t* power,
+ int16_t* log_energy) {
+ int shfts = 0, shfts2 = 0;
+ int16_t energy_s16 = 0;
+ int16_t zeros = 0, frac = 0, log2 = 0;
+ int32_t energy = WebRtcSpl_Energy(vector, vector_length, &shfts);
+
+ if (energy > 0) {
+
+ shfts2 = 16 - WebRtcSpl_NormW32(energy);
+ shfts += shfts2;
+ // "shfts" is the total number of right shifts that has been done to
+ // energy_s16.
+ energy_s16 = (int16_t) WEBRTC_SPL_SHIFT_W32(energy, -shfts2);
+
+ // Find:
+ // 160*log10(energy_s16*2^shfts) = 160*log10(2)*log2(energy_s16*2^shfts) =
+ // 160*log10(2)*(log2(energy_s16) + log2(2^shfts)) =
+ // 160*log10(2)*(log2(energy_s16) + shfts)
+
+ zeros = WebRtcSpl_NormU32(energy_s16);
+ frac = (int16_t) (((uint32_t) ((int32_t) (energy_s16) << zeros)
+ & 0x7FFFFFFF) >> 21);
+ log2 = (int16_t) (((31 - zeros) << 10) + frac);
+
+ *log_energy = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLogConst, log2, 19)
+ + (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(shfts, kLogConst, 9);
+
+ if (*log_energy < 0) {
+ *log_energy = 0;
+ }
+ } else {
+ *log_energy = 0;
+ shfts = -15;
+ energy_s16 = 0;
+ }
+
+ *log_energy += offset;
+
+ // Total power in frame
+ if (*power <= MIN_ENERGY) {
+ if (shfts > 0) {
+ *power += MIN_ENERGY + 1;
+ } else if (WEBRTC_SPL_SHIFT_W16(energy_s16, shfts) > MIN_ENERGY) {
+ *power += MIN_ENERGY + 1;
+ } else {
+ *power += WEBRTC_SPL_SHIFT_W16(energy_s16, shfts);
+ }
+ }
+}
diff --git a/src/common_audio/vad/main/source/vad_filterbank.h b/src/common_audio/vad/vad_filterbank.h
similarity index 73%
rename from src/common_audio/vad/main/source/vad_filterbank.h
rename to src/common_audio/vad/vad_filterbank.h
index a5507ea..1285c47 100644
--- a/src/common_audio/vad/main/source/vad_filterbank.h
+++ b/src/common_audio/vad/vad_filterbank.h
@@ -8,17 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-
/*
* This header file includes the description of the internal VAD call
* WebRtcVad_GaussianProbability.
*/
-#ifndef WEBRTC_VAD_FILTERBANK_H_
-#define WEBRTC_VAD_FILTERBANK_H_
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+#include "typedefs.h"
#include "vad_core.h"
+// TODO(bjornv): Move local functions to vad_filterbank.c and make static.
/****************************************************************************
* WebRtcVad_HpOutput(...)
*
@@ -34,10 +35,10 @@
* - filter_state : Updated state of the filter
*
*/
-void WebRtcVad_HpOutput(WebRtc_Word16* in_vector,
- WebRtc_Word16 in_vector_length,
- WebRtc_Word16* out_vector,
- WebRtc_Word16* filter_state);
+void WebRtcVad_HpOutput(int16_t* in_vector,
+ int in_vector_length,
+ int16_t* filter_state,
+ int16_t* out_vector);
/****************************************************************************
* WebRtcVad_Allpass(...)
@@ -58,11 +59,11 @@
* - filter_state : Updated state of the filter (Q(-1))
*
*/
-void WebRtcVad_Allpass(WebRtc_Word16* in_vector,
- WebRtc_Word16* outw16,
- WebRtc_Word16 filter_coefficients,
+void WebRtcVad_Allpass(int16_t* in_vector,
+ int16_t filter_coefficients,
int vector_length,
- WebRtc_Word16* filter_state);
+ int16_t* filter_state,
+ int16_t* outw16);
/****************************************************************************
* WebRtcVad_SplitFilter(...)
@@ -83,12 +84,12 @@
* - lower_state : Updated state of the lower filter
*
*/
-void WebRtcVad_SplitFilter(WebRtc_Word16* in_vector,
- WebRtc_Word16* out_vector_hp,
- WebRtc_Word16* out_vector_lp,
- WebRtc_Word16* upper_state,
- WebRtc_Word16* lower_state,
- int in_vector_length);
+void WebRtcVad_SplitFilter(int16_t* in_vector,
+ int in_vector_length,
+ int16_t* upper_state,
+ int16_t* lower_state,
+ int16_t* out_vector_hp,
+ int16_t* out_vector_lp);
/****************************************************************************
* WebRtcVad_get_features(...)
@@ -113,10 +114,10 @@
* Return: total power in the signal (NOTE! This value is not exact since it
* is only used in a comparison.
*/
-WebRtc_Word16 WebRtcVad_get_features(VadInstT* inst,
- WebRtc_Word16* in_vector,
- int frame_size,
- WebRtc_Word16* out_vector);
+int16_t WebRtcVad_get_features(VadInstT* inst,
+ int16_t* in_vector,
+ int frame_size,
+ int16_t* out_vector);
/****************************************************************************
* WebRtcVad_LogOfEnergy(...)
@@ -129,15 +130,15 @@
* - vector_length : Length of input vector
*
* Output:
- * - enerlogval : 10*log10(energy);
+ * - log_energy : 10*log10(energy);
* - power : Update total power in speech frame. NOTE! This value
* is not exact since it is only used in a comparison.
*
*/
-void WebRtcVad_LogOfEnergy(WebRtc_Word16* vector,
- WebRtc_Word16* enerlogval,
- WebRtc_Word16* power,
- WebRtc_Word16 offset,
- int vector_length);
+void WebRtcVad_LogOfEnergy(int16_t* vector,
+ int vector_length,
+ int16_t offset,
+ int16_t* power,
+ int16_t* log_energy);
-#endif // WEBRTC_VAD_FILTERBANK_H_
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/src/common_audio/vad/vad_gmm.c b/src/common_audio/vad/vad_gmm.c
new file mode 100644
index 0000000..20a703a
--- /dev/null
+++ b/src/common_audio/vad/vad_gmm.c
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_gmm.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+
+static const int32_t kCompVar = 22005;
+static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.
+
+// For a normal distribution, the probability of |input| is calculated and
+// returned (in Q20). The formula for normal distributed probability is
+//
+// 1 / s * exp(-(x - m)^2 / (2 * s^2))
+//
+// where the parameters are given in the following Q domains:
+// m = |mean| (Q7)
+// s = |std| (Q7)
+// x = |input| (Q4)
+// in addition to the probability we output |delta| (in Q11) used when updating
+// the noise/speech model.
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+ int16_t mean,
+ int16_t std,
+ int16_t* delta) {
+ int16_t tmp16, inv_std, inv_std2, exp_value = 0;
+ int32_t tmp32;
+
+ // Calculate |inv_std| = 1 / s, in Q10.
+ // 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
+ // Q-domain: Q17 / Q7 = Q10.
+ tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
+ inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
+
+ // Calculate |inv_std2| = 1 / s^2, in Q14.
+ tmp16 = (inv_std >> 2); // Q10 -> Q8.
+ // Q-domain: (Q8 * Q8) >> 2 = Q14.
+ inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2);
+ // TODO(bjornv): Investigate if changing to
+ // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6);
+ // gives better accuracy.
+
+ tmp16 = (input << 3); // Q4 -> Q7
+ tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
+
+ // To be used later, when updating noise/speech model.
+ // |delta| = (x - m) / s^2, in Q11.
+ // Q-domain: (Q14 * Q7) >> 10 = Q11.
+ *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inv_std2, tmp16, 10);
+
+ // Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
+ // division by two with one shift.
+ // Q-domain: (Q11 * Q7) >> 8 = Q10.
+ tmp32 = WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9);
+
+ // If the exponent is small enough to give a non-zero probability we calculate
+ // |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
+ // ~= exp2(-log2(exp(1)) * |tmp32|).
+ if (tmp32 < kCompVar) {
+ // Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
+ // Q-domain: (Q12 * Q10) >> 12 = Q10.
+ tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLog2Exp, (int16_t) tmp32, 12);
+ tmp16 = -tmp16;
+ exp_value = (0x0400 | (tmp16 & 0x03FF));
+ tmp16 ^= 0xFFFF;
+ tmp16 >>= 10;
+ tmp16 += 1;
+ // Get |exp_value| = exp(-|tmp32|) in Q10.
+ exp_value >>= tmp16;
+ }
+
+ // Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
+ // Q-domain: Q10 * Q10 = Q20.
+ return WEBRTC_SPL_MUL_16_16(inv_std, exp_value);
+}
diff --git a/src/common_audio/vad/vad_gmm.h b/src/common_audio/vad/vad_gmm.h
new file mode 100644
index 0000000..2333af7
--- /dev/null
+++ b/src/common_audio/vad/vad_gmm.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Gaussian probability calculations internally used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+
+#include "typedefs.h"
+
+// Calculates the probability for |input|, given that |input| comes from a
+// normal distribution with mean and standard deviation (|mean|, |std|).
+//
+// Inputs:
+// - input : input sample in Q4.
+// - mean : mean input in the statistical model, Q7.
+// - std : standard deviation, Q7.
+//
+// Output:
+//
+// - delta : input used when updating the model, Q11.
+// |delta| = (|input| - |mean|) / |std|^2.
+//
+// Return:
+// (probability for |input|) =
+// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+ int16_t mean,
+ int16_t std,
+ int16_t* delta);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
diff --git a/src/common_audio/vad/vad_sp.c b/src/common_audio/vad/vad_sp.c
new file mode 100644
index 0000000..4fface3
--- /dev/null
+++ b/src/common_audio/vad/vad_sp.c
@@ -0,0 +1,181 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_sp.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+
+// Allpass filter coefficients, upper and lower, in Q13.
+// Upper: 0.64, Lower: 0.17.
+static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13
+
+// TODO(bjornv): Move this function to vad_filterbank.c.
+// Downsampling filter based on splitting filter and allpass functions.
+void WebRtcVad_Downsampling(int16_t* signal_in,
+ int16_t* signal_out,
+ int32_t* filter_state,
+ int in_length) {
+ int16_t tmp16_1 = 0, tmp16_2 = 0;
+ int32_t tmp32_1 = filter_state[0];
+ int32_t tmp32_2 = filter_state[1];
+ int n = 0;
+ int half_length = (in_length >> 1); // Downsampling by 2 gives half length.
+
+ // Filter coefficients in Q13, filter state in Q0.
+ for (n = 0; n < half_length; n++) {
+ // All-pass filtering upper branch.
+ tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
+ *signal_out = tmp16_1;
+ tmp32_1 = (int32_t) (*signal_in++) -
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
+
+ // All-pass filtering lower branch.
+ tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
+ *signal_out++ += tmp16_2;
+ tmp32_2 = (int32_t) (*signal_in++) -
+ WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
+ }
+ // Store the filter states.
+ filter_state[0] = tmp32_1;
+ filter_state[1] = tmp32_2;
+}
+
+// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
+// smallest values the last 100 frames. Then calculates and returns the median
+// of the five smallest values.
+int16_t WebRtcVad_FindMinimum(VadInstT* self,
+ int16_t feature_value,
+ int channel) {
+ int i = 0, j = 0;
+ int position = -1;
+ // Offset to beginning of the 16 minimum values in memory.
+ int offset = (channel << 4);
+ int16_t current_median = 1600;
+ int16_t alpha = 0;
+ int32_t tmp32 = 0;
+ // Pointer to memory for the 16 minimum values and the age of each value of
+ // the |channel|.
+ int16_t* age_ptr = &self->index_vector[offset];
+ int16_t* value_ptr = &self->low_value_vector[offset];
+ int16_t *p1, *p2, *p3;
+
+ assert(channel < NUM_CHANNELS);
+
+ // Each value in |low_value_vector| is getting 1 loop older.
+ // Update age of each value in |age_ptr|, and remove old values.
+ for (i = 0; i < 16; i++) {
+ p3 = age_ptr + i;
+ if (*p3 != 100) {
+ *p3 += 1;
+ } else {
+ p1 = value_ptr + i + 1;
+ p2 = p3 + 1;
+ for (j = i; j < 16; j++) {
+ *(value_ptr + j) = *p1++;
+ *(age_ptr + j) = *p2++;
+ }
+ *(age_ptr + 15) = 101;
+ *(value_ptr + 15) = 10000;
+ }
+ }
+
+ // Check if |feature_value| is smaller than any of the values in
+ // |low_value_vector|. If so, find the |position| where to insert the new
+ // value.
+ if (feature_value < *(value_ptr + 7)) {
+ if (feature_value < *(value_ptr + 3)) {
+ if (feature_value < *(value_ptr + 1)) {
+ if (feature_value < *value_ptr) {
+ position = 0;
+ } else {
+ position = 1;
+ }
+ } else if (feature_value < *(value_ptr + 2)) {
+ position = 2;
+ } else {
+ position = 3;
+ }
+ } else if (feature_value < *(value_ptr + 5)) {
+ if (feature_value < *(value_ptr + 4)) {
+ position = 4;
+ } else {
+ position = 5;
+ }
+ } else if (feature_value < *(value_ptr + 6)) {
+ position = 6;
+ } else {
+ position = 7;
+ }
+ } else if (feature_value < *(value_ptr + 15)) {
+ if (feature_value < *(value_ptr + 11)) {
+ if (feature_value < *(value_ptr + 9)) {
+ if (feature_value < *(value_ptr + 8)) {
+ position = 8;
+ } else {
+ position = 9;
+ }
+ } else if (feature_value < *(value_ptr + 10)) {
+ position = 10;
+ } else {
+ position = 11;
+ }
+ } else if (feature_value < *(value_ptr + 13)) {
+ if (feature_value < *(value_ptr + 12)) {
+ position = 12;
+ } else {
+ position = 13;
+ }
+ } else if (feature_value < *(value_ptr + 14)) {
+ position = 14;
+ } else {
+ position = 15;
+ }
+ }
+
+ // If we have a new small value, put it in the correct position and shift
+ // larger values up.
+ if (position > -1) {
+ for (i = 15; i > position; i--) {
+ j = i - 1;
+ *(value_ptr + i) = *(value_ptr + j);
+ *(age_ptr + i) = *(age_ptr + j);
+ }
+ *(value_ptr + position) = feature_value;
+ *(age_ptr + position) = 1;
+ }
+
+ // Get |current_median|.
+ if (self->frame_counter > 2) {
+ current_median = *(value_ptr + 2);
+ } else if (self->frame_counter > 0) {
+ current_median = *value_ptr;
+ }
+
+ // Smooth the median value.
+ if (self->frame_counter > 0) {
+ if (current_median < self->mean_value[channel]) {
+ alpha = (int16_t) ALPHA1; // 0.2 in Q15.
+ } else {
+ alpha = (int16_t) ALPHA2; // 0.99 in Q15.
+ }
+ }
+ tmp32 = WEBRTC_SPL_MUL_16_16(alpha + 1, self->mean_value[channel]);
+ tmp32 += WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, current_median);
+ tmp32 += 16384;
+ self->mean_value[channel] = (int16_t) (tmp32 >> 15);
+
+ return self->mean_value[channel];
+}
diff --git a/src/common_audio/vad/vad_sp.h b/src/common_audio/vad/vad_sp.h
new file mode 100644
index 0000000..95c3b4c
--- /dev/null
+++ b/src/common_audio/vad/vad_sp.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This file includes specific signal processing tools used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+
+#include "typedefs.h"
+#include "vad_core.h"
+
+// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
+//
+// Inputs:
+// - signal_in : Input signal.
+// - in_length : Length of input signal in samples.
+//
+// Input & Output:
+// - filter_state : Current filter states of the two all-pass filters. The
+// |filter_state| is updated after all samples have been
+// processed.
+//
+// Output:
+// - signal_out : Downsampled signal (of length |in_length| / 2).
+void WebRtcVad_Downsampling(int16_t* signal_in,
+ int16_t* signal_out,
+ int32_t* filter_state,
+ int in_length);
+
+// Updates and returns the smoothed feature minimum. As minimum we use the
+// median of the five smallest feature values in a 100 frames long window.
+//
+// Inputs:
+// - feature_value : New feature value to update with.
+// - channel : Channel number.
+//
+// Input & Output:
+// - handle : State information of the VAD.
+//
+// Returns:
+// : Smoothed minimum value for a moving window.
+int16_t WebRtcVad_FindMinimum(VadInstT* handle,
+ int16_t feature_value,
+ int channel);
+
+#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
diff --git a/src/common_audio/vad/vad_unittest.cc b/src/common_audio/vad/vad_unittest.cc
new file mode 100644
index 0000000..54a397a
--- /dev/null
+++ b/src/common_audio/vad/vad_unittest.cc
@@ -0,0 +1,234 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h> // size_t
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "webrtc_vad.h"
+
+// TODO(bjornv): Move the internal unit tests to separate files.
+extern "C" {
+#include "vad_core.h"
+#include "vad_gmm.h"
+#include "vad_sp.h"
+}
+
+namespace webrtc {
+namespace {
+const int16_t kModes[] = { 0, 1, 2, 3 };
+const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
+
+// Rates we support.
+const int16_t kRates[] = { 8000, 12000, 16000, 24000, 32000 };
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+// Frame lengths we support.
+const int16_t kMaxFrameLength = 960;
+const int16_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640,
+ kMaxFrameLength };
+const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
+
+// Returns true if the rate and frame length combination is valid.
+bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) {
+ if (rate == 8000) {
+ if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
+ return true;
+ }
+ return false;
+ } else if (rate == 16000) {
+ if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
+ return true;
+ }
+ return false;
+ }
+ if (rate == 32000) {
+ if (frame_length == 320 || frame_length == 640 || frame_length == 960) {
+ return true;
+ }
+ return false;
+ }
+
+ return false;
+}
+
+class VadTest : public ::testing::Test {
+ protected:
+ VadTest();
+ virtual void SetUp();
+ virtual void TearDown();
+};
+
+VadTest::VadTest() {
+}
+
+void VadTest::SetUp() {
+}
+
+void VadTest::TearDown() {
+}
+
+TEST_F(VadTest, ApiTest) {
+ // This API test runs through the APIs for all possible valid and invalid
+ // combinations.
+
+ VadInst* handle = NULL;
+ int16_t zeros[kMaxFrameLength] = { 0 };
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ int16_t speech[kMaxFrameLength];
+ for (int16_t i = 0; i < kMaxFrameLength; i++) {
+ speech[i] = (i * i);
+ }
+
+ // WebRtcVad_get_version() tests
+ char version[32];
+ EXPECT_EQ(-1, WebRtcVad_get_version(NULL, sizeof(version)));
+ EXPECT_EQ(-1, WebRtcVad_get_version(version, 1));
+ EXPECT_EQ(0, WebRtcVad_get_version(version, sizeof(version)));
+
+ // Null instance tests
+ EXPECT_EQ(-1, WebRtcVad_Create(NULL));
+ EXPECT_EQ(-1, WebRtcVad_Init(NULL));
+ EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
+ EXPECT_EQ(-1, WebRtcVad_Free(NULL));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, kModes[0]));
+ EXPECT_EQ(-1, WebRtcVad_Process(NULL, kRates[0], speech, kFrameLengths[0]));
+
+ // WebRtcVad_AssignSize tests
+ int handle_size_bytes = 0;
+ EXPECT_EQ(0, WebRtcVad_AssignSize(&handle_size_bytes));
+ EXPECT_EQ(576, handle_size_bytes);
+
+ // WebRtcVad_Assign tests
+ void* tmp_handle = malloc(handle_size_bytes);
+ EXPECT_EQ(-1, WebRtcVad_Assign(&handle, NULL));
+ EXPECT_EQ(0, WebRtcVad_Assign(&handle, tmp_handle));
+ EXPECT_EQ(handle, tmp_handle);
+ free(tmp_handle);
+
+ // WebRtcVad_Create()
+ ASSERT_EQ(0, WebRtcVad_Create(&handle));
+
+ // Not initialized tests
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], speech, kFrameLengths[0]));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0]));
+
+ // WebRtcVad_Init() test
+ ASSERT_EQ(0, WebRtcVad_Init(handle));
+
+ // WebRtcVad_set_mode() invalid modes tests
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0] - 1));
+ EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[kModesSize - 1] + 1));
+
+ // WebRtcVad_Process() tests
+ // NULL speech pointer
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], NULL, kFrameLengths[0]));
+ // Invalid sampling rate
+ EXPECT_EQ(-1, WebRtcVad_Process(handle, 9999, speech, kFrameLengths[0]));
+ // All zeros as input should work
+ EXPECT_EQ(0, WebRtcVad_Process(handle, kRates[0], zeros, kFrameLengths[0]));
+ for (size_t k = 0; k < kModesSize; k++) {
+ // Test valid modes
+ EXPECT_EQ(0, WebRtcVad_set_mode(handle, kModes[k]));
+ // Loop through sampling rate and frame length combinations
+ for (size_t i = 0; i < kRatesSize; i++) {
+ for (size_t j = 0; j < kFrameLengthsSize; j++) {
+ if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
+ EXPECT_EQ(1, WebRtcVad_Process(handle,
+ kRates[i],
+ speech,
+ kFrameLengths[j]));
+ } else {
+ EXPECT_EQ(-1, WebRtcVad_Process(handle,
+ kRates[i],
+ speech,
+ kFrameLengths[j]));
+ }
+ }
+ }
+ }
+
+ EXPECT_EQ(0, WebRtcVad_Free(handle));
+}
+
+TEST_F(VadTest, GMMTests) {
+ int16_t delta = 0;
+ // Input value at mean.
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(0, 0, 128, &delta));
+ EXPECT_EQ(0, delta);
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(16, 128, 128, &delta));
+ EXPECT_EQ(0, delta);
+ EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(-16, -128, 128, &delta));
+ EXPECT_EQ(0, delta);
+
+ // Largest possible input to give non-zero probability.
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(59, 0, 128, &delta));
+ EXPECT_EQ(7552, delta);
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(75, 128, 128, &delta));
+ EXPECT_EQ(7552, delta);
+ EXPECT_EQ(1024, WebRtcVad_GaussianProbability(-75, -128, 128, &delta));
+ EXPECT_EQ(-7552, delta);
+
+ // Too large input, should give zero probability.
+ EXPECT_EQ(0, WebRtcVad_GaussianProbability(105, 0, 128, &delta));
+ EXPECT_EQ(13440, delta);
+}
+
+TEST_F(VadTest, SPTests) {
+ VadInstT* handle = (VadInstT*) malloc(sizeof(VadInstT));
+ int16_t zeros[kMaxFrameLength] = { 0 };
+ int32_t state[2] = { 0 };
+ int16_t data_in[kMaxFrameLength];
+ int16_t data_out[kMaxFrameLength];
+
+ const int16_t kReferenceMin[32] = {
+ 1600, 720, 509, 512, 532, 552, 570, 588,
+ 606, 624, 642, 659, 675, 691, 707, 723,
+ 1600, 544, 502, 522, 542, 561, 579, 597,
+ 615, 633, 651, 667, 683, 699, 715, 731
+ };
+
+ // Construct a speech signal that will trigger the VAD in all modes. It is
+ // known that (i * i) will wrap around, but that doesn't matter in this case.
+ for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+ data_in[i] = (i * i);
+ }
+ // Input values all zeros, expect all zeros out.
+ WebRtcVad_Downsampling(zeros, data_out, state, (int) kMaxFrameLength);
+ EXPECT_EQ(0, state[0]);
+ EXPECT_EQ(0, state[1]);
+ for (int16_t i = 0; i < kMaxFrameLength / 2; ++i) {
+ EXPECT_EQ(0, data_out[i]);
+ }
+ // Make a simple non-zero data test.
+ WebRtcVad_Downsampling(data_in, data_out, state, (int) kMaxFrameLength);
+ EXPECT_EQ(207, state[0]);
+ EXPECT_EQ(2270, state[1]);
+
+ ASSERT_EQ(0, WebRtcVad_InitCore(handle, 0));
+ for (int16_t i = 0; i < 16; ++i) {
+ int16_t value = 500 * (i + 1);
+ for (int j = 0; j < NUM_CHANNELS; ++j) {
+ // Use values both above and below initialized value.
+ EXPECT_EQ(kReferenceMin[i], WebRtcVad_FindMinimum(handle, value, j));
+ EXPECT_EQ(kReferenceMin[i + 16], WebRtcVad_FindMinimum(handle, 12000, j));
+ }
+ handle->frame_counter++;
+ }
+
+ free(handle);
+}
+
+// TODO(bjornv): Add a process test, run on file.
+
+} // namespace
+} // namespace webrtc
diff --git a/src/common_audio/vad/main/source/webrtc_vad.c b/src/common_audio/vad/webrtc_vad.c
similarity index 100%
rename from src/common_audio/vad/main/source/webrtc_vad.c
rename to src/common_audio/vad/webrtc_vad.c
diff --git a/src/common_types.h b/src/common_types.h
index 8b0b8a5..02d712e 100644
--- a/src/common_types.h
+++ b/src/common_types.h
@@ -251,6 +251,8 @@
WebRtc_UWord16 currentBufferSize;
// preferred (optimal) buffer size in ms
WebRtc_UWord16 preferredBufferSize;
+ // adding extra delay due to "peaky jitter"
+ bool jitterPeaksFound;
// loss rate (network + late) in percent (in Q14)
WebRtc_UWord16 currentPacketLossRate;
// late loss rate in percent (in Q14)
@@ -263,58 +265,14 @@
WebRtc_UWord16 currentPreemptiveRate;
// fraction of data removed through acceleration (in Q14)
WebRtc_UWord16 currentAccelerateRate;
-};
-
-struct JitterStatistics
-{
- // smallest Jitter Buffer size during call in ms
- WebRtc_UWord32 jbMinSize;
- // largest Jitter Buffer size during call in ms
- WebRtc_UWord32 jbMaxSize;
- // the average JB size, measured over time - ms
- WebRtc_UWord32 jbAvgSize;
- // number of times the Jitter Buffer changed (using Accelerate or
- // Pre-emptive Expand)
- WebRtc_UWord32 jbChangeCount;
- // amount (in ms) of audio data received late
- WebRtc_UWord32 lateLossMs;
- // milliseconds removed to reduce jitter buffer size
- WebRtc_UWord32 accelerateMs;
- // milliseconds discarded through buffer flushing
- WebRtc_UWord32 flushedMs;
- // milliseconds of generated silence
- WebRtc_UWord32 generatedSilentMs;
- // milliseconds of synthetic audio data (non-background noise)
- WebRtc_UWord32 interpolatedVoiceMs;
- // milliseconds of synthetic audio data (background noise level)
- WebRtc_UWord32 interpolatedSilentMs;
- // count of tiny expansions in output audio
- WebRtc_UWord32 countExpandMoreThan120ms;
- // count of small expansions in output audio
- WebRtc_UWord32 countExpandMoreThan250ms;
- // count of medium expansions in output audio
- WebRtc_UWord32 countExpandMoreThan500ms;
- // count of long expansions in output audio
- WebRtc_UWord32 countExpandMoreThan2000ms;
- // duration of longest audio drop-out
- WebRtc_UWord32 longestExpandDurationMs;
- // count of times we got small network outage (inter-arrival time in
- // [500, 1000) ms)
- WebRtc_UWord32 countIAT500ms;
- // count of times we got medium network outage (inter-arrival time in
- // [1000, 2000) ms)
- WebRtc_UWord32 countIAT1000ms;
- // count of times we got large network outage (inter-arrival time >=
- // 2000 ms)
- WebRtc_UWord32 countIAT2000ms;
- // longest packet inter-arrival time in ms
- WebRtc_UWord32 longestIATms;
- // min time incoming Packet "waited" to be played
- WebRtc_UWord32 minPacketDelayMs;
- // max time incoming Packet "waited" to be played
- WebRtc_UWord32 maxPacketDelayMs;
- // avg time incoming Packet "waited" to be played
- WebRtc_UWord32 avgPacketDelayMs;
+ // clock-drift in parts-per-million (negative or positive)
+ int32_t clockDriftPPM;
+ // average packet waiting time in the jitter buffer (ms)
+ int meanWaitingTimeMs;
+ // median packet waiting time in the jitter buffer (ms)
+ int medianWaitingTimeMs;
+ // max packet waiting time in the jitter buffer (ms)
+ int maxWaitingTimeMs;
};
typedef struct
@@ -479,12 +437,15 @@
kVideoMJPEG = 10,
kVideoNV12 = 11,
kVideoNV21 = 12,
+ kVideoBGRA = 13,
kVideoUnknown = 99
};
// Video codec
enum { kConfigParameterSize = 128};
enum { kPayloadNameSize = 32};
+enum { kMaxSimulcastStreams = 4};
+enum { kMaxTemporalStreams = 4};
// H.263 specific
struct VideoCodecH263
@@ -513,6 +474,17 @@
kProfileMain = 0x01
};
+enum VP8ResilienceMode {
+ kResilienceOff, // The stream produced by the encoder requires a
+ // recovery frame (typically a key frame) to be
+ // decodable after a packet loss.
+ kResilientStream, // A stream produced by the encoder is resilient to
+ // packet losses, but packets within a frame subsequent
+ // to a loss can't be decoded.
+ kResilientFrames // Same as kResilientStream but with added resilience
+ // within a frame.
+};
+
struct VideoCodecH264
{
H264Packetization packetization;
@@ -530,9 +502,11 @@
// VP8 specific
struct VideoCodecVP8
{
- bool pictureLossIndicationOn;
- bool feedbackModeOn;
- VideoCodecComplexity complexity;
+ bool pictureLossIndicationOn;
+ bool feedbackModeOn;
+ VideoCodecComplexity complexity;
+ VP8ResilienceMode resilience;
+ unsigned char numberOfTemporalLayers;
};
// MPEG-4 specific
@@ -570,6 +544,19 @@
VideoCodecGeneric Generic;
};
+/*
+* Simulcast is when the same stream is encoded multiple times with different
+* settings such as resolution.
+*/
+struct SimulcastStream
+{
+ unsigned short width;
+ unsigned short height;
+ unsigned char numberOfTemporalLayers;
+ unsigned int maxBitrate;
+ unsigned int qpMax; // minimum quality
+};
+
// Common video codec properties
struct VideoCodec
{
@@ -588,8 +575,8 @@
VideoCodecUnion codecSpecific;
unsigned int qpMax;
+ unsigned char numberOfSimulcastStreams;
+ SimulcastStream simulcastStream[kMaxSimulcastStreams];
};
-
} // namespace webrtc
-
#endif // WEBRTC_COMMON_TYPES_H
diff --git a/src/modules/audio_processing/Android.mk b/src/modules/audio_processing/Android.mk
new file mode 100644
index 0000000..9ca2aee
--- /dev/null
+++ b/src/modules/audio_processing/Android.mk
@@ -0,0 +1,143 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_apm
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := \
+ $(call all-proto-files-under, .) \
+ audio_buffer.cc \
+ audio_processing_impl.cc \
+ echo_cancellation_impl.cc \
+ echo_control_mobile_impl.cc \
+ gain_control_impl.cc \
+ high_pass_filter_impl.cc \
+ level_estimator_impl.cc \
+ noise_suppression_impl.cc \
+ splitting_filter.cc \
+ processing_component.cc \
+ voice_detection_impl.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_NS_FIXED'
+# floating point
+# -DWEBRTC_NS_FLOAT'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/aec/interface \
+ $(LOCAL_PATH)/aecm/interface \
+ $(LOCAL_PATH)/agc/interface \
+ $(LOCAL_PATH)/ns/interface \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../common_audio/vad/include \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ external/protobuf/src
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+# apm process test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ $(call all-proto-files-under, .) \
+ test/process_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ external/gtest/include
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libutils \
+ libstlport \
+ libwebrtc_audio_preprocessing
+
+LOCAL_MODULE:= webrtc_apm_process_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
+
+# apm unit test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ $(call all-proto-files-under, test) \
+ test/unit_test.cc \
+ ../../../test/testsupport/fileutils.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ '-DWEBRTC_APM_UNIT_TEST_FIXED_PROFILE'
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../../../test \
+ $(LOCAL_PATH)/../../system_wrappers/interface \
+ $(LOCAL_PATH)/../../common_audio/signal_processing/include \
+ external/gtest/include \
+ external/protobuf/src
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest \
+ libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libwebrtc_audio_preprocessing
+
+LOCAL_MODULE:= webrtc_apm_unit_test
+
+ifdef NDK_ROOT
+include $(BUILD_EXECUTABLE)
+else
+include external/stlport/libstlport.mk
+include $(BUILD_NATIVE_TEST)
+endif
diff --git a/src/modules/audio_processing/OWNERS b/src/modules/audio_processing/OWNERS
index aecf56e..5a25634 100644
--- a/src/modules/audio_processing/OWNERS
+++ b/src/modules/audio_processing/OWNERS
@@ -1,2 +1,2 @@
-ajm@google.com
-bjornv@google.com
+andrew@webrtc.org
+bjornv@webrtc.org
diff --git a/src/modules/audio_processing/aec/Android.mk b/src/modules/audio_processing/aec/Android.mk
new file mode 100644
index 0000000..698755a
--- /dev/null
+++ b/src/modules/audio_processing/aec/Android.mk
@@ -0,0 +1,45 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aec
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ echo_cancellation.c \
+ aec_resampler.c \
+ aec_core.c \
+ aec_rdft.c \
+ aec_core_sse2.c \
+ aec_rdft_sse2.c
+
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/aec/main/source/aec.gyp b/src/modules/audio_processing/aec/aec.gypi
similarity index 63%
rename from src/modules/audio_processing/aec/main/source/aec.gyp
rename to src/modules/audio_processing/aec/aec.gypi
index 0427e00..7e86a90 100644
--- a/src/modules/audio_processing/aec/main/source/aec.gyp
+++ b/src/modules/audio_processing/aec/aec.gypi
@@ -7,43 +7,43 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../../common_settings.gypi',
- ],
'targets': [
{
'target_name': 'aec',
'type': '<(library)',
+ 'variables': {
+ # Outputs some low-level debug files.
+ 'aec_debug_dump%': 0,
+ },
'dependencies': [
- '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
- '../../../utility/util.gyp:apm_util'
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ 'apm_util'
],
'include_dirs': [
- '../interface',
+ 'interface',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'interface',
],
},
'sources': [
- '../interface/echo_cancellation.h',
+ 'interface/echo_cancellation.h',
'echo_cancellation.c',
+ 'aec_core.h',
'aec_core.c',
'aec_core_sse2.c',
'aec_rdft.h',
'aec_rdft.c',
'aec_rdft_sse2.c',
- 'aec_core.h',
- 'resampler.c',
- 'resampler.h',
+ 'aec_resampler.h',
+ 'aec_resampler.c',
+ ],
+ 'conditions': [
+ ['aec_debug_dump==1', {
+ 'defines': [ 'WEBRTC_AEC_DEBUG_DUMP', ],
+ }],
],
},
],
}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/aec/main/source/aec_core.c b/src/modules/audio_processing/aec/aec_core.c
similarity index 67%
rename from src/modules/audio_processing/aec/main/source/aec_core.c
rename to src/modules/audio_processing/aec/aec_core.c
index 81197ea..6718dec 100644
--- a/src/modules/audio_processing/aec/main/source/aec_core.c
+++ b/src/modules/audio_processing/aec/aec_core.c
@@ -12,14 +12,22 @@
* The core AEC algorithm, which is presented with time-aligned signals.
*/
+#include "aec_core.h"
+
+#include <assert.h>
#include <math.h>
+#include <stddef.h> // size_t
#include <stdlib.h>
#include <string.h>
-#include "aec_core.h"
#include "aec_rdft.h"
+#include "delay_estimator_wrapper.h"
#include "ring_buffer.h"
#include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "typedefs.h"
+
+// Buffer size (samples)
+static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
// Noise suppression
static const int converged = 250;
@@ -34,26 +42,9 @@
// Initial bin for averaging nlp gain in low band
static const int freqAvgIc = PART_LEN / 2;
-/* Matlab code to produce table:
-win = sqrt(hanning(63)); win = [0 ; win(1:32)];
-fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
-*/
-/*
-static const float sqrtHanning[33] = {
- 0.00000000000000, 0.04906767432742, 0.09801714032956,
- 0.14673047445536, 0.19509032201613, 0.24298017990326,
- 0.29028467725446, 0.33688985339222, 0.38268343236509,
- 0.42755509343028, 0.47139673682600, 0.51410274419322,
- 0.55557023301960, 0.59569930449243, 0.63439328416365,
- 0.67155895484702, 0.70710678118655, 0.74095112535496,
- 0.77301045336274, 0.80320753148064, 0.83146961230255,
- 0.85772861000027, 0.88192126434835, 0.90398929312344,
- 0.92387953251129, 0.94154406518302, 0.95694033573221,
- 0.97003125319454, 0.98078528040323, 0.98917650996478,
- 0.99518472667220, 0.99879545620517, 1.00000000000000
-};
-*/
-
+// Matlab code to produce table:
+// win = sqrt(hanning(63)); win = [0 ; win(1:32)];
+// fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
static const float sqrtHanning[65] = {
0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
@@ -79,10 +70,9 @@
0.99969881869620f, 1.00000000000000f
};
-/* Matlab code to produce table:
-weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
-fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
-*/
+// Matlab code to produce table:
+// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
const float WebRtcAec_weightCurve[65] = {
0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f,
0.1845f, 0.1926f, 0.2000f, 0.2069f, 0.2134f, 0.2195f,
@@ -97,10 +87,9 @@
0.3903f, 0.3928f, 0.3952f, 0.3976f, 0.4000f
};
-/* Matlab code to produce table:
-overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
-fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
-*/
+// Matlab code to produce table:
+// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
const float WebRtcAec_overDriveCurve[65] = {
1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f,
1.3062f, 1.3307f, 1.3536f, 1.3750f, 1.3953f, 1.4146f,
@@ -116,12 +105,7 @@
};
// "Private" function prototypes.
-static void ProcessBlock(aec_t *aec, const short *farend,
- const short *nearend, const short *nearendH,
- short *out, short *outH);
-
-static void BufferFar(aec_t *aec, const short *farend, int farLen);
-static void FetchFar(aec_t *aec, short *farend, int farLen, int knownDelay);
+static void ProcessBlock(aec_t* aec);
static void NonLinearProcessing(aec_t *aec, short *output, short *outputH);
@@ -134,8 +118,13 @@
static void WebRtcAec_InitLevel(power_level_t *level);
static void WebRtcAec_InitStats(stats_t *stats);
-static void UpdateLevel(power_level_t *level, const short *in);
+static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]);
static void UpdateMetrics(aec_t *aec);
+// Convert from time domain to frequency domain. Note that |time_data| are
+// overwritten.
+static void TimeToFrequency(float time_data[PART_LEN2],
+ float freq_data[2][PART_LEN1],
+ int window);
__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
{
@@ -163,35 +152,67 @@
return -1;
}
- if (WebRtcApm_CreateBuffer(&aec->farFrBuf, FRAME_LEN + PART_LEN) == -1) {
+ if (WebRtc_CreateBuffer(&aec->nearFrBuf,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtcApm_CreateBuffer(&aec->nearFrBuf, FRAME_LEN + PART_LEN) == -1) {
+ if (WebRtc_CreateBuffer(&aec->outFrBuf,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtcApm_CreateBuffer(&aec->outFrBuf, FRAME_LEN + PART_LEN) == -1) {
+ if (WebRtc_CreateBuffer(&aec->nearFrBufH,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtcApm_CreateBuffer(&aec->nearFrBufH, FRAME_LEN + PART_LEN) == -1) {
+ if (WebRtc_CreateBuffer(&aec->outFrBufH,
+ FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
- if (WebRtcApm_CreateBuffer(&aec->outFrBufH, FRAME_LEN + PART_LEN) == -1) {
+ // Create far-end buffers.
+ if (WebRtc_CreateBuffer(&aec->far_buf, kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1) == -1) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}
+ if (WebRtc_CreateBuffer(&aec->far_buf_windowed, kBufSizePartitions,
+ sizeof(float) * 2 * PART_LEN1) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_CreateBuffer(&aec->far_time_buf, kBufSizePartitions,
+ sizeof(int16_t) * PART_LEN) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+#endif
+ if (WebRtc_CreateDelayEstimator(&aec->delay_estimator,
+ PART_LEN1,
+ kMaxDelayBlocks,
+ kLookaheadBlocks) == -1) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
return 0;
}
@@ -202,12 +223,18 @@
return -1;
}
- WebRtcApm_FreeBuffer(aec->farFrBuf);
- WebRtcApm_FreeBuffer(aec->nearFrBuf);
- WebRtcApm_FreeBuffer(aec->outFrBuf);
+ WebRtc_FreeBuffer(aec->nearFrBuf);
+ WebRtc_FreeBuffer(aec->outFrBuf);
- WebRtcApm_FreeBuffer(aec->nearFrBufH);
- WebRtcApm_FreeBuffer(aec->outFrBufH);
+ WebRtc_FreeBuffer(aec->nearFrBufH);
+ WebRtc_FreeBuffer(aec->outFrBufH);
+
+ WebRtc_FreeBuffer(aec->far_buf);
+ WebRtc_FreeBuffer(aec->far_buf_windowed);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_FreeBuffer(aec->far_time_buf);
+#endif
+ WebRtc_FreeDelayEstimator(aec->delay_estimator);
free(aec);
return 0;
@@ -255,6 +282,32 @@
}
}
+// Time-unconstrined filter adaptation.
+// TODO(andrew): consider for a low-complexity mode.
+//static void FilterAdaptationUnconstrained(aec_t *aec, float *fft,
+// float ef[2][PART_LEN1]) {
+// int i, j;
+// for (i = 0; i < NR_PART; i++) {
+// int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+// int pos;
+// // Check for wrap
+// if (i + aec->xfBufBlockPos >= NR_PART) {
+// xPos -= NR_PART * PART_LEN1;
+// }
+//
+// pos = i * PART_LEN1;
+//
+// for (j = 0; j < PART_LEN1; j++) {
+// aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
+// aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
+// }
+// }
+//}
+
static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
int i, j;
for (i = 0; i < NR_PART; i++) {
@@ -267,16 +320,6 @@
pos = i * PART_LEN1;
-#ifdef UNCONSTR
- for (j = 0; j < PART_LEN1; j++) {
- aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
- -aec->xfBuf[xPos + j][1],
- ef[j][0], ef[j][1]);
- aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
- -aec->xfBuf[xPos + j][1],
- ef[j][0], ef[j][1]);
- }
-#else
for (j = 0; j < PART_LEN; j++) {
fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
@@ -309,7 +352,6 @@
aec->wfBuf[0][pos + j] += fft[2 * j];
aec->wfBuf[1][pos + j] += fft[2 * j + 1];
}
-#endif // UNCONSTR
}
}
@@ -355,25 +397,41 @@
aec->errThresh = 1.5e-6f;
}
- if (WebRtcApm_InitBuffer(aec->farFrBuf) == -1) {
+ if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
return -1;
}
- if (WebRtcApm_InitBuffer(aec->nearFrBuf) == -1) {
+ if (WebRtc_InitBuffer(aec->outFrBuf) == -1) {
return -1;
}
- if (WebRtcApm_InitBuffer(aec->outFrBuf) == -1) {
+ if (WebRtc_InitBuffer(aec->nearFrBufH) == -1) {
return -1;
}
- if (WebRtcApm_InitBuffer(aec->nearFrBufH) == -1) {
+ if (WebRtc_InitBuffer(aec->outFrBufH) == -1) {
return -1;
}
- if (WebRtcApm_InitBuffer(aec->outFrBufH) == -1) {
+ // Initialize far-end buffers.
+ if (WebRtc_InitBuffer(aec->far_buf) == -1) {
return -1;
}
+ if (WebRtc_InitBuffer(aec->far_buf_windowed) == -1) {
+ return -1;
+ }
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_InitBuffer(aec->far_time_buf) == -1) {
+ return -1;
+ }
+#endif
+ aec->system_delay = 0;
+
+ if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
+ return -1;
+ }
+ aec->delay_logging_enabled = 0;
+ memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
// Default target suppression level
aec->targetSupp = -11.5;
@@ -396,8 +454,6 @@
aec->knownDelay = 0;
// Initialize buffers
- memset(aec->farBuf, 0, sizeof(aec->farBuf));
- memset(aec->xBuf, 0, sizeof(aec->xBuf));
memset(aec->dBuf, 0, sizeof(aec->dBuf));
memset(aec->eBuf, 0, sizeof(aec->eBuf));
// For H band
@@ -451,13 +507,6 @@
aec->seed = 777;
aec->delayEstCtr = 0;
- // Features on by default (G.167)
-#ifdef G167
- aec->adaptToggle = 1;
- aec->nlpToggle = 1;
- aec->cnToggle = 1;
-#endif
-
// Metrics disabled by default
aec->metricsMode = 0;
WebRtcAec_InitMetrics(aec);
@@ -468,7 +517,7 @@
WebRtcAec_FilterAdaptation = FilterAdaptation;
WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
+#if defined(WEBRTC_USE_SSE2)
WebRtcAec_InitAec_SSE2();
#endif
}
@@ -492,86 +541,108 @@
}
-void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
- const short *nearend, const short *nearendH,
- short *out, short *outH,
- int knownDelay)
-{
- short farBl[PART_LEN], nearBl[PART_LEN], outBl[PART_LEN];
- short farFr[FRAME_LEN];
- // For H band
- short nearBlH[PART_LEN], outBlH[PART_LEN];
+void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend) {
+ float fft[PART_LEN2];
+ float xf[2][PART_LEN1];
- int size = 0;
+ // Check if the buffer is full, and in that case flush the oldest data.
+ if (WebRtc_available_write(aec->far_buf) < 1) {
+ WebRtc_MoveReadPtr(aec->far_buf, 1);
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, 1);
+ aec->system_delay -= PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, 1);
+#endif
+ }
+ // Convert far-end partition to the frequency domain without windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 0);
+ WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
- // initialize: only used for SWB
- memset(nearBlH, 0, sizeof(nearBlH));
- memset(outBlH, 0, sizeof(outBlH));
-
- // Buffer the current frame.
- // Fetch an older one corresponding to the delay.
- BufferFar(aec, farend, FRAME_LEN);
- FetchFar(aec, farFr, FRAME_LEN, knownDelay);
-
- // Buffer the synchronized far and near frames,
- // to pass the smaller blocks individually.
- WebRtcApm_WriteBuffer(aec->farFrBuf, farFr, FRAME_LEN);
- WebRtcApm_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtcApm_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
- }
-
- // Process as many blocks as possible.
- while (WebRtcApm_get_buffer_size(aec->farFrBuf) >= PART_LEN) {
-
- WebRtcApm_ReadBuffer(aec->farFrBuf, farBl, PART_LEN);
- WebRtcApm_ReadBuffer(aec->nearFrBuf, nearBl, PART_LEN);
-
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtcApm_ReadBuffer(aec->nearFrBufH, nearBlH, PART_LEN);
- }
-
- ProcessBlock(aec, farBl, nearBl, nearBlH, outBl, outBlH);
-
- WebRtcApm_WriteBuffer(aec->outFrBuf, outBl, PART_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtcApm_WriteBuffer(aec->outFrBufH, outBlH, PART_LEN);
- }
- }
-
- // Stuff the out buffer if we have less than a frame to output.
- // This should only happen for the first frame.
- size = WebRtcApm_get_buffer_size(aec->outFrBuf);
- if (size < FRAME_LEN) {
- WebRtcApm_StuffBuffer(aec->outFrBuf, FRAME_LEN - size);
- if (aec->sampFreq == 32000) {
- WebRtcApm_StuffBuffer(aec->outFrBufH, FRAME_LEN - size);
- }
- }
-
- // Obtain an output frame.
- WebRtcApm_ReadBuffer(aec->outFrBuf, out, FRAME_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtcApm_ReadBuffer(aec->outFrBufH, outH, FRAME_LEN);
- }
+ // Convert far-end partition to the frequency domain with windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 1);
+ WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
}
-static void ProcessBlock(aec_t *aec, const short *farend,
- const short *nearend, const short *nearendH,
- short *output, short *outputH)
+void WebRtcAec_ProcessFrame(aec_t *aec,
+ const short *nearend,
+ const short *nearendH,
+ int knownDelay)
{
+ // For each frame the process is as follows:
+ // 1) If the system_delay indicates on being too small for processing a
+ // frame we stuff the buffer with enough data for 10 ms.
+ // 2) Adjust the buffer to the system delay, by moving the read pointer.
+ // 3) If we can't move read pointer due to buffer size limitations we
+ // flush/stuff the buffer.
+ // 4) Process as many partitions as possible.
+ // 5) Update the |system_delay| with respect to a full frame of FRAME_LEN
+ // samples. Even though we will have data left to process (we work with
+ // partitions) we consider updating a whole frame, since that's the
+ // amount of data we input and output in audio_processing.
+
+ // TODO(bjornv): Investigate how we should round the delay difference; right
+ // now we know that incoming |knownDelay| is underestimated when it's less
+ // than |aec->knownDelay|. We therefore, round (-32) in that direction. In
+ // the other direction, we don't have this situation, but might flush one
+ // partition too little. This can cause non-causality, which should be
+ // investigated. Maybe, allow for a non-symmetric rounding, like -16.
+ int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
+ int moved_elements = 0;
+
+ // TODO(bjornv): Change the near-end buffer handling to be the same as for
+ // far-end, that is, with a near_pre_buf.
+ // Buffer the near-end frame.
+ WebRtc_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
+ // For H band
+ if (aec->sampFreq == 32000) {
+ WebRtc_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
+ }
+
+ // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
+ // have enough far-end data for that by stuffing the buffer if the
+ // |system_delay| indicates others.
+ if (aec->system_delay < FRAME_LEN) {
+ // We don't have enough data so we rewind 10 ms.
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, -(aec->mult + 1));
+ aec->system_delay -= WebRtc_MoveReadPtr(aec->far_buf, -(aec->mult + 1)) *
+ PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, -(aec->mult + 1));
+#endif
+ }
+
+ // 2) Compensate for a possible change in the system delay.
+
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
+ moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
+ aec->knownDelay -= moved_elements * PART_LEN;
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+#endif
+
+ // 4) Process as many blocks as possible.
+ while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
+ ProcessBlock(aec);
+ }
+
+ // 5) Update system delay with respect to the entire frame.
+ aec->system_delay -= FRAME_LEN;
+}
+
+static void ProcessBlock(aec_t* aec) {
int i;
float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
- short eInt16[PART_LEN];
float scale;
float fft[PART_LEN2];
float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
- complex_t df[PART_LEN1];
+ float df[2][PART_LEN1];
+ float far_spectrum = 0.0f;
+ float near_spectrum = 0.0f;
+ float abs_far_spectrum[PART_LEN1];
+ float abs_near_spectrum[PART_LEN1];
const float gPow[2] = {0.9f, 0.1f};
@@ -581,66 +652,64 @@
const float ramp = 1.0002f;
const float gInitNoise[2] = {0.999f, 0.001f};
-#ifdef AEC_DEBUG
- fwrite(farend, sizeof(short), PART_LEN, aec->farFile);
- fwrite(nearend, sizeof(short), PART_LEN, aec->nearFile);
-#endif
+ int16_t nearend[PART_LEN];
+ int16_t* nearend_ptr = NULL;
+ int16_t output[PART_LEN];
+ int16_t outputH[PART_LEN];
+
+ float* xf_ptr = NULL;
memset(dH, 0, sizeof(dH));
+ if (aec->sampFreq == 32000) {
+ // Get the upper band first so we can reuse |nearend|.
+ WebRtc_ReadBuffer(aec->nearFrBufH,
+ (void**) &nearend_ptr,
+ nearend,
+ PART_LEN);
+ for (i = 0; i < PART_LEN; i++) {
+ dH[i] = (float) (nearend_ptr[i]);
+ }
+ memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
+ }
+ WebRtc_ReadBuffer(aec->nearFrBuf, (void**) &nearend_ptr, nearend, PART_LEN);
// ---------- Ooura fft ----------
- // Concatenate old and new farend blocks.
+ // Concatenate old and new nearend blocks.
for (i = 0; i < PART_LEN; i++) {
- aec->xBuf[i + PART_LEN] = (float)farend[i];
- d[i] = (float)nearend[i];
+ d[i] = (float) (nearend_ptr[i]);
}
-
- if (aec->sampFreq == 32000) {
- for (i = 0; i < PART_LEN; i++) {
- dH[i] = (float)nearendH[i];
- }
- }
-
-
- memcpy(fft, aec->xBuf, sizeof(float) * PART_LEN2);
memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t farend[PART_LEN];
+ int16_t* farend_ptr = NULL;
+ WebRtc_ReadBuffer(aec->far_time_buf, (void**) &farend_ptr, farend, 1);
+ fwrite(farend_ptr, sizeof(int16_t), PART_LEN, aec->farFile);
+ fwrite(nearend_ptr, sizeof(int16_t), PART_LEN, aec->nearFile);
}
+#endif
- aec_rdft_forward_128(fft);
-
- // Far fft
- xf[1][0] = 0;
- xf[1][PART_LEN] = 0;
- xf[0][0] = fft[0];
- xf[0][PART_LEN] = fft[1];
-
- for (i = 1; i < PART_LEN; i++) {
- xf[0][i] = fft[2 * i];
- xf[1][i] = fft[2 * i + 1];
- }
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf) > 0);
+ WebRtc_ReadBuffer(aec->far_buf, (void**) &xf_ptr, &xf[0][0], 1);
// Near fft
memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
- aec_rdft_forward_128(fft);
- df[0][1] = 0;
- df[PART_LEN][1] = 0;
- df[0][0] = fft[0];
- df[PART_LEN][0] = fft[1];
-
- for (i = 1; i < PART_LEN; i++) {
- df[i][0] = fft[2 * i];
- df[i][1] = fft[2 * i + 1];
- }
+ TimeToFrequency(fft, df, 0);
// Power smoothing
for (i = 0; i < PART_LEN1; i++) {
- aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART *
- (xf[0][i] * xf[0][i] + xf[1][i] * xf[1][i]);
- aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] *
- (df[i][0] * df[i][0] + df[i][1] * df[i][1]);
+ far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
+ (xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
+ aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART * far_spectrum;
+ // Calculate absolute spectra
+ abs_far_spectrum[i] = sqrtf(far_spectrum);
+
+ near_spectrum = df[0][i] * df[0][i] + df[1][i] * df[1][i];
+ aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] * near_spectrum;
+ // Calculate absolute spectra
+ abs_near_spectrum[i] = sqrtf(near_spectrum);
}
// Estimate noise power. Wait until dPow is more stable.
@@ -675,6 +744,19 @@
aec->noisePow = aec->dMinPow;
}
+ // Block wise delay estimation used for logging
+ if (aec->delay_logging_enabled) {
+ int delay_estimate = 0;
+ // Estimate the delay
+ delay_estimate = WebRtc_DelayEstimatorProcessFloat(aec->delay_estimator,
+ abs_far_spectrum,
+ abs_near_spectrum,
+ PART_LEN1);
+ if (delay_estimate >= 0) {
+ // Update delay estimate buffer.
+ aec->delay_histogram[delay_estimate]++;
+ }
+ }
// Update the xfBuf block position.
aec->xfBufBlockPos--;
@@ -683,9 +765,9 @@
}
// Buffer xf
- memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1, xf[0],
+ memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1, xf_ptr,
sizeof(float) * PART_LEN1);
- memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, xf[1],
+ memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, &xf_ptr[PART_LEN1],
sizeof(float) * PART_LEN1);
memset(yf[0], 0, sizeof(float) * (PART_LEN1 * 2));
@@ -715,6 +797,7 @@
memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
memset(fft, 0, sizeof(float) * PART_LEN);
memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
+ // TODO(bjornv): Change to use TimeToFrequency().
aec_rdft_forward_128(fft);
ef[1][0] = 0;
@@ -726,55 +809,49 @@
ef[1][i] = fft[2 * i + 1];
}
+ if (aec->metricsMode == 1) {
+ // Note that the first PART_LEN samples in fft (before transformation) are
+ // zero. Hence, the scaling by two in UpdateLevel() should not be
+ // performed. That scaling is taken care of in UpdateMetrics() instead.
+ UpdateLevel(&aec->linoutlevel, ef);
+ }
+
// Scale error signal inversely with far power.
WebRtcAec_ScaleErrorSignal(aec, ef);
-#ifdef G167
- if (aec->adaptToggle) {
-#endif
- // Filter adaptation
- WebRtcAec_FilterAdaptation(aec, fft, ef);
-#ifdef G167
- }
-#endif
-
+ WebRtcAec_FilterAdaptation(aec, fft, ef);
NonLinearProcessing(aec, output, outputH);
-#if defined(AEC_DEBUG) || defined(G167)
- for (i = 0; i < PART_LEN; i++) {
- eInt16[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
- WEBRTC_SPL_WORD16_MIN);
- }
-#endif
-#ifdef G167
- if (aec->nlpToggle == 0) {
- memcpy(output, eInt16, sizeof(eInt16));
- }
-#endif
-
if (aec->metricsMode == 1) {
- for (i = 0; i < PART_LEN; i++) {
- eInt16[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
- WEBRTC_SPL_WORD16_MIN);
- }
-
// Update power levels and echo metrics
- UpdateLevel(&aec->farlevel, farend);
- UpdateLevel(&aec->nearlevel, nearend);
- UpdateLevel(&aec->linoutlevel, eInt16);
- UpdateLevel(&aec->nlpoutlevel, output);
+ UpdateLevel(&aec->farlevel, (float (*)[PART_LEN1]) xf_ptr);
+ UpdateLevel(&aec->nearlevel, df);
UpdateMetrics(aec);
}
-#ifdef AEC_DEBUG
- fwrite(eInt16, sizeof(short), PART_LEN, aec->outLpFile);
- fwrite(output, sizeof(short), PART_LEN, aec->outFile);
+ // Store the output block.
+ WebRtc_WriteBuffer(aec->outFrBuf, output, PART_LEN);
+ // For H band
+ if (aec->sampFreq == 32000) {
+ WebRtc_WriteBuffer(aec->outFrBufH, outputH, PART_LEN);
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t eInt16[PART_LEN];
+ for (i = 0; i < PART_LEN; i++) {
+ eInt16[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
+ WEBRTC_SPL_WORD16_MIN);
+ }
+
+ fwrite(eInt16, sizeof(int16_t), PART_LEN, aec->outLinearFile);
+ fwrite(output, sizeof(int16_t), PART_LEN, aec->outFile);
+ }
#endif
}
static void NonLinearProcessing(aec_t *aec, short *output, short *outputH)
{
- float efw[2][PART_LEN1], dfw[2][PART_LEN1];
- complex_t xfw[PART_LEN1];
+ float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1];
complex_t comfortNoiseHband[PART_LEN1];
float fft[PART_LEN2];
float scale, dtmp;
@@ -798,10 +875,12 @@
const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
const float *ptrGCoh = gCoh[aec->mult - 1];
- // Filter energey
+ // Filter energy
float wfEnMax = 0, wfEn = 0;
const int delayEstInterval = 10 * aec->mult;
+ float* xfw_ptr = NULL;
+
aec->delayEstCtr++;
if (aec->delayEstCtr == delayEstInterval) {
aec->delayEstCtr = 0;
@@ -832,25 +911,15 @@
}
}
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
// NLP
- // Windowed far fft
- for (i = 0; i < PART_LEN; i++) {
- fft[i] = aec->xBuf[i] * sqrtHanning[i];
- fft[PART_LEN + i] = aec->xBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
- }
- aec_rdft_forward_128(fft);
+ WebRtc_ReadBuffer(aec->far_buf_windowed, (void**) &xfw_ptr, &xfw[0][0], 1);
- xfw[0][1] = 0;
- xfw[PART_LEN][1] = 0;
- xfw[0][0] = fft[0];
- xfw[PART_LEN][0] = fft[1];
- for (i = 1; i < PART_LEN; i++) {
- xfw[i][0] = fft[2 * i];
- xfw[i][1] = fft[2 * i + 1];
- }
-
+ // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
+ // |xfwBuf|.
// Buffer far.
- memcpy(aec->xfwBuf, xfw, sizeof(xfw));
+ memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
// Use delayed far.
memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1, sizeof(xfw));
@@ -897,7 +966,7 @@
// adverse interaction with the algorithm's tuning.
// TODO: investigate further why this is so sensitive.
aec->sx[i] = ptrGCoh[0] * aec->sx[i] + ptrGCoh[1] *
- WEBRTC_SPL_MAX(xfw[i][0] * xfw[i][0] + xfw[i][1] * xfw[i][1], 15);
+ WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i], 15);
aec->sde[i][0] = ptrGCoh[0] * aec->sde[i][0] + ptrGCoh[1] *
(dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
@@ -905,9 +974,9 @@
(dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
aec->sxd[i][0] = ptrGCoh[0] * aec->sxd[i][0] + ptrGCoh[1] *
- (dfw[0][i] * xfw[i][0] + dfw[1][i] * xfw[i][1]);
+ (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
aec->sxd[i][1] = ptrGCoh[0] * aec->sxd[i][1] + ptrGCoh[1] *
- (dfw[0][i] * xfw[i][1] - dfw[1][i] * xfw[i][0]);
+ (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
sdSum += aec->sd[i];
seSum += aec->se[i];
@@ -1036,15 +1105,18 @@
WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
-#ifdef G167
- if (aec->cnToggle) {
- ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
- }
-#else
// Add comfort noise.
ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
-#endif
+ // TODO(bjornv): Investigate how to take the windowing below into account if
+ // needed.
+ if (aec->metricsMode == 1) {
+ // Note that we have a scaling by two in the time domain |eBuf|.
+ // In addition the time domain signal is windowed before transformation,
+ // losing half the energy on the average. We take care of the first
+ // scaling only in UpdateMetrics().
+ UpdateLevel(&aec->nlpoutlevel, efw);
+ }
// Inverse error fft.
fft[0] = efw[0][0];
fft[1] = efw[0][PART_LEN];
@@ -1107,7 +1179,6 @@
}
// Copy the current block to the old position.
- memcpy(aec->xBuf, aec->xBuf + PART_LEN, sizeof(float) * PART_LEN);
memcpy(aec->dBuf, aec->dBuf + PART_LEN, sizeof(float) * PART_LEN);
memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
@@ -1216,60 +1287,6 @@
}
}
-// Buffer the farend to account for knownDelay
-static void BufferFar(aec_t *aec, const short *farend, int farLen)
-{
- int writeLen = farLen, writePos = 0;
-
- // Check if the write position must be wrapped.
- while (aec->farBufWritePos + writeLen > FAR_BUF_LEN) {
-
- // Write to remaining buffer space before wrapping.
- writeLen = FAR_BUF_LEN - aec->farBufWritePos;
- memcpy(aec->farBuf + aec->farBufWritePos, farend + writePos,
- sizeof(short) * writeLen);
- aec->farBufWritePos = 0;
- writePos = writeLen;
- writeLen = farLen - writeLen;
- }
-
- memcpy(aec->farBuf + aec->farBufWritePos, farend + writePos,
- sizeof(short) * writeLen);
- aec->farBufWritePos += writeLen;
-}
-
-static void FetchFar(aec_t *aec, short *farend, int farLen, int knownDelay)
-{
- int readLen = farLen, readPos = 0, delayChange = knownDelay - aec->knownDelay;
-
- aec->farBufReadPos -= delayChange;
-
- // Check if delay forces a read position wrap.
- while(aec->farBufReadPos < 0) {
- aec->farBufReadPos += FAR_BUF_LEN;
- }
- while(aec->farBufReadPos > FAR_BUF_LEN - 1) {
- aec->farBufReadPos -= FAR_BUF_LEN;
- }
-
- aec->knownDelay = knownDelay;
-
- // Check if read position must be wrapped.
- while (aec->farBufReadPos + readLen > FAR_BUF_LEN) {
-
- // Read from remaining buffer space before wrapping.
- readLen = FAR_BUF_LEN - aec->farBufReadPos;
- memcpy(farend + readPos, aec->farBuf + aec->farBufReadPos,
- sizeof(short) * readLen);
- aec->farBufReadPos = 0;
- readPos = readLen;
- readLen = farLen - readLen;
- }
- memcpy(farend + readPos, aec->farBuf + aec->farBufReadPos,
- sizeof(short) * readLen);
- aec->farBufReadPos += readLen;
-}
-
static void WebRtcAec_InitLevel(power_level_t *level)
{
const float bigFloat = 1E17f;
@@ -1296,42 +1313,68 @@
stats->hicounter = 0;
}
-static void UpdateLevel(power_level_t *level, const short *in)
-{
- int k;
+static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]) {
+ // Do the energy calculation in the frequency domain. The FFT is performed on
+ // a segment of PART_LEN2 samples due to overlap, but we only want the energy
+ // of half that data (the last PART_LEN samples). Parseval's relation states
+ // that the energy is preserved according to
+ //
+ // \sum_{n=0}^{N-1} |x(n)|^2 = 1/N * \sum_{n=0}^{N-1} |X(n)|^2
+ // = ENERGY,
+ //
+ // where N = PART_LEN2. Since we are only interested in calculating the energy
+ // for the last PART_LEN samples we approximate by calculating ENERGY and
+ // divide by 2,
+ //
+ // \sum_{n=N/2}^{N-1} |x(n)|^2 ~= ENERGY / 2
+ //
+ // Since we deal with real valued time domain signals we only store frequency
+ // bins [0, PART_LEN], which is what |in| consists of. To calculate ENERGY we
+ // need to add the contribution from the missing part in
+ // [PART_LEN+1, PART_LEN2-1]. These values are, up to a phase shift, identical
+ // with the values in [1, PART_LEN-1], hence multiply those values by 2. This
+ // is the values in the for loop below, but multiplication by 2 and division
+ // by 2 cancel.
- for (k = 0; k < PART_LEN; k++) {
- level->sfrsum += in[k] * in[k];
+ // TODO(bjornv): Investigate reusing energy calculations performed at other
+ // places in the code.
+ int k = 1;
+ // Imaginary parts are zero at end points and left out of the calculation.
+ float energy = (in[0][0] * in[0][0]) / 2;
+ energy += (in[0][PART_LEN] * in[0][PART_LEN]) / 2;
+
+ for (k = 1; k < PART_LEN; k++) {
+ energy += (in[0][k] * in[0][k] + in[1][k] * in[1][k]);
+ }
+ energy /= PART_LEN2;
+
+ level->sfrsum += energy;
+ level->sfrcounter++;
+
+ if (level->sfrcounter > subCountLen) {
+ level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
+ level->sfrsum = 0;
+ level->sfrcounter = 0;
+ if (level->framelevel > 0) {
+ if (level->framelevel < level->minlevel) {
+ level->minlevel = level->framelevel; // New minimum.
+ } else {
+ level->minlevel *= (1 + 0.001f); // Small increase.
+ }
}
- level->sfrcounter++;
-
- if (level->sfrcounter > subCountLen) {
- level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
- level->sfrsum = 0;
- level->sfrcounter = 0;
-
- if (level->framelevel > 0) {
- if (level->framelevel < level->minlevel) {
- level->minlevel = level->framelevel; // New minimum
- } else {
- level->minlevel *= (1 + 0.001f); // Small increase
- }
- }
- level->frcounter++;
- level->frsum += level->framelevel;
-
- if (level->frcounter > countLen) {
- level->averagelevel = level->frsum / countLen;
- level->frsum = 0;
- level->frcounter = 0;
- }
-
+ level->frcounter++;
+ level->frsum += level->framelevel;
+ if (level->frcounter > countLen) {
+ level->averagelevel = level->frsum / countLen;
+ level->frsum = 0;
+ level->frcounter = 0;
}
+ }
}
static void UpdateMetrics(aec_t *aec)
{
- float dtmp, dtmp2, dtmp3;
+ float dtmp, dtmp2;
const float actThresholdNoisy = 8.0f;
const float actThresholdClean = 40.0f;
@@ -1345,7 +1388,7 @@
aec->stateCounter++;
}
- if (aec->farlevel.frcounter == countLen) {
+ if (aec->farlevel.frcounter == 0) {
if (aec->farlevel.minlevel < noisyPower) {
actThreshold = actThresholdClean;
@@ -1391,13 +1434,13 @@
// A_NLP
dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
- aec->linoutlevel.averagelevel + 1e-10f);
+ (2 * aec->linoutlevel.averagelevel) + 1e-10f);
// subtract noise power
- suppressedEcho = aec->linoutlevel.averagelevel - safety * aec->linoutlevel.minlevel;
+ suppressedEcho = 2 * (aec->linoutlevel.averagelevel -
+ safety * aec->linoutlevel.minlevel);
dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
- dtmp3 = 10 * (float)log10(aec->nearlevel.averagelevel / suppressedEcho + 1e-10f);
aec->aNlp.instant = dtmp2;
if (dtmp > aec->aNlp.max) {
@@ -1422,10 +1465,11 @@
// ERLE
// subtract noise power
- suppressedEcho = aec->nlpoutlevel.averagelevel - safety * aec->nlpoutlevel.minlevel;
+ suppressedEcho = 2 * (aec->nlpoutlevel.averagelevel -
+ safety * aec->nlpoutlevel.minlevel);
dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
- aec->nlpoutlevel.averagelevel + 1e-10f);
+ (2 * aec->nlpoutlevel.averagelevel) + 1e-10f);
dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
dtmp = dtmp2;
@@ -1454,3 +1498,27 @@
}
}
+static void TimeToFrequency(float time_data[PART_LEN2],
+ float freq_data[2][PART_LEN1],
+ int window) {
+ int i = 0;
+
+ // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
+ if (window) {
+ for (i = 0; i < PART_LEN; i++) {
+ time_data[i] *= sqrtHanning[i];
+ time_data[PART_LEN + i] *= sqrtHanning[PART_LEN - i];
+ }
+ }
+
+ aec_rdft_forward_128(time_data);
+ // Reorder.
+ freq_data[1][0] = 0;
+ freq_data[1][PART_LEN] = 0;
+ freq_data[0][0] = time_data[0];
+ freq_data[0][PART_LEN] = time_data[1];
+ for (i = 1; i < PART_LEN; i++) {
+ freq_data[0][i] = time_data[2 * i];
+ freq_data[1][i] = time_data[2 * i + 1];
+ }
+}
diff --git a/src/modules/audio_processing/aec/main/source/aec_core.h b/src/modules/audio_processing/aec/aec_core.h
similarity index 76%
rename from src/modules/audio_processing/aec/main/source/aec_core.h
rename to src/modules/audio_processing/aec/aec_core.h
index 3386b92..1b9828a 100644
--- a/src/modules/audio_processing/aec/main/source/aec_core.h
+++ b/src/modules/audio_processing/aec/aec_core.h
@@ -16,24 +16,21 @@
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
#include <stdio.h>
-#include "typedefs.h"
-#include "signal_processing_library.h"
-//#define G167 // for running G167 tests
-//#define UNCONSTR // time-unconstrained filter
-//#define AEC_DEBUG // for recording files
+#include "signal_processing_library.h"
+#include "typedefs.h"
#define FRAME_LEN 80
#define PART_LEN 64 // Length of partition
#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
-#define NR_PART 12 // Number of partitions
-#define FILT_LEN (PART_LEN * NR_PART) // Filter length
-#define FILT_LEN2 (FILT_LEN * 2) // Double filter length
-#define FAR_BUF_LEN (FILT_LEN2 * 2)
+#define NR_PART 12 // Number of partitions in filter.
#define PREF_BAND_SIZE 24
-#define BLOCKL_MAX FRAME_LEN
+// Delay estimator constants, used for logging.
+enum { kMaxDelayBlocks = 60 };
+enum { kLookaheadBlocks = 15 };
+enum { kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks };
typedef float complex_t[2];
// For performance reasons, some arrays of complex numbers are replaced by twice
@@ -76,12 +73,11 @@
int inSamples, outSamples;
int delayEstCtr;
- void *farFrBuf, *nearFrBuf, *outFrBuf;
+ void *nearFrBuf, *outFrBuf;
void *nearFrBufH;
void *outFrBufH;
- float xBuf[PART_LEN2]; // farend
float dBuf[PART_LEN2]; // nearend
float eBuf[PART_LEN2]; // error
@@ -92,21 +88,13 @@
float dMinPow[PART_LEN1];
float dInitMinPow[PART_LEN1];
float *noisePow;
-#ifdef FFTW
- float fftR[PART_LEN2];
- fftw_complex fftC[PART_LEN2];
- fftw_plan fftPlan, ifftPlan;
- fftw_complex xfBuf[NR_PART * PART_LEN1];
- fftw_complex wfBuf[NR_PART * PART_LEN1];
- fftw_complex sde[PART_LEN1];
-#else
float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
complex_t sde[PART_LEN1]; // cross-psd of nearend and error
complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
-#endif
+
float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
float hNs[PART_LEN1];
float hNlFbMin, hNlFbLocalMin;
@@ -122,9 +110,11 @@
int xfBufBlockPos;
- short farBuf[FILT_LEN2 * 2];
+ void* far_buf;
+ void* far_buf_windowed;
+ int system_delay; // Current system delay buffered in AEC.
- short mult; // sampling frequency multiple
+ int mult; // sampling frequency multiple
int sampFreq;
WebRtc_UWord32 seed;
@@ -133,13 +123,6 @@
int noiseEstCtr;
- // Toggles for G.167 testing
-#ifdef G167
- short adaptToggle; // Filter adaptation
- short nlpToggle; // Nonlinear processing
- short cnToggle; // Comfort noise
-#endif
-
power_level_t farlevel;
power_level_t nearlevel;
power_level_t linoutlevel;
@@ -157,11 +140,16 @@
int flag_Hband_cn; //for comfort noise
float cn_scale_Hband; //scale for comfort noise in H band
-#ifdef AEC_DEBUG
+ int delay_histogram[kHistorySizeBlocks];
+ int delay_logging_enabled;
+ void* delay_estimator;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ void* far_time_buf;
FILE *farFile;
FILE *nearFile;
FILE *outFile;
- FILE *outLpFile;
+ FILE *outLinearFile;
#endif
} aec_t;
@@ -169,8 +157,6 @@
extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-#define IP_LEN PART_LEN // this must be at least ceil(2 + sqrt(PART_LEN))
-#define W_LEN PART_LEN
typedef void (*WebRtcAec_FilterAdaptation_t)
(aec_t *aec, float *fft, float ef[2][PART_LEN1]);
extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
@@ -184,10 +170,10 @@
void WebRtcAec_InitAec_SSE2(void);
void WebRtcAec_InitMetrics(aec_t *aec);
-void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
- const short *nearend, const short *nearendH,
- short *out, short *outH,
- int knownDelay);
+void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend);
+void WebRtcAec_ProcessFrame(aec_t* aec,
+ const short *nearend,
+ const short *nearendH,
+ int knownDelay);
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
diff --git a/src/modules/audio_processing/aec/main/source/aec_core_sse2.c b/src/modules/audio_processing/aec/aec_core_sse2.c
similarity index 94%
rename from src/modules/audio_processing/aec/main/source/aec_core_sse2.c
rename to src/modules/audio_processing/aec/aec_core_sse2.c
index 524669f..8894f28 100644
--- a/src/modules/audio_processing/aec/main/source/aec_core_sse2.c
+++ b/src/modules/audio_processing/aec/aec_core_sse2.c
@@ -12,7 +12,9 @@
* The core AEC algorithm, SSE2 version of speed-critical functions.
*/
-#if defined(__SSE2__)
+#include "typedefs.h"
+
+#if defined(WEBRTC_USE_SSE2)
#include <emmintrin.h>
#include <math.h>
@@ -136,16 +138,6 @@
xPos -= NR_PART * PART_LEN1;
}
-#ifdef UNCONSTR
- for (j = 0; j < PART_LEN1; j++) {
- aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
- -aec->xfBuf[xPos + j][1],
- ef[j][0], ef[j][1]);
- aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
- -aec->xfBuf[xPos + j][1],
- ef[j][0], ef[j][1]);
- }
-#else
// Process the whole array...
for (j = 0; j < PART_LEN; j+= 4) {
// Load xfBuf and ef.
@@ -206,18 +198,9 @@
}
aec->wfBuf[1][pos] = wt1;
}
-#endif // UNCONSTR
}
}
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
static __m128 mm_pow_ps(__m128 a, __m128 b)
{
// a^b = exp2(b * log2(a))
@@ -252,10 +235,9 @@
{0x43BF8000, 0x43BF8000, 0x43BF8000, 0x43BF8000};
static const int shift_exponent_into_top_mantissa = 8;
const __m128 two_n = _mm_and_ps(a, *((__m128 *)float_exponent_mask));
- const __m128 n_1 = (__m128)_mm_srli_epi32((__m128i)two_n,
- shift_exponent_into_top_mantissa);
- const __m128 n_0 = _mm_or_ps(
- (__m128)n_1, *((__m128 *)eight_biased_exponent));
+ const __m128 n_1 = _mm_castsi128_ps(_mm_srli_epi32(_mm_castps_si128(two_n),
+ shift_exponent_into_top_mantissa));
+ const __m128 n_0 = _mm_or_ps(n_1, *((__m128 *)eight_biased_exponent));
const __m128 n = _mm_sub_ps(n_0, *((__m128 *)implicit_leading_one));
// Compute y.
@@ -334,8 +316,8 @@
static const int float_exponent_shift = 23;
const __m128i two_n_exponent = _mm_add_epi32(
x_minus_half_floor, *((__m128i *)float_exponent_bias));
- const __m128 two_n = (__m128)_mm_slli_epi32(
- two_n_exponent, float_exponent_shift);
+ const __m128 two_n = _mm_castsi128_ps(_mm_slli_epi32(
+ two_n_exponent, float_exponent_shift));
// Compute y.
const __m128 y = _mm_sub_ps(x_max, _mm_cvtepi32_ps(x_minus_half_floor));
// Approximate 2^y ~= C2 * y^2 + C1 * y + C0.
@@ -432,4 +414,4 @@
WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
}
-#endif //__SSE2__
+#endif // WEBRTC_USE_SSE2
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft.c b/src/modules/audio_processing/aec/aec_rdft.c
similarity index 65%
rename from src/modules/audio_processing/aec/main/source/aec_rdft.c
rename to src/modules/audio_processing/aec/aec_rdft.c
index 072a1c4..9222334 100644
--- a/src/modules/audio_processing/aec/main/source/aec_rdft.c
+++ b/src/modules/audio_processing/aec/aec_rdft.c
@@ -19,12 +19,27 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "aec_rdft.h"
+
#include <math.h>
-#include "aec_rdft.h"
#include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "typedefs.h"
+// constants shared by all paths (C, SSE2).
float rdft_w[64];
+// constants used by the C path.
+float rdft_wk3ri_first[32];
+float rdft_wk3ri_second[32];
+// constants used by SSE2 but initialized in C path.
+ALIGN16_BEG float ALIGN16_END rdft_wk1r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk2r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk3r[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk1i[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk2i[32];
+ALIGN16_BEG float ALIGN16_END rdft_wk3i[32];
+ALIGN16_BEG float ALIGN16_END cftmdl_wk1r[4];
+
static int ip[16];
static void bitrv2_32or128(int n, int *ip, float *a) {
@@ -101,7 +116,7 @@
}
}
-static void makewt_32() {
+static void makewt_32(void) {
const int nw = 32;
int j, nwh;
float delta, x, y;
@@ -123,9 +138,59 @@
rdft_w[nw - j + 1] = x;
}
bitrv2_32or128(nw, ip + 2, rdft_w);
+
+ // pre-calculate constants used by cft1st_128 and cftmdl_128...
+ cftmdl_wk1r[0] = rdft_w[2];
+ cftmdl_wk1r[1] = rdft_w[2];
+ cftmdl_wk1r[2] = rdft_w[2];
+ cftmdl_wk1r[3] = -rdft_w[2];
+ {
+ int k1;
+
+ for (k1 = 0, j = 0; j < 128; j += 16, k1 += 2) {
+ const int k2 = 2 * k1;
+ const float wk2r = rdft_w[k1 + 0];
+ const float wk2i = rdft_w[k1 + 1];
+ float wk1r, wk1i;
+ // ... scalar version.
+ wk1r = rdft_w[k2 + 0];
+ wk1i = rdft_w[k2 + 1];
+ rdft_wk3ri_first[k1 + 0] = wk1r - 2 * wk2i * wk1i;
+ rdft_wk3ri_first[k1 + 1] = 2 * wk2i * wk1r - wk1i;
+ wk1r = rdft_w[k2 + 2];
+ wk1i = rdft_w[k2 + 3];
+ rdft_wk3ri_second[k1 + 0] = wk1r - 2 * wk2r * wk1i;
+ rdft_wk3ri_second[k1 + 1] = 2 * wk2r * wk1r - wk1i;
+ // ... vector version.
+ rdft_wk1r[k2 + 0] = rdft_w[k2 + 0];
+ rdft_wk1r[k2 + 1] = rdft_w[k2 + 0];
+ rdft_wk1r[k2 + 2] = rdft_w[k2 + 2];
+ rdft_wk1r[k2 + 3] = rdft_w[k2 + 2];
+ rdft_wk2r[k2 + 0] = rdft_w[k1 + 0];
+ rdft_wk2r[k2 + 1] = rdft_w[k1 + 0];
+ rdft_wk2r[k2 + 2] = -rdft_w[k1 + 1];
+ rdft_wk2r[k2 + 3] = -rdft_w[k1 + 1];
+ rdft_wk3r[k2 + 0] = rdft_wk3ri_first[k1 + 0];
+ rdft_wk3r[k2 + 1] = rdft_wk3ri_first[k1 + 0];
+ rdft_wk3r[k2 + 2] = rdft_wk3ri_second[k1 + 0];
+ rdft_wk3r[k2 + 3] = rdft_wk3ri_second[k1 + 0];
+ rdft_wk1i[k2 + 0] = -rdft_w[k2 + 1];
+ rdft_wk1i[k2 + 1] = rdft_w[k2 + 1];
+ rdft_wk1i[k2 + 2] = -rdft_w[k2 + 3];
+ rdft_wk1i[k2 + 3] = rdft_w[k2 + 3];
+ rdft_wk2i[k2 + 0] = -rdft_w[k1 + 1];
+ rdft_wk2i[k2 + 1] = rdft_w[k1 + 1];
+ rdft_wk2i[k2 + 2] = -rdft_w[k1 + 0];
+ rdft_wk2i[k2 + 3] = rdft_w[k1 + 0];
+ rdft_wk3i[k2 + 0] = -rdft_wk3ri_first[k1 + 1];
+ rdft_wk3i[k2 + 1] = rdft_wk3ri_first[k1 + 1];
+ rdft_wk3i[k2 + 2] = -rdft_wk3ri_second[k1 + 1];
+ rdft_wk3i[k2 + 3] = rdft_wk3ri_second[k1 + 1];
+ }
+ }
}
-static void makect_32() {
+static void makect_32(void) {
float *c = rdft_w + 32;
const int nc = 32;
int j, nch;
@@ -142,7 +207,7 @@
}
}
-static void cft1st_128(float *a) {
+static void cft1st_128_C(float *a) {
const int n = 128;
int j, k1, k2;
float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
@@ -189,21 +254,21 @@
for (j = 16; j < n; j += 16) {
k1 += 2;
k2 = 2 * k1;
- wk2r = rdft_w[k1];
+ wk2r = rdft_w[k1 + 0];
wk2i = rdft_w[k1 + 1];
- wk1r = rdft_w[k2];
+ wk1r = rdft_w[k2 + 0];
wk1i = rdft_w[k2 + 1];
- wk3r = wk1r - 2 * wk2i * wk1i;
- wk3i = 2 * wk2i * wk1r - wk1i;
- x0r = a[j] + a[j + 2];
+ wk3r = rdft_wk3ri_first[k1 + 0];
+ wk3i = rdft_wk3ri_first[k1 + 1];
+ x0r = a[j + 0] + a[j + 2];
x0i = a[j + 1] + a[j + 3];
- x1r = a[j] - a[j + 2];
+ x1r = a[j + 0] - a[j + 2];
x1i = a[j + 1] - a[j + 3];
x2r = a[j + 4] + a[j + 6];
x2i = a[j + 5] + a[j + 7];
x3r = a[j + 4] - a[j + 6];
x3i = a[j + 5] - a[j + 7];
- a[j] = x0r + x2r;
+ a[j + 0] = x0r + x2r;
a[j + 1] = x0i + x2i;
x0r -= x2r;
x0i -= x2i;
@@ -219,8 +284,8 @@
a[j + 7] = wk3r * x0i + wk3i * x0r;
wk1r = rdft_w[k2 + 2];
wk1i = rdft_w[k2 + 3];
- wk3r = wk1r - 2 * wk2r * wk1i;
- wk3i = 2 * wk2r * wk1r - wk1i;
+ wk3r = rdft_wk3ri_second[k1 + 0];
+ wk3i = rdft_wk3ri_second[k1 + 1];
x0r = a[j + 8] + a[j + 10];
x0i = a[j + 9] + a[j + 11];
x1r = a[j + 8] - a[j + 10];
@@ -246,58 +311,59 @@
}
}
-static void cftmdl_128(int l, float *a) {
+static void cftmdl_128_C(float *a) {
+ const int l = 8;
const int n = 128;
- int j, j1, j2, j3, k, k1, k2, m, m2;
+ const int m = 32;
+ int j0, j1, j2, j3, k, k1, k2, m2;
float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
- m = l << 2;
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
+ for (j0 = 0; j0 < l; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
+ x3r = a[j2 + 0] - a[j3 + 0];
x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x0r - x2r;
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ a[j2 + 0] = x0r - x2r;
a[j2 + 1] = x0i - x2i;
- a[j1] = x1r - x3i;
+ a[j1 + 0] = x1r - x3i;
a[j1 + 1] = x1i + x3r;
- a[j3] = x1r + x3i;
+ a[j3 + 0] = x1r + x3i;
a[j3 + 1] = x1i - x3r;
}
wk1r = rdft_w[2];
- for (j = m; j < l + m; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
+ for (j0 = m; j0 < l + m; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
+ x3r = a[j2 + 0] - a[j3 + 0];
x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x2i - x0i;
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
+ a[j2 + 0] = x2i - x0i;
a[j2 + 1] = x0r - x2r;
x0r = x1r - x3i;
x0i = x1i + x3r;
- a[j1] = wk1r * (x0r - x0i);
+ a[j1 + 0] = wk1r * (x0r - x0i);
a[j1 + 1] = wk1r * (x0r + x0i);
x0r = x3i + x1r;
x0i = x3r - x1i;
- a[j3] = wk1r * (x0i - x0r);
+ a[j3 + 0] = wk1r * (x0i - x0r);
a[j3 + 1] = wk1r * (x0i + x0r);
}
k1 = 0;
@@ -305,68 +371,68 @@
for (k = m2; k < n; k += m2) {
k1 += 2;
k2 = 2 * k1;
- wk2r = rdft_w[k1];
+ wk2r = rdft_w[k1 + 0];
wk2i = rdft_w[k1 + 1];
- wk1r = rdft_w[k2];
+ wk1r = rdft_w[k2 + 0];
wk1i = rdft_w[k2 + 1];
- wk3r = wk1r - 2 * wk2i * wk1i;
- wk3i = 2 * wk2i * wk1r - wk1i;
- for (j = k; j < l + k; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
+ wk3r = rdft_wk3ri_first[k1 + 0];
+ wk3i = rdft_wk3ri_first[k1 + 1];
+ for (j0 = k; j0 < l + k; j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
+ x3r = a[j2 + 0] - a[j3 + 0];
x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
x0r -= x2r;
x0i -= x2i;
- a[j2] = wk2r * x0r - wk2i * x0i;
+ a[j2 + 0] = wk2r * x0r - wk2i * x0i;
a[j2 + 1] = wk2r * x0i + wk2i * x0r;
x0r = x1r - x3i;
x0i = x1i + x3r;
- a[j1] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 0] = wk1r * x0r - wk1i * x0i;
a[j1 + 1] = wk1r * x0i + wk1i * x0r;
x0r = x1r + x3i;
x0i = x1i - x3r;
- a[j3] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 0] = wk3r * x0r - wk3i * x0i;
a[j3 + 1] = wk3r * x0i + wk3i * x0r;
}
wk1r = rdft_w[k2 + 2];
wk1i = rdft_w[k2 + 3];
- wk3r = wk1r - 2 * wk2r * wk1i;
- wk3i = 2 * wk2r * wk1r - wk1i;
- for (j = k + m; j < l + (k + m); j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
+ wk3r = rdft_wk3ri_second[k1 + 0];
+ wk3i = rdft_wk3ri_second[k1 + 1];
+ for (j0 = k + m; j0 < l + (k + m); j0 += 2) {
+ j1 = j0 + 8;
+ j2 = j0 + 16;
+ j3 = j0 + 24;
+ x0r = a[j0 + 0] + a[j1 + 0];
+ x0i = a[j0 + 1] + a[j1 + 1];
+ x1r = a[j0 + 0] - a[j1 + 0];
+ x1i = a[j0 + 1] - a[j1 + 1];
+ x2r = a[j2 + 0] + a[j3 + 0];
x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
+ x3r = a[j2 + 0] - a[j3 + 0];
x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
+ a[j0 + 0] = x0r + x2r;
+ a[j0 + 1] = x0i + x2i;
x0r -= x2r;
x0i -= x2i;
- a[j2] = -wk2i * x0r - wk2r * x0i;
+ a[j2 + 0] = -wk2i * x0r - wk2r * x0i;
a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
x0r = x1r - x3i;
x0i = x1i + x3r;
- a[j1] = wk1r * x0r - wk1i * x0i;
+ a[j1 + 0] = wk1r * x0r - wk1i * x0i;
a[j1 + 1] = wk1r * x0i + wk1i * x0r;
x0r = x1r + x3i;
x0i = x1i - x3r;
- a[j3] = wk3r * x0r - wk3i * x0i;
+ a[j3 + 0] = wk3r * x0r - wk3i * x0i;
a[j3 + 1] = wk3r * x0i + wk3i * x0r;
}
}
@@ -377,7 +443,7 @@
float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
cft1st_128(a);
- cftmdl_128(8, a);
+ cftmdl_128(a);
l = 32;
for (j = 0; j < l; j += 2) {
j1 = j + l;
@@ -407,7 +473,7 @@
float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
cft1st_128(a);
- cftmdl_128(8, a);
+ cftmdl_128(a);
l = 32;
for (j = 0; j < l; j += 2) {
@@ -479,10 +545,8 @@
void aec_rdft_forward_128(float *a) {
const int n = 128;
- int nw;
float xi;
- nw = ip[0];
bitrv2_32or128(n, ip + 2, a);
cftfsub_128(a);
rftfsub_128(a);
@@ -493,10 +557,7 @@
void aec_rdft_inverse_128(float *a) {
const int n = 128;
- int nw;
- float xi;
- nw = ip[0];
a[1] = 0.5f * (a[0] - a[1]);
a[0] -= a[1];
rftbsub_128(a);
@@ -505,14 +566,18 @@
}
// code path selection
+rft_sub_128_t cft1st_128;
+rft_sub_128_t cftmdl_128;
rft_sub_128_t rftfsub_128;
rft_sub_128_t rftbsub_128;
void aec_rdft_init(void) {
+ cft1st_128 = cft1st_128_C;
+ cftmdl_128 = cftmdl_128_C;
rftfsub_128 = rftfsub_128_C;
rftbsub_128 = rftbsub_128_C;
if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
+#if defined(WEBRTC_USE_SSE2)
aec_rdft_init_sse2();
#endif
}
diff --git a/src/modules/audio_processing/aec/aec_rdft.h b/src/modules/audio_processing/aec/aec_rdft.h
new file mode 100644
index 0000000..91bedc9
--- /dev/null
+++ b/src/modules/audio_processing/aec/aec_rdft.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+
+// These intrinsics were unavailable before VS 2008.
+// TODO(andrew): move to a common file.
+#if defined(_MSC_VER) && _MSC_VER < 1500
+#include <emmintrin.h>
+static __inline __m128 _mm_castsi128_ps(__m128i a) { return *(__m128*)&a; }
+static __inline __m128i _mm_castps_si128(__m128 a) { return *(__m128i*)&a; }
+#endif
+
+#ifdef _MSC_VER /* visual c++ */
+# define ALIGN16_BEG __declspec(align(16))
+# define ALIGN16_END
+#else /* gcc or icc */
+# define ALIGN16_BEG
+# define ALIGN16_END __attribute__((aligned(16)))
+#endif
+
+// constants shared by all paths (C, SSE2).
+extern float rdft_w[64];
+// constants used by the C path.
+extern float rdft_wk3ri_first[32];
+extern float rdft_wk3ri_second[32];
+// constants used by SSE2 but initialized in C path.
+extern float rdft_wk1r[32];
+extern float rdft_wk2r[32];
+extern float rdft_wk3r[32];
+extern float rdft_wk1i[32];
+extern float rdft_wk2i[32];
+extern float rdft_wk3i[32];
+extern float cftmdl_wk1r[4];
+
+// code path selection function pointers
+typedef void (*rft_sub_128_t)(float *a);
+extern rft_sub_128_t rftfsub_128;
+extern rft_sub_128_t rftbsub_128;
+extern rft_sub_128_t cft1st_128;
+extern rft_sub_128_t cftmdl_128;
+
+// entry points
+void aec_rdft_init(void);
+void aec_rdft_init_sse2(void);
+void aec_rdft_forward_128(float *a);
+void aec_rdft_inverse_128(float *a);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
diff --git a/src/modules/audio_processing/aec/aec_rdft_sse2.c b/src/modules/audio_processing/aec/aec_rdft_sse2.c
new file mode 100644
index 0000000..f936e2a
--- /dev/null
+++ b/src/modules/audio_processing/aec/aec_rdft_sse2.c
@@ -0,0 +1,431 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "typedefs.h"
+
+#if defined(WEBRTC_USE_SSE2)
+#include <emmintrin.h>
+
+#include "aec_rdft.h"
+
+static const ALIGN16_BEG float ALIGN16_END k_swap_sign[4] =
+ {-1.f, 1.f, -1.f, 1.f};
+
+static void cft1st_128_SSE2(float *a) {
+ const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
+ int j, k2;
+
+ for (k2 = 0, j = 0; j < 128; j += 16, k2 += 4) {
+ __m128 a00v = _mm_loadu_ps(&a[j + 0]);
+ __m128 a04v = _mm_loadu_ps(&a[j + 4]);
+ __m128 a08v = _mm_loadu_ps(&a[j + 8]);
+ __m128 a12v = _mm_loadu_ps(&a[j + 12]);
+ __m128 a01v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 a23v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(3, 2, 3 ,2));
+ __m128 a45v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 a67v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(3, 2, 3 ,2));
+
+ const __m128 wk1rv = _mm_load_ps(&rdft_wk1r[k2]);
+ const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2]);
+ const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2]);
+ const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2]);
+ const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2]);
+ const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2]);
+ __m128 x0v = _mm_add_ps(a01v, a23v);
+ const __m128 x1v = _mm_sub_ps(a01v, a23v);
+ const __m128 x2v = _mm_add_ps(a45v, a67v);
+ const __m128 x3v = _mm_sub_ps(a45v, a67v);
+ __m128 x0w;
+ a01v = _mm_add_ps(x0v, x2v);
+ x0v = _mm_sub_ps(x0v, x2v);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ {
+ const __m128 a45_0v = _mm_mul_ps(wk2rv, x0v);
+ const __m128 a45_1v = _mm_mul_ps(wk2iv, x0w);
+ a45v = _mm_add_ps(a45_0v, a45_1v);
+ }
+ {
+ __m128 a23_0v, a23_1v;
+ const __m128 x3w = _mm_shuffle_ps(x3v, x3v, _MM_SHUFFLE(2, 3, 0 ,1));
+ const __m128 x3s = _mm_mul_ps(mm_swap_sign, x3w);
+ x0v = _mm_add_ps(x1v, x3s);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ a23_0v = _mm_mul_ps(wk1rv, x0v);
+ a23_1v = _mm_mul_ps(wk1iv, x0w);
+ a23v = _mm_add_ps(a23_0v, a23_1v);
+
+ x0v = _mm_sub_ps(x1v, x3s);
+ x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
+ }
+ {
+ const __m128 a67_0v = _mm_mul_ps(wk3rv, x0v);
+ const __m128 a67_1v = _mm_mul_ps(wk3iv, x0w);
+ a67v = _mm_add_ps(a67_0v, a67_1v);
+ }
+
+ a00v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(1, 0, 1 ,0));
+ a04v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(1, 0, 1 ,0));
+ a08v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(3, 2, 3 ,2));
+ a12v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(3, 2, 3 ,2));
+ _mm_storeu_ps(&a[j + 0], a00v);
+ _mm_storeu_ps(&a[j + 4], a04v);
+ _mm_storeu_ps(&a[j + 8], a08v);
+ _mm_storeu_ps(&a[j + 12], a12v);
+ }
+}
+
+static void cftmdl_128_SSE2(float *a) {
+ const int l = 8;
+ const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
+ int j0;
+
+ __m128 wk1rv = _mm_load_ps(cftmdl_wk1r);
+ for (j0 = 0; j0 < l; j0 += 2) {
+ const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
+ const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
+ const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
+ const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
+ const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
+ _mm_castsi128_ps(a_32),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
+ _mm_castsi128_ps(a_40),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
+ const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
+
+ const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
+ const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
+ const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
+ const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
+ const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
+ _mm_castsi128_ps(a_48),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
+ _mm_castsi128_ps(a_56),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
+ const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
+
+ const __m128 xx0 = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+
+ const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
+ _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
+ _MM_SHUFFLE(2, 3, 0, 1)));
+ const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
+ const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+ const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+
+ const __m128 yy0 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
+ _MM_SHUFFLE(2, 2, 2 ,2));
+ const __m128 yy1 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
+ _MM_SHUFFLE(3, 3, 3 ,3));
+ const __m128 yy2 = _mm_mul_ps(mm_swap_sign, yy1);
+ const __m128 yy3 = _mm_add_ps(yy0, yy2);
+ const __m128 yy4 = _mm_mul_ps(wk1rv, yy3);
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx0));
+ _mm_storel_epi64((__m128i*)&a[j0 + 32],
+ _mm_shuffle_epi32(_mm_castps_si128(xx0),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx1));
+ _mm_storel_epi64((__m128i*)&a[j0 + 48],
+ _mm_shuffle_epi32(_mm_castps_si128(xx1),
+ _MM_SHUFFLE(2, 3, 2, 3)));
+ a[j0 + 48] = -a[j0 + 48];
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(x1_x3_add));
+ _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(x1_x3_sub));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 40], _mm_castps_si128(yy4));
+ _mm_storel_epi64((__m128i*)&a[j0 + 56],
+ _mm_shuffle_epi32(_mm_castps_si128(yy4),
+ _MM_SHUFFLE(2, 3, 2, 3)));
+ }
+
+ {
+ int k = 64;
+ int k1 = 2;
+ int k2 = 2 * k1;
+ const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2+0]);
+ const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2+0]);
+ const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2+0]);
+ const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2+0]);
+ const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2+0]);
+ wk1rv = _mm_load_ps(&rdft_wk1r[k2+0]);
+ for (j0 = k; j0 < l + k; j0 += 2) {
+ const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
+ const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
+ const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
+ const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
+ const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
+ _mm_castsi128_ps(a_32),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
+ _mm_castsi128_ps(a_40),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
+ const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
+
+ const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
+ const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
+ const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
+ const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
+ const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
+ _mm_castsi128_ps(a_48),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
+ _mm_castsi128_ps(a_56),
+ _MM_SHUFFLE(1, 0, 1 ,0));
+ const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
+ const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
+
+ const __m128 xx = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
+ const __m128 xx2 = _mm_mul_ps(xx1 , wk2rv);
+ const __m128 xx3 = _mm_mul_ps(wk2iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(xx1),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx4 = _mm_add_ps(xx2, xx3);
+
+ const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
+ _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
+ _MM_SHUFFLE(2, 3, 0, 1)));
+ const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
+ const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+ const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
+
+ const __m128 xx10 = _mm_mul_ps(x1_x3_add, wk1rv);
+ const __m128 xx11 = _mm_mul_ps(wk1iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_add),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx12 = _mm_add_ps(xx10, xx11);
+
+ const __m128 xx20 = _mm_mul_ps(x1_x3_sub, wk3rv);
+ const __m128 xx21 = _mm_mul_ps(wk3iv,
+ _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_sub),
+ _MM_SHUFFLE(2, 3, 0, 1))));
+ const __m128 xx22 = _mm_add_ps(xx20, xx21);
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx));
+ _mm_storel_epi64((__m128i*)&a[j0 + 32],
+ _mm_shuffle_epi32(_mm_castps_si128(xx),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx4));
+ _mm_storel_epi64((__m128i*)&a[j0 + 48],
+ _mm_shuffle_epi32(_mm_castps_si128(xx4),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(xx12));
+ _mm_storel_epi64((__m128i*)&a[j0 + 40],
+ _mm_shuffle_epi32(_mm_castps_si128(xx12),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+
+ _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(xx22));
+ _mm_storel_epi64((__m128i*)&a[j0 + 56],
+ _mm_shuffle_epi32(_mm_castps_si128(xx22),
+ _MM_SHUFFLE(3, 2, 3, 2)));
+ }
+ }
+}
+
+static void rftfsub_128_SSE2(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ static const ALIGN16_BEG float ALIGN16_END k_half[4] =
+ {0.5f, 0.5f, 0.5f, 0.5f};
+ const __m128 mm_half = _mm_load_ps(k_half);
+
+ // Vectorized code (four at once).
+ // Note: commented number are indexes for the first iteration of the loop.
+ for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
+ // Load 'wk'.
+ const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
+ const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
+ const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
+ const __m128 wkr_ =
+ _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
+ const __m128 wki_ = c_j1; // 1, 2, 3, 4,
+ // Load and shuffle 'a'.
+ const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
+ const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
+ const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
+ const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
+ const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
+ const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
+ const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
+ const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
+ // Calculate 'x'.
+ const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
+ // 2-126, 4-124, 6-122, 8-120,
+ const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
+ // 3-127, 5-125, 7-123, 9-121,
+ // Calculate product into 'y'.
+ // yr = wkr * xr - wki * xi;
+ // yi = wkr * xi + wki * xr;
+ const __m128 a_ = _mm_mul_ps(wkr_, xr_);
+ const __m128 b_ = _mm_mul_ps(wki_, xi_);
+ const __m128 c_ = _mm_mul_ps(wkr_, xi_);
+ const __m128 d_ = _mm_mul_ps(wki_, xr_);
+ const __m128 yr_ = _mm_sub_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
+ const __m128 yi_ = _mm_add_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
+ // Update 'a'.
+ // a[j2 + 0] -= yr;
+ // a[j2 + 1] -= yi;
+ // a[k2 + 0] += yr;
+ // a[k2 + 1] -= yi;
+ const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
+ const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_); // 3, 5, 7, 9,
+ const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
+ const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_); // 127, 125, 123, 121,
+ // Shuffle in right order and store.
+ const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
+ // 2, 3, 4, 5,
+ const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
+ // 6, 7, 8, 9,
+ const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
+ // 122, 123, 120, 121,
+ const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
+ // 126, 127, 124, 125,
+ const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
+ const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
+ _mm_storeu_ps(&a[0 + j2], a_j2_0n);
+ _mm_storeu_ps(&a[4 + j2], a_j2_4n);
+ _mm_storeu_ps(&a[122 - j2], a_k2_0n);
+ _mm_storeu_ps(&a[126 - j2], a_k2_4n);
+ }
+ // Scalar code for the remaining items.
+ for (; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr - wki * xi;
+ yi = wkr * xi + wki * xr;
+ a[j2 + 0] -= yr;
+ a[j2 + 1] -= yi;
+ a[k2 + 0] += yr;
+ a[k2 + 1] -= yi;
+ }
+}
+
+static void rftbsub_128_SSE2(float *a) {
+ const float *c = rdft_w + 32;
+ int j1, j2, k1, k2;
+ float wkr, wki, xr, xi, yr, yi;
+
+ static const ALIGN16_BEG float ALIGN16_END k_half[4] =
+ {0.5f, 0.5f, 0.5f, 0.5f};
+ const __m128 mm_half = _mm_load_ps(k_half);
+
+ a[1] = -a[1];
+ // Vectorized code (four at once).
+ // Note: commented number are indexes for the first iteration of the loop.
+ for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
+ // Load 'wk'.
+ const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
+ const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
+ const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
+ const __m128 wkr_ =
+ _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
+ const __m128 wki_ = c_j1; // 1, 2, 3, 4,
+ // Load and shuffle 'a'.
+ const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
+ const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
+ const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
+ const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
+ const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
+ const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
+ _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
+ const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
+ const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
+ _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
+ // Calculate 'x'.
+ const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
+ // 2-126, 4-124, 6-122, 8-120,
+ const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
+ // 3-127, 5-125, 7-123, 9-121,
+ // Calculate product into 'y'.
+ // yr = wkr * xr + wki * xi;
+ // yi = wkr * xi - wki * xr;
+ const __m128 a_ = _mm_mul_ps(wkr_, xr_);
+ const __m128 b_ = _mm_mul_ps(wki_, xi_);
+ const __m128 c_ = _mm_mul_ps(wkr_, xi_);
+ const __m128 d_ = _mm_mul_ps(wki_, xr_);
+ const __m128 yr_ = _mm_add_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
+ const __m128 yi_ = _mm_sub_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
+ // Update 'a'.
+ // a[j2 + 0] = a[j2 + 0] - yr;
+ // a[j2 + 1] = yi - a[j2 + 1];
+ // a[k2 + 0] = yr + a[k2 + 0];
+ // a[k2 + 1] = yi - a[k2 + 1];
+ const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
+ const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1); // 3, 5, 7, 9,
+ const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
+ const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1); // 127, 125, 123, 121,
+ // Shuffle in right order and store.
+ const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
+ // 2, 3, 4, 5,
+ const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
+ // 6, 7, 8, 9,
+ const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
+ // 122, 123, 120, 121,
+ const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
+ // 126, 127, 124, 125,
+ const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
+ const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
+ _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
+ _mm_storeu_ps(&a[0 + j2], a_j2_0n);
+ _mm_storeu_ps(&a[4 + j2], a_j2_4n);
+ _mm_storeu_ps(&a[122 - j2], a_k2_0n);
+ _mm_storeu_ps(&a[126 - j2], a_k2_4n);
+ }
+ // Scalar code for the remaining items.
+ for (; j2 < 64; j1 += 1, j2 += 2) {
+ k2 = 128 - j2;
+ k1 = 32 - j1;
+ wkr = 0.5f - c[k1];
+ wki = c[j1];
+ xr = a[j2 + 0] - a[k2 + 0];
+ xi = a[j2 + 1] + a[k2 + 1];
+ yr = wkr * xr + wki * xi;
+ yi = wkr * xi - wki * xr;
+ a[j2 + 0] = a[j2 + 0] - yr;
+ a[j2 + 1] = yi - a[j2 + 1];
+ a[k2 + 0] = yr + a[k2 + 0];
+ a[k2 + 1] = yi - a[k2 + 1];
+ }
+ a[65] = -a[65];
+}
+
+void aec_rdft_init_sse2(void) {
+ cft1st_128 = cft1st_128_SSE2;
+ cftmdl_128 = cftmdl_128_SSE2;
+ rftfsub_128 = rftfsub_128_SSE2;
+ rftbsub_128 = rftbsub_128_SSE2;
+}
+
+#endif // WEBRTC_USE_SS2
diff --git a/src/modules/audio_processing/aec/main/source/resampler.c b/src/modules/audio_processing/aec/aec_resampler.c
similarity index 94%
rename from src/modules/audio_processing/aec/main/source/resampler.c
rename to src/modules/audio_processing/aec/aec_resampler.c
index 4caa6f4..ea980cd 100644
--- a/src/modules/audio_processing/aec/main/source/resampler.c
+++ b/src/modules/audio_processing/aec/aec_resampler.c
@@ -12,19 +12,19 @@
* skew by resampling the farend signal.
*/
+#include "aec_resampler.h"
+
#include <assert.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
-#include "resampler.h"
#include "aec_core.h"
-enum { kFrameBufferSize = FRAME_LEN * 4 };
enum { kEstimateLengthFrames = 400 };
typedef struct {
- short buffer[kFrameBufferSize];
+ short buffer[kResamplerBufferSize];
float position;
int deviceSampleRateHz;
@@ -127,7 +127,7 @@
// Shift buffer
memmove(obj->buffer,
&obj->buffer[size],
- (kFrameBufferSize - size) * sizeof(short));
+ (kResamplerBufferSize - size) * sizeof(short));
return outsize;
}
@@ -157,8 +157,8 @@
}
int EstimateSkew(const int* rawSkew,
- const int size,
- const int deviceSampleRateHz,
+ int size,
+ int deviceSampleRateHz,
float *skewEst)
{
const int absLimitOuter = (int)(0.04f * deviceSampleRateHz);
@@ -176,7 +176,6 @@
float y = 0;
float xy = 0;
float xAvg = 0;
- float yAvg = 0;
float denom = 0;
float skew = 0;
@@ -223,7 +222,6 @@
}
assert(n > 0);
xAvg = x / n;
- yAvg = y / n;
denom = x2 - xAvg*x;
if (denom != 0) {
diff --git a/src/modules/audio_processing/aec/main/source/resampler.h b/src/modules/audio_processing/aec/aec_resampler.h
similarity index 81%
rename from src/modules/audio_processing/aec/main/source/resampler.h
rename to src/modules/audio_processing/aec/aec_resampler.h
index 9cb2837..ab4cc6e 100644
--- a/src/modules/audio_processing/aec/main/source/resampler.h
+++ b/src/modules/audio_processing/aec/aec_resampler.h
@@ -8,10 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+
+#include "aec_core.h"
enum { kResamplingDelay = 1 };
+enum { kResamplerBufferSize = FRAME_LEN * 4 };
// Unless otherwise specified, functions return 0 on success and -1 on error
int WebRtcAec_CreateResampler(void **resampInst);
@@ -29,4 +32,4 @@
float skew,
short *outspeech);
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
diff --git a/src/modules/audio_processing/aec/echo_cancellation.c b/src/modules/audio_processing/aec/echo_cancellation.c
new file mode 100644
index 0000000..66c9b97
--- /dev/null
+++ b/src/modules/audio_processing/aec/echo_cancellation.c
@@ -0,0 +1,940 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Contains the API functions for the AEC.
+ */
+#include "echo_cancellation.h"
+
+#include <math.h>
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#endif
+#include <stdlib.h>
+#include <string.h>
+
+#include "aec_core.h"
+#include "aec_resampler.h"
+#include "ring_buffer.h"
+#include "typedefs.h"
+
+// Maximum length of resampled signal. Must be an integer multiple of frames
+// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
+// The factor of 2 handles wb, and the + 1 is as a safety margin
+// TODO(bjornv): Replace with kResamplerBufferSize
+#define MAX_RESAMP_LEN (5 * FRAME_LEN)
+
+static const int kMaxBufSizeStart = 62; // In partitions
+static const int sampMsNb = 8; // samples per ms in nb
+// Target suppression levels for nlp modes
+// log{0.001, 0.00001, 0.00000001}
+static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
+static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
+static const int initCheck = 42;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+static int instance_count = 0;
+#endif
+
+typedef struct {
+ int delayCtr;
+ int sampFreq;
+ int splitSampFreq;
+ int scSampFreq;
+ float sampFactor; // scSampRate / sampFreq
+ short nlpMode;
+ short autoOnOff;
+ short activity;
+ short skewMode;
+ int bufSizeStart;
+ //short bufResetCtr; // counts number of noncausal frames
+ int knownDelay;
+
+ short initFlag; // indicates if AEC has been initialized
+
+ // Variables used for averaging far end buffer size
+ short counter;
+ int sum;
+ short firstVal;
+ short checkBufSizeCtr;
+
+ // Variables used for delay shifts
+ short msInSndCardBuf;
+ short filtDelay; // Filtered delay estimate.
+ int timeForDelayChange;
+ int ECstartup;
+ int checkBuffSize;
+ short lastDelayDiff;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ void* far_pre_buf_s16; // Time domain far-end pre-buffer in int16_t.
+ FILE *bufFile;
+ FILE *delayFile;
+ FILE *skewFile;
+#endif
+
+ // Structures
+ void *resampler;
+
+ int skewFrCtr;
+ int resample; // if the skew is small enough we don't resample
+ int highSkewCtr;
+ float skew;
+
+ void* far_pre_buf; // Time domain far-end pre-buffer.
+
+ int lastError;
+
+ aec_t *aec;
+} aecpc_t;
+
+// Estimates delay to set the position of the far-end buffer read pointer
+// (controlled by knownDelay)
+static int EstBufDelay(aecpc_t *aecInst);
+
+WebRtc_Word32 WebRtcAec_Create(void **aecInst)
+{
+ aecpc_t *aecpc;
+ if (aecInst == NULL) {
+ return -1;
+ }
+
+ aecpc = malloc(sizeof(aecpc_t));
+ *aecInst = aecpc;
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+
+ if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+ // Create far-end pre-buffer. The buffer size has to be large enough for
+ // largest possible drift compensation (kResamplerBufferSize) + "almost" an
+ // FFT buffer (PART_LEN2 - 1).
+ if (WebRtc_CreateBuffer(&aecpc->far_pre_buf,
+ PART_LEN2 + kResamplerBufferSize,
+ sizeof(float)) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+
+ aecpc->initFlag = 0;
+ aecpc->lastError = 0;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_CreateBuffer(&aecpc->far_pre_buf_s16,
+ PART_LEN2 + kResamplerBufferSize,
+ sizeof(int16_t)) == -1) {
+ WebRtcAec_Free(aecpc);
+ aecpc = NULL;
+ return -1;
+ }
+ {
+ char filename[64];
+ sprintf(filename, "aec_far%d.pcm", instance_count);
+ aecpc->aec->farFile = fopen(filename, "wb");
+ sprintf(filename, "aec_near%d.pcm", instance_count);
+ aecpc->aec->nearFile = fopen(filename, "wb");
+ sprintf(filename, "aec_out%d.pcm", instance_count);
+ aecpc->aec->outFile = fopen(filename, "wb");
+ sprintf(filename, "aec_out_linear%d.pcm", instance_count);
+ aecpc->aec->outLinearFile = fopen(filename, "wb");
+ sprintf(filename, "aec_buf%d.dat", instance_count);
+ aecpc->bufFile = fopen(filename, "wb");
+ sprintf(filename, "aec_skew%d.dat", instance_count);
+ aecpc->skewFile = fopen(filename, "wb");
+ sprintf(filename, "aec_delay%d.dat", instance_count);
+ aecpc->delayFile = fopen(filename, "wb");
+ instance_count++;
+ }
+#endif
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_Free(void *aecInst)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ WebRtc_FreeBuffer(aecpc->far_pre_buf);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_FreeBuffer(aecpc->far_pre_buf_s16);
+ fclose(aecpc->aec->farFile);
+ fclose(aecpc->aec->nearFile);
+ fclose(aecpc->aec->outFile);
+ fclose(aecpc->aec->outLinearFile);
+ fclose(aecpc->bufFile);
+ fclose(aecpc->skewFile);
+ fclose(aecpc->delayFile);
+#endif
+
+ WebRtcAec_FreeAec(aecpc->aec);
+ WebRtcAec_FreeResampler(aecpc->resampler);
+ free(aecpc);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
+{
+ aecpc_t *aecpc = aecInst;
+ AecConfig aecConfig;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->sampFreq = sampFreq;
+
+ if (scSampFreq < 1 || scSampFreq > 96000) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->scSampFreq = scSampFreq;
+
+ // Initialize echo canceller core
+ if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+ if (WebRtc_InitBuffer(aecpc->far_pre_buf) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap.
+
+ aecpc->initFlag = initCheck; // indicates that initialization has been done
+
+ if (aecpc->sampFreq == 32000) {
+ aecpc->splitSampFreq = 16000;
+ }
+ else {
+ aecpc->splitSampFreq = sampFreq;
+ }
+
+ aecpc->skewFrCtr = 0;
+ aecpc->activity = 0;
+
+ aecpc->delayCtr = 0;
+
+ aecpc->sum = 0;
+ aecpc->counter = 0;
+ aecpc->checkBuffSize = 1;
+ aecpc->firstVal = 0;
+
+ aecpc->ECstartup = 1;
+ aecpc->bufSizeStart = 0;
+ aecpc->checkBufSizeCtr = 0;
+ aecpc->filtDelay = 0;
+ aecpc->timeForDelayChange = 0;
+ aecpc->knownDelay = 0;
+ aecpc->lastDelayDiff = 0;
+
+ aecpc->skew = 0;
+ aecpc->resample = kAecFalse;
+ aecpc->highSkewCtr = 0;
+ aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
+
+ // Default settings.
+ aecConfig.nlpMode = kAecNlpModerate;
+ aecConfig.skewMode = kAecFalse;
+ aecConfig.metricsMode = kAecFalse;
+ aecConfig.delay_logging = kAecFalse;
+
+ if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ if (WebRtc_InitBuffer(aecpc->far_pre_buf_s16) == -1) {
+ aecpc->lastError = AEC_UNSPECIFIED_ERROR;
+ return -1;
+ }
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN); // Start overlap.
+#endif
+
+ return 0;
+}
+
+// only buffer L band for farend
+WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
+ WebRtc_Word16 nrOfSamples)
+{
+ aecpc_t *aecpc = aecInst;
+ WebRtc_Word32 retVal = 0;
+ int newNrOfSamples = (int) nrOfSamples;
+ short newFarend[MAX_RESAMP_LEN];
+ const int16_t* farend_ptr = farend;
+ float tmp_farend[MAX_RESAMP_LEN];
+ const float* farend_float = tmp_farend;
+ float skew;
+ int i = 0;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (farend == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // number of samples == 160 for SWB input
+ if (nrOfSamples != 80 && nrOfSamples != 160) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ skew = aecpc->skew;
+
+ if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
+ // Resample and get a new number of samples
+ newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
+ farend,
+ nrOfSamples,
+ skew,
+ newFarend);
+ farend_ptr = (const int16_t*) newFarend;
+ }
+
+ aecpc->aec->system_delay += newNrOfSamples;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_WriteBuffer(aecpc->far_pre_buf_s16, farend_ptr,
+ (size_t) newNrOfSamples);
+#endif
+ // Cast to float and write the time-domain data to |far_pre_buf|.
+ for (i = 0; i < newNrOfSamples; i++) {
+ tmp_farend[i] = (float) farend_ptr[i];
+ }
+ WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_float,
+ (size_t) newNrOfSamples);
+
+ // Transform to frequency domain if we have enough data.
+ while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
+ // We have enough data to pass to the FFT, hence read PART_LEN2 samples.
+ WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**) &farend_float, tmp_farend,
+ PART_LEN2);
+
+ WebRtcAec_BufferFarendPartition(aecpc->aec, farend_float);
+
+ // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_ReadBuffer(aecpc->far_pre_buf_s16, (void**) &farend_ptr, newFarend,
+ PART_LEN2);
+ WebRtc_WriteBuffer(aecpc->aec->far_time_buf, &farend_ptr[PART_LEN], 1);
+ WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN);
+#endif
+ }
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
+ const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
+ WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
+{
+ aecpc_t *aecpc = aecInst;
+ WebRtc_Word32 retVal = 0;
+ short i;
+ short nBlocks10ms;
+ short nFrames;
+ // Limit resampling to doubling/halving of signal
+ const float minSkewEst = -0.5f;
+ const float maxSkewEst = 1.0f;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (nearend == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (out == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // number of samples == 160 for SWB input
+ if (nrOfSamples != 80 && nrOfSamples != 160) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+
+ // Check for valid pointers based on sampling rate
+ if (aecpc->sampFreq == 32000 && nearendH == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (msInSndCardBuf < 0) {
+ msInSndCardBuf = 0;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ }
+ else if (msInSndCardBuf > 500) {
+ msInSndCardBuf = 500;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ retVal = -1;
+ }
+ // TODO(andrew): we need to investigate if this +10 is really wanted.
+ msInSndCardBuf += 10;
+ aecpc->msInSndCardBuf = msInSndCardBuf;
+
+ if (aecpc->skewMode == kAecTrue) {
+ if (aecpc->skewFrCtr < 25) {
+ aecpc->skewFrCtr++;
+ }
+ else {
+ retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
+ if (retVal == -1) {
+ aecpc->skew = 0;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ }
+
+ aecpc->skew /= aecpc->sampFactor*nrOfSamples;
+
+ if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
+ aecpc->resample = kAecFalse;
+ }
+ else {
+ aecpc->resample = kAecTrue;
+ }
+
+ if (aecpc->skew < minSkewEst) {
+ aecpc->skew = minSkewEst;
+ }
+ else if (aecpc->skew > maxSkewEst) {
+ aecpc->skew = maxSkewEst;
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
+#endif
+ }
+ }
+
+ nFrames = nrOfSamples / FRAME_LEN;
+ nBlocks10ms = nFrames / aecpc->aec->mult;
+
+ if (aecpc->ECstartup) {
+ if (nearend != out) {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, nearend, sizeof(short) * nrOfSamples);
+ }
+
+ // The AEC is in the start up mode
+ // AEC is disabled until the system delay is OK
+
+ // Mechanism to ensure that the system delay is reasonably stable.
+ if (aecpc->checkBuffSize) {
+ aecpc->checkBufSizeCtr++;
+ // Before we fill up the far-end buffer we require the system delay
+ // to be stable (+/-8 ms) compared to the first value. This
+ // comparison is made during the following 6 consecutive 10 ms
+ // blocks. If it seems to be stable then we start to fill up the
+ // far-end buffer.
+ if (aecpc->counter == 0) {
+ aecpc->firstVal = aecpc->msInSndCardBuf;
+ aecpc->sum = 0;
+ }
+
+ if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
+ WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
+ aecpc->sum += aecpc->msInSndCardBuf;
+ aecpc->counter++;
+ }
+ else {
+ aecpc->counter = 0;
+ }
+
+ if (aecpc->counter * nBlocks10ms >= 6) {
+ // The far-end buffer size is determined in partitions of
+ // PART_LEN samples. Use 75% of the average value of the system
+ // delay as buffer size to start with.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
+ aecpc->aec->mult * 8) / (4 * aecpc->counter * PART_LEN),
+ kMaxBufSizeStart);
+ // Buffer size has now been determined.
+ aecpc->checkBuffSize = 0;
+ }
+
+ if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
+ // For really bad systems, don't disable the echo canceller for
+ // more than 0.5 sec.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
+ aecpc->aec->mult * 3) / 40, kMaxBufSizeStart);
+ aecpc->checkBuffSize = 0;
+ }
+ }
+
+ // If |checkBuffSize| changed in the if-statement above.
+ if (!aecpc->checkBuffSize) {
+ // The system delay is now reasonably stable (or has been unstable
+ // for too long). When the far-end buffer is filled with
+ // approximately the same amount of data as reported by the system
+ // we end the startup phase.
+ int overhead_elements = aecpc->aec->system_delay / PART_LEN -
+ aecpc->bufSizeStart;
+ if (overhead_elements == 0) {
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ } else if (overhead_elements > 0) {
+ WebRtc_MoveReadPtr(aecpc->aec->far_buf_windowed,
+ overhead_elements);
+ WebRtc_MoveReadPtr(aecpc->aec->far_buf, overhead_elements);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aecpc->aec->far_time_buf, overhead_elements);
+#endif
+ // TODO(bjornv): Do we need a check on how much we actually
+ // moved the read pointer? It should always be possible to move
+ // the pointer |overhead_elements| since we have only added data
+ // to the buffer and no delay compensation nor AEC processing
+ // has been done.
+ aecpc->aec->system_delay -= overhead_elements * PART_LEN;
+
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ }
+ }
+ } else {
+ // AEC is enabled.
+
+ int out_elements = 0;
+
+ EstBufDelay(aecpc);
+
+ // Note that 1 frame is supported for NB and 2 frames for WB.
+ for (i = 0; i < nFrames; i++) {
+ int16_t* out_ptr = NULL;
+ int16_t out_tmp[FRAME_LEN];
+
+ // Call the AEC.
+ WebRtcAec_ProcessFrame(aecpc->aec,
+ &nearend[FRAME_LEN * i],
+ &nearendH[FRAME_LEN * i],
+ aecpc->knownDelay);
+ // TODO(bjornv): Re-structure such that we don't have to pass
+ // |aecpc->knownDelay| as input. Change name to something like
+ // |system_buffer_diff|.
+
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ out_elements = (int) WebRtc_available_read(aecpc->aec->outFrBuf);
+ if (out_elements < FRAME_LEN) {
+ WebRtc_MoveReadPtr(aecpc->aec->outFrBuf,
+ out_elements - FRAME_LEN);
+ if (aecpc->sampFreq == 32000) {
+ WebRtc_MoveReadPtr(aecpc->aec->outFrBufH,
+ out_elements - FRAME_LEN);
+ }
+ }
+
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aecpc->aec->outFrBuf, (void**) &out_ptr,
+ out_tmp, FRAME_LEN);
+ memcpy(&out[FRAME_LEN * i], out_ptr, sizeof(int16_t) * FRAME_LEN);
+ // For H band
+ if (aecpc->sampFreq == 32000) {
+ WebRtc_ReadBuffer(aecpc->aec->outFrBufH, (void**) &out_ptr,
+ out_tmp, FRAME_LEN);
+ memcpy(&outH[FRAME_LEN * i], out_ptr,
+ sizeof(int16_t) * FRAME_LEN);
+ }
+ }
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ int16_t far_buf_size_ms = (int16_t) (aecpc->aec->system_delay /
+ (sampMsNb * aecpc->aec->mult));
+ fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
+ fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
+ }
+#endif
+
+ return retVal;
+}
+
+WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->skewMode = config.skewMode;
+
+ if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
+ kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->nlpMode = config.nlpMode;
+ aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
+ aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
+
+ if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->aec->metricsMode = config.metricsMode;
+ if (aecpc->aec->metricsMode == kAecTrue) {
+ WebRtcAec_InitMetrics(aecpc->aec);
+ }
+
+ if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
+ aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ aecpc->aec->delay_logging_enabled = config.delay_logging;
+ if (aecpc->aec->delay_logging_enabled == kAecTrue) {
+ memset(aecpc->aec->delay_histogram, 0, sizeof(aecpc->aec->delay_histogram));
+ }
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (config == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ config->nlpMode = aecpc->nlpMode;
+ config->skewMode = aecpc->skewMode;
+ config->metricsMode = aecpc->aec->metricsMode;
+ config->delay_logging = aecpc->aec->delay_logging_enabled;
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (status == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ *status = aecpc->aec->echoState;
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
+{
+ const float upweight = 0.7f;
+ float dtmp;
+ short stmp;
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ if (metrics == NULL) {
+ aecpc->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+
+ if (aecpc->initFlag != initCheck) {
+ aecpc->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ // ERL
+ metrics->erl.instant = (short) aecpc->aec->erl.instant;
+
+ if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
+ metrics->erl.average = (short) dtmp;
+ }
+ else {
+ metrics->erl.average = offsetLevel;
+ }
+
+ metrics->erl.max = (short) aecpc->aec->erl.max;
+
+ if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
+ metrics->erl.min = (short) aecpc->aec->erl.min;
+ }
+ else {
+ metrics->erl.min = offsetLevel;
+ }
+
+ // ERLE
+ metrics->erle.instant = (short) aecpc->aec->erle.instant;
+
+ if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
+ metrics->erle.average = (short) dtmp;
+ }
+ else {
+ metrics->erle.average = offsetLevel;
+ }
+
+ metrics->erle.max = (short) aecpc->aec->erle.max;
+
+ if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
+ metrics->erle.min = (short) aecpc->aec->erle.min;
+ } else {
+ metrics->erle.min = offsetLevel;
+ }
+
+ // RERL
+ if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
+ stmp = metrics->erl.average + metrics->erle.average;
+ }
+ else {
+ stmp = offsetLevel;
+ }
+ metrics->rerl.average = stmp;
+
+ // No other statistics needed, but returned for completeness
+ metrics->rerl.instant = stmp;
+ metrics->rerl.max = stmp;
+ metrics->rerl.min = stmp;
+
+ // A_NLP
+ metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
+
+ if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
+ // Use a mix between regular average and upper part average
+ dtmp = upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
+ metrics->aNlp.average = (short) dtmp;
+ }
+ else {
+ metrics->aNlp.average = offsetLevel;
+ }
+
+ metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
+
+ if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
+ metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
+ }
+ else {
+ metrics->aNlp.min = offsetLevel;
+ }
+
+ return 0;
+}
+
+int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
+ aecpc_t* self = handle;
+ int i = 0;
+ int delay_values = 0;
+ int num_delay_values = 0;
+ int my_median = 0;
+ const int kMsPerBlock = (PART_LEN * 1000) / self->splitSampFreq;
+ float l1_norm = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (median == NULL) {
+ self->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (std == NULL) {
+ self->lastError = AEC_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (self->initFlag != initCheck) {
+ self->lastError = AEC_UNINITIALIZED_ERROR;
+ return -1;
+ }
+ if (self->aec->delay_logging_enabled == 0) {
+ // Logging disabled
+ self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR;
+ return -1;
+ }
+
+ // Get number of delay values since last update
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ num_delay_values += self->aec->delay_histogram[i];
+ }
+ if (num_delay_values == 0) {
+ // We have no new delay value data. Even though -1 is a valid estimate, it
+ // will practically never be used since multiples of |kMsPerBlock| will
+ // always be returned.
+ *median = -1;
+ *std = -1;
+ return 0;
+ }
+
+ delay_values = num_delay_values >> 1; // Start value for median count down
+ // Get median of delay values since last update
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ delay_values -= self->aec->delay_histogram[i];
+ if (delay_values < 0) {
+ my_median = i;
+ break;
+ }
+ }
+ // Account for lookahead.
+ *median = (my_median - kLookaheadBlocks) * kMsPerBlock;
+
+ // Calculate the L1 norm, with median value as central moment
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ l1_norm += (float) (fabs(i - my_median) * self->aec->delay_histogram[i]);
+ }
+ *std = (int) (l1_norm / (float) num_delay_values + 0.5f) * kMsPerBlock;
+
+ // Reset histogram
+ memset(self->aec->delay_histogram, 0, sizeof(self->aec->delay_histogram));
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
+{
+ const char version[] = "AEC 2.5.0";
+ const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
+
+ if (versionStr == NULL) {
+ return -1;
+ }
+
+ if (versionLen > len) {
+ return -1;
+ }
+
+ strncpy(versionStr, version, versionLen);
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
+{
+ aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
+ return aecpc->lastError;
+}
+
+static int EstBufDelay(aecpc_t* aecpc) {
+ int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
+ int current_delay = nSampSndCard - aecpc->aec->system_delay;
+ int delay_difference = 0;
+
+ // Before we proceed with the delay estimate filtering we:
+ // 1) Compensate for the frame that will be read.
+ // 2) Compensate for drift resampling.
+
+ // 1) Compensating for the frame(s) that will be read/processed.
+ current_delay += FRAME_LEN * aecpc->aec->mult;
+
+ // 2) Account for resampling frame delay.
+ if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
+ current_delay -= kResamplingDelay;
+ }
+
+ aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
+ 0.2 * current_delay));
+
+ delay_difference = aecpc->filtDelay - aecpc->knownDelay;
+ if (delay_difference > 224) {
+ if (aecpc->lastDelayDiff < 96) {
+ aecpc->timeForDelayChange = 0;
+ } else {
+ aecpc->timeForDelayChange++;
+ }
+ } else if (delay_difference < 96 && aecpc->knownDelay > 0) {
+ if (aecpc->lastDelayDiff > 224) {
+ aecpc->timeForDelayChange = 0;
+ } else {
+ aecpc->timeForDelayChange++;
+ }
+ } else {
+ aecpc->timeForDelayChange = 0;
+ }
+ aecpc->lastDelayDiff = delay_difference;
+
+ if (aecpc->timeForDelayChange > 25) {
+ aecpc->knownDelay = WEBRTC_SPL_MAX((int) aecpc->filtDelay - 160, 0);
+ }
+
+ return 0;
+}
diff --git a/src/modules/audio_processing/aec/main/interface/echo_cancellation.h b/src/modules/audio_processing/aec/interface/echo_cancellation.h
similarity index 92%
rename from src/modules/audio_processing/aec/main/interface/echo_cancellation.h
rename to src/modules/audio_processing/aec/interface/echo_cancellation.h
index 883357d..4da6e73 100644
--- a/src/modules/audio_processing/aec/main/interface/echo_cancellation.h
+++ b/src/modules/audio_processing/aec/interface/echo_cancellation.h
@@ -38,6 +38,7 @@
WebRtc_Word16 nlpMode; // default kAecNlpModerate
WebRtc_Word16 skewMode; // default kAecFalse
WebRtc_Word16 metricsMode; // default kAecFalse
+ int delay_logging; // default kAecFalse
//float realSkew;
} AecConfig;
@@ -66,7 +67,7 @@
* Inputs Description
* -------------------------------------------------------------------
* void **aecInst Pointer to the AEC instance to be created
- * and initilized
+ * and initialized
*
* Outputs Description
* -------------------------------------------------------------------
@@ -226,6 +227,23 @@
WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
/*
+ * Gets the current delay metrics for the session.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* handle Pointer to the AEC instance
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * int* median Delay median value.
+ * int* std Delay standard deviation.
+ *
+ * int return 0: OK
+ * -1: error
+ */
+int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std);
+
+/*
* Gets the last error code.
*
* Inputs Description
diff --git a/src/modules/audio_processing/aec/main/matlab/fullaec.m b/src/modules/audio_processing/aec/main/matlab/fullaec.m
deleted file mode 100644
index 0f86a8c..0000000
--- a/src/modules/audio_processing/aec/main/matlab/fullaec.m
+++ /dev/null
@@ -1,953 +0,0 @@
-% Partitioned block frequency domain adaptive filtering NLMS and
-% standard time-domain sample-based NLMS
-%fid=fopen('aecFar-samsung.pcm', 'rb'); % Load far end
-fid=fopen('aecFar.pcm', 'rb'); % Load far end
-%fid=fopen(farFile, 'rb'); % Load far end
-rrin=fread(fid,inf,'int16');
-fclose(fid);
-%rrin=loadsl('data/far_me2.pcm'); % Load far end
-%fid=fopen('aecNear-samsung.pcm', 'rb'); % Load near end
-fid=fopen('aecNear.pcm', 'rb'); % Load near end
-%fid=fopen(nearFile, 'rb'); % Load near end
-ssin=fread(fid,inf,'int16');
-%ssin = [zeros(1024,1) ; ssin(1:end-1024)];
-
-fclose(fid);
-rand('state',13);
-fs=16000;
-mult=fs/8000;
-%rrin=rrin(fs*0+1:round(fs*120));
-%ssin=ssin(fs*0+1:round(fs*120));
-if fs == 8000
- cohRange = 2:3;
-elseif fs==16000
- cohRange = 2;
-end
-
-% Flags
-NLPon=1; % NLP
-CNon=1; % Comfort noise
-PLTon=1; % Plotting
-
-M = 16; % Number of partitions
-N = 64; % Partition length
-L = M*N; % Filter length
-if fs == 8000
- mufb = 0.6;
-else
- mufb = 0.5;
-end
-%mufb=1;
-VADtd=48;
-alp = 0.1; % Power estimation factor alc = 0.1; % Coherence estimation factor
-beta = 0.9; % Plotting factor
-%% Changed a little %%
-step = 0.3;%0.1875; % Downward step size
-%%
-if fs == 8000
- threshold=2e-6; % DTrob threshold
-else
- %threshold=0.7e-6;
- threshold=1.5e-6; end
-
-if fs == 8000
- echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
- %echoBandRange = ceil(1500*2/fs*N):floor(2500*2/fs*N);
-else
- echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
- %echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
-end
-%echoBandRange = ceil(1600*2/fs*N):floor(1900*2/fs*N);
-%echoBandRange = ceil(2000*2/fs*N):floor(4000*2/fs*N);
-suppState = 1;
-transCtr = 0;
-
-Nt=1;
-vt=1;
-
-ramp = 1.0003; % Upward ramp
-rampd = 0.999; % Downward ramp
-cvt = 20; % Subband VAD threshold;
-nnthres = 20; % Noise threshold
-
-shh=logspace(-1.3,-2.2,N+1)';
-sh=[shh;flipud(shh(2:end-1))]; % Suppression profile
-
-len=length(ssin);
-w=zeros(L,1); % Sample-based TD NLMS
-WFb=zeros(N+1,M); % Block-based FD NLMS
-WFbOld=zeros(N+1,M); % Block-based FD NLMS
-YFb=zeros(N+1,M);
-erfb=zeros(len,1);
-erfb3=zeros(len,1);
-
-ercn=zeros(len,1);
-zm=zeros(N,1);
-XFm=zeros(N+1,M);
-YFm=zeros(N+1,M);
-pn0=10*ones(N+1,1);
-pn=zeros(N+1,1);
-NN=len;
-Nb=floor(NN/N)-M;
-erifb=zeros(Nb+1,1)+0.1;
-erifb3=zeros(Nb+1,1)+0.1;
-ericn=zeros(Nb+1,1)+0.1;
-dri=zeros(Nb+1,1)+0.1;
-start=1;
-xo=zeros(N,1);
-do=xo;
-eo=xo;
-
-echoBands=zeros(Nb+1,1);
-cohxdAvg=zeros(Nb+1,1);
-cohxdSlow=zeros(Nb+1,N+1);
-cohedSlow=zeros(Nb+1,N+1);
-%overdriveM=zeros(Nb+1,N+1);
-cohxdFastAvg=zeros(Nb+1,1);
-cohxdAvgBad=zeros(Nb+1,1);
-cohedAvg=zeros(Nb+1,1);
-cohedFastAvg=zeros(Nb+1,1);
-hnledAvg=zeros(Nb+1,1);
-hnlxdAvg=zeros(Nb+1,1);
-ovrdV=zeros(Nb+1,1);
-dIdxV=zeros(Nb+1,1);
-SLxV=zeros(Nb+1,1);
-hnlSortQV=zeros(Nb+1,1);
-hnlPrefAvgV=zeros(Nb+1,1);
-mutInfAvg=zeros(Nb+1,1);
-%overdrive=zeros(Nb+1,1);
-hnled = zeros(N+1, 1);
-weight=zeros(N+1,1);
-hnlMax = zeros(N+1, 1);
-hnl = zeros(N+1, 1);
-overdrive = ones(1, N+1);
-xfwm=zeros(N+1,M);
-dfm=zeros(N+1,M);
-WFbD=ones(N+1,1);
-
-fbSupp = 0;
-hnlLocalMin = 1;
-cohxdLocalMin = 1;
-hnlLocalMinV=zeros(Nb+1,1);
-cohxdLocalMinV=zeros(Nb+1,1);
-hnlMinV=zeros(Nb+1,1);
-dkEnV=zeros(Nb+1,1);
-ekEnV=zeros(Nb+1,1);
-ovrd = 2;
-ovrdPos = floor((N+1)/4);
-ovrdSm = 2;
-hnlMin = 1;
-minCtr = 0;
-SeMin = 0;
-SdMin = 0;
-SeLocalAvg = 0;
-SeMinSm = 0;
-divergeFact = 1;
-dIdx = 1;
-hnlMinCtr = 0;
-hnlNewMin = 0;
-divergeState = 0;
-
-Sy=ones(N+1,1);
-Sym=1e7*ones(N+1,1);
-
-wins=[0;sqrt(hanning(2*N-1))];
-ubufn=zeros(2*N,1);
-ebuf=zeros(2*N,1);
-ebuf2=zeros(2*N,1);
-ebuf4=zeros(2*N,1);
-mbuf=zeros(2*N,1);
-
-cohedFast = zeros(N+1,1);
-cohxdFast = zeros(N+1,1);
-cohxd = zeros(N+1,1);
-Se = zeros(N+1,1);
-Sd = zeros(N+1,1);
-Sx = zeros(N+1,1);
-SxBad = zeros(N+1,1);
-Sed = zeros(N+1,1);
-Sxd = zeros(N+1,1);
-SxdBad = zeros(N+1,1);
-hnledp=[];
-
-cohxdMax = 0;
-
-%hh=waitbar(0,'Please wait...');
-progressbar(0);
-
-%spaces = ' ';
-%spaces = repmat(spaces, 50, 1);
-%spaces = ['[' ; spaces ; ']'];
-%fprintf(1, spaces);
-%fprintf(1, '\n');
-
-for kk=1:Nb
- pos = N * (kk-1) + start;
-
- % FD block method
- % ---------------------- Organize data
- xk = rrin(pos:pos+N-1);
- dk = ssin(pos:pos+N-1);
-
- xx = [xo;xk];
- xo = xk;
- tmp = fft(xx);
- XX = tmp(1:N+1);
-
- dd = [do;dk]; % Overlap
- do = dk;
- tmp = fft(dd); % Frequency domain
- DD = tmp(1:N+1);
-
- % ------------------------ Power estimation
- pn0 = (1 - alp) * pn0 + alp * real(XX.* conj(XX));
- pn = pn0;
- %pn = (1 - alp) * pn + alp * M * pn0;
- if (CNon)
- Yp = real(conj(DD).*DD); % Instantaneous power
- Sy = (1 - alp) * Sy + alp * Yp; % Averaged power
-
- mm = min(Sy,Sym);
- diff = Sym - mm;
- if (kk>50)
- Sym = (mm + step*diff) * ramp; % Estimated background noise power
- end
- end
-
- % ---------------------- Filtering
- XFm(:,1) = XX;
- for mm=0:(M-1)
- m=mm+1;
- YFb(:,m) = XFm(:,m) .* WFb(:,m);
- end
- yfk = sum(YFb,2);
- tmp = [yfk ; flipud(conj(yfk(2:N)))];
- ykt = real(ifft(tmp));
- ykfb = ykt(end-N+1:end);
-
- % ---------------------- Error estimation
- ekfb = dk - ykfb;
- %if sum(abs(ekfb)) < sum(abs(dk))
- %ekfb = dk - ykfb;
- % erfb(pos:pos+N-1) = ekfb;
- %else
- %ekfb = dk;
- % erfb(pos:pos+N-1) = dk;
- %end
- %(kk-1)*(N*2)+1
- erfb(pos:pos+N-1) = ekfb;
- tmp = fft([zm;ekfb]); % FD version for cancelling part (overlap-save)
- Ek = tmp(1:N+1);
-
- % ------------------------ Adaptation
- Ek2 = Ek ./(M*pn + 0.001); % Normalized error
- %Ek2 = Ek ./(pn + 0.001); % Normalized error
- %Ek2 = Ek ./(100*pn + 0.001); % Normalized error
-
- absEf = max(abs(Ek2), threshold);
- absEf = ones(N+1,1)*threshold./absEf;
- Ek2 = Ek2.*absEf;
-
- mEk = mufb.*Ek2;
- PP = conj(XFm).*(ones(M,1) * mEk')';
- tmp = [PP ; flipud(conj(PP(2:N,:)))];
- IFPP = real(ifft(tmp));
- PH = IFPP(1:N,:);
- tmp = fft([PH;zeros(N,M)]);
- FPH = tmp(1:N+1,:);
- WFb = WFb + FPH;
-
- if mod(kk, 10*mult) == 0
- WFbEn = sum(real(WFb.*conj(WFb)));
- %WFbEn = sum(abs(WFb));
- [tmp, dIdx] = max(WFbEn);
-
- WFbD = sum(abs(WFb(:, dIdx)),2);
- %WFbD = WFbD / (mean(WFbD) + 1e-10);
- WFbD = min(max(WFbD, 0.5), 4);
- end
- dIdxV(kk) = dIdx;
-
- % NLP
- if (NLPon)
-
- ee = [eo;ekfb];
- eo = ekfb;
- window = wins;
- if fs == 8000
- %gamma = 0.88;
- gamma = 0.9;
- else
- %gamma = 0.92;
- gamma = 0.93;
- end
- %gamma = 0.9;
-
- tmp = fft(xx.*window);
- xf = tmp(1:N+1);
- tmp = fft(dd.*window);
- df = tmp(1:N+1);
- tmp = fft(ee.*window);
- ef = tmp(1:N+1);
-
- xfwm(:,1) = xf;
- xf = xfwm(:,dIdx);
- %fprintf(1,'%d: %f\n', kk, xf(4));
- dfm(:,1) = df;
-
- SxOld = Sx;
-
- Se = gamma*Se + (1-gamma)*real(ef.*conj(ef));
- Sd = gamma*Sd + (1-gamma)*real(df.*conj(df));
- Sx = gamma*Sx + (1 - gamma)*real(xf.*conj(xf));
-
- %xRatio = real(xfwm(:,1).*conj(xfwm(:,1))) ./ ...
- % (real(xfwm(:,2).*conj(xfwm(:,2))) + 1e-10);
- %xRatio = Sx ./ (SxOld + 1e-10);
- %SLx = log(1/(N+1)*sum(xRatio)) - 1/(N+1)*sum(log(xRatio));
- %SLxV(kk) = SLx;
-
- %freqSm = 0.9;
- %Sx = filter(freqSm, [1 -(1-freqSm)], Sx);
- %Sx(end:1) = filter(freqSm, [1 -(1-freqSm)], Sx(end:1));
- %Se = filter(freqSm, [1 -(1-freqSm)], Se);
- %Se(end:1) = filter(freqSm, [1 -(1-freqSm)], Se(end:1));
- %Sd = filter(freqSm, [1 -(1-freqSm)], Sd);
- %Sd(end:1) = filter(freqSm, [1 -(1-freqSm)], Sd(end:1));
-
- %SeFast = ef.*conj(ef);
- %SdFast = df.*conj(df);
- %SxFast = xf.*conj(xf);
- %cohedFast = 0.9*cohedFast + 0.1*SeFast ./ (SdFast + 1e-10);
- %cohedFast(find(cohedFast > 1)) = 1;
- %cohedFast(find(cohedFast > 1)) = 1 ./ cohedFast(find(cohedFast>1));
- %cohedFastAvg(kk) = mean(cohedFast(echoBandRange));
- %cohedFastAvg(kk) = min(cohedFast);
-
- %cohxdFast = 0.8*cohxdFast + 0.2*log(SdFast ./ (SxFast + 1e-10));
- %cohxdFastAvg(kk) = mean(cohxdFast(echoBandRange));
-
- % coherence
- Sxd = gamma*Sxd + (1 - gamma)*xf.*conj(df);
- Sed = gamma*Sed + (1-gamma)*ef.*conj(df);
-
- %Sxd = filter(freqSm, [1 -(1-freqSm)], Sxd);
- %Sxd(end:1) = filter(freqSm, [1 -(1-freqSm)], Sxd(end:1));
- %Sed = filter(freqSm, [1 -(1-freqSm)], Sed);
- %Sed(end:1) = filter(freqSm, [1 -(1-freqSm)], Sed(end:1));
-
- cohed = real(Sed.*conj(Sed))./(Se.*Sd + 1e-10);
- %cohedAvg(kk) = mean(cohed(echoBandRange));
- %cohedAvg(kk) = cohed(6);
- %cohedAvg(kk) = min(cohed);
-
- cohxd = real(Sxd.*conj(Sxd))./(Sx.*Sd + 1e-10);
- %freqSm = 0.5;
- %cohxd(3:end) = filter(freqSm, [1 -(1-freqSm)], cohxd(3:end));
- %cohxd(end:3) = filter(freqSm, [1 -(1-freqSm)], cohxd(end:3));
- %cohxdAvg(kk) = mean(cohxd(echoBandRange));
- %cohxdAvg(kk) = (cohxd(32));
- %cohxdAvg(kk) = max(cohxd);
-
- %xf = xfm(:,dIdx);
- %SxBad = gamma*SxBad + (1 - gamma)*real(xf.*conj(xf));
- %SxdBad = gamma*SxdBad + (1 - gamma)*xf.*conj(df);
- %cohxdBad = real(SxdBad.*conj(SxdBad))./(SxBad.*Sd + 0.01);
- %cohxdAvgBad(kk) = mean(cohxdBad);
-
- %for j=1:N+1
- % mutInf(j) = 0.9*mutInf(j) + 0.1*information(abs(xfm(j,:)), abs(dfm(j,:)));
- %end
- %mutInfAvg(kk) = mean(mutInf);
-
- %hnled = cohedFast;
- %xIdx = find(cohxd > 1 - cohed);
- %hnled(xIdx) = 1 - cohxd(xIdx);
- %hnled = 1 - max(cohxd, 1-cohedFast);
- hnled = min(1 - cohxd, cohed);
- %hnled = 1 - cohxd;
- %hnled = max(1 - (cohxd + (1-cohedFast)), 0);
- %hnled = 1 - max(cohxd, 1-cohed);
-
- if kk > 1
- cohxdSlow(kk,:) = 0.99*cohxdSlow(kk-1,:) + 0.01*cohxd';
- cohedSlow(kk,:) = 0.99*cohedSlow(kk-1,:) + 0.01*(1-cohed)';
- end
-
-
- if 0
- %if kk > 50
- %idx = find(hnled > 0.3);
- hnlMax = hnlMax*0.9999;
- %hnlMax(idx) = max(hnlMax(idx), hnled(idx));
- hnlMax = max(hnlMax, hnled);
- %overdrive(idx) = max(log(hnlMax(idx))/log(0.99), 1);
- avgHnl = mean(hnlMax(echoBandRange));
- if avgHnl > 0.3
- overdrive = max(log(avgHnl)/log(0.99), 1);
- end
- weight(4:end) = max(hnlMax) - hnlMax(4:end);
- end
-
-
-
- %[hg, gidx] = max(hnled);
- %fnrg = Sx(gidx) / (Sd(gidx) + 1e-10);
-
- %[tmp, bidx] = find((Sx / Sd + 1e-10) > fnrg);
- %hnled(bidx) = hg;
-
-
- %cohed1 = mean(cohed(cohRange)); % range depends on bandwidth
- %cohed1 = cohed1^2;
- %echoBands(kk) = length(find(cohed(echoBandRange) < 0.25))/length(echoBandRange);
-
- %if (fbSupp == 0)
- % if (echoBands(kk) > 0.8)
- % fbSupp = 1;
- % end
- %else
- % if (echoBands(kk) < 0.6)
- % fbSupp = 0;
- % end
- %end
- %overdrive(kk) = 7.5*echoBands(kk) + 0.5;
-
- % Factor by which to weight other bands
- %if (cohed1 < 0.1)
- % w = 0.8 - cohed1*10*0.4;
- %else
- % w = 0.4;
- %end
-
- % Weight coherence subbands
- %hnled = w*cohed1 + (1 - w)*cohed;
- %hnled = (hnled).^2;
- %cohed(floor(N/2):end) = cohed(floor(N/2):end).^2;
- %if fbSupp == 1
- % cohed = zeros(size(cohed));
- %end
- %cohed = cohed.^overdrive(kk);
-
- %hnled = gamma*hnled + (1 - gamma)*cohed;
- % Additional hf suppression
- %hnledp = [hnledp ; mean(hnled)];
- %hnled(floor(N/2):end) = hnled(floor(N/2):end).^2;
- %ef = ef.*((weight*(min(1 - hnled)).^2 + (1 - weight).*(1 - hnled)).^2);
-
- cohedMean = mean(cohed(echoBandRange));
- %aggrFact = 4*(1-mean(hnled(echoBandRange))) + 1;
- %[hnlSort, hnlSortIdx] = sort(hnled(echoBandRange));
- [hnlSort, hnlSortIdx] = sort(1-cohxd(echoBandRange));
- [xSort, xSortIdx] = sort(Sx);
- %aggrFact = (1-mean(hnled(echoBandRange)));
- %hnlSortQ = hnlSort(qIdx);
- hnlSortQ = mean(1 - cohxd(echoBandRange));
- %hnlSortQ = mean(1 - cohxd);
-
- [hnlSort2, hnlSortIdx2] = sort(hnled(echoBandRange));
- %[hnlSort2, hnlSortIdx2] = sort(hnled);
- hnlQuant = 0.75;
- hnlQuantLow = 0.5;
- qIdx = floor(hnlQuant*length(hnlSort2));
- qIdxLow = floor(hnlQuantLow*length(hnlSort2));
- hnlPrefAvg = hnlSort2(qIdx);
- hnlPrefAvgLow = hnlSort2(qIdxLow);
- %hnlPrefAvgLow = mean(hnled);
- %hnlPrefAvg = max(hnlSort2);
- %hnlPrefAvgLow = min(hnlSort2);
-
- %hnlPref = hnled(echoBandRange);
- %hnlPrefAvg = mean(hnlPref(xSortIdx((0.5*length(xSortIdx)):end)));
-
- %hnlPrefAvg = min(hnlPrefAvg, hnlSortQ);
-
- %hnlSortQIdx = hnlSortIdx(qIdx);
- %SeQ = Se(qIdx + echoBandRange(1) - 1);
- %SdQ = Sd(qIdx + echoBandRange(1) - 1);
- %SeQ = Se(qIdxLow + echoBandRange(1) - 1);
- %SdQ = Sd(qIdxLow + echoBandRange(1) - 1);
- %propLow = length(find(hnlSort < 0.1))/length(hnlSort);
- %aggrFact = min((1 - hnlSortQ)/2, 0.5);
- %aggrTerm = 1/aggrFact;
-
- %hnlg = mean(hnled(echoBandRange));
- %hnlg = hnlSortQ;
- %if suppState == 0
- % if hnlg < 0.05
- % suppState = 2;
- % transCtr = 0;
- % elseif hnlg < 0.75
- % suppState = 1;
- % transCtr = 0;
- % end
- %elseif suppState == 1
- % if hnlg > 0.8
- % suppState = 0;
- % transCtr = 0;
- % elseif hnlg < 0.05
- % suppState = 2;
- % transCtr = 0;
- % end
- %else
- % if hnlg > 0.8
- % suppState = 0;
- % transCtr = 0;
- % elseif hnlg > 0.25
- % suppState = 1;
- % transCtr = 0;
- % end
- %end
- %if kk > 50
-
- if cohedMean > 0.98 & hnlSortQ > 0.9
- %if suppState == 1
- % hnled = 0.5*hnled + 0.5*cohed;
- % %hnlSortQ = 0.5*hnlSortQ + 0.5*cohedMean;
- % hnlPrefAvg = 0.5*hnlPrefAvg + 0.5*cohedMean;
- %else
- % hnled = cohed;
- % %hnlSortQ = cohedMean;
- % hnlPrefAvg = cohedMean;
- %end
- suppState = 0;
- elseif cohedMean < 0.95 | hnlSortQ < 0.8
- %if suppState == 0
- % hnled = 0.5*hnled + 0.5*cohed;
- % %hnlSortQ = 0.5*hnlSortQ + 0.5*cohedMean;
- % hnlPrefAvg = 0.5*hnlPrefAvg + 0.5*cohedMean;
- %end
- suppState = 1;
- end
-
- if hnlSortQ < cohxdLocalMin & hnlSortQ < 0.75
- cohxdLocalMin = hnlSortQ;
- end
-
- if cohxdLocalMin == 1
- ovrd = 3;
- hnled = 1-cohxd;
- hnlPrefAvg = hnlSortQ;
- hnlPrefAvgLow = hnlSortQ;
- end
-
- if suppState == 0
- hnled = cohed;
- hnlPrefAvg = cohedMean;
- hnlPrefAvgLow = cohedMean;
- end
-
- %if hnlPrefAvg < hnlLocalMin & hnlPrefAvg < 0.6
- if hnlPrefAvgLow < hnlLocalMin & hnlPrefAvgLow < 0.6
- %hnlLocalMin = hnlPrefAvg;
- %hnlMin = hnlPrefAvg;
- hnlLocalMin = hnlPrefAvgLow;
- hnlMin = hnlPrefAvgLow;
- hnlNewMin = 1;
- hnlMinCtr = 0;
- %if hnlMinCtr == 0
- % hnlMinCtr = hnlMinCtr + 1;
- %else
- % hnlMinCtr = 0;
- % hnlMin = hnlLocalMin;
- %SeLocalMin = SeQ;
- %SdLocalMin = SdQ;
- %SeLocalAvg = 0;
- %minCtr = 0;
- % ovrd = max(log(0.0001)/log(hnlMin), 2);
- %divergeFact = hnlLocalMin;
- end
-
- if hnlNewMin == 1
- hnlMinCtr = hnlMinCtr + 1;
- end
- if hnlMinCtr == 2
- hnlNewMin = 0;
- hnlMinCtr = 0;
- %ovrd = max(log(0.0001)/log(hnlMin), 2);
- ovrd = max(log(0.00001)/(log(hnlMin + 1e-10) + 1e-10), 3);
- %ovrd = max(log(0.00000001)/(log(hnlMin + 1e-10) + 1e-10), 5);
- %ovrd = max(log(0.0001)/log(hnlPrefAvg), 2);
- %ovrd = max(log(0.001)/log(hnlMin), 2);
- end
- hnlLocalMin = min(hnlLocalMin + 0.0008/mult, 1);
- cohxdLocalMin = min(cohxdLocalMin + 0.0004/mult, 1);
- %divergeFact = hnlSortQ;
-
-
- %if minCtr > 0 & hnlLocalMin < 1
- % hnlMin = hnlLocalMin;
- % %SeMin = 0.9*SeMin + 0.1*sqrt(SeLocalMin);
- % SdMin = sqrt(SdLocalMin);
- % %SeMin = sqrt(SeLocalMin)*hnlSortQ;
- % SeMin = sqrt(SeLocalMin);
- % %ovrd = log(100/SeMin)/log(hnlSortQ);
- % %ovrd = log(100/SeMin)/log(hnlSortQ);
- % ovrd = log(0.01)/log(hnlMin);
- % ovrd = max(ovrd, 2);
- % ovrdPos = hnlSortQIdx;
- % %ovrd = max(ovrd, 1);
- % %SeMin = sqrt(SeLocalAvg/5);
- % minCtr = 0;
- %else
- % %SeLocalMin = 0.9*SeLocalMin +0.1*SeQ;
- % SeLocalAvg = SeLocalAvg + SeQ;
- % minCtr = minCtr + 1;
- %end
-
- if ovrd < ovrdSm
- ovrdSm = 0.99*ovrdSm + 0.01*ovrd;
- else
- ovrdSm = 0.9*ovrdSm + 0.1*ovrd;
- end
- %end
-
- %ekEn = sum(real(ekfb.^2));
- %dkEn = sum(real(dk.^2));
- ekEn = sum(Se);
- dkEn = sum(Sd);
-
- if divergeState == 0
- if ekEn > dkEn
- ef = df;
- divergeState = 1;
- %hnlPrefAvg = hnlSortQ;
- %hnled = (1 - cohxd);
- end
- else
- %if ekEn*1.1 < dkEn
- %if ekEn*1.26 < dkEn
- if ekEn*1.05 < dkEn
- divergeState = 0;
- else
- ef = df;
- end
- end
-
- if ekEn > dkEn*19.95
- WFb=zeros(N+1,M); % Block-based FD NLMS
- end
-
- ekEnV(kk) = ekEn;
- dkEnV(kk) = dkEn;
-
- hnlLocalMinV(kk) = hnlLocalMin;
- cohxdLocalMinV(kk) = cohxdLocalMin;
- hnlMinV(kk) = hnlMin;
- %cohxdMaxLocal = max(cohxdSlow(kk,:));
- %if kk > 50
- %cohxdMaxLocal = 1-hnlSortQ;
- %if cohxdMaxLocal > 0.5
- % %if cohxdMaxLocal > cohxdMax
- % odScale = max(log(cohxdMaxLocal)/log(0.95), 1);
- % %overdrive(7:end) = max(log(cohxdSlow(kk,7:end))/log(0.9), 1);
- % cohxdMax = cohxdMaxLocal;
- % end
- %end
- %end
- %cohxdMax = cohxdMax*0.999;
-
- %overdriveM(kk,:) = max(overdrive, 1);
- %aggrFact = 0.25;
- aggrFact = 0.3;
- %aggrFact = 0.5*propLow;
- %if fs == 8000
- % wCurve = [0 ; 0 ; aggrFact*sqrt(linspace(0,1,N-1))' + 0.1];
- %else
- % wCurve = [0; 0; 0; aggrFact*sqrt(linspace(0,1,N-2))' + 0.1];
- %end
- wCurve = [0; aggrFact*sqrt(linspace(0,1,N))' + 0.1];
- % For sync with C
- %if fs == 8000
- % wCurve = wCurve(2:end);
- %else
- % wCurve = wCurve(1:end-1);
- %end
- %weight = aggrFact*(sqrt(linspace(0,1,N+1)'));
- %weight = aggrFact*wCurve;
- weight = wCurve;
- %weight = aggrFact*ones(N+1,1);
- %weight = zeros(N+1,1);
- %hnled = weight.*min(hnled) + (1 - weight).*hnled;
- %hnled = weight.*min(mean(hnled(echoBandRange)), hnled) + (1 - weight).*hnled;
- %hnled = weight.*min(hnlSortQ, hnled) + (1 - weight).*hnled;
-
- %hnlSortQV(kk) = mean(hnled);
- %hnlPrefAvgV(kk) = mean(hnled(echoBandRange));
-
- hnled = weight.*min(hnlPrefAvg, hnled) + (1 - weight).*hnled;
-
- %od = aggrFact*(sqrt(linspace(0,1,N+1)') + aggrTerm);
- %od = 4*(sqrt(linspace(0,1,N+1)') + 1/4);
-
- %ovrdFact = (ovrdSm - 1) / sqrt(ovrdPos/(N+1));
- %ovrdFact = ovrdSm / sqrt(echoBandRange(floor(length(echoBandRange)/2))/(N+1));
- %od = ovrdFact*sqrt(linspace(0,1,N+1))' + 1;
- %od = ovrdSm*ones(N+1,1).*abs(WFb(:,dIdx))/(max(abs(WFb(:,dIdx)))+1e-10);
-
- %od = ovrdSm*ones(N+1,1);
- %od = ovrdSm*WFbD.*(sqrt(linspace(0,1,N+1))' + 1);
-
- od = ovrdSm*(sqrt(linspace(0,1,N+1))' + 1);
- %od = 4*(sqrt(linspace(0,1,N+1))' + 1);
-
- %od = 2*ones(N+1,1);
- %od = 2*ones(N+1,1);
- %sshift = ((1-hnled)*2-1).^3+1;
- sshift = ones(N+1,1);
-
- hnled = hnled.^(od.*sshift);
-
- %if hnlg > 0.75
- %if (suppState ~= 0)
- % transCtr = 0;
- %end
- % suppState = 0;
- %elseif hnlg < 0.6 & hnlg > 0.2
- % suppState = 1;
- %elseif hnlg < 0.1
- %hnled = zeros(N+1, 1);
- %if (suppState ~= 2)
- % transCtr = 0;
- %end
- % suppState = 2;
- %else
- % if (suppState ~= 2)
- % transCtr = 0;
- % end
- % suppState = 2;
- %end
- %if suppState == 0
- % hnled = ones(N+1, 1);
- %elseif suppState == 2
- % hnled = zeros(N+1, 1);
- %end
- %hnled(find(hnled < 0.1)) = 0;
- %hnled = hnled.^2;
- %if transCtr < 5
- %hnl = 0.75*hnl + 0.25*hnled;
- % transCtr = transCtr + 1;
- %else
- hnl = hnled;
- %end
- %hnled(find(hnled < 0.05)) = 0;
- ef = ef.*(hnl);
-
- %ef = ef.*(min(1 - cohxd, cohed).^2);
- %ef = ef.*((1-cohxd).^2);
-
- ovrdV(kk) = ovrdSm;
- %ovrdV(kk) = dIdx;
- %ovrdV(kk) = divergeFact;
- %hnledAvg(kk) = 1-mean(1-cohedFast(echoBandRange));
- hnledAvg(kk) = 1-mean(1-cohed(echoBandRange));
- hnlxdAvg(kk) = 1-mean(cohxd(echoBandRange));
- %hnlxdAvg(kk) = cohxd(5);
- %hnlSortQV(kk) = mean(hnled);
- hnlSortQV(kk) = hnlPrefAvgLow;
- hnlPrefAvgV(kk) = hnlPrefAvg;
- %hnlAvg(kk) = propLow;
- %ef(N/2:end) = 0;
- %ner = (sum(Sd) ./ (sum(Se.*(hnl.^2)) + 1e-10));
-
- % Comfort noise
- if (CNon)
- snn=sqrt(Sym);
- snn(1)=0; % Reject LF noise
- Un=snn.*exp(j*2*pi.*[0;rand(N-1,1);0]);
-
- % Weight comfort noise by suppression
- Un = sqrt(1-hnled.^2).*Un;
- Fmix = ef + Un;
- else
- Fmix = ef;
- end
-
- % Overlap and add in time domain for smoothness
- tmp = [Fmix ; flipud(conj(Fmix(2:N)))];
- mixw = wins.*real(ifft(tmp));
- mola = mbuf(end-N+1:end) + mixw(1:N);
- mbuf = mixw;
- ercn(pos:pos+N-1) = mola;
- end % NLPon
-
- % Filter update
- %Ek2 = Ek ./(12*pn + 0.001); % Normalized error
- %Ek2 = Ek2 * divergeFact;
- %Ek2 = Ek ./(pn + 0.001); % Normalized error
- %Ek2 = Ek ./(100*pn + 0.001); % Normalized error
-
- %divergeIdx = find(abs(Ek) > abs(DD));
- %divergeIdx = find(Se > Sd);
- %threshMod = threshold*ones(N+1,1);
- %if length(divergeIdx) > 0
- %if sum(abs(Ek)) > sum(abs(DD))
- %WFb(divergeIdx,:) = WFb(divergeIdx,:) .* repmat(sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10))),1,M);
- %Ek2(divergeIdx) = Ek2(divergeIdx) .* sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10));
- %Ek2(divergeIdx) = Ek2(divergeIdx) .* abs(DD(divergeIdx))./(abs(Ek(divergeIdx))+1e-10);
- %WFb(divergeIdx,:) = WFbOld(divergeIdx,:);
- %WFb = WFbOld;
- %threshMod(divergeIdx) = threshMod(divergeIdx) .* abs(DD(divergeIdx))./(abs(Ek(divergeIdx))+1e-10);
- % threshMod(divergeIdx) = threshMod(divergeIdx) .* sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10));
- %end
-
- %absEf = max(abs(Ek2), threshold);
- %absEf = ones(N+1,1)*threshold./absEf;
- %absEf = max(abs(Ek2), threshMod);
- %absEf = threshMod./absEf;
- %Ek2 = Ek2.*absEf;
-
- %if sum(Se) <= sum(Sd)
-
- % mEk = mufb.*Ek2;
- % PP = conj(XFm).*(ones(M,1) * mEk')';
- % tmp = [PP ; flipud(conj(PP(2:N,:)))];
- % IFPP = real(ifft(tmp));
- % PH = IFPP(1:N,:);
- % tmp = fft([PH;zeros(N,M)]);
- % FPH = tmp(1:N+1,:);
- % %WFbOld = WFb;
- % WFb = WFb + FPH;
-
- %else
- % WF = WFbOld;
- %end
-
- % Shift old FFTs
- %for m=M:-1:2
- % XFm(:,m) = XFm(:,m-1);
- % YFm(:,m) = YFm(:,m-1);
- %end
- XFm(:,2:end) = XFm(:,1:end-1);
- YFm(:,2:end) = YFm(:,1:end-1);
- xfwm(:,2:end) = xfwm(:,1:end-1);
- dfm(:,2:end) = dfm(:,1:end-1);
-
- %if mod(kk, floor(Nb/50)) == 0
- % fprintf(1, '.');
- %end
-
- if mod(kk, floor(Nb/100)) == 0
- %if mod(kk, floor(Nb/500)) == 0
- progressbar(kk/Nb);
- %figure(5)
- %plot(abs(WFb));
- %legend('1','2','3','4','5','6','7','8','9','10','11','12');
- %title(kk*N/fs);
- %figure(6)
- %plot(WFbD);
- %figure(6)
- %plot(threshMod)
- %if length(divergeIdx) > 0
- % plot(abs(DD))
- % hold on
- % plot(abs(Ek), 'r')
- % hold off
- %plot(min(sqrt(Sd./(Se+1e-10)),1))
- %axis([0 N 0 1]);
- %end
- %figure(6)
- %plot(cohedFast);
- %axis([1 N+1 0 1]);
- %plot(WFbEn);
-
- %figure(7)
- %plot(weight);
- %plot([cohxd 1-cohed]);
- %plot([cohxd 1-cohed 1-cohedFast hnled]);
- %plot([cohxd cohxdFast/max(cohxdFast)]);
- %legend('cohxd', '1-cohed', '1-cohedFast');
- %axis([1 65 0 1]);
- %pause(0.5);
- %overdrive
- end
-end
-progressbar(1);
-
-%figure(2);
-%plot([feat(:,1) feat(:,2)+1 feat(:,3)+2 mfeat+3]);
-%plot([feat(:,1) mfeat+1]);
-
-%figure(3);
-%plot(10*log10([dri erifb erifb3 ericn]));
-%legend('Near-end','Error','Post NLP','Final',4);
-% Compensate for delay
-%ercn=[ercn(N+1:end);zeros(N,1)];
-%ercn_=[ercn_(N+1:end);zeros(N,1)];
-
-%figure(11);
-%plot(cohxdSlow);
-
-%figure(12);
-%surf(cohxdSlow);
-%shading interp;
-
-%figure(13);
-%plot(overdriveM);
-
-%figure(14);
-%surf(overdriveM);
-%shading interp;
-
-figure(10);
-t = (0:Nb)*N/fs;
-rrinSubSamp = rrin(N*(1:(Nb+1)));
-plot(t, rrinSubSamp/max(abs(rrinSubSamp)),'b');
-hold on
-plot(t, hnledAvg, 'r');
-plot(t, hnlxdAvg, 'g');
-plot(t, hnlSortQV, 'y');
-plot(t, hnlLocalMinV, 'k');
-plot(t, cohxdLocalMinV, 'c');
-plot(t, hnlPrefAvgV, 'm');
-%plot(t, cohxdAvg, 'r');
-%plot(cohxdFastAvg, 'r');
-%plot(cohxdAvgBad, 'k');
-%plot(t, cohedAvg, 'k');
-%plot(t, 1-cohedFastAvg, 'k');
-%plot(ssin(N*(1:floor(length(ssin)/N)))/max(abs(ssin)));
-%plot(echoBands,'r');
-%plot(overdrive, 'g');
-%plot(erfb(N*(1:floor(length(erfb)/N)))/max(abs(erfb)));
-hold off
-tightx;
-
-figure(11)
-plot(t, ovrdV);
-tightx;
-%plot(mfeat,'r');
-%plot(1-cohxyp_,'r');
-%plot(Hnlxydp,'y');
-%plot(hnledp,'k');
-%plot(Hnlxydp, 'c');
-%plot(ccohpd_,'k');
-%plot(supplot_, 'g');
-%plot(ones(length(mfeat),1)*rr1_, 'k');
-%plot(ones(length(mfeat),1)*rr2_, 'k');
-%plot(N*(1:length(feat)), feat);
-%plot(Sep_,'r');
-%axis([1 floor(length(erfb)/N) -1 1])
-%hold off
-%plot(10*log10([Se_, Sx_, Seu_, real(sf_.*conj(sf_))]));
-%legend('Se','Sx','Seu','S');
-%figure(5)
-%plot([ercn ercn_]);
-
-figure(12)
-plot(t, dIdxV);
-%plot(t, SLxV);
-tightx;
-
-%figure(13)
-%plot(t, [ekEnV dkEnV]);
-%plot(t, dkEnV./(ekEnV+1e-10));
-%tightx;
-
-%close(hh);
-%spclab(fs,ssin,erfb,ercn,'outxd.pcm');
-%spclab(fs,rrin,ssin,erfb,1.78*ercn,'vqeOut-1.pcm');
-%spclab(fs,erfb,'aecOutLp.pcm');
-%spclab(fs,rrin,ssin,erfb,1.78*ercn,'aecOut25.pcm','vqeOut-1.pcm');
-%spclab(fs,rrin,ssin,erfb,ercn,'aecOut-mba.pcm');
-%spclab(fs,rrin,ssin,erfb,ercn,'aecOut.pcm');
-%spclab(fs, ssin, erfb, ercn, 'out0.pcm');
diff --git a/src/modules/audio_processing/aec/main/source/Android.mk b/src/modules/audio_processing/aec/main/source/Android.mk
deleted file mode 100644
index f16f26b..0000000
--- a/src/modules/audio_processing/aec/main/source/Android.mk
+++ /dev/null
@@ -1,61 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aec
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- echo_cancellation.c \
- resampler.c \
- aec_core.c \
- aec_rdft.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-else
-LOCAL_SRC_FILES += \
- aec_core_sse2.c \
- aec_rdft_sse2.c
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../utility \
- $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft.h b/src/modules/audio_processing/aec/main/source/aec_rdft.h
deleted file mode 100644
index cf90882..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_rdft.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// constants shared by all paths (C, SSE2).
-extern float rdft_w[64];
-
-// code path selection function pointers
-typedef void (*rft_sub_128_t)(float *a);
-extern rft_sub_128_t rftfsub_128;
-extern rft_sub_128_t rftbsub_128;
-
-// entry points
-void aec_rdft_init(void);
-void aec_rdft_init_sse2(void);
-void aec_rdft_forward_128(float *a);
-void aec_rdft_inverse_128(float *a);
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c b/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c
deleted file mode 100644
index 901a1b1..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <emmintrin.h>
-
-#include "aec_rdft.h"
-
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
-static void rftfsub_128_SSE2(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- static const ALIGN16_BEG float ALIGN16_END k_half[4] =
- {0.5f, 0.5f, 0.5f, 0.5f};
- const __m128 mm_half = _mm_load_ps(k_half);
-
- // Vectorized code (four at once).
- // Note: commented number are indexes for the first iteration of the loop.
- for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
- // Load 'wk'.
- const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
- const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
- const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
- const __m128 wkr_ =
- _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
- const __m128 wki_ = c_j1; // 1, 2, 3, 4,
- // Load and shuffle 'a'.
- const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
- const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
- const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
- const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
- const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
- const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
- const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
- const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
- // Calculate 'x'.
- const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
- // 2-126, 4-124, 6-122, 8-120,
- const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
- // 3-127, 5-125, 7-123, 9-121,
- // Calculate product into 'y'.
- // yr = wkr * xr - wki * xi;
- // yi = wkr * xi + wki * xr;
- const __m128 a_ = _mm_mul_ps(wkr_, xr_);
- const __m128 b_ = _mm_mul_ps(wki_, xi_);
- const __m128 c_ = _mm_mul_ps(wkr_, xi_);
- const __m128 d_ = _mm_mul_ps(wki_, xr_);
- const __m128 yr_ = _mm_sub_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
- const __m128 yi_ = _mm_add_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
- // Update 'a'.
- // a[j2 + 0] -= yr;
- // a[j2 + 1] -= yi;
- // a[k2 + 0] += yr;
- // a[k2 + 1] -= yi;
- const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
- const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_); // 3, 5, 7, 9,
- const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
- const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_); // 127, 125, 123, 121,
- // Shuffle in right order and store.
- const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
- // 2, 3, 4, 5,
- const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
- // 6, 7, 8, 9,
- const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
- // 122, 123, 120, 121,
- const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
- // 126, 127, 124, 125,
- const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
- const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
- _mm_storeu_ps(&a[0 + j2], a_j2_0n);
- _mm_storeu_ps(&a[4 + j2], a_j2_4n);
- _mm_storeu_ps(&a[122 - j2], a_k2_0n);
- _mm_storeu_ps(&a[126 - j2], a_k2_4n);
- }
- // Scalar code for the remaining items.
- for (; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr - wki * xi;
- yi = wkr * xi + wki * xr;
- a[j2 + 0] -= yr;
- a[j2 + 1] -= yi;
- a[k2 + 0] += yr;
- a[k2 + 1] -= yi;
- }
-}
-
-static void rftbsub_128_SSE2(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- static const ALIGN16_BEG float ALIGN16_END k_half[4] =
- {0.5f, 0.5f, 0.5f, 0.5f};
- const __m128 mm_half = _mm_load_ps(k_half);
-
- a[1] = -a[1];
- // Vectorized code (four at once).
- // Note: commented number are indexes for the first iteration of the loop.
- for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
- // Load 'wk'.
- const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
- const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
- const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
- const __m128 wkr_ =
- _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
- const __m128 wki_ = c_j1; // 1, 2, 3, 4,
- // Load and shuffle 'a'.
- const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
- const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
- const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
- const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
- const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
- const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
- const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
- const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
- // Calculate 'x'.
- const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
- // 2-126, 4-124, 6-122, 8-120,
- const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
- // 3-127, 5-125, 7-123, 9-121,
- // Calculate product into 'y'.
- // yr = wkr * xr + wki * xi;
- // yi = wkr * xi - wki * xr;
- const __m128 a_ = _mm_mul_ps(wkr_, xr_);
- const __m128 b_ = _mm_mul_ps(wki_, xi_);
- const __m128 c_ = _mm_mul_ps(wkr_, xi_);
- const __m128 d_ = _mm_mul_ps(wki_, xr_);
- const __m128 yr_ = _mm_add_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
- const __m128 yi_ = _mm_sub_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
- // Update 'a'.
- // a[j2 + 0] = a[j2 + 0] - yr;
- // a[j2 + 1] = yi - a[j2 + 1];
- // a[k2 + 0] = yr + a[k2 + 0];
- // a[k2 + 1] = yi - a[k2 + 1];
- const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
- const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1); // 3, 5, 7, 9,
- const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
- const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1); // 127, 125, 123, 121,
- // Shuffle in right order and store.
- // Shuffle in right order and store.
- const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
- // 2, 3, 4, 5,
- const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
- // 6, 7, 8, 9,
- const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
- // 122, 123, 120, 121,
- const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
- // 126, 127, 124, 125,
- const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
- const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
- _mm_storeu_ps(&a[0 + j2], a_j2_0n);
- _mm_storeu_ps(&a[4 + j2], a_j2_4n);
- _mm_storeu_ps(&a[122 - j2], a_k2_0n);
- _mm_storeu_ps(&a[126 - j2], a_k2_4n);
- }
- // Scalar code for the remaining items.
- for (; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr + wki * xi;
- yi = wkr * xi - wki * xr;
- a[j2 + 0] = a[j2 + 0] - yr;
- a[j2 + 1] = yi - a[j2 + 1];
- a[k2 + 0] = yr + a[k2 + 0];
- a[k2 + 1] = yi - a[k2 + 1];
- }
- a[65] = -a[65];
-}
-
-void aec_rdft_init_sse2(void) {
- rftfsub_128 = rftfsub_128_SSE2;
- rftbsub_128 = rftbsub_128_SSE2;
-}
diff --git a/src/modules/audio_processing/aec/main/source/echo_cancellation.c b/src/modules/audio_processing/aec/main/source/echo_cancellation.c
deleted file mode 100644
index 1313e35..0000000
--- a/src/modules/audio_processing/aec/main/source/echo_cancellation.c
+++ /dev/null
@@ -1,821 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains the API functions for the AEC.
- */
-#include <stdlib.h>
-#include <string.h>
-
-#include "echo_cancellation.h"
-#include "aec_core.h"
-#include "ring_buffer.h"
-#include "resampler.h"
-#ifdef AEC_DEBUG
- #include <stdio.h>
-#endif
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int bufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int sampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
-static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
-static const int initCheck = 42;
-
-typedef struct {
- int delayCtr;
- int sampFreq;
- int splitSampFreq;
- int scSampFreq;
- float sampFactor; // scSampRate / sampFreq
- short nlpMode;
- short autoOnOff;
- short activity;
- short skewMode;
- short bufSizeStart;
- //short bufResetCtr; // counts number of noncausal frames
- int knownDelay;
-
- // Stores the last frame added to the farend buffer
- short farendOld[2][FRAME_LEN];
- short initFlag; // indicates if AEC has been initialized
-
- // Variables used for averaging far end buffer size
- short counter;
- short sum;
- short firstVal;
- short checkBufSizeCtr;
-
- // Variables used for delay shifts
- short msInSndCardBuf;
- short filtDelay;
- int timeForDelayChange;
- int ECstartup;
- int checkBuffSize;
- int delayChange;
- short lastDelayDiff;
-
-#ifdef AEC_DEBUG
- FILE *bufFile;
- FILE *delayFile;
- FILE *skewFile;
- FILE *preCompFile;
- FILE *postCompFile;
-#endif // AEC_DEBUG
-
- // Structures
- void *farendBuf;
- void *resampler;
-
- int skewFrCtr;
- int resample; // if the skew is small enough we don't resample
- int highSkewCtr;
- float skew;
-
- int lastError;
-
- aec_t *aec;
-} aecpc_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int EstBufDelay(aecpc_t *aecInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int DelayComp(aecpc_t *aecInst);
-
-WebRtc_Word32 WebRtcAec_Create(void **aecInst)
-{
- aecpc_t *aecpc;
- if (aecInst == NULL) {
- return -1;
- }
-
- aecpc = malloc(sizeof(aecpc_t));
- *aecInst = aecpc;
- if (aecpc == NULL) {
- return -1;
- }
-
- if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
-
- if (WebRtcApm_CreateBuffer(&aecpc->farendBuf, bufSizeSamp) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
-
- if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
-
- aecpc->initFlag = 0;
- aecpc->lastError = 0;
-
-#ifdef AEC_DEBUG
- aecpc->aec->farFile = fopen("aecFar.pcm","wb");
- aecpc->aec->nearFile = fopen("aecNear.pcm","wb");
- aecpc->aec->outFile = fopen("aecOut.pcm","wb");
- aecpc->aec->outLpFile = fopen("aecOutLp.pcm","wb");
-
- aecpc->bufFile = fopen("aecBuf.dat", "wb");
- aecpc->skewFile = fopen("aecSkew.dat", "wb");
- aecpc->delayFile = fopen("aecDelay.dat", "wb");
- aecpc->preCompFile = fopen("preComp.pcm", "wb");
- aecpc->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Free(void *aecInst)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
-#ifdef AEC_DEBUG
- fclose(aecpc->aec->farFile);
- fclose(aecpc->aec->nearFile);
- fclose(aecpc->aec->outFile);
- fclose(aecpc->aec->outLpFile);
-
- fclose(aecpc->bufFile);
- fclose(aecpc->skewFile);
- fclose(aecpc->delayFile);
- fclose(aecpc->preCompFile);
- fclose(aecpc->postCompFile);
-#endif // AEC_DEBUG
-
- WebRtcAec_FreeAec(aecpc->aec);
- WebRtcApm_FreeBuffer(aecpc->farendBuf);
- WebRtcAec_FreeResampler(aecpc->resampler);
- free(aecpc);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
-{
- aecpc_t *aecpc = aecInst;
- AecConfig aecConfig;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->sampFreq = sampFreq;
-
- if (scSampFreq < 1 || scSampFreq > 96000) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->scSampFreq = scSampFreq;
-
- // Initialize echo canceller core
- if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- // Initialize farend buffer
- if (WebRtcApm_InitBuffer(aecpc->farendBuf) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- aecpc->initFlag = initCheck; // indicates that initilisation has been done
-
- if (aecpc->sampFreq == 32000) {
- aecpc->splitSampFreq = 16000;
- }
- else {
- aecpc->splitSampFreq = sampFreq;
- }
-
- aecpc->skewFrCtr = 0;
- aecpc->activity = 0;
-
- aecpc->delayChange = 1;
- aecpc->delayCtr = 0;
-
- aecpc->sum = 0;
- aecpc->counter = 0;
- aecpc->checkBuffSize = 1;
- aecpc->firstVal = 0;
-
- aecpc->ECstartup = 1;
- aecpc->bufSizeStart = 0;
- aecpc->checkBufSizeCtr = 0;
- aecpc->filtDelay = 0;
- aecpc->timeForDelayChange =0;
- aecpc->knownDelay = 0;
- aecpc->lastDelayDiff = 0;
-
- aecpc->skew = 0;
- aecpc->resample = kAecFalse;
- aecpc->highSkewCtr = 0;
- aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
-
- memset(&aecpc->farendOld[0][0], 0, 160);
-
- // Default settings.
- aecConfig.nlpMode = kAecNlpModerate;
- aecConfig.skewMode = kAecFalse;
- aecConfig.metricsMode = kAecFalse;
-
- if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- return 0;
-}
-
-// only buffer L band for farend
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
- WebRtc_Word16 nrOfSamples)
-{
- aecpc_t *aecpc = aecInst;
- WebRtc_Word32 retVal = 0;
- short newNrOfSamples;
- short newFarend[MAX_RESAMP_LEN];
- float skew;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (farend == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // number of samples == 160 for SWB input
- if (nrOfSamples != 80 && nrOfSamples != 160) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- skew = aecpc->skew;
-
- // TODO: Is this really a good idea?
- if (!aecpc->ECstartup) {
- DelayComp(aecpc);
- }
-
- if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
- // Resample and get a new number of samples
- newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
- farend,
- nrOfSamples,
- skew,
- newFarend);
- WebRtcApm_WriteBuffer(aecpc->farendBuf, newFarend, newNrOfSamples);
-
-#ifdef AEC_DEBUG
- fwrite(farend, 2, nrOfSamples, aecpc->preCompFile);
- fwrite(newFarend, 2, newNrOfSamples, aecpc->postCompFile);
-#endif
- }
- else {
- WebRtcApm_WriteBuffer(aecpc->farendBuf, farend, nrOfSamples);
- }
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
- const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
- WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
-{
- aecpc_t *aecpc = aecInst;
- WebRtc_Word32 retVal = 0;
- short i;
- short farend[FRAME_LEN];
- short nmbrOfFilledBuffers;
- short nBlocks10ms;
- short nFrames;
-#ifdef AEC_DEBUG
- short msInAECBuf;
-#endif
- // Limit resampling to doubling/halving of signal
- const float minSkewEst = -0.5f;
- const float maxSkewEst = 1.0f;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (nearend == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (out == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // number of samples == 160 for SWB input
- if (nrOfSamples != 80 && nrOfSamples != 160) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- // Check for valid pointers based on sampling rate
- if (aecpc->sampFreq == 32000 && nearendH == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (msInSndCardBuf < 0) {
- msInSndCardBuf = 0;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- retVal = -1;
- }
- else if (msInSndCardBuf > 500) {
- msInSndCardBuf = 500;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- retVal = -1;
- }
- msInSndCardBuf += 10;
- aecpc->msInSndCardBuf = msInSndCardBuf;
-
- if (aecpc->skewMode == kAecTrue) {
- if (aecpc->skewFrCtr < 25) {
- aecpc->skewFrCtr++;
- }
- else {
- retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
- if (retVal == -1) {
- aecpc->skew = 0;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- }
-
- aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
- if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
- aecpc->resample = kAecFalse;
- }
- else {
- aecpc->resample = kAecTrue;
- }
-
- if (aecpc->skew < minSkewEst) {
- aecpc->skew = minSkewEst;
- }
- else if (aecpc->skew > maxSkewEst) {
- aecpc->skew = maxSkewEst;
- }
-
-#ifdef AEC_DEBUG
- fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
- }
- }
-
- nFrames = nrOfSamples / FRAME_LEN;
- nBlocks10ms = nFrames / aecpc->aec->mult;
-
- if (aecpc->ECstartup) {
- memcpy(out, nearend, sizeof(short) * nrOfSamples);
- nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
- // The AEC is in the start up mode
- // AEC is disabled until the soundcard buffer and farend buffers are OK
-
- // Mechanism to ensure that the soundcard buffer is reasonably stable.
- if (aecpc->checkBuffSize) {
-
- aecpc->checkBufSizeCtr++;
- // Before we fill up the far end buffer we require the amount of data on the
- // sound card to be stable (+/-8 ms) compared to the first value. This
- // comparison is made during the following 4 consecutive frames. If it seems
- // to be stable then we start to fill up the far end buffer.
-
- if (aecpc->counter == 0) {
- aecpc->firstVal = aecpc->msInSndCardBuf;
- aecpc->sum = 0;
- }
-
- if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
- WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
- aecpc->sum += aecpc->msInSndCardBuf;
- aecpc->counter++;
- }
- else {
- aecpc->counter = 0;
- }
-
- if (aecpc->counter*nBlocks10ms >= 6) {
- // The farend buffer size is determined in blocks of 80 samples
- // Use 75% of the average value of the soundcard buffer
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->sum *
- aecpc->aec->mult) / (aecpc->counter * 10)), BUF_SIZE_FRAMES);
- // buffersize has now been determined
- aecpc->checkBuffSize = 0;
- }
-
- if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
- // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->msInSndCardBuf *
- aecpc->aec->mult) / 10), BUF_SIZE_FRAMES);
- aecpc->checkBuffSize = 0;
- }
- }
-
- // if checkBuffSize changed in the if-statement above
- if (!aecpc->checkBuffSize) {
- // soundcard buffer is now reasonably stable
- // When the far end buffer is filled with approximately the same amount of
- // data as the amount on the sound card we end the start up phase and start
- // to cancel echoes.
-
- if (nmbrOfFilledBuffers == aecpc->bufSizeStart) {
- aecpc->ECstartup = 0; // Enable the AEC
- }
- else if (nmbrOfFilledBuffers > aecpc->bufSizeStart) {
- WebRtcApm_FlushBuffer(aecpc->farendBuf, WebRtcApm_get_buffer_size(aecpc->farendBuf) -
- aecpc->bufSizeStart * FRAME_LEN);
- aecpc->ECstartup = 0;
- }
- }
-
- }
- else {
- // AEC is enabled
-
- // Note only 1 block supported for nb and 2 blocks for wb
- for (i = 0; i < nFrames; i++) {
- nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
- // Check that there is data in the far end buffer
- if (nmbrOfFilledBuffers > 0) {
- // Get the next 80 samples from the farend buffer
- WebRtcApm_ReadBuffer(aecpc->farendBuf, farend, FRAME_LEN);
-
- // Always store the last frame for use when we run out of data
- memcpy(&(aecpc->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
- }
- else {
- // We have no data so we use the last played frame
- memcpy(farend, &(aecpc->farendOld[i][0]), FRAME_LEN * sizeof(short));
- }
-
- // Call buffer delay estimator when all data is extracted,
- // i.e. i = 0 for NB and i = 1 for WB or SWB
- if ((i == 0 && aecpc->splitSampFreq == 8000) ||
- (i == 1 && (aecpc->splitSampFreq == 16000))) {
- EstBufDelay(aecpc, aecpc->msInSndCardBuf);
- }
-
- // Call the AEC
- WebRtcAec_ProcessFrame(aecpc->aec, farend, &nearend[FRAME_LEN * i], &nearendH[FRAME_LEN * i],
- &out[FRAME_LEN * i], &outH[FRAME_LEN * i], aecpc->knownDelay);
- }
- }
-
-#ifdef AEC_DEBUG
- msInAECBuf = WebRtcApm_get_buffer_size(aecpc->farendBuf) / (sampMsNb*aecpc->aec->mult);
- fwrite(&msInAECBuf, 2, 1, aecpc->bufFile);
- fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
-#endif
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->skewMode = config.skewMode;
-
- if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
- kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->nlpMode = config.nlpMode;
- aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
- aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
-
- if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->aec->metricsMode = config.metricsMode;
- if (aecpc->aec->metricsMode == kAecTrue) {
- WebRtcAec_InitMetrics(aecpc->aec);
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (config == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- config->nlpMode = aecpc->nlpMode;
- config->skewMode = aecpc->skewMode;
- config->metricsMode = aecpc->aec->metricsMode;
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (status == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- *status = aecpc->aec->echoState;
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
-{
- const float upweight = 0.7f;
- float dtmp;
- short stmp;
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (metrics == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // ERL
- metrics->erl.instant = (short) aecpc->aec->erl.instant;
-
- if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
- metrics->erl.average = (short) dtmp;
- }
- else {
- metrics->erl.average = offsetLevel;
- }
-
- metrics->erl.max = (short) aecpc->aec->erl.max;
-
- if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
- metrics->erl.min = (short) aecpc->aec->erl.min;
- }
- else {
- metrics->erl.min = offsetLevel;
- }
-
- // ERLE
- metrics->erle.instant = (short) aecpc->aec->erle.instant;
-
- if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
- metrics->erle.average = (short) dtmp;
- }
- else {
- metrics->erle.average = offsetLevel;
- }
-
- metrics->erle.max = (short) aecpc->aec->erle.max;
-
- if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
- metrics->erle.min = (short) aecpc->aec->erle.min;
- } else {
- metrics->erle.min = offsetLevel;
- }
-
- // RERL
- if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
- stmp = metrics->erl.average + metrics->erle.average;
- }
- else {
- stmp = offsetLevel;
- }
- metrics->rerl.average = stmp;
-
- // No other statistics needed, but returned for completeness
- metrics->rerl.instant = stmp;
- metrics->rerl.max = stmp;
- metrics->rerl.min = stmp;
-
- // A_NLP
- metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
-
- if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
- metrics->aNlp.average = (short) dtmp;
- }
- else {
- metrics->aNlp.average = offsetLevel;
- }
-
- metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
-
- if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
- metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
- }
- else {
- metrics->aNlp.min = offsetLevel;
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
-{
- const char version[] = "AEC 2.5.0";
- const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
- if (versionStr == NULL) {
- return -1;
- }
-
- if (versionLen > len) {
- return -1;
- }
-
- strncpy(versionStr, version, versionLen);
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- return aecpc->lastError;
-}
-
-static int EstBufDelay(aecpc_t *aecpc, short msInSndCardBuf)
-{
- short delayNew, nSampFar, nSampSndCard;
- short diff;
-
- nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
- nSampSndCard = msInSndCardBuf * sampMsNb * aecpc->aec->mult;
-
- delayNew = nSampSndCard - nSampFar;
-
- // Account for resampling frame delay
- if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
- delayNew -= kResamplingDelay;
- }
-
- if (delayNew < FRAME_LEN) {
- WebRtcApm_FlushBuffer(aecpc->farendBuf, FRAME_LEN);
- delayNew += FRAME_LEN;
- }
-
- aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short)(0.8*aecpc->filtDelay + 0.2*delayNew));
-
- diff = aecpc->filtDelay - aecpc->knownDelay;
- if (diff > 224) {
- if (aecpc->lastDelayDiff < 96) {
- aecpc->timeForDelayChange = 0;
- }
- else {
- aecpc->timeForDelayChange++;
- }
- }
- else if (diff < 96 && aecpc->knownDelay > 0) {
- if (aecpc->lastDelayDiff > 224) {
- aecpc->timeForDelayChange = 0;
- }
- else {
- aecpc->timeForDelayChange++;
- }
- }
- else {
- aecpc->timeForDelayChange = 0;
- }
- aecpc->lastDelayDiff = diff;
-
- if (aecpc->timeForDelayChange > 25) {
- aecpc->knownDelay = WEBRTC_SPL_MAX((int)aecpc->filtDelay - 160, 0);
- }
- return 0;
-}
-
-static int DelayComp(aecpc_t *aecpc)
-{
- int nSampFar, nSampSndCard, delayNew, nSampAdd;
- const int maxStuffSamp = 10 * FRAME_LEN;
-
- nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
- nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
- delayNew = nSampSndCard - nSampFar;
-
- // Account for resampling frame delay
- if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
- delayNew -= kResamplingDelay;
- }
-
- if (delayNew > FAR_BUF_LEN - FRAME_LEN*aecpc->aec->mult) {
- // The difference of the buffersizes is larger than the maximum
- // allowed known delay. Compensate by stuffing the buffer.
- nSampAdd = (int)(WEBRTC_SPL_MAX((int)(0.5 * nSampSndCard - nSampFar),
- FRAME_LEN));
- nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
- WebRtcApm_StuffBuffer(aecpc->farendBuf, nSampAdd);
- aecpc->delayChange = 1; // the delay needs to be updated
- }
-
- return 0;
-}
diff --git a/src/modules/audio_processing/aecm/Android.mk b/src/modules/audio_processing/aecm/Android.mk
new file mode 100644
index 0000000..10c38ca
--- /dev/null
+++ b/src/modules/audio_processing/aecm/Android.mk
@@ -0,0 +1,78 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+#############################
+# Build the non-neon library.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aecm
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ echo_control_mobile.c \
+ aecm_core.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#########################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_aecm_neon
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := aecm_core_neon.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ -mfpu=neon \
+ -mfloat-abi=softfp \
+ -flax-vector-conversions
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/src/modules/audio_processing/aecm/main/source/aecm.gyp b/src/modules/audio_processing/aecm/aecm.gypi
similarity index 64%
rename from src/modules/audio_processing/aecm/main/source/aecm.gyp
rename to src/modules/audio_processing/aecm/aecm.gypi
index a535d2b..bf520bf 100644
--- a/src/modules/audio_processing/aecm/main/source/aecm.gyp
+++ b/src/modules/audio_processing/aecm/aecm.gypi
@@ -7,27 +7,25 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../../common_settings.gypi',
- ],
'targets': [
{
'target_name': 'aecm',
'type': '<(library)',
'dependencies': [
- '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
- '../../../utility/util.gyp:apm_util'
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ 'apm_util'
],
'include_dirs': [
- '../interface',
+ 'interface',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'interface',
],
},
'sources': [
- '../interface/echo_control_mobile.h',
+ 'interface/echo_control_mobile.h',
'echo_control_mobile.c',
'aecm_core.c',
'aecm_core.h',
@@ -35,9 +33,3 @@
},
],
}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/aecm/aecm_core.c b/src/modules/audio_processing/aecm/aecm_core.c
new file mode 100644
index 0000000..9bf5c4a
--- /dev/null
+++ b/src/modules/audio_processing/aecm/aecm_core.c
@@ -0,0 +1,2126 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aecm_core.h"
+
+#include <assert.h>
+#include <stdlib.h>
+
+#include "cpu_features_wrapper.h"
+#include "delay_estimator_wrapper.h"
+#include "echo_control_mobile.h"
+#include "ring_buffer.h"
+#include "typedefs.h"
+
+#ifdef ARM_WINM_LOG
+#include <stdio.h>
+#include <windows.h>
+#endif
+
+#ifdef AEC_DEBUG
+FILE *dfile;
+FILE *testfile;
+#endif
+
+#ifdef _MSC_VER // visual c++
+#define ALIGN8_BEG __declspec(align(8))
+#define ALIGN8_END
+#else // gcc or icc
+#define ALIGN8_BEG
+#define ALIGN8_END __attribute__((aligned(8)))
+#endif
+
+#ifdef AECM_SHORT
+
+// Square root of Hanning window in Q14
+const WebRtc_Word16 WebRtcAecm_kSqrtHanning[] =
+{
+ 0, 804, 1606, 2404, 3196, 3981, 4756, 5520,
+ 6270, 7005, 7723, 8423, 9102, 9760, 10394, 11003,
+ 11585, 12140, 12665, 13160, 13623, 14053, 14449, 14811,
+ 15137, 15426, 15679, 15893, 16069, 16207, 16305, 16364,
+ 16384
+};
+
+#else
+
+// Square root of Hanning window in Q14
+const ALIGN8_BEG WebRtc_Word16 WebRtcAecm_kSqrtHanning[] ALIGN8_END =
+{
+ 0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
+ 3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019, 8364,
+ 8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514, 11795, 12068, 12335,
+ 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189, 14384, 14571, 14749, 14918,
+ 15079, 15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034, 16111, 16179, 16237,
+ 16286, 16325, 16354, 16373, 16384
+};
+
+#endif
+
+//Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha1 = 32584;
+//Q15 beta = 0.12967166976970 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta1 = 4249;
+//Q15 alpha = 0.94234827210087 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha2 = 30879;
+//Q15 beta = 0.33787806009150 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta2 = 11072;
+//Q15 alpha = 0.82247698684306 const Factor for magnitude approximation
+static const WebRtc_UWord16 kAlpha3 = 26951;
+//Q15 beta = 0.57762063060713 const Factor for magnitude approximation
+static const WebRtc_UWord16 kBeta3 = 18927;
+
+// Initialization table for echo channel in 8 kHz
+static const WebRtc_Word16 kChannelStored8kHz[PART_LEN1] = {
+ 2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418,
+ 1451, 1506, 1562, 1644, 1726, 1804, 1882, 1918,
+ 1953, 1982, 2010, 2025, 2040, 2034, 2027, 2021,
+ 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683,
+ 1635, 1604, 1572, 1545, 1517, 1481, 1444, 1405,
+ 1367, 1331, 1294, 1270, 1245, 1239, 1233, 1247,
+ 1260, 1282, 1303, 1338, 1373, 1407, 1441, 1470,
+ 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649,
+ 1676
+};
+
+// Initialization table for echo channel in 16 kHz
+static const WebRtc_Word16 kChannelStored16kHz[PART_LEN1] = {
+ 2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882,
+ 1953, 2010, 2040, 2027, 2014, 1980, 1869, 1732,
+ 1635, 1572, 1517, 1444, 1367, 1294, 1245, 1233,
+ 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621,
+ 1676, 1741, 1802, 1861, 1921, 1983, 2040, 2102,
+ 2170, 2265, 2375, 2515, 2651, 2781, 2922, 3075,
+ 3253, 3471, 3738, 3976, 4151, 4258, 4308, 4288,
+ 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484,
+ 3153
+};
+
+static const WebRtc_Word16 kCosTable[] = {
+ 8192, 8190, 8187, 8180, 8172, 8160, 8147, 8130, 8112,
+ 8091, 8067, 8041, 8012, 7982, 7948, 7912, 7874, 7834,
+ 7791, 7745, 7697, 7647, 7595, 7540, 7483, 7424, 7362,
+ 7299, 7233, 7164, 7094, 7021, 6947, 6870, 6791, 6710,
+ 6627, 6542, 6455, 6366, 6275, 6182, 6087, 5991, 5892,
+ 5792, 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
+ 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971, 3845,
+ 3719, 3591, 3462, 3331, 3200, 3068, 2935, 2801, 2667,
+ 2531, 2395, 2258, 2120, 1981, 1842, 1703, 1563, 1422,
+ 1281, 1140, 998, 856, 713, 571, 428, 285, 142,
+ 0, -142, -285, -428, -571, -713, -856, -998, -1140,
+ -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
+ -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
+ -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
+ -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
+ -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
+ -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
+ -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
+ -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
+ -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
+ -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
+ -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
+ -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
+ -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
+ -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
+ -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
+ -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
+ -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
+ -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
+ -1281, -1140, -998, -856, -713, -571, -428, -285, -142,
+ 0, 142, 285, 428, 571, 713, 856, 998, 1140,
+ 1281, 1422, 1563, 1703, 1842, 1981, 2120, 2258, 2395,
+ 2531, 2667, 2801, 2935, 3068, 3200, 3331, 3462, 3591,
+ 3719, 3845, 3971, 4095, 4219, 4341, 4461, 4580, 4698,
+ 4815, 4930, 5043, 5155, 5265, 5374, 5481, 5586, 5690,
+ 5792, 5892, 5991, 6087, 6182, 6275, 6366, 6455, 6542,
+ 6627, 6710, 6791, 6870, 6947, 7021, 7094, 7164, 7233,
+ 7299, 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
+ 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041, 8067,
+ 8091, 8112, 8130, 8147, 8160, 8172, 8180, 8187, 8190
+};
+
+static const WebRtc_Word16 kSinTable[] = {
+ 0, 142, 285, 428, 571, 713, 856, 998,
+ 1140, 1281, 1422, 1563, 1703, 1842, 1981, 2120,
+ 2258, 2395, 2531, 2667, 2801, 2935, 3068, 3200,
+ 3331, 3462, 3591, 3719, 3845, 3971, 4095, 4219,
+ 4341, 4461, 4580, 4698, 4815, 4930, 5043, 5155,
+ 5265, 5374, 5481, 5586, 5690, 5792, 5892, 5991,
+ 6087, 6182, 6275, 6366, 6455, 6542, 6627, 6710,
+ 6791, 6870, 6947, 7021, 7094, 7164, 7233, 7299,
+ 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
+ 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041,
+ 8067, 8091, 8112, 8130, 8147, 8160, 8172, 8180,
+ 8187, 8190, 8191, 8190, 8187, 8180, 8172, 8160,
+ 8147, 8130, 8112, 8091, 8067, 8041, 8012, 7982,
+ 7948, 7912, 7874, 7834, 7791, 7745, 7697, 7647,
+ 7595, 7540, 7483, 7424, 7362, 7299, 7233, 7164,
+ 7094, 7021, 6947, 6870, 6791, 6710, 6627, 6542,
+ 6455, 6366, 6275, 6182, 6087, 5991, 5892, 5792,
+ 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
+ 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971,
+ 3845, 3719, 3591, 3462, 3331, 3200, 3068, 2935,
+ 2801, 2667, 2531, 2395, 2258, 2120, 1981, 1842,
+ 1703, 1563, 1422, 1281, 1140, 998, 856, 713,
+ 571, 428, 285, 142, 0, -142, -285, -428,
+ -571, -713, -856, -998, -1140, -1281, -1422, -1563,
+ -1703, -1842, -1981, -2120, -2258, -2395, -2531, -2667,
+ -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719,
+ -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
+ -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586,
+ -5690, -5792, -5892, -5991, -6087, -6182, -6275, -6366,
+ -6455, -6542, -6627, -6710, -6791, -6870, -6947, -7021,
+ -7094, -7164, -7233, -7299, -7362, -7424, -7483, -7540,
+ -7595, -7647, -7697, -7745, -7791, -7834, -7874, -7912,
+ -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130,
+ -8147, -8160, -8172, -8180, -8187, -8190, -8191, -8190,
+ -8187, -8180, -8172, -8160, -8147, -8130, -8112, -8091,
+ -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
+ -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424,
+ -7362, -7299, -7233, -7164, -7094, -7021, -6947, -6870,
+ -6791, -6710, -6627, -6542, -6455, -6366, -6275, -6182,
+ -6087, -5991, -5892, -5792, -5690, -5586, -5481, -5374,
+ -5265, -5155, -5043, -4930, -4815, -4698, -4580, -4461,
+ -4341, -4219, -4096, -3971, -3845, -3719, -3591, -3462,
+ -3331, -3200, -3068, -2935, -2801, -2667, -2531, -2395,
+ -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
+ -1140, -998, -856, -713, -571, -428, -285, -142
+};
+
+static const WebRtc_Word16 kNoiseEstQDomain = 15;
+static const WebRtc_Word16 kNoiseEstIncCount = 5;
+
+static void ComfortNoise(AecmCore_t* aecm,
+ const WebRtc_UWord16* dfa,
+ complex16_t* out,
+ const WebRtc_Word16* lambda);
+
+static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm);
+
+// Moves the pointer to the next entry and inserts |far_spectrum| and
+// corresponding Q-domain in its buffer.
+//
+// Inputs:
+// - self : Pointer to the delay estimation instance
+// - far_spectrum : Pointer to the far end spectrum
+// - far_q : Q-domain of far end spectrum
+//
+static void UpdateFarHistory(AecmCore_t* self,
+ uint16_t* far_spectrum,
+ int far_q) {
+ // Get new buffer position
+ self->far_history_pos++;
+ if (self->far_history_pos >= MAX_DELAY) {
+ self->far_history_pos = 0;
+ }
+ // Update Q-domain buffer
+ self->far_q_domains[self->far_history_pos] = far_q;
+ // Update far end spectrum buffer
+ memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]),
+ far_spectrum,
+ sizeof(uint16_t) * PART_LEN1);
+}
+
+// Returns a pointer to the far end spectrum aligned to current near end
+// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
+// called before AlignedFarend(...). Otherwise, you get the pointer to the
+// previous frame. The memory is only valid until the next call of
+// WebRtc_DelayEstimatorProcessFix(...).
+//
+// Inputs:
+// - self : Pointer to the AECM instance.
+// - delay : Current delay estimate.
+//
+// Output:
+// - far_q : The Q-domain of the aligned far end spectrum
+//
+// Return value:
+// - far_spectrum : Pointer to the aligned far end spectrum
+// NULL - Error
+//
+static const uint16_t* AlignedFarend(AecmCore_t* self, int* far_q, int delay) {
+ int buffer_position = 0;
+ assert(self != NULL);
+ buffer_position = self->far_history_pos - delay;
+
+ // Check buffer position
+ if (buffer_position < 0) {
+ buffer_position += MAX_DELAY;
+ }
+ // Get Q-domain
+ *far_q = self->far_q_domains[buffer_position];
+ // Return far end spectrum
+ return &(self->far_history[buffer_position * PART_LEN1]);
+}
+
+#ifdef ARM_WINM_LOG
+HANDLE logFile = NULL;
+#endif
+
+// Declare function pointers.
+CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+WindowAndFFT WebRtcAecm_WindowAndFFT;
+InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
+
+int WebRtcAecm_CreateCore(AecmCore_t **aecmInst)
+{
+ AecmCore_t *aecm = malloc(sizeof(AecmCore_t));
+ *aecmInst = aecm;
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN,
+ sizeof(int16_t)) == -1)
+ {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ if (WebRtc_CreateDelayEstimator(&aecm->delay_estimator,
+ PART_LEN1,
+ MAX_DELAY,
+ 0) == -1) {
+ WebRtcAecm_FreeCore(aecm);
+ aecm = NULL;
+ return -1;
+ }
+
+ // Init some aecm pointers. 16 and 32 byte alignment is only necessary
+ // for Neon code currently.
+ aecm->xBuf = (WebRtc_Word16*) (((uintptr_t)aecm->xBuf_buf + 31) & ~ 31);
+ aecm->dBufClean = (WebRtc_Word16*) (((uintptr_t)aecm->dBufClean_buf + 31) & ~ 31);
+ aecm->dBufNoisy = (WebRtc_Word16*) (((uintptr_t)aecm->dBufNoisy_buf + 31) & ~ 31);
+ aecm->outBuf = (WebRtc_Word16*) (((uintptr_t)aecm->outBuf_buf + 15) & ~ 15);
+ aecm->channelStored = (WebRtc_Word16*) (((uintptr_t)
+ aecm->channelStored_buf + 15) & ~ 15);
+ aecm->channelAdapt16 = (WebRtc_Word16*) (((uintptr_t)
+ aecm->channelAdapt16_buf + 15) & ~ 15);
+ aecm->channelAdapt32 = (WebRtc_Word32*) (((uintptr_t)
+ aecm->channelAdapt32_buf + 31) & ~ 31);
+
+ return 0;
+}
+
+void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path)
+{
+ int i = 0;
+
+ // Reset the stored channel
+ memcpy(aecm->channelStored, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
+ // Reset the adapted channels
+ memcpy(aecm->channelAdapt16, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)(aecm->channelAdapt16[i]), 16);
+ }
+
+ // Reset channel storing variables
+ aecm->mseAdaptOld = 1000;
+ aecm->mseStoredOld = 1000;
+ aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
+ aecm->mseChannelCount = 0;
+}
+
+static void WindowAndFFTC(WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling)
+{
+ int i, j;
+
+ memset(fft, 0, sizeof(WebRtc_Word16) * PART_LEN4);
+ // FFT of signal
+ for (i = 0, j = 0; i < PART_LEN; i++, j += 2)
+ {
+ // Window time domain signal and insert into real part of
+ // transformation array |fft|
+ fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (time_signal[i] << time_signal_scaling),
+ WebRtcAecm_kSqrtHanning[i],
+ 14);
+ fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ (time_signal[i + PART_LEN] << time_signal_scaling),
+ WebRtcAecm_kSqrtHanning[PART_LEN - i],
+ 14);
+ // Inserting zeros in imaginary parts not necessary since we
+ // initialized the array with all zeros
+ }
+
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the first PART_LEN2 samples
+ for (i = 0, j = 0; j < PART_LEN2; i += 1, j += 2)
+ {
+ freq_signal[i].real = fft[j];
+
+ // The imaginary part has to switch sign
+ freq_signal[i].imag = - fft[j+1];
+ }
+}
+
+static void InverseFFTAndWindowC(AecmCore_t* aecm,
+ WebRtc_Word16* fft,
+ complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean)
+{
+ int i, j, outCFFT;
+ WebRtc_Word32 tmp32no1;
+
+ // Synthesis
+ for (i = 1; i < PART_LEN; i++)
+ {
+ j = WEBRTC_SPL_LSHIFT_W32(i, 1);
+ fft[j] = efw[i].real;
+
+ // mirrored data, even
+ fft[PART_LEN4 - j] = efw[i].real;
+ fft[j + 1] = -efw[i].imag;
+
+ //mirrored data, odd
+ fft[PART_LEN4 - (j - 1)] = efw[i].imag;
+ }
+ fft[0] = efw[0].real;
+ fft[1] = -efw[0].imag;
+
+ fft[PART_LEN2] = efw[PART_LEN].real;
+ fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
+
+ // inverse FFT, result should be scaled with outCFFT
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
+
+ //take only the real values and scale with outCFFT
+ for (i = 0; i < PART_LEN2; i++)
+ {
+ j = WEBRTC_SPL_LSHIFT_W32(i, 1);
+ fft[i] = fft[j];
+ }
+
+ for (i = 0; i < PART_LEN; i++)
+ {
+ fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ fft[i],
+ WebRtcAecm_kSqrtHanning[i],
+ 14);
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
+ outCFFT - aecm->dfaCleanQDomain);
+ fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ tmp32no1 + aecm->outBuf[i],
+ WEBRTC_SPL_WORD16_MIN);
+ output[i] = fft[i];
+
+ tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
+ fft[PART_LEN + i],
+ WebRtcAecm_kSqrtHanning[PART_LEN - i],
+ 14);
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
+ outCFFT - aecm->dfaCleanQDomain);
+ aecm->outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX,
+ tmp32no1,
+ WEBRTC_SPL_WORD16_MIN);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+#endif
+
+ // Copy the current block to the old position (aecm->outBuf is shifted elsewhere)
+ memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ if (nearendClean != NULL)
+ {
+ memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
+ }
+}
+
+static void CalcLinearEnergiesC(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored)
+{
+ int i;
+
+ // Get energy for the delayed far end signal and estimated
+ // echo using both stored and adapted channels.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+ (*far_energy) += (WebRtc_UWord32)(far_spectrum[i]);
+ (*echo_energy_adapt) += WEBRTC_SPL_UMUL_16_16(aecm->channelAdapt16[i],
+ far_spectrum[i]);
+ (*echo_energy_stored) += (WebRtc_UWord32)echo_est[i];
+ }
+}
+
+static void StoreAdaptiveChannelC(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est)
+{
+ int i;
+
+ // During startup we store the channel every block.
+ memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
+ // Recalculate echo estimate
+ for (i = 0; i < PART_LEN; i += 4)
+ {
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+ echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
+ far_spectrum[i + 1]);
+ echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
+ far_spectrum[i + 2]);
+ echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
+ far_spectrum[i + 3]);
+ }
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+ far_spectrum[i]);
+}
+
+static void ResetAdaptiveChannelC(AecmCore_t* aecm)
+{
+ int i;
+
+ // The stored channel has a significantly lower MSE than the adaptive one for
+ // two consecutive calculations. Reset the adaptive channel.
+ memcpy(aecm->channelAdapt16, aecm->channelStored,
+ sizeof(WebRtc_Word16) * PART_LEN1);
+ // Restore the W32 channel
+ for (i = 0; i < PART_LEN; i += 4)
+ {
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i], 16);
+ aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 1], 16);
+ aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 2], 16);
+ aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i + 3], 16);
+ }
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
+}
+
+// WebRtcAecm_InitCore(...)
+//
+// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
+// Input:
+// - aecm : Pointer to the Echo Suppression instance
+// - samplingFreq : Sampling Frequency
+//
+// Output:
+// - aecm : Initialized instance
+//
+// Return value : 0 - Ok
+// -1 - Error
+//
+int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq)
+{
+ int i = 0;
+ WebRtc_Word32 tmp32 = PART_LEN1 * PART_LEN1;
+ WebRtc_Word16 tmp16 = PART_LEN1;
+
+ if (samplingFreq != 8000 && samplingFreq != 16000)
+ {
+ samplingFreq = 8000;
+ return -1;
+ }
+ // sanity check of sampling frequency
+ aecm->mult = (WebRtc_Word16)samplingFreq / 8000;
+
+ aecm->farBufWritePos = 0;
+ aecm->farBufReadPos = 0;
+ aecm->knownDelay = 0;
+ aecm->lastKnownDelay = 0;
+
+ WebRtc_InitBuffer(aecm->farFrameBuf);
+ WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
+ WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
+ WebRtc_InitBuffer(aecm->outFrameBuf);
+
+ memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
+ memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
+ memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
+ memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
+
+ aecm->seed = 666;
+ aecm->totCount = 0;
+
+ if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
+ return -1;
+ }
+ // Set far end histories to zero
+ memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
+ memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
+ aecm->far_history_pos = MAX_DELAY;
+
+ aecm->nlpFlag = 1;
+ aecm->fixedDelay = -1;
+
+ aecm->dfaCleanQDomain = 0;
+ aecm->dfaCleanQDomainOld = 0;
+ aecm->dfaNoisyQDomain = 0;
+ aecm->dfaNoisyQDomainOld = 0;
+
+ memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
+ aecm->farLogEnergy = 0;
+ memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
+ memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
+
+ // Initialize the echo channels with a stored shape.
+ if (samplingFreq == 8000)
+ {
+ WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
+ }
+ else
+ {
+ WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
+ }
+
+ memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
+ memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
+ aecm->noiseEstCtr = 0;
+
+ aecm->cngMode = AecmTrue;
+
+ memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
+ memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
+ // Shape the initial noise level to an approximate pink noise.
+ for (i = 0; i < (PART_LEN1 >> 1) - 1; i++)
+ {
+ aecm->noiseEst[i] = (tmp32 << 8);
+ tmp16--;
+ tmp32 -= (WebRtc_Word32)((tmp16 << 1) + 1);
+ }
+ for (; i < PART_LEN1; i++)
+ {
+ aecm->noiseEst[i] = (tmp32 << 8);
+ }
+
+ aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
+ aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
+ aecm->farEnergyMaxMin = 0;
+ aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
+ // beginning.
+ aecm->farEnergyMSE = 0;
+ aecm->currentVADValue = 0;
+ aecm->vadUpdateCount = 0;
+ aecm->firstVAD = 1;
+
+ aecm->startupState = 0;
+ aecm->supGain = SUPGAIN_DEFAULT;
+ aecm->supGainOld = SUPGAIN_DEFAULT;
+
+ aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
+ aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
+ aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
+ aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
+
+ assert(PART_LEN % 16 == 0);
+
+ // Initialize function pointers.
+ WebRtcAecm_WindowAndFFT = WindowAndFFTC;
+ WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowC;
+ WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
+ WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
+ WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
+
+#ifdef WEBRTC_DETECT_ARM_NEON
+ uint64_t features = WebRtc_GetCPUFeaturesARM();
+ if ((features & kCPUFeatureNEON) != 0)
+ {
+ WebRtcAecm_InitNeon();
+ }
+#elif defined(WEBRTC_ARCH_ARM_NEON)
+ WebRtcAecm_InitNeon();
+#endif
+
+ return 0;
+}
+
+// TODO(bjornv): This function is currently not used. Add support for these
+// parameters from a higher level
+int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag)
+{
+ aecm->nlpFlag = nlpFlag;
+ aecm->fixedDelay = delay;
+
+ return 0;
+}
+
+int WebRtcAecm_FreeCore(AecmCore_t *aecm)
+{
+ if (aecm == NULL)
+ {
+ return -1;
+ }
+
+ WebRtc_FreeBuffer(aecm->farFrameBuf);
+ WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
+ WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
+ WebRtc_FreeBuffer(aecm->outFrameBuf);
+
+ WebRtc_FreeDelayEstimator(aecm->delay_estimator);
+ free(aecm);
+
+ return 0;
+}
+
+int WebRtcAecm_ProcessFrame(AecmCore_t * aecm,
+ const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * out)
+{
+ WebRtc_Word16 outBlock_buf[PART_LEN + 8]; // Align buffer to 8-byte boundary.
+ WebRtc_Word16* outBlock = (WebRtc_Word16*) (((uintptr_t) outBlock_buf + 15) & ~ 15);
+
+ WebRtc_Word16 farFrame[FRAME_LEN];
+ const int16_t* out_ptr = NULL;
+ int size = 0;
+
+ // Buffer the current frame.
+ // Fetch an older one corresponding to the delay.
+ WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
+ WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
+
+ // Buffer the synchronized far and near frames,
+ // to pass the smaller blocks individually.
+ WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
+ WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
+ if (nearendClean != NULL)
+ {
+ WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
+ }
+
+ // Process as many blocks as possible.
+ while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN)
+ {
+ int16_t far_block[PART_LEN];
+ const int16_t* far_block_ptr = NULL;
+ int16_t near_noisy_block[PART_LEN];
+ const int16_t* near_noisy_block_ptr = NULL;
+
+ WebRtc_ReadBuffer(aecm->farFrameBuf, (void**) &far_block_ptr, far_block,
+ PART_LEN);
+ WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf,
+ (void**) &near_noisy_block_ptr,
+ near_noisy_block,
+ PART_LEN);
+ if (nearendClean != NULL)
+ {
+ int16_t near_clean_block[PART_LEN];
+ const int16_t* near_clean_block_ptr = NULL;
+
+ WebRtc_ReadBuffer(aecm->nearCleanFrameBuf,
+ (void**) &near_clean_block_ptr,
+ near_clean_block,
+ PART_LEN);
+ if (WebRtcAecm_ProcessBlock(aecm,
+ far_block_ptr,
+ near_noisy_block_ptr,
+ near_clean_block_ptr,
+ outBlock) == -1)
+ {
+ return -1;
+ }
+ } else
+ {
+ if (WebRtcAecm_ProcessBlock(aecm,
+ far_block_ptr,
+ near_noisy_block_ptr,
+ NULL,
+ outBlock) == -1)
+ {
+ return -1;
+ }
+ }
+
+ WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
+ }
+
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ size = (int) WebRtc_available_read(aecm->outFrameBuf);
+ if (size < FRAME_LEN)
+ {
+ WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
+ }
+
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aecm->outFrameBuf, (void**) &out_ptr, out, FRAME_LEN);
+ if (out_ptr != out) {
+ // ReadBuffer() hasn't copied to |out| in this case.
+ memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
+ }
+
+ return 0;
+}
+
+// WebRtcAecm_AsymFilt(...)
+//
+// Performs asymmetric filtering.
+//
+// Inputs:
+// - filtOld : Previous filtered value.
+// - inVal : New input value.
+// - stepSizePos : Step size when we have a positive contribution.
+// - stepSizeNeg : Step size when we have a negative contribution.
+//
+// Output:
+//
+// Return: - Filtered value.
+//
+WebRtc_Word16 WebRtcAecm_AsymFilt(const WebRtc_Word16 filtOld, const WebRtc_Word16 inVal,
+ const WebRtc_Word16 stepSizePos,
+ const WebRtc_Word16 stepSizeNeg)
+{
+ WebRtc_Word16 retVal;
+
+ if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
+ {
+ return inVal;
+ }
+ retVal = filtOld;
+ if (filtOld > inVal)
+ {
+ retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
+ } else
+ {
+ retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
+ }
+
+ return retVal;
+}
+
+// WebRtcAecm_CalcEnergies(...)
+//
+// This function calculates the log of energies for nearend, farend and estimated
+// echoes. There is also an update of energy decision levels, i.e. internal VAD.
+//
+//
+// @param aecm [i/o] Handle of the AECM instance.
+// @param far_spectrum [in] Pointer to farend spectrum.
+// @param far_q [in] Q-domain of farend spectrum.
+// @param nearEner [in] Near end energy for current block in
+// Q(aecm->dfaQDomain).
+// @param echoEst [out] Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
+ const WebRtc_UWord16* far_spectrum,
+ const WebRtc_Word16 far_q,
+ const WebRtc_UWord32 nearEner,
+ WebRtc_Word32 * echoEst)
+{
+ // Local variables
+ WebRtc_UWord32 tmpAdapt = 0;
+ WebRtc_UWord32 tmpStored = 0;
+ WebRtc_UWord32 tmpFar = 0;
+
+ int i;
+
+ WebRtc_Word16 zeros, frac;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word16 increase_max_shifts = 4;
+ WebRtc_Word16 decrease_max_shifts = 11;
+ WebRtc_Word16 increase_min_shifts = 11;
+ WebRtc_Word16 decrease_min_shifts = 3;
+ WebRtc_Word16 kLogLowValue = WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
+
+ // Get log of near end energy and store in buffer
+
+ // Shift buffer
+ memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+
+ // Logarithm of integrated magnitude spectrum (nearEner)
+ tmp16 = kLogLowValue;
+ if (nearEner)
+ {
+ zeros = WebRtcSpl_NormU32(nearEner);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
+ (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
+ 23);
+ // log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
+ }
+ aecm->nearLogEnergy[0] = tmp16;
+ // END: Get log of near end energy
+
+ WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt, &tmpStored);
+
+ // Shift buffers
+ memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+ memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
+ sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
+
+ // Logarithm of delayed far end energy
+ tmp16 = kLogLowValue;
+ if (tmpFar)
+ {
+ zeros = WebRtcSpl_NormU32(tmpFar);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
+ & 0x7FFFFFFF), 23);
+ // log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8);
+ }
+ aecm->farLogEnergy = tmp16;
+
+ // Logarithm of estimated echo energy through adapted channel
+ tmp16 = kLogLowValue;
+ if (tmpAdapt)
+ {
+ zeros = WebRtcSpl_NormU32(tmpAdapt);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
+ & 0x7FFFFFFF), 23);
+ //log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
+ }
+ aecm->echoAdaptLogEnergy[0] = tmp16;
+
+ // Logarithm of estimated echo energy through stored channel
+ tmp16 = kLogLowValue;
+ if (tmpStored)
+ {
+ zeros = WebRtcSpl_NormU32(tmpStored);
+ frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
+ & 0x7FFFFFFF), 23);
+ //log2 in Q8
+ tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
+ tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
+ }
+ aecm->echoStoredLogEnergy[0] = tmp16;
+
+ // Update farend energy levels (min, max, vad, mse)
+ if (aecm->farLogEnergy > FAR_ENERGY_MIN)
+ {
+ if (aecm->startupState == 0)
+ {
+ increase_max_shifts = 2;
+ decrease_min_shifts = 2;
+ increase_min_shifts = 8;
+ }
+
+ aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
+ increase_min_shifts, decrease_min_shifts);
+ aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
+ increase_max_shifts, decrease_max_shifts);
+ aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
+
+ // Dynamic VAD region size
+ tmp16 = 2560 - aecm->farEnergyMin;
+ if (tmp16 > 0)
+ {
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9);
+ } else
+ {
+ tmp16 = 0;
+ }
+ tmp16 += FAR_ENERGY_VAD_REGION;
+
+ if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
+ {
+ // In startup phase or VAD update halted
+ aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
+ } else
+ {
+ if (aecm->farEnergyVAD > aecm->farLogEnergy)
+ {
+ aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy +
+ tmp16 -
+ aecm->farEnergyVAD,
+ 6);
+ aecm->vadUpdateCount = 0;
+ } else
+ {
+ aecm->vadUpdateCount++;
+ }
+ }
+ // Put MSE threshold higher than VAD
+ aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
+ }
+
+ // Update VAD variables
+ if (aecm->farLogEnergy > aecm->farEnergyVAD)
+ {
+ if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
+ {
+ // We are in startup or have significant dynamics in input speech level
+ aecm->currentVADValue = 1;
+ }
+ } else
+ {
+ aecm->currentVADValue = 0;
+ }
+ if ((aecm->currentVADValue) && (aecm->firstVAD))
+ {
+ aecm->firstVAD = 0;
+ if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
+ {
+ // The estimated echo has higher energy than the near end signal.
+ // This means that the initialization was too aggressive. Scale
+ // down by a factor 8
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ aecm->channelAdapt16[i] >>= 3;
+ }
+ // Compensate the adapted echo energy level accordingly.
+ aecm->echoAdaptLogEnergy[0] -= (3 << 8);
+ aecm->firstVAD = 1;
+ }
+ }
+}
+
+// WebRtcAecm_CalcStepSize(...)
+//
+// This function calculates the step size used in channel estimation
+//
+//
+// @param aecm [in] Handle of the AECM instance.
+// @param mu [out] (Return value) Stepsize in log2(), i.e. number of shifts.
+//
+//
+WebRtc_Word16 WebRtcAecm_CalcStepSize(AecmCore_t * const aecm)
+{
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word16 mu = MU_MAX;
+
+ // Here we calculate the step size mu used in the
+ // following NLMS based Channel estimation algorithm
+ if (!aecm->currentVADValue)
+ {
+ // Far end energy level too low, no channel update
+ mu = 0;
+ } else if (aecm->startupState > 0)
+ {
+ if (aecm->farEnergyMin >= aecm->farEnergyMax)
+ {
+ mu = MU_MIN;
+ } else
+ {
+ tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
+ tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, MU_DIFF);
+ tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
+ mu = MU_MIN - 1 - (WebRtc_Word16)(tmp32);
+ // The -1 is an alternative to rounding. This way we get a larger
+ // stepsize, so we in some sense compensate for truncation in NLMS
+ }
+ if (mu < MU_MAX)
+ {
+ mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
+ }
+ }
+
+ return mu;
+}
+
+// WebRtcAecm_UpdateChannel(...)
+//
+// This function performs channel estimation. NLMS and decision on channel storage.
+//
+//
+// @param aecm [i/o] Handle of the AECM instance.
+// @param far_spectrum [in] Absolute value of the farend signal in Q(far_q)
+// @param far_q [in] Q-domain of the farend signal
+// @param dfa [in] Absolute value of the nearend signal (Q[aecm->dfaQDomain])
+// @param mu [in] NLMS step size.
+// @param echoEst [i/o] Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_UpdateChannel(AecmCore_t * aecm,
+ const WebRtc_UWord16* far_spectrum,
+ const WebRtc_Word16 far_q,
+ const WebRtc_UWord16 * const dfa,
+ const WebRtc_Word16 mu,
+ WebRtc_Word32 * echoEst)
+{
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 mseStored;
+ WebRtc_Word32 mseAdapt;
+
+ int i;
+
+ WebRtc_Word16 zerosFar, zerosNum, zerosCh, zerosDfa;
+ WebRtc_Word16 shiftChFar, shiftNum, shift2ResChan;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 xfaQ, dfaQ;
+
+ // This is the channel estimation algorithm. It is base on NLMS but has a variable step
+ // length, which was calculated above.
+ if (mu)
+ {
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Determine norm of channel and farend to make sure we don't get overflow in
+ // multiplication
+ zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
+ zerosFar = WebRtcSpl_NormU32((WebRtc_UWord32)far_spectrum[i]);
+ if (zerosCh + zerosFar > 31)
+ {
+ // Multiplication is safe
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
+ far_spectrum[i]);
+ shiftChFar = 0;
+ } else
+ {
+ // We need to shift down before multiplication
+ shiftChFar = 32 - zerosCh - zerosFar;
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], shiftChFar),
+ far_spectrum[i]);
+ }
+ // Determine Q-domain of numerator
+ zerosNum = WebRtcSpl_NormU32(tmpU32no1);
+ if (dfa[i])
+ {
+ zerosDfa = WebRtcSpl_NormU32((WebRtc_UWord32)dfa[i]);
+ } else
+ {
+ zerosDfa = 32;
+ }
+ tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain -
+ RESOLUTION_CHANNEL32 - far_q + shiftChFar;
+ if (zerosNum > tmp16no1 + 1)
+ {
+ xfaQ = tmp16no1;
+ dfaQ = zerosDfa - 2;
+ } else
+ {
+ xfaQ = zerosNum - 2;
+ dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
+ shiftChFar + xfaQ;
+ }
+ // Add in the same Q-domain
+ tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
+ tmpU32no2 = WEBRTC_SPL_SHIFT_W32((WebRtc_UWord32)dfa[i], dfaQ);
+ tmp32no1 = (WebRtc_Word32)tmpU32no2 - (WebRtc_Word32)tmpU32no1;
+ zerosNum = WebRtcSpl_NormW32(tmp32no1);
+ if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q)))
+ {
+ //
+ // Update is needed
+ //
+ // This is what we would like to compute
+ //
+ // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
+ // tmp32norm = (i + 1)
+ // aecm->channelAdapt[i] += (2^mu) * tmp32no1
+ // / (tmp32norm * far_spectrum[i])
+ //
+
+ // Make sure we don't get overflow in multiplication.
+ if (zerosNum + zerosFar > 31)
+ {
+ if (tmp32no1 > 0)
+ {
+ tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmp32no1,
+ far_spectrum[i]);
+ } else
+ {
+ tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
+ far_spectrum[i]);
+ }
+ shiftNum = 0;
+ } else
+ {
+ shiftNum = 32 - (zerosNum + zerosFar);
+ if (tmp32no1 > 0)
+ {
+ tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(tmp32no1, shiftNum),
+ far_spectrum[i]);
+ } else
+ {
+ tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
+ WEBRTC_SPL_RSHIFT_W32(-tmp32no1, shiftNum),
+ far_spectrum[i]);
+ }
+ }
+ // Normalize with respect to frequency bin
+ tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
+ // Make sure we are in the right Q-domain
+ shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
+ if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
+ {
+ tmp32no2 = WEBRTC_SPL_WORD32_MAX;
+ } else
+ {
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
+ }
+ aecm->channelAdapt32[i] = WEBRTC_SPL_ADD_SAT_W32(aecm->channelAdapt32[i],
+ tmp32no2);
+ if (aecm->channelAdapt32[i] < 0)
+ {
+ // We can never have negative channel gain
+ aecm->channelAdapt32[i] = 0;
+ }
+ aecm->channelAdapt16[i]
+ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], 16);
+ }
+ }
+ }
+ // END: Adaptive channel update
+
+ // Determine if we should store or restore the channel
+ if ((aecm->startupState == 0) & (aecm->currentVADValue))
+ {
+ // During startup we store the channel every block,
+ // and we recalculate echo estimate
+ WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+ } else
+ {
+ if (aecm->farLogEnergy < aecm->farEnergyMSE)
+ {
+ aecm->mseChannelCount = 0;
+ } else
+ {
+ aecm->mseChannelCount++;
+ }
+ // Enough data for validation. Store channel if we can.
+ if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
+ {
+ // We have enough data.
+ // Calculate MSE of "Adapt" and "Stored" versions.
+ // It is actually not MSE, but average absolute error.
+ mseStored = 0;
+ mseAdapt = 0;
+ for (i = 0; i < MIN_MSE_COUNT; i++)
+ {
+ tmp32no1 = ((WebRtc_Word32)aecm->echoStoredLogEnergy[i]
+ - (WebRtc_Word32)aecm->nearLogEnergy[i]);
+ tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+ mseStored += tmp32no2;
+
+ tmp32no1 = ((WebRtc_Word32)aecm->echoAdaptLogEnergy[i]
+ - (WebRtc_Word32)aecm->nearLogEnergy[i]);
+ tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+ mseAdapt += tmp32no2;
+ }
+ if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
+ & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
+ * aecm->mseAdaptOld)))
+ {
+ // The stored channel has a significantly lower MSE than the adaptive one for
+ // two consecutive calculations. Reset the adaptive channel.
+ WebRtcAecm_ResetAdaptiveChannel(aecm);
+ } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
+ < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
+ {
+ // The adaptive channel has a significantly lower MSE than the stored one.
+ // The MSE for the adaptive channel has also been low for two consecutive
+ // calculations. Store the adaptive channel.
+ WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+
+ // Update threshold
+ if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
+ {
+ aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
+ } else
+ {
+ aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt
+ - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8);
+ }
+
+ }
+
+ // Reset counter
+ aecm->mseChannelCount = 0;
+
+ // Store the MSE values.
+ aecm->mseStoredOld = mseStored;
+ aecm->mseAdaptOld = mseAdapt;
+ }
+ }
+ // END: Determine if we should store or reset channel estimate.
+}
+
+// CalcSuppressionGain(...)
+//
+// This function calculates the suppression gain that is used in the Wiener filter.
+//
+//
+// @param aecm [i/n] Handle of the AECM instance.
+// @param supGain [out] (Return value) Suppression gain with which to scale the noise
+// level (Q14).
+//
+//
+static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm)
+{
+ WebRtc_Word32 tmp32no1;
+
+ WebRtc_Word16 supGain = SUPGAIN_DEFAULT;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 dE = 0;
+
+ // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
+ // end energy and echo estimation error.
+ // Adjust for the far end signal level. A low signal level indicates no far end signal,
+ // hence we set the suppression gain to 0
+ if (!aecm->currentVADValue)
+ {
+ supGain = 0;
+ } else
+ {
+ // Adjust for possible double talk. If we have large variations in estimation error we
+ // likely have double talk (or poor channel).
+ tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
+ dE = WEBRTC_SPL_ABS_W16(tmp16no1);
+
+ if (dE < ENERGY_DEV_TOL)
+ {
+ // Likely no double talk. The better estimation, the more we can suppress signal.
+ // Update counters
+ if (dE < SUPGAIN_EPC_DT)
+ {
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffAB, dE);
+ tmp32no1 += (SUPGAIN_EPC_DT >> 1);
+ tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
+ supGain = aecm->supGainErrParamA - tmp16no1;
+ } else
+ {
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffBD,
+ (ENERGY_DEV_TOL - dE));
+ tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
+ tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
+ - SUPGAIN_EPC_DT));
+ supGain = aecm->supGainErrParamD + tmp16no1;
+ }
+ } else
+ {
+ // Likely in double talk. Use default value
+ supGain = aecm->supGainErrParamD;
+ }
+ }
+
+ if (supGain > aecm->supGainOld)
+ {
+ tmp16no1 = supGain;
+ } else
+ {
+ tmp16no1 = aecm->supGainOld;
+ }
+ aecm->supGainOld = supGain;
+ if (tmp16no1 < aecm->supGain)
+ {
+ aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
+ } else
+ {
+ aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
+ }
+
+ // END: Update suppression gain
+
+ return aecm->supGain;
+}
+
+// Transforms a time domain signal into the frequency domain, outputting the
+// complex valued signal, absolute value and sum of absolute values.
+//
+// time_signal [in] Pointer to time domain signal
+// freq_signal_real [out] Pointer to real part of frequency domain array
+// freq_signal_imag [out] Pointer to imaginary part of frequency domain
+// array
+// freq_signal_abs [out] Pointer to absolute value of frequency domain
+// array
+// freq_signal_sum_abs [out] Pointer to the sum of all absolute values in
+// the frequency domain array
+// return value The Q-domain of current frequency values
+//
+static int TimeToFrequencyDomain(const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ WebRtc_UWord16* freq_signal_abs,
+ WebRtc_UWord32* freq_signal_sum_abs)
+{
+ int i = 0;
+ int time_signal_scaling = 0;
+
+ WebRtc_Word32 tmp32no1;
+ WebRtc_Word32 tmp32no2;
+
+ // In fft_buf, +16 for 32-byte alignment.
+ WebRtc_Word16 fft_buf[PART_LEN4 + 16];
+ WebRtc_Word16 *fft = (WebRtc_Word16 *) (((uintptr_t) fft_buf + 31) & ~31);
+
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 tmp16no2;
+#ifdef AECM_WITH_ABS_APPROX
+ WebRtc_Word16 max_value = 0;
+ WebRtc_Word16 min_value = 0;
+ WebRtc_UWord16 alpha = 0;
+ WebRtc_UWord16 beta = 0;
+#endif
+
+#ifdef AECM_DYNAMIC_Q
+ tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
+ time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
+#endif
+
+ WebRtcAecm_WindowAndFFT(fft, time_signal, freq_signal, time_signal_scaling);
+
+ // Extract imaginary and real part, calculate the magnitude for all frequency bins
+ freq_signal[0].imag = 0;
+ freq_signal[PART_LEN].imag = 0;
+ freq_signal[PART_LEN].real = fft[PART_LEN2];
+ freq_signal_abs[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[0].real);
+ freq_signal_abs[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[PART_LEN].real);
+ (*freq_signal_sum_abs) = (WebRtc_UWord32)(freq_signal_abs[0]) +
+ (WebRtc_UWord32)(freq_signal_abs[PART_LEN]);
+
+ for (i = 1; i < PART_LEN; i++)
+ {
+ if (freq_signal[i].real == 0)
+ {
+ freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[i].imag);
+ }
+ else if (freq_signal[i].imag == 0)
+ {
+ freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
+ freq_signal[i].real);
+ }
+ else
+ {
+ // Approximation for magnitude of complex fft output
+ // magn = sqrt(real^2 + imag^2)
+ // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
+ //
+ // The parameters alpha and beta are stored in Q15
+
+#ifdef AECM_WITH_ABS_APPROX
+ tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+ tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+
+ if(tmp16no1 > tmp16no2)
+ {
+ max_value = tmp16no1;
+ min_value = tmp16no2;
+ } else
+ {
+ max_value = tmp16no2;
+ min_value = tmp16no1;
+ }
+
+ // Magnitude in Q(-6)
+ if ((max_value >> 2) > min_value)
+ {
+ alpha = kAlpha1;
+ beta = kBeta1;
+ } else if ((max_value >> 1) > min_value)
+ {
+ alpha = kAlpha2;
+ beta = kBeta2;
+ } else
+ {
+ alpha = kAlpha3;
+ beta = kBeta3;
+ }
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(max_value,
+ alpha,
+ 15);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(min_value,
+ beta,
+ 15);
+ freq_signal_abs[i] = (WebRtc_UWord16)tmp16no1 +
+ (WebRtc_UWord16)tmp16no2;
+#else
+#ifdef WEBRTC_ARCH_ARM_V7A
+ __asm__("smulbb %0, %1, %2" : "=r"(tmp32no1) : "r"(freq_signal[i].real),
+ "r"(freq_signal[i].real));
+ __asm__("smlabb %0, %1, %2, %3" :: "r"(tmp32no2), "r"(freq_signal[i].imag),
+ "r"(freq_signal[i].imag), "r"(tmp32no1));
+#else
+ tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+ tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+ tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
+ tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
+#endif // WEBRTC_ARCH_ARM_V7A
+ tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
+
+ freq_signal_abs[i] = (WebRtc_UWord16)tmp32no1;
+#endif // AECM_WITH_ABS_APPROX
+ }
+ (*freq_signal_sum_abs) += (WebRtc_UWord32)freq_signal_abs[i];
+ }
+
+ return time_signal_scaling;
+}
+
+int WebRtcAecm_ProcessBlock(AecmCore_t * aecm,
+ const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * output)
+{
+ int i;
+
+ WebRtc_UWord32 xfaSum;
+ WebRtc_UWord32 dfaNoisySum;
+ WebRtc_UWord32 dfaCleanSum;
+ WebRtc_UWord32 echoEst32Gained;
+ WebRtc_UWord32 tmpU32;
+
+ WebRtc_Word32 tmp32no1;
+
+ WebRtc_UWord16 xfa[PART_LEN1];
+ WebRtc_UWord16 dfaNoisy[PART_LEN1];
+ WebRtc_UWord16 dfaClean[PART_LEN1];
+ WebRtc_UWord16* ptrDfaClean = dfaClean;
+ const WebRtc_UWord16* far_spectrum_ptr = NULL;
+
+ // 32 byte aligned buffers (with +8 or +16).
+ // TODO (kma): define fft with complex16_t.
+ WebRtc_Word16 fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
+ WebRtc_Word32 echoEst32_buf[PART_LEN1 + 8];
+ WebRtc_Word32 dfw_buf[PART_LEN1 + 8];
+ WebRtc_Word32 efw_buf[PART_LEN1 + 8];
+
+ WebRtc_Word16* fft = (WebRtc_Word16*) (((uintptr_t) fft_buf + 31) & ~ 31);
+ WebRtc_Word32* echoEst32 = (WebRtc_Word32*) (((uintptr_t) echoEst32_buf + 31) & ~ 31);
+ complex16_t* dfw = (complex16_t*) (((uintptr_t) dfw_buf + 31) & ~ 31);
+ complex16_t* efw = (complex16_t*) (((uintptr_t) efw_buf + 31) & ~ 31);
+
+ WebRtc_Word16 hnl[PART_LEN1];
+ WebRtc_Word16 numPosCoef = 0;
+ WebRtc_Word16 nlpGain = ONE_Q14;
+ int delay;
+ WebRtc_Word16 tmp16no1;
+ WebRtc_Word16 tmp16no2;
+ WebRtc_Word16 mu;
+ WebRtc_Word16 supGain;
+ WebRtc_Word16 zeros32, zeros16;
+ WebRtc_Word16 zerosDBufNoisy, zerosDBufClean, zerosXBuf;
+ int far_q;
+ WebRtc_Word16 resolutionDiff, qDomainDiff;
+
+ const int kMinPrefBand = 4;
+ const int kMaxPrefBand = 24;
+ WebRtc_Word32 avgHnl32 = 0;
+
+#ifdef ARM_WINM_LOG_
+ DWORD temp;
+ static int flag0 = 0;
+ __int64 freq, start, end, diff__;
+ unsigned int milliseconds;
+#endif
+
+ // Determine startup state. There are three states:
+ // (0) the first CONV_LEN blocks
+ // (1) another CONV_LEN blocks
+ // (2) the rest
+
+ if (aecm->startupState < 2)
+ {
+ aecm->startupState = (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
+ }
+ // END: Determine startup state
+
+ // Buffer near and far end signals
+ memcpy(aecm->xBuf + PART_LEN, farend, sizeof(WebRtc_Word16) * PART_LEN);
+ memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(WebRtc_Word16) * PART_LEN);
+ if (nearendClean != NULL)
+ {
+ memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(WebRtc_Word16) * PART_LEN);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick start
+ QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Transform far end signal from time domain to frequency domain.
+ far_q = TimeToFrequencyDomain(aecm->xBuf,
+ dfw,
+ xfa,
+ &xfaSum);
+
+ // Transform noisy near end signal from time domain to frequency domain.
+ zerosDBufNoisy = TimeToFrequencyDomain(aecm->dBufNoisy,
+ dfw,
+ dfaNoisy,
+ &dfaNoisySum);
+ aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
+ aecm->dfaNoisyQDomain = (WebRtc_Word16)zerosDBufNoisy;
+
+
+ if (nearendClean == NULL)
+ {
+ ptrDfaClean = dfaNoisy;
+ aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
+ aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
+ dfaCleanSum = dfaNoisySum;
+ } else
+ {
+ // Transform clean near end signal from time domain to frequency domain.
+ zerosDBufClean = TimeToFrequencyDomain(aecm->dBufClean,
+ dfw,
+ dfaClean,
+ &dfaCleanSum);
+ aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
+ aecm->dfaCleanQDomain = (WebRtc_Word16)zerosDBufClean;
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Get the delay
+ // Save far-end history and estimate delay
+ UpdateFarHistory(aecm, xfa, far_q);
+ delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator,
+ xfa,
+ dfaNoisy,
+ PART_LEN1,
+ far_q,
+ zerosDBufNoisy);
+ if (delay == -1)
+ {
+ return -1;
+ }
+ else if (delay == -2)
+ {
+ // If the delay is unknown, we assume zero.
+ // NOTE: this will have to be adjusted if we ever add lookahead.
+ delay = 0;
+ }
+
+ if (aecm->fixedDelay >= 0)
+ {
+ // Use fixed delay
+ delay = aecm->fixedDelay;
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+ // Get aligned far end spectrum
+ far_spectrum_ptr = AlignedFarend(aecm, &far_q, delay);
+ zerosXBuf = (WebRtc_Word16) far_q;
+ if (far_spectrum_ptr == NULL)
+ {
+ return -1;
+ }
+
+ // Calculate log(energy) and update energy threshold levels
+ WebRtcAecm_CalcEnergies(aecm,
+ far_spectrum_ptr,
+ zerosXBuf,
+ dfaNoisySum,
+ echoEst32);
+
+ // Calculate stepsize
+ mu = WebRtcAecm_CalcStepSize(aecm);
+
+ // Update counters
+ aecm->totCount++;
+
+ // This is the channel estimation algorithm.
+ // It is base on NLMS but has a variable step length, which was calculated above.
+ WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu, echoEst32);
+ supGain = CalcSuppressionGain(aecm);
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Calculate Wiener filter hnl[]
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Far end signal through channel estimate in Q8
+ // How much can we shift right to preserve resolution
+ tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
+ aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
+
+ zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
+ zeros16 = WebRtcSpl_NormW16(supGain) + 1;
+ if (zeros32 + zeros16 > 16)
+ {
+ // Multiplication is safe
+ // Result in Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
+ (WebRtc_UWord16)supGain);
+ resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+ resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+ } else
+ {
+ tmp16no1 = 17 - zeros32 - zeros16;
+ resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+ resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+ if (zeros32 > tmp16no1)
+ {
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
+ (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W16(supGain,
+ tmp16no1)); // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+ } else
+ {
+ // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+ echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
+ (WebRtc_UWord32)WEBRTC_SPL_RSHIFT_W32(aecm->echoFilt[i], tmp16no1),
+ (WebRtc_UWord16)supGain);
+ }
+ }
+
+ zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
+ if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld))
+ & (aecm->nearFilt[i]))
+ {
+ tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16);
+ qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld;
+ } else
+ {
+ tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i],
+ aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld);
+ qDomainDiff = 0;
+ }
+ tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff);
+ tmp32no1 = (WebRtc_Word32)(tmp16no2 - tmp16no1);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 4);
+ tmp16no2 += tmp16no1;
+ zeros16 = WebRtcSpl_NormW16(tmp16no2);
+ if ((tmp16no2) & (-qDomainDiff > zeros16))
+ {
+ aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
+ } else
+ {
+ aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff);
+ }
+
+ // Wiener filter coefficients, resulting hnl in Q14
+ if (echoEst32Gained == 0)
+ {
+ hnl[i] = ONE_Q14;
+ } else if (aecm->nearFilt[i] == 0)
+ {
+ hnl[i] = 0;
+ } else
+ {
+ // Multiply the suppression gain
+ // Rounding
+ echoEst32Gained += (WebRtc_UWord32)(aecm->nearFilt[i] >> 1);
+ tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained, (WebRtc_UWord16)aecm->nearFilt[i]);
+
+ // Current resolution is
+ // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN - max(0, 17 - zeros16 - zeros32))
+ // Make sure we are in Q14
+ tmp32no1 = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
+ if (tmp32no1 > ONE_Q14)
+ {
+ hnl[i] = 0;
+ } else if (tmp32no1 < 0)
+ {
+ hnl[i] = ONE_Q14;
+ } else
+ {
+ // 1-echoEst/dfa
+ hnl[i] = ONE_Q14 - (WebRtc_Word16)tmp32no1;
+ if (hnl[i] < 0)
+ {
+ hnl[i] = 0;
+ }
+ }
+ }
+ if (hnl[i])
+ {
+ numPosCoef++;
+ }
+ }
+ // Only in wideband. Prevent the gain in upper band from being larger than
+ // in lower band.
+ if (aecm->mult == 2)
+ {
+ // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
+ // speech distortion in double-talk.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], hnl[i], 14);
+ }
+
+ for (i = kMinPrefBand; i <= kMaxPrefBand; i++)
+ {
+ avgHnl32 += (WebRtc_Word32)hnl[i];
+ }
+ assert(kMaxPrefBand - kMinPrefBand + 1 > 0);
+ avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
+
+ for (i = kMaxPrefBand; i < PART_LEN1; i++)
+ {
+ if (hnl[i] > (WebRtc_Word16)avgHnl32)
+ {
+ hnl[i] = (WebRtc_Word16)avgHnl32;
+ }
+ }
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ // Calculate NLP gain, result is in Q14
+ if (aecm->nlpFlag)
+ {
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ // Truncate values close to zero and one.
+ if (hnl[i] > NLP_COMP_HIGH)
+ {
+ hnl[i] = ONE_Q14;
+ } else if (hnl[i] < NLP_COMP_LOW)
+ {
+ hnl[i] = 0;
+ }
+
+ // Remove outliers
+ if (numPosCoef < 3)
+ {
+ nlpGain = 0;
+ } else
+ {
+ nlpGain = ONE_Q14;
+ }
+
+ // NLP
+ if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
+ {
+ hnl[i] = ONE_Q14;
+ } else
+ {
+ hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
+ }
+
+ // multiply with Wiener coefficients
+ efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
+ hnl[i], 14));
+ efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
+ hnl[i], 14));
+ }
+ }
+ else
+ {
+ // multiply with Wiener coefficients
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
+ hnl[i], 14));
+ efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
+ hnl[i], 14));
+ }
+ }
+
+ if (aecm->cngMode == AecmTrue)
+ {
+ ComfortNoise(aecm, ptrDfaClean, efw, hnl);
+ }
+
+#ifdef ARM_WINM_LOG_
+ // measure tick end
+ QueryPerformanceCounter((LARGE_INTEGER*)&end);
+ diff__ = ((end - start) * 1000) / (freq/1000);
+ milliseconds = (unsigned int)(diff__ & 0xffffffff);
+ WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
+ // measure tick start
+ QueryPerformanceCounter((LARGE_INTEGER*)&start);
+#endif
+
+ WebRtcAecm_InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
+
+ return 0;
+}
+
+
+// Generate comfort noise and add to output signal.
+//
+// \param[in] aecm Handle of the AECM instance.
+// \param[in] dfa Absolute value of the nearend signal (Q[aecm->dfaQDomain]).
+// \param[in,out] outReal Real part of the output signal (Q[aecm->dfaQDomain]).
+// \param[in,out] outImag Imaginary part of the output signal (Q[aecm->dfaQDomain]).
+// \param[in] lambda Suppression gain with which to scale the noise level (Q14).
+//
+static void ComfortNoise(AecmCore_t* aecm,
+ const WebRtc_UWord16* dfa,
+ complex16_t* out,
+ const WebRtc_Word16* lambda)
+{
+ WebRtc_Word16 i;
+ WebRtc_Word16 tmp16;
+ WebRtc_Word32 tmp32;
+
+ WebRtc_Word16 randW16[PART_LEN];
+ WebRtc_Word16 uReal[PART_LEN1];
+ WebRtc_Word16 uImag[PART_LEN1];
+ WebRtc_Word32 outLShift32;
+ WebRtc_Word16 noiseRShift16[PART_LEN1];
+
+ WebRtc_Word16 shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
+ WebRtc_Word16 minTrackShift;
+
+ assert(shiftFromNearToNoise >= 0);
+ assert(shiftFromNearToNoise < 16);
+
+ if (aecm->noiseEstCtr < 100)
+ {
+ // Track the minimum more quickly initially.
+ aecm->noiseEstCtr++;
+ minTrackShift = 6;
+ } else
+ {
+ minTrackShift = 9;
+ }
+
+ // Estimate noise power.
+ for (i = 0; i < PART_LEN1; i++)
+ {
+
+ // Shift to the noise domain.
+ tmp32 = (WebRtc_Word32)dfa[i];
+ outLShift32 = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+
+ if (outLShift32 < aecm->noiseEst[i])
+ {
+ // Reset "too low" counter
+ aecm->noiseEstTooLowCtr[i] = 0;
+ // Track the minimum.
+ if (aecm->noiseEst[i] < (1 << minTrackShift))
+ {
+ // For small values, decrease noiseEst[i] every
+ // |kNoiseEstIncCount| block. The regular approach below can not
+ // go further down due to truncation.
+ aecm->noiseEstTooHighCtr[i]++;
+ if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount)
+ {
+ aecm->noiseEst[i]--;
+ aecm->noiseEstTooHighCtr[i] = 0; // Reset the counter
+ }
+ }
+ else
+ {
+ aecm->noiseEst[i] -= ((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
+ }
+ } else
+ {
+ // Reset "too high" counter
+ aecm->noiseEstTooHighCtr[i] = 0;
+ // Ramp slowly upwards until we hit the minimum again.
+ if ((aecm->noiseEst[i] >> 19) > 0)
+ {
+ // Avoid overflow.
+ // Multiplication with 2049 will cause wrap around. Scale
+ // down first and then multiply
+ aecm->noiseEst[i] >>= 11;
+ aecm->noiseEst[i] *= 2049;
+ }
+ else if ((aecm->noiseEst[i] >> 11) > 0)
+ {
+ // Large enough for relative increase
+ aecm->noiseEst[i] *= 2049;
+ aecm->noiseEst[i] >>= 11;
+ }
+ else
+ {
+ // Make incremental increases based on size every
+ // |kNoiseEstIncCount| block
+ aecm->noiseEstTooLowCtr[i]++;
+ if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount)
+ {
+ aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
+ aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
+ }
+ }
+ }
+ }
+
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(aecm->noiseEst[i], shiftFromNearToNoise);
+ if (tmp32 > 32767)
+ {
+ tmp32 = 32767;
+ aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+ }
+ noiseRShift16[i] = (WebRtc_Word16)tmp32;
+
+ tmp16 = ONE_Q14 - lambda[i];
+ noiseRShift16[i]
+ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, noiseRShift16[i], 14);
+ }
+
+ // Generate a uniform random array on [0 2^15-1].
+ WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
+
+ // Generate noise according to estimated energy.
+ uReal[0] = 0; // Reject LF noise.
+ uImag[0] = 0;
+ for (i = 1; i < PART_LEN1; i++)
+ {
+ // Get a random index for the cos and sin tables over [0 359].
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
+
+ // Tables are in Q13.
+ uReal[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i],
+ kCosTable[tmp16], 13);
+ uImag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i],
+ kSinTable[tmp16], 13);
+ }
+ uImag[PART_LEN] = 0;
+
+#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
+ for (i = 0; i < PART_LEN1; i++)
+ {
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ }
+#else
+ for (i = 0; i < PART_LEN1 -1; )
+ {
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ i++;
+
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+ i++;
+ }
+ out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
+ out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
+#endif
+}
+
+void WebRtcAecm_BufferFarFrame(AecmCore_t* const aecm,
+ const WebRtc_Word16* const farend,
+ const int farLen)
+{
+ int writeLen = farLen, writePos = 0;
+
+ // Check if the write position must be wrapped
+ while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
+ {
+ // Write to remaining buffer space before wrapping
+ writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
+ memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+ sizeof(WebRtc_Word16) * writeLen);
+ aecm->farBufWritePos = 0;
+ writePos = writeLen;
+ writeLen = farLen - writeLen;
+ }
+
+ memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+ sizeof(WebRtc_Word16) * writeLen);
+ aecm->farBufWritePos += writeLen;
+}
+
+void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
+ const int farLen, const int knownDelay)
+{
+ int readLen = farLen;
+ int readPos = 0;
+ int delayChange = knownDelay - aecm->lastKnownDelay;
+
+ aecm->farBufReadPos -= delayChange;
+
+ // Check if delay forces a read position wrap
+ while (aecm->farBufReadPos < 0)
+ {
+ aecm->farBufReadPos += FAR_BUF_LEN;
+ }
+ while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
+ {
+ aecm->farBufReadPos -= FAR_BUF_LEN;
+ }
+
+ aecm->lastKnownDelay = knownDelay;
+
+ // Check if read position must be wrapped
+ while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
+ {
+
+ // Read from remaining buffer space before wrapping
+ readLen = FAR_BUF_LEN - aecm->farBufReadPos;
+ memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+ sizeof(WebRtc_Word16) * readLen);
+ aecm->farBufReadPos = 0;
+ readPos = readLen;
+ readLen = farLen - readLen;
+ }
+ memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+ sizeof(WebRtc_Word16) * readLen);
+ aecm->farBufReadPos += readLen;
+}
+
diff --git a/src/modules/audio_processing/aecm/main/source/aecm_core.h b/src/modules/audio_processing/aecm/aecm_core.h
similarity index 72%
rename from src/modules/audio_processing/aecm/main/source/aecm_core.h
rename to src/modules/audio_processing/aecm/aecm_core.h
index 5defbe4..0ec62ec 100644
--- a/src/modules/audio_processing/aecm/main/source/aecm_core.h
+++ b/src/modules/audio_processing/aecm/aecm_core.h
@@ -17,14 +17,8 @@
//#define AECM_WITH_ABS_APPROX
//#define AECM_SHORT // for 32 sample partition length (otherwise 64)
-// TODO(bjornv): These defines will be removed in final version.
-//#define STORE_CHANNEL_DATA
-//#define VAD_DATA
-
#include "typedefs.h"
#include "signal_processing_library.h"
-// TODO(bjornv): Will be removed in final version.
-//#include <stdio.h>
// Algorithm parameters
@@ -103,6 +97,13 @@
#define NLP_COMP_LOW 3277 // 0.2 in Q14
#define NLP_COMP_HIGH ONE_Q14 // 1 in Q14
+extern const WebRtc_Word16 WebRtcAecm_kSqrtHanning[];
+
+typedef struct {
+ WebRtc_Word16 real;
+ WebRtc_Word16 imag;
+} complex16_t;
+
typedef struct
{
int farBufWritePos;
@@ -116,50 +117,60 @@
void *nearCleanFrameBuf;
void *outFrameBuf;
- WebRtc_Word16 xBuf[PART_LEN2]; // farend
- WebRtc_Word16 dBufClean[PART_LEN2]; // nearend
- WebRtc_Word16 dBufNoisy[PART_LEN2]; // nearend
- WebRtc_Word16 outBuf[PART_LEN];
-
WebRtc_Word16 farBuf[FAR_BUF_LEN];
WebRtc_Word16 mult;
WebRtc_UWord32 seed;
// Delay estimation variables
- WebRtc_UWord16 medianYlogspec[PART_LEN1];
- WebRtc_UWord16 medianXlogspec[PART_LEN1];
- WebRtc_UWord16 medianBCount[MAX_DELAY];
- WebRtc_UWord16 xfaHistory[PART_LEN1][MAX_DELAY];
- WebRtc_Word16 delHistoryPos;
- WebRtc_UWord32 bxHistory[MAX_DELAY];
+ void* delay_estimator;
WebRtc_UWord16 currentDelay;
- WebRtc_UWord16 previousDelay;
- WebRtc_Word16 delayAdjust;
+ // Far end history variables
+ // TODO(bjornv): Replace |far_history| with ring_buffer.
+ uint16_t far_history[PART_LEN1 * MAX_DELAY];
+ int far_history_pos;
+ int far_q_domains[MAX_DELAY];
WebRtc_Word16 nlpFlag;
WebRtc_Word16 fixedDelay;
WebRtc_UWord32 totCount;
- WebRtc_Word16 xfaQDomainBuf[MAX_DELAY];
WebRtc_Word16 dfaCleanQDomain;
WebRtc_Word16 dfaCleanQDomainOld;
WebRtc_Word16 dfaNoisyQDomain;
WebRtc_Word16 dfaNoisyQDomainOld;
WebRtc_Word16 nearLogEnergy[MAX_BUF_LEN];
- WebRtc_Word16 farLogEnergy[MAX_BUF_LEN];
+ WebRtc_Word16 farLogEnergy;
WebRtc_Word16 echoAdaptLogEnergy[MAX_BUF_LEN];
WebRtc_Word16 echoStoredLogEnergy[MAX_BUF_LEN];
- WebRtc_Word16 channelAdapt16[PART_LEN1];
- WebRtc_Word32 channelAdapt32[PART_LEN1];
- WebRtc_Word16 channelStored[PART_LEN1];
+ // The extra 16 or 32 bytes in the following buffers are for alignment based Neon code.
+ // It's designed this way since the current GCC compiler can't align a buffer in 16 or 32
+ // byte boundaries properly.
+ WebRtc_Word16 channelStored_buf[PART_LEN1 + 8];
+ WebRtc_Word16 channelAdapt16_buf[PART_LEN1 + 8];
+ WebRtc_Word32 channelAdapt32_buf[PART_LEN1 + 8];
+ WebRtc_Word16 xBuf_buf[PART_LEN2 + 16]; // farend
+ WebRtc_Word16 dBufClean_buf[PART_LEN2 + 16]; // nearend
+ WebRtc_Word16 dBufNoisy_buf[PART_LEN2 + 16]; // nearend
+ WebRtc_Word16 outBuf_buf[PART_LEN + 8];
+
+ // Pointers to the above buffers
+ WebRtc_Word16 *channelStored;
+ WebRtc_Word16 *channelAdapt16;
+ WebRtc_Word32 *channelAdapt32;
+ WebRtc_Word16 *xBuf;
+ WebRtc_Word16 *dBufClean;
+ WebRtc_Word16 *dBufNoisy;
+ WebRtc_Word16 *outBuf;
+
WebRtc_Word32 echoFilt[PART_LEN1];
WebRtc_Word16 nearFilt[PART_LEN1];
WebRtc_Word32 noiseEst[PART_LEN1];
- WebRtc_Word16 noiseEstQDomain[PART_LEN1];
+ int noiseEstTooLowCtr[PART_LEN1];
+ int noiseEstTooHighCtr[PART_LEN1];
WebRtc_Word16 noiseEstCtr;
WebRtc_Word16 cngMode;
@@ -172,46 +183,19 @@
WebRtc_Word16 farEnergyMaxMin;
WebRtc_Word16 farEnergyVAD;
WebRtc_Word16 farEnergyMSE;
- WebRtc_Word16 currentVADValue;
+ int currentVADValue;
WebRtc_Word16 vadUpdateCount;
- WebRtc_Word16 delayHistogram[MAX_DELAY];
- WebRtc_Word16 delayVadCount;
- WebRtc_Word16 maxDelayHistIdx;
- WebRtc_Word16 lastMinPos;
-
WebRtc_Word16 startupState;
WebRtc_Word16 mseChannelCount;
- WebRtc_Word16 delayCount;
- WebRtc_Word16 newDelayCorrData;
- WebRtc_Word16 lastDelayUpdateCount;
- WebRtc_Word16 delayCorrelation[CORR_BUF_LEN];
WebRtc_Word16 supGain;
WebRtc_Word16 supGainOld;
- WebRtc_Word16 delayOffsetFlag;
WebRtc_Word16 supGainErrParamA;
WebRtc_Word16 supGainErrParamD;
WebRtc_Word16 supGainErrParamDiffAB;
WebRtc_Word16 supGainErrParamDiffBD;
- // TODO(bjornv): Will be removed after final version has been committed.
-#ifdef VAD_DATA
- FILE *vad_file;
- FILE *delay_file;
- FILE *far_file;
- FILE *far_cur_file;
- FILE *far_min_file;
- FILE *far_max_file;
- FILE *far_vad_file;
-#endif
-
- // TODO(bjornv): Will be removed after final version has been committed.
-#ifdef STORE_CHANNEL_DATA
- FILE *channel_file;
- FILE *channel_file_init;
-#endif
-
#ifdef AEC_DEBUG
FILE *farFile;
FILE *nearFile;
@@ -265,7 +249,20 @@
//
int WebRtcAecm_FreeCore(AecmCore_t *aecm);
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag, int delayOffsetFlag);
+int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag);
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_InitEchoPathCore(...)
+//
+// This function resets the echo channel adaptation with the specified channel.
+// Input:
+// - aecm : Pointer to the AECM instance
+// - echo_path : Pointer to the data that should initialize the echo path
+//
+// Output:
+// - aecm : Initialized instance
+//
+void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path);
///////////////////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_ProcessFrame(...)
@@ -282,10 +279,10 @@
// - out : Out buffer, one frame of nearend signal :
//
//
-void WebRtcAecm_ProcessFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const WebRtc_Word16 * const nearendNoisy,
- const WebRtc_Word16 * const nearendClean,
- WebRtc_Word16 * const out);
+int WebRtcAecm_ProcessFrame(AecmCore_t * aecm, const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * nearendClean,
+ WebRtc_Word16 * out);
///////////////////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_ProcessBlock(...)
@@ -303,10 +300,10 @@
// - out : Out buffer, one block of nearend signal :
//
//
-void WebRtcAecm_ProcessBlock(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const WebRtc_Word16 * const nearendNoisy,
- const WebRtc_Word16 * const noisyClean,
- WebRtc_Word16 * const out);
+int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, const WebRtc_Word16 * farend,
+ const WebRtc_Word16 * nearendNoisy,
+ const WebRtc_Word16 * noisyClean,
+ WebRtc_Word16 * out);
///////////////////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_BufferFarFrame()
@@ -335,4 +332,44 @@
void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
const int farLen, const int knownDelay);
+///////////////////////////////////////////////////////////////////////////////
+// Some function pointers, for internal functions shared by ARM NEON and
+// generic C code.
+//
+typedef void (*CalcLinearEnergies)(
+ AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echoEst,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored);
+extern CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+
+typedef void (*StoreAdaptiveChannel)(
+ AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est);
+extern StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+
+typedef void (*ResetAdaptiveChannel)(AecmCore_t* aecm);
+extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+
+typedef void (*WindowAndFFT)(
+ WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling);
+extern WindowAndFFT WebRtcAecm_WindowAndFFT;
+
+typedef void (*InverseFFTAndWindow)(
+ AecmCore_t* aecm,
+ WebRtc_Word16* fft, complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean);
+extern InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
+
+// Initialization of the above function pointers for ARM Neon.
+void WebRtcAecm_InitNeon(void);
+
+
#endif
diff --git a/src/modules/audio_processing/aecm/aecm_core_neon.c b/src/modules/audio_processing/aecm/aecm_core_neon.c
new file mode 100644
index 0000000..ab448b4
--- /dev/null
+++ b/src/modules/audio_processing/aecm/aecm_core_neon.c
@@ -0,0 +1,303 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "aecm_core.h"
+
+#include <arm_neon.h>
+#include <assert.h>
+
+
+// Square root of Hanning window in Q14.
+static const WebRtc_Word16 kSqrtHanningReversed[] __attribute__((aligned(8))) = {
+ 16384, 16373, 16354, 16325,
+ 16286, 16237, 16179, 16111,
+ 16034, 15947, 15851, 15746,
+ 15631, 15506, 15373, 15231,
+ 15079, 14918, 14749, 14571,
+ 14384, 14189, 13985, 13773,
+ 13553, 13325, 13089, 12845,
+ 12594, 12335, 12068, 11795,
+ 11514, 11227, 10933, 10633,
+ 10326, 10013, 9695, 9370,
+ 9040, 8705, 8364, 8019,
+ 7668, 7313, 6954, 6591,
+ 6224, 5853, 5478, 5101,
+ 4720, 4337, 3951, 3562,
+ 3172, 2780, 2386, 1990,
+ 1594, 1196, 798, 399
+};
+
+static void WindowAndFFTNeon(WebRtc_Word16* fft,
+ const WebRtc_Word16* time_signal,
+ complex16_t* freq_signal,
+ int time_signal_scaling) {
+ int i, j;
+
+ int16x4_t tmp16x4_scaling = vdup_n_s16(time_signal_scaling);
+ __asm__("vmov.i16 d21, #0" ::: "d21");
+
+ for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int32x4_t tmp32x4_0;
+
+ /* Window near end */
+ // fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((time_signal[i]
+ // << time_signal_scaling), WebRtcAecm_kSqrtHanning[i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i]));
+ tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
+
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
+ tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
+
+ __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
+ __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[j]) : "q10");
+
+ // fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ // (time_signal[PART_LEN + i] << time_signal_scaling),
+ // WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i + PART_LEN]));
+ tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
+
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
+ tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
+
+ __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
+ __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[PART_LEN2 + j]) : "q10");
+ }
+
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the first PART_LEN2 samples, and switch the sign of the imaginary part.
+ for (i = 0, j = 0; j < PART_LEN2; i += 8, j += 16) {
+ __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
+ __asm__("vneg.s16 d22, d22" : : : "q10");
+ __asm__("vneg.s16 d23, d23" : : : "q11");
+ __asm__("vst2.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&freq_signal[i].real): "q10", "q11");
+ }
+}
+
+static void InverseFFTAndWindowNeon(AecmCore_t* aecm,
+ WebRtc_Word16* fft,
+ complex16_t* efw,
+ WebRtc_Word16* output,
+ const WebRtc_Word16* nearendClean) {
+ int i, j, outCFFT;
+ WebRtc_Word32 tmp32no1;
+
+ // Synthesis
+ for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
+ // We overwrite two more elements in fft[], but it's ok.
+ __asm__("vld2.16 {d20, d21}, [%0, :128]" : : "r"(&(efw[i].real)) : "q10");
+ __asm__("vmov q11, q10" : : : "q10", "q11");
+
+ __asm__("vneg.s16 d23, d23" : : : "q11");
+ __asm__("vst2.16 {d22, d23}, [%0, :128]" : : "r"(&fft[j]): "q11");
+
+ __asm__("vrev64.16 q10, q10" : : : "q10");
+ __asm__("vst2.16 {d20, d21}, [%0]" : : "r"(&fft[PART_LEN4 - j - 6]): "q10");
+ }
+
+ fft[PART_LEN2] = efw[PART_LEN].real;
+ fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
+
+ // Inverse FFT, result should be scaled with outCFFT.
+ WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
+ outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
+
+ // Take only the real values and scale with outCFFT.
+ for (i = 0, j = 0; i < PART_LEN2; i += 8, j += 16) {
+ __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
+ __asm__("vst1.16 {d20, d21}, [%0, :128]" : : "r"(&fft[i]): "q10");
+ }
+
+ int32x4_t tmp32x4_2;
+ __asm__("vdup.32 %q0, %1" : "=w"(tmp32x4_2) : "r"((WebRtc_Word32)
+ (outCFFT - aecm->dfaCleanQDomain)));
+ for (i = 0; i < PART_LEN; i += 4) {
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int32x4_t tmp32x4_0;
+ int32x4_t tmp32x4_1;
+
+ // fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // fft[i], WebRtcAecm_kSqrtHanning[i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[i]));
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
+ __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
+ __asm__("vrshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
+
+ // tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
+ // outCFFT - aecm->dfaCleanQDomain);
+ __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
+
+ // fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ // tmp32no1 + outBuf[i], WEBRTC_SPL_WORD16_MIN);
+ // output[i] = fft[i];
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&aecm->outBuf[i]));
+ __asm__("vmovl.s16 %q0, %P1" : "=w"(tmp32x4_1) : "w"(tmp16x4_0));
+ __asm__("vadd.i32 %q0, %q1" : : "w"(tmp32x4_0), "w"(tmp32x4_1));
+ __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&fft[i]));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&output[i]));
+
+ // tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
+ // fft[PART_LEN + i], WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[PART_LEN + i]));
+ __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
+ __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
+ __asm__("vshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
+
+ // tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
+ __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
+ // outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
+ // WEBRTC_SPL_WORD16_MAX, tmp32no1, WEBRTC_SPL_WORD16_MIN);
+ __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
+ __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&aecm->outBuf[i]));
+ }
+
+ // Copy the current block to the old position (outBuf is shifted elsewhere).
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->xBuf[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&aecm->xBuf[i]): "q10");
+ }
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufNoisy[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufNoisy[i]): "q10");
+ }
+ if (nearendClean != NULL) {
+ for (i = 0; i < PART_LEN; i += 16) {
+ __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufClean[i + PART_LEN]) : "q10");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->dBufClean[i]): "q10");
+ }
+ }
+}
+
+static void CalcLinearEnergiesNeon(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est,
+ WebRtc_UWord32* far_energy,
+ WebRtc_UWord32* echo_energy_adapt,
+ WebRtc_UWord32* echo_energy_stored) {
+ int i;
+
+ register WebRtc_UWord32 far_energy_r;
+ register WebRtc_UWord32 echo_energy_stored_r;
+ register WebRtc_UWord32 echo_energy_adapt_r;
+ uint32x4_t tmp32x4_0;
+
+ __asm__("vmov.i32 q14, #0" : : : "q14"); // far_energy
+ __asm__("vmov.i32 q8, #0" : : : "q8"); // echo_energy_stored
+ __asm__("vmov.i32 q9, #0" : : : "q9"); // echo_energy_adapt
+
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // far_energy += (WebRtc_UWord32)(far_spectrum[i]);
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
+ __asm__("vaddw.u16 q14, q14, d26" : : : "q14", "q13");
+ __asm__("vaddw.u16 q14, q14, d27" : : : "q14", "q13");
+
+ // Get estimated echo energies for adaptive channel and stored channel.
+ // echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vst1.32 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&echo_est[i]):
+ "q10", "q11");
+
+ // echo_energy_stored += (WebRtc_UWord32)echoEst[i];
+ __asm__("vadd.u32 q8, q10" : : : "q10", "q8");
+ __asm__("vadd.u32 q8, q11" : : : "q11", "q8");
+
+ // echo_energy_adapt += WEBRTC_SPL_UMUL_16_16(
+ // aecm->channelAdapt16[i], far_spectrum[i]);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vadd.u32 q9, q10" : : : "q9", "q15");
+ __asm__("vadd.u32 q9, q11" : : : "q9", "q11");
+ }
+
+ __asm__("vadd.u32 d28, d29" : : : "q14");
+ __asm__("vpadd.u32 d28, d28" : : : "q14");
+ __asm__("vmov.32 %0, d28[0]" : "=r"(far_energy_r): : "q14");
+
+ __asm__("vadd.u32 d18, d19" : : : "q9");
+ __asm__("vpadd.u32 d18, d18" : : : "q9");
+ __asm__("vmov.32 %0, d18[0]" : "=r"(echo_energy_adapt_r): : "q9");
+
+ __asm__("vadd.u32 d16, d17" : : : "q8");
+ __asm__("vpadd.u32 d16, d16" : : : "q8");
+ __asm__("vmov.32 %0, d16[0]" : "=r"(echo_energy_stored_r): : "q8");
+
+ // Get estimated echo energies for adaptive channel and stored channel.
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ *echo_energy_stored = echo_energy_stored_r + (WebRtc_UWord32)echo_est[i];
+ *far_energy = far_energy_r + (WebRtc_UWord32)(far_spectrum[i]);
+ *echo_energy_adapt = echo_energy_adapt_r + WEBRTC_SPL_UMUL_16_16(
+ aecm->channelAdapt16[i], far_spectrum[i]);
+}
+
+static void StoreAdaptiveChannelNeon(AecmCore_t* aecm,
+ const WebRtc_UWord16* far_spectrum,
+ WebRtc_Word32* echo_est) {
+ int i;
+
+ // During startup we store the channel every block.
+ // Recalculate echo estimate.
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // aecm->channelStored[i] = acem->channelAdapt16[i];
+ // echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+ __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vst1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
+ __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&echo_est[i]) : "q10", "q11");
+ }
+ aecm->channelStored[i] = aecm->channelAdapt16[i];
+ echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+}
+
+static void ResetAdaptiveChannelNeon(AecmCore_t* aecm) {
+ int i;
+
+ for (i = 0; i < PART_LEN - 7; i += 8) {
+ // aecm->channelAdapt16[i] = aecm->channelStored[i];
+ // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)
+ // aecm->channelStored[i], 16);
+ __asm__("vld1.16 {d24, d25}, [%0, :128]" : :
+ "r"(&aecm->channelStored[i]) : "q12");
+ __asm__("vst1.16 {d24, d25}, [%0, :128]" : :
+ "r"(&aecm->channelAdapt16[i]) : "q12");
+ __asm__("vshll.s16 q10, d24, #16" : : : "q12", "q13", "q10");
+ __asm__("vshll.s16 q11, d25, #16" : : : "q12", "q13", "q11");
+ __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
+ "r"(&aecm->channelAdapt32[i]): "q10", "q11");
+ }
+ aecm->channelAdapt16[i] = aecm->channelStored[i];
+ aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+ (WebRtc_Word32)aecm->channelStored[i], 16);
+}
+
+void WebRtcAecm_InitNeon(void) {
+ WebRtcAecm_WindowAndFFT = WindowAndFFTNeon;
+ WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowNeon;
+ WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesNeon;
+ WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelNeon;
+ WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelNeon;
+}
diff --git a/src/modules/audio_processing/aecm/main/source/echo_control_mobile.c b/src/modules/audio_processing/aecm/echo_control_mobile.c
similarity index 83%
rename from src/modules/audio_processing/aecm/main/source/echo_control_mobile.c
rename to src/modules/audio_processing/aecm/echo_control_mobile.c
index f9d84f0..49798b7 100644
--- a/src/modules/audio_processing/aecm/main/source/echo_control_mobile.c
+++ b/src/modules/audio_processing/aecm/echo_control_mobile.c
@@ -31,7 +31,7 @@
// The factor of 2 handles wb, and the + 1 is as a safety margin
#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-static const int kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
+static const size_t kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
static const int kSampMsNb = 8; // samples per ms in nb
// Target suppression levels for nlp modes
// log{0.001, 0.00001, 0.00000001}
@@ -108,7 +108,8 @@
return -1;
}
- if (WebRtcApm_CreateBuffer(&aecm->farendBuf, kBufSizeSamp) == -1)
+ if (WebRtc_CreateBuffer(&aecm->farendBuf, kBufSizeSamp,
+ sizeof(int16_t)) == -1)
{
WebRtcAecm_Free(aecm);
aecm = NULL;
@@ -153,13 +154,13 @@
fclose(aecm->postCompFile);
#endif // AEC_DEBUG
WebRtcAecm_FreeCore(aecm->aecmCore);
- WebRtcApm_FreeBuffer(aecm->farendBuf);
+ WebRtc_FreeBuffer(aecm->farendBuf);
free(aecm);
return 0;
}
-WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
+WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq)
{
aecmob_t *aecm = aecmInst;
AecmConfig aecConfig;
@@ -176,13 +177,6 @@
}
aecm->sampFreq = sampFreq;
- if (scSampFreq < 1 || scSampFreq > 96000)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecm->scSampFreq = scSampFreq;
-
// Initialize AECM core
if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
{
@@ -191,7 +185,7 @@
}
// Initialize farend buffer
- if (WebRtcApm_InitBuffer(aecm->farendBuf) == -1)
+ if (WebRtc_InitBuffer(aecm->farendBuf) == -1)
{
aecm->lastError = AECM_UNSPECIFIED_ERROR;
return -1;
@@ -264,7 +258,7 @@
WebRtcAecm_DelayComp(aecm);
}
- WebRtcApm_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
+ WebRtc_WriteBuffer(aecm->farendBuf, farend, (size_t) nrOfSamples);
return retVal;
}
@@ -276,7 +270,6 @@
aecmob_t *aecm = aecmInst;
WebRtc_Word32 retVal = 0;
short i;
- short farend[FRAME_LEN];
short nmbrOfFilledBuffers;
short nBlocks10ms;
short nFrames;
@@ -352,7 +345,8 @@
memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
}
- nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
+ nmbrOfFilledBuffers =
+ (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
// The AECM is in the start up mode
// AECM is disabled until the soundcard buffer and farend buffers are OK
@@ -414,10 +408,9 @@
aecm->ECstartup = 0; // Enable the AECM
} else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
{
- WebRtcApm_FlushBuffer(
- aecm->farendBuf,
- WebRtcApm_get_buffer_size(aecm->farendBuf)
- - aecm->bufSizeStart * FRAME_LEN);
+ WebRtc_MoveReadPtr(aecm->farendBuf,
+ (int) WebRtc_available_read(aecm->farendBuf)
+ - (int) aecm->bufSizeStart * FRAME_LEN);
aecm->ECstartup = 0;
}
}
@@ -429,20 +422,27 @@
// Note only 1 block supported for nb and 2 blocks for wb
for (i = 0; i < nFrames; i++)
{
- nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
+ int16_t farend[FRAME_LEN];
+ const int16_t* farend_ptr = NULL;
+
+ nmbrOfFilledBuffers =
+ (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
// Check that there is data in the far end buffer
if (nmbrOfFilledBuffers > 0)
{
// Get the next 80 samples from the farend buffer
- WebRtcApm_ReadBuffer(aecm->farendBuf, farend, FRAME_LEN);
+ WebRtc_ReadBuffer(aecm->farendBuf, (void**) &farend_ptr, farend,
+ FRAME_LEN);
// Always store the last frame for use when we run out of data
- memcpy(&(aecm->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
+ memcpy(&(aecm->farendOld[i][0]), farend_ptr,
+ FRAME_LEN * sizeof(short));
} else
{
// We have no data so we use the last played frame
memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
+ farend_ptr = farend;
}
// Call buffer delay estimator when all data is extracted,
@@ -465,12 +465,24 @@
&out[FRAME_LEN * i], aecm->knownDelay);*/
if (nearendClean == NULL)
{
- WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearendNoisy[FRAME_LEN * i],
- NULL, &out[FRAME_LEN * i]);
+ if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
+ farend_ptr,
+ &nearendNoisy[FRAME_LEN * i],
+ NULL,
+ &out[FRAME_LEN * i]) == -1)
+ {
+ return -1;
+ }
} else
{
- WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearendNoisy[FRAME_LEN * i],
- &nearendClean[FRAME_LEN * i], &out[FRAME_LEN * i]);
+ if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
+ farend_ptr,
+ &nearendNoisy[FRAME_LEN * i],
+ &nearendClean[FRAME_LEN * i],
+ &out[FRAME_LEN * i]) == -1)
+ {
+ return -1;
+ }
}
#ifdef ARM_WINM_LOG
@@ -510,7 +522,8 @@
}
#ifdef AEC_DEBUG
- msInAECBuf = WebRtcApm_get_buffer_size(aecm->farendBuf) / (kSampMsNb*aecm->aecmCore->mult);
+ msInAECBuf = (short) WebRtc_available_read(aecm->farendBuf) /
+ (kSampMsNb * aecm->aecmCore->mult);
fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
#endif
@@ -627,6 +640,68 @@
return 0;
}
+WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
+ const void* echo_path,
+ size_t size_bytes)
+{
+ aecmob_t *aecm = aecmInst;
+ const WebRtc_Word16* echo_path_ptr = echo_path;
+
+ if ((aecm == NULL) || (echo_path == NULL))
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (size_bytes != WebRtcAecm_echo_path_size_bytes())
+ {
+ // Input channel size does not match the size of AECM
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
+
+ return 0;
+}
+
+WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
+ void* echo_path,
+ size_t size_bytes)
+{
+ aecmob_t *aecm = aecmInst;
+ WebRtc_Word16* echo_path_ptr = echo_path;
+
+ if ((aecm == NULL) || (echo_path == NULL))
+ {
+ aecm->lastError = AECM_NULL_POINTER_ERROR;
+ return -1;
+ }
+ if (size_bytes != WebRtcAecm_echo_path_size_bytes())
+ {
+ // Input channel size does not match the size of AECM
+ aecm->lastError = AECM_BAD_PARAMETER_ERROR;
+ return -1;
+ }
+ if (aecm->initFlag != kInitCheck)
+ {
+ aecm->lastError = AECM_UNINITIALIZED_ERROR;
+ return -1;
+ }
+
+ memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
+ return 0;
+}
+
+size_t WebRtcAecm_echo_path_size_bytes()
+{
+ return (PART_LEN1 * sizeof(WebRtc_Word16));
+}
+
WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
{
const char version[] = "AECM 1.2.0";
@@ -660,17 +735,17 @@
static int WebRtcAecm_EstBufDelay(aecmob_t *aecm, short msInSndCardBuf)
{
- short delayNew, nSampFar, nSampSndCard;
+ short delayNew, nSampSndCard;
+ short nSampFar = (short) WebRtc_available_read(aecm->farendBuf);
short diff;
- nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
delayNew = nSampSndCard - nSampFar;
if (delayNew < FRAME_LEN)
{
- WebRtcApm_FlushBuffer(aecm->farendBuf, FRAME_LEN);
+ WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
delayNew += FRAME_LEN;
}
@@ -710,10 +785,10 @@
static int WebRtcAecm_DelayComp(aecmob_t *aecm)
{
- int nSampFar, nSampSndCard, delayNew, nSampAdd;
+ int nSampFar = (int) WebRtc_available_read(aecm->farendBuf);
+ int nSampSndCard, delayNew, nSampAdd;
const int maxStuffSamp = 10 * FRAME_LEN;
- nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
delayNew = nSampSndCard - nSampFar;
@@ -725,7 +800,7 @@
FRAME_LEN));
nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
- WebRtcApm_StuffBuffer(aecm->farendBuf, nSampAdd);
+ WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
aecm->delayChange = 1; // the delay needs to be updated
}
diff --git a/src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h b/src/modules/audio_processing/aecm/interface/echo_control_mobile.h
similarity index 80%
rename from src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h
rename to src/modules/audio_processing/aecm/interface/echo_control_mobile.h
index 26b1172..30bea7a 100644
--- a/src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h
+++ b/src/modules/audio_processing/aecm/interface/echo_control_mobile.h
@@ -74,16 +74,14 @@
* -------------------------------------------------------------------
* void *aecmInst Pointer to the AECM instance
* WebRtc_Word32 sampFreq Sampling frequency of data
- * WebRtc_Word32 scSampFreq Soundcard sampling frequency
*
* Outputs Description
* -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
+ * WebRtc_Word32 return 0: OK
* -1: error
*/
WebRtc_Word32 WebRtcAecm_Init(void* aecmInst,
- WebRtc_Word32 sampFreq,
- WebRtc_Word32 scSampFreq);
+ WebRtc_Word32 sampFreq);
/*
* Inserts an 80 or 160 sample block of data into the farend buffer.
@@ -171,6 +169,52 @@
AecmConfig *config);
/*
+ * This function enables the user to set the echo path on-the-fly.
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* aecmInst Pointer to the AECM instance
+ * void* echo_path Pointer to the echo path to be set
+ * size_t size_bytes Size in bytes of the echo path
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
+ const void* echo_path,
+ size_t size_bytes);
+
+/*
+ * This function enables the user to get the currently used echo path
+ * on-the-fly
+ *
+ * Inputs Description
+ * -------------------------------------------------------------------
+ * void* aecmInst Pointer to the AECM instance
+ * void* echo_path Pointer to echo path
+ * size_t size_bytes Size in bytes of the echo path
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return 0: OK
+ * -1: error
+ */
+WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
+ void* echo_path,
+ size_t size_bytes);
+
+/*
+ * This function enables the user to get the echo path size in bytes
+ *
+ * Outputs Description
+ * -------------------------------------------------------------------
+ * size_t return : size in bytes
+ */
+size_t WebRtcAecm_echo_path_size_bytes();
+
+/*
* Gets the last error code.
*
* Inputs Description
diff --git a/src/modules/audio_processing/aecm/main/matlab/compsup.m b/src/modules/audio_processing/aecm/main/matlab/compsup.m
deleted file mode 100644
index 9575ec4..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/compsup.m
+++ /dev/null
@@ -1,447 +0,0 @@
-function [emicrophone,aaa]=compsup(microphone,TheFarEnd,avtime,samplingfreq);
-% microphone = microphone signal
-% aaa = nonlinearity input variable
-% TheFarEnd = far end signal
-% avtime = interval to compute suppression from (seconds)
-% samplingfreq = sampling frequency
-
-%if(nargin==6)
-% fprintf(1,'suppress has received a delay sequence\n');
-%end
-
-
-Ap500=[ 1.00, -4.95, 9.801, -9.70299, 4.80298005, -0.9509900499];
-Bp500=[ 0.662743088639636, -2.5841655608125, 3.77668102146288, -2.45182477425154, 0.596566274575251, 0.0];
-
-
-Ap200=[ 1.00, -4.875, 9.50625, -9.26859375, 4.518439453125, -0.881095693359375];
-Bp200=[ 0.862545460994275, -3.2832804496114, 4.67892032308828, -2.95798023879133, 0.699796870041299, 0.0];
-
-maxDelay=0.4; %[s]
-histLen=1; %[s]
-
-
-% CONSTANTS THAT YOU CAN EXPERIMENT WITH
-A_GAIN=10.0; % for the suppress case
-oversampling = 2; % must be power of 2; minimum is 2; 4 works
-% fine for support=64, but for support=128,
-% 8 gives better results.
-support=64; %512 % fft support (frequency resolution; at low
-% settings you can hear more distortion
-% (e.g. pitch that is left-over from far-end))
-% 128 works well, 64 is ok)
-
-lowlevel = mean(abs(microphone))*0.0001;
-
-G_ol = 0; % Use overlapping sets of estimates
-
-% ECHO SUPPRESSION SPECIFIC PARAMETERS
-suppress_overdrive=1.0; % overdrive factor for suppression 1.4 is good
-gamma_echo=1.0; % same as suppress_overdrive but at different place
-de_echo_bound=0.0;
-mLim=10; % rank of matrix G
-%limBW = 1; % use bandwidth-limited response for G
-if mLim > (support/2+1)
- error('mLim in suppress.m too large\n');
-end
-
-
-dynrange=1.0000e-004;
-
-% other, constants
-hsupport = support/2;
-hsupport1 = hsupport+1;
-factor = 2 / oversampling;
-updatel = support/oversampling;
-win=sqrt(designwindow(0,support));
-estLen = round(avtime * samplingfreq/updatel)
-
-runningfmean =0.0;
-
-mLim = floor(hsupport1/2);
-V = sqrt(2/hsupport1)*cos(pi/hsupport1*(repmat((0:hsupport1-1) + 0.5, mLim, 1).* ...
- repmat((0:mLim-1)' + 0.5, 1, hsupport1)));
-
-fprintf(1,'updatel is %5.3f s\n', updatel/samplingfreq);
-
-
-
-bandfirst=8; bandlast=25;
-dosmooth=0; % to get rid of wavy bin counts (can be worse or better)
-
-% compute some constants
-blockLen = support/oversampling;
-maxDelayb = floor(samplingfreq*maxDelay/updatel); % in blocks
-histLenb = floor(samplingfreq*histLen/updatel); % in blocks
-
-x0=TheFarEnd;
-y0=microphone;
-
-
-%input
-tlength=min([length(microphone),length(TheFarEnd)]);
-updateno=floor(tlength/updatel);
-tlength=updatel*updateno;
-updateno = updateno - oversampling + 1;
-
-TheFarEnd =TheFarEnd(1:tlength);
-microphone =microphone(1:tlength);
-
-TheFarEnd =[zeros(hsupport,1);TheFarEnd(1:tlength)];
-microphone =[zeros(hsupport,1);microphone(1:tlength)];
-
-
-% signal length
-n = min([floor(length(x0)/support)*support,floor(length(y0)/support)*support]);
-nb = n/blockLen - oversampling + 1; % in blocks
-
-% initialize space
-win = sqrt([0 ; hanning(support-1)]);
-sxAll2 = zeros(hsupport1,nb);
-syAll2 = zeros(hsupport1,nb);
-
-z500=zeros(5,maxDelayb+1);
-z200=zeros(5,hsupport1);
-
-bxspectrum=uint32(zeros(nb,1));
-bxhist=uint32(zeros(maxDelayb+1,1));
-byspectrum=uint32(zeros(nb,1));
-bcount=zeros(1+maxDelayb,nb);
-fcount=zeros(1+maxDelayb,nb);
-fout=zeros(1+maxDelayb,nb);
-delay=zeros(nb,1);
-tdelay=zeros(nb,1);
-nlgains=zeros(nb,1);
-
-% create space (mainly for debugging)
-emicrophone=zeros(tlength,1);
-femicrophone=complex(zeros(hsupport1,updateno));
-thefilter=zeros(hsupport1,updateno);
-thelimiter=ones(hsupport1,updateno);
-fTheFarEnd=complex(zeros(hsupport1,updateno));
-afTheFarEnd=zeros(hsupport1,updateno);
-fmicrophone=complex(zeros(hsupport1,updateno));
-afmicrophone=zeros(hsupport1,updateno);
-
-G = zeros(hsupport1, hsupport1);
-zerovec = zeros(hsupport1,1);
-zeromat = zeros(hsupport1);
-
-% Reset sums
-mmxs_a = zerovec;
-mmys_a = zerovec;
-s2xs_a = zerovec;
-s2ys_a = zerovec;
-Rxxs_a = zeromat;
-Ryxs_a = zeromat;
-count_a = 1;
-
-mmxs_b = zerovec;
-mmys_b = zerovec;
-s2xs_b = zerovec;
-s2ys_b = zerovec;
-Rxxs_b = zeromat;
-Ryxs_b = zeromat;
-count_b = 1;
-
-nog=0;
-
-aaa=zeros(size(TheFarEnd));
-
-% loop over signal blocks
-fprintf(1,'.. Suppression; averaging G over %5.1f seconds; file length %5.1f seconds ..\n',avtime, length(microphone)/samplingfreq);
-fprintf(1,'.. SUPPRESSING ONLY AFTER %5.1f SECONDS! ..\n',avtime);
-fprintf(1,'.. 20 seconds is good ..\n');
-hh = waitbar_j(0,'Please wait...');
-
-
-for i=1:updateno
-
- sb = (i-1)*updatel + 1;
- se=sb+support-1;
-
- % analysis FFTs
- temp=fft(win .* TheFarEnd(sb:se));
- fTheFarEnd(:,i)=temp(1:hsupport1);
- xf=fTheFarEnd(:,i);
- afTheFarEnd(:,i)= abs(fTheFarEnd(:,i));
-
- temp=win .* microphone(sb:se);
-
- temp=fft(win .* microphone(sb:se));
- fmicrophone(:,i)=temp(1:hsupport1);
- yf=fmicrophone(:,i);
- afmicrophone(:,i)= abs(fmicrophone(:,i));
-
-
- ener_orig = afmicrophone(:,i)'*afmicrophone(:,i);
- if( ener_orig == 0)
- afmicrophone(:,i)=lowlevel*ones(size(afmicrophone(:,i)));
- end
-
-
- % use log domain (showed improved performance)
-xxf= sqrt(real(xf.*conj(xf))+1e-20);
-yyf= sqrt(real(yf.*conj(yf))+1e-20);
- sxAll2(:,i) = 20*log10(xxf);
- syAll2(:,i) = 20*log10(yyf);
-
- mD=min(i-1,maxDelayb);
- xthreshold = sum(sxAll2(:,i-mD:i),2)/(maxDelayb+1);
-
- [yout, z200] = filter(Bp200,Ap200,syAll2(:,i),z200,2);
- yout=yout/(maxDelayb+1);
- ythreshold = mean(syAll2(:,i-mD:i),2);
-
-
- bxspectrum(i)=getBspectrum(sxAll2(:,i),xthreshold,bandfirst,bandlast);
- byspectrum(i)=getBspectrum(syAll2(:,i),yout,bandfirst,bandlast);
-
- bxhist(end-mD:end)=bxspectrum(i-mD:i);
-
- bcount(:,i)=hisser2( ...
- byspectrum(i),flipud(bxhist),bandfirst,bandlast);
-
-
- [fout(:,i), z500] = filter(Bp500,Ap500,bcount(:,i),z500,2);
- fcount(:,i)=sum(bcount(:,max(1,i-histLenb+1):i),2); % using the history range
- fout(:,i)=round(fout(:,i));
- [value,delay(i)]=min(fout(:,i),[],1);
- tdelay(i)=(delay(i)-1)*support/(samplingfreq*oversampling);
-
- % compensate
-
- idel = max(i - delay(i) + 1,1);
-
-
- % echo suppression
-
- noisyspec = afmicrophone(:,i);
-
- % Estimate G using covariance matrices
-
- % Cumulative estimates
- xx = afTheFarEnd(:,idel);
- yy = afmicrophone(:,i);
-
- % Means
- mmxs_a = mmxs_a + xx;
- mmys_a = mmys_a + yy;
- if (G_ol)
- mmxs_b = mmxs_b + xx;
- mmys_b = mmys_b + yy;
- mmy = mean([mmys_a/count_a mmys_b/count_b],2);
- mmx = mean([mmxs_a/count_a mmxs_b/count_b],2);
- else
- mmx = mmxs_a/count_a;
- mmy = mmys_a/count_a;
- end
- count_a = count_a + 1;
- count_b = count_b + 1;
-
- % Mean removal
- xxm = xx - mmx;
- yym = yy - mmy;
-
- % Variances
- s2xs_a = s2xs_a + xxm .* xxm;
- s2ys_a = s2ys_a + yym .* yym;
- s2xs_b = s2xs_b + xxm .* xxm;
- s2ys_b = s2ys_b + yym .* yym;
-
- % Correlation matrices
- Rxxs_a = Rxxs_a + xxm * xxm';
- Ryxs_a = Ryxs_a + yym * xxm';
- Rxxs_b = Rxxs_b + xxm * xxm';
- Ryxs_b = Ryxs_b + yym * xxm';
-
-
- % Gain matrix A
-
- if mod(i, estLen) == 0
-
-
- % Cumulative based estimates
- Rxxf = Rxxs_a / (estLen - 1);
- Ryxf = Ryxs_a / (estLen - 1);
-
- % Variance normalization
- s2x2 = s2xs_a / (estLen - 1);
- s2x2 = sqrt(s2x2);
- % Sx = diag(max(s2x2,dynrange*max(s2x2)));
- Sx = diag(s2x2);
- if (sum(s2x2) > 0)
- iSx = inv(Sx);
- else
- iSx= Sx + 0.01;
- end
-
- s2y2 = s2ys_a / (estLen - 1);
- s2y2 = sqrt(s2y2);
- % Sy = diag(max(s2y2,dynrange*max(s2y2)));
- Sy = diag(s2y2);
- iSy = inv(Sy);
- rx = iSx * Rxxf * iSx;
- ryx = iSy * Ryxf * iSx;
-
-
-
- dbd= 7; % Us less than the full matrix
-
- % k x m
- % Bandlimited structure on G
- LSEon = 0; % Default is using MMSE
- if (LSEon)
- ryx = ryx*rx;
- rx = rx*rx;
- end
- p = dbd-1;
- gaj = min(min(hsupport1,2*p+1),min([p+(1:hsupport1); hsupport1+p+1-(1:hsupport1)]));
- cgaj = [0 cumsum(gaj)];
-
- G3 = zeros(hsupport1);
- for kk=1:hsupport1
- ki = max(0,kk-p-1);
- if (sum(sum(rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk))))>0)
- G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk))/rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk));
- else
- G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk));
- end
- end
- % End Bandlimited structure
-
- G = G3;
- G(abs(G)<0.01)=0;
- G = suppress_overdrive * Sy * G * iSx;
-
- if 1
- figure(32); mi=2;
- surf(max(min(G,mi),-mi)); view(2)
- title('Unscaled Masked Limited-bandwidth G');
- end
- pause(0.05);
-
- % Reset sums
- mmxs_a = zerovec;
- mmys_a = zerovec;
- s2xs_a = zerovec;
- s2ys_a = zerovec;
- Rxxs_a = zeromat;
- Ryxs_a = zeromat;
- count_a = 1;
-
- end
-
- if (G_ol)
- % Gain matrix B
-
- if ((mod((i-estLen/2), estLen) == 0) & i>estLen)
-
-
- % Cumulative based estimates
- Rxxf = Rxxs_b / (estLen - 1);
- Ryxf = Ryxs_b / (estLen - 1);
-
- % Variance normalization
- s2x2 = s2xs_b / (estLen - 1);
- s2x2 = sqrt(s2x2);
- Sx = diag(max(s2x2,dynrange*max(s2x2)));
- iSx = inv(Sx);
- s2y2 = s2ys_b / (estLen - 1);
- s2y2 = sqrt(s2y2);
- Sy = diag(max(s2y2,dynrange*max(s2y2)));
- iSy = inv(Sy);
- rx = iSx * Rxxf * iSx;
- ryx = iSy * Ryxf * iSx;
-
-
- % Bandlimited structure on G
- LSEon = 0; % Default is using MMSE
- if (LSEon)
- ryx = ryx*rx;
- rx = rx*rx;
- end
- p = dbd-1;
- gaj = min(min(hsupport1,2*p+1),min([p+(1:hsupport1); hsupport1+p+1-(1:hsupport1)]));
- cgaj = [0 cumsum(gaj)];
-
- G3 = zeros(hsupport1);
- for kk=1:hsupport1
- ki = max(0,kk-p-1);
- G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk))/rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk));
- end
- % End Bandlimited structure
-
- G = G3;
- G(abs(G)<0.01)=0;
- G = suppress_overdrive * Sy * G * iSx;
-
- if 1
- figure(32); mi=2;
- surf(max(min(G,mi),-mi)); view(2)
- title('Unscaled Masked Limited-bandwidth G');
- end
- pause(0.05);
-
-
- % Reset sums
- mmxs_b = zerovec;
- mmys_b = zerovec;
- s2xs_b = zerovec;
- s2ys_b = zerovec;
- Rxxs_b = zeromat;
- Ryxs_b = zeromat;
- count_b = 1;
-
- end
-
- end
-
- FECestimate2 = G*afTheFarEnd(:,idel);
-
- % compute Wiener filter and suppressor function
- thefilter(:,i) = (noisyspec - gamma_echo*FECestimate2) ./ noisyspec;
- ix0 = find(thefilter(:,i)<de_echo_bound); % bounding trick 1
- thefilter(ix0,i) = de_echo_bound; % bounding trick 2
- ix0 = find(thefilter(:,i)>1); % bounding in reasonable range
- thefilter(ix0,i) = 1;
-
- % NONLINEARITY
- nl_alpha=0.8; % memory; seems not very critical
- nlSeverity=0.3; % nonlinearity severity: 0 does nothing; 1 suppresses all
- thefmean=mean(thefilter(8:16,i));
- if (thefmean<1)
- disp('');
- end
- runningfmean = nl_alpha*runningfmean + (1-nl_alpha)*thefmean;
- aaa(sb+20+1:sb+20+updatel)=10000*runningfmean* ones(updatel,1); % debug
- slope0=1.0/(1.0-nlSeverity); %
- thegain = max(0.0,min(1.0,slope0*(runningfmean-nlSeverity)));
- % END NONLINEARITY
- thefilter(:,i) = thegain*thefilter(:,i);
-
-
- % Wiener filtering
- femicrophone(:,i) = fmicrophone(:,i) .* thefilter(:,i);
- thelimiter(:,i) = (noisyspec - A_GAIN*FECestimate2) ./ noisyspec;
- index = find(thelimiter(:,i)>1.0);
- thelimiter(index,i) = 1.0;
- index = find(thelimiter(:,i)<0.0);
- thelimiter(index,i) = 0.0;
-
- if (rem(i,floor(updateno/20))==0)
- fprintf(1,'.');
- end
- if mod(i,50)==0
- waitbar_j(i/updateno,hh);
- end
-
-
- % reconstruction; first make spectrum odd
- temp=[femicrophone(:,i);flipud(conj(femicrophone(2:hsupport,i)))];
- emicrophone(sb:se) = emicrophone(sb:se) + factor * win .* real(ifft(temp));
-
-end
-fprintf(1,'\n');
-
-close(hh);
\ No newline at end of file
diff --git a/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m b/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m
deleted file mode 100644
index a4a533d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m
+++ /dev/null
@@ -1,22 +0,0 @@
-function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% compute binary spectrum using threshold spectrum as pivot
-% bspectrum = binary spectrum (binary)
-% ps=current power spectrum (float)
-% threshold=threshold spectrum (float)
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% initialization stuff
- if( length(ps)<bandlast | bandlast>32 | length(ps)~=length(threshold))
- error('BinDelayEst:spectrum:invalid','Dimensionality error');
-end
-
-% get current binary spectrum
-diff = ps - threshold;
-bspectrum=uint32(0);
-for(i=bandfirst:bandlast)
- if( diff(i)>0 )
- bspectrum = bitset(bspectrum,i);
- end
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/hisser2.m b/src/modules/audio_processing/aecm/main/matlab/hisser2.m
deleted file mode 100644
index 5a414f9..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/hisser2.m
+++ /dev/null
@@ -1,21 +0,0 @@
-function bcount=hisser2(bs,bsr,bandfirst,bandlast)
-% function bcount=hisser(bspectrum,bandfirst,bandlast)
-% histogram for the binary spectra
-% bcount= array of bit counts
-% bs=binary spectrum (one int32 number each)
-% bsr=reference binary spectra (one int32 number each)
-% blockSize = histogram over blocksize blocks
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% weight all delays equally
-maxDelay = length(bsr);
-
-% compute counts (two methods; the first works better and is operational)
-bcount=zeros(maxDelay,1);
-for(i=1:maxDelay)
- % the delay should have low count for low-near&high-far and high-near&low-far
- bcount(i)= sum(bitget(bitxor(bs,bsr(i)),bandfirst:bandlast));
- % the delay should have low count for low-near&high-far (works less well)
-% bcount(i)= sum(bitget(bitand(bsr(i),bitxor(bs,bsr(i))),bandfirst:bandlast));
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/main2.m b/src/modules/audio_processing/aecm/main/matlab/main2.m
deleted file mode 100644
index 7e24c69..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/main2.m
+++ /dev/null
@@ -1,19 +0,0 @@
-
-fid=fopen('aecfar.pcm'); far=fread(fid,'short'); fclose(fid);
-fid=fopen('aecnear.pcm'); mic=fread(fid,'short'); fclose(fid);
-
-%fid=fopen('QA1far.pcm'); far=fread(fid,'short'); fclose(fid);
-%fid=fopen('QA1near.pcm'); mic=fread(fid,'short'); fclose(fid);
-
-start=0 * 8000+1;
-stop= 30 * 8000;
-microphone=mic(start:stop);
-TheFarEnd=far(start:stop);
-avtime=1;
-
-% 16000 to make it compatible with the C-version
-[emicrophone,tdel]=compsup(microphone,TheFarEnd,avtime,16000);
-
-spclab(8000,TheFarEnd,microphone,emicrophone);
-
-
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m b/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m
deleted file mode 100644
index 2d3e686..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m
+++ /dev/null
@@ -1,269 +0,0 @@
-function [femicrophone, aecmStructNew, enerNear, enerFar] = AECMobile(fmicrophone, afTheFarEnd, setupStruct, aecmStruct)
-global NEARENDFFT;
-global F;
-
-aecmStructNew = aecmStruct;
-
-% Magnitude spectrum of near end signal
-afmicrophone = abs(fmicrophone);
-%afmicrophone = NEARENDFFT(setupStruct.currentBlock,:)'/2^F(setupStruct.currentBlock,end);
-% Near end energy level
-ener_orig = afmicrophone'*afmicrophone;
-if( ener_orig == 0)
- lowlevel = 0.01;
- afmicrophone = lowlevel*ones(size(afmicrophone));
-end
-%adiff = max(abs(afmicrophone - afTheFarEnd));
-%if (adiff > 0)
-% disp([setupStruct.currentBlock adiff])
-%end
-
-% Store the near end energy
-%aecmStructNew.enerNear(setupStruct.currentBlock) = log(afmicrophone'*afmicrophone);
-aecmStructNew.enerNear(setupStruct.currentBlock) = log(sum(afmicrophone));
-% Store the far end energy
-%aecmStructNew.enerFar(setupStruct.currentBlock) = log(afTheFarEnd'*afTheFarEnd);
-aecmStructNew.enerFar(setupStruct.currentBlock) = log(sum(afTheFarEnd));
-
-% Update subbands (We currently use all frequency bins, hence .useSubBand is turned off)
-if aecmStructNew.useSubBand
- internalIndex = 1;
- for kk=1:setupStruct.subBandLength+1
- ySubBand(kk) = mean(afmicrophone(internalIndex:internalIndex+setupStruct.numInBand(kk)-1).^aecmStructNew.bandFactor);
- xSubBand(kk) = mean(afTheFarEnd(internalIndex:internalIndex+setupStruct.numInBand(kk)-1).^aecmStructNew.bandFactor);
- internalIndex = internalIndex + setupStruct.numInBand(kk);
- end
-else
- ySubBand = afmicrophone.^aecmStructNew.bandFactor;
- xSubBand = afTheFarEnd.^aecmStructNew.bandFactor;
-end
-
-% Estimated echo energy
-if (aecmStructNew.bandFactor == 1)
- %aecmStructNew.enerEcho(setupStruct.currentBlock) = log((aecmStructNew.H.*xSubBand)'*(aecmStructNew.H.*xSubBand));
- %aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log((aecmStructNew.HStored.*xSubBand)'*(aecmStructNew.HStored.*xSubBand));
- aecmStructNew.enerEcho(setupStruct.currentBlock) = log(sum(aecmStructNew.H.*xSubBand));
- aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log(sum(aecmStructNew.HStored.*xSubBand));
-elseif (aecmStructNew.bandFactor == 2)
- aecmStructNew.enerEcho(setupStruct.currentBlock) = log(aecmStructNew.H'*xSubBand);
- aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log(aecmStructNew.HStored'*xSubBand);
-end
-
-% Last 100 blocks of data, used for plotting
-n100 = max(1,setupStruct.currentBlock-99):setupStruct.currentBlock;
-enerError = aecmStructNew.enerNear(n100)-aecmStructNew.enerEcho(n100);
-enerErrorStored = aecmStructNew.enerNear(n100)-aecmStructNew.enerEchoStored(n100);
-
-% Store the far end sub band. This is needed if we use LSE instead of NLMS
-aecmStructNew.X = [xSubBand aecmStructNew.X(:,1:end-1)];
-
-% Update energy levels, which control the VAD
-if ((aecmStructNew.enerFar(setupStruct.currentBlock) < aecmStructNew.energyMin) & (aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStruct.FAR_ENERGY_MIN))
- aecmStructNew.energyMin = aecmStructNew.enerFar(setupStruct.currentBlock);
- %aecmStructNew.energyMin = max(aecmStructNew.energyMin,12);
- aecmStructNew.energyMin = max(aecmStructNew.energyMin,aecmStruct.FAR_ENERGY_MIN);
- aecmStructNew.energyLevel = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThres+aecmStructNew.energyMin;
- aecmStructNew.energyLevelMSE = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThresMSE+aecmStructNew.energyMin;
-end
-if (aecmStructNew.enerFar(setupStruct.currentBlock) > aecmStructNew.energyMax)
- aecmStructNew.energyMax = aecmStructNew.enerFar(setupStruct.currentBlock);
- aecmStructNew.energyLevel = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThres+aecmStructNew.energyMin;
- aecmStructNew.energyLevelMSE = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThresMSE+aecmStructNew.energyMin;
-end
-
-% Calculate current energy error in near end (estimated echo vs. near end)
-dE = aecmStructNew.enerNear(setupStruct.currentBlock)-aecmStructNew.enerEcho(setupStruct.currentBlock);
-
-%%%%%%%%
-% Calculate step size used in LMS algorithm, based on current far end energy and near end energy error (dE)
-%%%%%%%%
-if setupStruct.stepSize_flag
- [mu, aecmStructNew] = calcStepSize(aecmStructNew.enerFar(setupStruct.currentBlock), dE, aecmStructNew, setupStruct.currentBlock, 1);
-else
- mu = 0.25;
-end
-aecmStructNew.muLog(setupStruct.currentBlock) = mu; % Store the step size
-
-% Estimate Echo Spectral Shape
-[U, aecmStructNew.H] = fallerEstimator(ySubBand,aecmStructNew.X,aecmStructNew.H,mu);
-
-%%%%%
-% Determine if we should store or restore the channel
-%%%%%
-if ((setupStruct.currentBlock <= aecmStructNew.convLength) | (~setupStruct.channelUpdate_flag))
- aecmStructNew.HStored = aecmStructNew.H; % Store what you have after startup
-elseif ((setupStruct.currentBlock > aecmStructNew.convLength) & (setupStruct.channelUpdate_flag))
- if ((aecmStructNew.enerFar(setupStruct.currentBlock) < aecmStructNew.energyLevelMSE) & (aecmStructNew.enerFar(setupStruct.currentBlock-1) >= aecmStructNew.energyLevelMSE))
- xxx = aecmStructNew.countMseH;
- if (xxx > 20)
- mseStored = mean(abs(aecmStructNew.enerEchoStored(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)-aecmStructNew.enerNear(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)));
- mseLatest = mean(abs(aecmStructNew.enerEcho(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)-aecmStructNew.enerNear(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)));
- %fprintf('Stored: %4f Latest: %4f\n', mseStored, mseLatest) % Uncomment if you want to display the MSE values
- if ((mseStored < 0.8*mseLatest) & (aecmStructNew.mseHStoredOld < 0.8*aecmStructNew.mseHLatestOld))
- aecmStructNew.H = aecmStructNew.HStored;
- fprintf('Restored H at block %d\n',setupStruct.currentBlock)
- elseif (((0.8*mseStored > mseLatest) & (mseLatest < aecmStructNew.mseHThreshold) & (aecmStructNew.mseHLatestOld < aecmStructNew.mseHThreshold)) | (mseStored == Inf))
- aecmStructNew.HStored = aecmStructNew.H;
- fprintf('Stored new H at block %d\n',setupStruct.currentBlock)
- end
- aecmStructNew.mseHStoredOld = mseStored;
- aecmStructNew.mseHLatestOld = mseLatest;
- end
- elseif ((aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStructNew.energyLevelMSE) & (aecmStructNew.enerFar(setupStruct.currentBlock-1) < aecmStructNew.energyLevelMSE))
- aecmStructNew.countMseH = 1;
- elseif (aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStructNew.energyLevelMSE)
- aecmStructNew.countMseH = aecmStructNew.countMseH + 1;
- end
-end
-
-%%%%%
-% Check delay (calculate the delay offset (if we can))
-% The algorithm is not tuned and should be used with care. It runs separately from Bastiaan's algorithm.
-%%%%%
-yyy = 31; % Correlation buffer length (currently unfortunately hard coded)
-dxxx = 25; % Maximum offset (currently unfortunately hard coded)
-if (setupStruct.currentBlock > aecmStructNew.convLength)
- if (aecmStructNew.enerFar(setupStruct.currentBlock-(yyy+2*dxxx-1):setupStruct.currentBlock) > aecmStructNew.energyLevelMSE)
- for xxx = -dxxx:dxxx
- aecmStructNew.delayLatestS(xxx+dxxx+1) = sum(sign(aecmStructNew.enerEcho(setupStruct.currentBlock-(yyy+dxxx-xxx)+1:setupStruct.currentBlock+xxx-dxxx)-mean(aecmStructNew.enerEcho(setupStruct.currentBlock-(yyy++dxxx-xxx)+1:setupStruct.currentBlock+xxx-dxxx))).*sign(aecmStructNew.enerNear(setupStruct.currentBlock-yyy-dxxx+1:setupStruct.currentBlock-dxxx)-mean(aecmStructNew.enerNear(setupStruct.currentBlock-yyy-dxxx+1:setupStruct.currentBlock-dxxx))));
- end
- aecmStructNew.newDelayCurve = 1;
- end
-end
-if ((setupStruct.currentBlock > 2*aecmStructNew.convLength) & ~rem(setupStruct.currentBlock,yyy*2) & aecmStructNew.newDelayCurve)
- [maxV,maxP] = max(aecmStructNew.delayLatestS);
- if ((maxP > 2) & (maxP < 2*dxxx))
- maxVLeft = aecmStructNew.delayLatestS(max(1,maxP-4));
- maxVRight = aecmStructNew.delayLatestS(min(2*dxxx+1,maxP+4));
- %fprintf('Max %d, Left %d, Right %d\n',maxV,maxVLeft,maxVRight) % Uncomment if you want to see max value
- if ((maxV > 24) & (maxVLeft < maxV - 10) & (maxVRight < maxV - 10))
- aecmStructNew.feedbackDelay = maxP-dxxx-1;
- aecmStructNew.newDelayCurve = 0;
- aecmStructNew.feedbackDelayUpdate = 1;
- fprintf('Feedback Update at block %d\n',setupStruct.currentBlock)
- end
- end
-end
-% End of "Check delay"
-%%%%%%%%
-
-%%%%%
-% Calculate suppression gain, based on far end energy and near end energy error (dE)
-if (setupStruct.supGain_flag)
- [gamma_echo, aecmStructNew.cntIn, aecmStructNew.cntOut] = calcFilterGain(aecmStructNew.enerFar(setupStruct.currentBlock), dE, aecmStructNew, setupStruct.currentBlock, aecmStructNew.convLength, aecmStructNew.cntIn, aecmStructNew.cntOut);
-else
- gamma_echo = 1;
-end
-aecmStructNew.gammaLog(setupStruct.currentBlock) = gamma_echo; % Store the gain
-gamma_use = gamma_echo;
-
-% Use the stored channel
-U = aecmStructNew.HStored.*xSubBand;
-
-% compute Wiener filter and suppressor function
-Iy = find(ySubBand);
-subBandFilter = zeros(size(ySubBand));
-if (aecmStructNew.bandFactor == 2)
- subBandFilter(Iy) = (1 - gamma_use*sqrt(U(Iy)./ySubBand(Iy))); % For Faller
-else
- subBandFilter(Iy) = (1 - gamma_use*(U(Iy)./ySubBand(Iy))); % For COV
-end
-ix0 = find(subBandFilter < 0); % bounding trick 1
-subBandFilter(ix0) = 0;
-ix0 = find(subBandFilter > 1); % bounding trick 1
-subBandFilter(ix0) = 1;
-
-% Interpolate back to normal frequency bins if we use sub bands
-if aecmStructNew.useSubBand
- thefilter = interp1(setupStruct.centerFreq,subBandFilter,linspace(0,setupStruct.samplingfreq/2,setupStruct.hsupport1)','nearest');
- testfilter = interp1(setupStruct.centerFreq,subBandFilter,linspace(0,setupStruct.samplingfreq/2,1000),'nearest');
- thefilter(end) = subBandFilter(end);
-
- internalIndex = 1;
- for kk=1:setupStruct.subBandLength+1
- internalIndex:internalIndex+setupStruct.numInBand(kk)-1;
- thefilter(internalIndex:internalIndex+setupStruct.numInBand(kk)-1) = subBandFilter(kk);
- internalIndex = internalIndex + setupStruct.numInBand(kk);
- end
-else
- thefilter = subBandFilter;
- testfilter = subBandFilter;
-end
-
-% Bound the filter
-ix0 = find(thefilter < setupStruct.de_echo_bound); % bounding trick 1
-thefilter(ix0) = setupStruct.de_echo_bound; % bounding trick 2
-ix0 = find(thefilter > 1); % bounding in reasonable range
-thefilter(ix0) = 1;
-
-%%%%
-% NLP
-%%%%
-thefmean = mean(thefilter(8:16));
-if (thefmean < 1)
- disp('');
-end
-aecmStructNew.runningfmean = setupStruct.nl_alpha*aecmStructNew.runningfmean + (1-setupStruct.nl_alpha)*thefmean;
-slope0 = 1.0/(1.0 - setupStruct.nlSeverity); %
-thegain = max(0.0, min(1.0, slope0*(aecmStructNew.runningfmean - setupStruct.nlSeverity)));
-if ~setupStruct.nlp_flag
- thegain = 1;
-end
-% END NONLINEARITY
-thefilter = thegain*thefilter;
-
-%%%%
-% The suppression
-%%%%
-femicrophone = fmicrophone .* thefilter;
-% Store the output energy (used for plotting)
-%aecmStructNew.enerOut(setupStruct.currentBlock) = log(abs(femicrophone)'*abs(femicrophone));
-aecmStructNew.enerOut(setupStruct.currentBlock) = log(sum(abs(femicrophone)));
-
-if aecmStructNew.plotIt
- figure(13)
- subplot(311)
- %plot(n100,enerFar(n100),'b-',n100,enerNear(n100),'k--',n100,enerEcho(n100),'r-',[n100(1) n100(end)],[1 1]*vadThNew,'b:',[n100(1) n100(end)],[1 1]*((energyMax-energyMin)/4+energyMin),'r-.',[n100(1) n100(end)],[1 1]*vadNearThNew,'g:',[n100(1) n100(end)],[1 1]*energyMax,'r-.',[n100(1) n100(end)],[1 1]*energyMin,'r-.','LineWidth',2)
- plot(n100,aecmStructNew.enerFar(n100),'b-',n100,aecmStructNew.enerNear(n100),'k--',n100,aecmStructNew.enerOut(n100),'r-.',n100,aecmStructNew.enerEcho(n100),'r-',n100,aecmStructNew.enerEchoStored(n100),'c-',[n100(1) n100(end)],[1 1]*((aecmStructNew.energyMax-aecmStructNew.energyMin)/4+aecmStructNew.energyMin),'g-.',[n100(1) n100(end)],[1 1]*aecmStructNew.energyMax,'g-.',[n100(1) n100(end)],[1 1]*aecmStructNew.energyMin,'g-.','LineWidth',2)
- %title(['Frame ',int2str(i),' av ',int2str(setupStruct.updateno),' State = ',int2str(speechState),' \mu = ',num2str(mu)])
- title(['\gamma = ',num2str(gamma_echo),' \mu = ',num2str(mu)])
- subplot(312)
- %plot(n100,enerError,'b-',[n100(1) n100(end)],[1 1]*vadNearTh,'r:',[n100(1) n100(end)],[-1.5 -1.5]*vadNearTh,'r:','LineWidth',2)
- %plot(n100,enerError,'b-',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
- plot(n100,enerError,'b-',n100,enerErrorStored,'c-',[n100(1) n100(end)],[1 1]*aecmStructNew.varMean,'k--',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
- % Plot mu
- %plot(n100,log2(aecmStructNew.muLog(n100)),'b-','LineWidth',2)
- %plot(n100,log2(aecmStructNew.HGain(n100)),'b-',[n100(1) n100(end)],[1 1]*log2(sum(aecmStructNew.HStored)),'r:','LineWidth',2)
- title(['Block ',int2str(setupStruct.currentBlock),' av ',int2str(setupStruct.updateno)])
- subplot(313)
- %plot(n100,enerVar(n100),'b-',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
- %plot(n100,enerVar(n100),'b-','LineWidth',2)
- % Plot correlation curve
-
- %plot(-25:25,aecmStructNew.delayStored/max(aecmStructNew.delayStored),'c-',-25:25,aecmStructNew.delayLatest/max(aecmStructNew.delayLatest),'r-',-25:25,(max(aecmStructNew.delayStoredS)-aecmStructNew.delayStoredS)/(max(aecmStructNew.delayStoredS)-min(aecmStructNew.delayStoredS)),'c:',-25:25,(max(aecmStructNew.delayLatestS)-aecmStructNew.delayLatestS)/(max(aecmStructNew.delayLatestS)-min(aecmStructNew.delayLatestS)),'r:','LineWidth',2)
- %plot(-25:25,aecmStructNew.delayStored,'c-',-25:25,aecmStructNew.delayLatest,'r-',-25:25,(max(aecmStructNew.delayStoredS)-aecmStructNew.delayStoredS)/(max(aecmStructNew.delayStoredS)-min(aecmStructNew.delayStoredS)),'c:',-25:25,(max(aecmStructNew.delayLatestS)-aecmStructNew.delayLatestS)/(max(aecmStructNew.delayLatestS)-min(aecmStructNew.delayLatestS)),'r:','LineWidth',2)
- %plot(-25:25,aecmStructNew.delayLatest,'r-',-25:25,(50-aecmStructNew.delayLatestS)/100,'r:','LineWidth',2)
- plot(-25:25,aecmStructNew.delayLatestS,'r:','LineWidth',2)
- %plot(-25:25,aecmStructNew.delayStored,'c-',-25:25,aecmStructNew.delayLatest,'r-','LineWidth',2)
- plot(0:32,aecmStruct.HStored,'bo-','LineWidth',2)
- %title(['\gamma | In = ',int2str(aecmStructNew.muStruct.countInInterval),' | Out High = ',int2str(aecmStructNew.muStruct.countOutHighInterval),' | Out Low = ',int2str(aecmStructNew.muStruct.countOutLowInterval)])
- pause(1)
- %if ((setupStruct.currentBlock == 860) | (setupStruct.currentBlock == 420) | (setupStruct.currentBlock == 960))
- if 0%(setupStruct.currentBlock == 960)
- figure(60)
- plot(n100,aecmStructNew.enerNear(n100),'k--',n100,aecmStructNew.enerEcho(n100),'k:','LineWidth',2)
- legend('Near End','Estimated Echo')
- title('Signal Energy witH offset compensation')
- figure(61)
- subplot(211)
- stem(sign(aecmStructNew.enerNear(n100)-mean(aecmStructNew.enerNear(n100))))
- title('Near End Energy Pattern (around mean value)')
- subplot(212)
- stem(sign(aecmStructNew.enerEcho(n100)-mean(aecmStructNew.enerEcho(n100))))
- title('Estimated Echo Energy Pattern (around mean value)')
- pause
- end
- drawnow%,pause
-elseif ~rem(setupStruct.currentBlock,100)
- fprintf('Block %d of %d\n',setupStruct.currentBlock,setupStruct.updateno)
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/align.m b/src/modules/audio_processing/aecm/main/matlab/matlab/align.m
deleted file mode 100644
index 9b9c0ba..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/align.m
+++ /dev/null
@@ -1,98 +0,0 @@
-function [delayStructNew] = align(xf, yf, delayStruct, i, trueDelay);
-
-%%%%%%%
-% Bastiaan's algorithm copied
-%%%%%%%
-Ap500 = [1.00, -4.95, 9.801, -9.70299, 4.80298005, -0.9509900499];
-Bp500 = [0.662743088639636, -2.5841655608125, 3.77668102146288, -2.45182477425154, 0.596566274575251, 0.0];
-Ap200 = [1.00, -4.875, 9.50625, -9.26859375, 4.518439453125, -0.881095693359375];
-Bp200 = [0.862545460994275, -3.2832804496114, 4.67892032308828, -2.95798023879133, 0.699796870041299, 0.0];
-
-oldMethod = 1; % Turn on or off the old method. The new one is Bastiaan's August 2008 updates
-THReSHoLD = 2.0; % ADJUSTABLE threshold factor; 4.0 seems good
-%%%%%%%%%%%%%%%%%%%
-% use log domain (showed improved performance)
-xxf = sqrt(real(xf.*conj(xf))+1e-20);
-yyf = sqrt(real(yf.*conj(yf))+1e-20);
-delayStruct.sxAll2(:,i) = 20*log10(xxf);
-delayStruct.syAll2(:,i) = 20*log10(yyf);
-
-mD = min(i-1,delayStruct.maxDelayb);
-if oldMethod
- factor = 1.0;
- histLenb = 250;
- xthreshold = factor*median(delayStruct.sxAll2(:,i-mD:i),2);
- ythreshold = factor*median(delayStruct.syAll2(:,i-mD:i),2);
-else
- xthreshold = sum(delayStruct.sxAll2(:,i-mD:i),2)/(delayStruct.maxDelayb+1);
-
- [yout, delayStruct.z200] = filter(Bp200, Ap200, delayStruct.syAll2(:,i), delayStruct.z200, 2);
- yout = yout/(delayStruct.maxDelayb+1);
- ythreshold = mean(delayStruct.syAll2(:,i-mD:i),2);
- ythreshold = yout;
-end
-
-delayStruct.bxspectrum(i) = getBspectrum(delayStruct.sxAll2(:,i), xthreshold, delayStruct.bandfirst, delayStruct.bandlast);
-delayStruct.byspectrum(i) = getBspectrum(delayStruct.syAll2(:,i), ythreshold, delayStruct.bandfirst, delayStruct.bandlast);
-
-delayStruct.bxhist(end-mD:end) = delayStruct.bxspectrum(i-mD:i);
-
-delayStruct.bcount(:,i) = hisser2(delayStruct.byspectrum(i), flipud(delayStruct.bxhist), delayStruct.bandfirst, delayStruct.bandlast);
-[delayStruct.fout(:,i), delayStruct.z500] = filter(Bp500, Ap500, delayStruct.bcount(:,i), delayStruct.z500, 2);
-if oldMethod
- %delayStruct.new(:,i) = sum(delayStruct.bcount(:,max(1,i-histLenb+1):i),2); % using the history range
- tmpVec = [delayStruct.fout(1,i)*ones(2,1); delayStruct.fout(:,i); delayStruct.fout(end,i)*ones(2,1)]; % using the history range
- tmpVec = filter(ones(1,5), 1, tmpVec);
- delayStruct.new(:,i) = tmpVec(5:end);
- %delayStruct.new(:,i) = delayStruct.fout(:,i); % using the history range
-else
- [delayStruct.fout(:,i), delayStruct.z500] = filter(Bp500, Ap500, delayStruct.bcount(:,i), delayStruct.z500, 2);
- % NEW CODE
- delayStruct.new(:,i) = filter([-1,-2,1,4,1,-2,-1], 1, delayStruct.fout(:,i)); %remv smth component
- delayStruct.new(1:end-3,i) = delayStruct.new(1+3:end,i);
- delayStruct.new(1:6,i) = 0.0;
- delayStruct.new(end-6:end,i) = 0.0; % ends are no good
-end
-[valuen, tempdelay] = min(delayStruct.new(:,i)); % find minimum
-if oldMethod
- threshold = valuen + (max(delayStruct.new(:,i)) - valuen)/4;
- thIndex = find(delayStruct.new(:,i) <= threshold);
- if (i > 1)
- delayDiff = abs(delayStruct.delay(i-1)-tempdelay+1);
- if (delayStruct.oneGoodEstimate & (max(diff(thIndex)) > 1) & (delayDiff < 10))
- % We consider this minimum to be significant, hence update the delay
- delayStruct.delay(i) = tempdelay;
- elseif (~delayStruct.oneGoodEstimate & (max(diff(thIndex)) > 1))
- delayStruct.delay(i) = tempdelay;
- if (i > histLenb)
- delayStruct.oneGoodEstimate = 1;
- end
- else
- delayStruct.delay(i) = delayStruct.delay(i-1);
- end
- else
- delayStruct.delay(i) = tempdelay;
- end
-else
- threshold = THReSHoLD*std(delayStruct.new(:,i)); % set updata threshold
- if ((-valuen > threshold) | (i < delayStruct.smlength)) % see if you want to update delay
- delayStruct.delay(i) = tempdelay;
- else
- delayStruct.delay(i) = delayStruct.delay(i-1);
- end
- % END NEW CODE
-end
-delayStructNew = delayStruct;
-
-% administrative and plotting stuff
-if( 0)
- figure(10);
- plot([1:length(delayStructNew.new(:,i))],delayStructNew.new(:,i),trueDelay*[1 1],[min(delayStructNew.new(:,i)),max(delayStructNew.new(:,i))],'r',[1 length(delayStructNew.new(:,i))],threshold*[1 1],'r:', 'LineWidth',2);
- %plot([1:length(delayStructNew.bcount(:,i))],delayStructNew.bcount(:,i),trueDelay*[1 1],[min(delayStructNew.bcount(:,i)),max(delayStructNew.bcount(:,i))],'r','LineWidth',2);
- %plot([thedelay,thedelay],[min(fcount(:,i)),max(fcount(:,i))],'r');
- %title(sprintf('bin count and known delay at time %5.1f s\n',(i-1)*(support/(fs*oversampling))));
- title(delayStructNew.oneGoodEstimate)
- xlabel('delay in frames');
- %hold off;
- drawnow
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m b/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m
deleted file mode 100644
index a09a7f2..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m
+++ /dev/null
@@ -1,88 +0,0 @@
-function [gam, cntIn2, cntOut2] = calcFilterGain(energy, dE, aecmStruct, t, T, cntIn, cntOut)
-
-defaultLevel = 1.2;
-cntIn2 = cntIn;
-cntOut2 = cntOut;
-if (t < T)
- gam = 1;
-else
- dE1 = -5;
- dE2 = 1;
- gamMid = 0.2;
- gam = max(0,min((energy - aecmStruct.energyMin)/(aecmStruct.energyLevel - aecmStruct.energyMin), 1-(1-gamMid)*(aecmStruct.energyMax-energy)/(aecmStruct.energyMax-aecmStruct.energyLevel)));
-
- dEOffset = -0.5;
- dEWidth = 1.5;
- %gam2 = max(1,2-((dE-dEOffset)/(dE2-dEOffset)).^2);
- gam2 = 1+(abs(dE-dEOffset)<(dE2-dEOffset));
-
- gam = gam*gam2;
-
-
- if (energy < aecmStruct.energyLevel)
- gam = 0;
- else
- gam = defaultLevel;
- end
- dEVec = aecmStruct.enerNear(t-63:t)-aecmStruct.enerEcho(t-63:t);
- %dEVec = aecmStruct.enerNear(t-20:t)-aecmStruct.enerEcho(t-20:t);
- numCross = 0;
- currentState = 0;
- for ii=1:64
- if (currentState == 0)
- currentState = (dEVec(ii) > dE2) - (dEVec(ii) < -2);
- elseif ((currentState == 1) & (dEVec(ii) < -2))
- numCross = numCross + 1;
- currentState = -1;
- elseif ((currentState == -1) & (dEVec(ii) > dE2))
- numCross = numCross + 1;
- currentState = 1;
- end
- end
- gam = max(0, gam - numCross/25);
- gam = 1;
-
- ener_A = 1;
- ener_B = 0.8;
- ener_C = aecmStruct.energyLevel + (aecmStruct.energyMax-aecmStruct.energyLevel)/5;
- dE_A = 4;%2;
- dE_B = 3.6;%1.8;
- dE_C = 0.9*dEWidth;
- dE_D = 1;
- timeFactorLength = 10;
- ddE = abs(dE-dEOffset);
- if (energy < aecmStruct.energyLevel)
- gam = 0;
- else
- gam = 1;
- gam2 = max(0, min(ener_B*(energy-aecmStruct.energyLevel)/(ener_C-aecmStruct.energyLevel), ener_B+(ener_A-ener_B)*(energy-ener_C)/(aecmStruct.energyMax-ener_C)));
- if (ddE < dEWidth)
- % Update counters
- cntIn2 = cntIn2 + 1;
- if (cntIn2 > 2)
- cntOut2 = 0;
- end
- gam3 = max(dE_D, min(dE_A-(dE_A-dE_B)*(ddE/dE_C), dE_D+(dE_B-dE_D)*(dEWidth-ddE)/(dEWidth-dE_C)));
- gam3 = dE_A;
- else
- % Update counters
- cntOut2 = cntOut2 + 1;
- if (cntOut2 > 2)
- cntIn2 = 0;
- end
- %gam2 = 1;
- gam3 = dE_D;
- end
- timeFactor = min(1, cntIn2/timeFactorLength);
- gam = gam*(1-timeFactor) + timeFactor*gam2*gam3;
- end
- %gam = gam/floor(numCross/2+1);
-end
-if isempty(gam)
- numCross
- timeFactor
- cntIn2
- cntOut2
- gam2
- gam3
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m b/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m
deleted file mode 100644
index ae1365f..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m
+++ /dev/null
@@ -1,105 +0,0 @@
-function [mu, aecmStructNew] = calcStepSize(energy, dE, aecmStruct, t, logscale)
-
-if (nargin < 4)
- t = 1;
- logscale = 1;
-elseif (nargin == 4)
- logscale = 1;
-end
-T = aecmStruct.convLength;
-
-if logscale
- currentMuMax = aecmStruct.MU_MIN + (aecmStruct.MU_MAX-aecmStruct.MU_MIN)*min(t,T)/T;
- if (aecmStruct.energyMin >= aecmStruct.energyMax)
- mu = aecmStruct.MU_MIN;
- else
- mu = (energy - aecmStruct.energyMin)/(aecmStruct.energyMax - aecmStruct.energyMin)*(currentMuMax-aecmStruct.MU_MIN) + aecmStruct.MU_MIN;
- end
- mu = 2^mu;
- if (energy < aecmStruct.energyLevel)
- mu = 0;
- end
-else
- muMin = 0;
- muMax = 0.5;
- currentMuMax = muMin + (muMax-muMin)*min(t,T)/T;
- if (aecmStruct.energyMin >= aecmStruct.energyMax)
- mu = muMin;
- else
- mu = (energy - aecmStruct.energyMin)/(aecmStruct.energyMax - aecmStruct.energyMin)*(currentMuMax-muMin) + muMin;
- end
-end
-dE2 = 1;
-dEOffset = -0.5;
-offBoost = 5;
-if (mu > 0)
- if (abs(dE-aecmStruct.ENERGY_DEV_OFFSET) > aecmStruct.ENERGY_DEV_TOL)
- aecmStruct.muStruct.countInInterval = 0;
- else
- aecmStruct.muStruct.countInInterval = aecmStruct.muStruct.countInInterval + 1;
- end
- if (dE < aecmStruct.ENERGY_DEV_OFFSET - aecmStruct.ENERGY_DEV_TOL)
- aecmStruct.muStruct.countOutLowInterval = aecmStruct.muStruct.countOutLowInterval + 1;
- else
- aecmStruct.muStruct.countOutLowInterval = 0;
- end
- if (dE > aecmStruct.ENERGY_DEV_OFFSET + aecmStruct.ENERGY_DEV_TOL)
- aecmStruct.muStruct.countOutHighInterval = aecmStruct.muStruct.countOutHighInterval + 1;
- else
- aecmStruct.muStruct.countOutHighInterval = 0;
- end
-end
-muVar = 2^min(-3,5/50*aecmStruct.muStruct.countInInterval-3);
-muOff = 2^max(offBoost,min(0,offBoost*(aecmStruct.muStruct.countOutLowInterval-aecmStruct.muStruct.minOutLowInterval)/(aecmStruct.muStruct.maxOutLowInterval-aecmStruct.muStruct.minOutLowInterval)));
-
-muLow = 1/64;
-muVar = 1;
-if (t < 2*T)
- muDT = 1;
- muVar = 1;
- mdEVec = 0;
- numCross = 0;
-else
- muDT = min(1,max(muLow,1-(1-muLow)*(dE-aecmStruct.ENERGY_DEV_OFFSET)/aecmStruct.ENERGY_DEV_TOL));
- dEVec = aecmStruct.enerNear(t-63:t)-aecmStruct.enerEcho(t-63:t);
- %dEVec = aecmStruct.enerNear(t-20:t)-aecmStruct.enerEcho(t-20:t);
- numCross = 0;
- currentState = 0;
- for ii=1:64
- if (currentState == 0)
- currentState = (dEVec(ii) > dE2) - (dEVec(ii) < -2);
- elseif ((currentState == 1) & (dEVec(ii) < -2))
- numCross = numCross + 1;
- currentState = -1;
- elseif ((currentState == -1) & (dEVec(ii) > dE2))
- numCross = numCross + 1;
- currentState = 1;
- end
- end
-
- %logicDEVec = (dEVec > dE2) - (dEVec < -2);
- %numCross = sum(abs(diff(logicDEVec)));
- %mdEVec = mean(abs(dEVec-dEOffset));
- %mdEVec = mean(abs(dEVec-mean(dEVec)));
- %mdEVec = max(dEVec)-min(dEVec);
- %if (mdEVec > 4)%1.5)
- % muVar = 0;
- %end
- muVar = 2^(-floor(numCross/2));
- muVar = 2^(-numCross);
-end
-%muVar = 1;
-
-
-% if (eStd > (dE2-dEOffset))
-% muVar = 1/8;
-% else
-% muVar = 1;
-% end
-
-%mu = mu*muDT*muVar*muOff;
-mu = mu*muDT*muVar;
-mu = min(mu,0.25);
-aecmStructNew = aecmStruct;
-%aecmStructNew.varMean = mdEVec;
-aecmStructNew.varMean = numCross;
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m b/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m
deleted file mode 100644
index d038b51..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m
+++ /dev/null
@@ -1,42 +0,0 @@
-function [U, Hnew] = fallerEstimator(Y, X, H, mu)
-
-% Near end signal is stacked frame by frame columnwise in matrix Y and far end in X
-%
-% Possible estimation procedures are
-% 1) LSE
-% 2) NLMS
-% 3) Separated numerator and denomerator filters
-regParam = 1;
-[numFreqs, numFrames] = size(Y);
-[numFreqs, Q] = size(X);
-U = zeros(numFreqs, 1);
-
-if ((nargin == 3) | (nargin == 5))
- dtd = 0;
-end
-if (nargin == 4)
- dtd = H;
-end
-Emax = 7;
-dEH = Emax-sum(sum(H));
-nu = 2*mu;
-% if (nargin < 5)
-% H = zeros(numFreqs, Q);
-% for kk = 1:numFreqs
-% Xmatrix = hankel(X(kk,1:Q),X(kk,Q:end));
-% y = Y(kk,1:end-Q+1)';
-% H(kk,:) = (y'*Xmatrix')*inv(Xmatrix*Xmatrix'+regParam);
-% U(kk,1) = H(kk,:)*Xmatrix(:,1);
-% end
-% else
- for kk = 1:numFreqs
- x = X(kk,1:Q)';
- y = Y(kk,1);
- Htmp = mu*(y-H(kk,:)*x)/(x'*x+regParam)*x;
- %Htmp = (mu*(y-H(kk,:)*x)/(x'*x+regParam) - nu/dEH)*x;
- H(kk,:) = H(kk,:) + Htmp';
- U(kk,1) = H(kk,:)*x;
- end
-% end
-
-Hnew = H;
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m b/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m
deleted file mode 100644
index a4a533d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m
+++ /dev/null
@@ -1,22 +0,0 @@
-function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% compute binary spectrum using threshold spectrum as pivot
-% bspectrum = binary spectrum (binary)
-% ps=current power spectrum (float)
-% threshold=threshold spectrum (float)
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% initialization stuff
- if( length(ps)<bandlast | bandlast>32 | length(ps)~=length(threshold))
- error('BinDelayEst:spectrum:invalid','Dimensionality error');
-end
-
-% get current binary spectrum
-diff = ps - threshold;
-bspectrum=uint32(0);
-for(i=bandfirst:bandlast)
- if( diff(i)>0 )
- bspectrum = bitset(bspectrum,i);
- end
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m b/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m
deleted file mode 100644
index 5a414f9..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m
+++ /dev/null
@@ -1,21 +0,0 @@
-function bcount=hisser2(bs,bsr,bandfirst,bandlast)
-% function bcount=hisser(bspectrum,bandfirst,bandlast)
-% histogram for the binary spectra
-% bcount= array of bit counts
-% bs=binary spectrum (one int32 number each)
-% bsr=reference binary spectra (one int32 number each)
-% blockSize = histogram over blocksize blocks
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% weight all delays equally
-maxDelay = length(bsr);
-
-% compute counts (two methods; the first works better and is operational)
-bcount=zeros(maxDelay,1);
-for(i=1:maxDelay)
- % the delay should have low count for low-near&high-far and high-near&low-far
- bcount(i)= sum(bitget(bitxor(bs,bsr(i)),bandfirst:bandlast));
- % the delay should have low count for low-near&high-far (works less well)
-% bcount(i)= sum(bitget(bitand(bsr(i),bitxor(bs,bsr(i))),bandfirst:bandlast));
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m b/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m
deleted file mode 100644
index eeb2aaa..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m
+++ /dev/null
@@ -1,283 +0,0 @@
-useHTC = 1; % Set this if you want to run a single file and set file names below. Otherwise use simEnvironment to run from several scenarios in a row
-delayCompensation_flag = 0; % Set this flag to one if you want to turn on the delay compensation/enhancement
-global FARENDFFT;
-global NEARENDFFT;
-global F;
-
-if useHTC
-% fid=fopen('./htcTouchHd/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid);
-% fid=fopen('./htcTouchHd/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid);
-% fid=fopen('./samsungBlackjack/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid);
-% fid=fopen('./samsungBlackjack/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid);
-% fid=fopen('aecFarPoor.pcm'); xFar=fread(fid,'short'); fclose(fid);
-% fid=fopen('aecNearPoor.pcm'); yNear=fread(fid,'short'); fclose(fid);
-% fid=fopen('out_aes.pcm'); outAES=fread(fid,'short'); fclose(fid);
- fid=fopen('aecFar4.pcm'); xFar=fread(fid,'short'); fclose(fid);
- fid=fopen('aecNear4.pcm'); yNear=fread(fid,'short'); fclose(fid);
- yNearSpeech = zeros(size(xFar));
- fs = 8000;
- frameSize = 64;
-% frameSize = 128;
- fs = 16000;
-% frameSize = 256;
-%F = load('fftValues.txt');
-%FARENDFFT = F(:,1:33);
-%NEARENDFFT = F(:,34:66);
-
-else
- loadFileFar = [speakerType, '_s_',scenario,'_far_b.wav'];
- [xFar,fs,nbits] = wavread(loadFileFar);
- xFar = xFar*2^(nbits-1);
- loadFileNear = [speakerType, '_s_',scenario,'_near_b.wav'];
- [yNear,fs,nbits] = wavread(loadFileNear);
- yNear = yNear*2^(nbits-1);
- loadFileNearSpeech = [speakerType, '_s_',scenario,'_nearSpeech_b.wav'];
- [yNearSpeech,fs,nbits] = wavread(loadFileNearSpeech);
- yNearSpeech = yNearSpeech*2^(nbits-1);
- frameSize = 256;
-end
-
-dtRegions = [];
-
-% General settings for the AECM
-setupStruct = struct(...
- 'stepSize_flag', 1,... % This flag turns on the step size calculation. If turned off, mu = 0.25.
- 'supGain_flag', 0,... % This flag turns on the suppression gain calculation. If turned off, gam = 1.
- 'channelUpdate_flag', 0,... % This flag turns on the channel update. If turned off, H is updated for convLength and then kept constant.
- 'nlp_flag', 0,... % Turn on/off NLP
- 'withVAD_flag', 0,... % Turn on/off NLP
- 'useSubBand', 0,... % Set to 1 if to use subBands
- 'useDelayEstimation', 1,... % Set to 1 if to use delay estimation
- 'support', frameSize,... % # of samples per frame
- 'samplingfreq',fs,... % Sampling frequency
- 'oversampling', 2,... % Overlap between blocks/frames
- 'updatel', 0,... % # of samples between blocks
- 'hsupport1', 0,... % # of bins in frequency domain
- 'factor', 0,... % synthesis window amplification
- 'tlength', 0,... % # of samples of entire file
- 'updateno', 0,... % # of updates
- 'nb', 1,... % # of blocks
- 'currentBlock', 0,... %
- 'win', zeros(frameSize,1),...% Window to apply for fft and synthesis
- 'avtime', 1,... % Time (in sec.) to perform averaging
- 'estLen', 0,... % Averaging in # of blocks
- 'A_GAIN', 10.0,... %
- 'suppress_overdrive', 1.0,... % overdrive factor for suppression 1.4 is good
- 'gamma_echo', 1.0,... % same as suppress_overdrive but at different place
- 'de_echo_bound', 0.0,... %
- 'nl_alpha', 0.4,... % memory; seems not very critical
- 'nlSeverity', 0.2,... % nonlinearity severity: 0 does nothing; 1 suppresses all
- 'numInBand', [],... % # of frequency bins in resp. subBand
- 'centerFreq', [],... % Center frequency of resp. subBand
- 'dtRegions', dtRegions,... % Regions where we have DT
- 'subBandLength', frameSize/2);%All bins
- %'subBandLength', 11); %Something's wrong when subBandLength even
- %'nl_alpha', 0.8,... % memory; seems not very critical
-
-delayStruct = struct(...
- 'bandfirst', 8,...
- 'bandlast', 25,...
- 'smlength', 600,...
- 'maxDelay', 0.4,...
- 'oneGoodEstimate', 0,...
- 'delayAdjust', 0,...
- 'maxDelayb', 0);
-% More parameters in delayStruct are constructed in "updateSettings" below
-
-% Make struct settings
-[setupStruct, delayStruct] = updateSettings(yNear, xFar, setupStruct, delayStruct);
-setupStruct.numInBand = ones(setupStruct.hsupport1,1);
-
-Q = 1; % Time diversity in channel
-% General settings for the step size calculation
-muStruct = struct(...
- 'countInInterval', 0,...
- 'countOutHighInterval', 0,...
- 'countOutLowInterval', 0,...
- 'minInInterval', 50,...
- 'minOutHighInterval', 10,...
- 'minOutLowInterval', 10,...
- 'maxOutLowInterval', 50);
-% General settings for the AECM
-aecmStruct = struct(...
- 'plotIt', 0,... % Set to 0 to turn off plotting
- 'useSubBand', 0,...
- 'bandFactor', 1,...
- 'H', zeros(setupStruct.subBandLength+1,Q),...
- 'HStored', zeros(setupStruct.subBandLength+1,Q),...
- 'X', zeros(setupStruct.subBandLength+1,Q),...
- 'energyThres', 0.28,...
- 'energyThresMSE', 0.4,...
- 'energyMin', inf,...
- 'energyMax', -inf,...
- 'energyLevel', 0,...
- 'energyLevelMSE', 0,...
- 'convLength', 100,...
- 'gammaLog', ones(setupStruct.updateno,1),...
- 'muLog', ones(setupStruct.updateno,1),...
- 'enerFar', zeros(setupStruct.updateno,1),...
- 'enerNear', zeros(setupStruct.updateno,1),...
- 'enerEcho', zeros(setupStruct.updateno,1),...
- 'enerEchoStored', zeros(setupStruct.updateno,1),...
- 'enerOut', zeros(setupStruct.updateno,1),...
- 'runningfmean', 0,...
- 'muStruct', muStruct,...
- 'varMean', 0,...
- 'countMseH', 0,...
- 'mseHThreshold', 1.1,...
- 'mseHStoredOld', inf,...
- 'mseHLatestOld', inf,...
- 'delayLatestS', zeros(1,51),...
- 'feedbackDelay', 0,...
- 'feedbackDelayUpdate', 0,...
- 'cntIn', 0,...
- 'cntOut', 0,...
- 'FAR_ENERGY_MIN', 1,...
- 'ENERGY_DEV_OFFSET', 0.5,...
- 'ENERGY_DEV_TOL', 1.5,...
- 'MU_MIN', -16,...
- 'MU_MAX', -2,...
- 'newDelayCurve', 0);
-
-% Adjust speech signals
-xFar = [zeros(setupStruct.hsupport1-1,1);xFar(1:setupStruct.tlength)];
-yNear = [zeros(setupStruct.hsupport1-1,1);yNear(1:setupStruct.tlength)];
-yNearSpeech = [zeros(setupStruct.hsupport1-1,1);yNearSpeech(1:setupStruct.tlength)];
-xFar = xFar(1:setupStruct.tlength);
-yNear = yNear(1:setupStruct.tlength);
-
-% Set figure settings
-if aecmStruct.plotIt
- figure(13)
- set(gcf,'doublebuffer','on')
-end
-%%%%%%%%%%
-% Here starts the algorithm
-% Dividing into frames and then estimating the near end speech
-%%%%%%%%%%
-fTheFarEnd = complex(zeros(setupStruct.hsupport1,1));
-afTheFarEnd = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fFar = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fmicrophone = complex(zeros(setupStruct.hsupport1,1));
-afmicrophone = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fNear = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-femicrophone = complex(zeros(setupStruct.hsupport1,1));
-emicrophone = zeros(setupStruct.tlength,1);
-
-if (setupStruct.useDelayEstimation == 2)
- delSamples = [1641 1895 2032 1895 2311 2000 2350 2222 NaN 2332 2330 2290 2401 2415 NaN 2393 2305 2381 2398];
- delBlocks = round(delSamples/setupStruct.updatel);
- delStarts = floor([25138 46844 105991 169901 195739 218536 241803 333905 347703 362660 373753 745135 765887 788078 806257 823835 842443 860139 881869]/setupStruct.updatel);
-else
- delStarts = [];
-end
-
-for i=1:setupStruct.updateno
- setupStruct.currentBlock = i;
-
- sb = (i-1)*setupStruct.updatel + 1;
- se = sb + setupStruct.support - 1;
-
- %%%%%%%
- % Analysis FFTs
- %%%%%%%
- % Far end signal
- temp = fft(setupStruct.win .* xFar(sb:se))/frameSize;
- fTheFarEnd = temp(1:setupStruct.hsupport1);
- afTheFarEnd(:,i) = abs(fTheFarEnd);
- fFar(:,i) = fTheFarEnd;
- % Near end signal
- temp = fft(setupStruct.win .* yNear(sb:se))/frameSize;%,pause
- fmicrophone = temp(1:setupStruct.hsupport1);
- afmicrophone(:,i) = abs(fmicrophone);
- fNear(:,i) = fmicrophone;
- %abs(fmicrophone),pause
- % The true near end speaker (if we have such info)
- temp = fft(setupStruct.win .* yNearSpeech(sb:se));
- aftrueSpeech = abs(temp(1:setupStruct.hsupport1));
-
- if(i == 1000)
- %break;
- end
-
- % Perform delay estimation
- if (setupStruct.useDelayEstimation == 1)
- % Delay Estimation
- delayStruct = align(fTheFarEnd, fmicrophone, delayStruct, i);
- %delayStruct.delay(i) = 39;%19;
- idel = max(i - delayStruct.delay(i) + 1,1);
-
- if delayCompensation_flag
- % If we have a new delay estimate from Bastiaan's alg. update the offset
- if (delayStruct.delay(i) ~= delayStruct.delay(max(1,i-1)))
- delayStruct.delayAdjust = delayStruct.delayAdjust + delayStruct.delay(i) - delayStruct.delay(i-1);
- end
- % Store the compensated delay
- delayStruct.delayNew(i) = delayStruct.delay(i) - delayStruct.delayAdjust;
- if (delayStruct.delayNew(i) < 1)
- % Something's wrong
- pause,break
- end
- % Compensate with the offset estimate
- idel = idel + delayStruct.delayAdjust;
- end
- if 0%aecmStruct.plotIt
- figure(1)
- plot(1:i,delayStruct.delay(1:i),'k:',1:i,delayStruct.delayNew(1:i),'k--','LineWidth',2),drawnow
- end
- elseif (setupStruct.useDelayEstimation == 2)
- % Use "manual delay"
- delIndex = find(delStarts<i);
- if isempty(delIndex)
- idel = i;
- else
- idel = i - delBlocks(max(delIndex));
- if isnan(idel)
- idel = i - delBlocks(max(delIndex)-1);
- end
- end
- else
- % No delay estimation
- %idel = max(i - 18, 1);
- idel = max(i - 50, 1);
- end
-
- %%%%%%%%
- % This is the AECM algorithm
- %
- % Output is the new frequency domain signal (hopefully) echo compensated
- %%%%%%%%
- [femicrophone, aecmStruct] = AECMobile(fmicrophone, afTheFarEnd(:,idel), setupStruct, aecmStruct);
- %[femicrophone, aecmStruct] = AECMobile(fmicrophone, FARENDFFT(idel,:)'/2^F(idel,end-1), setupStruct, aecmStruct);
-
- if aecmStruct.feedbackDelayUpdate
- % If the feedback tells us there is a new offset out there update the enhancement
- delayStruct.delayAdjust = delayStruct.delayAdjust + aecmStruct.feedbackDelay;
- aecmStruct.feedbackDelayUpdate = 0;
- end
-
- % reconstruction; first make spectrum odd
- temp = [femicrophone; flipud(conj(femicrophone(2:(setupStruct.hsupport1-1))))];
- emicrophone(sb:se) = emicrophone(sb:se) + setupStruct.factor * setupStruct.win .* real(ifft(temp))*frameSize;
- if max(isnan(emicrophone(sb:se)))
- % Something's wrong with the output at block i
- i
- break
- end
-end
-
-
-if useHTC
- fid=fopen('aecOutMatlabC.pcm','w');fwrite(fid,int16(emicrophone),'short');fclose(fid);
- %fid=fopen('farendFFT.txt','w');fwrite(fid,int16(afTheFarEnd(:)),'short');fclose(fid);
- %fid=fopen('farendFFTreal.txt','w');fwrite(fid,int16(imag(fFar(:))),'short');fclose(fid);
- %fid=fopen('farendFFTimag.txt','w');fwrite(fid,int16(real(fFar(:))),'short');fclose(fid);
- %fid=fopen('nearendFFT.txt','w');fwrite(fid,int16(afmicrophone(:)),'short');fclose(fid);
- %fid=fopen('nearendFFTreal.txt','w');fwrite(fid,int16(real(fNear(:))),'short');fclose(fid);
- %fid=fopen('nearendFFTimag.txt','w');fwrite(fid,int16(imag(fNear(:))),'short');fclose(fid);
-end
-if useHTC
- %spclab(setupStruct.samplingfreq,xFar,yNear,emicrophone)
-else
- spclab(setupStruct.samplingfreq,xFar,yNear,emicrophone,yNearSpeech)
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m b/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m
deleted file mode 100644
index 3ebe701..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m
+++ /dev/null
@@ -1,15 +0,0 @@
-speakerType = 'fm';
-%for k=2:5
-%for k=[2 4 5]
-for k=3
- scenario = int2str(k);
- fprintf('Current scenario: %d\n',k)
- mainProgram
- %saveFile = [speakerType, '_s_',scenario,'_delayEst_v2_vad_man.wav'];
- %wavwrite(emic,fs,nbits,saveFile);
- %saveFile = ['P:\Engineering_share\BjornV\AECM\',speakerType, '_s_',scenario,'_delayEst_v2_vad_man.pcm'];
- %saveFile = [speakerType, '_s_',scenario,'_adaptMu_adaptGamma_withVar_gammFilt_HSt.pcm'];
- saveFile = ['scenario_',scenario,'_090417_backupH_nlp.pcm'];
- fid=fopen(saveFile,'w');fwrite(fid,int16(emicrophone),'short');fclose(fid);
- %pause
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m b/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m
deleted file mode 100644
index c805f1d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m
+++ /dev/null
@@ -1,94 +0,0 @@
-function [setupStructNew, delayStructNew] = updateSettings(microphone, TheFarEnd, setupStruct, delayStruct);
-
-% other, constants
-setupStruct.hsupport1 = setupStruct.support/2 + 1;
-setupStruct.factor = 2 / setupStruct.oversampling;
-setupStruct.updatel = setupStruct.support/setupStruct.oversampling;
-setupStruct.estLen = round(setupStruct.avtime * setupStruct.samplingfreq/setupStruct.updatel);
-
-% compute some constants
-blockLen = setupStruct.support/setupStruct.oversampling;
-delayStruct.maxDelayb = floor(setupStruct.samplingfreq*delayStruct.maxDelay/setupStruct.updatel); % in blocks
-
-%input
-tlength = min([length(microphone),length(TheFarEnd)]);
-updateno = floor(tlength/setupStruct.updatel);
-setupStruct.tlength = setupStruct.updatel*updateno;
-setupStruct.updateno = updateno - setupStruct.oversampling + 1;
-
-% signal length
-n = floor(min([length(TheFarEnd), length(microphone)])/setupStruct.support)*setupStruct.support;
-setupStruct.nb = n/blockLen - setupStruct.oversampling + 1; % in blocks
-
-setupStruct.win = sqrt([0 ; hanning(setupStruct.support-1)]);
-
-% Construct filterbank in Bark-scale
-
-K = setupStruct.subBandLength; %Something's wrong when K even
-erbs = 21.4*log10(0.00437*setupStruct.samplingfreq/2+1);
-fe = (10.^((0:K)'*erbs/K/21.4)-1)/0.00437;
-setupStruct.centerFreq = fe;
-H = diag(ones(1,K-1))+diag(ones(1,K-2),-1);
-Hinv = inv(H);
-aty = 2*Hinv(end,:)*fe(2:end-1);
-boundary = aty - (setupStruct.samplingfreq/2 + fe(end-1))/2;
-if rem(K,2)
- x1 = min([fe(2)/2, -boundary]);
-else
- x1 = max([0, boundary]);
-end
-%x1
-g = fe(2:end-1);
-g(1) = g(1) - x1/2;
-x = 2*Hinv*g;
-x = [x1;x];
-%figure(42), clf
-xy = zeros((K+1)*4,1);
-yy = zeros((K+1)*4,1);
-xy(1:4) = [fe(1) fe(1) x(1) x(1)]';
-yy(1:4) = [0 1 1 0]'/x(1);
-for kk=2:K
- xy((kk-1)*4+(1:4)) = [x(kk-1) x(kk-1) x(kk) x(kk)]';
- yy((kk-1)*4+(1:4)) = [0 1 1 0]'/(x(kk)-x(kk-1));
-end
-xy(end-3:end) = [x(K) x(K) fe(end) fe(end)]';
-yy(end-3:end) = [0 1 1 0]'/(fe(end)*2-2*x(K));
-%plot(xy,yy,'LineWidth',2)
-%fill(xy,yy,'y')
-
-x = [0;x];
-xk = x*setupStruct.hsupport1/setupStruct.samplingfreq*2;
-%setupStruct.erbBoundaries = xk;
-numInBand = zeros(length(xk),1);
-xh = (0:setupStruct.hsupport1-1);
-
-for kk=1:length(xk)
- if (kk==length(xk))
- numInBand(kk) = length(find(xh>=xk(kk)));
- else
- numInBand(kk) = length(intersect(find(xh>=xk(kk)),find(xh<xk(kk+1))));
- end
-end
-setupStruct.numInBand = numInBand;
-
-setupStructNew = setupStruct;
-
-delayStructNew = struct(...
- 'sxAll2',zeros(setupStructNew.hsupport1,setupStructNew.nb),...
- 'syAll2',zeros(setupStructNew.hsupport1,setupStructNew.nb),...
- 'z200',zeros(5,setupStructNew.hsupport1),...
- 'z500',zeros(5,delayStruct.maxDelayb+1),...
- 'bxspectrum',uint32(zeros(setupStructNew.nb,1)),...
- 'byspectrum',uint32(zeros(setupStructNew.nb,1)),...
- 'bandfirst',delayStruct.bandfirst,'bandlast',delayStruct.bandlast,...
- 'bxhist',uint32(zeros(delayStruct.maxDelayb+1,1)),...
- 'bcount',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
- 'fout',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
- 'new',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
- 'smlength',delayStruct.smlength,...
- 'maxDelay', delayStruct.maxDelay,...
- 'maxDelayb', delayStruct.maxDelayb,...
- 'oneGoodEstimate', 0,...
- 'delayAdjust', 0,...
- 'delayNew',zeros(setupStructNew.nb,1),...
- 'delay',zeros(setupStructNew.nb,1));
diff --git a/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m b/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m
deleted file mode 100644
index 50b9ccf..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m
+++ /dev/null
@@ -1,234 +0,0 @@
-function fout = waitbar_j(x,whichbar, varargin)
-%WAITBAR Display wait bar.
-% H = WAITBAR(X,'title', property, value, property, value, ...)
-% creates and displays a waitbar of fractional length X. The
-% handle to the waitbar figure is returned in H.
-% X should be between 0 and 1. Optional arguments property and
-% value allow to set corresponding waitbar figure properties.
-% Property can also be an action keyword 'CreateCancelBtn', in
-% which case a cancel button will be added to the figure, and
-% the passed value string will be executed upon clicking on the
-% cancel button or the close figure button.
-%
-% WAITBAR(X) will set the length of the bar in the most recently
-% created waitbar window to the fractional length X.
-%
-% WAITBAR(X,H) will set the length of the bar in waitbar H
-% to the fractional length X.
-%
-% WAITBAR(X,H,'updated title') will update the title text in
-% the waitbar figure, in addition to setting the fractional
-% length to X.
-%
-% WAITBAR is typically used inside a FOR loop that performs a
-% lengthy computation. A sample usage is shown below:
-%
-% h = waitbar(0,'Please wait...');
-% for i=1:100,
-% % computation here %
-% waitbar(i/100,h)
-% end
-% close(h)
-
-% Clay M. Thompson 11-9-92
-% Vlad Kolesnikov 06-7-99
-% Copyright 1984-2001 The MathWorks, Inc.
-% $Revision: 1.22 $ $Date: 2001/04/15 12:03:29 $
-
-if nargin>=2
- if ischar(whichbar)
- type=2; %we are initializing
- name=whichbar;
- elseif isnumeric(whichbar)
- type=1; %we are updating, given a handle
- f=whichbar;
- else
- error(['Input arguments of type ' class(whichbar) ' not valid.'])
- end
-elseif nargin==1
- f = findobj(allchild(0),'flat','Tag','TMWWaitbar');
-
- if isempty(f)
- type=2;
- name='Waitbar';
- else
- type=1;
- f=f(1);
- end
-else
- error('Input arguments not valid.');
-end
-
-x = max(0,min(100*x,100));
-
-switch type
- case 1, % waitbar(x) update
- p = findobj(f,'Type','patch');
- l = findobj(f,'Type','line');
- if isempty(f) | isempty(p) | isempty(l),
- error('Couldn''t find waitbar handles.');
- end
- xpatch = get(p,'XData');
- xpatch = [0 x x 0];
- set(p,'XData',xpatch)
- xline = get(l,'XData');
- set(l,'XData',xline);
-
- if nargin>2,
- % Update waitbar title:
- hAxes = findobj(f,'type','axes');
- hTitle = get(hAxes,'title');
- set(hTitle,'string',varargin{1});
- end
-
- case 2, % waitbar(x,name) initialize
- vertMargin = 0;
- if nargin > 2,
- % we have optional arguments: property-value pairs
- if rem (nargin, 2 ) ~= 0
- error( 'Optional initialization arguments must be passed in pairs' );
- end
- end
-
- oldRootUnits = get(0,'Units');
-
- set(0, 'Units', 'points');
- screenSize = get(0,'ScreenSize');
-
- axFontSize=get(0,'FactoryAxesFontSize');
-
- pointsPerPixel = 72/get(0,'ScreenPixelsPerInch');
-
- width = 360 * pointsPerPixel;
- height = 75 * pointsPerPixel;
- pos = [screenSize(3)/2-width/2 screenSize(4)/2-height/2 width height];
-
-%pos= [501.75 589.5 393.75 52.5];
- f = figure(...
- 'Units', 'points', ...
- 'BusyAction', 'queue', ...
- 'Position', pos, ...
- 'Resize','on', ...
- 'CreateFcn','', ...
- 'NumberTitle','off', ...
- 'IntegerHandle','off', ...
- 'MenuBar', 'none', ...
- 'Tag','TMWWaitbar',...
- 'Interruptible', 'off', ...
- 'Visible','on');
-
- %%%%%%%%%%%%%%%%%%%%%
- % set figure properties as passed to the fcn
- % pay special attention to the 'cancel' request
- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
- if nargin > 2,
- propList = varargin(1:2:end);
- valueList = varargin(2:2:end);
- cancelBtnCreated = 0;
- for ii = 1:length( propList )
- try
- if strcmp(lower(propList{ii}), 'createcancelbtn' ) & ~cancelBtnCreated
- cancelBtnHeight = 23 * pointsPerPixel;
- cancelBtnWidth = 60 * pointsPerPixel;
- newPos = pos;
- vertMargin = vertMargin + cancelBtnHeight;
- newPos(4) = newPos(4)+vertMargin;
- callbackFcn = [valueList{ii}];
- set( f, 'Position', newPos, 'CloseRequestFcn', callbackFcn );
- cancelButt = uicontrol('Parent',f, ...
- 'Units','points', ...
- 'Callback',callbackFcn, ...
- 'ButtonDownFcn', callbackFcn, ...
- 'Enable','on', ...
- 'Interruptible','off', ...
- 'Position', [pos(3)-cancelBtnWidth*1.4, 7, ...
- cancelBtnWidth, cancelBtnHeight], ...
- 'String','Cancel', ...
- 'Tag','TMWWaitbarCancelButton');
- cancelBtnCreated = 1;
- else
- % simply set the prop/value pair of the figure
- set( f, propList{ii}, valueList{ii});
- end
- catch
- disp ( ['Warning: could not set property ''' propList{ii} ''' with value ''' num2str(valueList{ii}) '''' ] );
- end
- end
- end
-
- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-
-
- colormap([]);
-
- axNorm=[.05 .3 .9 .2];
- % axNorm=[1 1 1 1];
- axPos=axNorm.*[pos(3:4),pos(3:4)] + [0 vertMargin 0 0];
-
- h = axes('XLim',[0 100],...
- 'YLim',[0 1],...
- 'Box','on', ...
- 'Units','Points',...
- 'FontSize', axFontSize,...
- 'Position',axPos,...
- 'XTickMode','manual',...
- 'YTickMode','manual',...
- 'XTick',[],...
- 'YTick',[],...
- 'XTickLabelMode','manual',...
- 'XTickLabel',[],...
- 'YTickLabelMode','manual',...
- 'YTickLabel',[]);
-
- tHandle=title(name);
- tHandle=get(h,'title');
- oldTitleUnits=get(tHandle,'Units');
- set(tHandle,...
- 'Units', 'points',...
- 'String', name);
-
- tExtent=get(tHandle,'Extent');
- set(tHandle,'Units',oldTitleUnits);
-
- titleHeight=tExtent(4)+axPos(2)+axPos(4)+5;
- if titleHeight>pos(4)
- pos(4)=titleHeight;
- pos(2)=screenSize(4)/2-pos(4)/2;
- figPosDirty=logical(1);
- else
- figPosDirty=logical(0);
- end
-
- if tExtent(3)>pos(3)*1.10;
- pos(3)=min(tExtent(3)*1.10,screenSize(3));
- pos(1)=screenSize(3)/2-pos(3)/2;
-
- axPos([1,3])=axNorm([1,3])*pos(3);
- set(h,'Position',axPos);
-
- figPosDirty=logical(1);
- end
-
- if figPosDirty
- set(f,'Position',pos);
- end
-
- xpatch = [0 x x 0];
- ypatch = [0 0 1 1];
- xline = [100 0 0 100 100];
- yline = [0 0 1 1 0];
-
- p = patch(xpatch,ypatch,'r','EdgeColor','r','EraseMode','none');
- l = line(xline,yline,'EraseMode','none');
- set(l,'Color',get(gca,'XColor'));
-
-
- set(f,'HandleVisibility','callback','visible','on', 'resize','off');
-
- set(0, 'Units', oldRootUnits);
-end % case
-drawnow;
-
-if nargout==1,
- fout = f;
-end
diff --git a/src/modules/audio_processing/aecm/main/source/Android.mk b/src/modules/audio_processing/aecm/main/source/Android.mk
deleted file mode 100644
index 7ed9f36..0000000
--- a/src/modules/audio_processing/aecm/main/source/Android.mk
+++ /dev/null
@@ -1,55 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aecm
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := echo_control_mobile.c \
- aecm_core.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../utility \
- $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/aecm/main/source/aecm_core.c b/src/modules/audio_processing/aecm/main/source/aecm_core.c
deleted file mode 100644
index f17f1bf..0000000
--- a/src/modules/audio_processing/aecm/main/source/aecm_core.c
+++ /dev/null
@@ -1,2534 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-
-#include "aecm_core.h"
-#include "ring_buffer.h"
-#include "echo_control_mobile.h"
-#include "typedefs.h"
-
-// TODO(bjornv): Will be removed in final version.
-//#include <stdio.h>
-
-#ifdef ARM_WINM_LOG
-#include <stdio.h>
-#include <windows.h>
-#endif
-
-// BANDLAST - BANDFIRST must be < 32
-#define BANDFIRST 12 // Only bit BANDFIRST through bit BANDLAST are processed
-#define BANDLAST 43
-
-#ifdef ARM_WINM
-#define WebRtcSpl_AddSatW32(a,b) _AddSatInt(a,b)
-#define WebRtcSpl_SubSatW32(a,b) _SubSatInt(a,b)
-#endif
-// 16 instructions on most risc machines for 32-bit bitcount !
-
-#ifdef AEC_DEBUG
-FILE *dfile;
-FILE *testfile;
-#endif
-
-#ifdef AECM_SHORT
-
-// Square root of Hanning window in Q14
-static const WebRtc_Word16 kSqrtHanning[] =
-{
- 0, 804, 1606, 2404, 3196, 3981, 4756, 5520,
- 6270, 7005, 7723, 8423, 9102, 9760, 10394, 11003,
- 11585, 12140, 12665, 13160, 13623, 14053, 14449, 14811,
- 15137, 15426, 15679, 15893, 16069, 16207, 16305, 16364,
- 16384
-};
-
-#else
-
-// Square root of Hanning window in Q14
-static const WebRtc_Word16 kSqrtHanning[] = {0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
- 3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019, 8364,
- 8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514, 11795, 12068, 12335,
- 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189, 14384, 14571, 14749, 14918,
- 15079, 15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034, 16111, 16179, 16237,
- 16286, 16325, 16354, 16373, 16384};
-
-#endif
-
-//Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha1 = 32584;
-//Q15 beta = 0.12967166976970 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta1 = 4249;
-//Q15 alpha = 0.94234827210087 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha2 = 30879;
-//Q15 beta = 0.33787806009150 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta2 = 11072;
-//Q15 alpha = 0.82247698684306 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha3 = 26951;
-//Q15 beta = 0.57762063060713 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta3 = 18927;
-
-// Initialization table for echo channel in 8 kHz
-static const WebRtc_Word16 kChannelStored8kHz[PART_LEN1] = {
- 2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418,
- 1451, 1506, 1562, 1644, 1726, 1804, 1882, 1918,
- 1953, 1982, 2010, 2025, 2040, 2034, 2027, 2021,
- 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683,
- 1635, 1604, 1572, 1545, 1517, 1481, 1444, 1405,
- 1367, 1331, 1294, 1270, 1245, 1239, 1233, 1247,
- 1260, 1282, 1303, 1338, 1373, 1407, 1441, 1470,
- 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649,
- 1676
-};
-
-// Initialization table for echo channel in 16 kHz
-static const WebRtc_Word16 kChannelStored16kHz[PART_LEN1] = {
- 2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882,
- 1953, 2010, 2040, 2027, 2014, 1980, 1869, 1732,
- 1635, 1572, 1517, 1444, 1367, 1294, 1245, 1233,
- 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621,
- 1676, 1741, 1802, 1861, 1921, 1983, 2040, 2102,
- 2170, 2265, 2375, 2515, 2651, 2781, 2922, 3075,
- 3253, 3471, 3738, 3976, 4151, 4258, 4308, 4288,
- 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484,
- 3153
-};
-
-#ifdef ARM_WINM_LOG
-HANDLE logFile = NULL;
-#endif
-
-static void WebRtcAecm_ComfortNoise(AecmCore_t* const aecm, const WebRtc_UWord16 * const dfa,
- WebRtc_Word16 * const outReal,
- WebRtc_Word16 * const outImag,
- const WebRtc_Word16 * const lambda);
-
-static __inline WebRtc_UWord32 WebRtcAecm_SetBit(WebRtc_UWord32 in, WebRtc_Word32 pos)
-{
- WebRtc_UWord32 mask, out;
-
- mask = WEBRTC_SPL_SHIFT_W32(1, pos);
- out = (in | mask);
-
- return out;
-}
-
-// WebRtcAecm_Hisser(...)
-//
-// This function compares the binary vector specvec with all rows of the binary matrix specmat
-// and counts per row the number of times they have the same value.
-// Input:
-// - specvec : binary "vector" that is stored in a long
-// - specmat : binary "matrix" that is stored as a vector of long
-// Output:
-// - bcount : "Vector" stored as a long, containing for each row the number of times
-// the matrix row and the input vector have the same value
-//
-//
-void WebRtcAecm_Hisser(const WebRtc_UWord32 specvec, const WebRtc_UWord32 * const specmat,
- WebRtc_UWord32 * const bcount)
-{
- int n;
- WebRtc_UWord32 a, b;
- register WebRtc_UWord32 tmp;
-
- a = specvec;
- // compare binary vector specvec with all rows of the binary matrix specmat
- for (n = 0; n < MAX_DELAY; n++)
- {
- b = specmat[n];
- a = (specvec ^ b);
- // Returns bit counts in tmp
- tmp = a - ((a >> 1) & 033333333333) - ((a >> 2) & 011111111111);
- tmp = ((tmp + (tmp >> 3)) & 030707070707);
- tmp = (tmp + (tmp >> 6));
- tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
-
- bcount[n] = tmp;
- }
-}
-
-// WebRtcAecm_BSpectrum(...)
-//
-// Computes the binary spectrum by comparing the input spectrum with a threshold spectrum.
-//
-// Input:
-// - spectrum : Spectrum of which the binary spectrum should be calculated.
-// - thresvec : Threshold spectrum with which the input spectrum is compared.
-// Return:
-// - out : Binary spectrum
-//
-WebRtc_UWord32 WebRtcAecm_BSpectrum(const WebRtc_UWord16 * const spectrum,
- const WebRtc_UWord16 * const thresvec)
-{
- int k;
- WebRtc_UWord32 out;
-
- out = 0;
- for (k = BANDFIRST; k <= BANDLAST; k++)
- {
- if (spectrum[k] > thresvec[k])
- {
- out = WebRtcAecm_SetBit(out, k - BANDFIRST);
- }
- }
-
- return out;
-}
-
-// WebRtcAecm_MedianEstimator(...)
-//
-// Calculates the median recursively.
-//
-// Input:
-// - newVal : new additional value
-// - medianVec : vector with current medians
-// - factor : factor for smoothing
-//
-// Output:
-// - medianVec : vector with updated median
-//
-int WebRtcAecm_MedianEstimator(const WebRtc_UWord16 newVal, WebRtc_UWord16 * const medianVec,
- const int factor)
-{
- WebRtc_Word32 median;
- WebRtc_Word32 diff;
-
- median = (WebRtc_Word32)medianVec[0];
-
- //median = median + ((newVal-median)>>factor);
- diff = (WebRtc_Word32)newVal - median;
- diff = WEBRTC_SPL_SHIFT_W32(diff, -factor);
- median = median + diff;
-
- medianVec[0] = (WebRtc_UWord16)median;
-
- return 0;
-}
-
-int WebRtcAecm_CreateCore(AecmCore_t **aecmInst)
-{
- AecmCore_t *aecm = malloc(sizeof(AecmCore_t));
- *aecmInst = aecm;
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (WebRtcApm_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtcApm_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtcApm_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtcApm_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- return 0;
-}
-
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-// - aecm : Pointer to the Echo Suppression instance
-// - samplingFreq : Sampling Frequency
-//
-// Output:
-// - aecm : Initialized instance
-//
-// Return value : 0 - Ok
-// -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq)
-{
- int retVal = 0;
- WebRtc_Word16 i;
- WebRtc_Word16 tmp16;
-
- if (samplingFreq != 8000 && samplingFreq != 16000)
- {
- samplingFreq = 8000;
- retVal = -1;
- }
- // sanity check of sampling frequency
- aecm->mult = (WebRtc_Word16)samplingFreq / 8000;
-
- aecm->farBufWritePos = 0;
- aecm->farBufReadPos = 0;
- aecm->knownDelay = 0;
- aecm->lastKnownDelay = 0;
-
- WebRtcApm_InitBuffer(aecm->farFrameBuf);
- WebRtcApm_InitBuffer(aecm->nearNoisyFrameBuf);
- WebRtcApm_InitBuffer(aecm->nearCleanFrameBuf);
- WebRtcApm_InitBuffer(aecm->outFrameBuf);
-
- memset(aecm->xBuf, 0, sizeof(aecm->xBuf));
- memset(aecm->dBufClean, 0, sizeof(aecm->dBufClean));
- memset(aecm->dBufNoisy, 0, sizeof(aecm->dBufNoisy));
- memset(aecm->outBuf, 0, sizeof(WebRtc_Word16) * PART_LEN);
-
- aecm->seed = 666;
- aecm->totCount = 0;
-
- memset(aecm->xfaHistory, 0, sizeof(WebRtc_UWord16) * (PART_LEN1) * MAX_DELAY);
-
- aecm->delHistoryPos = MAX_DELAY;
-
- memset(aecm->medianYlogspec, 0, sizeof(WebRtc_UWord16) * PART_LEN1);
- memset(aecm->medianXlogspec, 0, sizeof(WebRtc_UWord16) * PART_LEN1);
- memset(aecm->medianBCount, 0, sizeof(WebRtc_UWord16) * MAX_DELAY);
- memset(aecm->bxHistory, 0, sizeof(aecm->bxHistory));
-
- // Initialize to reasonable values
- aecm->currentDelay = 8;
- aecm->previousDelay = 8;
- aecm->delayAdjust = 0;
-
- aecm->nlpFlag = 1;
- aecm->fixedDelay = -1;
-
- memset(aecm->xfaQDomainBuf, 0, sizeof(WebRtc_Word16) * MAX_DELAY);
- aecm->dfaCleanQDomain = 0;
- aecm->dfaCleanQDomainOld = 0;
- aecm->dfaNoisyQDomain = 0;
- aecm->dfaNoisyQDomainOld = 0;
-
- memset(aecm->nearLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
- memset(aecm->farLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
- memset(aecm->echoAdaptLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
- memset(aecm->echoStoredLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
-
- // Initialize the echo channels with a stored shape.
- if (samplingFreq == 8000)
- {
- memcpy(aecm->channelAdapt16, kChannelStored8kHz, sizeof(WebRtc_Word16) * PART_LEN1);
- }
- else
- {
- memcpy(aecm->channelAdapt16, kChannelStored16kHz, sizeof(WebRtc_Word16) * PART_LEN1);
- }
- memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)(aecm->channelAdapt16[i]), 16);
- }
-
- memset(aecm->echoFilt, 0, sizeof(WebRtc_Word32) * PART_LEN1);
- memset(aecm->nearFilt, 0, sizeof(WebRtc_Word16) * PART_LEN1);
- aecm->noiseEstCtr = 0;
-
- aecm->cngMode = AecmTrue;
-
- // Increase the noise Q domain with increasing frequency, to correspond to the
- // expected energy levels.
- // Also shape the initial noise level with this consideration.
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- if (i < PART_LEN1 >> 2)
- {
- aecm->noiseEstQDomain[i] = 10;
- tmp16 = PART_LEN1 - i;
- aecm->noiseEst[i] = (tmp16 * tmp16) << 4;
- } else if (i < PART_LEN1 >> 1)
- {
- aecm->noiseEstQDomain[i] = 11;
- tmp16 = PART_LEN1 - i;
- aecm->noiseEst[i] = ((tmp16 * tmp16) << 4) << 1;
- } else
- {
- aecm->noiseEstQDomain[i] = 12;
- aecm->noiseEst[i] = aecm->noiseEst[(PART_LEN1 >> 1) - 1] << 1;
- }
- }
-#else
- for (i = 0; i < PART_LEN1 >> 2; i++)
- {
- aecm->noiseEstQDomain[i] = 10;
- tmp16 = PART_LEN1 - i;
- aecm->noiseEst[i] = (tmp16 * tmp16) << 4;
- }
- for (; i < PART_LEN1 >> 1; i++)
- {
- aecm->noiseEstQDomain[i] = 11;
- tmp16 = PART_LEN1 - i;
- aecm->noiseEst[i] = ((tmp16 * tmp16) << 4) << 1;
- }
- for (; i < PART_LEN1; i++)
- {
- aecm->noiseEstQDomain[i] = 12;
- aecm->noiseEst[i] = aecm->noiseEst[(PART_LEN1 >> 1) - 1] << 1;
- }
-#endif
-
- aecm->mseAdaptOld = 1000;
- aecm->mseStoredOld = 1000;
- aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
-
- aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
- aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
- aecm->farEnergyMaxMin = 0;
- aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
- // beginning.
- aecm->farEnergyMSE = 0;
- aecm->currentVADValue = 0;
- aecm->vadUpdateCount = 0;
- aecm->firstVAD = 1;
-
- aecm->delayCount = 0;
- aecm->newDelayCorrData = 0;
- aecm->lastDelayUpdateCount = 0;
- memset(aecm->delayCorrelation, 0, sizeof(WebRtc_Word16) * ((CORR_MAX << 1) + 1));
-
- aecm->startupState = 0;
- aecm->mseChannelCount = 0;
- aecm->supGain = SUPGAIN_DEFAULT;
- aecm->supGainOld = SUPGAIN_DEFAULT;
- aecm->delayOffsetFlag = 0;
-
- memset(aecm->delayHistogram, 0, sizeof(aecm->delayHistogram));
- aecm->delayVadCount = 0;
- aecm->maxDelayHistIdx = 0;
- aecm->lastMinPos = 0;
-
- aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
- aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
- aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
- aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
-
- return 0;
-}
-
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag, int delayOffsetFlag)
-{
- aecm->nlpFlag = nlpFlag;
- aecm->fixedDelay = delay;
- aecm->delayOffsetFlag = delayOffsetFlag;
-
- return 0;
-}
-
-// WebRtcAecm_GetNewDelPos(...)
-//
-// Moves the pointer to the next entry. Returns to zero if max position reached.
-//
-// Input:
-// - aecm : Pointer to the AECM instance
-// Return:
-// - pos : New position in the history.
-//
-//
-WebRtc_Word16 WebRtcAecm_GetNewDelPos(AecmCore_t * const aecm)
-{
- WebRtc_Word16 pos;
-
- pos = aecm->delHistoryPos;
- pos++;
- if (pos >= MAX_DELAY)
- {
- pos = 0;
- }
- aecm->delHistoryPos = pos;
-
- return pos;
-}
-
-// WebRtcAecm_EstimateDelay(...)
-//
-// Estimate the delay of the echo signal.
-//
-// Inputs:
-// - aecm : Pointer to the AECM instance
-// - farSpec : Delayed farend magnitude spectrum
-// - nearSpec : Nearend magnitude spectrum
-// - stages : Q-domain of xxFIX and yyFIX (without dynamic Q-domain)
-// - xfaQ : normalization factor, i.e., Q-domain before FFT
-// Return:
-// - delay : Estimated delay
-//
-WebRtc_Word16 WebRtcAecm_EstimateDelay(AecmCore_t * const aecm,
- const WebRtc_UWord16 * const farSpec,
- const WebRtc_UWord16 * const nearSpec,
- const WebRtc_Word16 xfaQ)
-{
- WebRtc_UWord32 bxspectrum, byspectrum;
- WebRtc_UWord32 bcount[MAX_DELAY];
-
- int i, res;
-
- WebRtc_UWord16 xmean[PART_LEN1], ymean[PART_LEN1];
- WebRtc_UWord16 dtmp1;
- WebRtc_Word16 fcount[MAX_DELAY];
-
- //WebRtc_Word16 res;
- WebRtc_Word16 histpos;
- WebRtc_Word16 maxHistLvl;
- WebRtc_UWord16 *state;
- WebRtc_Word16 minpos = -1;
-
- enum
- {
- kVadCountThreshold = 25
- };
- enum
- {
- kMaxHistogram = 600
- };
-
- histpos = WebRtcAecm_GetNewDelPos(aecm);
-
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->xfaHistory[i][histpos] = farSpec[i];
-
- state = &(aecm->medianXlogspec[i]);
- res = WebRtcAecm_MedianEstimator(farSpec[i], state, 6);
-
- state = &(aecm->medianYlogspec[i]);
- res = WebRtcAecm_MedianEstimator(nearSpec[i], state, 6);
-
- // Mean:
- // FLOAT:
- // ymean = dtmp2/MAX_DELAY
- //
- // FIX:
- // input: dtmp2FIX in Q0
- // output: ymeanFIX in Q8
- // 20 = 1/MAX_DELAY in Q13 = 1/MAX_DELAY * 2^13
- xmean[i] = (aecm->medianXlogspec[i]);
- ymean[i] = (aecm->medianYlogspec[i]);
-
- }
- // Update Q-domain buffer
- aecm->xfaQDomainBuf[histpos] = xfaQ;
-
- // Get binary spectra
- // FLOAT:
- // bxspectrum = bspectrum(xlogspec, xmean);
- //
- // FIX:
- // input: xlogspecFIX,ylogspecFIX in Q8
- // xmeanFIX, ymeanFIX in Q8
- // output: unsigned long bxspectrum, byspectrum in Q0
- bxspectrum = WebRtcAecm_BSpectrum(farSpec, xmean);
- byspectrum = WebRtcAecm_BSpectrum(nearSpec, ymean);
-
- // Shift binary spectrum history
- memmove(&(aecm->bxHistory[1]), &(aecm->bxHistory[0]),
- (MAX_DELAY - 1) * sizeof(WebRtc_UWord32));
-
- aecm->bxHistory[0] = bxspectrum;
-
- // Compare with delayed spectra
- WebRtcAecm_Hisser(byspectrum, aecm->bxHistory, bcount);
-
- for (i = 0; i < MAX_DELAY; i++)
- {
- // Update sum
- // bcount is constrained to [0, 32], meaning we can smooth with a factor up to 2^11.
- dtmp1 = (WebRtc_UWord16)bcount[i];
- dtmp1 = WEBRTC_SPL_LSHIFT_W16(dtmp1, 9);
- state = &(aecm->medianBCount[i]);
- res = WebRtcAecm_MedianEstimator(dtmp1, state, 9);
- fcount[i] = (aecm->medianBCount[i]);
- }
-
- // Find minimum
- minpos = WebRtcSpl_MinIndexW16(fcount, MAX_DELAY);
-
- // If the farend has been active sufficiently long, begin accumulating a histogram
- // of the minimum positions. Search for the maximum bin to determine the delay.
- if (aecm->currentVADValue == 1)
- {
- if (aecm->delayVadCount >= kVadCountThreshold)
- {
- // Increment the histogram at the current minimum position.
- if (aecm->delayHistogram[minpos] < kMaxHistogram)
- {
- aecm->delayHistogram[minpos] += 3;
- }
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- // Decrement the entire histogram.
- for (i = 0; i < MAX_DELAY; i++)
- {
- if (aecm->delayHistogram[i] > 0)
- {
- aecm->delayHistogram[i]--;
- }
- }
-
- // Select the histogram index corresponding to the maximum bin as the delay.
- maxHistLvl = 0;
- aecm->maxDelayHistIdx = 0;
- for (i = 0; i < MAX_DELAY; i++)
- {
- if (aecm->delayHistogram[i] > maxHistLvl)
- {
- maxHistLvl = aecm->delayHistogram[i];
- aecm->maxDelayHistIdx = i;
- }
- }
-#else
- maxHistLvl = 0;
- aecm->maxDelayHistIdx = 0;
-
- for (i = 0; i < MAX_DELAY; i++)
- {
- WebRtc_Word16 tempVar = aecm->delayHistogram[i];
-
- // Decrement the entire histogram.
- if (tempVar > 0)
- {
- tempVar--;
- aecm->delayHistogram[i] = tempVar;
-
- // Select the histogram index corresponding to the maximum bin as the delay.
- if (tempVar > maxHistLvl)
- {
- maxHistLvl = tempVar;
- aecm->maxDelayHistIdx = i;
- }
- }
- }
-#endif
- } else
- {
- aecm->delayVadCount++;
- }
- } else
- {
- aecm->delayVadCount = 0;
- }
-
- return aecm->maxDelayHistIdx;
-}
-
-int WebRtcAecm_FreeCore(AecmCore_t *aecm)
-{
- if (aecm == NULL)
- {
- return -1;
- }
-
- WebRtcApm_FreeBuffer(aecm->farFrameBuf);
- WebRtcApm_FreeBuffer(aecm->nearNoisyFrameBuf);
- WebRtcApm_FreeBuffer(aecm->nearCleanFrameBuf);
- WebRtcApm_FreeBuffer(aecm->outFrameBuf);
-
- free(aecm);
-
- return 0;
-}
-
-void WebRtcAecm_ProcessFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const WebRtc_Word16 * const nearendNoisy,
- const WebRtc_Word16 * const nearendClean,
- WebRtc_Word16 * const out)
-{
- WebRtc_Word16 farBlock[PART_LEN];
- WebRtc_Word16 nearNoisyBlock[PART_LEN];
- WebRtc_Word16 nearCleanBlock[PART_LEN];
- WebRtc_Word16 outBlock[PART_LEN];
- WebRtc_Word16 farFrame[FRAME_LEN];
- int size = 0;
-
- // Buffer the current frame.
- // Fetch an older one corresponding to the delay.
- WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
- WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
-
- // Buffer the synchronized far and near frames,
- // to pass the smaller blocks individually.
- WebRtcApm_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
- WebRtcApm_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
- if (nearendClean != NULL)
- {
- WebRtcApm_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
- }
-
- // Process as many blocks as possible.
- while (WebRtcApm_get_buffer_size(aecm->farFrameBuf) >= PART_LEN)
- {
- WebRtcApm_ReadBuffer(aecm->farFrameBuf, farBlock, PART_LEN);
- WebRtcApm_ReadBuffer(aecm->nearNoisyFrameBuf, nearNoisyBlock, PART_LEN);
- if (nearendClean != NULL)
- {
- WebRtcApm_ReadBuffer(aecm->nearCleanFrameBuf, nearCleanBlock, PART_LEN);
- WebRtcAecm_ProcessBlock(aecm, farBlock, nearNoisyBlock, nearCleanBlock, outBlock);
- } else
- {
- WebRtcAecm_ProcessBlock(aecm, farBlock, nearNoisyBlock, NULL, outBlock);
- }
-
- WebRtcApm_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
- }
-
- // Stuff the out buffer if we have less than a frame to output.
- // This should only happen for the first frame.
- size = WebRtcApm_get_buffer_size(aecm->outFrameBuf);
- if (size < FRAME_LEN)
- {
- WebRtcApm_StuffBuffer(aecm->outFrameBuf, FRAME_LEN - size);
- }
-
- // Obtain an output frame.
- WebRtcApm_ReadBuffer(aecm->outFrameBuf, out, FRAME_LEN);
-}
-
-// WebRtcAecm_AsymFilt(...)
-//
-// Performs asymmetric filtering.
-//
-// Inputs:
-// - filtOld : Previous filtered value.
-// - inVal : New input value.
-// - stepSizePos : Step size when we have a positive contribution.
-// - stepSizeNeg : Step size when we have a negative contribution.
-//
-// Output:
-//
-// Return: - Filtered value.
-//
-WebRtc_Word16 WebRtcAecm_AsymFilt(const WebRtc_Word16 filtOld, const WebRtc_Word16 inVal,
- const WebRtc_Word16 stepSizePos,
- const WebRtc_Word16 stepSizeNeg)
-{
- WebRtc_Word16 retVal;
-
- if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
- {
- return inVal;
- }
- retVal = filtOld;
- if (filtOld > inVal)
- {
- retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
- } else
- {
- retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
- }
-
- return retVal;
-}
-
-// WebRtcAecm_CalcEnergies(...)
-//
-// This function calculates the log of energies for nearend, farend and estimated
-// echoes. There is also an update of energy decision levels, i.e. internl VAD.
-//
-//
-// @param aecm [i/o] Handle of the AECM instance.
-// @param delayDiff [in] Delay position in farend buffer.
-// @param nearEner [in] Near end energy for current block (Q[aecm->dfaQDomain]).
-// @param echoEst [i/o] Estimated echo
-// (Q[aecm->xfaQDomain[delayDiff]+RESOLUTION_CHANNEL16]).
-//
-void WebRtcAecm_CalcEnergies(AecmCore_t * const aecm, const WebRtc_Word16 delayDiff,
- const WebRtc_UWord32 nearEner, WebRtc_Word32 * const echoEst)
-{
- // Local variables
- WebRtc_UWord32 tmpAdapt, tmpStored, tmpFar;
-
- int i;
-
- WebRtc_Word16 zeros, frac;
- WebRtc_Word16 tmp16;
- WebRtc_Word16 increase_max_shifts = 4;
- WebRtc_Word16 decrease_max_shifts = 11;
- WebRtc_Word16 increase_min_shifts = 11;
- WebRtc_Word16 decrease_min_shifts = 3;
-
- // Get log of near end energy and store in buffer
-
- // Shift buffer
- memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
- // Logarithm of integrated magnitude spectrum (nearEner)
- if (nearEner)
- {
- zeros = WebRtcSpl_NormU32(nearEner);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
- (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
- 23);
- // log2 in Q8
- aecm->nearLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- aecm->nearLogEnergy[0] -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
- } else
- {
- aecm->nearLogEnergy[0] = 0;
- }
- aecm->nearLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
- // END: Get log of near end energy
-
- // Get energy for the delayed far end signal and estimated
- // echo using both stored and adapted channels.
- tmpAdapt = 0;
- tmpStored = 0;
- tmpFar = 0;
-
- for (i = 0; i < PART_LEN1; i++)
- {
- // Get estimated echo energies for adaptive channel and stored channel
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- tmpFar += (WebRtc_UWord32)(aecm->xfaHistory[i][delayDiff]);
- tmpAdapt += WEBRTC_SPL_UMUL_16_16(aecm->channelAdapt16[i],
- aecm->xfaHistory[i][delayDiff]);
- tmpStored += (WebRtc_UWord32)echoEst[i];
- }
- // Shift buffers
- memmove(aecm->farLogEnergy + 1, aecm->farLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
- memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
- memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
- // Logarithm of delayed far end energy
- if (tmpFar)
- {
- zeros = WebRtcSpl_NormU32(tmpFar);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
- & 0x7FFFFFFF), 23);
- // log2 in Q8
- aecm->farLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- aecm->farLogEnergy[0] -= WEBRTC_SPL_LSHIFT_W16(aecm->xfaQDomainBuf[delayDiff], 8);
- } else
- {
- aecm->farLogEnergy[0] = 0;
- }
- aecm->farLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
- // Logarithm of estimated echo energy through adapted channel
- if (tmpAdapt)
- {
- zeros = WebRtcSpl_NormU32(tmpAdapt);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
- & 0x7FFFFFFF), 23);
- //log2 in Q8
- aecm->echoAdaptLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- aecm->echoAdaptLogEnergy[0]
- -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + aecm->xfaQDomainBuf[delayDiff], 8);
- } else
- {
- aecm->echoAdaptLogEnergy[0] = 0;
- }
- aecm->echoAdaptLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
- // Logarithm of estimated echo energy through stored channel
- if (tmpStored)
- {
- zeros = WebRtcSpl_NormU32(tmpStored);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
- & 0x7FFFFFFF), 23);
- //log2 in Q8
- aecm->echoStoredLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- aecm->echoStoredLogEnergy[0]
- -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + aecm->xfaQDomainBuf[delayDiff], 8);
- } else
- {
- aecm->echoStoredLogEnergy[0] = 0;
- }
- aecm->echoStoredLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
- // Update farend energy levels (min, max, vad, mse)
- if (aecm->farLogEnergy[0] > FAR_ENERGY_MIN)
- {
- if (aecm->startupState == 0)
- {
- increase_max_shifts = 2;
- decrease_min_shifts = 2;
- increase_min_shifts = 8;
- }
-
- aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy[0],
- increase_min_shifts, decrease_min_shifts);
- aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy[0],
- increase_max_shifts, decrease_max_shifts);
- aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
-
- // Dynamic VAD region size
- tmp16 = 2560 - aecm->farEnergyMin;
- if (tmp16 > 0)
- {
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9);
- } else
- {
- tmp16 = 0;
- }
- tmp16 += FAR_ENERGY_VAD_REGION;
-
- if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
- {
- // In startup phase or VAD update halted
- aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
- } else
- {
- if (aecm->farEnergyVAD > aecm->farLogEnergy[0])
- {
- aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy[0] + tmp16
- - aecm->farEnergyVAD, 6);
- aecm->vadUpdateCount = 0;
- } else
- {
- aecm->vadUpdateCount++;
- }
- }
- // Put MSE threshold higher than VAD
- aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
- }
-
- // Update VAD variables
- if (aecm->farLogEnergy[0] > aecm->farEnergyVAD)
- {
- if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
- {
- // We are in startup or have significant dynamics in input speech level
- aecm->currentVADValue = 1;
- }
- } else
- {
- aecm->currentVADValue = 0;
- }
- if ((aecm->currentVADValue) && (aecm->firstVAD))
- {
- aecm->firstVAD = 0;
- if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
- {
- // The estimated echo has higher energy than the near end signal. This means that
- // the initialization was too aggressive. Scale down by a factor 8
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->channelAdapt16[i] >>= 3;
- }
- // Compensate the adapted echo energy level accordingly.
- aecm->echoAdaptLogEnergy[0] -= (3 << 8);
- aecm->firstVAD = 1;
- }
- }
- // END: Energies of delayed far, echo estimates
- // TODO(bjornv): Will be removed in final version.
-#ifdef VAD_DATA
- fwrite(&(aecm->currentVADValue), sizeof(WebRtc_Word16), 1, aecm->vad_file);
- fwrite(&(aecm->currentDelay), sizeof(WebRtc_Word16), 1, aecm->delay_file);
- fwrite(&(aecm->farLogEnergy[0]), sizeof(WebRtc_Word16), 1, aecm->far_cur_file);
- fwrite(&(aecm->farEnergyMin), sizeof(WebRtc_Word16), 1, aecm->far_min_file);
- fwrite(&(aecm->farEnergyMax), sizeof(WebRtc_Word16), 1, aecm->far_max_file);
- fwrite(&(aecm->farEnergyVAD), sizeof(WebRtc_Word16), 1, aecm->far_vad_file);
-#endif
-}
-
-// WebRtcAecm_CalcStepSize(...)
-//
-// This function calculates the step size used in channel estimation
-//
-//
-// @param aecm [in] Handle of the AECM instance.
-// @param mu [out] (Return value) Stepsize in log2(), i.e. number of shifts.
-//
-//
-WebRtc_Word16 WebRtcAecm_CalcStepSize(AecmCore_t * const aecm)
-{
-
- WebRtc_Word32 tmp32;
- WebRtc_Word16 tmp16;
- WebRtc_Word16 mu;
-
- // Here we calculate the step size mu used in the
- // following NLMS based Channel estimation algorithm
- mu = MU_MAX;
- if (!aecm->currentVADValue)
- {
- // Far end energy level too low, no channel update
- mu = 0;
- } else if (aecm->startupState > 0)
- {
- if (aecm->farEnergyMin >= aecm->farEnergyMax)
- {
- mu = MU_MIN;
- } else
- {
- tmp16 = (aecm->farLogEnergy[0] - aecm->farEnergyMin);
- tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, MU_DIFF);
- tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
- mu = MU_MIN - 1 - (WebRtc_Word16)(tmp32);
- // The -1 is an alternative to rounding. This way we get a larger
- // stepsize, so we in some sense compensate for truncation in NLMS
- }
- if (mu < MU_MAX)
- {
- mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
- }
- }
- // END: Update step size
-
- return mu;
-}
-
-// WebRtcAecm_UpdateChannel(...)
-//
-// This function performs channel estimation. NLMS and decision on channel storage.
-//
-//
-// @param aecm [i/o] Handle of the AECM instance.
-// @param dfa [in] Absolute value of the nearend signal (Q[aecm->dfaQDomain])
-// @param delayDiff [in] Delay position in farend buffer.
-// @param mu [in] NLMS step size.
-// @param echoEst [i/o] Estimated echo
-// (Q[aecm->xfaQDomain[delayDiff]+RESOLUTION_CHANNEL16]).
-//
-void WebRtcAecm_UpdateChannel(AecmCore_t * const aecm, const WebRtc_UWord16 * const dfa,
- const WebRtc_Word16 delayDiff, const WebRtc_Word16 mu,
- WebRtc_Word32 * const echoEst)
-{
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 mseStored;
- WebRtc_Word32 mseAdapt;
-
- int i;
-
- WebRtc_Word16 zerosFar, zerosNum, zerosCh, zerosDfa;
- WebRtc_Word16 shiftChFar, shiftNum, shift2ResChan;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 xfaQ, dfaQ;
-
- // This is the channel estimation algorithm. It is base on NLMS but has a variable step
- // length, which was calculated above.
- if (mu)
- {
- for (i = 0; i < PART_LEN1; i++)
- {
- // Determine norm of channel and farend to make sure we don't get overflow in
- // multiplication
- zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
- zerosFar = WebRtcSpl_NormU32((WebRtc_UWord32)aecm->xfaHistory[i][delayDiff]);
- if (zerosCh + zerosFar > 31)
- {
- // Multiplication is safe
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
- aecm->xfaHistory[i][delayDiff]);
- shiftChFar = 0;
- } else
- {
- // We need to shift down before multiplication
- shiftChFar = 32 - zerosCh - zerosFar;
- tmpU32no1
- = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i],
- shiftChFar),
- aecm->xfaHistory[i][delayDiff]);
- }
- // Determine Q-domain of numerator
- zerosNum = WebRtcSpl_NormU32(tmpU32no1);
- if (dfa[i])
- {
- zerosDfa = WebRtcSpl_NormU32((WebRtc_UWord32)dfa[i]);
- } else
- {
- zerosDfa = 32;
- }
- tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain - RESOLUTION_CHANNEL32
- - aecm->xfaQDomainBuf[delayDiff] + shiftChFar;
- if (zerosNum > tmp16no1 + 1)
- {
- xfaQ = tmp16no1;
- dfaQ = zerosDfa - 2;
- } else
- {
- xfaQ = zerosNum - 2;
- dfaQ = RESOLUTION_CHANNEL32 + aecm->xfaQDomainBuf[delayDiff]
- - aecm->dfaNoisyQDomain - shiftChFar + xfaQ;
- }
- // Add in the same Q-domain
- tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
- tmpU32no2 = WEBRTC_SPL_SHIFT_W32((WebRtc_UWord32)dfa[i], dfaQ);
- tmp32no1 = (WebRtc_Word32)tmpU32no2 - (WebRtc_Word32)tmpU32no1;
- zerosNum = WebRtcSpl_NormW32(tmp32no1);
- if ((tmp32no1) && (aecm->xfaHistory[i][delayDiff] > (CHANNEL_VAD
- << aecm->xfaQDomainBuf[delayDiff])))
- {
- //
- // Update is needed
- //
- // This is what we would like to compute
- //
- // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * aecm->xfaHistory[i][delayDiff])
- // tmp32norm = (i + 1)
- // aecm->channelAdapt[i] += (2^mu) * tmp32no1
- // / (tmp32norm * aecm->xfaHistory[i][delayDiff])
- //
-
- // Make sure we don't get overflow in multiplication.
- if (zerosNum + zerosFar > 31)
- {
- if (tmp32no1 > 0)
- {
- tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmp32no1,
- aecm->xfaHistory[i][delayDiff]);
- } else
- {
- tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
- aecm->xfaHistory[i][delayDiff]);
- }
- shiftNum = 0;
- } else
- {
- shiftNum = 32 - (zerosNum + zerosFar);
- if (tmp32no1 > 0)
- {
- tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
- WEBRTC_SPL_RSHIFT_W32(tmp32no1, shiftNum),
- aecm->xfaHistory[i][delayDiff]);
- } else
- {
- tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
- WEBRTC_SPL_RSHIFT_W32(-tmp32no1, shiftNum),
- aecm->xfaHistory[i][delayDiff]);
- }
- }
- // Normalize with respect to frequency bin
- tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
- // Make sure we are in the right Q-domain
- shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
- if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
- {
- tmp32no2 = WEBRTC_SPL_WORD32_MAX;
- } else
- {
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
- }
- aecm->channelAdapt32[i] = WEBRTC_SPL_ADD_SAT_W32(aecm->channelAdapt32[i],
- tmp32no2);
- if (aecm->channelAdapt32[i] < 0)
- {
- // We can never have negative channel gain
- aecm->channelAdapt32[i] = 0;
- }
- aecm->channelAdapt16[i]
- = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], 16);
- }
- }
- }
- // END: Adaptive channel update
-
- // Determine if we should store or restore the channel
- if ((aecm->startupState == 0) & (aecm->currentVADValue))
- {
- // During startup we store the channel every block.
- memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
- // TODO(bjornv): Will be removed in final version.
-#ifdef STORE_CHANNEL_DATA
- fwrite(aecm->channelStored, sizeof(WebRtc_Word16), PART_LEN1, aecm->channel_file_init);
-#endif
- // Recalculate echo estimate
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- }
-#else
- for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
- {
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
- i++;
- }
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- aecm->xfaHistory[i][delayDiff]);
-#endif
- } else
- {
- if (aecm->farLogEnergy[0] < aecm->farEnergyMSE)
- {
- aecm->mseChannelCount = 0;
- aecm->delayCount = 0;
- } else
- {
- aecm->mseChannelCount++;
- aecm->delayCount++;
- }
- // Enough data for validation. Store channel if we can.
- if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
- {
- // We have enough data.
- // Calculate MSE of "Adapt" and "Stored" versions.
- // It is actually not MSE, but average absolute error.
- mseStored = 0;
- mseAdapt = 0;
- for (i = 0; i < MIN_MSE_COUNT; i++)
- {
- tmp32no1 = ((WebRtc_Word32)aecm->echoStoredLogEnergy[i]
- - (WebRtc_Word32)aecm->nearLogEnergy[i]);
- tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
- mseStored += tmp32no2;
-
- tmp32no1 = ((WebRtc_Word32)aecm->echoAdaptLogEnergy[i]
- - (WebRtc_Word32)aecm->nearLogEnergy[i]);
- tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
- mseAdapt += tmp32no2;
- }
- if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
- & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
- * aecm->mseAdaptOld)))
- {
- // The stored channel has a significantly lower MSE than the adaptive one for
- // two consecutive calculations. Reset the adaptive channel.
- memcpy(aecm->channelAdapt16, aecm->channelStored,
- sizeof(WebRtc_Word16) * PART_LEN1);
- // Restore the W32 channel
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->channelAdapt32[i]
- = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
- }
-#else
- for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
- {
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
- i++;
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
- i++;
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
- i++;
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
- i++;
- }
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-#endif
-
- } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
- < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
- {
- // The adaptive channel has a significantly lower MSE than the stored one.
- // The MSE for the adaptive channel has also been low for two consecutive
- // calculations. Store the adaptive channel.
- memcpy(aecm->channelStored, aecm->channelAdapt16,
- sizeof(WebRtc_Word16) * PART_LEN1);
- // TODO(bjornv): Will be removed in final version.
-#ifdef STORE_CHANNEL_DATA
- fwrite(aecm->channelStored, sizeof(WebRtc_Word16), PART_LEN1,
- aecm->channel_file);
-#endif
-// Recalculate echo estimate
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- echoEst[i]
- = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
- }
-#else
- for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
- {
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
- i++;
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
- i++;
- }
- echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-#endif
- // Update threshold
- if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
- {
- aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
- } else
- {
- aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt
- - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8);
- }
-
- }
-
- // Reset counter
- aecm->mseChannelCount = 0;
-
- // Store the MSE values.
- aecm->mseStoredOld = mseStored;
- aecm->mseAdaptOld = mseAdapt;
- }
- }
- // END: Determine if we should store or reset channel estimate.
-}
-
-// WebRtcAecm_CalcSuppressionGain(...)
-//
-// This function calculates the suppression gain that is used in the Wiener filter.
-//
-//
-// @param aecm [i/n] Handle of the AECM instance.
-// @param supGain [out] (Return value) Suppression gain with which to scale the noise
-// level (Q14).
-//
-//
-WebRtc_Word16 WebRtcAecm_CalcSuppressionGain(AecmCore_t * const aecm)
-{
- WebRtc_Word32 tmp32no1;
-
- WebRtc_Word16 supGain;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 dE = 0;
-
- // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
- // end energy and echo estimation error.
- supGain = SUPGAIN_DEFAULT;
- // Adjust for the far end signal level. A low signal level indicates no far end signal,
- // hence we set the suppression gain to 0
- if (!aecm->currentVADValue)
- {
- supGain = 0;
- } else
- {
- // Adjust for possible double talk. If we have large variations in estimation error we
- // likely have double talk (or poor channel).
- tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
- dE = WEBRTC_SPL_ABS_W16(tmp16no1);
-
- if (dE < ENERGY_DEV_TOL)
- {
- // Likely no double talk. The better estimation, the more we can suppress signal.
- // Update counters
- if (dE < SUPGAIN_EPC_DT)
- {
- tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffAB, dE);
- tmp32no1 += (SUPGAIN_EPC_DT >> 1);
- tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
- supGain = aecm->supGainErrParamA - tmp16no1;
- } else
- {
- tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffBD,
- (ENERGY_DEV_TOL - dE));
- tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
- tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
- - SUPGAIN_EPC_DT));
- supGain = aecm->supGainErrParamD + tmp16no1;
- }
- } else
- {
- // Likely in double talk. Use default value
- supGain = aecm->supGainErrParamD;
- }
- }
-
- if (supGain > aecm->supGainOld)
- {
- tmp16no1 = supGain;
- } else
- {
- tmp16no1 = aecm->supGainOld;
- }
- aecm->supGainOld = supGain;
- if (tmp16no1 < aecm->supGain)
- {
- aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
- } else
- {
- aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
- }
-
- // END: Update suppression gain
-
- return aecm->supGain;
-}
-
-// WebRtcAecm_DelayCompensation(...)
-//
-// Secondary delay estimation that can be used as a backup or for validation. This function is
-// still under construction and not activated in current version.
-//
-//
-// @param aecm [i/o] Handle of the AECM instance.
-//
-//
-void WebRtcAecm_DelayCompensation(AecmCore_t * const aecm)
-{
- int i, j;
- WebRtc_Word32 delayMeanEcho[CORR_BUF_LEN];
- WebRtc_Word32 delayMeanNear[CORR_BUF_LEN];
- WebRtc_Word16 sumBitPattern, bitPatternEcho, bitPatternNear, maxPos, maxValue,
- maxValueLeft, maxValueRight;
-
- // Check delay (calculate the delay offset (if we can)).
- if ((aecm->startupState > 0) & (aecm->delayCount >= CORR_MAX_BUF) & aecm->delayOffsetFlag)
- {
- // Calculate mean values
- for (i = 0; i < CORR_BUF_LEN; i++)
- {
- delayMeanEcho[i] = 0;
- delayMeanNear[i] = 0;
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (j = 0; j < CORR_WIDTH; j++)
- {
- delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
- delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
- }
-#else
- for (j = 0; j < CORR_WIDTH -1; )
- {
- delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
- delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
- j++;
- delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
- delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
- j++;
- }
- delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
- delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
-#endif
- }
- // Calculate correlation values
- for (i = 0; i < CORR_BUF_LEN; i++)
- {
- sumBitPattern = 0;
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (j = 0; j < CORR_WIDTH; j++)
- {
- bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
- + j] * CORR_WIDTH > delayMeanEcho[i]);
- bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
- + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
- sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
- }
-#else
- for (j = 0; j < CORR_WIDTH -1; )
- {
- bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
- + j] * CORR_WIDTH > delayMeanEcho[i]);
- bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
- + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
- sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
- j++;
- bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
- + j] * CORR_WIDTH > delayMeanEcho[i]);
- bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
- + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
- sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
- j++;
- }
- bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i + j]
- * CORR_WIDTH > delayMeanEcho[i]);
- bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX + j]
- * CORR_WIDTH > delayMeanNear[CORR_MAX]);
- sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
-#endif
- aecm->delayCorrelation[i] = sumBitPattern;
- }
- aecm->newDelayCorrData = 1; // Indicate we have new correlation data to evaluate
- }
- if ((aecm->startupState == 2) & (aecm->lastDelayUpdateCount > (CORR_WIDTH << 1))
- & aecm->newDelayCorrData)
- {
- // Find maximum value and maximum position as well as values on the sides.
- maxPos = 0;
- maxValue = aecm->delayCorrelation[0];
- maxValueLeft = maxValue;
- maxValueRight = aecm->delayCorrelation[CORR_DEV];
- for (i = 1; i < CORR_BUF_LEN; i++)
- {
- if (aecm->delayCorrelation[i] > maxValue)
- {
- maxValue = aecm->delayCorrelation[i];
- maxPos = i;
- if (maxPos < CORR_DEV)
- {
- maxValueLeft = aecm->delayCorrelation[0];
- maxValueRight = aecm->delayCorrelation[i + CORR_DEV];
- } else if (maxPos > (CORR_MAX << 1) - CORR_DEV)
- {
- maxValueLeft = aecm->delayCorrelation[i - CORR_DEV];
- maxValueRight = aecm->delayCorrelation[(CORR_MAX << 1)];
- } else
- {
- maxValueLeft = aecm->delayCorrelation[i - CORR_DEV];
- maxValueRight = aecm->delayCorrelation[i + CORR_DEV];
- }
- }
- }
- if ((maxPos > 0) & (maxPos < (CORR_MAX << 1)))
- {
- // Avoid maximum at boundaries. The maximum peak has to be higher than
- // CORR_MAX_LEVEL. It also has to be sharp, i.e. the value CORR_DEV bins off should
- // be CORR_MAX_LOW lower than the maximum.
- if ((maxValue > CORR_MAX_LEVEL) & (maxValueLeft < maxValue - CORR_MAX_LOW)
- & (maxValueRight < maxValue - CORR_MAX_LOW))
- {
- aecm->delayAdjust += CORR_MAX - maxPos;
- aecm->newDelayCorrData = 0;
- aecm->lastDelayUpdateCount = 0;
- }
- }
- }
- // END: "Check delay"
-}
-
-void WebRtcAecm_ProcessBlock(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const WebRtc_Word16 * const nearendNoisy,
- const WebRtc_Word16 * const nearendClean,
- WebRtc_Word16 * const output)
-{
- int i, j;
-
- WebRtc_UWord32 xfaSum;
- WebRtc_UWord32 dfaNoisySum;
- WebRtc_UWord32 echoEst32Gained;
- WebRtc_UWord32 tmpU32;
-
- WebRtc_Word32 tmp32no1;
- WebRtc_Word32 tmp32no2;
- WebRtc_Word32 echoEst32[PART_LEN1];
-
- WebRtc_UWord16 xfa[PART_LEN1];
- WebRtc_UWord16 dfaNoisy[PART_LEN1];
- WebRtc_UWord16 dfaClean[PART_LEN1];
- WebRtc_UWord16* ptrDfaClean = dfaClean;
-
- int outCFFT;
-
- WebRtc_Word16 fft[PART_LEN4];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- WebRtc_Word16 postFft[PART_LEN4];
-#else
- WebRtc_Word16 postFft[PART_LEN2];
-#endif
- WebRtc_Word16 dfwReal[PART_LEN1];
- WebRtc_Word16 dfwImag[PART_LEN1];
- WebRtc_Word16 xfwReal[PART_LEN1];
- WebRtc_Word16 xfwImag[PART_LEN1];
- WebRtc_Word16 efwReal[PART_LEN1];
- WebRtc_Word16 efwImag[PART_LEN1];
- WebRtc_Word16 hnl[PART_LEN1];
- WebRtc_Word16 numPosCoef;
- WebRtc_Word16 nlpGain;
- WebRtc_Word16 delay, diff, diffMinusOne;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 tmp16no2;
-#ifdef AECM_WITH_ABS_APPROX
- WebRtc_Word16 maxValue;
- WebRtc_Word16 minValue;
-#endif
- WebRtc_Word16 mu;
- WebRtc_Word16 supGain;
- WebRtc_Word16 zeros32, zeros16;
- WebRtc_Word16 zerosDBufNoisy, zerosDBufClean, zerosXBuf;
- WebRtc_Word16 resolutionDiff, qDomainDiff;
-
-#ifdef ARM_WINM_LOG_
- DWORD temp;
- static int flag0 = 0;
- __int64 freq, start, end, diff__;
- unsigned int milliseconds;
-#endif
-
-#ifdef AECM_WITH_ABS_APPROX
- WebRtc_UWord16 alpha, beta;
-#endif
-
- // Determine startup state. There are three states:
- // (0) the first CONV_LEN blocks
- // (1) another CONV_LEN blocks
- // (2) the rest
-
- if (aecm->startupState < 2)
- {
- aecm->startupState = (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
- }
- // END: Determine startup state
-
- // Buffer near and far end signals
- memcpy(aecm->xBuf + PART_LEN, farend, sizeof(WebRtc_Word16) * PART_LEN);
- memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(WebRtc_Word16) * PART_LEN);
- if (nearendClean != NULL)
- {
- memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(WebRtc_Word16) * PART_LEN);
- }
- // TODO(bjornv): Will be removed in final version.
-#ifdef VAD_DATA
- fwrite(aecm->xBuf, sizeof(WebRtc_Word16), PART_LEN, aecm->far_file);
-#endif
-
-#ifdef AECM_DYNAMIC_Q
- tmp16no1 = WebRtcSpl_MaxAbsValueW16(aecm->dBufNoisy, PART_LEN2);
- tmp16no2 = WebRtcSpl_MaxAbsValueW16(aecm->xBuf, PART_LEN2);
- zerosDBufNoisy = WebRtcSpl_NormW16(tmp16no1);
- zerosXBuf = WebRtcSpl_NormW16(tmp16no2);
-#else
- zerosDBufNoisy = 0;
- zerosXBuf = 0;
-#endif
- aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
- aecm->dfaNoisyQDomain = zerosDBufNoisy;
-
- if (nearendClean != NULL)
- {
-#ifdef AECM_DYNAMIC_Q
- tmp16no1 = WebRtcSpl_MaxAbsValueW16(aecm->dBufClean, PART_LEN2);
- zerosDBufClean = WebRtcSpl_NormW16(tmp16no1);
-#else
- zerosDBufClean = 0;
-#endif
- aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
- aecm->dfaCleanQDomain = zerosDBufClean;
- } else
- {
- zerosDBufClean = zerosDBufNoisy;
- aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
- aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // FFT of noisy near end signal
- for (i = 0; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- // Window near end
- fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufNoisy[i]
- << zerosDBufNoisy), kSqrtHanning[i], 14);
- fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (aecm->dBufNoisy[PART_LEN + i] << zerosDBufNoisy),
- kSqrtHanning[PART_LEN - i], 14);
- // Inserting zeros in imaginary parts
- fft[j + 1] = 0;
- fft[PART_LEN2 + j + 1] = 0;
- }
-
- // Fourier transformation of near end signal.
- // The result is scaled with 1/PART_LEN2, that is, the result is in Q(-6) for PART_LEN = 32
-
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
- // The imaginary part has to switch sign
- for(i = 1; i < PART_LEN2-1;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- postFft[i] = -postFft[i];
- i += 2;
- }
-#else
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the first PART_LEN2 samples
- for (i = 0; i < PART_LEN2; i++)
- {
- postFft[i] = fft[i];
- }
- // The imaginary part has to switch sign
- for (i = 1; i < PART_LEN2;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- }
-#endif
-
- // Extract imaginary and real part, calculate the magnitude for all frequency bins
- dfwImag[0] = 0;
- dfwImag[PART_LEN] = 0;
- dfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- dfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
- dfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
- dfaNoisy[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[0]);
- dfaNoisy[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[PART_LEN]);
- dfaNoisySum = (WebRtc_UWord32)(dfaNoisy[0]);
- dfaNoisySum += (WebRtc_UWord32)(dfaNoisy[PART_LEN]);
-
- for (i = 1; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- dfwReal[i] = postFft[j];
- dfwImag[i] = postFft[j + 1];
-
- if (dfwReal[i] == 0 || dfwImag[i] == 0)
- {
- dfaNoisy[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[i] + dfwImag[i]);
- } else
- {
- // Approximation for magnitude of complex fft output
- // magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
- //
- // The parameters alpha and beta are stored in Q15
-
- tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
- tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
- if(tmp16no1 > tmp16no2)
- {
- maxValue = tmp16no1;
- minValue = tmp16no2;
- } else
- {
- maxValue = tmp16no2;
- minValue = tmp16no1;
- }
-
- // Magnitude in Q-6
- if ((maxValue >> 2) > minValue)
- {
- alpha = kAlpha1;
- beta = kBeta1;
- } else if ((maxValue >> 1) > minValue)
- {
- alpha = kAlpha2;
- beta = kBeta2;
- } else
- {
- alpha = kAlpha3;
- beta = kBeta3;
- }
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
- dfaNoisy[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
- tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
- tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
- tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
- dfaNoisy[i] = (WebRtc_UWord16)tmp32no1;
-#endif
- }
- dfaNoisySum += (WebRtc_UWord32)dfaNoisy[i];
- }
- // END: FFT of noisy near end signal
-
- if (nearendClean == NULL)
- {
- ptrDfaClean = dfaNoisy;
- } else
- {
- // FFT of clean near end signal
- for (i = 0; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- // Window near end
- fft[j]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufClean[i] << zerosDBufClean), kSqrtHanning[i], 14);
- fft[PART_LEN2 + j]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufClean[PART_LEN + i] << zerosDBufClean), kSqrtHanning[PART_LEN - i], 14);
- // Inserting zeros in imaginary parts
- fft[j + 1] = 0;
- fft[PART_LEN2 + j + 1] = 0;
- }
-
- // Fourier transformation of near end signal.
- // The result is scaled with 1/PART_LEN2, that is, in Q(-6) for PART_LEN = 32
-
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
- // The imaginary part has to switch sign
- for(i = 1; i < PART_LEN2-1;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- postFft[i] = -postFft[i];
- i += 2;
- }
-#else
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the first PART_LEN2 samples
- for (i = 0; i < PART_LEN2; i++)
- {
- postFft[i] = fft[i];
- }
- // The imaginary part has to switch sign
- for (i = 1; i < PART_LEN2;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- }
-#endif
-
- // Extract imaginary and real part, calculate the magnitude for all frequency bins
- dfwImag[0] = 0;
- dfwImag[PART_LEN] = 0;
- dfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- dfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
- dfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
- dfaClean[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[0]);
- dfaClean[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[PART_LEN]);
-
- for (i = 1; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- dfwReal[i] = postFft[j];
- dfwImag[i] = postFft[j + 1];
-
- if (dfwReal[i] == 0 || dfwImag[i] == 0)
- {
- dfaClean[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[i] + dfwImag[i]);
- } else
- {
- // Approximation for magnitude of complex fft output
- // magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
- //
- // The parameters alpha and beta are stored in Q15
-
- tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
- tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
- if(tmp16no1 > tmp16no2)
- {
- maxValue = tmp16no1;
- minValue = tmp16no2;
- } else
- {
- maxValue = tmp16no2;
- minValue = tmp16no1;
- }
-
- // Magnitude in Q-6
- if ((maxValue >> 2) > minValue)
- {
- alpha = kAlpha1;
- beta = kBeta1;
- } else if ((maxValue >> 1) > minValue)
- {
- alpha = kAlpha2;
- beta = kBeta2;
- } else
- {
- alpha = kAlpha3;
- beta = kBeta3;
- }
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
- dfaClean[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
- tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
- tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
- tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
- dfaClean[i] = (WebRtc_UWord16)tmp32no1;
-#endif
- }
- }
- }
- // END: FFT of clean near end signal
-
- // FFT of far end signal
- for (i = 0; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- // Window farend
- fft[j]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->xBuf[i] << zerosXBuf), kSqrtHanning[i], 14);
- fft[PART_LEN2 + j]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->xBuf[PART_LEN + i] << zerosXBuf), kSqrtHanning[PART_LEN - i], 14);
- // Inserting zeros in imaginary parts
- fft[j + 1] = 0;
- fft[PART_LEN2 + j + 1] = 0;
- }
- // Fourier transformation of far end signal.
- // The result is scaled with 1/PART_LEN2, that is the result is in Q(-6) for PART_LEN = 32
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
- // The imaginary part has to switch sign
- for(i = 1; i < PART_LEN2-1;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- postFft[i] = -postFft[i];
- i += 2;
- }
-#else
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the first PART_LEN2 samples
- for (i = 0; i < PART_LEN2; i++)
- {
- postFft[i] = fft[i];
- }
- // The imaginary part has to switch sign
- for (i = 1; i < PART_LEN2;)
- {
- postFft[i] = -postFft[i];
- i += 2;
- }
-#endif
-
- // Extract imaginary and real part, calculate the magnitude for all frequency bins
- xfwImag[0] = 0;
- xfwImag[PART_LEN] = 0;
- xfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
- xfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
- xfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
- xfa[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[0]);
- xfa[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[PART_LEN]);
- xfaSum = (WebRtc_UWord32)(xfa[0]) + (WebRtc_UWord32)(xfa[PART_LEN]);
-
- for (i = 1; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i,1);
- xfwReal[i] = postFft[j];
- xfwImag[i] = postFft[j + 1];
-
- if (xfwReal[i] == 0 || xfwImag[i] == 0)
- {
- xfa[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[i] + xfwImag[i]);
- } else
- {
- // Approximation for magnitude of complex fft output
- // magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
- //
- // The parameters alpha and beta are stored in Q15
-
- tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
- tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
- if(tmp16no1 > xfwImag[i])
- {
- maxValue = tmp16no1;
- minValue = tmp16no2;
- } else
- {
- maxValue = tmp16no2;
- minValue = tmp16no1;
- }
- // Magnitude in Q-6
- if ((maxValue >> 2) > minValue)
- {
- alpha = kAlpha1;
- beta = kBeta1;
- } else if ((maxValue >> 1) > minValue)
- {
- alpha = kAlpha2;
- beta = kBeta2;
- } else
- {
- alpha = kAlpha3;
- beta = kBeta3;
- }
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
- xfa[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
- tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
- tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
- tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
- xfa[i] = (WebRtc_UWord16)tmp32no1;
-#endif
- }
- xfaSum += (WebRtc_UWord32)xfa[i];
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
- // END: FFT of far end signal
-
- // Get the delay
-
- // Fixed delay estimation
- // input: dfaFIX, xfaFIX in Q-stages
- // output: delay in Q0
- //
- // comment on the fixed point accuracy of estimate_delayFIX
- // -> due to rounding the fixed point variables xfa and dfa contain a lot more zeros
- // than the corresponding floating point variables this results in big differences
- // between the floating point and the fixed point logarithmic spectra for small values
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Save far-end history and estimate delay
- delay = WebRtcAecm_EstimateDelay(aecm, xfa, dfaNoisy, zerosXBuf);
-
- if (aecm->fixedDelay >= 0)
- {
- // Use fixed delay
- delay = aecm->fixedDelay;
- }
-
- aecm->currentDelay = delay;
-
- if ((aecm->delayOffsetFlag) & (aecm->startupState > 0)) // If delay compensation is on
- {
- // If the delay estimate changed from previous block, update the offset
- if ((aecm->currentDelay != aecm->previousDelay) & !aecm->currentDelay
- & !aecm->previousDelay)
- {
- aecm->delayAdjust += (aecm->currentDelay - aecm->previousDelay);
- }
- // Compensate with the offset estimate
- aecm->currentDelay -= aecm->delayAdjust;
- aecm->previousDelay = delay;
- }
-
- diff = aecm->delHistoryPos - aecm->currentDelay;
- if (diff < 0)
- {
- diff = diff + MAX_DELAY;
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
- // END: Get the delay
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
- // Calculate log(energy) and update energy threshold levels
- WebRtcAecm_CalcEnergies(aecm, diff, dfaNoisySum, echoEst32);
-
- // Calculate stepsize
- mu = WebRtcAecm_CalcStepSize(aecm);
-
- // Update counters
- aecm->totCount++;
- aecm->lastDelayUpdateCount++;
-
- // This is the channel estimation algorithm.
- // It is base on NLMS but has a variable step length, which was calculated above.
- WebRtcAecm_UpdateChannel(aecm, dfaNoisy, diff, mu, echoEst32);
- WebRtcAecm_DelayCompensation(aecm);
- supGain = WebRtcAecm_CalcSuppressionGain(aecm);
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Calculate Wiener filter hnl[]
- numPosCoef = 0;
- diffMinusOne = diff - 1;
- if (diff == 0)
- {
- diffMinusOne = MAX_DELAY;
- }
- for (i = 0; i < PART_LEN1; i++)
- {
- // Far end signal through channel estimate in Q8
- // How much can we shift right to preserve resolution
- tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
- aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
-
- zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
- zeros16 = WebRtcSpl_NormW16(supGain) + 1;
- if (zeros32 + zeros16 > 16)
- {
- // Multiplication is safe
- // Result in Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
- (WebRtc_UWord16)supGain);
- resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
- resolutionDiff += (aecm->dfaCleanQDomain - aecm->xfaQDomainBuf[diff]);
- } else
- {
- tmp16no1 = 17 - zeros32 - zeros16;
- resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
- resolutionDiff += (aecm->dfaCleanQDomain - aecm->xfaQDomainBuf[diff]);
- if (zeros32 > tmp16no1)
- {
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
- (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W16(supGain,
- tmp16no1)); // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
- } else
- {
- // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
- (WebRtc_UWord32)WEBRTC_SPL_RSHIFT_W32(aecm->echoFilt[i], tmp16no1),
- (WebRtc_UWord16)supGain);
- }
- }
-
- zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
- if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld))
- & (aecm->nearFilt[i]))
- {
- tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16);
- qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld;
- } else
- {
- tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], aecm->dfaCleanQDomain
- - aecm->dfaCleanQDomainOld);
- qDomainDiff = 0;
- }
- tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no2 - tmp16no1, 1, 4);
- tmp16no2 += tmp16no1;
- zeros16 = WebRtcSpl_NormW16(tmp16no2);
- if ((tmp16no2) & (-qDomainDiff > zeros16))
- {
- aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
- } else
- {
- aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff);
- }
-
- // Wiener filter coefficients, resulting hnl in Q14
- if (echoEst32Gained == 0)
- {
- hnl[i] = ONE_Q14;
- } else if (aecm->nearFilt[i] == 0)
- {
- hnl[i] = 0;
- } else
- {
- // Multiply the suppression gain
- // Rounding
- echoEst32Gained += (WebRtc_UWord32)(aecm->nearFilt[i] >> 1);
- tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained, (WebRtc_UWord16)aecm->nearFilt[i]);
-
- // Current resolution is
- // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN - max(0, 17 - zeros16 - zeros32))
- // Make sure we are in Q14
- tmp32no1 = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
- if (tmp32no1 > ONE_Q14)
- {
- hnl[i] = 0;
- } else if (tmp32no1 < 0)
- {
- hnl[i] = ONE_Q14;
- } else
- {
- // 1-echoEst/dfa
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- hnl[i] = ONE_Q14 - (WebRtc_Word16)tmp32no1;
- if (hnl[i] < 0)
- {
- hnl[i] = 0;
- }
-#else
- hnl[i] = ((ONE_Q14 - (WebRtc_Word16)tmp32no1) > 0) ? (ONE_Q14 - (WebRtc_Word16)tmp32no1) : 0;
-#endif
- }
- }
- if (hnl[i])
- {
- numPosCoef++;
- }
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Calculate NLP gain, result is in Q14
- for (i = 0; i < PART_LEN1; i++)
- {
- if (aecm->nlpFlag)
- {
- // Truncate values close to zero and one.
- if (hnl[i] > NLP_COMP_HIGH)
- {
- hnl[i] = ONE_Q14;
- } else if (hnl[i] < NLP_COMP_LOW)
- {
- hnl[i] = 0;
- }
-
- // Remove outliers
- if (numPosCoef < 3)
- {
- nlpGain = 0;
- } else
- {
- nlpGain = ONE_Q14;
- }
- // NLP
- if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
- {
- hnl[i] = ONE_Q14;
- } else
- {
- hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
- }
- }
-
- // multiply with Wiener coefficients
- efwReal[i] = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfwReal[i], hnl[i],
- 14));
- efwImag[i] = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfwImag[i], hnl[i],
- 14));
- }
-
- if (aecm->cngMode == AecmTrue)
- {
- WebRtcAecm_ComfortNoise(aecm, ptrDfaClean, efwReal, efwImag, hnl);
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Synthesis
- for (i = 1; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- fft[j] = efwReal[i];
-
- // mirrored data, even
- fft[PART_LEN4 - j] = efwReal[i];
- fft[j + 1] = -efwImag[i];
-
- //mirrored data, odd
- fft[PART_LEN4 - (j - 1)] = efwImag[i];
- }
- fft[0] = efwReal[0];
- fft[1] = -efwImag[0];
-
- fft[PART_LEN2] = efwReal[PART_LEN];
- fft[PART_LEN2 + 1] = -efwImag[PART_LEN];
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- // inverse FFT, result should be scaled with outCFFT
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
-
- //take only the real values and scale with outCFFT
- for (i = 0; i < PART_LEN2; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- fft[i] = fft[j];
- }
-#else
- outCFFT = WebRtcSpl_ComplexIFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
- //take only the real values and scale with outCFFT
- for(i = 0, j = 0; i < PART_LEN2;)
- {
- fft[i] = postFft[j];
- i += 1;
- j += 2;
- fft[i] = postFft[j];
- i += 1;
- j += 2;
- }
-#endif
-
- for (i = 0; i < PART_LEN; i++)
- {
- fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- fft[i],
- kSqrtHanning[i],
- 14);
- tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
- outCFFT - aecm->dfaCleanQDomain);
- fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
- tmp32no1 + aecm->outBuf[i],
- WEBRTC_SPL_WORD16_MIN);
- output[i] = fft[i];
-
- tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
- fft[PART_LEN + i],
- kSqrtHanning[PART_LEN - i],
- 14);
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
- outCFFT - aecm->dfaCleanQDomain);
- aecm->outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
- WEBRTC_SPL_WORD16_MAX,
- tmp32no1,
- WEBRTC_SPL_WORD16_MIN);
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
- // Copy the current block to the old position (outBuf is shifted elsewhere)
- memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- if (nearendClean != NULL)
- {
- memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- }
-}
-
-// Generate comfort noise and add to output signal.
-//
-// \param[in] aecm Handle of the AECM instance.
-// \param[in] dfa Absolute value of the nearend signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outReal Real part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outImag Imaginary part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in] lambda Suppression gain with which to scale the noise level (Q14).
-//
-static void WebRtcAecm_ComfortNoise(AecmCore_t * const aecm, const WebRtc_UWord16 * const dfa,
- WebRtc_Word16 * const outReal,
- WebRtc_Word16 * const outImag,
- const WebRtc_Word16 * const lambda)
-{
- WebRtc_Word16 i;
- WebRtc_Word16 tmp16;
- WebRtc_Word32 tmp32;
-
- WebRtc_Word16 randW16[PART_LEN];
- WebRtc_Word16 uReal[PART_LEN1];
- WebRtc_Word16 uImag[PART_LEN1];
- WebRtc_Word32 outLShift32[PART_LEN1];
- WebRtc_Word16 noiseRShift16[PART_LEN1];
-
- WebRtc_Word16 shiftFromNearToNoise[PART_LEN1];
- WebRtc_Word16 minTrackShift;
- WebRtc_Word32 upper32;
- WebRtc_Word32 lower32;
-
- if (aecm->noiseEstCtr < 100)
- {
- // Track the minimum more quickly initially.
- aecm->noiseEstCtr++;
- minTrackShift = 7;
- } else
- {
- minTrackShift = 9;
- }
-
- // Estimate noise power.
- for (i = 0; i < PART_LEN1; i++)
- {
- shiftFromNearToNoise[i] = aecm->noiseEstQDomain[i] - aecm->dfaCleanQDomain;
-
- // Shift to the noise domain.
- tmp32 = (WebRtc_Word32)dfa[i];
- outLShift32[i] = WEBRTC_SPL_SHIFT_W32(tmp32, shiftFromNearToNoise[i]);
-
- if (outLShift32[i] < aecm->noiseEst[i])
- {
- // Track the minimum.
- aecm->noiseEst[i] += ((outLShift32[i] - aecm->noiseEst[i]) >> minTrackShift);
- } else
- {
- // Ramp slowly upwards until we hit the minimum again.
-
- // Avoid overflow.
- if (aecm->noiseEst[i] < 2146435583)
- {
- // Store the fractional portion.
- upper32 = (aecm->noiseEst[i] & 0xffff0000) >> 16;
- lower32 = aecm->noiseEst[i] & 0x0000ffff;
- upper32 = ((upper32 * 2049) >> 11);
- lower32 = ((lower32 * 2049) >> 11);
- aecm->noiseEst[i] = WEBRTC_SPL_ADD_SAT_W32(upper32 << 16, lower32);
- }
- }
- }
-
- for (i = 0; i < PART_LEN1; i++)
- {
- tmp32 = WEBRTC_SPL_SHIFT_W32(aecm->noiseEst[i], -shiftFromNearToNoise[i]);
- if (tmp32 > 32767)
- {
- tmp32 = 32767;
- aecm->noiseEst[i] = WEBRTC_SPL_SHIFT_W32(tmp32, shiftFromNearToNoise[i]);
- }
- noiseRShift16[i] = (WebRtc_Word16)tmp32;
-
- tmp16 = ONE_Q14 - lambda[i];
- noiseRShift16[i]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, noiseRShift16[i], 14);
- }
-
- // Generate a uniform random array on [0 2^15-1].
- WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
-
- // Generate noise according to estimated energy.
- uReal[0] = 0; // Reject LF noise.
- uImag[0] = 0;
- for (i = 1; i < PART_LEN1; i++)
- {
- // Get a random index for the cos and sin tables over [0 359].
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
-
- // Tables are in Q13.
- uReal[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i],
- WebRtcSpl_kCosTable[tmp16], 13);
- uImag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i],
- WebRtcSpl_kSinTable[tmp16], 13);
- }
- uImag[PART_LEN] = 0;
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
- outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
- }
-#else
- for (i = 0; i < PART_LEN1 -1; )
- {
- outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
- outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
- i++;
-
- outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
- outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
- i++;
- }
- outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
- outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
-#endif
-}
-
-void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const int farLen)
-{
- int writeLen = farLen, writePos = 0;
-
- // Check if the write position must be wrapped
- while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
- {
- // Write to remaining buffer space before wrapping
- writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
- memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
- sizeof(WebRtc_Word16) * writeLen);
- aecm->farBufWritePos = 0;
- writePos = writeLen;
- writeLen = farLen - writeLen;
- }
-
- memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
- sizeof(WebRtc_Word16) * writeLen);
- aecm->farBufWritePos += writeLen;
-}
-
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
- const int farLen, const int knownDelay)
-{
- int readLen = farLen;
- int readPos = 0;
- int delayChange = knownDelay - aecm->lastKnownDelay;
-
- aecm->farBufReadPos -= delayChange;
-
- // Check if delay forces a read position wrap
- while (aecm->farBufReadPos < 0)
- {
- aecm->farBufReadPos += FAR_BUF_LEN;
- }
- while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
- {
- aecm->farBufReadPos -= FAR_BUF_LEN;
- }
-
- aecm->lastKnownDelay = knownDelay;
-
- // Check if read position must be wrapped
- while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
- {
-
- // Read from remaining buffer space before wrapping
- readLen = FAR_BUF_LEN - aecm->farBufReadPos;
- memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
- sizeof(WebRtc_Word16) * readLen);
- aecm->farBufReadPos = 0;
- readPos = readLen;
- readLen = farLen - readLen;
- }
- memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
- sizeof(WebRtc_Word16) * readLen);
- aecm->farBufReadPos += readLen;
-}
diff --git a/src/modules/audio_processing/agc/Android.mk b/src/modules/audio_processing/agc/Android.mk
new file mode 100644
index 0000000..546128d
--- /dev/null
+++ b/src/modules/audio_processing/agc/Android.mk
@@ -0,0 +1,40 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_agc
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+ analog_agc.c \
+ digital_agc.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/agc/main/source/agc.gyp b/src/modules/audio_processing/agc/agc.gypi
similarity index 66%
rename from src/modules/audio_processing/agc/main/source/agc.gyp
rename to src/modules/audio_processing/agc/agc.gypi
index e28a4c8..78288b7 100644
--- a/src/modules/audio_processing/agc/main/source/agc.gyp
+++ b/src/modules/audio_processing/agc/agc.gypi
@@ -7,26 +7,23 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../../common_settings.gypi', # Common settings
- ],
'targets': [
{
'target_name': 'agc',
'type': '<(library)',
'dependencies': [
- '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
],
'include_dirs': [
- '../interface',
+ 'interface',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'interface',
],
},
'sources': [
- '../interface/gain_control.h',
+ 'interface/gain_control.h',
'analog_agc.c',
'analog_agc.h',
'digital_agc.c',
@@ -35,9 +32,3 @@
},
],
}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/agc/main/source/analog_agc.c b/src/modules/audio_processing/agc/analog_agc.c
similarity index 98%
rename from src/modules/audio_processing/agc/main/source/analog_agc.c
rename to src/modules/audio_processing/agc/analog_agc.c
index e52fd66..40c5566 100644
--- a/src/modules/audio_processing/agc/main/source/analog_agc.c
+++ b/src/modules/audio_processing/agc/analog_agc.c
@@ -200,6 +200,10 @@
/* apply slowly varying digital gain */
if (stt->micVol > stt->maxAnalog)
{
+ /* |maxLevel| is strictly >= |micVol|, so this condition should be
+ * satisfied here, ensuring there is no divide-by-zero. */
+ assert(stt->maxLevel > stt->maxAnalog);
+
/* Q1 */
tmp16 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog);
tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
@@ -921,7 +925,8 @@
stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
/* Circular buffer */
- if (++(stt->Rxx16pos) == RXX_BUFFER_LEN)
+ stt->Rxx16pos++;
+ if (stt->Rxx16pos == RXX_BUFFER_LEN)
{
stt->Rxx16pos = 0;
}
@@ -1316,10 +1321,19 @@
*outMicLevel = inMicLevel;
inMicLevelTmp = inMicLevel;
- memcpy(out, in_near, samples * sizeof(WebRtc_Word16));
+ // TODO(andrew): clearly we don't need input and output pointers...
+ // Change the interface to take a shared input/output.
+ if (in_near != out)
+ {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, in_near, samples * sizeof(WebRtc_Word16));
+ }
if (stt->fs == 32000)
{
- memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16));
+ if (in_near_H != out_H)
+ {
+ memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16));
+ }
}
#ifdef AGC_DEBUG//test log
diff --git a/src/modules/audio_processing/agc/main/source/analog_agc.h b/src/modules/audio_processing/agc/analog_agc.h
similarity index 100%
rename from src/modules/audio_processing/agc/main/source/analog_agc.h
rename to src/modules/audio_processing/agc/analog_agc.h
diff --git a/src/modules/audio_processing/agc/main/source/digital_agc.c b/src/modules/audio_processing/agc/digital_agc.c
similarity index 96%
rename from src/modules/audio_processing/agc/main/source/digital_agc.c
rename to src/modules/audio_processing/agc/digital_agc.c
index 2966586..3b4b39b 100644
--- a/src/modules/audio_processing/agc/main/source/digital_agc.c
+++ b/src/modules/audio_processing/agc/digital_agc.c
@@ -12,11 +12,14 @@
*
*/
+#include "digital_agc.h"
+
+#include <assert.h>
#include <string.h>
#ifdef AGC_DEBUG
#include <stdio.h>
#endif
-#include "digital_agc.h"
+
#include "gain_control.h"
// To generate the gaintable, copy&paste the following lines to a Matlab window:
@@ -33,7 +36,8 @@
// zoom on;
// Generator table for y=log2(1+e^x) in Q8.
-static const WebRtc_UWord16 kGenFuncTable[128] = {
+enum { kGenFuncTableSize = 128 };
+static const WebRtc_UWord16 kGenFuncTable[kGenFuncTableSize] = {
256, 485, 786, 1126, 1484, 1849, 2217, 2586,
2955, 3324, 3693, 4063, 4432, 4801, 5171, 5540,
5909, 6279, 6648, 7017, 7387, 7756, 8125, 8495,
@@ -102,8 +106,9 @@
// = (compRatio-1)*digCompGaindB/compRatio
tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
- if (diffGain < 0)
+ if (diffGain < 0 || diffGain >= kGenFuncTableSize)
{
+ assert(0);
return -1;
}
@@ -185,8 +190,15 @@
numFIX -= WEBRTC_SPL_MUL_32_16((WebRtc_Word32)logApprox, diffGain); // Q14
// Calculate ratio
- // Shift numFIX as much as possible
- zeros = WebRtcSpl_NormW32(numFIX);
+ // Shift |numFIX| as much as possible.
+ // Ensure we avoid wrap-around in |den| as well.
+ if (numFIX > (den >> 8)) // |den| is Q8.
+ {
+ zeros = WebRtcSpl_NormW32(numFIX);
+ } else
+ {
+ zeros = WebRtcSpl_NormW32(den) + 8;
+ }
numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
// Shift den so we end up in Qy1
@@ -327,10 +339,18 @@
return -1;
}
- memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
+ // TODO(andrew): again, we don't need input and output pointers...
+ if (in_near != out)
+ {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
+ }
if (FS == 32000)
{
- memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
+ if (in_near_H != out_H)
+ {
+ memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
+ }
}
// VAD for near end
logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
@@ -652,11 +672,9 @@
WebRtc_Word16 buf2[4];
WebRtc_Word16 HPstate;
WebRtc_Word16 zeros, dB;
- WebRtc_Word16 *buf1_ptr;
// process in 10 sub frames of 1 ms (to save on memory)
nrg = 0;
- buf1_ptr = &buf1[0];
HPstate = state->HPstate;
for (subfr = 0; subfr < 10; subfr++)
{
diff --git a/src/modules/audio_processing/agc/main/source/digital_agc.h b/src/modules/audio_processing/agc/digital_agc.h
similarity index 100%
rename from src/modules/audio_processing/agc/main/source/digital_agc.h
rename to src/modules/audio_processing/agc/digital_agc.h
diff --git a/src/modules/audio_processing/agc/main/interface/gain_control.h b/src/modules/audio_processing/agc/interface/gain_control.h
similarity index 100%
rename from src/modules/audio_processing/agc/main/interface/gain_control.h
rename to src/modules/audio_processing/agc/interface/gain_control.h
diff --git a/src/modules/audio_processing/agc/main/matlab/getGains.m b/src/modules/audio_processing/agc/main/matlab/getGains.m
deleted file mode 100644
index e0234b8..0000000
--- a/src/modules/audio_processing/agc/main/matlab/getGains.m
+++ /dev/null
@@ -1,32 +0,0 @@
-% Outputs a file for testing purposes.
-%
-% Adjust the following parameters to suit. Their purpose becomes more clear on
-% viewing the gain plots.
-% MaxGain: Max gain in dB
-% MinGain: Min gain at overload (0 dBov) in dB
-% CompRatio: Compression ratio, essentially determines the slope of the gain
-% function between the max and min gains
-% Knee: The smoothness of the transition to max gain (smaller is smoother)
-MaxGain = 5; MinGain = 0; CompRatio = 3; Knee = 1;
-
-% Compute gains
-zeros = 0:31; lvl = 2.^(1-zeros);
-A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
-B = MaxGain - MinGain;
-gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
-fprintf(1, '\t%i, %i, %i, %i,\n', gains);
-
-% Save gains to file
-fid = fopen('gains', 'wb');
-if fid == -1
- error(sprintf('Unable to open file %s', filename));
- return
-end
-fwrite(fid, gains, 'int32');
-fclose(fid);
-
-% Plotting
-in = 10*log10(lvl); out = 20*log10(gains/65536);
-subplot(121); plot(in, out); axis([-60, 0, -5, 30]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
-subplot(122); plot(in, in+out); axis([-60, 0, -60, 10]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
-zoom on;
diff --git a/src/modules/audio_processing/agc/main/source/Android.mk b/src/modules/audio_processing/agc/main/source/Android.mk
deleted file mode 100644
index e045839..0000000
--- a/src/modules/audio_processing/agc/main/source/Android.mk
+++ /dev/null
@@ -1,49 +0,0 @@
-# This file is generated by gyp; do not edit. This means you!
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_agc
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := analog_agc.c \
- digital_agc.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-# Duplicate the static libraries to fix circular references
-LOCAL_STATIC_LIBRARIES += $(LOCAL_STATIC_LIBRARIES)
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/apm_tests.gypi b/src/modules/audio_processing/apm_tests.gypi
new file mode 100644
index 0000000..f9b21d2
--- /dev/null
+++ b/src/modules/audio_processing/apm_tests.gypi
@@ -0,0 +1,75 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audioproc_unittest',
+ 'type': 'executable',
+ 'conditions': [
+ ['prefer_fixed_point==1', {
+ 'defines': [ 'WEBRTC_APM_UNIT_TEST_FIXED_PROFILE' ],
+ }, {
+ 'defines': [ 'WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE' ],
+ }],
+ ['enable_protobuf==1', {
+ 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
+ }],
+ ],
+ 'dependencies': [
+ 'audio_processing',
+ 'audioproc_unittest_proto',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../test/test.gyp:test_support',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [ 'test/unit_test.cc', ],
+ },
+ {
+ 'target_name': 'audioproc_unittest_proto',
+ 'type': 'static_library',
+ 'sources': [ 'test/unittest.proto', ],
+ 'variables': {
+ 'proto_in_dir': 'test',
+ # Workaround to protect against gyp's pathname relativization when this
+ # file is included by modules.gyp.
+ 'proto_out_protected': 'webrtc/audio_processing',
+ 'proto_out_dir': '<(proto_out_protected)',
+ },
+ 'includes': [ '../../build/protoc.gypi', ],
+ },
+ ],
+ 'conditions': [
+ ['enable_protobuf==1', {
+ 'targets': [
+ {
+ 'target_name': 'audioproc',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_processing',
+ 'audioproc_debug_proto',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [ 'test/process_test.cc', ],
+ },
+ {
+ 'target_name': 'unpack_aecdump',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audioproc_debug_proto',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
+ ],
+ 'sources': [ 'test/unpack.cc', ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/src/modules/audio_processing/audio_buffer.cc b/src/modules/audio_processing/audio_buffer.cc
new file mode 100644
index 0000000..a7fb04d
--- /dev/null
+++ b/src/modules/audio_processing/audio_buffer.cc
@@ -0,0 +1,306 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_buffer.h"
+
+#include "signal_processing_library.h"
+
+namespace webrtc {
+namespace {
+
+enum {
+ kSamplesPer8kHzChannel = 80,
+ kSamplesPer16kHzChannel = 160,
+ kSamplesPer32kHzChannel = 320
+};
+
+void StereoToMono(const int16_t* left, const int16_t* right,
+ int16_t* out, int samples_per_channel) {
+ assert(left != NULL && right != NULL && out != NULL);
+ for (int i = 0; i < samples_per_channel; i++) {
+ int32_t data32 = (static_cast<int32_t>(left[i]) +
+ static_cast<int32_t>(right[i])) >> 1;
+
+ out[i] = WebRtcSpl_SatW32ToW16(data32);
+ }
+}
+} // namespace
+
+struct AudioChannel {
+ AudioChannel() {
+ memset(data, 0, sizeof(data));
+ }
+
+ int16_t data[kSamplesPer32kHzChannel];
+};
+
+struct SplitAudioChannel {
+ SplitAudioChannel() {
+ memset(low_pass_data, 0, sizeof(low_pass_data));
+ memset(high_pass_data, 0, sizeof(high_pass_data));
+ memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
+ memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
+ memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
+ memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
+ }
+
+ int16_t low_pass_data[kSamplesPer16kHzChannel];
+ int16_t high_pass_data[kSamplesPer16kHzChannel];
+
+ WebRtc_Word32 analysis_filter_state1[6];
+ WebRtc_Word32 analysis_filter_state2[6];
+ WebRtc_Word32 synthesis_filter_state1[6];
+ WebRtc_Word32 synthesis_filter_state2[6];
+};
+
+// TODO(andrew): check range of input parameters?
+AudioBuffer::AudioBuffer(int max_num_channels,
+ int samples_per_channel)
+ : max_num_channels_(max_num_channels),
+ num_channels_(0),
+ num_mixed_channels_(0),
+ num_mixed_low_pass_channels_(0),
+ data_was_mixed_(false),
+ samples_per_channel_(samples_per_channel),
+ samples_per_split_channel_(samples_per_channel),
+ reference_copied_(false),
+ activity_(AudioFrame::kVadUnknown),
+ is_muted_(false),
+ data_(NULL),
+ channels_(NULL),
+ split_channels_(NULL),
+ mixed_channels_(NULL),
+ mixed_low_pass_channels_(NULL),
+ low_pass_reference_channels_(NULL) {
+ if (max_num_channels_ > 1) {
+ channels_.reset(new AudioChannel[max_num_channels_]);
+ mixed_channels_.reset(new AudioChannel[max_num_channels_]);
+ mixed_low_pass_channels_.reset(new AudioChannel[max_num_channels_]);
+ }
+ low_pass_reference_channels_.reset(new AudioChannel[max_num_channels_]);
+
+ if (samples_per_channel_ == kSamplesPer32kHzChannel) {
+ split_channels_.reset(new SplitAudioChannel[max_num_channels_]);
+ samples_per_split_channel_ = kSamplesPer16kHzChannel;
+ }
+}
+
+AudioBuffer::~AudioBuffer() {}
+
+int16_t* AudioBuffer::data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (data_ != NULL) {
+ return data_;
+ }
+
+ return channels_[channel].data;
+}
+
+int16_t* AudioBuffer::low_pass_split_data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (split_channels_.get() == NULL) {
+ return data(channel);
+ }
+
+ return split_channels_[channel].low_pass_data;
+}
+
+int16_t* AudioBuffer::high_pass_split_data(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (split_channels_.get() == NULL) {
+ return NULL;
+ }
+
+ return split_channels_[channel].high_pass_data;
+}
+
+int16_t* AudioBuffer::mixed_data(int channel) const {
+ assert(channel >= 0 && channel < num_mixed_channels_);
+
+ return mixed_channels_[channel].data;
+}
+
+int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
+ assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
+
+ return mixed_low_pass_channels_[channel].data;
+}
+
+int16_t* AudioBuffer::low_pass_reference(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ if (!reference_copied_) {
+ return NULL;
+ }
+
+ return low_pass_reference_channels_[channel].data;
+}
+
+WebRtc_Word32* AudioBuffer::analysis_filter_state1(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].analysis_filter_state1;
+}
+
+WebRtc_Word32* AudioBuffer::analysis_filter_state2(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].analysis_filter_state2;
+}
+
+WebRtc_Word32* AudioBuffer::synthesis_filter_state1(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].synthesis_filter_state1;
+}
+
+WebRtc_Word32* AudioBuffer::synthesis_filter_state2(int channel) const {
+ assert(channel >= 0 && channel < num_channels_);
+ return split_channels_[channel].synthesis_filter_state2;
+}
+
+void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
+ activity_ = activity;
+}
+
+AudioFrame::VADActivity AudioBuffer::activity() const {
+ return activity_;
+}
+
+bool AudioBuffer::is_muted() const {
+ return is_muted_;
+}
+
+int AudioBuffer::num_channels() const {
+ return num_channels_;
+}
+
+int AudioBuffer::samples_per_channel() const {
+ return samples_per_channel_;
+}
+
+int AudioBuffer::samples_per_split_channel() const {
+ return samples_per_split_channel_;
+}
+
+// TODO(andrew): Do deinterleaving and mixing in one step?
+void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
+ assert(frame->_audioChannel <= max_num_channels_);
+ assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
+
+ num_channels_ = frame->_audioChannel;
+ data_was_mixed_ = false;
+ num_mixed_channels_ = 0;
+ num_mixed_low_pass_channels_ = 0;
+ reference_copied_ = false;
+ activity_ = frame->_vadActivity;
+ is_muted_ = false;
+ if (frame->_energy == 0) {
+ is_muted_ = true;
+ }
+
+ if (num_channels_ == 1) {
+ // We can get away with a pointer assignment in this case.
+ data_ = frame->_payloadData;
+ return;
+ }
+
+ int16_t* interleaved = frame->_payloadData;
+ for (int i = 0; i < num_channels_; i++) {
+ int16_t* deinterleaved = channels_[i].data;
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel_; j++) {
+ deinterleaved[j] = interleaved[interleaved_idx];
+ interleaved_idx += num_channels_;
+ }
+ }
+}
+
+void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
+ assert(frame->_audioChannel == num_channels_);
+ assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
+ frame->_vadActivity = activity_;
+
+ if (!data_changed) {
+ return;
+ }
+
+ if (num_channels_ == 1) {
+ if (data_was_mixed_) {
+ memcpy(frame->_payloadData,
+ channels_[0].data,
+ sizeof(int16_t) * samples_per_channel_);
+ } else {
+ // These should point to the same buffer in this case.
+ assert(data_ == frame->_payloadData);
+ }
+
+ return;
+ }
+
+ int16_t* interleaved = frame->_payloadData;
+ for (int i = 0; i < num_channels_; i++) {
+ int16_t* deinterleaved = channels_[i].data;
+ int interleaved_idx = i;
+ for (int j = 0; j < samples_per_channel_; j++) {
+ interleaved[interleaved_idx] = deinterleaved[j];
+ interleaved_idx += num_channels_;
+ }
+ }
+}
+
+// TODO(andrew): would be good to support the no-mix case with pointer
+// assignment.
+// TODO(andrew): handle mixing to multiple channels?
+void AudioBuffer::Mix(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(channels_[0].data,
+ channels_[1].data,
+ channels_[0].data,
+ samples_per_channel_);
+
+ num_channels_ = num_mixed_channels;
+ data_was_mixed_ = true;
+}
+
+void AudioBuffer::CopyAndMix(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(channels_[0].data,
+ channels_[1].data,
+ mixed_channels_[0].data,
+ samples_per_channel_);
+
+ num_mixed_channels_ = num_mixed_channels;
+}
+
+void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
+ // We currently only support the stereo to mono case.
+ assert(num_channels_ == 2);
+ assert(num_mixed_channels == 1);
+
+ StereoToMono(low_pass_split_data(0),
+ low_pass_split_data(1),
+ mixed_low_pass_channels_[0].data,
+ samples_per_split_channel_);
+
+ num_mixed_low_pass_channels_ = num_mixed_channels;
+}
+
+void AudioBuffer::CopyLowPassToReference() {
+ reference_copied_ = true;
+ for (int i = 0; i < num_channels_; i++) {
+ memcpy(low_pass_reference_channels_[i].data,
+ low_pass_split_data(i),
+ sizeof(int16_t) * samples_per_split_channel_);
+ }
+}
+} // namespace webrtc
diff --git a/src/modules/audio_processing/audio_buffer.h b/src/modules/audio_processing/audio_buffer.h
new file mode 100644
index 0000000..87d6972
--- /dev/null
+++ b/src/modules/audio_processing/audio_buffer.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
+
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+struct AudioChannel;
+struct SplitAudioChannel;
+
+class AudioBuffer {
+ public:
+ AudioBuffer(int max_num_channels, int samples_per_channel);
+ virtual ~AudioBuffer();
+
+ int num_channels() const;
+ int samples_per_channel() const;
+ int samples_per_split_channel() const;
+
+ int16_t* data(int channel) const;
+ int16_t* low_pass_split_data(int channel) const;
+ int16_t* high_pass_split_data(int channel) const;
+ int16_t* mixed_data(int channel) const;
+ int16_t* mixed_low_pass_data(int channel) const;
+ int16_t* low_pass_reference(int channel) const;
+
+ int32_t* analysis_filter_state1(int channel) const;
+ int32_t* analysis_filter_state2(int channel) const;
+ int32_t* synthesis_filter_state1(int channel) const;
+ int32_t* synthesis_filter_state2(int channel) const;
+
+ void set_activity(AudioFrame::VADActivity activity);
+ AudioFrame::VADActivity activity() const;
+
+ bool is_muted() const;
+
+ void DeinterleaveFrom(AudioFrame* audioFrame);
+ void InterleaveTo(AudioFrame* audioFrame) const;
+ // If |data_changed| is false, only the non-audio data members will be copied
+ // to |frame|.
+ void InterleaveTo(AudioFrame* frame, bool data_changed) const;
+ void Mix(int num_mixed_channels);
+ void CopyAndMix(int num_mixed_channels);
+ void CopyAndMixLowPass(int num_mixed_channels);
+ void CopyLowPassToReference();
+
+ private:
+ const int max_num_channels_;
+ int num_channels_;
+ int num_mixed_channels_;
+ int num_mixed_low_pass_channels_;
+ // Whether the original data was replaced with mixed data.
+ bool data_was_mixed_;
+ const int samples_per_channel_;
+ int samples_per_split_channel_;
+ bool reference_copied_;
+ AudioFrame::VADActivity activity_;
+ bool is_muted_;
+
+ int16_t* data_;
+ scoped_array<AudioChannel> channels_;
+ scoped_array<SplitAudioChannel> split_channels_;
+ scoped_array<AudioChannel> mixed_channels_;
+ // TODO(andrew): improve this, we don't need the full 32 kHz space here.
+ scoped_array<AudioChannel> mixed_low_pass_channels_;
+ scoped_array<AudioChannel> low_pass_reference_channels_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
diff --git a/src/modules/audio_processing/audio_processing.gypi b/src/modules/audio_processing/audio_processing.gypi
new file mode 100644
index 0000000..2a22a79
--- /dev/null
+++ b/src/modules/audio_processing/audio_processing.gypi
@@ -0,0 +1,91 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'audio_processing',
+ 'type': '<(library)',
+ 'conditions': [
+ ['prefer_fixed_point==1', {
+ 'dependencies': [ 'ns_fix' ],
+ 'defines': [ 'WEBRTC_NS_FIXED' ],
+ }, {
+ 'dependencies': [ 'ns' ],
+ 'defines': [ 'WEBRTC_NS_FLOAT' ],
+ }],
+ ['enable_protobuf==1', {
+ 'dependencies': [ 'audioproc_debug_proto' ],
+ 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
+ }],
+ ],
+ 'dependencies': [
+ 'aec',
+ 'aecm',
+ 'agc',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/common_audio/common_audio.gyp:vad',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'include_dirs': [
+ 'interface',
+ '../interface',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'interface',
+ '../interface',
+ ],
+ },
+ 'sources': [
+ 'interface/audio_processing.h',
+ 'audio_buffer.cc',
+ 'audio_buffer.h',
+ 'audio_processing_impl.cc',
+ 'audio_processing_impl.h',
+ 'echo_cancellation_impl.cc',
+ 'echo_cancellation_impl.h',
+ 'echo_control_mobile_impl.cc',
+ 'echo_control_mobile_impl.h',
+ 'gain_control_impl.cc',
+ 'gain_control_impl.h',
+ 'high_pass_filter_impl.cc',
+ 'high_pass_filter_impl.h',
+ 'level_estimator_impl.cc',
+ 'level_estimator_impl.h',
+ 'noise_suppression_impl.cc',
+ 'noise_suppression_impl.h',
+ 'splitting_filter.cc',
+ 'splitting_filter.h',
+ 'processing_component.cc',
+ 'processing_component.h',
+ 'voice_detection_impl.cc',
+ 'voice_detection_impl.h',
+ ],
+ },
+ ],
+ 'conditions': [
+ ['enable_protobuf==1', {
+ 'targets': [
+ {
+ 'target_name': 'audioproc_debug_proto',
+ 'type': 'static_library',
+ 'sources': [ 'debug.proto', ],
+ 'variables': {
+ 'proto_in_dir': '.',
+ # Workaround to protect against gyp's pathname relativization when
+ # this file is included by modules.gyp.
+ 'proto_out_protected': 'webrtc/audio_processing',
+ 'proto_out_dir': '<(proto_out_protected)',
+ },
+ 'includes': [ '../../build/protoc.gypi', ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.cc b/src/modules/audio_processing/audio_processing_impl.cc
similarity index 66%
rename from src/modules/audio_processing/main/source/audio_processing_impl.cc
rename to src/modules/audio_processing/audio_processing_impl.cc
index 6440e36..9702e9e 100644
--- a/src/modules/audio_processing/main/source/audio_processing_impl.cc
+++ b/src/modules/audio_processing/audio_processing_impl.cc
@@ -10,36 +10,32 @@
#include "audio_processing_impl.h"
-#include <cassert>
-
-#include "module_common_types.h"
-
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
+#include <assert.h>
#include "audio_buffer.h"
+#include "critical_section_wrapper.h"
#include "echo_cancellation_impl.h"
#include "echo_control_mobile_impl.h"
+#include "file_wrapper.h"
#include "high_pass_filter_impl.h"
#include "gain_control_impl.h"
#include "level_estimator_impl.h"
+#include "module_common_types.h"
#include "noise_suppression_impl.h"
#include "processing_component.h"
#include "splitting_filter.h"
#include "voice_detection_impl.h"
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
namespace webrtc {
-namespace {
-
-enum Events {
- kInitializeEvent,
- kRenderEvent,
- kCaptureEvent
-};
-
-const char kMagicNumber[] = "#!vqetrace1.2";
-} // namespace
-
AudioProcessing* AudioProcessing::Create(int id) {
/*WEBRTC_TRACE(webrtc::kTraceModuleCall,
webrtc::kTraceAudioProcessing,
@@ -68,18 +64,21 @@
level_estimator_(NULL),
noise_suppression_(NULL),
voice_detection_(NULL),
- debug_file_(FileWrapper::Create()),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
render_audio_(NULL),
capture_audio_(NULL),
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ debug_file_(FileWrapper::Create()),
+ event_msg_(new audioproc::Event()),
+#endif
sample_rate_hz_(kSampleRate16kHz),
split_sample_rate_hz_(kSampleRate16kHz),
samples_per_channel_(sample_rate_hz_ / 100),
stream_delay_ms_(0),
was_stream_delay_set_(false),
- num_render_input_channels_(1),
- num_capture_input_channels_(1),
- num_capture_output_channels_(1) {
+ num_reverse_channels_(1),
+ num_input_channels_(1),
+ num_output_channels_(1) {
echo_cancellation_ = new EchoCancellationImpl(this);
component_list_.push_back(echo_cancellation_);
@@ -111,21 +110,21 @@
component_list_.pop_front();
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
debug_file_->CloseFile();
}
- delete debug_file_;
- debug_file_ = NULL;
+#endif
delete crit_;
crit_ = NULL;
- if (render_audio_ != NULL) {
+ if (render_audio_) {
delete render_audio_;
render_audio_ = NULL;
}
- if (capture_audio_ != NULL) {
+ if (capture_audio_) {
delete capture_audio_;
capture_audio_ = NULL;
}
@@ -155,9 +154,9 @@
capture_audio_ = NULL;
}
- render_audio_ = new AudioBuffer(num_render_input_channels_,
+ render_audio_ = new AudioBuffer(num_reverse_channels_,
samples_per_channel_);
- capture_audio_ = new AudioBuffer(num_capture_input_channels_,
+ capture_audio_ = new AudioBuffer(num_input_channels_,
samples_per_channel_);
was_stream_delay_set_ = false;
@@ -171,6 +170,15 @@
}
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+#endif
+
return kNoError;
}
@@ -205,13 +213,13 @@
return kBadParameterError;
}
- num_render_input_channels_ = channels;
+ num_reverse_channels_ = channels;
return InitializeLocked();
}
int AudioProcessingImpl::num_reverse_channels() const {
- return num_render_input_channels_;
+ return num_reverse_channels_;
}
int AudioProcessingImpl::set_num_channels(
@@ -231,18 +239,18 @@
return kBadParameterError;
}
- num_capture_input_channels_ = input_channels;
- num_capture_output_channels_ = output_channels;
+ num_input_channels_ = input_channels;
+ num_output_channels_ = output_channels;
return InitializeLocked();
}
int AudioProcessingImpl::num_input_channels() const {
- return num_capture_input_channels_;
+ return num_input_channels_;
}
int AudioProcessingImpl::num_output_channels() const {
- return num_capture_output_channels_;
+ return num_output_channels_;
}
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
@@ -253,12 +261,11 @@
return kNullPointerError;
}
- if (frame->_frequencyInHz !=
- static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
+ if (frame->_frequencyInHz != sample_rate_hz_) {
return kBadSampleRateError;
}
- if (frame->_audioChannel != num_capture_input_channels_) {
+ if (frame->_audioChannel != num_input_channels_) {
return kBadNumberChannelsError;
}
@@ -266,45 +273,31 @@
return kBadDataLengthError;
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
- WebRtc_UWord8 event = kCaptureEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_frequencyInHz,
- sizeof(frame->_frequencyInHz))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_audioChannel,
- sizeof(frame->_audioChannel))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
- sizeof(frame->_payloadDataLengthInSamples))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(frame->_payloadData,
- sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
- frame->_audioChannel)) {
- return kFileError;
- }
+ event_msg_->set_type(audioproc::Event::STREAM);
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_input_data(frame->_payloadData, data_size);
+ msg->set_delay(stream_delay_ms_);
+ msg->set_drift(echo_cancellation_->stream_drift_samples());
+ msg->set_level(gain_control_->stream_analog_level());
}
+#endif
capture_audio_->DeinterleaveFrom(frame);
// TODO(ajm): experiment with mixing and AEC placement.
- if (num_capture_output_channels_ < num_capture_input_channels_) {
- capture_audio_->Mix(num_capture_output_channels_);
-
- frame->_audioChannel = num_capture_output_channels_;
+ if (num_output_channels_ < num_input_channels_) {
+ capture_audio_->Mix(num_output_channels_);
+ frame->_audioChannel = num_output_channels_;
}
- if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_capture_input_channels_; i++) {
+ bool data_changed = stream_data_changed();
+ if (analysis_needed(data_changed)) {
+ for (int i = 0; i < num_output_channels_; i++) {
// Split into a low and high band.
SplittingFilterAnalysis(capture_audio_->data(i),
capture_audio_->low_pass_split_data(i),
@@ -354,13 +347,8 @@
return err;
}
- //err = level_estimator_->ProcessCaptureAudio(capture_audio_);
- //if (err != kNoError) {
- // return err;
- //}
-
- if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_capture_output_channels_; i++) {
+ if (synthesis_needed(data_changed)) {
+ for (int i = 0; i < num_output_channels_; i++) {
// Recombine low and high bands.
SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
capture_audio_->high_pass_split_data(i),
@@ -370,8 +358,29 @@
}
}
- capture_audio_->InterleaveTo(frame);
+ // The level estimator operates on the recombined data.
+ err = level_estimator_->ProcessStream(capture_audio_);
+ if (err != kNoError) {
+ return err;
+ }
+ capture_audio_->InterleaveTo(frame, data_changed);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ if (debug_file_->Open()) {
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_output_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+#endif
+
+ was_stream_delay_set_ = false;
return kNoError;
}
@@ -383,12 +392,11 @@
return kNullPointerError;
}
- if (frame->_frequencyInHz !=
- static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
+ if (frame->_frequencyInHz != sample_rate_hz_) {
return kBadSampleRateError;
}
- if (frame->_audioChannel != num_render_input_channels_) {
+ if (frame->_audioChannel != num_reverse_channels_) {
return kBadNumberChannelsError;
}
@@ -396,39 +404,26 @@
return kBadDataLengthError;
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
- WebRtc_UWord8 event = kRenderEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_frequencyInHz,
- sizeof(frame->_frequencyInHz))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_audioChannel,
- sizeof(frame->_audioChannel))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
- sizeof(frame->_payloadDataLengthInSamples))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(frame->_payloadData,
- sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
- frame->_audioChannel)) {
- return kFileError;
+ event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
+ audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+ const size_t data_size = sizeof(int16_t) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
}
}
+#endif
render_audio_->DeinterleaveFrom(frame);
// TODO(ajm): turn the splitting filter into a component?
if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_render_input_channels_; i++) {
+ for (int i = 0; i < num_reverse_channels_; i++) {
// Split into low and high band.
SplittingFilterAnalysis(render_audio_->data(i),
render_audio_->low_pass_split_data(i),
@@ -454,12 +449,6 @@
return err;
}
- //err = level_estimator_->AnalyzeReverseStream(render_audio_);
- //if (err != kNoError) {
- // return err;
- //}
-
- was_stream_delay_set_ = false;
return err; // TODO(ajm): this is for returning warnings; necessary?
}
@@ -496,6 +485,7 @@
return kNullPointerError;
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stop any ongoing recording.
if (debug_file_->Open()) {
if (debug_file_->CloseFile() == -1) {
@@ -508,35 +498,30 @@
return kFileError;
}
- if (debug_file_->WriteText("%s\n", kMagicNumber) == -1) {
- debug_file_->CloseFile();
- return kFileError;
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
}
-
- // TODO(ajm): should we do this? If so, we need the number of channels etc.
- // Record the default sample rate.
- WebRtc_UWord8 event = kInitializeEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&sample_rate_hz_, sizeof(sample_rate_hz_))) {
- return kFileError;
- }
-
return kNoError;
+#else
+ return kUnsupportedFunctionError;
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
int AudioProcessingImpl::StopDebugRecording() {
CriticalSectionScoped crit_scoped(*crit_);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// We just return if recording hasn't started.
if (debug_file_->Open()) {
if (debug_file_->CloseFile() == -1) {
return kFileError;
}
}
-
return kNoError;
+#else
+ return kUnsupportedFunctionError;
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
@@ -567,61 +552,6 @@
return voice_detection_;
}
-WebRtc_Word32 AudioProcessingImpl::Version(WebRtc_Word8* version,
- WebRtc_UWord32& bytes_remaining, WebRtc_UWord32& position) const {
- if (version == NULL) {
- /*WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioProcessing,
- -1,
- "Null version pointer");*/
- return kNullPointerError;
- }
- memset(&version[position], 0, bytes_remaining);
-
- WebRtc_Word8 my_version[] = "AudioProcessing 1.0.0";
- // Includes null termination.
- WebRtc_UWord32 length = static_cast<WebRtc_UWord32>(strlen(my_version));
- if (bytes_remaining < length) {
- /*WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioProcessing,
- -1,
- "Buffer of insufficient length");*/
- return kBadParameterError;
- }
- memcpy(&version[position], my_version, length);
- bytes_remaining -= length;
- position += length;
-
- std::list<ProcessingComponent*>::const_iterator it;
- for (it = component_list_.begin(); it != component_list_.end(); it++) {
- char component_version[256];
- strcpy(component_version, "\n");
- int err = (*it)->get_version(&component_version[1],
- sizeof(component_version) - 1);
- if (err != kNoError) {
- return err;
- }
- if (strncmp(&component_version[1], "\0", 1) == 0) {
- // Assume empty if first byte is NULL.
- continue;
- }
-
- length = static_cast<WebRtc_UWord32>(strlen(component_version));
- if (bytes_remaining < length) {
- /*WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioProcessing,
- -1,
- "Buffer of insufficient length");*/
- return kBadParameterError;
- }
- memcpy(&version[position], component_version, length);
- bytes_remaining -= length;
- position += length;
- }
-
- return kNoError;
-}
-
WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
CriticalSectionScoped crit_scoped(*crit_);
/*WEBRTC_TRACE(webrtc::kTraceModuleCall,
@@ -633,4 +563,90 @@
return kNoError;
}
+
+bool AudioProcessingImpl::stream_data_changed() const {
+ int enabled_count = 0;
+ std::list<ProcessingComponent*>::const_iterator it;
+ for (it = component_list_.begin(); it != component_list_.end(); it++) {
+ if ((*it)->is_component_enabled()) {
+ enabled_count++;
+ }
+ }
+
+ // Data is unchanged if no components are enabled, or if only level_estimator_
+ // or voice_detection_ is enabled.
+ if (enabled_count == 0) {
+ return false;
+ } else if (enabled_count == 1) {
+ if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
+ return false;
+ }
+ } else if (enabled_count == 2) {
+ if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool AudioProcessingImpl::synthesis_needed(bool stream_data_changed) const {
+ return (stream_data_changed && sample_rate_hz_ == kSampleRate32kHz);
+}
+
+bool AudioProcessingImpl::analysis_needed(bool stream_data_changed) const {
+ if (!stream_data_changed && !voice_detection_->is_enabled()) {
+ // Only level_estimator_ is enabled.
+ return false;
+ } else if (sample_rate_hz_ == kSampleRate32kHz) {
+ // Something besides level_estimator_ is enabled, and we have super-wb.
+ return true;
+ }
+ return false;
+}
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+int AudioProcessingImpl::WriteMessageToDebugFile() {
+ int32_t size = event_msg_->ByteSize();
+ if (size <= 0) {
+ return kUnspecifiedError;
+ }
+#if defined(WEBRTC_BIG_ENDIAN)
+ // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
+ // pretty safe in assuming little-endian.
+#endif
+
+ if (!event_msg_->SerializeToString(&event_str_)) {
+ return kUnspecifiedError;
+ }
+
+ // Write message preceded by its size.
+ if (!debug_file_->Write(&size, sizeof(int32_t))) {
+ return kFileError;
+ }
+ if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
+ return kFileError;
+ }
+
+ event_msg_->Clear();
+
+ return 0;
+}
+
+int AudioProcessingImpl::WriteInitMessage() {
+ event_msg_->set_type(audioproc::Event::INIT);
+ audioproc::Init* msg = event_msg_->mutable_init();
+ msg->set_sample_rate(sample_rate_hz_);
+ msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz());
+ msg->set_num_input_channels(num_input_channels_);
+ msg->set_num_output_channels(num_output_channels_);
+ msg->set_num_reverse_channels(num_reverse_channels_);
+
+ int err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+
+ return kNoError;
+}
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
} // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.h b/src/modules/audio_processing/audio_processing_impl.h
similarity index 79%
rename from src/modules/audio_processing/main/source/audio_processing_impl.h
rename to src/modules/audio_processing/audio_processing_impl.h
index 9707bde..c1ab476 100644
--- a/src/modules/audio_processing/main/source/audio_processing_impl.h
+++ b/src/modules/audio_processing/audio_processing_impl.h
@@ -11,17 +11,19 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-#include <list>
-
#include "audio_processing.h"
-namespace webrtc {
-class CriticalSectionWrapper;
-class FileWrapper;
+#include <list>
+#include <string>
+#include "scoped_ptr.h"
+
+namespace webrtc {
class AudioBuffer;
+class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
+class FileWrapper;
class GainControlImpl;
class HighPassFilterImpl;
class LevelEstimatorImpl;
@@ -29,6 +31,14 @@
class ProcessingComponent;
class VoiceDetectionImpl;
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+namespace audioproc {
+
+class Event;
+
+} // namespace audioproc
+#endif
+
class AudioProcessingImpl : public AudioProcessing {
public:
enum {
@@ -70,12 +80,13 @@
virtual VoiceDetection* voice_detection() const;
// Module methods.
- virtual WebRtc_Word32 Version(WebRtc_Word8* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position) const;
virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
private:
+ bool stream_data_changed() const;
+ bool synthesis_needed(bool stream_data_changed) const;
+ bool analysis_needed(bool stream_data_changed) const;
+
int id_;
EchoCancellationImpl* echo_cancellation_;
@@ -87,12 +98,18 @@
VoiceDetectionImpl* voice_detection_;
std::list<ProcessingComponent*> component_list_;
-
- FileWrapper* debug_file_;
CriticalSectionWrapper* crit_;
-
AudioBuffer* render_audio_;
AudioBuffer* capture_audio_;
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // TODO(andrew): make this more graceful. Ideally we would split this stuff
+ // out into a separate class with an "enabled" and "disabled" implementation.
+ int WriteMessageToDebugFile();
+ int WriteInitMessage();
+ scoped_ptr<FileWrapper> debug_file_;
+ scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
+ std::string event_str_; // Memory for protobuf serialization.
+#endif
int sample_rate_hz_;
int split_sample_rate_hz_;
@@ -100,9 +117,9 @@
int stream_delay_ms_;
bool was_stream_delay_set_;
- int num_render_input_channels_;
- int num_capture_input_channels_;
- int num_capture_output_channels_;
+ int num_reverse_channels_;
+ int num_input_channels_;
+ int num_output_channels_;
};
} // namespace webrtc
diff --git a/src/modules/audio_processing/debug.proto b/src/modules/audio_processing/debug.proto
new file mode 100644
index 0000000..4b3a163
--- /dev/null
+++ b/src/modules/audio_processing/debug.proto
@@ -0,0 +1,37 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+message Init {
+ optional int32 sample_rate = 1;
+ optional int32 device_sample_rate = 2;
+ optional int32 num_input_channels = 3;
+ optional int32 num_output_channels = 4;
+ optional int32 num_reverse_channels = 5;
+}
+
+message ReverseStream {
+ optional bytes data = 1;
+}
+
+message Stream {
+ optional bytes input_data = 1;
+ optional bytes output_data = 2;
+ optional int32 delay = 3;
+ optional sint32 drift = 4;
+ optional int32 level = 5;
+}
+
+message Event {
+ enum Type {
+ INIT = 0;
+ REVERSE_STREAM = 1;
+ STREAM = 2;
+ }
+
+ required Type type = 1;
+
+ optional Init init = 2;
+ optional ReverseStream reverse_stream = 3;
+ optional Stream stream = 4;
+}
diff --git a/src/modules/audio_processing/main/source/echo_cancellation_impl.cc b/src/modules/audio_processing/echo_cancellation_impl.cc
similarity index 90%
rename from src/modules/audio_processing/main/source/echo_cancellation_impl.cc
rename to src/modules/audio_processing/echo_cancellation_impl.cc
index 886d5f1..61940b1 100644
--- a/src/modules/audio_processing/main/source/echo_cancellation_impl.cc
+++ b/src/modules/audio_processing/echo_cancellation_impl.cc
@@ -66,7 +66,8 @@
device_sample_rate_hz_(48000),
stream_drift_samples_(0),
was_stream_drift_set_(false),
- stream_has_echo_(false) {}
+ stream_has_echo_(false),
+ delay_logging_enabled_(false) {}
EchoCancellationImpl::~EchoCancellationImpl() {}
@@ -283,6 +284,39 @@
return stream_has_echo_;
}
+int EchoCancellationImpl::enable_delay_logging(bool enable) {
+ CriticalSectionScoped crit_scoped(*apm_->crit());
+ delay_logging_enabled_ = enable;
+ return Configure();
+}
+
+bool EchoCancellationImpl::is_delay_logging_enabled() const {
+ return delay_logging_enabled_;
+}
+
+// TODO(bjornv): How should we handle the multi-channel case?
+int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
+ CriticalSectionScoped crit_scoped(*apm_->crit());
+ if (median == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (std == NULL) {
+ return apm_->kNullPointerError;
+ }
+
+ if (!is_component_enabled() || !delay_logging_enabled_) {
+ return apm_->kNotEnabledError;
+ }
+
+ Handle* my_handle = static_cast<Handle*>(handle(0));
+ if (WebRtcAec_GetDelayMetrics(my_handle, median, std) !=
+ apm_->kNoError) {
+ return GetHandleError(my_handle);
+ }
+
+ return apm_->kNoError;
+}
+
int EchoCancellationImpl::Initialize() {
int err = ProcessingComponent::Initialize();
if (err != apm_->kNoError || !is_component_enabled()) {
@@ -332,6 +366,7 @@
config.metricsMode = metrics_enabled_;
config.nlpMode = MapSetting(suppression_level_);
config.skewMode = drift_compensation_enabled_;
+ config.delay_logging = delay_logging_enabled_;
return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
}
diff --git a/src/modules/audio_processing/main/source/echo_cancellation_impl.h b/src/modules/audio_processing/echo_cancellation_impl.h
similarity index 93%
rename from src/modules/audio_processing/main/source/echo_cancellation_impl.h
rename to src/modules/audio_processing/echo_cancellation_impl.h
index 380a698..a483a3a 100644
--- a/src/modules/audio_processing/main/source/echo_cancellation_impl.h
+++ b/src/modules/audio_processing/echo_cancellation_impl.h
@@ -29,6 +29,8 @@
// EchoCancellation implementation.
virtual bool is_enabled() const;
+ virtual int device_sample_rate_hz() const;
+ virtual int stream_drift_samples() const;
// ProcessingComponent implementation.
virtual int Initialize();
@@ -40,15 +42,16 @@
virtual int enable_drift_compensation(bool enable);
virtual bool is_drift_compensation_enabled() const;
virtual int set_device_sample_rate_hz(int rate);
- virtual int device_sample_rate_hz() const;
virtual int set_stream_drift_samples(int drift);
- virtual int stream_drift_samples() const;
virtual int set_suppression_level(SuppressionLevel level);
virtual SuppressionLevel suppression_level() const;
virtual int enable_metrics(bool enable);
virtual bool are_metrics_enabled() const;
virtual bool stream_has_echo() const;
virtual int GetMetrics(Metrics* metrics);
+ virtual int enable_delay_logging(bool enable);
+ virtual bool is_delay_logging_enabled() const;
+ virtual int GetDelayMetrics(int* median, int* std);
// ProcessingComponent implementation.
virtual void* CreateHandle() const;
@@ -66,6 +69,7 @@
int stream_drift_samples_;
bool was_stream_drift_set_;
bool stream_has_echo_;
+ bool delay_logging_enabled_;
};
} // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/echo_control_mobile_impl.cc b/src/modules/audio_processing/echo_control_mobile_impl.cc
similarity index 74%
rename from src/modules/audio_processing/main/source/echo_control_mobile_impl.cc
rename to src/modules/audio_processing/echo_control_mobile_impl.cc
index 1cd2502..ff15255 100644
--- a/src/modules/audio_processing/main/source/echo_control_mobile_impl.cc
+++ b/src/modules/audio_processing/echo_control_mobile_impl.cc
@@ -11,6 +11,7 @@
#include "echo_control_mobile_impl.h"
#include <cassert>
+#include <cstring>
#include "critical_section_wrapper.h"
#include "echo_control_mobile.h"
@@ -44,26 +45,37 @@
switch (err) {
case AECM_UNSUPPORTED_FUNCTION_ERROR:
return AudioProcessing::kUnsupportedFunctionError;
+ case AECM_NULL_POINTER_ERROR:
+ return AudioProcessing::kNullPointerError;
case AECM_BAD_PARAMETER_ERROR:
return AudioProcessing::kBadParameterError;
case AECM_BAD_PARAMETER_WARNING:
return AudioProcessing::kBadStreamParameterWarning;
default:
- // AECMOBFIX_UNSPECIFIED_ERROR
- // AECMOBFIX_UNINITIALIZED_ERROR
- // AECMOBFIX_NULL_POINTER_ERROR
+ // AECM_UNSPECIFIED_ERROR
+ // AECM_UNINITIALIZED_ERROR
return AudioProcessing::kUnspecifiedError;
}
}
} // namespace
+size_t EchoControlMobile::echo_path_size_bytes() {
+ return WebRtcAecm_echo_path_size_bytes();
+}
+
EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessingImpl* apm)
: ProcessingComponent(apm),
apm_(apm),
routing_mode_(kSpeakerphone),
- comfort_noise_enabled_(true) {}
+ comfort_noise_enabled_(true),
+ external_echo_path_(NULL) {}
-EchoControlMobileImpl::~EchoControlMobileImpl() {}
+EchoControlMobileImpl::~EchoControlMobileImpl() {
+ if (external_echo_path_ != NULL) {
+ delete [] external_echo_path_;
+ external_echo_path_ = NULL;
+ }
+}
int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
if (!is_component_enabled()) {
@@ -181,6 +193,48 @@
return comfort_noise_enabled_;
}
+int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
+ size_t size_bytes) {
+ CriticalSectionScoped crit_scoped(*apm_->crit());
+ if (echo_path == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (size_bytes != echo_path_size_bytes()) {
+ // Size mismatch
+ return apm_->kBadParameterError;
+ }
+
+ if (external_echo_path_ == NULL) {
+ external_echo_path_ = new unsigned char[size_bytes];
+ }
+ memcpy(external_echo_path_, echo_path, size_bytes);
+
+ return Initialize();
+}
+
+int EchoControlMobileImpl::GetEchoPath(void* echo_path,
+ size_t size_bytes) const {
+ CriticalSectionScoped crit_scoped(*apm_->crit());
+ if (echo_path == NULL) {
+ return apm_->kNullPointerError;
+ }
+ if (size_bytes != echo_path_size_bytes()) {
+ // Size mismatch
+ return apm_->kBadParameterError;
+ }
+ if (!is_component_enabled()) {
+ return apm_->kNotEnabledError;
+ }
+
+ // Get the echo path from the first channel
+ Handle* my_handle = static_cast<Handle*>(handle(0));
+ if (WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes) != 0) {
+ return GetHandleError(my_handle);
+ }
+
+ return apm_->kNoError;
+}
+
int EchoControlMobileImpl::Initialize() {
if (!is_component_enabled()) {
return apm_->kNoError;
@@ -197,7 +251,7 @@
int EchoControlMobileImpl::get_version(char* version,
int version_len_bytes) const {
if (WebRtcAecm_get_version(version, version_len_bytes) != 0) {
- return apm_->kBadParameterError;
+ return apm_->kBadParameterError;
}
return apm_->kNoError;
@@ -219,10 +273,20 @@
}
int EchoControlMobileImpl::InitializeHandle(void* handle) const {
- return WebRtcAecm_Init(static_cast<Handle*>(handle),
- apm_->sample_rate_hz(),
- 48000); // Dummy value. This isn't actually
- // required by AECM.
+ assert(handle != NULL);
+ Handle* my_handle = static_cast<Handle*>(handle);
+ if (WebRtcAecm_Init(my_handle, apm_->sample_rate_hz()) != 0) {
+ return GetHandleError(my_handle);
+ }
+ if (external_echo_path_ != NULL) {
+ if (WebRtcAecm_InitEchoPath(my_handle,
+ external_echo_path_,
+ echo_path_size_bytes()) != 0) {
+ return GetHandleError(my_handle);
+ }
+ }
+
+ return apm_->kNoError;
}
int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
diff --git a/src/modules/audio_processing/main/source/echo_control_mobile_impl.h b/src/modules/audio_processing/echo_control_mobile_impl.h
similarity index 91%
rename from src/modules/audio_processing/main/source/echo_control_mobile_impl.h
rename to src/modules/audio_processing/echo_control_mobile_impl.h
index 2fd6248..6314e66 100644
--- a/src/modules/audio_processing/main/source/echo_control_mobile_impl.h
+++ b/src/modules/audio_processing/echo_control_mobile_impl.h
@@ -41,6 +41,8 @@
virtual RoutingMode routing_mode() const;
virtual int enable_comfort_noise(bool enable);
virtual bool is_comfort_noise_enabled() const;
+ virtual int SetEchoPath(const void* echo_path, size_t size_bytes);
+ virtual int GetEchoPath(void* echo_path, size_t size_bytes) const;
// ProcessingComponent implementation.
virtual void* CreateHandle() const;
@@ -53,6 +55,7 @@
const AudioProcessingImpl* apm_;
RoutingMode routing_mode_;
bool comfort_noise_enabled_;
+ unsigned char* external_echo_path_;
};
} // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/gain_control_impl.cc b/src/modules/audio_processing/gain_control_impl.cc
similarity index 100%
rename from src/modules/audio_processing/main/source/gain_control_impl.cc
rename to src/modules/audio_processing/gain_control_impl.cc
diff --git a/src/modules/audio_processing/main/source/gain_control_impl.h b/src/modules/audio_processing/gain_control_impl.h
similarity index 99%
rename from src/modules/audio_processing/main/source/gain_control_impl.h
rename to src/modules/audio_processing/gain_control_impl.h
index a11d606..7b6987e 100644
--- a/src/modules/audio_processing/main/source/gain_control_impl.h
+++ b/src/modules/audio_processing/gain_control_impl.h
@@ -36,12 +36,12 @@
// GainControl implementation.
virtual bool is_enabled() const;
+ virtual int stream_analog_level();
private:
// GainControl implementation.
virtual int Enable(bool enable);
virtual int set_stream_analog_level(int level);
- virtual int stream_analog_level();
virtual int set_mode(Mode mode);
virtual Mode mode() const;
virtual int set_target_level_dbfs(int level);
diff --git a/src/modules/audio_processing/main/source/high_pass_filter_impl.cc b/src/modules/audio_processing/high_pass_filter_impl.cc
similarity index 100%
rename from src/modules/audio_processing/main/source/high_pass_filter_impl.cc
rename to src/modules/audio_processing/high_pass_filter_impl.cc
diff --git a/src/modules/audio_processing/main/source/high_pass_filter_impl.h b/src/modules/audio_processing/high_pass_filter_impl.h
similarity index 100%
rename from src/modules/audio_processing/main/source/high_pass_filter_impl.h
rename to src/modules/audio_processing/high_pass_filter_impl.h
diff --git a/src/modules/audio_processing/main/interface/audio_processing.h b/src/modules/audio_processing/interface/audio_processing.h
similarity index 89%
rename from src/modules/audio_processing/main/interface/audio_processing.h
rename to src/modules/audio_processing/interface/audio_processing.h
index dc9c232..ee4d06f 100644
--- a/src/modules/audio_processing/main/interface/audio_processing.h
+++ b/src/modules/audio_processing/interface/audio_processing.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
+#include <stddef.h> // size_t
+
#include "typedefs.h"
#include "module.h"
@@ -111,7 +113,10 @@
// for each far-end stream which requires processing. On the server-side,
// this would typically be one instance for every incoming stream.
static AudioProcessing* Create(int id);
+ virtual ~AudioProcessing() {};
+ // TODO(andrew): remove this method. We now allow users to delete instances
+ // directly, useful for scoped_ptr.
// Destroys a |apm| instance.
static void Destroy(AudioProcessing* apm);
@@ -186,7 +191,7 @@
// a NULL-terminated string. If there is an ongoing recording, the old file
// will be closed, and recording will continue in the newly specified file.
// An already existing file will be overwritten without warning.
- static const int kMaxFilenameSize = 1024;
+ static const size_t kMaxFilenameSize = 1024;
virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
// Stops recording debugging information, and closes the file. Recording
@@ -238,9 +243,6 @@
// Inherited from Module.
virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; };
virtual WebRtc_Word32 Process() { return -1; };
-
- protected:
- virtual ~AudioProcessing() {};
};
// The acoustic echo cancellation (AEC) component provides better performance
@@ -320,6 +322,16 @@
// TODO(ajm): discuss the metrics update period.
virtual int GetMetrics(Metrics* metrics) = 0;
+ // Enables computation and logging of delay values. Statistics are obtained
+ // through |GetDelayMetrics()|.
+ virtual int enable_delay_logging(bool enable) = 0;
+ virtual bool is_delay_logging_enabled() const = 0;
+
+ // The delay metrics consists of the delay |median| and the delay standard
+ // deviation |std|. The values are averaged over the time period since the
+ // last call to |GetDelayMetrics()|.
+ virtual int GetDelayMetrics(int* median, int* std) = 0;
+
protected:
virtual ~EchoCancellation() {};
};
@@ -356,6 +368,26 @@
virtual int enable_comfort_noise(bool enable) = 0;
virtual bool is_comfort_noise_enabled() const = 0;
+ // A typical use case is to initialize the component with an echo path from a
+ // previous call. The echo path is retrieved using |GetEchoPath()|, typically
+ // at the end of a call. The data can then be stored for later use as an
+ // initializer before the next call, using |SetEchoPath()|.
+ //
+ // Controlling the echo path this way requires the data |size_bytes| to match
+ // the internal echo path size. This size can be acquired using
+ // |echo_path_size_bytes()|. |SetEchoPath()| causes an entire reset, worth
+ // noting if it is to be called during an ongoing call.
+ //
+ // It is possible that version incompatibilities may result in a stored echo
+ // path of the incorrect size. In this case, the stored path should be
+ // discarded.
+ virtual int SetEchoPath(const void* echo_path, size_t size_bytes) = 0;
+ virtual int GetEchoPath(void* echo_path, size_t size_bytes) const = 0;
+
+ // The returned path size is guaranteed not to change for the lifetime of
+ // the application.
+ static size_t echo_path_size_bytes();
+
protected:
virtual ~EchoControlMobile() {};
};
@@ -469,21 +501,18 @@
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
- // The metrics are reported in dBFs calculated as:
- // Level = 10log_10(P_s / P_max) [dBFs], where
- // P_s is the signal power and P_max is the maximum possible (or peak)
- // power. With 16-bit signals, P_max = (2^15)^2.
- struct Metrics {
- AudioProcessing::Statistic signal; // Overall signal level.
- AudioProcessing::Statistic speech; // Speech level.
- AudioProcessing::Statistic noise; // Noise level.
- };
-
- virtual int GetMetrics(Metrics* metrics, Metrics* reverse_metrics) = 0;
-
- //virtual int enable_noise_warning(bool enable) = 0;
- //bool is_noise_warning_enabled() const = 0;
- //virtual bool stream_has_high_noise() const = 0;
+ // Returns the root mean square (RMS) level in dBFs (decibels from digital
+ // full-scale), or alternately dBov. It is computed over all primary stream
+ // frames since the last call to RMS(). The returned value is positive but
+ // should be interpreted as negative. It is constrained to [0, 127].
+ //
+ // The computation follows:
+ // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-05
+ // with the intent that it can provide the RTP audio level indication.
+ //
+ // Frames passed to ProcessStream() with an |_energy| of zero are considered
+ // to have been muted. The RMS of the frame will be interpreted as -127.
+ virtual int RMS() = 0;
protected:
virtual ~LevelEstimator() {};
@@ -517,6 +546,10 @@
// The voice activity detection (VAD) component analyzes the stream to
// determine if voice is present. A facility is also provided to pass in an
// external VAD decision.
+//
+// In addition to |stream_has_voice()| the VAD decision is provided through the
+// |AudioFrame| passed to |ProcessStream()|. The |_vadActivity| member will be
+// modified to reflect the current decision.
class VoiceDetection {
public:
virtual int Enable(bool enable) = 0;
diff --git a/src/modules/audio_processing/level_estimator_impl.cc b/src/modules/audio_processing/level_estimator_impl.cc
new file mode 100644
index 0000000..f127d4a
--- /dev/null
+++ b/src/modules/audio_processing/level_estimator_impl.cc
@@ -0,0 +1,172 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "level_estimator_impl.h"
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+
+#include "audio_processing_impl.h"
+#include "audio_buffer.h"
+#include "critical_section_wrapper.h"
+
+namespace webrtc {
+namespace {
+
+const double kMaxSquaredLevel = 32768.0 * 32768.0;
+
+class Level {
+ public:
+ static const int kMinLevel = 127;
+
+ Level()
+ : sum_square_(0.0),
+ sample_count_(0) {}
+ ~Level() {}
+
+ void Init() {
+ sum_square_ = 0.0;
+ sample_count_ = 0;
+ }
+
+ void Process(int16_t* data, int length) {
+ assert(data != NULL);
+ assert(length > 0);
+ sum_square_ += SumSquare(data, length);
+ sample_count_ += length;
+ }
+
+ void ProcessMuted(int length) {
+ assert(length > 0);
+ sample_count_ += length;
+ }
+
+ int RMS() {
+ if (sample_count_ == 0 || sum_square_ == 0.0) {
+ Init();
+ return kMinLevel;
+ }
+
+ // Normalize by the max level.
+ double rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
+ // 20log_10(x^0.5) = 10log_10(x)
+ rms = 10 * log10(rms);
+ if (rms > 0)
+ rms = 0;
+ else if (rms < -kMinLevel)
+ rms = -kMinLevel;
+
+ rms = -rms;
+ Init();
+ return static_cast<int>(rms + 0.5);
+ }
+
+ private:
+ static double SumSquare(int16_t* data, int length) {
+ double sum_square = 0.0;
+ for (int i = 0; i < length; ++i) {
+ double data_d = static_cast<double>(data[i]);
+ sum_square += data_d * data_d;
+ }
+ return sum_square;
+ }
+
+ double sum_square_;
+ int sample_count_;
+};
+} // namespace
+
+LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
+ : ProcessingComponent(apm),
+ apm_(apm) {}
+
+LevelEstimatorImpl::~LevelEstimatorImpl() {}
+
+int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+ if (!is_component_enabled()) {
+ return apm_->kNoError;
+ }
+
+ Level* level = static_cast<Level*>(handle(0));
+ if (audio->is_muted()) {
+ level->ProcessMuted(audio->samples_per_channel());
+ return apm_->kNoError;
+ }
+
+ int16_t* mixed_data = audio->data(0);
+ if (audio->num_channels() > 1) {
+ audio->CopyAndMix(1);
+ mixed_data = audio->mixed_data(0);
+ }
+
+ level->Process(mixed_data, audio->samples_per_channel());
+
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::Enable(bool enable) {
+ CriticalSectionScoped crit_scoped(*apm_->crit());
+ return EnableComponent(enable);
+}
+
+bool LevelEstimatorImpl::is_enabled() const {
+ return is_component_enabled();
+}
+
+int LevelEstimatorImpl::RMS() {
+ if (!is_component_enabled()) {
+ return apm_->kNotEnabledError;
+ }
+
+ Level* level = static_cast<Level*>(handle(0));
+ return level->RMS();
+}
+
+int LevelEstimatorImpl::get_version(char* version,
+ int version_len_bytes) const {
+ // An empty string is used to indicate no version information.
+ memset(version, 0, version_len_bytes);
+ return apm_->kNoError;
+}
+
+void* LevelEstimatorImpl::CreateHandle() const {
+ return new Level;
+}
+
+int LevelEstimatorImpl::DestroyHandle(void* handle) const {
+ assert(handle != NULL);
+ Level* level = static_cast<Level*>(handle);
+ delete level;
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::InitializeHandle(void* handle) const {
+ assert(handle != NULL);
+ Level* level = static_cast<Level*>(handle);
+ level->Init();
+
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
+ return apm_->kNoError;
+}
+
+int LevelEstimatorImpl::num_handles_required() const {
+ return 1;
+}
+
+int LevelEstimatorImpl::GetHandleError(void* handle) const {
+ // The component has no detailed errors.
+ assert(handle != NULL);
+ return apm_->kUnspecifiedError;
+}
+} // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/level_estimator_impl.h b/src/modules/audio_processing/level_estimator_impl.h
similarity index 90%
rename from src/modules/audio_processing/main/source/level_estimator_impl.h
rename to src/modules/audio_processing/level_estimator_impl.h
index 1515722..c9b7e02 100644
--- a/src/modules/audio_processing/main/source/level_estimator_impl.h
+++ b/src/modules/audio_processing/level_estimator_impl.h
@@ -24,8 +24,7 @@
explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
virtual ~LevelEstimatorImpl();
- int AnalyzeReverseStream(AudioBuffer* audio);
- int ProcessCaptureAudio(AudioBuffer* audio);
+ int ProcessStream(AudioBuffer* audio);
// LevelEstimator implementation.
virtual bool is_enabled() const;
@@ -36,7 +35,7 @@
private:
// LevelEstimator implementation.
virtual int Enable(bool enable);
- virtual int GetMetrics(Metrics* metrics, Metrics* reverse_metrics);
+ virtual int RMS();
// ProcessingComponent implementation.
virtual void* CreateHandle() const;
diff --git a/src/modules/audio_processing/main/apm_tests.gyp b/src/modules/audio_processing/main/apm_tests.gyp
deleted file mode 100644
index 441abeb..0000000
--- a/src/modules/audio_processing/main/apm_tests.gyp
+++ /dev/null
@@ -1,60 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../../../common_settings.gypi',
- ],
- 'targets': [
- {
- 'target_name': 'unit_test',
- 'type': 'executable',
- 'dependencies': [
- 'source/apm.gyp:audio_processing',
- '../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-
- '../../../../testing/gtest.gyp:gtest',
- '../../../../testing/gtest.gyp:gtest_main',
- '../../../../third_party/protobuf/protobuf.gyp:protobuf_lite',
- ],
- 'include_dirs': [
- '../../../../testing/gtest/include',
- ],
- 'sources': [
- 'test/unit_test/unit_test.cc',
- 'test/unit_test/audio_processing_unittest.pb.cc',
- 'test/unit_test/audio_processing_unittest.pb.h',
- ],
- },
- {
- 'target_name': 'process_test',
- 'type': 'executable',
- 'dependencies': [
- 'source/apm.gyp:audio_processing',
- '../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
-
- '../../../../testing/gtest.gyp:gtest',
- '../../../../testing/gtest.gyp:gtest_main',
- ],
- 'include_dirs': [
- '../../../../testing/gtest/include',
- ],
- 'sources': [
- 'test/process_test/process_test.cc',
- ],
- },
-
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/main/source/Android.mk b/src/modules/audio_processing/main/source/Android.mk
deleted file mode 100644
index 634ad6a..0000000
--- a/src/modules/audio_processing/main/source/Android.mk
+++ /dev/null
@@ -1,75 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_apm
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := audio_buffer.cc \
- audio_processing_impl.cc \
- echo_cancellation_impl.cc \
- echo_control_mobile_impl.cc \
- gain_control_impl.cc \
- high_pass_filter_impl.cc \
- level_estimator_impl.cc \
- noise_suppression_impl.cc \
- splitting_filter.cc \
- processing_component.cc \
- voice_detection_impl.cc
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_NS_FIXED'
-# floating point
-# -DWEBRTC_NS_FLOAT'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface \
- $(LOCAL_PATH)/../../aec/main/interface \
- $(LOCAL_PATH)/../../aecm/main/interface \
- $(LOCAL_PATH)/../../agc/main/interface \
- $(LOCAL_PATH)/../../ns/main/interface \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing_library/main/interface \
- $(LOCAL_PATH)/../../../../common_audio/vad/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
-
diff --git a/src/modules/audio_processing/main/source/apm.gyp b/src/modules/audio_processing/main/source/apm.gyp
deleted file mode 100644
index 93811c7..0000000
--- a/src/modules/audio_processing/main/source/apm.gyp
+++ /dev/null
@@ -1,77 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../../../../common_settings.gypi', # Common settings
- ],
- 'targets': [
- {
- 'target_name': 'audio_processing',
- 'type': '<(library)',
- 'conditions': [
- ['prefer_fixed_point==1', {
- 'dependencies': ['../../ns/main/source/ns.gyp:ns_fix'],
- 'defines': ['WEBRTC_NS_FIXED'],
- }, { # else: prefer_fixed_point==0
- 'dependencies': ['../../ns/main/source/ns.gyp:ns'],
- 'defines': ['WEBRTC_NS_FLOAT'],
- }],
- ],
- 'dependencies': [
- '../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '../../aec/main/source/aec.gyp:aec',
- '../../aecm/main/source/aecm.gyp:aecm',
- '../../agc/main/source/agc.gyp:agc',
- '../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
- '../../../../common_audio/vad/main/source/vad.gyp:vad',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- },
- 'sources': [
- '../interface/audio_processing.h',
- 'audio_buffer.cc',
- 'audio_buffer.h',
- 'audio_processing_impl.cc',
- 'audio_processing_impl.h',
- 'echo_cancellation_impl.cc',
- 'echo_cancellation_impl.h',
- 'echo_control_mobile_impl.cc',
- 'echo_control_mobile_impl.h',
- 'gain_control_impl.cc',
- 'gain_control_impl.h',
- 'high_pass_filter_impl.cc',
- 'high_pass_filter_impl.h',
- 'level_estimator_impl.cc',
- 'level_estimator_impl.h',
- 'noise_suppression_impl.cc',
- 'noise_suppression_impl.h',
- 'splitting_filter.cc',
- 'splitting_filter.h',
- 'processing_component.cc',
- 'processing_component.h',
- 'voice_detection_impl.cc',
- 'voice_detection_impl.h',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/main/source/audio_buffer.cc b/src/modules/audio_processing/main/source/audio_buffer.cc
deleted file mode 100644
index 6b20fce..0000000
--- a/src/modules/audio_processing/main/source/audio_buffer.cc
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_buffer.h"
-
-#include "module_common_types.h"
-
-namespace webrtc {
-namespace {
-
-enum {
- kSamplesPer8kHzChannel = 80,
- kSamplesPer16kHzChannel = 160,
- kSamplesPer32kHzChannel = 320
-};
-
-void StereoToMono(const WebRtc_Word16* left, const WebRtc_Word16* right,
- WebRtc_Word16* out, int samples_per_channel) {
- WebRtc_Word32 data_int32 = 0;
- for (int i = 0; i < samples_per_channel; i++) {
- data_int32 = (left[i] + right[i]) >> 1;
- if (data_int32 > 32767) {
- data_int32 = 32767;
- } else if (data_int32 < -32768) {
- data_int32 = -32768;
- }
-
- out[i] = static_cast<WebRtc_Word16>(data_int32);
- }
-}
-} // namespace
-
-struct AudioChannel {
- AudioChannel() {
- memset(data, 0, sizeof(data));
- }
-
- WebRtc_Word16 data[kSamplesPer32kHzChannel];
-};
-
-struct SplitAudioChannel {
- SplitAudioChannel() {
- memset(low_pass_data, 0, sizeof(low_pass_data));
- memset(high_pass_data, 0, sizeof(high_pass_data));
- memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
- memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
- memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
- memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
- }
-
- WebRtc_Word16 low_pass_data[kSamplesPer16kHzChannel];
- WebRtc_Word16 high_pass_data[kSamplesPer16kHzChannel];
-
- WebRtc_Word32 analysis_filter_state1[6];
- WebRtc_Word32 analysis_filter_state2[6];
- WebRtc_Word32 synthesis_filter_state1[6];
- WebRtc_Word32 synthesis_filter_state2[6];
-};
-
-// TODO(am): check range of input parameters?
-AudioBuffer::AudioBuffer(WebRtc_Word32 max_num_channels,
- WebRtc_Word32 samples_per_channel)
- : max_num_channels_(max_num_channels),
- num_channels_(0),
- num_mixed_channels_(0),
- num_mixed_low_pass_channels_(0),
- samples_per_channel_(samples_per_channel),
- samples_per_split_channel_(samples_per_channel),
- reference_copied_(false),
- data_(NULL),
- channels_(NULL),
- split_channels_(NULL),
- mixed_low_pass_channels_(NULL),
- low_pass_reference_channels_(NULL) {
- if (max_num_channels_ > 1) {
- channels_ = new AudioChannel[max_num_channels_];
- mixed_low_pass_channels_ = new AudioChannel[max_num_channels_];
- }
- low_pass_reference_channels_ = new AudioChannel[max_num_channels_];
-
- if (samples_per_channel_ == kSamplesPer32kHzChannel) {
- split_channels_ = new SplitAudioChannel[max_num_channels_];
- samples_per_split_channel_ = kSamplesPer16kHzChannel;
- }
-}
-
-AudioBuffer::~AudioBuffer() {
- if (channels_ != NULL) {
- delete [] channels_;
- }
-
- if (mixed_low_pass_channels_ != NULL) {
- delete [] mixed_low_pass_channels_;
- }
-
- if (low_pass_reference_channels_ != NULL) {
- delete [] low_pass_reference_channels_;
- }
-
- if (split_channels_ != NULL) {
- delete [] split_channels_;
- }
-}
-
-WebRtc_Word16* AudioBuffer::data(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (data_ != NULL) {
- return data_;
- }
-
- return channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_split_data(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (split_channels_ == NULL) {
- return data(channel);
- }
-
- return split_channels_[channel].low_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::high_pass_split_data(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (split_channels_ == NULL) {
- return NULL;
- }
-
- return split_channels_[channel].high_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::mixed_low_pass_data(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
-
- return mixed_low_pass_channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_reference(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (!reference_copied_) {
- return NULL;
- }
-
- return low_pass_reference_channels_[channel].data;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state1(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].analysis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state2(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].analysis_filter_state2;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state1(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].synthesis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state2(WebRtc_Word32 channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].synthesis_filter_state2;
-}
-
-WebRtc_Word32 AudioBuffer::num_channels() const {
- return num_channels_;
-}
-
-WebRtc_Word32 AudioBuffer::samples_per_channel() const {
- return samples_per_channel_;
-}
-
-WebRtc_Word32 AudioBuffer::samples_per_split_channel() const {
- return samples_per_split_channel_;
-}
-
-// TODO(ajm): Do deinterleaving and mixing in one step?
-void AudioBuffer::DeinterleaveFrom(AudioFrame* audioFrame) {
- assert(audioFrame->_audioChannel <= max_num_channels_);
- assert(audioFrame->_payloadDataLengthInSamples == samples_per_channel_);
-
- num_channels_ = audioFrame->_audioChannel;
- num_mixed_channels_ = 0;
- num_mixed_low_pass_channels_ = 0;
- reference_copied_ = false;
-
- if (num_channels_ == 1) {
- // We can get away with a pointer assignment in this case.
- data_ = audioFrame->_payloadData;
- return;
- }
-
- for (int i = 0; i < num_channels_; i++) {
- WebRtc_Word16* deinterleaved = channels_[i].data;
- WebRtc_Word16* interleaved = audioFrame->_payloadData;
- WebRtc_Word32 interleaved_idx = i;
- for (int j = 0; j < samples_per_channel_; j++) {
- deinterleaved[j] = interleaved[interleaved_idx];
- interleaved_idx += num_channels_;
- }
- }
-}
-
-void AudioBuffer::InterleaveTo(AudioFrame* audioFrame) const {
- assert(audioFrame->_audioChannel == num_channels_);
- assert(audioFrame->_payloadDataLengthInSamples == samples_per_channel_);
-
- if (num_channels_ == 1) {
- if (num_mixed_channels_ == 1) {
- memcpy(audioFrame->_payloadData,
- channels_[0].data,
- sizeof(WebRtc_Word16) * samples_per_channel_);
- } else {
- // These should point to the same buffer in this case.
- assert(data_ == audioFrame->_payloadData);
- }
-
- return;
- }
-
- for (int i = 0; i < num_channels_; i++) {
- WebRtc_Word16* deinterleaved = channels_[i].data;
- WebRtc_Word16* interleaved = audioFrame->_payloadData;
- WebRtc_Word32 interleaved_idx = i;
- for (int j = 0; j < samples_per_channel_; j++) {
- interleaved[interleaved_idx] = deinterleaved[j];
- interleaved_idx += num_channels_;
- }
- }
-}
-
-// TODO(ajm): would be good to support the no-mix case with pointer assignment.
-// TODO(ajm): handle mixing to multiple channels?
-void AudioBuffer::Mix(WebRtc_Word32 num_mixed_channels) {
- // We currently only support the stereo to mono case.
- assert(num_channels_ == 2);
- assert(num_mixed_channels == 1);
-
- StereoToMono(channels_[0].data,
- channels_[1].data,
- channels_[0].data,
- samples_per_channel_);
-
- num_channels_ = num_mixed_channels;
- num_mixed_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels) {
- // We currently only support the stereo to mono case.
- assert(num_channels_ == 2);
- assert(num_mixed_channels == 1);
-
- StereoToMono(low_pass_split_data(0),
- low_pass_split_data(1),
- mixed_low_pass_channels_[0].data,
- samples_per_split_channel_);
-
- num_mixed_low_pass_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyLowPassToReference() {
- reference_copied_ = true;
- for (int i = 0; i < num_channels_; i++) {
- memcpy(low_pass_reference_channels_[i].data,
- low_pass_split_data(i),
- sizeof(WebRtc_Word16) * samples_per_split_channel_);
- }
-}
-} // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/audio_buffer.h b/src/modules/audio_processing/main/source/audio_buffer.h
deleted file mode 100644
index 15f850b..0000000
--- a/src/modules/audio_processing/main/source/audio_buffer.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-
-#include "typedefs.h"
-
-
-namespace webrtc {
-
-struct AudioChannel;
-struct SplitAudioChannel;
-class AudioFrame;
-
-class AudioBuffer {
- public:
- AudioBuffer(WebRtc_Word32 max_num_channels, WebRtc_Word32 samples_per_channel);
- virtual ~AudioBuffer();
-
- WebRtc_Word32 num_channels() const;
- WebRtc_Word32 samples_per_channel() const;
- WebRtc_Word32 samples_per_split_channel() const;
-
- WebRtc_Word16* data(WebRtc_Word32 channel) const;
- WebRtc_Word16* low_pass_split_data(WebRtc_Word32 channel) const;
- WebRtc_Word16* high_pass_split_data(WebRtc_Word32 channel) const;
- WebRtc_Word16* mixed_low_pass_data(WebRtc_Word32 channel) const;
- WebRtc_Word16* low_pass_reference(WebRtc_Word32 channel) const;
-
- WebRtc_Word32* analysis_filter_state1(WebRtc_Word32 channel) const;
- WebRtc_Word32* analysis_filter_state2(WebRtc_Word32 channel) const;
- WebRtc_Word32* synthesis_filter_state1(WebRtc_Word32 channel) const;
- WebRtc_Word32* synthesis_filter_state2(WebRtc_Word32 channel) const;
-
- void DeinterleaveFrom(AudioFrame* audioFrame);
- void InterleaveTo(AudioFrame* audioFrame) const;
- void Mix(WebRtc_Word32 num_mixed_channels);
- void CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels);
- void CopyLowPassToReference();
-
- private:
- const WebRtc_Word32 max_num_channels_;
- WebRtc_Word32 num_channels_;
- WebRtc_Word32 num_mixed_channels_;
- WebRtc_Word32 num_mixed_low_pass_channels_;
- const WebRtc_Word32 samples_per_channel_;
- WebRtc_Word32 samples_per_split_channel_;
- bool reference_copied_;
-
- WebRtc_Word16* data_;
- // TODO(ajm): Prefer to make these vectors if permitted...
- AudioChannel* channels_;
- SplitAudioChannel* split_channels_;
- // TODO(ajm): improve this, we don't need the full 32 kHz space here.
- AudioChannel* mixed_low_pass_channels_;
- AudioChannel* low_pass_reference_channels_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
diff --git a/src/modules/audio_processing/main/source/level_estimator_impl.cc b/src/modules/audio_processing/main/source/level_estimator_impl.cc
deleted file mode 100644
index 799a962..0000000
--- a/src/modules/audio_processing/main/source/level_estimator_impl.cc
+++ /dev/null
@@ -1,182 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "level_estimator_impl.h"
-
-#include <cassert>
-#include <cstring>
-
-#include "critical_section_wrapper.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-// TODO(ajm): implement the underlying level estimator component.
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-/*int EstimateLevel(AudioBuffer* audio, Handle* my_handle) {
- assert(audio->samples_per_split_channel() <= 160);
-
- WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
- if (audio->num_channels() > 1) {
- audio->CopyAndMixLowPass(1);
- mixed_data = audio->mixed_low_pass_data(0);
- }
-
- int err = UpdateLvlEst(my_handle,
- mixed_data,
- audio->samples_per_split_channel());
- if (err != AudioProcessing::kNoError) {
- return GetHandleError(my_handle);
- }
-
- return AudioProcessing::kNoError;
-}
-
-int GetMetricsLocal(Handle* my_handle, LevelEstimator::Metrics* metrics) {
- level_t levels;
- memset(&levels, 0, sizeof(levels));
-
- int err = ExportLevels(my_handle, &levels, 2);
- if (err != AudioProcessing::kNoError) {
- return err;
- }
- metrics->signal.instant = levels.instant;
- metrics->signal.average = levels.average;
- metrics->signal.maximum = levels.max;
- metrics->signal.minimum = levels.min;
-
- err = ExportLevels(my_handle, &levels, 1);
- if (err != AudioProcessing::kNoError) {
- return err;
- }
- metrics->speech.instant = levels.instant;
- metrics->speech.average = levels.average;
- metrics->speech.maximum = levels.max;
- metrics->speech.minimum = levels.min;
-
- err = ExportLevels(my_handle, &levels, 0);
- if (err != AudioProcessing::kNoError) {
- return err;
- }
- metrics->noise.instant = levels.instant;
- metrics->noise.average = levels.average;
- metrics->noise.maximum = levels.max;
- metrics->noise.minimum = levels.min;
-
- return AudioProcessing::kNoError;
-}*/
-} // namespace
-
-LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm) {}
-
-LevelEstimatorImpl::~LevelEstimatorImpl() {}
-
-int LevelEstimatorImpl::AnalyzeReverseStream(AudioBuffer* /*audio*/) {
- return apm_->kUnsupportedComponentError;
- /*if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- return EstimateLevel(audio, static_cast<Handle*>(handle(1)));*/
-}
-
-int LevelEstimatorImpl::ProcessCaptureAudio(AudioBuffer* /*audio*/) {
- return apm_->kUnsupportedComponentError;
- /*if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- return EstimateLevel(audio, static_cast<Handle*>(handle(0)));*/
-}
-
-int LevelEstimatorImpl::Enable(bool /*enable*/) {
- CriticalSectionScoped crit_scoped(*apm_->crit());
- return apm_->kUnsupportedComponentError;
- //return EnableComponent(enable);
-}
-
-bool LevelEstimatorImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int LevelEstimatorImpl::GetMetrics(LevelEstimator::Metrics* /*metrics*/,
- LevelEstimator::Metrics* /*reverse_metrics*/) {
- return apm_->kUnsupportedComponentError;
- /*if (!is_component_enabled()) {
- return apm_->kNotEnabledError;
- }
-
- int err = GetMetricsLocal(static_cast<Handle*>(handle(0)), metrics);
- if (err != apm_->kNoError) {
- return err;
- }
-
- err = GetMetricsLocal(static_cast<Handle*>(handle(1)), reverse_metrics);
- if (err != apm_->kNoError) {
- return err;
- }
-
- return apm_->kNoError;*/
-}
-
-int LevelEstimatorImpl::get_version(char* version,
- int version_len_bytes) const {
- // An empty string is used to indicate no version information.
- memset(version, 0, version_len_bytes);
- return apm_->kNoError;
-}
-
-void* LevelEstimatorImpl::CreateHandle() const {
- Handle* handle = NULL;
- /*if (CreateLvlEst(&handle) != apm_->kNoError) {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }*/
-
- return handle;
-}
-
-int LevelEstimatorImpl::DestroyHandle(void* /*handle*/) const {
- return apm_->kUnsupportedComponentError;
- //return FreeLvlEst(static_cast<Handle*>(handle));
-}
-
-int LevelEstimatorImpl::InitializeHandle(void* /*handle*/) const {
- return apm_->kUnsupportedComponentError;
- /*const double kIntervalSeconds = 1.5;
- return InitLvlEst(static_cast<Handle*>(handle),
- apm_->sample_rate_hz(),
- kIntervalSeconds);*/
-}
-
-int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
- return apm_->kUnsupportedComponentError;
- //return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::num_handles_required() const {
- return apm_->kUnsupportedComponentError;
- //return 2;
-}
-
-int LevelEstimatorImpl::GetHandleError(void* handle) const {
- // The component has no detailed errors.
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
diff --git a/src/modules/audio_processing/main/test/process_test/Android.mk b/src/modules/audio_processing/main/test/process_test/Android.mk
deleted file mode 100644
index 23080aa..0000000
--- a/src/modules/audio_processing/main/test/process_test/Android.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# apm test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- process_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-
-LOCAL_CPPFLAGS :=
-LOCAL_LDFLAGS :=
-LOCAL_C_INCLUDES := \
- external/gtest/include \
- $(LOCAL_PATH)/../../../../../system_wrappers/interface \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../../interface \
- $(LOCAL_PATH)/../../../../..
-
-LOCAL_STATIC_LIBRARIES := \
- libgtest
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libwebrtc_audio_preprocessing
-
-LOCAL_MODULE:= webrtc_apm_process_test
-
-include external/stlport/libstlport.mk
-include $(BUILD_EXECUTABLE)
diff --git a/src/modules/audio_processing/main/test/process_test/process_test.cc b/src/modules/audio_processing/main/test/process_test/process_test.cc
deleted file mode 100644
index c62345f..0000000
--- a/src/modules/audio_processing/main/test/process_test/process_test.cc
+++ /dev/null
@@ -1,628 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#ifdef WEBRTC_ANDROID
-#include <sys/stat.h>
-#endif
-
-#include "tick_util.h"
-#include "gtest/gtest.h"
-#include "module_common_types.h"
-
-#include "audio_processing.h"
-
-#include "cpu_features_wrapper.h"
-
-using webrtc::AudioFrame;
-using webrtc::TickInterval;
-using webrtc::TickTime;
-
-using webrtc::AudioProcessing;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-
-void usage() {
- printf(
- "Usage: process_test [options] [-ir REVERSE_FILE] [-i PRIMARY_FILE]\n");
- printf(
- " [-o OUT_FILE]\n");
- printf(
- "process_test is a test application for AudioProcessing.\n\n"
- "When -ir or -i is specified the files will be processed directly in a\n"
- "simulation mode. Otherwise the full set of test files is expected to be\n"
- "present in the working directory.\n");
- printf("\n");
- printf("Options\n");
- printf("General configuration:\n");
- printf(" -fs SAMPLE_RATE_HZ\n");
- printf(" -ch CHANNELS_IN CHANNELS_OUT\n");
- printf(" -rch REVERSE_CHANNELS\n");
- printf("\n");
- printf("Component configuration:\n");
- printf(
- "All components are disabled by default. Each block below begins with a\n"
- "flag to enable the component with default settings. The subsequent flags\n"
- "in the block are used to provide configuration settings.\n");
- printf("\n -aec Echo cancellation\n");
- printf(" --drift_compensation\n");
- printf(" --no_drift_compensation\n");
- printf("\n -aecm Echo control mobile\n");
- printf("\n -agc Gain control\n");
- printf(" --analog\n");
- printf(" --adaptive_digital\n");
- printf(" --fixed_digital\n");
- printf(" --target_level LEVEL\n");
- printf(" --compression_gain GAIN\n");
- printf(" --limiter\n");
- printf(" --no_limiter\n");
- printf("\n -hpf High pass filter\n");
- printf("\n -ns Noise suppression\n");
- printf(" --ns_low\n");
- printf(" --ns_moderate\n");
- printf(" --ns_high\n");
- printf(" --ns_very_high\n");
- printf("\n -vad Voice activity detection\n");
- printf(" --vad_out_file FILE");
- printf("\n");
- printf("Modifiers:\n");
- printf(" --perf Measure performance.\n");
- printf(" --quiet Suppress text output.\n");
- printf(" --no_progress Suppress progress.\n");
- printf(" --version Print version information and exit.\n");
-}
-
-// void function for gtest.
-void void_main(int argc, char* argv[]) {
- if (argc > 1 && strcmp(argv[1], "--help") == 0) {
- usage();
- return;
- }
-
- if (argc < 2) {
- printf("Did you mean to run without arguments?\n");
- printf("Try `process_test --help' for more information.\n\n");
- }
-
- AudioProcessing* apm = AudioProcessing::Create(0);
- ASSERT_TRUE(apm != NULL);
-
- WebRtc_Word8 version[1024];
- WebRtc_UWord32 version_bytes_remaining = sizeof(version);
- WebRtc_UWord32 version_position = 0;
-
- const char* far_filename = NULL;
- const char* near_filename = NULL;
- const char* out_filename = NULL;
- const char* vad_out_filename = NULL;
-
- int32_t sample_rate_hz = 16000;
- int32_t device_sample_rate_hz = 16000;
-
- int num_capture_input_channels = 1;
- int num_capture_output_channels = 1;
- int num_render_channels = 1;
-
- int samples_per_channel = sample_rate_hz / 100;
-
- bool simulating = false;
- bool perf_testing = false;
- bool verbose = true;
- bool progress = true;
- //bool interleaved = true;
-
- for (int i = 1; i < argc; i++) {
- if (strcmp(argv[i], "-ir") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -ir";
- far_filename = argv[i];
- simulating = true;
-
- } else if (strcmp(argv[i], "-i") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -i";
- near_filename = argv[i];
- simulating = true;
-
- } else if (strcmp(argv[i], "-o") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -o";
- out_filename = argv[i];
-
- } else if (strcmp(argv[i], "-fs") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify sample rate after -fs";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
- samples_per_channel = sample_rate_hz / 100;
-
- ASSERT_EQ(apm->kNoError,
- apm->set_sample_rate_hz(sample_rate_hz));
-
- } else if (strcmp(argv[i], "-ch") == 0) {
- i++;
- ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
- i++;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_num_channels(num_capture_input_channels,
- num_capture_output_channels));
-
- } else if (strcmp(argv[i], "-rch") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify number of channels after -rch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_num_reverse_channels(num_render_channels));
-
- } else if (strcmp(argv[i], "-aec") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
-
- } else if (strcmp(argv[i], "-noasm") == 0) {
- WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
-
- } else if (strcmp(argv[i], "--drift_compensation") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- // TODO(ajm): this is enabled in the VQE test app by default. Investigate
- // why it can give better performance despite passing zeros.
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_drift_compensation(true));
- } else if (strcmp(argv[i], "--no_drift_compensation") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_drift_compensation(false));
-
- } else if (strcmp(argv[i], "-aecm") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
-
- } else if (strcmp(argv[i], "-agc") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-
- } else if (strcmp(argv[i], "--analog") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
-
- } else if (strcmp(argv[i], "--adaptive_digital") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
-
- } else if (strcmp(argv[i], "--fixed_digital") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kFixedDigital));
-
- } else if (strcmp(argv[i], "--target_level") == 0) {
- i++;
- int level;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &level));
-
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_target_level_dbfs(level));
-
- } else if (strcmp(argv[i], "--compression_gain") == 0) {
- i++;
- int gain;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &gain));
-
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_compression_gain_db(gain));
-
- } else if (strcmp(argv[i], "--limiter") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->enable_limiter(true));
-
- } else if (strcmp(argv[i], "--no_limiter") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->enable_limiter(false));
-
- } else if (strcmp(argv[i], "-hpf") == 0) {
- ASSERT_EQ(apm->kNoError, apm->high_pass_filter()->Enable(true));
-
- } else if (strcmp(argv[i], "-ns") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-
- } else if (strcmp(argv[i], "--ns_low") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kLow));
-
- } else if (strcmp(argv[i], "--ns_moderate") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kModerate));
-
- } else if (strcmp(argv[i], "--ns_high") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kHigh));
-
- } else if (strcmp(argv[i], "--ns_very_high") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kVeryHigh));
-
- } else if (strcmp(argv[i], "-vad") == 0) {
- ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
-
- } else if (strcmp(argv[i], "--vad_out_file") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after --vad_out_file";
- vad_out_filename = argv[i];
-
- } else if (strcmp(argv[i], "--perf") == 0) {
- perf_testing = true;
-
- } else if (strcmp(argv[i], "--quiet") == 0) {
- verbose = false;
- progress = false;
-
- } else if (strcmp(argv[i], "--no_progress") == 0) {
- progress = false;
-
- } else if (strcmp(argv[i], "--version") == 0) {
- ASSERT_EQ(apm->kNoError, apm->Version(version,
- version_bytes_remaining,
- version_position));
- printf("%s\n", version);
- return;
-
- } else {
- FAIL() << "Unrecognized argument " << argv[i];
- }
- }
-
- if (verbose) {
- printf("Sample rate: %d Hz\n", sample_rate_hz);
- printf("Primary channels: %d (in), %d (out)\n",
- num_capture_input_channels,
- num_capture_output_channels);
- printf("Reverse channels: %d \n", num_render_channels);
- }
-
- const char far_file_default[] = "apm_far.pcm";
- const char near_file_default[] = "apm_near.pcm";
- const char out_file_default[] = "out.pcm";
- const char event_filename[] = "apm_event.dat";
- const char delay_filename[] = "apm_delay.dat";
- const char drift_filename[] = "apm_drift.dat";
- const char vad_file_default[] = "vad_out.dat";
-
- if (!simulating) {
- far_filename = far_file_default;
- near_filename = near_file_default;
- }
-
- if (out_filename == NULL) {
- out_filename = out_file_default;
- }
-
- if (vad_out_filename == NULL) {
- vad_out_filename = vad_file_default;
- }
-
- FILE* far_file = NULL;
- FILE* near_file = NULL;
- FILE* out_file = NULL;
- FILE* event_file = NULL;
- FILE* delay_file = NULL;
- FILE* drift_file = NULL;
- FILE* vad_out_file = NULL;
-
- if (far_filename != NULL) {
- far_file = fopen(far_filename, "rb");
- ASSERT_TRUE(NULL != far_file) << "Unable to open far-end audio file "
- << far_filename;
- }
-
- near_file = fopen(near_filename, "rb");
- ASSERT_TRUE(NULL != near_file) << "Unable to open near-end audio file "
- << near_filename;
- struct stat st;
- stat(near_filename, &st);
- int near_size_samples = st.st_size / sizeof(int16_t);
-
- out_file = fopen(out_filename, "wb");
- ASSERT_TRUE(NULL != out_file) << "Unable to open output audio file "
- << out_filename;
-
- if (!simulating) {
- event_file = fopen(event_filename, "rb");
- ASSERT_TRUE(NULL != event_file) << "Unable to open event file "
- << event_filename;
-
- delay_file = fopen(delay_filename, "rb");
- ASSERT_TRUE(NULL != delay_file) << "Unable to open buffer file "
- << delay_filename;
-
- drift_file = fopen(drift_filename, "rb");
- ASSERT_TRUE(NULL != drift_file) << "Unable to open drift file "
- << drift_filename;
- }
-
- if (apm->voice_detection()->is_enabled()) {
- vad_out_file = fopen(vad_out_filename, "wb");
- ASSERT_TRUE(NULL != vad_out_file) << "Unable to open VAD output file "
- << vad_out_file;
- }
-
- enum Events {
- kInitializeEvent,
- kRenderEvent,
- kCaptureEvent,
- kResetEventDeprecated
- };
- int16_t event = 0;
- size_t read_count = 0;
- int reverse_count = 0;
- int primary_count = 0;
- int near_read_samples = 0;
- TickInterval acc_ticks;
-
- AudioFrame far_frame;
- far_frame._frequencyInHz = sample_rate_hz;
-
- AudioFrame near_frame;
- near_frame._frequencyInHz = sample_rate_hz;
-
- int delay_ms = 0;
- int drift_samples = 0;
- int capture_level = 127;
- int8_t stream_has_voice = 0;
-
- TickTime t0 = TickTime::Now();
- TickTime t1 = t0;
- WebRtc_Word64 max_time_us = 0;
- WebRtc_Word64 max_time_reverse_us = 0;
- WebRtc_Word64 min_time_us = 1e6;
- WebRtc_Word64 min_time_reverse_us = 1e6;
-
- while (simulating || feof(event_file) == 0) {
- std::ostringstream trace_stream;
- trace_stream << "Processed frames: " << reverse_count << " (reverse), "
- << primary_count << " (primary)";
- SCOPED_TRACE(trace_stream.str());
-
-
- if (simulating) {
- if (far_file == NULL) {
- event = kCaptureEvent;
- } else {
- if (event == kRenderEvent) {
- event = kCaptureEvent;
- } else {
- event = kRenderEvent;
- }
- }
- } else {
- read_count = fread(&event, sizeof(event), 1, event_file);
- if (read_count != 1) {
- break;
- }
- //if (fread(&event, sizeof(event), 1, event_file) != 1) {
- // break; // This is expected.
- //}
- }
-
- if (event == kInitializeEvent || event == kResetEventDeprecated) {
- ASSERT_EQ(1u,
- fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file));
- samples_per_channel = sample_rate_hz / 100;
-
- ASSERT_EQ(1u,
- fread(&device_sample_rate_hz,
- sizeof(device_sample_rate_hz),
- 1,
- event_file));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_sample_rate_hz(sample_rate_hz));
-
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_device_sample_rate_hz(
- device_sample_rate_hz));
-
- far_frame._frequencyInHz = sample_rate_hz;
- near_frame._frequencyInHz = sample_rate_hz;
-
- if (verbose) {
- printf("Init at frame: %d (primary), %d (reverse)\n",
- primary_count, reverse_count);
- printf(" Sample rate: %d Hz\n", sample_rate_hz);
- }
-
- } else if (event == kRenderEvent) {
- reverse_count++;
- far_frame._audioChannel = num_render_channels;
- far_frame._payloadDataLengthInSamples =
- num_render_channels * samples_per_channel;
-
- read_count = fread(far_frame._payloadData,
- sizeof(WebRtc_Word16),
- far_frame._payloadDataLengthInSamples,
- far_file);
-
- if (simulating) {
- if (read_count != far_frame._payloadDataLengthInSamples) {
- break; // This is expected.
- }
- } else {
- ASSERT_EQ(read_count,
- far_frame._payloadDataLengthInSamples);
- }
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- ASSERT_EQ(apm->kNoError,
- apm->AnalyzeReverseStream(&far_frame));
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_reverse_us) {
- max_time_reverse_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_reverse_us) {
- min_time_reverse_us = tick_diff.Microseconds();
- }
- }
-
- } else if (event == kCaptureEvent) {
- primary_count++;
- near_frame._audioChannel = num_capture_input_channels;
- near_frame._payloadDataLengthInSamples =
- num_capture_input_channels * samples_per_channel;
-
- read_count = fread(near_frame._payloadData,
- sizeof(WebRtc_Word16),
- near_frame._payloadDataLengthInSamples,
- near_file);
-
- near_read_samples += read_count;
- if (progress && primary_count % 100 == 0) {
- printf("%.0f%% complete\r",
- (near_read_samples * 100.0) / near_size_samples);
- fflush(stdout);
- }
- if (simulating) {
- if (read_count != near_frame._payloadDataLengthInSamples) {
- break; // This is expected.
- }
-
- delay_ms = 0;
- drift_samples = 0;
- } else {
- ASSERT_EQ(read_count,
- near_frame._payloadDataLengthInSamples);
-
- // TODO(ajm): sizeof(delay_ms) for current files?
- ASSERT_EQ(1u,
- fread(&delay_ms, 2, 1, delay_file));
- ASSERT_EQ(1u,
- fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
- }
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- // TODO(ajm): fake an analog gain while simulating.
-
- int capture_level_in = capture_level;
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_stream_analog_level(capture_level));
- ASSERT_EQ(apm->kNoError,
- apm->set_stream_delay_ms(delay_ms));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
-
- int err = apm->ProcessStream(&near_frame);
- if (err == apm->kBadStreamParameterWarning) {
- printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
- }
- ASSERT_TRUE(err == apm->kNoError ||
- err == apm->kBadStreamParameterWarning);
-
- capture_level = apm->gain_control()->stream_analog_level();
-
- stream_has_voice =
- static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
- if (vad_out_file != NULL) {
- ASSERT_EQ(1u, fwrite(&stream_has_voice,
- sizeof(stream_has_voice),
- 1,
- vad_out_file));
- }
-
- if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
- ASSERT_EQ(capture_level_in, capture_level);
- }
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_us) {
- max_time_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_us) {
- min_time_us = tick_diff.Microseconds();
- }
- }
-
- ASSERT_EQ(near_frame._payloadDataLengthInSamples,
- fwrite(near_frame._payloadData,
- sizeof(WebRtc_Word16),
- near_frame._payloadDataLengthInSamples,
- out_file));
- }
- else {
- FAIL() << "Event " << event << " is unrecognized";
- }
- }
-
- if (verbose) {
- printf("\nProcessed frames: %d (primary), %d (reverse)\n",
- primary_count, reverse_count);
- }
-
- int8_t temp_int8;
- if (far_file != NULL) {
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, far_file);
- EXPECT_NE(0, feof(far_file)) << "Far-end file not fully processed";
- }
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, near_file);
- EXPECT_NE(0, feof(near_file)) << "Near-end file not fully processed";
-
- if (!simulating) {
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, event_file);
- EXPECT_NE(0, feof(event_file)) << "Event file not fully processed";
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, delay_file);
- EXPECT_NE(0, feof(delay_file)) << "Delay file not fully processed";
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, drift_file);
- EXPECT_NE(0, feof(drift_file)) << "Drift file not fully processed";
- }
-
- if (perf_testing) {
- if (primary_count > 0) {
- WebRtc_Word64 exec_time = acc_ticks.Milliseconds();
- printf("\nTotal time: %.3f s, file time: %.2f s\n",
- exec_time * 0.001, primary_count * 0.01);
- printf("Time per frame: %.3f ms (average), %.3f ms (max),"
- " %.3f ms (min)\n",
- (exec_time * 1.0) / primary_count,
- (max_time_us + max_time_reverse_us) / 1000.0,
- (min_time_us + min_time_reverse_us) / 1000.0);
- } else {
- printf("Warning: no capture frames\n");
- }
- }
-
- AudioProcessing::Destroy(apm);
- apm = NULL;
-}
-
-int main(int argc, char* argv[])
-{
- void_main(argc, argv);
-
- return 0;
-}
diff --git a/src/modules/audio_processing/main/test/unit_test/Android.mk b/src/modules/audio_processing/main/test/unit_test/Android.mk
deleted file mode 100644
index b2029cf..0000000
--- a/src/modules/audio_processing/main/test/unit_test/Android.mk
+++ /dev/null
@@ -1,49 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# apm test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- unit_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-
-LOCAL_CPPFLAGS :=
-LOCAL_LDFLAGS :=
-LOCAL_C_INCLUDES := \
- external/gtest/include \
- $(LOCAL_PATH)/../../../../../system_wrappers/interface \
- $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../../interface \
- $(LOCAL_PATH)/../../../../..
-
-LOCAL_STATIC_LIBRARIES := \
- libgtest
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libwebrtc_audio_preprocessing
-
-LOCAL_MODULE:= webrtc_apm_unit_test
-
-include external/stlport/libstlport.mk
-include $(BUILD_EXECUTABLE)
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc
deleted file mode 100644
index c82ffdb..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc
+++ /dev/null
@@ -1,1111 +0,0 @@
-// Generated by the protocol buffer compiler. DO NOT EDIT!
-
-#define INTERNAL_SUPPRESS_PROTOBUF_FIELD_DEPRECATION
-#include "audio_processing_unittest.pb.h"
-
-#include <algorithm>
-
-#include <google/protobuf/stubs/once.h>
-#include <google/protobuf/io/coded_stream.h>
-#include <google/protobuf/wire_format_lite_inl.h>
-// @@protoc_insertion_point(includes)
-
-namespace audio_processing_unittest {
-
-void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto() {
- delete Test::default_instance_;
- delete Test_Statistic::default_instance_;
- delete Test_EchoMetrics::default_instance_;
- delete OutputData::default_instance_;
-}
-
-void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto() {
- static bool already_here = false;
- if (already_here) return;
- already_here = true;
- GOOGLE_PROTOBUF_VERIFY_VERSION;
-
- Test::default_instance_ = new Test();
- Test_Statistic::default_instance_ = new Test_Statistic();
- Test_EchoMetrics::default_instance_ = new Test_EchoMetrics();
- OutputData::default_instance_ = new OutputData();
- Test::default_instance_->InitAsDefaultInstance();
- Test_Statistic::default_instance_->InitAsDefaultInstance();
- Test_EchoMetrics::default_instance_->InitAsDefaultInstance();
- OutputData::default_instance_->InitAsDefaultInstance();
- ::google::protobuf::internal::OnShutdown(&protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto);
-}
-
-// Force AddDescriptors() to be called at static initialization time.
-struct StaticDescriptorInitializer_audio_5fprocessing_5funittest_2eproto {
- StaticDescriptorInitializer_audio_5fprocessing_5funittest_2eproto() {
- protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
- }
-} static_descriptor_initializer_audio_5fprocessing_5funittest_2eproto_;
-
-
-// ===================================================================
-
-#ifndef _MSC_VER
-const int Test_Statistic::kInstantFieldNumber;
-const int Test_Statistic::kAverageFieldNumber;
-const int Test_Statistic::kMaximumFieldNumber;
-const int Test_Statistic::kMinimumFieldNumber;
-#endif // !_MSC_VER
-
-Test_Statistic::Test_Statistic()
- : ::google::protobuf::MessageLite() {
- SharedCtor();
-}
-
-void Test_Statistic::InitAsDefaultInstance() {
-}
-
-Test_Statistic::Test_Statistic(const Test_Statistic& from)
- : ::google::protobuf::MessageLite() {
- SharedCtor();
- MergeFrom(from);
-}
-
-void Test_Statistic::SharedCtor() {
- _cached_size_ = 0;
- instant_ = 0;
- average_ = 0;
- maximum_ = 0;
- minimum_ = 0;
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test_Statistic::~Test_Statistic() {
- SharedDtor();
-}
-
-void Test_Statistic::SharedDtor() {
- if (this != default_instance_) {
- }
-}
-
-void Test_Statistic::SetCachedSize(int size) const {
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test_Statistic& Test_Statistic::default_instance() {
- if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto(); return *default_instance_;
-}
-
-Test_Statistic* Test_Statistic::default_instance_ = NULL;
-
-Test_Statistic* Test_Statistic::New() const {
- return new Test_Statistic;
-}
-
-void Test_Statistic::Clear() {
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- instant_ = 0;
- average_ = 0;
- maximum_ = 0;
- minimum_ = 0;
- }
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test_Statistic::MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
- ::google::protobuf::uint32 tag;
- while ((tag = input->ReadTag()) != 0) {
- switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
- // optional int32 instant = 1;
- case 1: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &instant_)));
- set_has_instant();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(16)) goto parse_average;
- break;
- }
-
- // optional int32 average = 2;
- case 2: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_average:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &average_)));
- set_has_average();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(24)) goto parse_maximum;
- break;
- }
-
- // optional int32 maximum = 3;
- case 3: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_maximum:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &maximum_)));
- set_has_maximum();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(32)) goto parse_minimum;
- break;
- }
-
- // optional int32 minimum = 4;
- case 4: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_minimum:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &minimum_)));
- set_has_minimum();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectAtEnd()) return true;
- break;
- }
-
- default: {
- handle_uninterpreted:
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
- return true;
- }
- DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
- break;
- }
- }
- }
- return true;
-#undef DO_
-}
-
-void Test_Statistic::SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const {
- // optional int32 instant = 1;
- if (has_instant()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(1, this->instant(), output);
- }
-
- // optional int32 average = 2;
- if (has_average()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(2, this->average(), output);
- }
-
- // optional int32 maximum = 3;
- if (has_maximum()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(3, this->maximum(), output);
- }
-
- // optional int32 minimum = 4;
- if (has_minimum()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(4, this->minimum(), output);
- }
-
-}
-
-int Test_Statistic::ByteSize() const {
- int total_size = 0;
-
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- // optional int32 instant = 1;
- if (has_instant()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->instant());
- }
-
- // optional int32 average = 2;
- if (has_average()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->average());
- }
-
- // optional int32 maximum = 3;
- if (has_maximum()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->maximum());
- }
-
- // optional int32 minimum = 4;
- if (has_minimum()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->minimum());
- }
-
- }
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = total_size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
- return total_size;
-}
-
-void Test_Statistic::CheckTypeAndMergeFrom(
- const ::google::protobuf::MessageLite& from) {
- MergeFrom(*::google::protobuf::down_cast<const Test_Statistic*>(&from));
-}
-
-void Test_Statistic::MergeFrom(const Test_Statistic& from) {
- GOOGLE_CHECK_NE(&from, this);
- if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- if (from.has_instant()) {
- set_instant(from.instant());
- }
- if (from.has_average()) {
- set_average(from.average());
- }
- if (from.has_maximum()) {
- set_maximum(from.maximum());
- }
- if (from.has_minimum()) {
- set_minimum(from.minimum());
- }
- }
-}
-
-void Test_Statistic::CopyFrom(const Test_Statistic& from) {
- if (&from == this) return;
- Clear();
- MergeFrom(from);
-}
-
-bool Test_Statistic::IsInitialized() const {
-
- return true;
-}
-
-void Test_Statistic::Swap(Test_Statistic* other) {
- if (other != this) {
- std::swap(instant_, other->instant_);
- std::swap(average_, other->average_);
- std::swap(maximum_, other->maximum_);
- std::swap(minimum_, other->minimum_);
- std::swap(_has_bits_[0], other->_has_bits_[0]);
- std::swap(_cached_size_, other->_cached_size_);
- }
-}
-
-::std::string Test_Statistic::GetTypeName() const {
- return "audio_processing_unittest.Test.Statistic";
-}
-
-
-// -------------------------------------------------------------------
-
-#ifndef _MSC_VER
-const int Test_EchoMetrics::kResidualEchoReturnLossFieldNumber;
-const int Test_EchoMetrics::kEchoReturnLossFieldNumber;
-const int Test_EchoMetrics::kEchoReturnLossEnhancementFieldNumber;
-const int Test_EchoMetrics::kANlpFieldNumber;
-#endif // !_MSC_VER
-
-Test_EchoMetrics::Test_EchoMetrics()
- : ::google::protobuf::MessageLite() {
- SharedCtor();
-}
-
-void Test_EchoMetrics::InitAsDefaultInstance() {
- residualechoreturnloss_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
- echoreturnloss_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
- echoreturnlossenhancement_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
- anlp_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
-}
-
-Test_EchoMetrics::Test_EchoMetrics(const Test_EchoMetrics& from)
- : ::google::protobuf::MessageLite() {
- SharedCtor();
- MergeFrom(from);
-}
-
-void Test_EchoMetrics::SharedCtor() {
- _cached_size_ = 0;
- residualechoreturnloss_ = NULL;
- echoreturnloss_ = NULL;
- echoreturnlossenhancement_ = NULL;
- anlp_ = NULL;
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test_EchoMetrics::~Test_EchoMetrics() {
- SharedDtor();
-}
-
-void Test_EchoMetrics::SharedDtor() {
- if (this != default_instance_) {
- delete residualechoreturnloss_;
- delete echoreturnloss_;
- delete echoreturnlossenhancement_;
- delete anlp_;
- }
-}
-
-void Test_EchoMetrics::SetCachedSize(int size) const {
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test_EchoMetrics& Test_EchoMetrics::default_instance() {
- if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto(); return *default_instance_;
-}
-
-Test_EchoMetrics* Test_EchoMetrics::default_instance_ = NULL;
-
-Test_EchoMetrics* Test_EchoMetrics::New() const {
- return new Test_EchoMetrics;
-}
-
-void Test_EchoMetrics::Clear() {
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- if (has_residualechoreturnloss()) {
- if (residualechoreturnloss_ != NULL) residualechoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
- }
- if (has_echoreturnloss()) {
- if (echoreturnloss_ != NULL) echoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
- }
- if (has_echoreturnlossenhancement()) {
- if (echoreturnlossenhancement_ != NULL) echoreturnlossenhancement_->::audio_processing_unittest::Test_Statistic::Clear();
- }
- if (has_anlp()) {
- if (anlp_ != NULL) anlp_->::audio_processing_unittest::Test_Statistic::Clear();
- }
- }
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test_EchoMetrics::MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
- ::google::protobuf::uint32 tag;
- while ((tag = input->ReadTag()) != 0) {
- switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
- // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
- case 1: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, mutable_residualechoreturnloss()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(18)) goto parse_echoReturnLoss;
- break;
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
- case 2: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- parse_echoReturnLoss:
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, mutable_echoreturnloss()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(26)) goto parse_echoReturnLossEnhancement;
- break;
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
- case 3: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- parse_echoReturnLossEnhancement:
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, mutable_echoreturnlossenhancement()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(34)) goto parse_aNlp;
- break;
- }
-
- // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
- case 4: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- parse_aNlp:
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, mutable_anlp()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectAtEnd()) return true;
- break;
- }
-
- default: {
- handle_uninterpreted:
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
- return true;
- }
- DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
- break;
- }
- }
- }
- return true;
-#undef DO_
-}
-
-void Test_EchoMetrics::SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const {
- // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
- if (has_residualechoreturnloss()) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 1, this->residualechoreturnloss(), output);
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
- if (has_echoreturnloss()) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 2, this->echoreturnloss(), output);
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
- if (has_echoreturnlossenhancement()) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 3, this->echoreturnlossenhancement(), output);
- }
-
- // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
- if (has_anlp()) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 4, this->anlp(), output);
- }
-
-}
-
-int Test_EchoMetrics::ByteSize() const {
- int total_size = 0;
-
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
- if (has_residualechoreturnloss()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->residualechoreturnloss());
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
- if (has_echoreturnloss()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->echoreturnloss());
- }
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
- if (has_echoreturnlossenhancement()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->echoreturnlossenhancement());
- }
-
- // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
- if (has_anlp()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->anlp());
- }
-
- }
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = total_size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
- return total_size;
-}
-
-void Test_EchoMetrics::CheckTypeAndMergeFrom(
- const ::google::protobuf::MessageLite& from) {
- MergeFrom(*::google::protobuf::down_cast<const Test_EchoMetrics*>(&from));
-}
-
-void Test_EchoMetrics::MergeFrom(const Test_EchoMetrics& from) {
- GOOGLE_CHECK_NE(&from, this);
- if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- if (from.has_residualechoreturnloss()) {
- mutable_residualechoreturnloss()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.residualechoreturnloss());
- }
- if (from.has_echoreturnloss()) {
- mutable_echoreturnloss()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.echoreturnloss());
- }
- if (from.has_echoreturnlossenhancement()) {
- mutable_echoreturnlossenhancement()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.echoreturnlossenhancement());
- }
- if (from.has_anlp()) {
- mutable_anlp()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.anlp());
- }
- }
-}
-
-void Test_EchoMetrics::CopyFrom(const Test_EchoMetrics& from) {
- if (&from == this) return;
- Clear();
- MergeFrom(from);
-}
-
-bool Test_EchoMetrics::IsInitialized() const {
-
- return true;
-}
-
-void Test_EchoMetrics::Swap(Test_EchoMetrics* other) {
- if (other != this) {
- std::swap(residualechoreturnloss_, other->residualechoreturnloss_);
- std::swap(echoreturnloss_, other->echoreturnloss_);
- std::swap(echoreturnlossenhancement_, other->echoreturnlossenhancement_);
- std::swap(anlp_, other->anlp_);
- std::swap(_has_bits_[0], other->_has_bits_[0]);
- std::swap(_cached_size_, other->_cached_size_);
- }
-}
-
-::std::string Test_EchoMetrics::GetTypeName() const {
- return "audio_processing_unittest.Test.EchoMetrics";
-}
-
-
-// -------------------------------------------------------------------
-
-#ifndef _MSC_VER
-const int Test::kNumReverseChannelsFieldNumber;
-const int Test::kNumChannelsFieldNumber;
-const int Test::kSampleRateFieldNumber;
-const int Test::kHasEchoCountFieldNumber;
-const int Test::kHasVoiceCountFieldNumber;
-const int Test::kIsSaturatedCountFieldNumber;
-const int Test::kEchoMetricsFieldNumber;
-#endif // !_MSC_VER
-
-Test::Test()
- : ::google::protobuf::MessageLite() {
- SharedCtor();
-}
-
-void Test::InitAsDefaultInstance() {
- echometrics_ = const_cast< ::audio_processing_unittest::Test_EchoMetrics*>(&::audio_processing_unittest::Test_EchoMetrics::default_instance());
-}
-
-Test::Test(const Test& from)
- : ::google::protobuf::MessageLite() {
- SharedCtor();
- MergeFrom(from);
-}
-
-void Test::SharedCtor() {
- _cached_size_ = 0;
- numreversechannels_ = 0;
- numchannels_ = 0;
- samplerate_ = 0;
- hasechocount_ = 0;
- hasvoicecount_ = 0;
- issaturatedcount_ = 0;
- echometrics_ = NULL;
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test::~Test() {
- SharedDtor();
-}
-
-void Test::SharedDtor() {
- if (this != default_instance_) {
- delete echometrics_;
- }
-}
-
-void Test::SetCachedSize(int size) const {
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test& Test::default_instance() {
- if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto(); return *default_instance_;
-}
-
-Test* Test::default_instance_ = NULL;
-
-Test* Test::New() const {
- return new Test;
-}
-
-void Test::Clear() {
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- numreversechannels_ = 0;
- numchannels_ = 0;
- samplerate_ = 0;
- hasechocount_ = 0;
- hasvoicecount_ = 0;
- issaturatedcount_ = 0;
- if (has_echometrics()) {
- if (echometrics_ != NULL) echometrics_->::audio_processing_unittest::Test_EchoMetrics::Clear();
- }
- }
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test::MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
- ::google::protobuf::uint32 tag;
- while ((tag = input->ReadTag()) != 0) {
- switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
- // optional int32 numReverseChannels = 1;
- case 1: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &numreversechannels_)));
- set_has_numreversechannels();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(16)) goto parse_numChannels;
- break;
- }
-
- // optional int32 numChannels = 2;
- case 2: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_numChannels:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &numchannels_)));
- set_has_numchannels();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(24)) goto parse_sampleRate;
- break;
- }
-
- // optional int32 sampleRate = 3;
- case 3: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_sampleRate:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &samplerate_)));
- set_has_samplerate();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(32)) goto parse_hasEchoCount;
- break;
- }
-
- // optional int32 hasEchoCount = 4;
- case 4: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_hasEchoCount:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &hasechocount_)));
- set_has_hasechocount();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(40)) goto parse_hasVoiceCount;
- break;
- }
-
- // optional int32 hasVoiceCount = 5;
- case 5: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_hasVoiceCount:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &hasvoicecount_)));
- set_has_hasvoicecount();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(48)) goto parse_isSaturatedCount;
- break;
- }
-
- // optional int32 isSaturatedCount = 6;
- case 6: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
- parse_isSaturatedCount:
- DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
- ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
- input, &issaturatedcount_)));
- set_has_issaturatedcount();
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(58)) goto parse_echoMetrics;
- break;
- }
-
- // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
- case 7: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- parse_echoMetrics:
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, mutable_echometrics()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectAtEnd()) return true;
- break;
- }
-
- default: {
- handle_uninterpreted:
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
- return true;
- }
- DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
- break;
- }
- }
- }
- return true;
-#undef DO_
-}
-
-void Test::SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const {
- // optional int32 numReverseChannels = 1;
- if (has_numreversechannels()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(1, this->numreversechannels(), output);
- }
-
- // optional int32 numChannels = 2;
- if (has_numchannels()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(2, this->numchannels(), output);
- }
-
- // optional int32 sampleRate = 3;
- if (has_samplerate()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(3, this->samplerate(), output);
- }
-
- // optional int32 hasEchoCount = 4;
- if (has_hasechocount()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(4, this->hasechocount(), output);
- }
-
- // optional int32 hasVoiceCount = 5;
- if (has_hasvoicecount()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(5, this->hasvoicecount(), output);
- }
-
- // optional int32 isSaturatedCount = 6;
- if (has_issaturatedcount()) {
- ::google::protobuf::internal::WireFormatLite::WriteInt32(6, this->issaturatedcount(), output);
- }
-
- // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
- if (has_echometrics()) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 7, this->echometrics(), output);
- }
-
-}
-
-int Test::ByteSize() const {
- int total_size = 0;
-
- if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- // optional int32 numReverseChannels = 1;
- if (has_numreversechannels()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->numreversechannels());
- }
-
- // optional int32 numChannels = 2;
- if (has_numchannels()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->numchannels());
- }
-
- // optional int32 sampleRate = 3;
- if (has_samplerate()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->samplerate());
- }
-
- // optional int32 hasEchoCount = 4;
- if (has_hasechocount()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->hasechocount());
- }
-
- // optional int32 hasVoiceCount = 5;
- if (has_hasvoicecount()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->hasvoicecount());
- }
-
- // optional int32 isSaturatedCount = 6;
- if (has_issaturatedcount()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::Int32Size(
- this->issaturatedcount());
- }
-
- // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
- if (has_echometrics()) {
- total_size += 1 +
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->echometrics());
- }
-
- }
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = total_size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
- return total_size;
-}
-
-void Test::CheckTypeAndMergeFrom(
- const ::google::protobuf::MessageLite& from) {
- MergeFrom(*::google::protobuf::down_cast<const Test*>(&from));
-}
-
-void Test::MergeFrom(const Test& from) {
- GOOGLE_CHECK_NE(&from, this);
- if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
- if (from.has_numreversechannels()) {
- set_numreversechannels(from.numreversechannels());
- }
- if (from.has_numchannels()) {
- set_numchannels(from.numchannels());
- }
- if (from.has_samplerate()) {
- set_samplerate(from.samplerate());
- }
- if (from.has_hasechocount()) {
- set_hasechocount(from.hasechocount());
- }
- if (from.has_hasvoicecount()) {
- set_hasvoicecount(from.hasvoicecount());
- }
- if (from.has_issaturatedcount()) {
- set_issaturatedcount(from.issaturatedcount());
- }
- if (from.has_echometrics()) {
- mutable_echometrics()->::audio_processing_unittest::Test_EchoMetrics::MergeFrom(from.echometrics());
- }
- }
-}
-
-void Test::CopyFrom(const Test& from) {
- if (&from == this) return;
- Clear();
- MergeFrom(from);
-}
-
-bool Test::IsInitialized() const {
-
- return true;
-}
-
-void Test::Swap(Test* other) {
- if (other != this) {
- std::swap(numreversechannels_, other->numreversechannels_);
- std::swap(numchannels_, other->numchannels_);
- std::swap(samplerate_, other->samplerate_);
- std::swap(hasechocount_, other->hasechocount_);
- std::swap(hasvoicecount_, other->hasvoicecount_);
- std::swap(issaturatedcount_, other->issaturatedcount_);
- std::swap(echometrics_, other->echometrics_);
- std::swap(_has_bits_[0], other->_has_bits_[0]);
- std::swap(_cached_size_, other->_cached_size_);
- }
-}
-
-::std::string Test::GetTypeName() const {
- return "audio_processing_unittest.Test";
-}
-
-
-// ===================================================================
-
-#ifndef _MSC_VER
-const int OutputData::kTestFieldNumber;
-#endif // !_MSC_VER
-
-OutputData::OutputData()
- : ::google::protobuf::MessageLite() {
- SharedCtor();
-}
-
-void OutputData::InitAsDefaultInstance() {
-}
-
-OutputData::OutputData(const OutputData& from)
- : ::google::protobuf::MessageLite() {
- SharedCtor();
- MergeFrom(from);
-}
-
-void OutputData::SharedCtor() {
- _cached_size_ = 0;
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-OutputData::~OutputData() {
- SharedDtor();
-}
-
-void OutputData::SharedDtor() {
- if (this != default_instance_) {
- }
-}
-
-void OutputData::SetCachedSize(int size) const {
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const OutputData& OutputData::default_instance() {
- if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto(); return *default_instance_;
-}
-
-OutputData* OutputData::default_instance_ = NULL;
-
-OutputData* OutputData::New() const {
- return new OutputData;
-}
-
-void OutputData::Clear() {
- test_.Clear();
- ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool OutputData::MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
- ::google::protobuf::uint32 tag;
- while ((tag = input->ReadTag()) != 0) {
- switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
- // repeated .audio_processing_unittest.Test test = 1;
- case 1: {
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
- parse_test:
- DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
- input, add_test()));
- } else {
- goto handle_uninterpreted;
- }
- if (input->ExpectTag(10)) goto parse_test;
- if (input->ExpectAtEnd()) return true;
- break;
- }
-
- default: {
- handle_uninterpreted:
- if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
- ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
- return true;
- }
- DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
- break;
- }
- }
- }
- return true;
-#undef DO_
-}
-
-void OutputData::SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const {
- // repeated .audio_processing_unittest.Test test = 1;
- for (int i = 0; i < this->test_size(); i++) {
- ::google::protobuf::internal::WireFormatLite::WriteMessage(
- 1, this->test(i), output);
- }
-
-}
-
-int OutputData::ByteSize() const {
- int total_size = 0;
-
- // repeated .audio_processing_unittest.Test test = 1;
- total_size += 1 * this->test_size();
- for (int i = 0; i < this->test_size(); i++) {
- total_size +=
- ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
- this->test(i));
- }
-
- GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
- _cached_size_ = total_size;
- GOOGLE_SAFE_CONCURRENT_WRITES_END();
- return total_size;
-}
-
-void OutputData::CheckTypeAndMergeFrom(
- const ::google::protobuf::MessageLite& from) {
- MergeFrom(*::google::protobuf::down_cast<const OutputData*>(&from));
-}
-
-void OutputData::MergeFrom(const OutputData& from) {
- GOOGLE_CHECK_NE(&from, this);
- test_.MergeFrom(from.test_);
-}
-
-void OutputData::CopyFrom(const OutputData& from) {
- if (&from == this) return;
- Clear();
- MergeFrom(from);
-}
-
-bool OutputData::IsInitialized() const {
-
- return true;
-}
-
-void OutputData::Swap(OutputData* other) {
- if (other != this) {
- test_.Swap(&other->test_);
- std::swap(_has_bits_[0], other->_has_bits_[0]);
- std::swap(_cached_size_, other->_cached_size_);
- }
-}
-
-::std::string OutputData::GetTypeName() const {
- return "audio_processing_unittest.OutputData";
-}
-
-
-// @@protoc_insertion_point(namespace_scope)
-
-} // namespace audio_processing_unittest
-
-// @@protoc_insertion_point(global_scope)
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h
deleted file mode 100644
index 34c21b2..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h
+++ /dev/null
@@ -1,862 +0,0 @@
-// Generated by the protocol buffer compiler. DO NOT EDIT!
-// source: audio_processing_unittest.proto
-
-#ifndef PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
-#define PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
-
-#include <string>
-
-#include <google/protobuf/stubs/common.h>
-
-#if GOOGLE_PROTOBUF_VERSION < 2004000
-#error This file was generated by a newer version of protoc which is
-#error incompatible with your Protocol Buffer headers. Please update
-#error your headers.
-#endif
-#if 2004000 < GOOGLE_PROTOBUF_MIN_PROTOC_VERSION
-#error This file was generated by an older version of protoc which is
-#error incompatible with your Protocol Buffer headers. Please
-#error regenerate this file with a newer version of protoc.
-#endif
-
-#include <google/protobuf/generated_message_util.h>
-#include <google/protobuf/repeated_field.h>
-#include <google/protobuf/extension_set.h>
-// @@protoc_insertion_point(includes)
-
-namespace audio_processing_unittest {
-
-// Internal implementation detail -- do not call these.
-void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
-class Test;
-class Test_Statistic;
-class Test_EchoMetrics;
-class OutputData;
-
-// ===================================================================
-
-class Test_Statistic : public ::google::protobuf::MessageLite {
- public:
- Test_Statistic();
- virtual ~Test_Statistic();
-
- Test_Statistic(const Test_Statistic& from);
-
- inline Test_Statistic& operator=(const Test_Statistic& from) {
- CopyFrom(from);
- return *this;
- }
-
- static const Test_Statistic& default_instance();
-
- void Swap(Test_Statistic* other);
-
- // implements Message ----------------------------------------------
-
- Test_Statistic* New() const;
- void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
- void CopyFrom(const Test_Statistic& from);
- void MergeFrom(const Test_Statistic& from);
- void Clear();
- bool IsInitialized() const;
-
- int ByteSize() const;
- bool MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input);
- void SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const;
- int GetCachedSize() const { return _cached_size_; }
- private:
- void SharedCtor();
- void SharedDtor();
- void SetCachedSize(int size) const;
- public:
-
- ::std::string GetTypeName() const;
-
- // nested types ----------------------------------------------------
-
- // accessors -------------------------------------------------------
-
- // optional int32 instant = 1;
- inline bool has_instant() const;
- inline void clear_instant();
- static const int kInstantFieldNumber = 1;
- inline ::google::protobuf::int32 instant() const;
- inline void set_instant(::google::protobuf::int32 value);
-
- // optional int32 average = 2;
- inline bool has_average() const;
- inline void clear_average();
- static const int kAverageFieldNumber = 2;
- inline ::google::protobuf::int32 average() const;
- inline void set_average(::google::protobuf::int32 value);
-
- // optional int32 maximum = 3;
- inline bool has_maximum() const;
- inline void clear_maximum();
- static const int kMaximumFieldNumber = 3;
- inline ::google::protobuf::int32 maximum() const;
- inline void set_maximum(::google::protobuf::int32 value);
-
- // optional int32 minimum = 4;
- inline bool has_minimum() const;
- inline void clear_minimum();
- static const int kMinimumFieldNumber = 4;
- inline ::google::protobuf::int32 minimum() const;
- inline void set_minimum(::google::protobuf::int32 value);
-
- // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test.Statistic)
- private:
- inline void set_has_instant();
- inline void clear_has_instant();
- inline void set_has_average();
- inline void clear_has_average();
- inline void set_has_maximum();
- inline void clear_has_maximum();
- inline void set_has_minimum();
- inline void clear_has_minimum();
-
- ::google::protobuf::int32 instant_;
- ::google::protobuf::int32 average_;
- ::google::protobuf::int32 maximum_;
- ::google::protobuf::int32 minimum_;
-
- mutable int _cached_size_;
- ::google::protobuf::uint32 _has_bits_[(4 + 31) / 32];
-
- friend void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
- void InitAsDefaultInstance();
- static Test_Statistic* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class Test_EchoMetrics : public ::google::protobuf::MessageLite {
- public:
- Test_EchoMetrics();
- virtual ~Test_EchoMetrics();
-
- Test_EchoMetrics(const Test_EchoMetrics& from);
-
- inline Test_EchoMetrics& operator=(const Test_EchoMetrics& from) {
- CopyFrom(from);
- return *this;
- }
-
- static const Test_EchoMetrics& default_instance();
-
- void Swap(Test_EchoMetrics* other);
-
- // implements Message ----------------------------------------------
-
- Test_EchoMetrics* New() const;
- void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
- void CopyFrom(const Test_EchoMetrics& from);
- void MergeFrom(const Test_EchoMetrics& from);
- void Clear();
- bool IsInitialized() const;
-
- int ByteSize() const;
- bool MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input);
- void SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const;
- int GetCachedSize() const { return _cached_size_; }
- private:
- void SharedCtor();
- void SharedDtor();
- void SetCachedSize(int size) const;
- public:
-
- ::std::string GetTypeName() const;
-
- // nested types ----------------------------------------------------
-
- // accessors -------------------------------------------------------
-
- // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
- inline bool has_residualechoreturnloss() const;
- inline void clear_residualechoreturnloss();
- static const int kResidualEchoReturnLossFieldNumber = 1;
- inline const ::audio_processing_unittest::Test_Statistic& residualechoreturnloss() const;
- inline ::audio_processing_unittest::Test_Statistic* mutable_residualechoreturnloss();
- inline ::audio_processing_unittest::Test_Statistic* release_residualechoreturnloss();
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
- inline bool has_echoreturnloss() const;
- inline void clear_echoreturnloss();
- static const int kEchoReturnLossFieldNumber = 2;
- inline const ::audio_processing_unittest::Test_Statistic& echoreturnloss() const;
- inline ::audio_processing_unittest::Test_Statistic* mutable_echoreturnloss();
- inline ::audio_processing_unittest::Test_Statistic* release_echoreturnloss();
-
- // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
- inline bool has_echoreturnlossenhancement() const;
- inline void clear_echoreturnlossenhancement();
- static const int kEchoReturnLossEnhancementFieldNumber = 3;
- inline const ::audio_processing_unittest::Test_Statistic& echoreturnlossenhancement() const;
- inline ::audio_processing_unittest::Test_Statistic* mutable_echoreturnlossenhancement();
- inline ::audio_processing_unittest::Test_Statistic* release_echoreturnlossenhancement();
-
- // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
- inline bool has_anlp() const;
- inline void clear_anlp();
- static const int kANlpFieldNumber = 4;
- inline const ::audio_processing_unittest::Test_Statistic& anlp() const;
- inline ::audio_processing_unittest::Test_Statistic* mutable_anlp();
- inline ::audio_processing_unittest::Test_Statistic* release_anlp();
-
- // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test.EchoMetrics)
- private:
- inline void set_has_residualechoreturnloss();
- inline void clear_has_residualechoreturnloss();
- inline void set_has_echoreturnloss();
- inline void clear_has_echoreturnloss();
- inline void set_has_echoreturnlossenhancement();
- inline void clear_has_echoreturnlossenhancement();
- inline void set_has_anlp();
- inline void clear_has_anlp();
-
- ::audio_processing_unittest::Test_Statistic* residualechoreturnloss_;
- ::audio_processing_unittest::Test_Statistic* echoreturnloss_;
- ::audio_processing_unittest::Test_Statistic* echoreturnlossenhancement_;
- ::audio_processing_unittest::Test_Statistic* anlp_;
-
- mutable int _cached_size_;
- ::google::protobuf::uint32 _has_bits_[(4 + 31) / 32];
-
- friend void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
- void InitAsDefaultInstance();
- static Test_EchoMetrics* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class Test : public ::google::protobuf::MessageLite {
- public:
- Test();
- virtual ~Test();
-
- Test(const Test& from);
-
- inline Test& operator=(const Test& from) {
- CopyFrom(from);
- return *this;
- }
-
- static const Test& default_instance();
-
- void Swap(Test* other);
-
- // implements Message ----------------------------------------------
-
- Test* New() const;
- void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
- void CopyFrom(const Test& from);
- void MergeFrom(const Test& from);
- void Clear();
- bool IsInitialized() const;
-
- int ByteSize() const;
- bool MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input);
- void SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const;
- int GetCachedSize() const { return _cached_size_; }
- private:
- void SharedCtor();
- void SharedDtor();
- void SetCachedSize(int size) const;
- public:
-
- ::std::string GetTypeName() const;
-
- // nested types ----------------------------------------------------
-
- typedef Test_Statistic Statistic;
- typedef Test_EchoMetrics EchoMetrics;
-
- // accessors -------------------------------------------------------
-
- // optional int32 numReverseChannels = 1;
- inline bool has_numreversechannels() const;
- inline void clear_numreversechannels();
- static const int kNumReverseChannelsFieldNumber = 1;
- inline ::google::protobuf::int32 numreversechannels() const;
- inline void set_numreversechannels(::google::protobuf::int32 value);
-
- // optional int32 numChannels = 2;
- inline bool has_numchannels() const;
- inline void clear_numchannels();
- static const int kNumChannelsFieldNumber = 2;
- inline ::google::protobuf::int32 numchannels() const;
- inline void set_numchannels(::google::protobuf::int32 value);
-
- // optional int32 sampleRate = 3;
- inline bool has_samplerate() const;
- inline void clear_samplerate();
- static const int kSampleRateFieldNumber = 3;
- inline ::google::protobuf::int32 samplerate() const;
- inline void set_samplerate(::google::protobuf::int32 value);
-
- // optional int32 hasEchoCount = 4;
- inline bool has_hasechocount() const;
- inline void clear_hasechocount();
- static const int kHasEchoCountFieldNumber = 4;
- inline ::google::protobuf::int32 hasechocount() const;
- inline void set_hasechocount(::google::protobuf::int32 value);
-
- // optional int32 hasVoiceCount = 5;
- inline bool has_hasvoicecount() const;
- inline void clear_hasvoicecount();
- static const int kHasVoiceCountFieldNumber = 5;
- inline ::google::protobuf::int32 hasvoicecount() const;
- inline void set_hasvoicecount(::google::protobuf::int32 value);
-
- // optional int32 isSaturatedCount = 6;
- inline bool has_issaturatedcount() const;
- inline void clear_issaturatedcount();
- static const int kIsSaturatedCountFieldNumber = 6;
- inline ::google::protobuf::int32 issaturatedcount() const;
- inline void set_issaturatedcount(::google::protobuf::int32 value);
-
- // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
- inline bool has_echometrics() const;
- inline void clear_echometrics();
- static const int kEchoMetricsFieldNumber = 7;
- inline const ::audio_processing_unittest::Test_EchoMetrics& echometrics() const;
- inline ::audio_processing_unittest::Test_EchoMetrics* mutable_echometrics();
- inline ::audio_processing_unittest::Test_EchoMetrics* release_echometrics();
-
- // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test)
- private:
- inline void set_has_numreversechannels();
- inline void clear_has_numreversechannels();
- inline void set_has_numchannels();
- inline void clear_has_numchannels();
- inline void set_has_samplerate();
- inline void clear_has_samplerate();
- inline void set_has_hasechocount();
- inline void clear_has_hasechocount();
- inline void set_has_hasvoicecount();
- inline void clear_has_hasvoicecount();
- inline void set_has_issaturatedcount();
- inline void clear_has_issaturatedcount();
- inline void set_has_echometrics();
- inline void clear_has_echometrics();
-
- ::google::protobuf::int32 numreversechannels_;
- ::google::protobuf::int32 numchannels_;
- ::google::protobuf::int32 samplerate_;
- ::google::protobuf::int32 hasechocount_;
- ::google::protobuf::int32 hasvoicecount_;
- ::google::protobuf::int32 issaturatedcount_;
- ::audio_processing_unittest::Test_EchoMetrics* echometrics_;
-
- mutable int _cached_size_;
- ::google::protobuf::uint32 _has_bits_[(7 + 31) / 32];
-
- friend void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
- void InitAsDefaultInstance();
- static Test* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class OutputData : public ::google::protobuf::MessageLite {
- public:
- OutputData();
- virtual ~OutputData();
-
- OutputData(const OutputData& from);
-
- inline OutputData& operator=(const OutputData& from) {
- CopyFrom(from);
- return *this;
- }
-
- static const OutputData& default_instance();
-
- void Swap(OutputData* other);
-
- // implements Message ----------------------------------------------
-
- OutputData* New() const;
- void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
- void CopyFrom(const OutputData& from);
- void MergeFrom(const OutputData& from);
- void Clear();
- bool IsInitialized() const;
-
- int ByteSize() const;
- bool MergePartialFromCodedStream(
- ::google::protobuf::io::CodedInputStream* input);
- void SerializeWithCachedSizes(
- ::google::protobuf::io::CodedOutputStream* output) const;
- int GetCachedSize() const { return _cached_size_; }
- private:
- void SharedCtor();
- void SharedDtor();
- void SetCachedSize(int size) const;
- public:
-
- ::std::string GetTypeName() const;
-
- // nested types ----------------------------------------------------
-
- // accessors -------------------------------------------------------
-
- // repeated .audio_processing_unittest.Test test = 1;
- inline int test_size() const;
- inline void clear_test();
- static const int kTestFieldNumber = 1;
- inline const ::audio_processing_unittest::Test& test(int index) const;
- inline ::audio_processing_unittest::Test* mutable_test(int index);
- inline ::audio_processing_unittest::Test* add_test();
- inline const ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >&
- test() const;
- inline ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >*
- mutable_test();
-
- // @@protoc_insertion_point(class_scope:audio_processing_unittest.OutputData)
- private:
-
- ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test > test_;
-
- mutable int _cached_size_;
- ::google::protobuf::uint32 _has_bits_[(1 + 31) / 32];
-
- friend void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
- friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
- void InitAsDefaultInstance();
- static OutputData* default_instance_;
-};
-// ===================================================================
-
-
-// ===================================================================
-
-// Test_Statistic
-
-// optional int32 instant = 1;
-inline bool Test_Statistic::has_instant() const {
- return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test_Statistic::set_has_instant() {
- _has_bits_[0] |= 0x00000001u;
-}
-inline void Test_Statistic::clear_has_instant() {
- _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test_Statistic::clear_instant() {
- instant_ = 0;
- clear_has_instant();
-}
-inline ::google::protobuf::int32 Test_Statistic::instant() const {
- return instant_;
-}
-inline void Test_Statistic::set_instant(::google::protobuf::int32 value) {
- set_has_instant();
- instant_ = value;
-}
-
-// optional int32 average = 2;
-inline bool Test_Statistic::has_average() const {
- return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test_Statistic::set_has_average() {
- _has_bits_[0] |= 0x00000002u;
-}
-inline void Test_Statistic::clear_has_average() {
- _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test_Statistic::clear_average() {
- average_ = 0;
- clear_has_average();
-}
-inline ::google::protobuf::int32 Test_Statistic::average() const {
- return average_;
-}
-inline void Test_Statistic::set_average(::google::protobuf::int32 value) {
- set_has_average();
- average_ = value;
-}
-
-// optional int32 maximum = 3;
-inline bool Test_Statistic::has_maximum() const {
- return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test_Statistic::set_has_maximum() {
- _has_bits_[0] |= 0x00000004u;
-}
-inline void Test_Statistic::clear_has_maximum() {
- _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test_Statistic::clear_maximum() {
- maximum_ = 0;
- clear_has_maximum();
-}
-inline ::google::protobuf::int32 Test_Statistic::maximum() const {
- return maximum_;
-}
-inline void Test_Statistic::set_maximum(::google::protobuf::int32 value) {
- set_has_maximum();
- maximum_ = value;
-}
-
-// optional int32 minimum = 4;
-inline bool Test_Statistic::has_minimum() const {
- return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test_Statistic::set_has_minimum() {
- _has_bits_[0] |= 0x00000008u;
-}
-inline void Test_Statistic::clear_has_minimum() {
- _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test_Statistic::clear_minimum() {
- minimum_ = 0;
- clear_has_minimum();
-}
-inline ::google::protobuf::int32 Test_Statistic::minimum() const {
- return minimum_;
-}
-inline void Test_Statistic::set_minimum(::google::protobuf::int32 value) {
- set_has_minimum();
- minimum_ = value;
-}
-
-// -------------------------------------------------------------------
-
-// Test_EchoMetrics
-
-// optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-inline bool Test_EchoMetrics::has_residualechoreturnloss() const {
- return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test_EchoMetrics::set_has_residualechoreturnloss() {
- _has_bits_[0] |= 0x00000001u;
-}
-inline void Test_EchoMetrics::clear_has_residualechoreturnloss() {
- _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test_EchoMetrics::clear_residualechoreturnloss() {
- if (residualechoreturnloss_ != NULL) residualechoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
- clear_has_residualechoreturnloss();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::residualechoreturnloss() const {
- return residualechoreturnloss_ != NULL ? *residualechoreturnloss_ : *default_instance_->residualechoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_residualechoreturnloss() {
- set_has_residualechoreturnloss();
- if (residualechoreturnloss_ == NULL) residualechoreturnloss_ = new ::audio_processing_unittest::Test_Statistic;
- return residualechoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_residualechoreturnloss() {
- clear_has_residualechoreturnloss();
- ::audio_processing_unittest::Test_Statistic* temp = residualechoreturnloss_;
- residualechoreturnloss_ = NULL;
- return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-inline bool Test_EchoMetrics::has_echoreturnloss() const {
- return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test_EchoMetrics::set_has_echoreturnloss() {
- _has_bits_[0] |= 0x00000002u;
-}
-inline void Test_EchoMetrics::clear_has_echoreturnloss() {
- _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test_EchoMetrics::clear_echoreturnloss() {
- if (echoreturnloss_ != NULL) echoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
- clear_has_echoreturnloss();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::echoreturnloss() const {
- return echoreturnloss_ != NULL ? *echoreturnloss_ : *default_instance_->echoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_echoreturnloss() {
- set_has_echoreturnloss();
- if (echoreturnloss_ == NULL) echoreturnloss_ = new ::audio_processing_unittest::Test_Statistic;
- return echoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_echoreturnloss() {
- clear_has_echoreturnloss();
- ::audio_processing_unittest::Test_Statistic* temp = echoreturnloss_;
- echoreturnloss_ = NULL;
- return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-inline bool Test_EchoMetrics::has_echoreturnlossenhancement() const {
- return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test_EchoMetrics::set_has_echoreturnlossenhancement() {
- _has_bits_[0] |= 0x00000004u;
-}
-inline void Test_EchoMetrics::clear_has_echoreturnlossenhancement() {
- _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test_EchoMetrics::clear_echoreturnlossenhancement() {
- if (echoreturnlossenhancement_ != NULL) echoreturnlossenhancement_->::audio_processing_unittest::Test_Statistic::Clear();
- clear_has_echoreturnlossenhancement();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::echoreturnlossenhancement() const {
- return echoreturnlossenhancement_ != NULL ? *echoreturnlossenhancement_ : *default_instance_->echoreturnlossenhancement_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_echoreturnlossenhancement() {
- set_has_echoreturnlossenhancement();
- if (echoreturnlossenhancement_ == NULL) echoreturnlossenhancement_ = new ::audio_processing_unittest::Test_Statistic;
- return echoreturnlossenhancement_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_echoreturnlossenhancement() {
- clear_has_echoreturnlossenhancement();
- ::audio_processing_unittest::Test_Statistic* temp = echoreturnlossenhancement_;
- echoreturnlossenhancement_ = NULL;
- return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-inline bool Test_EchoMetrics::has_anlp() const {
- return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test_EchoMetrics::set_has_anlp() {
- _has_bits_[0] |= 0x00000008u;
-}
-inline void Test_EchoMetrics::clear_has_anlp() {
- _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test_EchoMetrics::clear_anlp() {
- if (anlp_ != NULL) anlp_->::audio_processing_unittest::Test_Statistic::Clear();
- clear_has_anlp();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::anlp() const {
- return anlp_ != NULL ? *anlp_ : *default_instance_->anlp_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_anlp() {
- set_has_anlp();
- if (anlp_ == NULL) anlp_ = new ::audio_processing_unittest::Test_Statistic;
- return anlp_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_anlp() {
- clear_has_anlp();
- ::audio_processing_unittest::Test_Statistic* temp = anlp_;
- anlp_ = NULL;
- return temp;
-}
-
-// -------------------------------------------------------------------
-
-// Test
-
-// optional int32 numReverseChannels = 1;
-inline bool Test::has_numreversechannels() const {
- return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test::set_has_numreversechannels() {
- _has_bits_[0] |= 0x00000001u;
-}
-inline void Test::clear_has_numreversechannels() {
- _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test::clear_numreversechannels() {
- numreversechannels_ = 0;
- clear_has_numreversechannels();
-}
-inline ::google::protobuf::int32 Test::numreversechannels() const {
- return numreversechannels_;
-}
-inline void Test::set_numreversechannels(::google::protobuf::int32 value) {
- set_has_numreversechannels();
- numreversechannels_ = value;
-}
-
-// optional int32 numChannels = 2;
-inline bool Test::has_numchannels() const {
- return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test::set_has_numchannels() {
- _has_bits_[0] |= 0x00000002u;
-}
-inline void Test::clear_has_numchannels() {
- _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test::clear_numchannels() {
- numchannels_ = 0;
- clear_has_numchannels();
-}
-inline ::google::protobuf::int32 Test::numchannels() const {
- return numchannels_;
-}
-inline void Test::set_numchannels(::google::protobuf::int32 value) {
- set_has_numchannels();
- numchannels_ = value;
-}
-
-// optional int32 sampleRate = 3;
-inline bool Test::has_samplerate() const {
- return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test::set_has_samplerate() {
- _has_bits_[0] |= 0x00000004u;
-}
-inline void Test::clear_has_samplerate() {
- _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test::clear_samplerate() {
- samplerate_ = 0;
- clear_has_samplerate();
-}
-inline ::google::protobuf::int32 Test::samplerate() const {
- return samplerate_;
-}
-inline void Test::set_samplerate(::google::protobuf::int32 value) {
- set_has_samplerate();
- samplerate_ = value;
-}
-
-// optional int32 hasEchoCount = 4;
-inline bool Test::has_hasechocount() const {
- return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test::set_has_hasechocount() {
- _has_bits_[0] |= 0x00000008u;
-}
-inline void Test::clear_has_hasechocount() {
- _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test::clear_hasechocount() {
- hasechocount_ = 0;
- clear_has_hasechocount();
-}
-inline ::google::protobuf::int32 Test::hasechocount() const {
- return hasechocount_;
-}
-inline void Test::set_hasechocount(::google::protobuf::int32 value) {
- set_has_hasechocount();
- hasechocount_ = value;
-}
-
-// optional int32 hasVoiceCount = 5;
-inline bool Test::has_hasvoicecount() const {
- return (_has_bits_[0] & 0x00000010u) != 0;
-}
-inline void Test::set_has_hasvoicecount() {
- _has_bits_[0] |= 0x00000010u;
-}
-inline void Test::clear_has_hasvoicecount() {
- _has_bits_[0] &= ~0x00000010u;
-}
-inline void Test::clear_hasvoicecount() {
- hasvoicecount_ = 0;
- clear_has_hasvoicecount();
-}
-inline ::google::protobuf::int32 Test::hasvoicecount() const {
- return hasvoicecount_;
-}
-inline void Test::set_hasvoicecount(::google::protobuf::int32 value) {
- set_has_hasvoicecount();
- hasvoicecount_ = value;
-}
-
-// optional int32 isSaturatedCount = 6;
-inline bool Test::has_issaturatedcount() const {
- return (_has_bits_[0] & 0x00000020u) != 0;
-}
-inline void Test::set_has_issaturatedcount() {
- _has_bits_[0] |= 0x00000020u;
-}
-inline void Test::clear_has_issaturatedcount() {
- _has_bits_[0] &= ~0x00000020u;
-}
-inline void Test::clear_issaturatedcount() {
- issaturatedcount_ = 0;
- clear_has_issaturatedcount();
-}
-inline ::google::protobuf::int32 Test::issaturatedcount() const {
- return issaturatedcount_;
-}
-inline void Test::set_issaturatedcount(::google::protobuf::int32 value) {
- set_has_issaturatedcount();
- issaturatedcount_ = value;
-}
-
-// optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-inline bool Test::has_echometrics() const {
- return (_has_bits_[0] & 0x00000040u) != 0;
-}
-inline void Test::set_has_echometrics() {
- _has_bits_[0] |= 0x00000040u;
-}
-inline void Test::clear_has_echometrics() {
- _has_bits_[0] &= ~0x00000040u;
-}
-inline void Test::clear_echometrics() {
- if (echometrics_ != NULL) echometrics_->::audio_processing_unittest::Test_EchoMetrics::Clear();
- clear_has_echometrics();
-}
-inline const ::audio_processing_unittest::Test_EchoMetrics& Test::echometrics() const {
- return echometrics_ != NULL ? *echometrics_ : *default_instance_->echometrics_;
-}
-inline ::audio_processing_unittest::Test_EchoMetrics* Test::mutable_echometrics() {
- set_has_echometrics();
- if (echometrics_ == NULL) echometrics_ = new ::audio_processing_unittest::Test_EchoMetrics;
- return echometrics_;
-}
-inline ::audio_processing_unittest::Test_EchoMetrics* Test::release_echometrics() {
- clear_has_echometrics();
- ::audio_processing_unittest::Test_EchoMetrics* temp = echometrics_;
- echometrics_ = NULL;
- return temp;
-}
-
-// -------------------------------------------------------------------
-
-// OutputData
-
-// repeated .audio_processing_unittest.Test test = 1;
-inline int OutputData::test_size() const {
- return test_.size();
-}
-inline void OutputData::clear_test() {
- test_.Clear();
-}
-inline const ::audio_processing_unittest::Test& OutputData::test(int index) const {
- return test_.Get(index);
-}
-inline ::audio_processing_unittest::Test* OutputData::mutable_test(int index) {
- return test_.Mutable(index);
-}
-inline ::audio_processing_unittest::Test* OutputData::add_test() {
- return test_.Add();
-}
-inline const ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >&
-OutputData::test() const {
- return test_;
-}
-inline ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >*
-OutputData::mutable_test() {
- return &test_;
-}
-
-
-// @@protoc_insertion_point(namespace_scope)
-
-} // namespace audio_processing_unittest
-
-// @@protoc_insertion_point(global_scope)
-
-#endif // PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto
deleted file mode 100644
index 8520e64..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto
+++ /dev/null
@@ -1,33 +0,0 @@
-package audio_processing_unittest;
-option optimize_for = LITE_RUNTIME;
-
-message Test {
- optional int32 numReverseChannels = 1;
- optional int32 numChannels = 2;
- optional int32 sampleRate = 3;
-
- optional int32 hasEchoCount = 4;
- optional int32 hasVoiceCount = 5;
- optional int32 isSaturatedCount = 6;
-
- message Statistic {
- optional int32 instant = 1;
- optional int32 average = 2;
- optional int32 maximum = 3;
- optional int32 minimum = 4;
- }
-
- message EchoMetrics {
- optional Statistic residualEchoReturnLoss = 1;
- optional Statistic echoReturnLoss = 2;
- optional Statistic echoReturnLossEnhancement = 3;
- optional Statistic aNlp = 4;
- }
-
- optional EchoMetrics echoMetrics = 7;
-}
-
-message OutputData {
- repeated Test test = 1;
-}
-
diff --git a/src/modules/audio_processing/main/test/unit_test/unit_test.cc b/src/modules/audio_processing/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 3a6fce5..0000000
--- a/src/modules/audio_processing/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,881 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-
-#include <gtest/gtest.h>
-
-#include "audio_processing.h"
-#include "audio_processing_unittest.pb.h"
-#include "event_wrapper.h"
-#include "module_common_types.h"
-#include "thread_wrapper.h"
-#include "trace.h"
-#include "signal_processing_library.h"
-
-using webrtc::AudioProcessing;
-using webrtc::AudioFrame;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-using webrtc::EchoCancellation;
-using webrtc::EventWrapper;
-using webrtc::Trace;
-using webrtc::LevelEstimator;
-using webrtc::EchoCancellation;
-using webrtc::EchoControlMobile;
-using webrtc::VoiceDetection;
-
-namespace {
-// When true, this will compare the output data with the results stored to
-// file. This is the typical case. When the file should be updated, it can
-// be set to false with the command-line switch --write_output_data.
-bool global_read_output_data = true;
-
-class ApmEnvironment : public ::testing::Environment {
- public:
- virtual void SetUp() {
- Trace::CreateTrace();
- ASSERT_EQ(0, Trace::SetTraceFile("apm_trace.txt"));
- }
-
- virtual void TearDown() {
- Trace::ReturnTrace();
- }
-};
-
-class ApmTest : public ::testing::Test {
- protected:
- ApmTest();
- virtual void SetUp();
- virtual void TearDown();
-
- webrtc::AudioProcessing* apm_;
- webrtc::AudioFrame* frame_;
- webrtc::AudioFrame* revframe_;
- FILE* far_file_;
- FILE* near_file_;
- bool update_output_data_;
-};
-
-ApmTest::ApmTest()
- : apm_(NULL),
- far_file_(NULL),
- near_file_(NULL),
- frame_(NULL),
- revframe_(NULL) {}
-
-void ApmTest::SetUp() {
- apm_ = AudioProcessing::Create(0);
- ASSERT_TRUE(apm_ != NULL);
-
- frame_ = new AudioFrame();
- revframe_ = new AudioFrame();
-
- ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
-
- frame_->_payloadDataLengthInSamples = 320;
- frame_->_audioChannel = 2;
- frame_->_frequencyInHz = 32000;
- revframe_->_payloadDataLengthInSamples = 320;
- revframe_->_audioChannel = 2;
- revframe_->_frequencyInHz = 32000;
-
- far_file_ = fopen("aec_far.pcm", "rb");
- ASSERT_TRUE(far_file_ != NULL) << "Could not open input file aec_far.pcm\n";
- near_file_ = fopen("aec_near.pcm", "rb");
- ASSERT_TRUE(near_file_ != NULL) << "Could not open input file aec_near.pcm\n";
-}
-
-void ApmTest::TearDown() {
- if (frame_) {
- delete frame_;
- }
- frame_ = NULL;
-
- if (revframe_) {
- delete revframe_;
- }
- revframe_ = NULL;
-
- if (far_file_) {
- ASSERT_EQ(0, fclose(far_file_));
- }
- far_file_ = NULL;
-
- if (near_file_) {
- ASSERT_EQ(0, fclose(near_file_));
- }
- near_file_ = NULL;
-
- if (apm_ != NULL) {
- AudioProcessing::Destroy(apm_);
- }
- apm_ = NULL;
-}
-
-void MixStereoToMono(WebRtc_Word16* stereo,
- WebRtc_Word16* mono,
- int numSamples) {
- for (int i = 0; i < numSamples; i++) {
- int int32 = (static_cast<int>(stereo[i * 2]) +
- static_cast<int>(stereo[i * 2 + 1])) >> 1;
- mono[i] = static_cast<WebRtc_Word16>(int32);
- }
-}
-
-void WriteMessageLiteToFile(const char* filename,
- const ::google::protobuf::MessageLite& message) {
- assert(filename != NULL);
-
- FILE* file = fopen(filename, "wb");
- ASSERT_TRUE(file != NULL) << "Could not open " << filename;
- int size = message.ByteSize();
- ASSERT_GT(size, 0);
- unsigned char* array = new unsigned char[size];
- ASSERT_TRUE(message.SerializeToArray(array, size));
-
- ASSERT_EQ(1, fwrite(&size, sizeof(int), 1, file));
- ASSERT_EQ(size, fwrite(array, sizeof(unsigned char), size, file));
-
- delete [] array;
- fclose(file);
-}
-
-void ReadMessageLiteFromFile(const char* filename,
- ::google::protobuf::MessageLite* message) {
- assert(filename != NULL);
- assert(message != NULL);
-
- FILE* file = fopen(filename, "rb");
- ASSERT_TRUE(file != NULL) << "Could not open " << filename;
- int size = 0;
- ASSERT_EQ(1, fread(&size, sizeof(int), 1, file));
- ASSERT_GT(size, 0);
- unsigned char* array = new unsigned char[size];
- ASSERT_EQ(size, fread(array, sizeof(unsigned char), size, file));
-
- ASSERT_TRUE(message->ParseFromArray(array, size));
-
- delete [] array;
- fclose(file);
-}
-
-struct ThreadData {
- ThreadData(int thread_num_, AudioProcessing* ap_)
- : thread_num(thread_num_),
- error(false),
- ap(ap_) {}
- int thread_num;
- bool error;
- AudioProcessing* ap;
-};
-
-// Don't use GTest here; non-thread-safe on Windows (as of 1.5.0).
-bool DeadlockProc(void* thread_object) {
- ThreadData* thread_data = static_cast<ThreadData*>(thread_object);
- AudioProcessing* ap = thread_data->ap;
- int err = ap->kNoError;
-
- AudioFrame primary_frame;
- AudioFrame reverse_frame;
- primary_frame._payloadDataLengthInSamples = 320;
- primary_frame._audioChannel = 2;
- primary_frame._frequencyInHz = 32000;
- reverse_frame._payloadDataLengthInSamples = 320;
- reverse_frame._audioChannel = 2;
- reverse_frame._frequencyInHz = 32000;
-
- ap->echo_cancellation()->Enable(true);
- ap->gain_control()->Enable(true);
- ap->high_pass_filter()->Enable(true);
- ap->level_estimator()->Enable(true);
- ap->noise_suppression()->Enable(true);
- ap->voice_detection()->Enable(true);
-
- if (thread_data->thread_num % 2 == 0) {
- err = ap->AnalyzeReverseStream(&reverse_frame);
- if (err != ap->kNoError) {
- printf("Error in AnalyzeReverseStream(): %d\n", err);
- thread_data->error = true;
- return false;
- }
- }
-
- if (thread_data->thread_num % 2 == 1) {
- ap->set_stream_delay_ms(0);
- ap->echo_cancellation()->set_stream_drift_samples(0);
- ap->gain_control()->set_stream_analog_level(0);
- err = ap->ProcessStream(&primary_frame);
- if (err == ap->kStreamParameterNotSetError) {
- printf("Expected kStreamParameterNotSetError in ProcessStream(): %d\n",
- err);
- } else if (err != ap->kNoError) {
- printf("Error in ProcessStream(): %d\n", err);
- thread_data->error = true;
- return false;
- }
- ap->gain_control()->stream_analog_level();
- }
-
- EventWrapper* event = EventWrapper::Create();
- event->Wait(1);
- delete event;
- event = NULL;
-
- return true;
-}
-
-/*TEST_F(ApmTest, Deadlock) {
- const int num_threads = 16;
- std::vector<ThreadWrapper*> threads(num_threads);
- std::vector<ThreadData*> thread_data(num_threads);
-
- ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
-
- for (int i = 0; i < num_threads; i++) {
- thread_data[i] = new ThreadData(i, apm_);
- threads[i] = ThreadWrapper::CreateThread(DeadlockProc,
- thread_data[i],
- kNormalPriority,
- 0);
- ASSERT_TRUE(threads[i] != NULL);
- unsigned int thread_id = 0;
- threads[i]->Start(thread_id);
- }
-
- EventWrapper* event = EventWrapper::Create();
- ASSERT_EQ(kEventTimeout, event->Wait(5000));
- delete event;
- event = NULL;
-
- for (int i = 0; i < num_threads; i++) {
- // This will return false if the thread has deadlocked.
- ASSERT_TRUE(threads[i]->Stop());
- ASSERT_FALSE(thread_data[i]->error);
- delete threads[i];
- threads[i] = NULL;
- delete thread_data[i];
- thread_data[i] = NULL;
- }
-}*/
-
-TEST_F(ApmTest, StreamParameters) {
- // No errors when the components are disabled.
- EXPECT_EQ(apm_->kNoError,
- apm_->ProcessStream(frame_));
-
- // Missing agc level
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-
- // Missing delay
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-
- // Missing drift
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
-
- // No stream parameters
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError,
- apm_->AnalyzeReverseStream(revframe_));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
-
- // All there
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-}
-
-TEST_F(ApmTest, Channels) {
- // Testing number of invalid channels
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
- // Testing number of valid channels
- for (int i = 1; i < 3; i++) {
- for (int j = 1; j < 3; j++) {
- if (j > i) {
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
- } else {
- EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
- EXPECT_EQ(j, apm_->num_output_channels());
- }
- }
- EXPECT_EQ(i, apm_->num_input_channels());
- EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
- EXPECT_EQ(i, apm_->num_reverse_channels());
- }
-}
-
-TEST_F(ApmTest, SampleRates) {
- // Testing invalid sample rates
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
- // Testing valid sample rates
- int fs[] = {8000, 16000, 32000};
- for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
- EXPECT_EQ(fs[i], apm_->sample_rate_hz());
- }
-}
-
-TEST_F(ApmTest, Process) {
- GOOGLE_PROTOBUF_VERIFY_VERSION;
- audio_processing_unittest::OutputData output_data;
-
- if (global_read_output_data) {
- ReadMessageLiteFromFile("output_data.pb", &output_data);
-
- } else {
- // We don't have a file; add the required tests to the protobuf.
- int rev_ch[] = {1, 2};
- int ch[] = {1, 2};
- int fs[] = {8000, 16000, 32000};
- for (size_t i = 0; i < sizeof(rev_ch) / sizeof(*rev_ch); i++) {
- for (size_t j = 0; j < sizeof(ch) / sizeof(*ch); j++) {
- for (size_t k = 0; k < sizeof(fs) / sizeof(*fs); k++) {
- audio_processing_unittest::Test* test = output_data.add_test();
- test->set_numreversechannels(rev_ch[i]);
- test->set_numchannels(ch[j]);
- test->set_samplerate(fs[k]);
- }
- }
- }
- }
-
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(true));
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(0, 255));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->high_pass_filter()->Enable(true));
-
- //EXPECT_EQ(apm_->kNoError,
- // apm_->level_estimator()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->noise_suppression()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->Enable(true));
-
- for (int i = 0; i < output_data.test_size(); i++) {
- printf("Running test %d of %d...\n", i + 1, output_data.test_size());
-
- audio_processing_unittest::Test* test = output_data.mutable_test(i);
- const int num_samples = test->samplerate() / 100;
- revframe_->_payloadDataLengthInSamples = num_samples;
- revframe_->_audioChannel = test->numreversechannels();
- revframe_->_frequencyInHz = test->samplerate();
- frame_->_payloadDataLengthInSamples = num_samples;
- frame_->_audioChannel = test->numchannels();
- frame_->_frequencyInHz = test->samplerate();
-
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(test->samplerate()));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(frame_->_audioChannel,
- frame_->_audioChannel));
- ASSERT_EQ(apm_->kNoError,
- apm_->set_num_reverse_channels(revframe_->_audioChannel));
-
-
- int has_echo_count = 0;
- int has_voice_count = 0;
- int is_saturated_count = 0;
-
- while (1) {
- WebRtc_Word16 temp_data[640];
- int analog_level = 127;
-
- // Read far-end frame
- size_t read_count = fread(temp_data,
- sizeof(WebRtc_Word16),
- num_samples * 2,
- far_file_);
- if (read_count != static_cast<size_t>(num_samples * 2)) {
- // Check that the file really ended.
- ASSERT_NE(0, feof(far_file_));
- break; // This is expected.
- }
-
- if (revframe_->_audioChannel == 1) {
- MixStereoToMono(temp_data, revframe_->_payloadData,
- revframe_->_payloadDataLengthInSamples);
- } else {
- memcpy(revframe_->_payloadData,
- &temp_data[0],
- sizeof(WebRtc_Word16) * read_count);
- }
-
- EXPECT_EQ(apm_->kNoError,
- apm_->AnalyzeReverseStream(revframe_));
-
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(analog_level));
-
- // Read near-end frame
- read_count = fread(temp_data,
- sizeof(WebRtc_Word16),
- num_samples * 2,
- near_file_);
- if (read_count != static_cast<size_t>(num_samples * 2)) {
- // Check that the file really ended.
- ASSERT_NE(0, feof(near_file_));
- break; // This is expected.
- }
-
- if (frame_->_audioChannel == 1) {
- MixStereoToMono(temp_data, frame_->_payloadData, num_samples);
- } else {
- memcpy(frame_->_payloadData,
- &temp_data[0],
- sizeof(WebRtc_Word16) * read_count);
- }
-
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-
- if (apm_->echo_cancellation()->stream_has_echo()) {
- has_echo_count++;
- }
-
- analog_level = apm_->gain_control()->stream_analog_level();
- if (apm_->gain_control()->stream_is_saturated()) {
- is_saturated_count++;
- }
- if (apm_->voice_detection()->stream_has_voice()) {
- has_voice_count++;
- }
- }
-
- //<-- Statistics -->
- //LevelEstimator::Metrics far_metrics;
- //LevelEstimator::Metrics near_metrics;
- //EchoCancellation::Metrics echo_metrics;
- //LevelEstimator::Metrics far_metrics_ref_;
- //LevelEstimator::Metrics near_metrics_ref_;
- //EchoCancellation::Metrics echo_metrics_ref_;
- //EXPECT_EQ(apm_->kNoError,
- // apm_->echo_cancellation()->GetMetrics(&echo_metrics));
- //EXPECT_EQ(apm_->kNoError,
- // apm_->level_estimator()->GetMetrics(&near_metrics,
-
- // TODO(ajm): check echo metrics and output audio.
- if (global_read_output_data) {
- EXPECT_EQ(has_echo_count,
- test->hasechocount());
- EXPECT_EQ(has_voice_count,
- test->hasvoicecount());
- EXPECT_EQ(is_saturated_count,
- test->issaturatedcount());
- } else {
- test->set_hasechocount(has_echo_count);
- test->set_hasvoicecount(has_voice_count);
- test->set_issaturatedcount(is_saturated_count);
- }
-
- rewind(far_file_);
- rewind(near_file_);
- }
-
- if (!global_read_output_data) {
- WriteMessageLiteToFile("output_data.pb", output_data);
- }
-
- google::protobuf::ShutdownProtobufLibrary();
-}
-
-TEST_F(ApmTest, EchoCancellation) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(false));
- EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(4000));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(100000));
-
- int rate[] = {16000, 44100, 48000};
- for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
- EXPECT_EQ(rate[i],
- apm_->echo_cancellation()->device_sample_rate_hz());
- }
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_suppression_level(
- static_cast<EchoCancellation::SuppressionLevel>(-1)));
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_suppression_level(
- static_cast<EchoCancellation::SuppressionLevel>(4)));
-
- EchoCancellation::SuppressionLevel level[] = {
- EchoCancellation::kLowSuppression,
- EchoCancellation::kModerateSuppression,
- EchoCancellation::kHighSuppression,
- };
- for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_suppression_level(level[i]));
- EXPECT_EQ(level[i],
- apm_->echo_cancellation()->suppression_level());
- }
-
- EchoCancellation::Metrics metrics;
- EXPECT_EQ(apm_->kNotEnabledError,
- apm_->echo_cancellation()->GetMetrics(&metrics));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(true));
- EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(false));
- EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
-
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
- EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
-}
-
-TEST_F(ApmTest, EchoControlMobile) {
- // AECM won't use super-wideband.
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
- // Turn AECM on (and AEC off)
- EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
- EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_control_mobile()->set_routing_mode(
- static_cast<EchoControlMobile::RoutingMode>(-1)));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_control_mobile()->set_routing_mode(
- static_cast<EchoControlMobile::RoutingMode>(5)));
-
- // Toggle routing modes
- EchoControlMobile::RoutingMode mode[] = {
- EchoControlMobile::kQuietEarpieceOrHeadset,
- EchoControlMobile::kEarpiece,
- EchoControlMobile::kLoudEarpiece,
- EchoControlMobile::kSpeakerphone,
- EchoControlMobile::kLoudSpeakerphone,
- };
- for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->set_routing_mode(mode[i]));
- EXPECT_EQ(mode[i],
- apm_->echo_control_mobile()->routing_mode());
- }
- // Turn comfort noise off/on
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->enable_comfort_noise(false));
- EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->enable_comfort_noise(true));
- EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
- // Turn AECM off
- EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
- EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
-}
-
-TEST_F(ApmTest, GainControl) {
- // Testing gain modes
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(-1)));
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(3)));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(
- apm_->gain_control()->mode()));
-
- GainControl::Mode mode[] = {
- GainControl::kAdaptiveAnalog,
- GainControl::kAdaptiveDigital,
- GainControl::kFixedDigital
- };
- for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(mode[i]));
- EXPECT_EQ(mode[i], apm_->gain_control()->mode());
- }
- // Testing invalid target levels
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_target_level_dbfs(-3));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_target_level_dbfs(-40));
- // Testing valid target levels
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(
- apm_->gain_control()->target_level_dbfs()));
-
- int level_dbfs[] = {0, 6, 31};
- for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
- EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
- }
-
- // Testing invalid compression gains
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_compression_gain_db(-1));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_compression_gain_db(100));
-
- // Testing valid compression gains
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(
- apm_->gain_control()->compression_gain_db()));
-
- int gain_db[] = {0, 10, 90};
- for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(gain_db[i]));
- EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
- }
-
- // Testing limiter off/on
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
- EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
- EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
-
- // Testing invalid level limits
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(-1, 512));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(100000, 512));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, -1));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, 100000));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, 255));
-
- // Testing valid level limits
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(
- apm_->gain_control()->analog_level_minimum(),
- apm_->gain_control()->analog_level_maximum()));
-
- int min_level[] = {0, 255, 1024};
- for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
- EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
- }
-
- int max_level[] = {0, 1024, 65535};
- for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
- EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
- }
-
- // TODO(ajm): stream_is_saturated() and stream_analog_level()
-
- // Turn AGC off
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
- EXPECT_FALSE(apm_->gain_control()->is_enabled());
-}
-
-TEST_F(ApmTest, NoiseSuppression) {
- // Tesing invalid suppression levels
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->noise_suppression()->set_level(
- static_cast<NoiseSuppression::Level>(-1)));
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->noise_suppression()->set_level(
- static_cast<NoiseSuppression::Level>(5)));
-
- // Tesing valid suppression levels
- NoiseSuppression::Level level[] = {
- NoiseSuppression::kLow,
- NoiseSuppression::kModerate,
- NoiseSuppression::kHigh,
- NoiseSuppression::kVeryHigh
- };
- for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->noise_suppression()->set_level(level[i]));
- EXPECT_EQ(level[i], apm_->noise_suppression()->level());
- }
-
- // Turing NS on/off
- EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
- EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
- EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
-}
-
-TEST_F(ApmTest, HighPassFilter) {
- // Turing HP filter on/off
- EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
- EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
- EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
-}
-
-TEST_F(ApmTest, LevelEstimator) {
- // Turing Level estimator on/off
- EXPECT_EQ(apm_->kUnsupportedComponentError,
- apm_->level_estimator()->Enable(true));
- EXPECT_FALSE(apm_->level_estimator()->is_enabled());
- EXPECT_EQ(apm_->kUnsupportedComponentError,
- apm_->level_estimator()->Enable(false));
- EXPECT_FALSE(apm_->level_estimator()->is_enabled());
-}
-
-TEST_F(ApmTest, VoiceDetection) {
- // Test external VAD
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_stream_has_voice(true));
- EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_stream_has_voice(false));
- EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
-
- // Tesing invalid likelihoods
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->voice_detection()->set_likelihood(
- static_cast<VoiceDetection::Likelihood>(-1)));
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->voice_detection()->set_likelihood(
- static_cast<VoiceDetection::Likelihood>(5)));
-
- // Tesing valid likelihoods
- VoiceDetection::Likelihood likelihood[] = {
- VoiceDetection::kVeryLowLikelihood,
- VoiceDetection::kLowLikelihood,
- VoiceDetection::kModerateLikelihood,
- VoiceDetection::kHighLikelihood
- };
- for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_likelihood(likelihood[i]));
- EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
- }
-
- /* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
- // Tesing invalid frame sizes
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->voice_detection()->set_frame_size_ms(12));
-
- // Tesing valid frame sizes
- for (int i = 10; i <= 30; i += 10) {
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_frame_size_ms(i));
- EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
- }
- */
-
- // Turing VAD on/off
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
- EXPECT_TRUE(apm_->voice_detection()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
- EXPECT_FALSE(apm_->voice_detection()->is_enabled());
-
- // TODO(bjornv): Add tests for streamed voice; stream_has_voice()
-}
-
-// Below are some ideas for tests from VPM.
-
-/*TEST_F(VideoProcessingModuleTest, GetVersionTest)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, HandleNullBuffer)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, HandleBadSize)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset)
-{
-}
-*/
-} // namespace
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
- ApmEnvironment* env = new ApmEnvironment; // GTest takes ownership.
- ::testing::AddGlobalTestEnvironment(env);
-
- for (int i = 1; i < argc; i++) {
- if (strcmp(argv[i], "--write_output_data") == 0) {
- global_read_output_data = false;
- }
- }
-
- return RUN_ALL_TESTS();
-}
diff --git a/src/modules/audio_processing/main/source/noise_suppression_impl.cc b/src/modules/audio_processing/noise_suppression_impl.cc
similarity index 100%
rename from src/modules/audio_processing/main/source/noise_suppression_impl.cc
rename to src/modules/audio_processing/noise_suppression_impl.cc
diff --git a/src/modules/audio_processing/main/source/noise_suppression_impl.h b/src/modules/audio_processing/noise_suppression_impl.h
similarity index 100%
rename from src/modules/audio_processing/main/source/noise_suppression_impl.h
rename to src/modules/audio_processing/noise_suppression_impl.h
diff --git a/src/modules/audio_processing/ns/Android.mk b/src/modules/audio_processing/ns/Android.mk
new file mode 100644
index 0000000..255f470
--- /dev/null
+++ b/src/modules/audio_processing/ns/Android.mk
@@ -0,0 +1,79 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+#############################
+# Build the non-neon library.
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_ns
+LOCAL_MODULE_TAGS := optional
+LOCAL_GENERATED_SOURCES :=
+LOCAL_SRC_FILES := \
+ noise_suppression_x.c \
+ nsx_core.c
+
+# Files for floating point.
+# noise_suppression.c ns_core.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../utility \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
+ $(LOCAL_PATH)/../../../system_wrappers/interface
+
+LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libdl \
+ libstlport
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#############################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_ns_neon
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := nsx_core_neon.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS) \
+ -mfpu=neon \
+ -mfloat-abi=softfp \
+ -flax-vector-conversions
+
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/interface \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/src/modules/audio_processing/ns/main/source/defines.h b/src/modules/audio_processing/ns/defines.h
similarity index 100%
rename from src/modules/audio_processing/ns/main/source/defines.h
rename to src/modules/audio_processing/ns/defines.h
diff --git a/src/modules/audio_processing/ns/main/interface/noise_suppression.h b/src/modules/audio_processing/ns/interface/noise_suppression.h
similarity index 85%
rename from src/modules/audio_processing/ns/main/interface/noise_suppression.h
rename to src/modules/audio_processing/ns/interface/noise_suppression.h
index b8983b0..907faf4 100644
--- a/src/modules/audio_processing/ns/main/interface/noise_suppression.h
+++ b/src/modules/audio_processing/ns/interface/noise_suppression.h
@@ -30,7 +30,7 @@
* Return value : 0 - Ok
* -1 - Error (probably length is not sufficient)
*/
-int WebRtcNs_get_version(char *version, short length);
+int WebRtcNs_get_version(char* version, short length);
/*
@@ -46,7 +46,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNs_Create(NsHandle **NS_inst);
+int WebRtcNs_Create(NsHandle** NS_inst);
/*
@@ -59,7 +59,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNs_Free(NsHandle *NS_inst);
+int WebRtcNs_Free(NsHandle* NS_inst);
/*
@@ -75,7 +75,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs);
+int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs);
/*
* This changes the aggressiveness of the noise suppression method.
@@ -90,7 +90,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNs_set_policy(NsHandle *NS_inst, int mode);
+int WebRtcNs_set_policy(NsHandle* NS_inst, int mode);
/*
@@ -98,7 +98,7 @@
* input and output signals should always be 10ms (80 or 160 samples).
*
* Input
- * - NS_inst : VAD Instance. Needs to be initiated before call.
+ * - NS_inst : NS Instance. Needs to be initiated before call.
* - spframe : Pointer to speech frame buffer for L band
* - spframe_H : Pointer to speech frame buffer for H band
* - fs : sampling frequency
@@ -111,11 +111,11 @@
* Return value : 0 - OK
* -1 - Error
*/
-int WebRtcNs_Process(NsHandle *NS_inst,
- short *spframe,
- short *spframe_H,
- short *outframe,
- short *outframe_H);
+int WebRtcNs_Process(NsHandle* NS_inst,
+ short* spframe,
+ short* spframe_H,
+ short* outframe,
+ short* outframe_H);
#ifdef __cplusplus
}
diff --git a/src/modules/audio_processing/ns/main/interface/noise_suppression_x.h b/src/modules/audio_processing/ns/interface/noise_suppression_x.h
similarity index 85%
rename from src/modules/audio_processing/ns/main/interface/noise_suppression_x.h
rename to src/modules/audio_processing/ns/interface/noise_suppression_x.h
index 35fea2f..14443fa 100644
--- a/src/modules/audio_processing/ns/main/interface/noise_suppression_x.h
+++ b/src/modules/audio_processing/ns/interface/noise_suppression_x.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
-#include "signal_processing_library.h"
+#include "typedefs.h"
typedef struct NsxHandleT NsxHandle;
@@ -30,7 +30,7 @@
* Return value : 0 - Ok
* -1 - Error (probably length is not sufficient)
*/
-int WebRtcNsx_get_version(char *version, short length);
+int WebRtcNsx_get_version(char* version, short length);
/*
@@ -46,7 +46,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNsx_Create(NsxHandle **nsxInst);
+int WebRtcNsx_Create(NsxHandle** nsxInst);
/*
@@ -59,7 +59,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNsx_Free(NsxHandle *nsxInst);
+int WebRtcNsx_Free(NsxHandle* nsxInst);
/*
@@ -75,7 +75,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs);
+int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs);
/*
* This changes the aggressiveness of the noise suppression method.
@@ -90,7 +90,7 @@
* Return value : 0 - Ok
* -1 - Error
*/
-int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode);
+int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode);
/*
* This functions does noise suppression for the inserted speech frame. The
@@ -110,11 +110,11 @@
* Return value : 0 - OK
* -1 - Error
*/
-int WebRtcNsx_Process(NsxHandle *nsxInst,
- short *speechFrame,
- short *speechFrameHB,
- short *outFrame,
- short *outFrameHB);
+int WebRtcNsx_Process(NsxHandle* nsxInst,
+ short* speechFrame,
+ short* speechFrameHB,
+ short* outFrame,
+ short* outFrameHB);
#ifdef __cplusplus
}
diff --git a/src/modules/audio_processing/ns/main/source/Android.mk b/src/modules/audio_processing/ns/main/source/Android.mk
deleted file mode 100644
index 07ec98e..0000000
--- a/src/modules/audio_processing/ns/main/source/Android.mk
+++ /dev/null
@@ -1,52 +0,0 @@
-# This file is generated by gyp; do not edit. This means you!
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_ns
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- noise_suppression_x.c \
- nsx_core.c
-
-# floating point
-# noise_suppression.c ns_core.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR'
-ifeq ($(TARGET_ARCH),arm)
-MY_DEFS += \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-endif
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../utility \
- $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
- libdl \
- libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/ns/main/source/noise_suppression.c b/src/modules/audio_processing/ns/main/source/noise_suppression.c
deleted file mode 100644
index aed10b1..0000000
--- a/src/modules/audio_processing/ns/main/source/noise_suppression.c
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "defines.h"
-
-int WebRtcNs_get_version(char *versionStr, short length)
-{
- const char version[] = "NS 2.2.0";
- const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
- if (versionStr == NULL) {
- return -1;
- }
-
- if (versionLen > length) {
- return -1;
- }
-
- strncpy(versionStr, version, versionLen);
-
- return 0;
-}
-
-int WebRtcNs_Create(NsHandle **NS_inst)
-{
- *NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
- if (*NS_inst!=NULL) {
- (*(NSinst_t**)NS_inst)->initFlag=0;
- return 0;
- } else {
- return -1;
- }
-
-}
-
-int WebRtcNs_Free(NsHandle *NS_inst)
-{
- free(NS_inst);
- return 0;
-}
-
-
-int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs)
-{
- return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
-}
-
-int WebRtcNs_set_policy(NsHandle *NS_inst, int mode)
-{
- return WebRtcNs_set_policy_core((NSinst_t*) NS_inst, mode);
-}
-
-
-int WebRtcNs_Process(NsHandle *NS_inst, short *spframe, short *spframe_H, short *outframe, short *outframe_H)
-{
- return WebRtcNs_ProcessCore((NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
-}
diff --git a/src/modules/audio_processing/ns/main/source/noise_suppression_x.c b/src/modules/audio_processing/ns/main/source/noise_suppression_x.c
deleted file mode 100644
index f1ad730..0000000
--- a/src/modules/audio_processing/ns/main/source/noise_suppression_x.c
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression_x.h"
-#include "nsx_core.h"
-#include "nsx_defines.h"
-
-int WebRtcNsx_get_version(char *versionStr, short length)
-{
- const char version[] = "NS\t3.1.0";
- const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
- if (versionStr == NULL)
- {
- return -1;
- }
-
- if (versionLen > length)
- {
- return -1;
- }
-
- strncpy(versionStr, version, versionLen);
-
- return 0;
-}
-
-int WebRtcNsx_Create(NsxHandle **nsxInst)
-{
- *nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
- if (*nsxInst != NULL)
- {
- (*(NsxInst_t**)nsxInst)->initFlag = 0;
- return 0;
- } else
- {
- return -1;
- }
-
-}
-
-int WebRtcNsx_Free(NsxHandle *nsxInst)
-{
- free(nsxInst);
- return 0;
-}
-
-int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs)
-{
- return WebRtcNsx_InitCore((NsxInst_t*)nsxInst, fs);
-}
-
-int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode)
-{
- return WebRtcNsx_set_policy_core((NsxInst_t*)nsxInst, mode);
-}
-
-int WebRtcNsx_Process(NsxHandle *nsxInst, short *speechFrame, short *speechFrameHB,
- short *outFrame, short *outFrameHB)
-{
- return WebRtcNsx_ProcessCore((NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame,
- outFrameHB);
-}
-
diff --git a/src/modules/audio_processing/ns/main/source/ns_core.c b/src/modules/audio_processing/ns/main/source/ns_core.c
deleted file mode 100644
index 10a1b83..0000000
--- a/src/modules/audio_processing/ns/main/source/ns_core.c
+++ /dev/null
@@ -1,1500 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include <math.h>
-//#include <stdio.h>
-#include <stdlib.h>
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "windows_private.h"
-#include "fft4g.h"
-#include "signal_processing_library.h"
-
-// Set Feature Extraction Parameters
-void WebRtcNs_set_feature_extraction_parameters(NSinst_t *inst)
-{
- //bin size of histogram
- inst->featureExtractionParams.binSizeLrt = (float)0.1;
- inst->featureExtractionParams.binSizeSpecFlat = (float)0.05;
- inst->featureExtractionParams.binSizeSpecDiff = (float)0.1;
-
- //range of histogram over which lrt threshold is computed
- inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0;
-
- //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
- // thresholds for prior model
- inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff
- inst->featureExtractionParams.factor2ModelPars = (float)0.9; //for spectral_flatness:
- // used when noise is flatter than speech
-
- //peak limit for spectral flatness (varies between 0 and 1)
- inst->featureExtractionParams.thresPosSpecFlat = (float)0.6;
-
- //limit on spacing of two highest peaks in histogram: spacing determined by bin size
- inst->featureExtractionParams.limitPeakSpacingSpecFlat = 2
- * inst->featureExtractionParams.binSizeSpecFlat;
- inst->featureExtractionParams.limitPeakSpacingSpecDiff = 2
- * inst->featureExtractionParams.binSizeSpecDiff;
-
- //limit on relevance of second peak:
- inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5;
- inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5;
-
- // fluctuation limit of lrt feature
- inst->featureExtractionParams.thresFluctLrt = (float)0.05;
-
- //limit on the max and min values for the feature thresholds
- inst->featureExtractionParams.maxLrt = (float)1.0;
- inst->featureExtractionParams.minLrt = (float)0.20;
-
- inst->featureExtractionParams.maxSpecFlat = (float)0.95;
- inst->featureExtractionParams.minSpecFlat = (float)0.10;
-
- inst->featureExtractionParams.maxSpecDiff = (float)1.0;
- inst->featureExtractionParams.minSpecDiff = (float)0.16;
-
- //criteria of weight of histogram peak to accept/reject feature
- inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3
- * (inst->modelUpdatePars[1])); //for spectral flatness
- inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3
- * (inst->modelUpdatePars[1])); //for spectral difference
-}
-
-// Initialize state
-int WebRtcNs_InitCore(NSinst_t *inst, WebRtc_UWord32 fs)
-{
- int i;
- //We only support 10ms frames
-
- //check for valid pointer
- if (inst == NULL)
- {
- return -1;
- }
-
- // Initialization of struct
- if (fs == 8000 || fs == 16000 || fs == 32000)
- {
- inst->fs = fs;
- }
- else
- {
- return -1;
- }
- inst->windShift = 0;
- if (fs == 8000)
- {
- // We only support 10ms frames
- inst->blockLen = 80;
- inst->blockLen10ms = 80;
- inst->anaLen = 128;
- inst->window = kBlocks80w128;
- inst->outLen = 0;
- }
- else if (fs == 16000)
- {
- // We only support 10ms frames
- inst->blockLen = 160;
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->window = kBlocks160w256;
- inst->outLen = 0;
- }
- else if (fs==32000)
- {
- // We only support 10ms frames
- inst->blockLen = 160;
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->window = kBlocks160w256;
- inst->outLen = 0;
- }
- inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
-
- // Initialize fft work arrays.
- inst->ip[0] = 0; // Setting this triggers initialization.
- memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
- rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft);
-
- memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
- memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
- //for HB processing
- memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
- //for quantile noise estimation
- memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
- for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++)
- {
- inst->lquantile[i] = (float)8.0;
- inst->density[i] = (float)0.3;
- }
-
- for (i = 0; i < SIMULT; i++)
- {
- inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
- }
-
- inst->updates = 0;
-
- // Wiener filter initialization
- for (i = 0; i < HALF_ANAL_BLOCKL; i++)
- {
- inst->smooth[i] = (float)1.0;
- }
-
- // Set the aggressiveness: default
- inst->aggrMode = 0;
-
- //initialize variables for new method
- inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise
- for (i = 0; i < HALF_ANAL_BLOCKL; i++)
- {
- inst->magnPrev[i] = (float)0.0; //previous mag spectrum
- inst->noisePrev[i] = (float)0.0; //previous noise-spectrum
- inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold)
- inst->magnAvgPause[i] = (float)0.0; //conservative noise spectrum estimate
- inst->speechProbHB[i] = (float)0.0; //for estimation of HB in second pass
- inst->initMagnEst[i] = (float)0.0; //initial average mag spectrum
- }
-
- //feature quantities
- inst->featureData[0] = SF_FEATURE_THR; //spectral flatness (start on threshold)
- inst->featureData[1] = (float)0.0; //spectral entropy: not used in this version
- inst->featureData[2] = (float)0.0; //spectral variance: not used in this version
- inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold)
- inst->featureData[4] = SF_FEATURE_THR; //spectral template diff (start on threshold)
- inst->featureData[5] = (float)0.0; //normalization for spectral-diff
- inst->featureData[6] = (float)0.0; //window time-average of input magnitude spectrum
-
- //histogram quantities: used to estimate/update thresholds for features
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- inst->histLrt[i] = 0;
- inst->histSpecFlat[i] = 0;
- inst->histSpecDiff[i] = 0;
- }
-
- inst->blockInd = -1; //frame counter
- inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature
- inst->priorModelPars[1] = (float)0.5; //threshold for spectral flatness:
- // determined on-line
- inst->priorModelPars[2] = (float)1.0; //sgn_map par for spectral measure:
- // 1 for flatness measure
- inst->priorModelPars[3] = (float)0.5; //threshold for template-difference feature:
- // determined on-line
- inst->priorModelPars[4] = (float)1.0; //default weighting parameter for lrt feature
- inst->priorModelPars[5] = (float)0.0; //default weighting parameter for
- // spectral flatness feature
- inst->priorModelPars[6] = (float)0.0; //default weighting parameter for
- // spectral difference feature
-
- inst->modelUpdatePars[0] = 2; //update flag for parameters:
- // 0 no update, 1=update once, 2=update every window
- inst->modelUpdatePars[1] = 500; //window for update
- inst->modelUpdatePars[2] = 0; //counter for update of conservative noise spectrum
- //counter if the feature thresholds are updated during the sequence
- inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
-
- inst->signalEnergy = 0.0;
- inst->sumMagn = 0.0;
- inst->whiteNoiseLevel = 0.0;
- inst->pinkNoiseNumerator = 0.0;
- inst->pinkNoiseExp = 0.0;
-
- WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration
-
- //default mode
- WebRtcNs_set_policy_core(inst, 0);
-
-
- memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
-
- inst->initFlag = 1;
- return 0;
-}
-
-int WebRtcNs_set_policy_core(NSinst_t *inst, int mode)
-{
- // allow for modes:0,1,2,3
- if (mode < 0 || mode > 3)
- {
- return (-1);
- }
-
- inst->aggrMode = mode;
- if (mode == 0)
- {
- inst->overdrive = (float)1.0;
- inst->denoiseBound = (float)0.5;
- inst->gainmap = 0;
- }
- else if (mode == 1)
- {
- //inst->overdrive = (float)1.25;
- inst->overdrive = (float)1.0;
- inst->denoiseBound = (float)0.25;
- inst->gainmap = 1;
- }
- else if (mode == 2)
- {
- //inst->overdrive = (float)1.25;
- inst->overdrive = (float)1.1;
- inst->denoiseBound = (float)0.125;
- inst->gainmap = 1;
- }
- else if (mode == 3)
- {
- //inst->overdrive = (float)1.30;
- inst->overdrive = (float)1.25;
- inst->denoiseBound = (float)0.09;
- inst->gainmap = 1;
- }
- return 0;
-}
-
-// Estimate noise
-void WebRtcNs_NoiseEstimation(NSinst_t *inst, float *magn, float *noise)
-{
- int i, s, offset;
- float lmagn[HALF_ANAL_BLOCKL], delta;
-
- if (inst->updates < END_STARTUP_LONG)
- {
- inst->updates++;
- }
-
- for (i = 0; i < inst->magnLen; i++)
- {
- lmagn[i] = (float)log(magn[i]);
- }
-
- // loop over simultaneous estimates
- for (s = 0; s < SIMULT; s++)
- {
- offset = s * inst->magnLen;
-
- // newquantest(...)
- for (i = 0; i < inst->magnLen; i++)
- {
- // compute delta
- if (inst->density[offset + i] > 1.0)
- {
- delta = FACTOR * (float)1.0 / inst->density[offset + i];
- }
- else
- {
- delta = FACTOR;
- }
-
- // update log quantile estimate
- if (lmagn[i] > inst->lquantile[offset + i])
- {
- inst->lquantile[offset + i] += QUANTILE * delta
- / (float)(inst->counter[s] + 1);
- }
- else
- {
- inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta
- / (float)(inst->counter[s] + 1);
- }
-
- // update density estimate
- if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH)
- {
- inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
- + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s]
- + 1);
- }
- } // end loop over magnitude spectrum
-
- if (inst->counter[s] >= END_STARTUP_LONG)
- {
- inst->counter[s] = 0;
- if (inst->updates >= END_STARTUP_LONG)
- {
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
- }
- }
- }
-
- inst->counter[s]++;
- } // end loop over simultaneous estimates
-
- // Sequentially update the noise during startup
- if (inst->updates < END_STARTUP_LONG)
- {
- // Use the last "s" to get noise during startup that differ from zero.
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
- }
- }
-
- for (i = 0; i < inst->magnLen; i++)
- {
- noise[i] = inst->quantile[i];
- }
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by inst->modelUpdatePars[1])
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNs_FeatureParameterExtraction(NSinst_t *inst, int flag)
-{
- int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
- int maxPeak1, maxPeak2;
- int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff;
-
- float binMid, featureSum;
- float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
- float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
-
- //3 features: lrt, flatness, difference
- //lrt_feature = inst->featureData[3];
- //flat_feature = inst->featureData[0];
- //diff_feature = inst->featureData[4];
-
- //update histograms
- if (flag == 0)
- {
- // LRT
- if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt)
- && (inst->featureData[3] >= 0.0))
- {
- i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt);
- inst->histLrt[i]++;
- }
- // Spectral flatness
- if ((inst->featureData[0] < HIST_PAR_EST
- * inst->featureExtractionParams.binSizeSpecFlat)
- && (inst->featureData[0] >= 0.0))
- {
- i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat);
- inst->histSpecFlat[i]++;
- }
- // Spectral difference
- if ((inst->featureData[4] < HIST_PAR_EST
- * inst->featureExtractionParams.binSizeSpecDiff)
- && (inst->featureData[4] >= 0.0))
- {
- i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff);
- inst->histSpecDiff[i]++;
- }
- }
-
- // extract parameters for speech/noise probability
- if (flag == 1)
- {
- //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt
- avgHistLrt = 0.0;
- avgHistLrtCompl = 0.0;
- avgSquareHistLrt = 0.0;
- numHistLrt = 0;
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt;
- if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt)
- {
- avgHistLrt += inst->histLrt[i] * binMid;
- numHistLrt += inst->histLrt[i];
- }
- avgSquareHistLrt += inst->histLrt[i] * binMid * binMid;
- avgHistLrtCompl += inst->histLrt[i] * binMid;
- }
- if (numHistLrt > 0)
- {
- avgHistLrt = avgHistLrt / ((float)numHistLrt);
- }
- avgHistLrtCompl = avgHistLrtCompl / ((float)inst->modelUpdatePars[1]);
- avgSquareHistLrt = avgSquareHistLrt / ((float)inst->modelUpdatePars[1]);
- fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
- // get threshold for lrt feature:
- if (fluctLrt < inst->featureExtractionParams.thresFluctLrt)
- {
- //very low fluct, so likely noise
- inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
- }
- else
- {
- inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars
- * avgHistLrt;
- // check if value is within min/max range
- if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt)
- {
- inst->priorModelPars[0] = inst->featureExtractionParams.minLrt;
- }
- if (inst->priorModelPars[0] > inst->featureExtractionParams.maxLrt)
- {
- inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
- }
- }
- // done with lrt feature
-
- //
- // for spectral flatness and spectral difference: compute the main peaks of histogram
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecFlat = 0.0;
- posPeak2SpecFlat = 0.0;
- weightPeak1SpecFlat = 0;
- weightPeak2SpecFlat = 0;
-
- // peaks for flatness
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat;
- if (inst->histSpecFlat[i] > maxPeak1)
- {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecFlat = weightPeak1SpecFlat;
- posPeak2SpecFlat = posPeak1SpecFlat;
-
- maxPeak1 = inst->histSpecFlat[i];
- weightPeak1SpecFlat = inst->histSpecFlat[i];
- posPeak1SpecFlat = binMid;
- }
- else if (inst->histSpecFlat[i] > maxPeak2)
- {
- // Found new "second" peak
- maxPeak2 = inst->histSpecFlat[i];
- weightPeak2SpecFlat = inst->histSpecFlat[i];
- posPeak2SpecFlat = binMid;
- }
- }
-
- //compute two peaks for spectral difference
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecDiff = 0.0;
- posPeak2SpecDiff = 0.0;
- weightPeak1SpecDiff = 0;
- weightPeak2SpecDiff = 0;
- // peaks for spectral difference
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff;
- if (inst->histSpecDiff[i] > maxPeak1)
- {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecDiff = weightPeak1SpecDiff;
- posPeak2SpecDiff = posPeak1SpecDiff;
-
- maxPeak1 = inst->histSpecDiff[i];
- weightPeak1SpecDiff = inst->histSpecDiff[i];
- posPeak1SpecDiff = binMid;
- }
- else if (inst->histSpecDiff[i] > maxPeak2)
- {
- // Found new "second" peak
- maxPeak2 = inst->histSpecDiff[i];
- weightPeak2SpecDiff = inst->histSpecDiff[i];
- posPeak2SpecDiff = binMid;
- }
- }
-
- // for spectrum flatness feature
- useFeatureSpecFlat = 1;
- // merge the two peaks if they are close
- if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat)
- < inst->featureExtractionParams.limitPeakSpacingSpecFlat)
- && (weightPeak2SpecFlat
- > inst->featureExtractionParams.limitPeakWeightsSpecFlat
- * weightPeak1SpecFlat))
- {
- weightPeak1SpecFlat += weightPeak2SpecFlat;
- posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat);
- }
- //reject if weight of peaks is not large enough, or peak value too small
- if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat
- || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat)
- {
- useFeatureSpecFlat = 0;
- }
- // if selected, get the threshold
- if (useFeatureSpecFlat == 1)
- {
- // compute the threshold
- inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars
- * posPeak1SpecFlat;
- //check if value is within min/max range
- if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat)
- {
- inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat;
- }
- if (inst->priorModelPars[1] > inst->featureExtractionParams.maxSpecFlat)
- {
- inst->priorModelPars[1] = inst->featureExtractionParams.maxSpecFlat;
- }
- }
- // done with flatness feature
-
- // for template feature
- useFeatureSpecDiff = 1;
- // merge the two peaks if they are close
- if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff)
- < inst->featureExtractionParams.limitPeakSpacingSpecDiff)
- && (weightPeak2SpecDiff
- > inst->featureExtractionParams.limitPeakWeightsSpecDiff
- * weightPeak1SpecDiff))
- {
- weightPeak1SpecDiff += weightPeak2SpecDiff;
- posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff);
- }
- // get the threshold value
- inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars
- * posPeak1SpecDiff;
- //reject if weight of peaks is not large enough
- if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff)
- {
- useFeatureSpecDiff = 0;
- }
- //check if value is within min/max range
- if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff)
- {
- inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff;
- }
- if (inst->priorModelPars[3] > inst->featureExtractionParams.maxSpecDiff)
- {
- inst->priorModelPars[3] = inst->featureExtractionParams.maxSpecDiff;
- }
- // done with spectral difference feature
-
- // don't use template feature if fluctuation of lrt feature is very low:
- // most likely just noise state
- if (fluctLrt < inst->featureExtractionParams.thresFluctLrt)
- {
- useFeatureSpecDiff = 0;
- }
-
- // select the weights between the features
- // inst->priorModelPars[4] is weight for lrt: always selected
- // inst->priorModelPars[5] is weight for spectral flatness
- // inst->priorModelPars[6] is weight for spectral difference
- featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
- inst->priorModelPars[4] = (float)1.0 / featureSum;
- inst->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
- inst->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
-
- // set hists to zero for next update
- if (inst->modelUpdatePars[0] >= 1)
- {
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- inst->histLrt[i] = 0;
- inst->histSpecFlat[i] = 0;
- inst->histSpecDiff[i] = 0;
- }
- }
- } // end of flag == 1
-}
-
-// Compute spectral flatness on input spectrum
-// magnIn is the magnitude spectrum
-// spectral flatness is returned in inst->featureData[0]
-void WebRtcNs_ComputeSpectralFlatness(NSinst_t *inst, float *magnIn)
-{
- int i;
- int shiftLP = 1; //option to remove first bin(s) from spectral measures
- float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
-
- // comute spectral measures
- // for flatness
- avgSpectralFlatnessNum = 0.0;
- avgSpectralFlatnessDen = inst->sumMagn;
- for (i = 0; i < shiftLP; i++)
- {
- avgSpectralFlatnessDen -= magnIn[i];
- }
- // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
- for (i = shiftLP; i < inst->magnLen; i++)
- {
- if (magnIn[i] > 0.0)
- {
- avgSpectralFlatnessNum += (float)log(magnIn[i]);
- }
- else
- {
- inst->featureData[0] -= SPECT_FL_TAVG * inst->featureData[0];
- return;
- }
- }
- //normalize
- avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen;
- avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen;
-
- //ratio and inverse log: check for case of log(0)
- spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
-
- //time-avg update of spectral flatness feature
- inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]);
- // done with flatness feature
-}
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magnIn is the input spectrum
-// the reference/template spectrum is inst->magnAvgPause[i]
-// returns (normalized) spectral difference in inst->featureData[4]
-void WebRtcNs_ComputeSpectralDifference(NSinst_t *inst, float *magnIn)
-{
- // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
- int i;
- float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
-
- avgPause = 0.0;
- avgMagn = inst->sumMagn;
- // compute average quantities
- for (i = 0; i < inst->magnLen; i++)
- {
- //conservative smooth noise spectrum from pause frames
- avgPause += inst->magnAvgPause[i];
- }
- avgPause = avgPause / ((float)inst->magnLen);
- avgMagn = avgMagn / ((float)inst->magnLen);
-
- covMagnPause = 0.0;
- varPause = 0.0;
- varMagn = 0.0;
- // compute variance and covariance quantities
- for (i = 0; i < inst->magnLen; i++)
- {
- covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause);
- varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause);
- varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
- }
- covMagnPause = covMagnPause / ((float)inst->magnLen);
- varPause = varPause / ((float)inst->magnLen);
- varMagn = varMagn / ((float)inst->magnLen);
- // update of average magnitude spectrum
- inst->featureData[6] += inst->signalEnergy;
-
- avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001);
- // normalize and compute time-avg update of difference feature
- avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001));
- inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]);
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//magn is the input magnitude spectrum
-//noise is the noise spectrum
-//snrLocPrior is the prior snr for each freq.
-//snr loc_post is the post snr for each freq.
-void WebRtcNs_SpeechNoiseProb(NSinst_t *inst, float *probSpeechFinal, float *snrLocPrior,
- float *snrLocPost)
-{
- int i, sgnMap;
- float invLrt, gainPrior, indPrior;
- float logLrtTimeAvgKsum, besselTmp;
- float indicator0, indicator1, indicator2;
- float tmpFloat1, tmpFloat2;
- float weightIndPrior0, weightIndPrior1, weightIndPrior2;
- float threshPrior0, threshPrior1, threshPrior2;
- float widthPrior, widthPrior0, widthPrior1, widthPrior2;
-
- widthPrior0 = WIDTH_PR_MAP;
- widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region:
- // lower range, so increase width in tanh map
- widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure
-
- //threshold parameters for features
- threshPrior0 = inst->priorModelPars[0];
- threshPrior1 = inst->priorModelPars[1];
- threshPrior2 = inst->priorModelPars[3];
-
- //sign for flatness feature
- sgnMap = (int)(inst->priorModelPars[2]);
-
- //weight parameters for features
- weightIndPrior0 = inst->priorModelPars[4];
- weightIndPrior1 = inst->priorModelPars[5];
- weightIndPrior2 = inst->priorModelPars[6];
-
- // compute feature based on average LR factor
- // this is the average over all frequencies of the smooth log lrt
- logLrtTimeAvgKsum = 0.0;
- for (i = 0; i < inst->magnLen; i++)
- {
- tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i];
- tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001);
- besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2;
- inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1)
- - inst->logLrtTimeAvg[i]);
- logLrtTimeAvgKsum += inst->logLrtTimeAvg[i];
- }
- logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen);
- inst->featureData[3] = logLrtTimeAvgKsum;
- // done with computation of LR factor
-
- //
- //compute the indicator functions
- //
-
- // average lrt feature
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (logLrtTimeAvgKsum < threshPrior0)
- {
- widthPrior = widthPrior1;
- }
- // compute indicator function: sigmoid map
- indicator0 = (float)0.5 * ((float)tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0))
- + (float)1.0);
-
- //spectral flatness feature
- tmpFloat1 = inst->featureData[0];
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (sgnMap == 1 && (tmpFloat1 > threshPrior1))
- {
- widthPrior = widthPrior1;
- }
- if (sgnMap == -1 && (tmpFloat1 < threshPrior1))
- {
- widthPrior = widthPrior1;
- }
- // compute indicator function: sigmoid map
- indicator1 = (float)0.5 * ((float)tanh(
- (float)sgnMap * widthPrior * (threshPrior1
- - tmpFloat1)) + (float)1.0);
-
- //for template spectrum-difference
- tmpFloat1 = inst->featureData[4];
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (tmpFloat1 < threshPrior2)
- {
- widthPrior = widthPrior2;
- }
- // compute indicator function: sigmoid map
- indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2))
- + (float)1.0);
-
- //combine the indicator function with the feature weights
- indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2
- * indicator2;
- // done with computing indicator function
-
- //compute the prior probability
- inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb);
- // make sure probabilities are within range: keep floor to 0.01
- if (inst->priorSpeechProb > 1.0)
- {
- inst->priorSpeechProb = (float)1.0;
- }
- if (inst->priorSpeechProb < 0.01)
- {
- inst->priorSpeechProb = (float)0.01;
- }
-
- //final speech probability: combine prior model with LR factor:
- gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001);
- for (i = 0; i < inst->magnLen; i++)
- {
- invLrt = (float)exp(-inst->logLrtTimeAvg[i]);
- invLrt = (float)gainPrior * invLrt;
- probSpeechFinal[i] = (float)1.0 / ((float)1.0 + invLrt);
- }
-}
-
-int WebRtcNs_ProcessCore(NSinst_t *inst,
- short *speechFrame,
- short *speechFrameHB,
- short *outFrame,
- short *outFrameHB)
-{
- // main routine for noise reduction
-
- int flagHB = 0;
- int i;
- const int kStartBand = 5; // Skip first frequency bins during estimation.
- int updateParsFlag;
-
- float energy1, energy2, gain, factor, factor1, factor2;
- float signalEnergy, sumMagn;
- float snrPrior, currentEstimateStsa;
- float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
- float gammaNoiseTmp, gammaNoiseOld;
- float noiseUpdateTmp, fTmp, dTmp;
- float fin[BLOCKL_MAX], fout[BLOCKL_MAX];
- float winData[ANAL_BLOCKL_MAX];
- float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
- float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
- float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
- float probSpeechFinal[HALF_ANAL_BLOCKL], previousEstimateStsa[HALF_ANAL_BLOCKL];
- float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
- // Variables during startup
- float sum_log_i = 0.0;
- float sum_log_i_square = 0.0;
- float sum_log_magn = 0.0;
- float sum_log_i_log_magn = 0.0;
- float parametric_noise = 0.0;
- float parametric_exp = 0.0;
- float parametric_num = 0.0;
-
- // SWB variables
- int deltaBweHB = 1;
- int deltaGainHB = 1;
- float decayBweHB = 1.0;
- float gainMapParHB = 1.0;
- float gainTimeDomainHB = 1.0;
- float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
-
- // Check that initiation has been done
- if (inst->initFlag != 1)
- {
- return (-1);
- }
- // Check for valid pointers based on sampling rate
- if (inst->fs == 32000)
- {
- if (speechFrameHB == NULL)
- {
- return -1;
- }
- flagHB = 1;
- // range for averaging low band quantities for H band gain
- deltaBweHB = (int)inst->magnLen / 4;
- deltaGainHB = deltaBweHB;
- }
- //
- updateParsFlag = inst->modelUpdatePars[0];
- //
-
- //for LB do all processing
- // convert to float
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- fin[i] = (float)speechFrame[i];
- }
- // update analysis buffer for L band
- memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
- sizeof(float) * inst->blockLen10ms);
-
- if (flagHB == 1)
- {
- // convert to float
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- fin[i] = (float)speechFrameHB[i];
- }
- // update analysis buffer for H band
- memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
- sizeof(float) * inst->blockLen10ms);
- }
-
- // check if processing needed
- if (inst->outLen == 0)
- {
- // windowing
- energy1 = 0.0;
- for (i = 0; i < inst->anaLen; i++)
- {
- winData[i] = inst->window[i] * inst->dataBuf[i];
- energy1 += winData[i] * winData[i];
- }
- if (energy1 == 0.0)
- {
- // synthesize the special case of zero input
- // we want to avoid updating statistics in this case:
- // Updating feature statistics when we have zeros only will cause thresholds to
- // move towards zero signal situations. This in turn has the effect that once the
- // signal is "turned on" (non-zero values) everything will be treated as speech
- // and there is no noise suppression effect. Depending on the duration of the
- // inactive signal it takes a considerable amount of time for the system to learn
- // what is noise and what is speech.
-
- // read out fully processed segment
- for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++)
- {
- fout[i - inst->windShift] = inst->syntBuf[i];
- }
- // update synthesis buffer
- memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
- sizeof(float) * (inst->anaLen - inst->blockLen));
- memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
- sizeof(float) * inst->blockLen);
-
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms)
- {
- for (i = 0; i < inst->outLen; i++)
- {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
- }
- // convert to short
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- dTmp = fout[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN)
- {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- }
- else if (dTmp > WEBRTC_SPL_WORD16_MAX)
- {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrame[i] = (short)dTmp;
- }
-
- // for time-domain gain of HB
- if (flagHB == 1)
- {
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- dTmp = inst->dataBufHB[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN)
- {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- }
- else if (dTmp > WEBRTC_SPL_WORD16_MAX)
- {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrameHB[i] = (short)dTmp;
- }
- } // end of H band gain computation
- //
- return 0;
- }
-
- //
- inst->blockInd++; // Update the block index only when we process a block.
- // FFT
- rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
-
- imag[0] = 0;
- real[0] = winData[0];
- magn[0] = (float)(fabs(real[0]) + 1.0f);
- imag[inst->magnLen - 1] = 0;
- real[inst->magnLen - 1] = winData[1];
- magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
- signalEnergy = (float)(real[0] * real[0]) + (float)(real[inst->magnLen - 1]
- * real[inst->magnLen - 1]);
- sumMagn = magn[0] + magn[inst->magnLen - 1];
- if (inst->blockInd < END_STARTUP_SHORT)
- {
- inst->initMagnEst[0] += magn[0];
- inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
- tmpFloat2 = log((float)(inst->magnLen - 1));
- sum_log_i = tmpFloat2;
- sum_log_i_square = tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[inst->magnLen - 1]);
- sum_log_magn = tmpFloat1;
- sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
- }
- for (i = 1; i < inst->magnLen - 1; i++)
- {
- real[i] = winData[2 * i];
- imag[i] = winData[2 * i + 1];
- // magnitude spectrum
- fTmp = real[i] * real[i];
- fTmp += imag[i] * imag[i];
- signalEnergy += fTmp;
- magn[i] = ((float)sqrt(fTmp)) + 1.0f;
- sumMagn += magn[i];
- if (inst->blockInd < END_STARTUP_SHORT)
- {
- inst->initMagnEst[i] += magn[i];
- if (i >= kStartBand)
- {
- tmpFloat2 = log((float)i);
- sum_log_i += tmpFloat2;
- sum_log_i_square += tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[i]);
- sum_log_magn += tmpFloat1;
- sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
- }
- }
- }
- signalEnergy = signalEnergy / ((float)inst->magnLen);
- inst->signalEnergy = signalEnergy;
- inst->sumMagn = sumMagn;
-
- //compute spectral flatness on input spectrum
- WebRtcNs_ComputeSpectralFlatness(inst, magn);
- // quantile noise estimate
- WebRtcNs_NoiseEstimation(inst, magn, noise);
- //compute simplified noise model during startup
- if (inst->blockInd < END_STARTUP_SHORT)
- {
- // Estimate White noise
- inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
- // Estimate Pink noise parameters
- tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
- tmpFloat1 -= (sum_log_i * sum_log_i);
- tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the estimated spectrum to be positive
- if (tmpFloat3 < 0.0f)
- {
- tmpFloat3 = 0.0f;
- }
- inst->pinkNoiseNumerator += tmpFloat3;
- tmpFloat2 = (sum_log_i * sum_log_magn);
- tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the pink noise power to be in the interval [0, 1];
- if (tmpFloat3 < 0.0f)
- {
- tmpFloat3 = 0.0f;
- }
- if (tmpFloat3 > 1.0f)
- {
- tmpFloat3 = 1.0f;
- }
- inst->pinkNoiseExp += tmpFloat3;
-
- // Calculate frequency independent parts of parametric noise estimate.
- if (inst->pinkNoiseExp == 0.0f)
- {
- // Use white noise estimate
- parametric_noise = inst->whiteNoiseLevel;
- }
- else
- {
- // Use pink noise estimate
- parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
- parametric_num *= (float)(inst->blockInd + 1);
- parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
- parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp);
- }
- for (i = 0; i < inst->magnLen; i++)
- {
- // Estimate the background noise using the white and pink noise parameters
- if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand))
- {
- // Use pink noise estimate
- parametric_noise = parametric_num / pow((float)i, parametric_exp);
- }
- theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise);
- theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
- // Weight quantile noise with modeled noise
- noise[i] *= (inst->blockInd);
- tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd);
- noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
- noise[i] /= END_STARTUP_SHORT;
- }
- }
- //compute average signal during END_STARTUP_LONG time:
- // used to normalize spectral difference measure
- if (inst->blockInd < END_STARTUP_LONG)
- {
- inst->featureData[5] *= inst->blockInd;
- inst->featureData[5] += signalEnergy;
- inst->featureData[5] /= (inst->blockInd + 1);
- }
-
-#ifdef PROCESS_FLOW_0
- if (inst->blockInd > END_STARTUP_LONG)
- {
- //option: average the quantile noise: for check with AEC2
- for (i = 0; i < inst->magnLen; i++)
- {
- noise[i] = (float)0.6 * inst->noisePrev[i] + (float)0.4 * noise[i];
- }
- for (i = 0; i < inst->magnLen; i++)
- {
- // Wiener with over sub-substraction:
- theFilter[i] = (magn[i] - inst->overdrive * noise[i]) / (magn[i] + (float)0.0001);
- }
- }
-#else
- //start processing at frames == converged+1
- //
- // STEP 1: compute prior and post snr based on quantile noise est
- //
-
- // compute DD estimate of prior SNR: needed for new method
- for (i = 0; i < inst->magnLen; i++)
- {
- // post snr
- snrLocPost[i] = (float)0.0;
- if (magn[i] > noise[i])
- {
- snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
- }
- // previous post snr
- // previous estimate: based on previous frame with gain filter
- previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001)
- * (inst->smooth[i]);
- // DD estimate is sum of two terms: current estimate and previous estimate
- // directed decision update of snrPrior
- snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
- * snrLocPost[i];
- // post and prior snr needed for step 2
- } // end of loop over freqs
-#ifdef PROCESS_FLOW_1
- for (i = 0; i < inst->magnLen; i++)
- {
- // gain filter
- tmpFloat1 = inst->overdrive + snrLocPrior[i];
- tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
- theFilter[i] = (float)tmpFloat2;
- } // end of loop over freqs
-#endif
- // done with step 1: dd computation of prior and post snr
-
- //
- //STEP 2: compute speech/noise likelihood
- //
-#ifdef PROCESS_FLOW_2
- // compute difference of input spectrum with learned/estimated noise spectrum
- WebRtcNs_ComputeSpectralDifference(inst, magn);
- // compute histograms for parameter decisions (thresholds and weights for features)
- // parameters are extracted once every window time (=inst->modelUpdatePars[1])
- if (updateParsFlag >= 1)
- {
- // counter update
- inst->modelUpdatePars[3]--;
- // update histogram
- if (inst->modelUpdatePars[3] > 0)
- {
- WebRtcNs_FeatureParameterExtraction(inst, 0);
- }
- // compute model parameters
- if (inst->modelUpdatePars[3] == 0)
- {
- WebRtcNs_FeatureParameterExtraction(inst, 1);
- inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
- // if wish to update only once, set flag to zero
- if (updateParsFlag == 1)
- {
- inst->modelUpdatePars[0] = 0;
- }
- else
- {
- // update every window:
- // get normalization for spectral difference for next window estimate
- inst->featureData[6] = inst->featureData[6]
- / ((float)inst->modelUpdatePars[1]);
- inst->featureData[5] = (float)0.5 * (inst->featureData[6]
- + inst->featureData[5]);
- inst->featureData[6] = (float)0.0;
- }
- }
- }
- // compute speech/noise probability
- WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost);
- // time-avg parameter for noise update
- gammaNoiseTmp = NOISE_UPDATE;
- for (i = 0; i < inst->magnLen; i++)
- {
- probSpeech = probSpeechFinal[i];
- probNonSpeech = (float)1.0 - probSpeech;
- // temporary noise update:
- // use it for speech frames if update value is less than previous
- noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
- * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- //
- // time-constant based on speech/noise state
- gammaNoiseOld = gammaNoiseTmp;
- gammaNoiseTmp = NOISE_UPDATE;
- // increase gamma (i.e., less noise update) for frame likely to be speech
- if (probSpeech > PROB_RANGE)
- {
- gammaNoiseTmp = SPEECH_UPDATE;
- }
- // conservative noise update
- if (probSpeech < PROB_RANGE)
- {
- inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
- }
- // noise update
- if (gammaNoiseTmp == gammaNoiseOld)
- {
- noise[i] = noiseUpdateTmp;
- }
- else
- {
- noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
- * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- // allow for noise update downwards:
- // if noise update decreases the noise, it is safe, so allow it to happen
- if (noiseUpdateTmp < noise[i])
- {
- noise[i] = noiseUpdateTmp;
- }
- }
- } // end of freq loop
- // done with step 2: noise update
-
- //
- // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
- //
- for (i = 0; i < inst->magnLen; i++)
- {
- // post and prior snr
- currentEstimateStsa = (float)0.0;
- if (magn[i] > noise[i])
- {
- currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
- }
- // DD estimate is sume of two terms: current estimate and previous estimate
- // directed decision update of snrPrior
- snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
- * currentEstimateStsa;
- // gain filter
- tmpFloat1 = inst->overdrive + snrPrior;
- tmpFloat2 = (float)snrPrior / tmpFloat1;
- theFilter[i] = (float)tmpFloat2;
- } // end of loop over freqs
- // done with step3
-#endif
-#endif
-
- for (i = 0; i < inst->magnLen; i++)
- {
- // flooring bottom
- if (theFilter[i] < inst->denoiseBound)
- {
- theFilter[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilter[i] > (float)1.0)
- {
- theFilter[i] = 1.0;
- }
- if (inst->blockInd < END_STARTUP_SHORT)
- {
- // flooring bottom
- if (theFilterTmp[i] < inst->denoiseBound)
- {
- theFilterTmp[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilterTmp[i] > (float)1.0)
- {
- theFilterTmp[i] = 1.0;
- }
- // Weight the two suppression filters
- theFilter[i] *= (inst->blockInd);
- theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
- theFilter[i] += theFilterTmp[i];
- theFilter[i] /= (END_STARTUP_SHORT);
- }
- // smoothing
-#ifdef PROCESS_FLOW_0
- inst->smooth[i] *= SMOOTH; // value set to 0.7 in define.h file
- inst->smooth[i] += ((float)1.0 - SMOOTH) * theFilter[i];
-#else
- inst->smooth[i] = theFilter[i];
-#endif
- real[i] *= inst->smooth[i];
- imag[i] *= inst->smooth[i];
- }
- // keep track of noise and magn spectrum for next frame
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->noisePrev[i] = noise[i];
- inst->magnPrev[i] = magn[i];
- }
- // back to time domain
- winData[0] = real[0];
- winData[1] = real[inst->magnLen - 1];
- for (i = 1; i < inst->magnLen - 1; i++)
- {
- winData[2 * i] = real[i];
- winData[2 * i + 1] = imag[i];
- }
- rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
-
- for (i = 0; i < inst->anaLen; i++)
- {
- real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
- }
-
- //scale factor: only do it after END_STARTUP_LONG time
- factor = (float)1.0;
- if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG)
- {
- factor1 = (float)1.0;
- factor2 = (float)1.0;
-
- energy2 = 0.0;
- for (i = 0; i < inst->anaLen;i++)
- {
- energy2 += (float)real[i] * (float)real[i];
- }
- gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
-
-#ifdef PROCESS_FLOW_2
- // scaling for new version
- if (gain > B_LIM)
- {
- factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
- if (gain * factor1 > (float)1.0)
- {
- factor1 = (float)1.0 / gain;
- }
- }
- if (gain < B_LIM)
- {
- //don't reduce scale too much for pause regions:
- // attenuation here should be controlled by flooring
- if (gain <= inst->denoiseBound)
- {
- gain = inst->denoiseBound;
- }
- factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
- }
- //combine both scales with speech/noise prob:
- // note prior (priorSpeechProb) is not frequency dependent
- factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb)
- * factor2;
-#else
- if (gain > B_LIM)
- {
- factor = (float)1.0 + (float)1.3 * (gain - B_LIM);
- }
- else
- {
- factor = (float)1.0 + (float)2.0 * (gain - B_LIM);
- }
- if (gain * factor > (float)1.0)
- {
- factor = (float)1.0 / gain;
- }
-#endif
- } // out of inst->gainmap==1
-
- // synthesis
- for (i = 0; i < inst->anaLen; i++)
- {
- inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
- }
- // read out fully processed segment
- for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++)
- {
- fout[i - inst->windShift] = inst->syntBuf[i];
- }
- // update synthesis buffer
- memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
- sizeof(float) * (inst->anaLen - inst->blockLen));
- memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
- sizeof(float) * inst->blockLen);
-
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms)
- {
- for (i = 0; i < inst->outLen; i++)
- {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
- }
- } // end of if out.len==0
- else
- {
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- fout[i] = inst->outBuf[i];
- }
- memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms,
- sizeof(float) * (inst->outLen - inst->blockLen10ms));
- memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0,
- sizeof(float) * inst->blockLen10ms);
- inst->outLen -= inst->blockLen10ms;
- }
-
- // convert to short
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- dTmp = fout[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN)
- {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- }
- else if (dTmp > WEBRTC_SPL_WORD16_MAX)
- {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrame[i] = (short)dTmp;
- }
-
- // for time-domain gain of HB
- if (flagHB == 1)
- {
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->speechProbHB[i] = probSpeechFinal[i];
- }
- if (inst->blockInd > END_STARTUP_LONG)
- {
- // average speech prob from low band
- // avg over second half (i.e., 4->8kHz) of freq. spectrum
- avgProbSpeechHB = 0.0;
- for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++)
- {
- avgProbSpeechHB += inst->speechProbHB[i];
- }
- avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
- // average filter gain from low band
- // average over second half (i.e., 4->8kHz) of freq. spectrum
- avgFilterGainHB = 0.0;
- for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++)
- {
- avgFilterGainHB += inst->smooth[i];
- }
- avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
- avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
- // gain based on speech prob:
- gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
- //combine gain with low band gain
- gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
- if (avgProbSpeechHB >= (float)0.5)
- {
- gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
- }
- gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
- } // end of converged
- //make sure gain is within flooring range
- // flooring bottom
- if (gainTimeDomainHB < inst->denoiseBound)
- {
- gainTimeDomainHB = inst->denoiseBound;
- }
- // flooring top
- if (gainTimeDomainHB > (float)1.0)
- {
- gainTimeDomainHB = 1.0;
- }
- //apply gain
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- dTmp = gainTimeDomainHB * inst->dataBufHB[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN)
- {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- }
- else if (dTmp > WEBRTC_SPL_WORD16_MAX)
- {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrameHB[i] = (short)dTmp;
- }
- } // end of H band gain computation
- //
-
- return 0;
-}
diff --git a/src/modules/audio_processing/ns/main/source/ns_core.h b/src/modules/audio_processing/ns/main/source/ns_core.h
deleted file mode 100644
index f72e22b..0000000
--- a/src/modules/audio_processing/ns/main/source/ns_core.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-
-#include "defines.h"
-
-typedef struct NSParaExtract_t_ {
-
- //bin size of histogram
- float binSizeLrt;
- float binSizeSpecFlat;
- float binSizeSpecDiff;
- //range of histogram over which lrt threshold is computed
- float rangeAvgHistLrt;
- //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
- //thresholds for prior model
- float factor1ModelPars; //for lrt and spectral difference
- float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech
- //peak limit for spectral flatness (varies between 0 and 1)
- float thresPosSpecFlat;
- //limit on spacing of two highest peaks in histogram: spacing determined by bin size
- float limitPeakSpacingSpecFlat;
- float limitPeakSpacingSpecDiff;
- //limit on relevance of second peak:
- float limitPeakWeightsSpecFlat;
- float limitPeakWeightsSpecDiff;
- //limit on fluctuation of lrt feature
- float thresFluctLrt;
- //limit on the max and min values for the feature thresholds
- float maxLrt;
- float minLrt;
- float maxSpecFlat;
- float minSpecFlat;
- float maxSpecDiff;
- float minSpecDiff;
- //criteria of weight of histogram peak to accept/reject feature
- int thresWeightSpecFlat;
- int thresWeightSpecDiff;
-
-} NSParaExtract_t;
-
-typedef struct NSinst_t_ {
-
- WebRtc_UWord32 fs;
- int blockLen;
- int blockLen10ms;
- int windShift;
- int outLen;
- int anaLen;
- int magnLen;
- int aggrMode;
- const float* window;
- float dataBuf[ANAL_BLOCKL_MAX];
- float syntBuf[ANAL_BLOCKL_MAX];
- float outBuf[3 * BLOCKL_MAX];
-
- int initFlag;
- // parameters for quantile noise estimation
- float density[SIMULT * HALF_ANAL_BLOCKL];
- float lquantile[SIMULT * HALF_ANAL_BLOCKL];
- float quantile[HALF_ANAL_BLOCKL];
- int counter[SIMULT];
- int updates;
- // parameters for Wiener filter
- float smooth[HALF_ANAL_BLOCKL];
- float overdrive;
- float denoiseBound;
- int gainmap;
- // fft work arrays.
- int ip[IP_LENGTH];
- float wfft[W_LENGTH];
-
- // parameters for new method: some not needed, will reduce/cleanup later
- WebRtc_Word32 blockInd; //frame index counter
- int modelUpdatePars[4]; //parameters for updating or estimating
- // thresholds/weights for prior model
- float priorModelPars[7]; //parameters for prior model
- float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
- float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame
- float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing
- float priorSpeechProb; //prior speech/noise probability
- float featureData[7]; //data for features
- float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate
- float signalEnergy; //energy of magn
- float sumMagn; //sum of magn
- float whiteNoiseLevel; //initial noise estimate
- float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate
- float pinkNoiseNumerator; //pink noise parameter: numerator
- float pinkNoiseExp; //pink noise parameter: power of freq
- NSParaExtract_t featureExtractionParams; //parameters for feature extraction
- //histograms for parameter estimation
- int histLrt[HIST_PAR_EST];
- int histSpecFlat[HIST_PAR_EST];
- int histSpecDiff[HIST_PAR_EST];
- //quantities for high band estimate
- float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT
- float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB
-
-} NSinst_t;
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/****************************************************************************
- * WebRtcNs_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- * - inst : Instance that should be initialized
- * - fs : Sampling frequency
- *
- * Output:
- * - inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_InitCore(NSinst_t *inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNs_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- * - NS_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_set_policy_core(NSinst_t *inst, int mode);
-
-/****************************************************************************
- * WebRtcNs_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - inFrameLow : Input speech frame for lower band
- * - inFrameHigh : Input speech frame for higher band
- *
- * Output:
- * - inst : Updated instance
- * - outFrameLow : Output speech frame for lower band
- * - outFrameHigh : Output speech frame for higher band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-
-
-int WebRtcNs_ProcessCore(NSinst_t *inst,
- short *inFrameLow,
- short *inFrameHigh,
- short *outFrameLow,
- short *outFrameHigh);
-
-
-#ifdef __cplusplus
-}
-#endif
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
diff --git a/src/modules/audio_processing/ns/main/source/nsx_core.c b/src/modules/audio_processing/ns/main/source/nsx_core.c
deleted file mode 100644
index 01d3e54..0000000
--- a/src/modules/audio_processing/ns/main/source/nsx_core.c
+++ /dev/null
@@ -1,2493 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "noise_suppression_x.h"
-
-#include <assert.h>
-#include <math.h>
-#include <string.h>
-#include <stdlib.h>
-
-#include "nsx_core.h"
-
-// Skip first frequency bins during estimation. (0 <= value < 64)
-static const int kStartBand = 5;
-
-// Rounding
-static const WebRtc_Word16 kRoundTable[16] = {0, 1, 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024,
- 2048, 4096, 8192, 16384};
-
-// Constants to compensate for shifting signal log(2^shifts).
-static const WebRtc_Word16 kLogTable[9] = {0, 177, 355, 532, 710, 887, 1065, 1242, 1420};
-
-static const WebRtc_Word16 kCounterDiv[201] = {32767, 16384, 10923, 8192, 6554, 5461, 4681,
- 4096, 3641, 3277, 2979, 2731, 2521, 2341, 2185, 2048, 1928, 1820, 1725, 1638, 1560,
- 1489, 1425, 1365, 1311, 1260, 1214, 1170, 1130, 1092, 1057, 1024, 993, 964, 936, 910,
- 886, 862, 840, 819, 799, 780, 762, 745, 728, 712, 697, 683, 669, 655, 643, 630, 618,
- 607, 596, 585, 575, 565, 555, 546, 537, 529, 520, 512, 504, 496, 489, 482, 475, 468,
- 462, 455, 449, 443, 437, 431, 426, 420, 415, 410, 405, 400, 395, 390, 386, 381, 377,
- 372, 368, 364, 360, 356, 352, 349, 345, 341, 338, 334, 331, 328, 324, 321, 318, 315,
- 312, 309, 306, 303, 301, 298, 295, 293, 290, 287, 285, 282, 280, 278, 275, 273, 271,
- 269, 266, 264, 262, 260, 258, 256, 254, 252, 250, 248, 246, 245, 243, 241, 239, 237,
- 236, 234, 232, 231, 229, 228, 226, 224, 223, 221, 220, 218, 217, 216, 214, 213, 211,
- 210, 209, 207, 206, 205, 204, 202, 201, 200, 199, 197, 196, 195, 194, 193, 192, 191,
- 189, 188, 187, 186, 185, 184, 183, 182, 181, 180, 179, 178, 177, 176, 175, 174, 173,
- 172, 172, 171, 170, 169, 168, 167, 166, 165, 165, 164, 163};
-
-static const WebRtc_Word16 kLogTableFrac[256] = {
- 0, 1, 3, 4, 6, 7, 9, 10, 11, 13, 14, 16, 17, 18, 20, 21,
- 22, 24, 25, 26, 28, 29, 30, 32, 33, 34, 36, 37, 38, 40, 41, 42,
- 44, 45, 46, 47, 49, 50, 51, 52, 54, 55, 56, 57, 59, 60, 61, 62,
- 63, 65, 66, 67, 68, 69, 71, 72, 73, 74, 75, 77, 78, 79, 80, 81,
- 82, 84, 85, 86, 87, 88, 89, 90, 92, 93, 94, 95, 96, 97, 98, 99,
- 100, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 116, 117,
- 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133,
- 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
- 150, 151, 152, 153, 154, 155, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
- 165, 166, 167, 168, 169, 169, 170, 171, 172, 173, 174, 175, 176, 177, 178, 178,
- 179, 180, 181, 182, 183, 184, 185, 185, 186, 187, 188, 189, 190, 191, 192, 192,
- 193, 194, 195, 196, 197, 198, 198, 199, 200, 201, 202, 203, 203, 204, 205, 206,
- 207, 208, 208, 209, 210, 211, 212, 212, 213, 214, 215, 216, 216, 217, 218, 219,
- 220, 220, 221, 222, 223, 224, 224, 225, 226, 227, 228, 228, 229, 230, 231, 231,
- 232, 233, 234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 244,
- 244, 245, 246, 247, 247, 248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
-};
-
-static const WebRtc_Word16 kPowTableFrac[1024] = {
- 0, 1, 1, 2, 3, 3, 4, 5,
- 6, 6, 7, 8, 8, 9, 10, 10,
- 11, 12, 13, 13, 14, 15, 15, 16,
- 17, 17, 18, 19, 20, 20, 21, 22,
- 22, 23, 24, 25, 25, 26, 27, 27,
- 28, 29, 30, 30, 31, 32, 32, 33,
- 34, 35, 35, 36, 37, 37, 38, 39,
- 40, 40, 41, 42, 42, 43, 44, 45,
- 45, 46, 47, 48, 48, 49, 50, 50,
- 51, 52, 53, 53, 54, 55, 56, 56,
- 57, 58, 58, 59, 60, 61, 61, 62,
- 63, 64, 64, 65, 66, 67, 67, 68,
- 69, 69, 70, 71, 72, 72, 73, 74,
- 75, 75, 76, 77, 78, 78, 79, 80,
- 81, 81, 82, 83, 84, 84, 85, 86,
- 87, 87, 88, 89, 90, 90, 91, 92,
- 93, 93, 94, 95, 96, 96, 97, 98,
- 99, 100, 100, 101, 102, 103, 103, 104,
- 105, 106, 106, 107, 108, 109, 109, 110,
- 111, 112, 113, 113, 114, 115, 116, 116,
- 117, 118, 119, 119, 120, 121, 122, 123,
- 123, 124, 125, 126, 126, 127, 128, 129,
- 130, 130, 131, 132, 133, 133, 134, 135,
- 136, 137, 137, 138, 139, 140, 141, 141,
- 142, 143, 144, 144, 145, 146, 147, 148,
- 148, 149, 150, 151, 152, 152, 153, 154,
- 155, 156, 156, 157, 158, 159, 160, 160,
- 161, 162, 163, 164, 164, 165, 166, 167,
- 168, 168, 169, 170, 171, 172, 173, 173,
- 174, 175, 176, 177, 177, 178, 179, 180,
- 181, 181, 182, 183, 184, 185, 186, 186,
- 187, 188, 189, 190, 190, 191, 192, 193,
- 194, 195, 195, 196, 197, 198, 199, 200,
- 200, 201, 202, 203, 204, 205, 205, 206,
- 207, 208, 209, 210, 210, 211, 212, 213,
- 214, 215, 215, 216, 217, 218, 219, 220,
- 220, 221, 222, 223, 224, 225, 225, 226,
- 227, 228, 229, 230, 231, 231, 232, 233,
- 234, 235, 236, 237, 237, 238, 239, 240,
- 241, 242, 243, 243, 244, 245, 246, 247,
- 248, 249, 249, 250, 251, 252, 253, 254,
- 255, 255, 256, 257, 258, 259, 260, 261,
- 262, 262, 263, 264, 265, 266, 267, 268,
- 268, 269, 270, 271, 272, 273, 274, 275,
- 276, 276, 277, 278, 279, 280, 281, 282,
- 283, 283, 284, 285, 286, 287, 288, 289,
- 290, 291, 291, 292, 293, 294, 295, 296,
- 297, 298, 299, 299, 300, 301, 302, 303,
- 304, 305, 306, 307, 308, 308, 309, 310,
- 311, 312, 313, 314, 315, 316, 317, 318,
- 318, 319, 320, 321, 322, 323, 324, 325,
- 326, 327, 328, 328, 329, 330, 331, 332,
- 333, 334, 335, 336, 337, 338, 339, 339,
- 340, 341, 342, 343, 344, 345, 346, 347,
- 348, 349, 350, 351, 352, 352, 353, 354,
- 355, 356, 357, 358, 359, 360, 361, 362,
- 363, 364, 365, 366, 367, 367, 368, 369,
- 370, 371, 372, 373, 374, 375, 376, 377,
- 378, 379, 380, 381, 382, 383, 384, 385,
- 385, 386, 387, 388, 389, 390, 391, 392,
- 393, 394, 395, 396, 397, 398, 399, 400,
- 401, 402, 403, 404, 405, 406, 407, 408,
- 409, 410, 410, 411, 412, 413, 414, 415,
- 416, 417, 418, 419, 420, 421, 422, 423,
- 424, 425, 426, 427, 428, 429, 430, 431,
- 432, 433, 434, 435, 436, 437, 438, 439,
- 440, 441, 442, 443, 444, 445, 446, 447,
- 448, 449, 450, 451, 452, 453, 454, 455,
- 456, 457, 458, 459, 460, 461, 462, 463,
- 464, 465, 466, 467, 468, 469, 470, 471,
- 472, 473, 474, 475, 476, 477, 478, 479,
- 480, 481, 482, 483, 484, 485, 486, 487,
- 488, 489, 490, 491, 492, 493, 494, 495,
- 496, 498, 499, 500, 501, 502, 503, 504,
- 505, 506, 507, 508, 509, 510, 511, 512,
- 513, 514, 515, 516, 517, 518, 519, 520,
- 521, 522, 523, 525, 526, 527, 528, 529,
- 530, 531, 532, 533, 534, 535, 536, 537,
- 538, 539, 540, 541, 542, 544, 545, 546,
- 547, 548, 549, 550, 551, 552, 553, 554,
- 555, 556, 557, 558, 560, 561, 562, 563,
- 564, 565, 566, 567, 568, 569, 570, 571,
- 572, 574, 575, 576, 577, 578, 579, 580,
- 581, 582, 583, 584, 585, 587, 588, 589,
- 590, 591, 592, 593, 594, 595, 596, 597,
- 599, 600, 601, 602, 603, 604, 605, 606,
- 607, 608, 610, 611, 612, 613, 614, 615,
- 616, 617, 618, 620, 621, 622, 623, 624,
- 625, 626, 627, 628, 630, 631, 632, 633,
- 634, 635, 636, 637, 639, 640, 641, 642,
- 643, 644, 645, 646, 648, 649, 650, 651,
- 652, 653, 654, 656, 657, 658, 659, 660,
- 661, 662, 664, 665, 666, 667, 668, 669,
- 670, 672, 673, 674, 675, 676, 677, 678,
- 680, 681, 682, 683, 684, 685, 687, 688,
- 689, 690, 691, 692, 693, 695, 696, 697,
- 698, 699, 700, 702, 703, 704, 705, 706,
- 708, 709, 710, 711, 712, 713, 715, 716,
- 717, 718, 719, 720, 722, 723, 724, 725,
- 726, 728, 729, 730, 731, 732, 733, 735,
- 736, 737, 738, 739, 741, 742, 743, 744,
- 745, 747, 748, 749, 750, 751, 753, 754,
- 755, 756, 757, 759, 760, 761, 762, 763,
- 765, 766, 767, 768, 770, 771, 772, 773,
- 774, 776, 777, 778, 779, 780, 782, 783,
- 784, 785, 787, 788, 789, 790, 792, 793,
- 794, 795, 796, 798, 799, 800, 801, 803,
- 804, 805, 806, 808, 809, 810, 811, 813,
- 814, 815, 816, 818, 819, 820, 821, 823,
- 824, 825, 826, 828, 829, 830, 831, 833,
- 834, 835, 836, 838, 839, 840, 841, 843,
- 844, 845, 846, 848, 849, 850, 851, 853,
- 854, 855, 857, 858, 859, 860, 862, 863,
- 864, 866, 867, 868, 869, 871, 872, 873,
- 874, 876, 877, 878, 880, 881, 882, 883,
- 885, 886, 887, 889, 890, 891, 893, 894,
- 895, 896, 898, 899, 900, 902, 903, 904,
- 906, 907, 908, 909, 911, 912, 913, 915,
- 916, 917, 919, 920, 921, 923, 924, 925,
- 927, 928, 929, 931, 932, 933, 935, 936,
- 937, 938, 940, 941, 942, 944, 945, 946,
- 948, 949, 950, 952, 953, 955, 956, 957,
- 959, 960, 961, 963, 964, 965, 967, 968,
- 969, 971, 972, 973, 975, 976, 977, 979,
- 980, 981, 983, 984, 986, 987, 988, 990,
- 991, 992, 994, 995, 996, 998, 999, 1001,
- 1002, 1003, 1005, 1006, 1007, 1009, 1010, 1012,
- 1013, 1014, 1016, 1017, 1018, 1020, 1021, 1023
-};
-
-static const WebRtc_Word16 kIndicatorTable[17] = {0, 2017, 3809, 5227, 6258, 6963, 7424, 7718,
- 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks80w128x[128] = {
- 0, 536, 1072, 1606, 2139, 2669, 3196, 3720, 4240, 4756, 5266,
- 5771, 6270, 6762, 7246, 7723, 8192, 8652, 9102, 9543, 9974, 10394,
- 10803, 11200, 11585, 11958, 12318, 12665, 12998, 13318, 13623, 13913, 14189,
- 14449, 14694, 14924, 15137, 15334, 15515, 15679, 15826, 15956, 16069, 16165,
- 16244, 16305, 16349, 16375, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16375, 16349, 16305, 16244, 16165, 16069, 15956,
- 15826, 15679, 15515, 15334, 15137, 14924, 14694, 14449, 14189, 13913, 13623,
- 13318, 12998, 12665, 12318, 11958, 11585, 11200, 10803, 10394, 9974, 9543,
- 9102, 8652, 8192, 7723, 7246, 6762, 6270, 5771, 5266, 4756, 4240,
- 3720, 3196, 2669, 2139, 1606, 1072, 536
-};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks160w256x[256] = {
- 0, 268, 536, 804, 1072, 1339, 1606, 1872,
- 2139, 2404, 2669, 2933, 3196, 3459, 3720, 3981,
- 4240, 4499, 4756, 5012, 5266, 5520, 5771, 6021,
- 6270, 6517, 6762, 7005, 7246, 7486, 7723, 7959,
- 8192, 8423, 8652, 8878, 9102, 9324, 9543, 9760,
- 9974, 10185, 10394, 10600, 10803, 11003, 11200, 11394,
-11585, 11773, 11958, 12140, 12318, 12493, 12665, 12833,
-12998, 13160, 13318, 13472, 13623, 13770, 13913, 14053,
-14189, 14321, 14449, 14574, 14694, 14811, 14924, 15032,
-15137, 15237, 15334, 15426, 15515, 15599, 15679, 15754,
-15826, 15893, 15956, 16015, 16069, 16119, 16165, 16207,
-16244, 16277, 16305, 16329, 16349, 16364, 16375, 16382,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16382, 16375, 16364, 16349, 16329, 16305, 16277,
-16244, 16207, 16165, 16119, 16069, 16015, 15956, 15893,
-15826, 15754, 15679, 15599, 15515, 15426, 15334, 15237,
-15137, 15032, 14924, 14811, 14694, 14574, 14449, 14321,
-14189, 14053, 13913, 13770, 13623, 13472, 13318, 13160,
-12998, 12833, 12665, 12493, 12318, 12140, 11958, 11773,
-11585, 11394, 11200, 11003, 10803, 10600, 10394, 10185,
- 9974, 9760, 9543, 9324, 9102, 8878, 8652, 8423,
- 8192, 7959, 7723, 7486, 7246, 7005, 6762, 6517,
- 6270, 6021, 5771, 5520, 5266, 5012, 4756, 4499,
- 4240, 3981, 3720, 3459, 3196, 2933, 2669, 2404,
- 2139, 1872, 1606, 1339, 1072, 804, 536, 268
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor1Table[257] = {
- 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8233, 8274, 8315, 8355, 8396, 8436, 8475, 8515, 8554, 8592, 8631, 8669,
- 8707, 8745, 8783, 8820, 8857, 8894, 8931, 8967, 9003, 9039, 9075, 9111, 9146, 9181,
- 9216, 9251, 9286, 9320, 9354, 9388, 9422, 9456, 9489, 9523, 9556, 9589, 9622, 9655,
- 9687, 9719, 9752, 9784, 9816, 9848, 9879, 9911, 9942, 9973, 10004, 10035, 10066,
- 10097, 10128, 10158, 10188, 10218, 10249, 10279, 10308, 10338, 10368, 10397, 10426,
- 10456, 10485, 10514, 10543, 10572, 10600, 10629, 10657, 10686, 10714, 10742, 10770,
- 10798, 10826, 10854, 10882, 10847, 10810, 10774, 10737, 10701, 10666, 10631, 10596,
- 10562, 10527, 10494, 10460, 10427, 10394, 10362, 10329, 10297, 10266, 10235, 10203,
- 10173, 10142, 10112, 10082, 10052, 10023, 9994, 9965, 9936, 9908, 9879, 9851, 9824,
- 9796, 9769, 9742, 9715, 9689, 9662, 9636, 9610, 9584, 9559, 9534, 9508, 9484, 9459,
- 9434, 9410, 9386, 9362, 9338, 9314, 9291, 9268, 9245, 9222, 9199, 9176, 9154, 9132,
- 9110, 9088, 9066, 9044, 9023, 9002, 8980, 8959, 8939, 8918, 8897, 8877, 8857, 8836,
- 8816, 8796, 8777, 8757, 8738, 8718, 8699, 8680, 8661, 8642, 8623, 8605, 8586, 8568,
- 8550, 8532, 8514, 8496, 8478, 8460, 8443, 8425, 8408, 8391, 8373, 8356, 8339, 8323,
- 8306, 8289, 8273, 8256, 8240, 8224, 8208, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness1[257] = {
- 7577, 7577, 7577, 7577, 7577, 7577,
- 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness2[257] = {
- 7270, 7270, 7270, 7270, 7270, 7306,
- 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness3[257] = {
- 7184, 7184, 7184, 7229, 7270, 7306,
- 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// sum of log2(i) from table index to inst->anaLen2 in Q5
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumLogIndex[66] = {
- 0, 22917, 22917, 22885, 22834, 22770, 22696, 22613,
- 22524, 22428, 22326, 22220, 22109, 21994, 21876, 21754,
- 21629, 21501, 21370, 21237, 21101, 20963, 20822, 20679,
- 20535, 20388, 20239, 20089, 19937, 19783, 19628, 19470,
- 19312, 19152, 18991, 18828, 18664, 18498, 18331, 18164,
- 17994, 17824, 17653, 17480, 17306, 17132, 16956, 16779,
- 16602, 16423, 16243, 16063, 15881, 15699, 15515, 15331,
- 15146, 14960, 14774, 14586, 14398, 14209, 14019, 13829,
- 13637, 13445
-};
-
-// sum of log2(i)^2 from table index to inst->anaLen2 in Q2
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumSquareLogIndex[66] = {
- 0, 16959, 16959, 16955, 16945, 16929, 16908, 16881,
- 16850, 16814, 16773, 16729, 16681, 16630, 16575, 16517,
- 16456, 16392, 16325, 16256, 16184, 16109, 16032, 15952,
- 15870, 15786, 15700, 15612, 15521, 15429, 15334, 15238,
- 15140, 15040, 14938, 14834, 14729, 14622, 14514, 14404,
- 14292, 14179, 14064, 13947, 13830, 13710, 13590, 13468,
- 13344, 13220, 13094, 12966, 12837, 12707, 12576, 12444,
- 12310, 12175, 12039, 11902, 11763, 11624, 11483, 11341,
- 11198, 11054
-};
-
-// log2(table index) in Q12
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kLogIndex[129] = {
- 0, 0, 4096, 6492, 8192, 9511, 10588, 11499,
- 12288, 12984, 13607, 14170, 14684, 15157, 15595, 16003,
- 16384, 16742, 17080, 17400, 17703, 17991, 18266, 18529,
- 18780, 19021, 19253, 19476, 19691, 19898, 20099, 20292,
- 20480, 20662, 20838, 21010, 21176, 21338, 21496, 21649,
- 21799, 21945, 22087, 22226, 22362, 22495, 22625, 22752,
- 22876, 22998, 23117, 23234, 23349, 23462, 23572, 23680,
- 23787, 23892, 23994, 24095, 24195, 24292, 24388, 24483,
- 24576, 24668, 24758, 24847, 24934, 25021, 25106, 25189,
- 25272, 25354, 25434, 25513, 25592, 25669, 25745, 25820,
- 25895, 25968, 26041, 26112, 26183, 26253, 26322, 26390,
- 26458, 26525, 26591, 26656, 26721, 26784, 26848, 26910,
- 26972, 27033, 27094, 27154, 27213, 27272, 27330, 27388,
- 27445, 27502, 27558, 27613, 27668, 27722, 27776, 27830,
- 27883, 27935, 27988, 28039, 28090, 28141, 28191, 28241,
- 28291, 28340, 28388, 28437, 28484, 28532, 28579, 28626,
- 28672
-};
-
-// determinant of estimation matrix in Q0 corresponding to the log2 tables above
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kDeterminantEstMatrix[66] = {
- 0, 29814, 25574, 22640, 20351, 18469, 16873, 15491,
- 14277, 13199, 12233, 11362, 10571, 9851, 9192, 8587,
- 8030, 7515, 7038, 6596, 6186, 5804, 5448, 5115,
- 4805, 4514, 4242, 3988, 3749, 3524, 3314, 3116,
- 2930, 2755, 2590, 2435, 2289, 2152, 2022, 1900,
- 1785, 1677, 1575, 1478, 1388, 1302, 1221, 1145,
- 1073, 1005, 942, 881, 825, 771, 721, 674,
- 629, 587, 547, 510, 475, 442, 411, 382,
- 355, 330
-};
-
-void WebRtcNsx_UpdateNoiseEstimate(NsxInst_t *inst, int offset)
-{
- WebRtc_Word32 tmp32no1 = 0;
- WebRtc_Word32 tmp32no2 = 0;
-
- WebRtc_Word16 tmp16no1 = 0;
- WebRtc_Word16 tmp16no2 = 0;
- WebRtc_Word16 exp2Const = 11819; // Q13
-
- int i = 0;
-
- tmp16no2 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset, inst->magnLen);
- inst->qNoise = 14
- - (int)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(exp2Const, tmp16no2, 21);
- for (i = 0; i < inst->magnLen; i++)
- {
- // inst->quantile[i]=exp(inst->lquantile[offset+i]);
- // in Q21
- tmp32no2 = WEBRTC_SPL_MUL_16_16(exp2Const, inst->noiseEstLogQuantile[offset + i]);
- tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF));
- tmp16no1 = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
- tmp16no1 += 21;// shift 21 to get result in Q0
- tmp16no1 -= (WebRtc_Word16)inst->qNoise; //shift to get result in Q(qNoise)
- if (tmp16no1 > 0)
- {
- inst->noiseEstQuantile[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1 +
- kRoundTable[tmp16no1], tmp16no1);
- }
- else
- {
- inst->noiseEstQuantile[i] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32no1,
- -tmp16no1);
- }
- }
-}
-
-void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t *inst,
- WebRtc_Word16 pink_noise_exp_avg,
- WebRtc_Word32 pink_noise_num_avg,
- int freq_index,
- WebRtc_UWord32 *noise_estimate,
- WebRtc_UWord32 *noise_estimate_avg)
-{
- WebRtc_Word32 tmp32no1 = 0;
- WebRtc_Word32 tmp32no2 = 0;
-
- WebRtc_Word16 int_part = 0;
- WebRtc_Word16 frac_part = 0;
-
- // Use pink noise estimate
- // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
- assert(freq_index > 0);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(pink_noise_exp_avg, kLogIndex[freq_index]); // Q26
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 15); // Q11
- tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
-
- // Calculate output: 2^tmp32no1
- // Output in Q(minNorm-stages)
- tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11);
- if (tmp32no1 > 0)
- {
- int_part = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
- frac_part = (WebRtc_Word16)(tmp32no1 & 0x000007ff); // Q11
- // Piecewise linear approximation of 'b' in
- // 2^(int_part+frac_part) = 2^int_part * (1 + b)
- // 'b' is given in Q11 and below stored in frac_part.
- if (WEBRTC_SPL_RSHIFT_W32(frac_part, 10))
- {
- // Upper fractional part
- tmp32no2 = WEBRTC_SPL_MUL_32_16(2048 - frac_part, 1244); // Q21
- tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
- }
- else
- {
- // Lower fractional part
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(frac_part, 804), 10);
- }
- // Shift fractional part to Q(minNorm-stages)
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
- *noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (WebRtc_UWord32)tmp32no2;
- // Scale up to initMagnEst, which is not block averaged
- *noise_estimate = (*noise_estimate_avg) * (WebRtc_UWord32)(inst->blockIndex + 1);
- }
-}
-
-// Initialize state
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t *inst, WebRtc_UWord32 fs)
-{
- int i;
-
- //check for valid pointer
- if (inst == NULL)
- {
- return -1;
- }
- //
-
- // Initialization of struct
- if (fs == 8000 || fs == 16000 || fs == 32000)
- {
- inst->fs = fs;
- } else
- {
- return -1;
- }
-
- if (fs == 8000)
- {
- inst->blockLen10ms = 80;
- inst->anaLen = 128;
- inst->stages = 7;
- inst->window = kBlocks80w128x;
- inst->thresholdLogLrt = 131072; //default threshold for LRT feature
- inst->maxLrt = 0x0040000;
- inst->minLrt = 52429;
- } else if (fs == 16000)
- {
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->stages = 8;
- inst->window = kBlocks160w256x;
- inst->thresholdLogLrt = 212644; //default threshold for LRT feature
- inst->maxLrt = 0x0080000;
- inst->minLrt = 104858;
- } else if (fs == 32000)
- {
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->stages = 8;
- inst->window = kBlocks160w256x;
- inst->thresholdLogLrt = 212644; //default threshold for LRT feature
- inst->maxLrt = 0x0080000;
- inst->minLrt = 104858;
- }
- inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
- inst->magnLen = inst->anaLen2 + 1;
-
- WebRtcSpl_ZerosArrayW16(inst->analysisBuffer, ANAL_BLOCKL_MAX);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer, ANAL_BLOCKL_MAX);
-
- // for HB processing
- WebRtcSpl_ZerosArrayW16(inst->dataBufHBFX, ANAL_BLOCKL_MAX);
- // for quantile noise estimation
- WebRtcSpl_ZerosArrayW16(inst->noiseEstQuantile, HALF_ANAL_BLOCKL);
- for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++)
- {
- inst->noiseEstLogQuantile[i] = 2048; // Q8
- inst->noiseEstDensity[i] = 153; // Q9
- }
- for (i = 0; i < SIMULT; i++)
- {
- inst->noiseEstCounter[i] = (WebRtc_Word16)(END_STARTUP_LONG * (i + 1)) / SIMULT;
- }
-
- // Initialize suppression filter with ones
- WebRtcSpl_MemSetW16((WebRtc_Word16*)inst->noiseSupFilter, 16384, HALF_ANAL_BLOCKL);
-
- // Set the aggressiveness: default
- inst->aggrMode = 0;
-
- //initialize variables for new method
- inst->priorNonSpeechProb = 8192; // Q14(0.5) prior probability for speech/noise
- for (i = 0; i < HALF_ANAL_BLOCKL; i++)
- {
- inst->prevMagnU16[i] = 0;
- inst->prevNoiseU32[i] = 0; //previous noise-spectrum
- inst->logLrtTimeAvgW32[i] = 0; //smooth LR ratio
- inst->avgMagnPause[i] = 0; //conservative noise spectrum estimate
- inst->initMagnEst[i] = 0; //initial average magnitude spectrum
- }
-
- //feature quantities
- inst->thresholdSpecDiff = 50; //threshold for difference feature: determined on-line
- inst->thresholdSpecFlat = 20480; //threshold for flatness: determined on-line
- inst->featureLogLrt = inst->thresholdLogLrt; //average LRT factor (= threshold)
- inst->featureSpecFlat = inst->thresholdSpecFlat; //spectral flatness (= threshold)
- inst->featureSpecDiff = inst->thresholdSpecDiff; //spectral difference (= threshold)
- inst->weightLogLrt = 6; //default weighting par for LRT feature
- inst->weightSpecFlat = 0; //default weighting par for spectral flatness feature
- inst->weightSpecDiff = 0; //default weighting par for spectral difference feature
-
- inst->curAvgMagnEnergy = 0; //window time-average of input magnitude spectrum
- inst->timeAvgMagnEnergy = 0; //normalization for spectral difference
- inst->timeAvgMagnEnergyTmp = 0; //normalization for spectral difference
-
- //histogram quantities: used to estimate/update thresholds for features
- WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
-
- inst->blockIndex = -1; //frame counter
-
- //inst->modelUpdate = 500; //window for update
- inst->modelUpdate = (1 << STAT_UPDATES); //window for update
- inst->cntThresUpdate = 0; //counter feature thresholds updates
-
- inst->sumMagn = 0;
- inst->magnEnergy = 0;
- inst->prevQMagn = 0;
- inst->qNoise = 0;
- inst->prevQNoise = 0;
-
- inst->energyIn = 0;
- inst->scaleEnergyIn = 0;
-
- inst->whiteNoiseLevel = 0;
- inst->pinkNoiseNumerator = 0;
- inst->pinkNoiseExp = 0;
- inst->minNorm = 15; // Start with full scale
- inst->zeroInputSignal = 0;
-
- //default mode
- WebRtcNsx_set_policy_core(inst, 0);
-
-#ifdef NS_FILEDEBUG
- inst->infile=fopen("indebug.pcm","wb");
- inst->outfile=fopen("outdebug.pcm","wb");
- inst->file1=fopen("file1.pcm","wb");
- inst->file2=fopen("file2.pcm","wb");
- inst->file3=fopen("file3.pcm","wb");
- inst->file4=fopen("file4.pcm","wb");
- inst->file5=fopen("file5.pcm","wb");
-#endif
-
- inst->initFlag = 1;
-
- return 0;
-}
-
-int WebRtcNsx_set_policy_core(NsxInst_t *inst, int mode)
-{
- // allow for modes:0,1,2,3
- if (mode < 0 || mode > 3)
- {
- return -1;
- }
-
- inst->aggrMode = mode;
- if (mode == 0)
- {
- inst->overdrive = 256; // Q8(1.0)
- inst->denoiseBound = 8192; // Q14(0.5)
- inst->gainMap = 0; // No gain compensation
- } else if (mode == 1)
- {
- inst->overdrive = 256; // Q8(1.0)
- inst->denoiseBound = 4096; // Q14(0.25)
- inst->factor2Table = kFactor2Aggressiveness1;
- inst->gainMap = 1;
- } else if (mode == 2)
- {
- inst->overdrive = 282; // ~= Q8(1.1)
- inst->denoiseBound = 2048; // Q14(0.125)
- inst->factor2Table = kFactor2Aggressiveness2;
- inst->gainMap = 1;
- } else if (mode == 3)
- {
- inst->overdrive = 320; // Q8(1.25)
- inst->denoiseBound = 1475; // ~= Q14(0.09)
- inst->factor2Table = kFactor2Aggressiveness3;
- inst->gainMap = 1;
- }
- return 0;
-}
-
-void WebRtcNsx_NoiseEstimation(NsxInst_t *inst, WebRtc_UWord16 *magn, WebRtc_UWord32 *noise,
- WebRtc_Word16 *qNoise)
-{
- WebRtc_Word32 numerator;
-
- WebRtc_Word16 lmagn[HALF_ANAL_BLOCKL], counter, countDiv, countProd, delta, zeros, frac;
- WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
- WebRtc_Word16 log2Const = 22713; // Q15
- WebRtc_Word16 widthFactor = 21845;
-
- int i, s, offset;
-
- numerator = FACTOR_Q16;
-
- tabind = inst->stages - inst->normData;
- if (tabind < 0)
- {
- logval = -kLogTable[-tabind];
- } else
- {
- logval = kLogTable[tabind];
- }
-
- // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
- // magn is in Q(-stages), and the real lmagn values are:
- // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
- // lmagn in Q8
- for (i = 0; i < inst->magnLen; i++)
- {
- if (magn[i])
- {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magn[i] << zeros) & 0x7FFFFFFF) >> 23);
- // log2(magn(i))
- log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
- // log2(magn(i))*log(2)
- lmagn[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2Const, 15);
- // + log(2^stages)
- lmagn[i] += logval;
- } else
- {
- lmagn[i] = logval;//0;
- }
- }
-
- // loop over simultaneous estimates
- for (s = 0; s < SIMULT; s++)
- {
- offset = s * inst->magnLen;
-
- // Get counter values from state
- counter = inst->noiseEstCounter[s];
- countDiv = kCounterDiv[counter];
- countProd = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(counter, countDiv);
-
- // quant_est(...)
- for (i = 0; i < inst->magnLen; i++)
- {
- // compute delta
- if (inst->noiseEstDensity[offset + i] > 512)
- {
- delta = WebRtcSpl_DivW32W16ResW16(numerator,
- inst->noiseEstDensity[offset + i]);
- } else
- {
- delta = FACTOR_Q7;
- }
-
- // update log quantile estimate
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
- if (lmagn[i] > inst->noiseEstLogQuantile[offset + i])
- {
- // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
- // CounterDiv=1/inst->counter[s] in Q15
- tmp16 += 2;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
- inst->noiseEstLogQuantile[offset + i] += tmp16no1;
- } else
- {
- tmp16 += 1;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
- // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
- inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
- }
-
- // update density estimate
- if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
- < WIDTH_Q8)
- {
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- inst->noiseEstDensity[offset + i], countProd, 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(widthFactor,
- countDiv, 15);
- inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
- }
- } // end loop over magnitude spectrum
-
- if (counter >= END_STARTUP_LONG)
- {
- inst->noiseEstCounter[s] = 0;
- if (inst->blockIndex >= END_STARTUP_LONG)
- {
- WebRtcNsx_UpdateNoiseEstimate(inst, offset);
- }
- }
- inst->noiseEstCounter[s]++;
-
- } // end loop over simultaneous estimates
-
- // Sequentially update the noise during startup
- if (inst->blockIndex < END_STARTUP_LONG)
- {
- WebRtcNsx_UpdateNoiseEstimate(inst, offset);
- }
-
- for (i = 0; i < inst->magnLen; i++)
- {
- noise[i] = (WebRtc_UWord32)(inst->noiseEstQuantile[i]); // Q(qNoise)
- }
- (*qNoise) = (WebRtc_Word16)inst->qNoise;
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by window_pars)
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNsx_FeatureParameterExtraction(NsxInst_t *inst, int flag)
-{
- WebRtc_UWord32 tmpU32;
- WebRtc_UWord32 histIndex;
- WebRtc_UWord32 posPeak1SpecFlatFX, posPeak2SpecFlatFX;
- WebRtc_UWord32 posPeak1SpecDiffFX, posPeak2SpecDiffFX;
-
- WebRtc_Word32 tmp32;
- WebRtc_Word32 fluctLrtFX, thresFluctLrtFX;
- WebRtc_Word32 avgHistLrtFX, avgSquareHistLrtFX, avgHistLrtComplFX;
-
- WebRtc_Word16 j;
- WebRtc_Word16 numHistLrt;
-
- int i;
- int useFeatureSpecFlat, useFeatureSpecDiff, featureSum;
- int maxPeak1, maxPeak2;
- int weightPeak1SpecFlat, weightPeak2SpecFlat;
- int weightPeak1SpecDiff, weightPeak2SpecDiff;
-
- //update histograms
- if (!flag)
- {
- // LRT
- // Type casting to UWord32 is safe since negative values will not be wrapped to larger
- // values than HIST_PAR_EST
- histIndex = (WebRtc_UWord32)(inst->featureLogLrt);
- if (histIndex < HIST_PAR_EST)
- {
- inst->histLrt[histIndex]++;
- }
- // Spectral flatness
- // (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
- histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
- if (histIndex < HIST_PAR_EST)
- {
- inst->histSpecFlat[histIndex]++;
- }
- // Spectral difference
- histIndex = HIST_PAR_EST;
- if (inst->timeAvgMagnEnergy)
- {
- // Guard against division by zero
- // If timeAvgMagnEnergy == 0 we have no normalizing statistics and therefore can't
- // update the histogram
- histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
- inst->timeAvgMagnEnergy);
- }
- if (histIndex < HIST_PAR_EST)
- {
- inst->histSpecDiff[histIndex]++;
- }
- }
-
- // extract parameters for speech/noise probability
- if (flag)
- {
- useFeatureSpecDiff = 1;
- //for LRT feature:
- // compute the average over inst->featureExtractionParams.rangeAvgHistLrt
- avgHistLrtFX = 0;
- avgSquareHistLrtFX = 0;
- numHistLrt = 0;
- for (i = 0; i < BIN_SIZE_LRT; i++)
- {
- j = (2 * i + 1);
- tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
- avgHistLrtFX += tmp32;
- numHistLrt += inst->histLrt[i];
- avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
- }
- avgHistLrtComplFX = avgHistLrtFX;
- for (; i < HIST_PAR_EST; i++)
- {
- j = (2 * i + 1);
- tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
- avgHistLrtComplFX += tmp32;
- avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
- }
- fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
- fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
- thresFluctLrtFX = THRES_FLUCT_LRT * numHistLrt;
- // get threshold for LRT feature:
- tmpU32 = (FACTOR_1_LRT_DIFF * (WebRtc_UWord32)avgHistLrtFX);
- if ((fluctLrtFX < thresFluctLrtFX) || (numHistLrt == 0) || (tmpU32
- > (WebRtc_UWord32)(100 * numHistLrt)))
- {
- inst->thresholdLogLrt = inst->maxLrt; //very low fluctuation, so likely noise
- } else
- {
- tmp32 = (WebRtc_Word32)((tmpU32 << (9 + inst->stages)) / numHistLrt / 25);
- // check if value is within min/max range
- inst->thresholdLogLrt = WEBRTC_SPL_SAT(inst->maxLrt, tmp32, inst->minLrt);
- }
- if (fluctLrtFX < thresFluctLrtFX)
- {
- // Do not use difference feature if fluctuation of LRT feature is very low:
- // most likely just noise state
- useFeatureSpecDiff = 0;
- }
-
- // for spectral flatness and spectral difference: compute the main peaks of histogram
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecFlatFX = 0;
- posPeak2SpecFlatFX = 0;
- weightPeak1SpecFlat = 0;
- weightPeak2SpecFlat = 0;
-
- // peaks for flatness
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- if (inst->histSpecFlat[i] > maxPeak1)
- {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecFlat = weightPeak1SpecFlat;
- posPeak2SpecFlatFX = posPeak1SpecFlatFX;
-
- maxPeak1 = inst->histSpecFlat[i];
- weightPeak1SpecFlat = inst->histSpecFlat[i];
- posPeak1SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
- } else if (inst->histSpecFlat[i] > maxPeak2)
- {
- // Found new "second" peak
- maxPeak2 = inst->histSpecFlat[i];
- weightPeak2SpecFlat = inst->histSpecFlat[i];
- posPeak2SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
- }
- }
-
- // for spectral flatness feature
- useFeatureSpecFlat = 1;
- // merge the two peaks if they are close
- if ((posPeak1SpecFlatFX - posPeak2SpecFlatFX < LIM_PEAK_SPACE_FLAT_DIFF)
- && (weightPeak2SpecFlat * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecFlat))
- {
- weightPeak1SpecFlat += weightPeak2SpecFlat;
- posPeak1SpecFlatFX = (posPeak1SpecFlatFX + posPeak2SpecFlatFX) >> 1;
- }
- //reject if weight of peaks is not large enough, or peak value too small
- if (weightPeak1SpecFlat < THRES_WEIGHT_FLAT_DIFF || posPeak1SpecFlatFX
- < THRES_PEAK_FLAT)
- {
- useFeatureSpecFlat = 0;
- } else // if selected, get the threshold
- {
- // compute the threshold and check if value is within min/max range
- inst->thresholdSpecFlat = WEBRTC_SPL_SAT(MAX_FLAT_Q10, FACTOR_2_FLAT_Q10
- * posPeak1SpecFlatFX, MIN_FLAT_Q10); //Q10
- }
- // done with flatness feature
-
- if (useFeatureSpecDiff)
- {
- //compute two peaks for spectral difference
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecDiffFX = 0;
- posPeak2SpecDiffFX = 0;
- weightPeak1SpecDiff = 0;
- weightPeak2SpecDiff = 0;
- // peaks for spectral difference
- for (i = 0; i < HIST_PAR_EST; i++)
- {
- if (inst->histSpecDiff[i] > maxPeak1)
- {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecDiff = weightPeak1SpecDiff;
- posPeak2SpecDiffFX = posPeak1SpecDiffFX;
-
- maxPeak1 = inst->histSpecDiff[i];
- weightPeak1SpecDiff = inst->histSpecDiff[i];
- posPeak1SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
- } else if (inst->histSpecDiff[i] > maxPeak2)
- {
- // Found new "second" peak
- maxPeak2 = inst->histSpecDiff[i];
- weightPeak2SpecDiff = inst->histSpecDiff[i];
- posPeak2SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
- }
- }
-
- // merge the two peaks if they are close
- if ((posPeak1SpecDiffFX - posPeak2SpecDiffFX < LIM_PEAK_SPACE_FLAT_DIFF)
- && (weightPeak2SpecDiff * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecDiff))
- {
- weightPeak1SpecDiff += weightPeak2SpecDiff;
- posPeak1SpecDiffFX = (posPeak1SpecDiffFX + posPeak2SpecDiffFX) >> 1;
- }
- // get the threshold value and check if value is within min/max range
- inst->thresholdSpecDiff = WEBRTC_SPL_SAT(MAX_DIFF, FACTOR_1_LRT_DIFF
- * posPeak1SpecDiffFX, MIN_DIFF); //5x bigger
- //reject if weight of peaks is not large enough
- if (weightPeak1SpecDiff < THRES_WEIGHT_FLAT_DIFF)
- {
- useFeatureSpecDiff = 0;
- }
- // done with spectral difference feature
- }
-
- // select the weights between the features
- // inst->priorModelPars[4] is weight for LRT: always selected
- featureSum = 6 / (1 + useFeatureSpecFlat + useFeatureSpecDiff);
- inst->weightLogLrt = featureSum;
- inst->weightSpecFlat = useFeatureSpecFlat * featureSum;
- inst->weightSpecDiff = useFeatureSpecDiff * featureSum;
-
- // set histograms to zero for next update
- WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
- } // end of flag == 1
-}
-
-
-// Compute spectral flatness on input spectrum
-// magn is the magnitude spectrum
-// spectral flatness is returned in inst->featureSpecFlat
-void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t *inst, WebRtc_UWord16 *magn)
-{
- WebRtc_UWord32 tmpU32;
- WebRtc_UWord32 avgSpectralFlatnessNum, avgSpectralFlatnessDen;
-
- WebRtc_Word32 tmp32;
- WebRtc_Word32 currentSpectralFlatness, logCurSpectralFlatness;
-
- WebRtc_Word16 zeros, frac, intPart;
-
- int i;
-
- // for flatness
- avgSpectralFlatnessNum = 0;
- avgSpectralFlatnessDen = inst->sumMagn - (WebRtc_UWord32)magn[0]; // Q(normData-stages)
-
- // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
- // flatness = exp( sum(log(magn[i]))/N - log(sum(magn[i])/N) )
- // = exp( sum(log(magn[i]))/N ) * N / sum(magn[i])
- // = 2^( sum(log2(magn[i]))/N - (log2(sum(magn[i])) - log2(N)) ) [This is used]
- for (i = 1; i < inst->magnLen; i++)
- {
- // First bin is excluded from spectrum measures. Number of bins is now a power of 2
- if (magn[i])
- {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
- frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_UWord32)(magn[i]) << zeros)
- & 0x7FFFFFFF) >> 23);
- // log2(magn(i))
- tmpU32 = (WebRtc_UWord32)(((31 - zeros) << 8) + kLogTableFrac[frac]); // Q8
- avgSpectralFlatnessNum += tmpU32; // Q8
- } else
- {
- //if at least one frequency component is zero, treat separately
- tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
- inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
- return;
- }
- }
- //ratio and inverse log: check for case of log(0)
- zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
- frac = (WebRtc_Word16)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
- // log2(avgSpectralFlatnessDen)
- tmp32 = (WebRtc_Word32)(((31 - zeros) << 8) + kLogTableFrac[frac]); // Q8
- logCurSpectralFlatness = (WebRtc_Word32)avgSpectralFlatnessNum;
- logCurSpectralFlatness += ((WebRtc_Word32)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
- logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
- logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
- tmp32 = (WebRtc_Word32)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
- & 0x0001FFFF)); //Q17
- intPart = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
- intPart += 7; // Shift 7 to get the output in Q10 (from Q17 = -17+10)
- if (intPart > 0)
- {
- currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
- } else
- {
- currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
- }
-
- //time average update of spectral flatness feature
- tmp32 = currentSpectralFlatness - (WebRtc_Word32)inst->featureSpecFlat; // Q10
- tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
- inst->featureSpecFlat = (WebRtc_UWord32)((WebRtc_Word32)inst->featureSpecFlat
- + WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
- // done with flatness feature
-}
-
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magn_tmp is the input spectrum
-// the reference/template spectrum is inst->magn_avg_pause[i]
-// returns (normalized) spectral difference in inst->featureSpecDiff
-void WebRtcNsx_ComputeSpectralDifference(NsxInst_t *inst, WebRtc_UWord16 *magnIn)
-{
- // This is to be calculated:
- // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
- WebRtc_UWord32 varMagnUFX, varPauseUFX, avgDiffNormMagnUFX;
-
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 avgPauseFX, avgMagnFX, covMagnPauseFX;
- WebRtc_Word32 maxPause, minPause;
-
- WebRtc_Word16 tmp16no1;
-
- int i, norm32, nShifts;
-
- avgPauseFX = 0;
- maxPause = 0;
- minPause = inst->avgMagnPause[0]; // Q(prevQMagn)
- // compute average quantities
- for (i = 0; i < inst->magnLen; i++)
- {
- // Compute mean of magn_pause
- avgPauseFX += inst->avgMagnPause[i]; // in Q(prevQMagn)
- maxPause = WEBRTC_SPL_MAX(maxPause, inst->avgMagnPause[i]);
- minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]);
- }
- // normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
- avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
- avgMagnFX = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
- // Largest possible deviation in magnPause for (co)var calculations
- tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
- // Get number of shifts to make sure we don't get wrap around in varPause
- nShifts = WEBRTC_SPL_MAX(0, 10 + inst->stages - WebRtcSpl_NormW32(tmp32no1));
-
- varMagnUFX = 0;
- varPauseUFX = 0;
- covMagnPauseFX = 0;
- for (i = 0; i < inst->magnLen; i++)
- {
- // Compute var and cov of magn and magn_pause
- tmp16no1 = (WebRtc_Word16)((WebRtc_Word32)magnIn[i] - avgMagnFX);
- tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
- varMagnUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
- covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
- varPauseUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
- }
- //update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
- inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
- + inst->stages - 1);
-
- avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
- if ((varPauseUFX) && (covMagnPauseFX))
- {
- tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
- norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
- if (norm32 > 0)
- {
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
- } else
- {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
- }
- tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
-
- nShifts += norm32;
- nShifts <<= 1;
- if (nShifts < 0)
- {
- varPauseUFX >>= (-nShifts); // Q(2*(qMagn+norm32+minPause))
- nShifts = 0;
- }
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX); // Q(2*(qMagn+norm32-16+minPause))
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
-
- avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1); // Q(2*qMagn)
- }
- //normalize and compute time average update of difference feature
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
- if (inst->featureSpecDiff > tmpU32no1)
- {
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
- SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
- inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
- } else
- {
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
- SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
- inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
- }
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//snrLocPrior is the prior SNR for each frequency (in Q11)
-//snrLocPost is the post SNR for each frequency (in Q11)
-void WebRtcNsx_SpeechNoiseProb(NsxInst_t *inst, WebRtc_UWord16 *nonSpeechProbFinal,
- WebRtc_UWord32 *priorLocSnr, WebRtc_UWord32 *postLocSnr)
-{
- WebRtc_UWord32 zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
-
- WebRtc_Word32 invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
- WebRtc_Word32 frac32, logTmp;
- WebRtc_Word32 logLrtTimeAvgKsumFX;
-
- WebRtc_Word16 indPriorFX16;
- WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
-
- int i, normTmp, normTmp2, nShifts;
-
- // compute feature based on average LR factor
- // this is the average over all frequencies of the smooth log LRT
- logLrtTimeAvgKsumFX = 0;
- for (i = 0; i < inst->magnLen; i++)
- {
- besselTmpFX32 = (WebRtc_Word32)postLocSnr[i]; // Q11
- normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
- num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp)
- if (normTmp > 10)
- {
- den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
- } else
- {
- den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
- }
- besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
-
- // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]);
- // Here, LRT_TAVG = 0.5
- zeros = WebRtcSpl_NormU32(priorLocSnr[i]);
- frac32 = (WebRtc_Word32)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19);
- tmp32 = WEBRTC_SPL_MUL(frac32, frac32);
- tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19);
- tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)frac32, 5412, 12);
- frac32 = tmp32 + 37;
- // tmp32 = log2(priorLocSnr[i])
- tmp32 = (WebRtc_Word32)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
- logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12
- inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
-
- logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12
- }
- inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2
- // done with computation of LR factor
-
- //
- //compute the indicator functions
- //
-
- // average LRT feature
- // FLOAT code
- // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0);
- tmpIndFX = 16384; // Q14(1.0)
- tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12
- nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5;
- //use larger width in tanh map for pause regions
- if (tmp32no1 < 0)
- {
- tmpIndFX = 0;
- tmp32no1 = -tmp32no1;
- //widthPrior = widthPrior * 2.0;
- nShifts++;
- }
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
- // compute indicator function: sigmoid map
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14);
- if ((tableIndex < 16) && (tableIndex >= 0))
- {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmp32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
- if (tmpIndFX == 0)
- {
- tmpIndFX = 8192 - tmp16no2; // Q14
- } else
- {
- tmpIndFX = 8192 + tmp16no2; // Q14
- }
- }
- indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14
-
- //spectral flatness feature
- if (inst->weightSpecFlat)
- {
- tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10
- tmpIndFX = 16384; // Q14(1.0)
- //use larger width in tanh map for pause regions
- tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10
- nShifts = 4;
- if (inst->thresholdSpecFlat < tmpU32no1)
- {
- tmpIndFX = 0;
- tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat;
- //widthPrior = widthPrior * 2.0;
- nShifts++;
- }
- tmp32no1 = (WebRtc_Word32)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
- nShifts), 25); //Q14
- tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14
- // compute indicator function: sigmoid map
- // FLOAT code
- // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0);
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
- if (tableIndex < 16)
- {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
- if (tmpIndFX)
- {
- tmpIndFX = 8192 + tmp16no2; // Q14
- } else
- {
- tmpIndFX = 8192 - tmp16no2; // Q14
- }
- }
- indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14
- }
-
- //for template spectral-difference
- if (inst->weightSpecDiff)
- {
- tmpU32no1 = 0;
- if (inst->featureSpecDiff)
- {
- normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
- WebRtcSpl_NormU32(inst->featureSpecDiff));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages
- - normTmp);
- if (tmpU32no2)
- {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q14?? Q(20 - inst->stages)
- } else
- {
- tmpU32no1 = (WebRtc_UWord32)(0x7fffffff);
- }
- }
- tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25);
- tmpU32no2 = tmpU32no1 - tmpU32no3;
- nShifts = 1;
- tmpIndFX = 16384; // Q14(1.0)
- //use larger width in tanh map for pause regions
- if (tmpU32no2 & 0x80000000)
- {
- tmpIndFX = 0;
- tmpU32no2 = tmpU32no3 - tmpU32no1;
- //widthPrior = widthPrior * 2.0;
- nShifts--;
- }
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
- // compute indicator function: sigmoid map
- /* FLOAT code
- indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
- */
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
- if (tableIndex < 16)
- {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16no1, frac,
- 14);
- if (tmpIndFX)
- {
- tmpIndFX = 8192 + tmp16no2;
- } else
- {
- tmpIndFX = 8192 - tmp16no2;
- }
- }
- indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14
- }
-
- //combine the indicator function with the feature weights
- // FLOAT code
- // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2);
- indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14
- // done with computing indicator function
-
- //compute the prior probability
- // FLOAT code
- // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb);
- tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14
- inst->priorNonSpeechProb += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(PRIOR_UPDATE_Q14,
- tmp16, 14); // Q14
-
- //final speech probability: combine prior model with LR factor:
- for (i = 0; i < inst->magnLen; i++)
- {
- // FLOAT code
- // invLrt = exp(inst->logLrtTimeAvg[i]);
- // invLrt = inst->priorSpeechProb * invLrt;
- // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt);
- // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt;
- // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt);
- nonSpeechProbFinal[i] = 0; // Q8
- if ((inst->logLrtTimeAvgW32[i] < 65300) && (inst->priorNonSpeechProb > 0))
- {
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637),
- 14); // Q12
- intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12);
- if (intPart < -8)
- {
- intPart = -8;
- }
- frac = (WebRtc_Word16)(tmp32no1 & 0x00000fff); // Q12
- // Quadratic approximation of 2^frac
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
- tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
- invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
- + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
-
- normTmp = WebRtcSpl_NormW32(invLrtFX);
- normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
- if (normTmp + normTmp2 < 15)
- {
- invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp); // Q(normTmp+normTmp2-7)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q(normTmp+normTmp2+7)
- invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14
- } else
- {
- tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22
- invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
- }
-
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22
- nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1,
- (WebRtc_Word32)inst->priorNonSpeechProb
- + invLrtFX); // Q8
- if (7 - normTmp - normTmp2 > 0)
- {
- nonSpeechProbFinal[i] = 0; // Q8
- }
- }
- }
-}
-
-// Transform input (speechFrame) to frequency domain magnitude (magnU16)
-void WebRtcNsx_DataAnalysis(NsxInst_t *inst, short *speechFrame, WebRtc_UWord16 *magnU16)
-{
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
-
- WebRtc_Word32 tmp_1_w32 = 0;
- WebRtc_Word32 tmp_2_w32 = 0;
- WebRtc_Word32 sum_log_magn = 0;
- WebRtc_Word32 sum_log_i_log_magn = 0;
-
- WebRtc_UWord16 sum_log_magn_u16 = 0;
- WebRtc_UWord16 tmp_u16 = 0;
-
- WebRtc_Word16 sum_log_i = 0;
- WebRtc_Word16 sum_log_i_square = 0;
- WebRtc_Word16 frac = 0;
- WebRtc_Word16 log2 = 0;
- WebRtc_Word16 matrix_determinant = 0;
- WebRtc_Word16 winData[ANAL_BLOCKL_MAX], maxWinData;
- WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
-
- int i, j;
- int outCFFT;
- int zeros;
- int net_norm = 0;
- int right_shifts_in_magnU16 = 0;
- int right_shifts_in_initMagnEst = 0;
-
- // For lower band do all processing
- // update analysis buffer for L band
- WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer + inst->anaLen - inst->blockLen10ms,
- speechFrame, inst->blockLen10ms);
-
- // Window data before FFT
- for (i = 0; i < inst->anaLen; i++)
- {
- winData[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(inst->window[i],
- inst->analysisBuffer[i],
- 14); // Q0
- }
- // Get input energy
- inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
-
- // Reset zero input flag
- inst->zeroInputSignal = 0;
- // Acquire norm for winData
- maxWinData = WebRtcSpl_MaxAbsValueW16(winData, inst->anaLen);
- inst->normData = WebRtcSpl_NormW16(maxWinData);
- if (maxWinData == 0)
- {
- // Treat zero input separately.
- inst->zeroInputSignal = 1;
- return;
- }
-
- // Determine the net normalization in the frequency domain
- net_norm = inst->stages - inst->normData;
- // Track lowest normalization factor and use it to prevent wrap around in shifting
- right_shifts_in_magnU16 = inst->normData - inst->minNorm;
- right_shifts_in_initMagnEst = WEBRTC_SPL_MAX(-right_shifts_in_magnU16, 0);
- inst->minNorm -= right_shifts_in_initMagnEst;
- right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0);
-
- // create realImag as winData interleaved with zeros (= imag. part), normalize it
- for (i = 0; i < inst->anaLen; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W16(i, 1);
- realImag[j] = WEBRTC_SPL_LSHIFT_W16(winData[i], inst->normData); // Q(normData)
- realImag[j + 1] = 0; // Insert zeros in imaginary part
- }
-
- // bit-reverse position of elements in array and FFT the array
- WebRtcSpl_ComplexBitReverse(realImag, inst->stages); // Q(normData-stages)
- outCFFT = WebRtcSpl_ComplexFFT(realImag, inst->stages, 1);
-
- inst->imag[0] = 0; // Q(normData-stages)
- inst->imag[inst->anaLen2] = 0;
- inst->real[0] = realImag[0]; // Q(normData-stages)
- inst->real[inst->anaLen2] = realImag[inst->anaLen];
- // Q(2*(normData-stages))
- inst->magnEnergy = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[0], inst->real[0]);
- inst->magnEnergy += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[inst->anaLen2],
- inst->real[inst->anaLen2]);
- magnU16[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[0]); // Q(normData-stages)
- magnU16[inst->anaLen2] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[inst->anaLen2]);
- inst->sumMagn = (WebRtc_UWord32)magnU16[0]; // Q(normData-stages)
- inst->sumMagn += (WebRtc_UWord32)magnU16[inst->anaLen2];
-
- // Gather information during startup for noise parameter estimation
- if (inst->blockIndex < END_STARTUP_SHORT)
- {
- // Switch initMagnEst to Q(minNorm-stages)
- inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
- right_shifts_in_initMagnEst);
- inst->initMagnEst[inst->anaLen2] =
- WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
- right_shifts_in_initMagnEst); // Q(minNorm-stages)
-
- // Shift magnU16 to same domain as initMagnEst
- tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[0],
- right_shifts_in_magnU16); // Q(minNorm-stages)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[inst->anaLen2],
- right_shifts_in_magnU16); // Q(minNorm-stages)
-
- // Update initMagnEst
- inst->initMagnEst[0] += tmpU32no1; // Q(minNorm-stages)
- inst->initMagnEst[inst->anaLen2] += tmpU32no2; // Q(minNorm-stages)
-
- log2 = 0;
- if (magnU16[inst->anaLen2])
- {
- // Calculate log2(magnU16[inst->anaLen2])
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[inst->anaLen2]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[inst->anaLen2] << zeros) &
- 0x7FFFFFFF) >> 23); // Q8
- // log2(magnU16(i)) in Q8
- log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
- }
-
- sum_log_magn = (WebRtc_Word32)log2; // Q8
- // sum_log_i_log_magn in Q17
- sum_log_i_log_magn = (WEBRTC_SPL_MUL_16_16(kLogIndex[inst->anaLen2], log2) >> 3);
- }
-
- for (i = 1; i < inst->anaLen2; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W16(i, 1);
- inst->real[i] = realImag[j];
- inst->imag[i] = -realImag[j + 1];
- // magnitude spectrum
- // energy in Q(2*(normData-stages))
- tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
- tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
- inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
-
- magnU16[i] = (WebRtc_UWord16)WebRtcSpl_Sqrt(tmpU32no1); // Q(normData-stages)
- inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
- if (inst->blockIndex < END_STARTUP_SHORT)
- {
- // Switch initMagnEst to Q(minNorm-stages)
- inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
- right_shifts_in_initMagnEst);
-
- // Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[i],
- right_shifts_in_magnU16);
- // Update initMagnEst
- inst->initMagnEst[i] += tmpU32no1; // Q(minNorm-stages)
-
- if (i >= kStartBand)
- {
- // For pink noise estimation. Collect data neglecting lower frequency band
- log2 = 0;
- if (magnU16[i])
- {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[i]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[i] << zeros) &
- 0x7FFFFFFF) >> 23);
- // log2(magnU16(i)) in Q8
- log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
- }
- sum_log_magn += (WebRtc_Word32)log2; // Q8
- // sum_log_i_log_magn in Q17
- sum_log_i_log_magn += (WEBRTC_SPL_MUL_16_16(kLogIndex[i], log2) >> 3);
- }
- }
- }
-
- //compute simplified noise model during startup
- if (inst->blockIndex < END_STARTUP_SHORT)
- {
- // Estimate White noise
- // Switch whiteNoiseLevel to Q(minNorm-stages)
- inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
- right_shifts_in_initMagnEst);
-
- // Update the average magnitude spectrum, used as noise estimate.
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
-
- // Replacing division above with 'stages' shifts
- // Shift to same Q-domain as whiteNoiseLevel
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
- // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
- assert(END_STARTUP_SHORT < 128);
- inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
-
- // Estimate Pink noise parameters
- // Denominator used in both parameter estimates.
- // The value is only dependent on the size of the frequency band (kStartBand)
- // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
- matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
- sum_log_i = kSumLogIndex[kStartBand]; // Q5
- sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
- if (inst->fs == 8000)
- {
- // Adjust values to shorter blocks in narrow band.
- tmp_1_w32 = (WebRtc_Word32)matrix_determinant;
- tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
- tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
- tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4);
- tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)(inst->magnLen
- - kStartBand), kSumSquareLogIndex[65], 2);
- matrix_determinant = (WebRtc_Word16)tmp_1_w32;
- sum_log_i -= kSumLogIndex[65]; // Q5
- sum_log_i_square -= kSumSquareLogIndex[65]; // Q2
- }
-
- // Necessary number of shifts to fit sum_log_magn in a word16
- zeros = 16 - WebRtcSpl_NormW32(sum_log_magn);
- if (zeros < 0)
- {
- zeros = 0;
- }
- tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
- sum_log_magn_u16 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
-
- // Calculate and update pinkNoiseNumerator. Result in Q11.
- tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((WebRtc_UWord32)sum_log_i_log_magn, 12); // Q5
-
- // Shift the largest value of sum_log_i and tmp32no3 before multiplication
- tmp_u16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)sum_log_i, 1); // Q6
- if ((WebRtc_UWord32)sum_log_i > tmpU32no1)
- {
- tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros);
- }
- else
- {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
- }
- tmp_2_w32 -= (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
- matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
- tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
- tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11
- if (tmp_2_w32 < 0)
- {
- tmp_2_w32 = 0;
- }
- inst->pinkNoiseNumerator += tmp_2_w32; // Q11
-
- // Calculate and update pinkNoiseExp. Result in Q14.
- tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i, sum_log_magn_u16); // Q(14-zeros)
- tmp_1_w32 = WEBRTC_SPL_RSHIFT_W32(sum_log_i_log_magn, 3 + zeros);
- tmp_1_w32 = WEBRTC_SPL_MUL((WebRtc_Word32)(inst->magnLen - kStartBand),
- tmp_1_w32);
- tmp_2_w32 -= tmp_1_w32; // Q(14-zeros)
- if (tmp_2_w32 > 0)
- {
- // If the exponential parameter is negative force it to zero, which means a
- // flat spectrum.
- tmp_1_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q14
- inst->pinkNoiseExp += WEBRTC_SPL_SAT(16384, tmp_1_w32, 0); // Q14
- }
- }
-}
-
-void WebRtcNsx_DataSynthesis(NsxInst_t *inst, short *outFrame)
-{
- WebRtc_Word32 tmp32no1;
- WebRtc_Word32 energyOut;
-
- WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
- WebRtc_Word16 tmp16no1, tmp16no2;
- WebRtc_Word16 energyRatio;
- WebRtc_Word16 gainFactor, gainFactor1, gainFactor2;
-
- int i, j;
- int outCIFFT;
- int scaleEnergyOut = 0;
-
- if (inst->zeroInputSignal)
- {
- // synthesize the special case of zero input
- // read out fully processed segment
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- outFrame[i] = inst->synthesisBuffer[i]; // Q0
- }
- // update synthesis buffer
- WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
- inst->synthesisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
- inst->blockLen10ms);
- return;
- }
- // Filter the data in the frequency domain
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
- (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- inst->imag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
- (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- }
- // back to time domain
- // Create spectrum
- realImag[0] = inst->real[0];
- realImag[1] = -inst->imag[0];
- for (i = 1; i < inst->anaLen2; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W16(i, 1);
- tmp16no1 = (inst->anaLen << 1) - j;
- realImag[j] = inst->real[i];
- realImag[j + 1] = -inst->imag[i];
- realImag[tmp16no1] = inst->real[i];
- realImag[tmp16no1 + 1] = inst->imag[i];
- }
- realImag[inst->anaLen] = inst->real[inst->anaLen2];
- realImag[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
-
- // bit-reverse position of elements in array and IFFT it
- WebRtcSpl_ComplexBitReverse(realImag, inst->stages);
- outCIFFT = WebRtcSpl_ComplexIFFT(realImag, inst->stages, 1);
-
- for (i = 0; i < inst->anaLen; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W16(i, 1);
- tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)realImag[j], outCIFFT - inst->normData);
- inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
- WEBRTC_SPL_WORD16_MIN);
- }
-
- //scale factor: only do it after END_STARTUP_LONG time
- gainFactor = 8192; // 8192 = Q13(1.0)
- if (inst->gainMap == 1 &&
- inst->blockIndex > END_STARTUP_LONG &&
- inst->energyIn > 0)
- {
- energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
- if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000))
- {
- energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
- - inst->scaleEnergyIn);
- } else
- {
- inst->energyIn = WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 8 + scaleEnergyOut
- - inst->scaleEnergyIn); // Q(-8-scaleEnergyOut)
- }
-
- assert(inst->energyIn > 0);
- energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut
- + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
-
- // // original FLOAT code
- // if (gain > blim) {
- // factor1=1.0+1.3*(gain-blim);
- // if (gain*factor1 > 1.0) { // FLOAT
- // factor1 = 1.0/gain; // FLOAT
- // }
- // }
- // else {
- // factor1=1.0; // FLOAT
- // }
- //
- // if (gain > blim) {
- // factor2=1.0; //FLOAT
- // }
- // else {
- // //don't reduce scale too much for pause regions: attenuation here should be controlled by flooring
- // factor2=1.0-0.3*(blim-gain); // FLOAT
- // if (gain <= inst->denoiseBound) {
- // factor2=1.0-0.3*(blim-inst->denoiseBound); // FLOAT
- // }
- // }
-
- // all done in lookup tables now
- gainFactor1 = kFactor1Table[energyRatio]; // Q8
- gainFactor2 = inst->factor2Table[energyRatio]; // Q8
-
- //combine both scales with speech/noise prob: note prior (priorSpeechProb) is not frequency dependent
-
- // factor = inst->priorSpeechProb*factor1 + (1.0-inst->priorSpeechProb)*factor2; // original code
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(16384 - inst->priorNonSpeechProb,
- gainFactor1, 14); // Q13 16384 = Q14(1.0)
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
- gainFactor2, 14); // Q13;
- gainFactor = tmp16no1 + tmp16no2; // Q13
- } // out of flag_gain_map==1
-
- // synthesis
- for (i = 0; i < inst->anaLen; i++)
- {
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(inst->window[i],
- inst->real[i], 14); // Q0, window in Q14
- tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16no1, gainFactor, 13); // Q0
- // Down shift with rounding
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
- WEBRTC_SPL_WORD16_MIN); // Q0
- inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(inst->synthesisBuffer[i], tmp16no2); // Q0
- }
-
- // read out fully processed segment
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- outFrame[i] = inst->synthesisBuffer[i]; // Q0
- }
- // update synthesis buffer
- WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer, inst->synthesisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
- inst->blockLen10ms);
-}
-
-int WebRtcNsx_ProcessCore(NsxInst_t *inst, short *speechFrame, short *speechFrameHB,
- short *outFrame, short *outFrameHB)
-{
- // main routine for noise suppression
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2, tmpU32no3;
- WebRtc_UWord32 satMax, maxNoiseU32;
- WebRtc_UWord32 tmpMagnU32, tmpNoiseU32;
- WebRtc_UWord32 nearMagnEst;
- WebRtc_UWord32 noiseUpdateU32;
- WebRtc_UWord32 noiseU32[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 postLocSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 priorLocSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 prevNearSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 curNearSnr;
- WebRtc_UWord32 priorSnr;
- WebRtc_UWord32 noise_estimate = 0;
- WebRtc_UWord32 noise_estimate_avg = 0;
- WebRtc_UWord32 numerator = 0;
-
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 pink_noise_num_avg = 0;
-
- WebRtc_UWord16 tmpU16no1;
- WebRtc_UWord16 magnU16[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 prevNoiseU16[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 nonSpeechProbFinal[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 gammaNoise, prevGammaNoise;
- WebRtc_UWord16 noiseSupFilterTmp[HALF_ANAL_BLOCKL];
-
- WebRtc_Word16 qMagn, qNoise;
- WebRtc_Word16 avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 int_part = 0;
- WebRtc_Word16 frac_part = 0;
- WebRtc_Word16 pink_noise_exp_avg = 0;
-
- int i;
- int nShifts, postShifts;
- int norm32no1, norm32no2;
- int flag, sign;
- int q_domain_to_use = 0;
-
-#ifdef NS_FILEDEBUG
- fwrite(spframe, sizeof(short), inst->blockLen10ms, inst->infile);
-#endif
-
- // Check that initialization has been done
- if (inst->initFlag != 1)
- {
- return -1;
- }
- // Check for valid pointers based on sampling rate
- if ((inst->fs == 32000) && (speechFrameHB == NULL))
- {
- return -1;
- }
-
- // Store speechFrame and transform to frequency domain
- WebRtcNsx_DataAnalysis(inst, speechFrame, magnU16);
-
- if (inst->zeroInputSignal)
- {
- WebRtcNsx_DataSynthesis(inst, outFrame);
-
- if (inst->fs == 32000)
- {
- // update analysis buffer for H band
- // append new data to buffer FX
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms,
- speechFrameHB, inst->blockLen10ms);
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
- }
- } // end of H band gain computation
- return 0;
- }
-
- // Update block index when we have something to process
- inst->blockIndex++;
- //
-
- // Norm of magn
- qMagn = inst->normData - inst->stages;
-
- // Compute spectral flatness on input spectrum
- WebRtcNsx_ComputeSpectralFlatness(inst, magnU16);
-
- // quantile noise estimate
- WebRtcNsx_NoiseEstimation(inst, magnU16, noiseU32, &qNoise);
-
- //noise estimate from previous frame
- for (i = 0; i < inst->magnLen; i++)
- {
- prevNoiseU16[i] = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
- }
-
- if (inst->blockIndex < END_STARTUP_SHORT)
- {
- // Noise Q-domain to be used later; see description at end of section.
- q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages);
-
- // Calculate frequency independent parts in parametric noise estimate and calculate
- // the estimate for the lower frequency band (same values for all frequency bins)
- if (inst->pinkNoiseExp)
- {
- pink_noise_exp_avg = (WebRtc_Word16)WebRtcSpl_DivW32W16(inst->pinkNoiseExp,
- (WebRtc_Word16)(inst->blockIndex + 1)); // Q14
- pink_noise_num_avg = WebRtcSpl_DivW32W16(inst->pinkNoiseNumerator,
- (WebRtc_Word16)(inst->blockIndex + 1)); // Q11
- WebRtcNsx_CalcParametricNoiseEstimate(inst,
- pink_noise_exp_avg,
- pink_noise_num_avg,
- kStartBand,
- &noise_estimate,
- &noise_estimate_avg);
- }
- else
- {
- // Use white noise estimate if we have poor pink noise parameter estimates
- noise_estimate = inst->whiteNoiseLevel; // Q(minNorm-stages)
- noise_estimate_avg = noise_estimate / (inst->blockIndex + 1); // Q(minNorm-stages)
- }
- for (i = 0; i < inst->magnLen; i++)
- {
- // Estimate the background noise using the pink noise parameters if permitted
- if ((inst->pinkNoiseExp) && (i >= kStartBand))
- {
- // Reset noise_estimate
- noise_estimate = 0;
- noise_estimate_avg = 0;
- // Calculate the parametric noise estimate for current frequency bin
- WebRtcNsx_CalcParametricNoiseEstimate(inst,
- pink_noise_exp_avg,
- pink_noise_num_avg,
- i,
- &noise_estimate,
- &noise_estimate_avg);
- }
- // Calculate parametric Wiener filter
- noiseSupFilterTmp[i] = inst->denoiseBound;
- if (inst->initMagnEst[i])
- {
- // numerator = (initMagnEst - noise_estimate * overdrive)
- // Result in Q(8+minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
- numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8);
- if (numerator > tmpU32no1)
- {
- // Suppression filter coefficient larger than zero, so calculate.
- numerator -= tmpU32no1;
-
- // Determine number of left shifts in numerator for best accuracy after
- // division
- nShifts = WebRtcSpl_NormU32(numerator);
- nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
-
- // Shift numerator to Q(nShifts+8+minNorm-stages)
- numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts);
-
- // Shift denominator to Q(nShifts-6+minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
- tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
- noiseSupFilterTmp[i] = (WebRtc_UWord16)WEBRTC_SPL_SAT(16384, tmpU32no2,
- (WebRtc_UWord32)(inst->denoiseBound)); // Q14
- }
- }
- // Weight quantile noise 'noiseU32' with modeled noise 'noise_estimate_avg'
- // 'noiseU32 is in Q(qNoise) and 'noise_estimate' in Q(minNorm-stages)
- // To guarantee that we do not get wrap around when shifting to the same domain
- // we use the lowest one. Furthermore, we need to save 6 bits for the weighting.
- // 'noise_estimate_avg' can handle this operation by construction, but 'noiseU32'
- // may not.
-
- // Shift 'noiseU32' to 'q_domain_to_use'
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
- // Shift 'noise_estimate_avg' to 'q_domain_to_use'
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
- - q_domain_to_use);
- // Make a simple check to see if we have enough room for weighting 'tmpU32no1'
- // without wrap around
- nShifts = 0;
- if (tmpU32no1 & 0xfc000000) {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
- nShifts = 6;
- }
- // Add them together and divide by startup length
- noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
- // Shift back if necessary
- noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
- }
- // Update new Q-domain for 'noiseU32'
- qNoise = q_domain_to_use;
- }
- // compute average signal during END_STARTUP_LONG time:
- // used to normalize spectral difference measure
- if (inst->blockIndex < END_STARTUP_LONG)
- {
- // substituting division with shift ending up in Q(-2*stages)
- inst->timeAvgMagnEnergyTmp
- += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
- 2 * inst->normData + inst->stages - 1);
- inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
- inst->blockIndex + 1);
- }
-
- //start processing at frames == converged+1
- // STEP 1: compute prior and post SNR based on quantile noise estimates
-
- // compute direct decision (DD) estimate of prior SNR: needed for new method
- satMax = (WebRtc_UWord32)1048575;// Largest possible value without getting overflow despite shifting 12 steps
- postShifts = 6 + qMagn - qNoise;
- nShifts = 5 - inst->prevQMagn + inst->prevQNoise;
- for (i = 0; i < inst->magnLen; i++)
- {
- // FLOAT:
- // post SNR
- // postLocSnr[i] = 0.0;
- // if (magn[i] > noise[i])
- // {
- // postLocSnr[i] = magn[i] / (noise[i] + 0.0001);
- // }
- // // previous post SNR
- // // previous estimate: based on previous frame with gain filter (smooth is previous filter)
- //
- // prevNearSnr[i] = inst->prevMagnU16[i] / (inst->noisePrev[i] + 0.0001) * (inst->smooth[i]);
- //
- // // DD estimate is sum of two terms: current estimate and previous estimate
- // // directed decision update of priorSnr (or we actually store [2*priorSnr+1])
- //
- // priorLocSnr[i] = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * (postLocSnr[i] - 1.0);
-
- // calculate post SNR: output in Q11
- postLocSnr[i] = 2048; // 1.0 in Q11
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], 6); // Q(6+qMagn)
- if (postShifts < 0)
- {
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
- } else
- {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
- }
- if (tmpU32no1 > tmpU32no2)
- {
- // Current magnitude larger than noise
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
- if (tmpU32no2)
- {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
- postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- } else
- {
- postLocSnr[i] = satMax;
- }
- }
-
- // calculate prevNearSnr[i] and save for later instead of recalculating it later
- nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
-
- if (tmpU32no2)
- {
- tmpU32no1 = WEBRTC_SPL_DIV(tmpU32no1, tmpU32no2); // Q11
- tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- } else
- {
- tmpU32no1 = satMax; // Q11
- }
- prevNearSnr[i] = tmpU32no1; // Q11
-
- //directed decision update of priorSnr
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
- priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
- // priorLocSnr = 1 + 2*priorSnr
- priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
- } // end of loop over frequencies
- // done with step 1: DD computation of prior and post SNR
-
- // STEP 2: compute speech/noise likelihood
-
- //compute difference of input spectrum with learned/estimated noise spectrum
- WebRtcNsx_ComputeSpectralDifference(inst, magnU16);
- //compute histograms for determination of parameters (thresholds and weights for features)
- //parameters are extracted once every window time (=inst->modelUpdate)
- //counter update
- inst->cntThresUpdate++;
- flag = (int)(inst->cntThresUpdate == inst->modelUpdate);
- //update histogram
- WebRtcNsx_FeatureParameterExtraction(inst, flag);
- //compute model parameters
- if (flag)
- {
- inst->cntThresUpdate = 0; // Reset counter
- //update every window:
- // get normalization for spectral difference for next window estimate
-
- // Shift to Q(-2*stages)
- inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
-
- tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
- // Update featureSpecDiff
- if ((tmpU32no1 != inst->timeAvgMagnEnergy) && (inst->featureSpecDiff))
- {
- norm32no1 = 0;
- tmpU32no3 = tmpU32no1;
- while (0xFFFF0000 & tmpU32no3)
- {
- tmpU32no3 >>= 1;
- norm32no1++;
- }
- tmpU32no2 = inst->featureSpecDiff;
- while (0xFFFF0000 & tmpU32no2)
- {
- tmpU32no2 >>= 1;
- norm32no1++;
- }
- tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
- tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
- if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1)
- {
- inst->featureSpecDiff = 0x007FFFFF;
- } else
- {
- inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
- WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
- }
- }
-
- inst->timeAvgMagnEnergy = tmpU32no1; // Q(-2*stages)
- inst->curAvgMagnEnergy = 0;
- }
-
- //compute speech/noise probability
- WebRtcNsx_SpeechNoiseProb(inst, nonSpeechProbFinal, priorLocSnr, postLocSnr);
-
- //time-avg parameter for noise update
- gammaNoise = NOISE_UPDATE_Q8; // Q8
-
- maxNoiseU32 = 0;
- postShifts = inst->prevQNoise - qMagn;
- nShifts = inst->prevQMagn - qMagn;
- for (i = 0; i < inst->magnLen; i++)
- {
- // temporary noise update: use it for speech frames if update value is less than previous
- // the formula has been rewritten into:
- // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
- if (postShifts < 0)
- {
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
- } else
- {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
- }
- if (prevNoiseU16[i] > tmpU32no2)
- {
- sign = -1;
- tmpU32no1 = prevNoiseU16[i] - tmpU32no2;
- } else
- {
- sign = 1;
- tmpU32no1 = tmpU32no2 - prevNoiseU16[i];
- }
- noiseUpdateU32 = inst->prevNoiseU32[i]; // Q(prevQNoise+11)
- tmpU32no3 = 0;
- if ((tmpU32no1) && (nonSpeechProbFinal[i]))
- {
- // This value will be used later, if gammaNoise changes
- tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
- if (0x7c000000 & tmpU32no3)
- {
- // Shifting required before multiplication
- tmpU32no2
- = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
- } else
- {
- // We can do shifting after multiplication
- tmpU32no2
- = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
- }
- if (sign > 0)
- {
- noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
- } else
- {
- // This operation is safe. We can never get wrap around, since worst
- // case scenario means magnU16 = 0
- noiseUpdateU32 -= tmpU32no2; // Q(prevQNoise+11)
- }
- }
-
- //increase gamma (i.e., less noise update) for frame likely to be speech
- prevGammaNoise = gammaNoise;
- gammaNoise = NOISE_UPDATE_Q8;
- //time-constant based on speech/noise state
- //increase gamma (i.e., less noise update) for frames likely to be speech
- if (nonSpeechProbFinal[i] < ONE_MINUS_PROB_RANGE_Q8)
- {
- gammaNoise = GAMMA_NOISE_TRANS_AND_SPEECH_Q8;
- }
-
- if (prevGammaNoise != gammaNoise)
- {
- // new noise update
- // this line is the same as above, only that the result is stored in a different variable and the gammaNoise
- // has changed
- //
- // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
- if (0x7c000000 & tmpU32no3)
- {
- // Shifting required before multiplication
- tmpU32no2
- = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
- } else
- {
- // We can do shifting after multiplication
- tmpU32no2
- = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
- }
- if (sign > 0)
- {
- tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
- } else
- {
- tmpU32no1 = inst->prevNoiseU32[i] - tmpU32no2; // Q(prevQNoise+11)
- }
- if (noiseUpdateU32 > tmpU32no1)
- {
- noiseUpdateU32 = tmpU32no1; // Q(prevQNoise+11)
- }
- }
- noiseU32[i] = noiseUpdateU32; // Q(prevQNoise+11)
- if (noiseUpdateU32 > maxNoiseU32)
- {
- maxNoiseU32 = noiseUpdateU32;
- }
-
- // conservative noise update
- // // original FLOAT code
- // if (prob_speech < PROB_RANGE) {
- // inst->avgMagnPause[i] = inst->avgMagnPause[i] + (1.0 - gamma_pause)*(magn[i] - inst->avgMagnPause[i]);
- // }
-
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(inst->avgMagnPause[i], -nShifts);
- if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8)
- {
- if (nShifts < 0)
- {
- tmp32no1 = (WebRtc_Word32)magnU16[i] - tmp32no2; // Q(qMagn)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
- } else
- {
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts)
- - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
- }
- tmp32no2 += tmp32no1; // Q(qMagn)
- }
- inst->avgMagnPause[i] = tmp32no2;
- } // end of frequency loop
-
- norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
- qNoise = inst->prevQNoise + norm32no1 - 5;
- // done with step 2: noise update
-
- // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
- nShifts = inst->prevQNoise + 11 - qMagn;
- for (i = 0; i < inst->magnLen; i++)
- {
- // FLOAT code
- // // post and prior SNR
- // curNearSnr = 0.0;
- // if (magn[i] > noise[i])
- // {
- // curNearSnr = magn[i] / (noise[i] + 0.0001) - 1.0;
- // }
- // // DD estimate is sum of two terms: current estimate and previous estimate
- // // directed decision update of snrPrior
- // snrPrior = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * curNearSnr;
- // // gain filter
- // tmpFloat1 = inst->overdrive + snrPrior;
- // tmpFloat2 = snrPrior / tmpFloat1;
- // theFilter[i] = tmpFloat2;
-
- // calculate curNearSnr again, this is necessary because a new noise estimate has been made since then. for the original
- curNearSnr = 0; // Q11
- if (nShifts < 0)
- {
- // This case is equivalent with magn < noise which implies curNearSnr = 0;
- tmpMagnU32 = (WebRtc_UWord32)magnU16[i]; // Q(qMagn)
- tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn)
- } else if (nShifts > 17)
- {
- tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17)
- tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
- } else
- {
- tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], nShifts); // Q(qNoise_prev+11)
- tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
- }
- if (tmpMagnU32 > tmpNoiseU32)
- {
- tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
- norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
- if (tmpU32no2)
- {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
- }
- curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- }
-
- //directed decision update of priorSnr
- // FLOAT
- // priorSnr = DD_PR_SNR * prevNearSnr + (1.0-DD_PR_SNR) * curNearSnr;
-
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(curNearSnr, ONE_MINUS_DD_PR_SNR_Q11); // Q22
- priorSnr = tmpU32no1 + tmpU32no2; // Q22
-
- //gain filter
- tmpU32no1 = (WebRtc_UWord32)(inst->overdrive)
- + WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
- tmpU16no1 = (WebRtc_UWord16)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
- inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
-
- // Weight in the parametric Wiener filter during startup
- if (inst->blockIndex < END_STARTUP_SHORT)
- {
- // Weight the two suppression filters
- tmpU32no1 = WEBRTC_SPL_UMUL_16_16(inst->noiseSupFilter[i],
- (WebRtc_UWord16)inst->blockIndex);
- tmpU32no2 = WEBRTC_SPL_UMUL_16_16(noiseSupFilterTmp[i],
- (WebRtc_UWord16)(END_STARTUP_SHORT
- - inst->blockIndex));
- tmpU32no1 += tmpU32no2;
- inst->noiseSupFilter[i] = (WebRtc_UWord16)WebRtcSpl_DivU32U16(tmpU32no1,
- END_STARTUP_SHORT);
- }
- } // end of loop over frequencies
- //done with step3
-
- // save noise and magnitude spectrum for next frame
- inst->prevQNoise = qNoise;
- inst->prevQMagn = qMagn;
- if (norm32no1 > 5)
- {
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11)
- inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
- }
- } else
- {
- for (i = 0; i < inst->magnLen; i++)
- {
- inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
- inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
- }
- }
-
- WebRtcNsx_DataSynthesis(inst, outFrame);
-#ifdef NS_FILEDEBUG
- fwrite(outframe, sizeof(short), inst->blockLen10ms, inst->outfile);
-#endif
-
- //for H band:
- // only update data buffer, then apply time-domain gain is applied derived from L band
- if (inst->fs == 32000)
- {
- // update analysis buffer for H band
- // append new data to buffer FX
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms, inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms, speechFrameHB, inst->blockLen10ms);
- // range for averaging low band quantities for H band gain
-
- gainTimeDomainHB = 16384; // 16384 = Q14(1.0)
- //average speech prob from low band
- //average filter gain from low band
- //avg over second half (i.e., 4->8kHz) of freq. spectrum
- tmpU32no1 = 0; // Q12
- tmpU16no1 = 0; // Q8
- for (i = inst->anaLen2 - (inst->anaLen2 >> 2); i < inst->anaLen2; i++)
- {
- tmpU16no1 += nonSpeechProbFinal[i]; // Q8
- tmpU32no1 += (WebRtc_UWord32)(inst->noiseSupFilter[i]); // Q14
- }
- avgProbSpeechHB = (WebRtc_Word16)(4096
- - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12
- avgFilterGainHB = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages - 3); // Q14
-
- // // original FLOAT code
- // // gain based on speech probability:
- // avg_prob_speech_tt=(float)2.0*avg_prob_speech-(float)1.0;
- // gain_mod=(float)0.5*((float)1.0+(float)tanh(avg_prob_speech_tt)); // between 0 and 1
-
- // gain based on speech probability:
- // original expression: "0.5 * (1 + tanh(2x-1))"
- // avgProbSpeechHB has been anyway saturated to a value between 0 and 1 so the other cases don't have to be dealt with
- // avgProbSpeechHB and gainModHB are in Q12, 3607 = Q12(0.880615234375) which is a zero point of
- // |0.5 * (1 + tanh(2x-1)) - x| - |0.5 * (1 + tanh(2x-1)) - 0.880615234375| meaning that from that point the error of approximating
- // the expression with f(x) = x would be greater than the error of approximating the expression with f(x) = 0.880615234375
- // error: "|0.5 * (1 + tanh(2x-1)) - x| from x=0 to 0.880615234375" -> http://www.wolframalpha.com/input/?i=|0.5+*+(1+%2B+tanh(2x-1))+-+x|+from+x%3D0+to+0.880615234375
- // and: "|0.5 * (1 + tanh(2x-1)) - 0.880615234375| from x=0.880615234375 to 1" -> http://www.wolframalpha.com/input/?i=+|0.5+*+(1+%2B+tanh(2x-1))+-+0.880615234375|+from+x%3D0.880615234375+to+1
- gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607);
-
- // // original FLOAT code
- // //combine gain with low band gain
- // if (avg_prob_speech < (float)0.5) {
- // gain_time_domain_HB=(float)0.5*gain_mod+(float)0.5*avg_filter_gain;
- // }
- // else {
- // gain_time_domain_HB=(float)0.25*gain_mod+(float)0.75*avg_filter_gain;
- // }
-
-
- //combine gain with low band gain
- if (avgProbSpeechHB < 2048)
- { // 2048 = Q12(0.5)
- // the next two lines in float are "gain_time_domain = 0.5 * gain_mod + 0.5 * avg_filter_gain"; Q2(0.5) = 2 equals one left shift
- gainTimeDomainHB = (gainModHB << 1) + (avgFilterGainHB >> 1); // Q14
- } else
- {
- // "gain_time_domain = 0.25 * gain_mod + 0.75 * agv_filter_gain;"
- gainTimeDomainHB = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, avgFilterGainHB, 2); // 3 = Q2(0.75); Q14
- gainTimeDomainHB += gainModHB; // Q14
- }
- //make sure gain is within flooring range
- gainTimeDomainHB
- = WEBRTC_SPL_SAT(16384, gainTimeDomainHB, (WebRtc_Word16)(inst->denoiseBound)); // 16384 = Q14(1.0)
-
-
- //apply gain
- for (i = 0; i < inst->blockLen10ms; i++)
- {
- outFrameHB[i]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
- }
- } // end of H band gain computation
-
- return 0;
-}
diff --git a/src/modules/audio_processing/ns/main/source/nsx_core.h b/src/modules/audio_processing/ns/main/source/nsx_core.h
deleted file mode 100644
index 2e74303..0000000
--- a/src/modules/audio_processing/ns/main/source/nsx_core.h
+++ /dev/null
@@ -1,169 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "nsx_defines.h"
-
-#ifdef NS_FILEDEBUG
-#include <stdio.h>
-#endif
-
-typedef struct NsxInst_t_
-{
- WebRtc_UWord32 fs;
-
- const WebRtc_Word16* window;
- WebRtc_Word16 analysisBuffer[ANAL_BLOCKL_MAX];
- WebRtc_Word16 synthesisBuffer[ANAL_BLOCKL_MAX];
- WebRtc_UWord16 noiseSupFilter[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 overdrive; /* Q8 */
- WebRtc_UWord16 denoiseBound; /* Q14 */
- const WebRtc_Word16* factor2Table;
- WebRtc_Word16 noiseEstLogQuantile[SIMULT * HALF_ANAL_BLOCKL];
- WebRtc_Word16 noiseEstDensity[SIMULT * HALF_ANAL_BLOCKL];
- WebRtc_Word16 noiseEstCounter[SIMULT];
- WebRtc_Word16 noiseEstQuantile[HALF_ANAL_BLOCKL];
-
- WebRtc_Word16 anaLen;
- int anaLen2;
- int magnLen;
- int aggrMode;
- int stages;
- int initFlag;
- int gainMap;
-
- WebRtc_Word32 maxLrt;
- WebRtc_Word32 minLrt;
- WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
- WebRtc_Word32 featureLogLrt;
- WebRtc_Word32 thresholdLogLrt;
- WebRtc_Word16 weightLogLrt;
-
- WebRtc_UWord32 featureSpecDiff;
- WebRtc_UWord32 thresholdSpecDiff;
- WebRtc_Word16 weightSpecDiff;
-
- WebRtc_UWord32 featureSpecFlat;
- WebRtc_UWord32 thresholdSpecFlat;
- WebRtc_Word16 weightSpecFlat;
-
- WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
- WebRtc_UWord32 magnEnergy;
- WebRtc_UWord32 sumMagn;
- WebRtc_UWord32 curAvgMagnEnergy;
- WebRtc_UWord32 timeAvgMagnEnergy;
- WebRtc_UWord32 timeAvgMagnEnergyTmp;
-
- WebRtc_UWord32 whiteNoiseLevel; //initial noise estimate
- WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
- WebRtc_Word32 pinkNoiseNumerator; //pink noise parameter: numerator
- WebRtc_Word32 pinkNoiseExp; //pink noise parameter: power of freq
- int minNorm; //smallest normalization factor
- int zeroInputSignal; //zero input signal flag
-
- WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
- WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
- WebRtc_Word16 priorNonSpeechProb; //prior speech/noise probability // Q14
-
- int blockIndex; //frame index counter
- int modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
- int cntThresUpdate;
-
- //histograms for parameter estimation
- WebRtc_Word16 histLrt[HIST_PAR_EST];
- WebRtc_Word16 histSpecFlat[HIST_PAR_EST];
- WebRtc_Word16 histSpecDiff[HIST_PAR_EST];
-
- //quantities for high band estimate
- WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
-
- int qNoise;
- int prevQNoise;
- int prevQMagn;
- int blockLen10ms;
-
- WebRtc_Word16 real[ANAL_BLOCKL_MAX];
- WebRtc_Word16 imag[ANAL_BLOCKL_MAX];
- WebRtc_Word32 energyIn;
- int scaleEnergyIn;
- int normData;
-
-} NsxInst_t;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcNsx_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- * - inst : Instance that should be initialized
- * - fs : Sampling frequency
- *
- * Output:
- * - inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t *inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNsx_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- * - NS_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_set_policy_core(NsxInst_t *inst, int mode);
-
-/****************************************************************************
- * WebRtcNsx_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - inFrameLow : Input speech frame for lower band
- * - inFrameHigh : Input speech frame for higher band
- *
- * Output:
- * - inst : Updated instance
- * - outFrameLow : Output speech frame for lower band
- * - outFrameHigh : Output speech frame for higher band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-int WebRtcNsx_ProcessCore(NsxInst_t *inst, short *inFrameLow, short *inFrameHigh,
- short *outFrameLow, short *outFrameHigh);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
diff --git a/src/modules/audio_processing/ns/main/source/windows_private.h b/src/modules/audio_processing/ns/main/source/windows_private.h
deleted file mode 100644
index 8f9006e..0000000
--- a/src/modules/audio_processing/ns/main/source/windows_private.h
+++ /dev/null
@@ -1,573 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-
-// Hanning window for 4ms 16kHz
-static const float kHanning64w128[128] = {
-0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
-0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
-0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
-0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
-0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
-0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
-0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
-0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
-0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
-0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
-0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
-0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
-0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
-0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
-0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
-0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
-0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
-0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
-0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
-0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
-0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
-0.99969881869620f, 1.00000000000000f,
-0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
-0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
-0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
-0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
-0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
-0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
-0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
-0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
-0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
-0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
-0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
-0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
-0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
-0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
-0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
-0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
-0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
-0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
-0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
-0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
-0.07356456359967f, 0.04906767432742f, 0.02454122852291f
-};
-
-
-
-// hybrib Hanning & flat window
-static const float kBlocks80w128[128] = {
-(float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
-(float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
-(float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
-(float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
-(float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
-(float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
-(float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
-(float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
-(float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
-(float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
-(float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
-(float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
-(float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
-(float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
-(float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
-(float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
-(float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
-(float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
-(float)0.09801714, (float)0.06540313, (float)0.03271908
-};
-
-// hybrib Hanning & flat window
-static const float kBlocks160w256[256] = {
-(float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
-(float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
-(float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
-(float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
-(float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
-(float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
-(float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
-(float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
-(float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
-(float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
-(float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
-(float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
-(float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
-(float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
-(float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
-(float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
-(float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
-(float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
-(float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
-(float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
-(float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
-(float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
-(float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
-(float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
-(float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
-(float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
-(float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
-(float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
-(float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
-(float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
-(float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
-(float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
-(float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
-(float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
-(float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
-(float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
-(float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
-(float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
-(float)0.01636173
-};
-
-// hybrib Hanning & flat window: for 20ms
-static const float kBlocks320w512[512] = {
-(float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
-(float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
-(float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
-(float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
-(float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
-(float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
-(float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
-(float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
-(float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
-(float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
-(float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
-(float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
-(float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
-(float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
-(float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
-(float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
-(float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
-(float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
-(float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
-(float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
-(float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
-(float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
-(float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
-(float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
-(float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
-(float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
-(float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
-(float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
-(float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
-(float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
-(float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
-(float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
-(float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
-(float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
-(float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
-(float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
-(float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
-(float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
-(float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
-(float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
-(float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
-(float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
-(float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
-(float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
-(float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
-(float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
-(float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
-(float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
-(float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
-(float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
-(float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
-(float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
-(float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
-(float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
-(float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
-(float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
-(float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
-(float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
-(float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
-(float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
-(float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
-(float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
-(float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
-(float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
-(float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
-(float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
-(float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
-(float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
-(float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
-(float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
-(float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
-(float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
-(float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
-(float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
-(float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
-(float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
-(float)0.01636173, (float)0.00818114
-};
-
-
-// Hanning window: for 15ms at 16kHz with symmetric zeros
-static const float kBlocks240w512[512] = {
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
-(float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
-(float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
-(float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
-(float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
-(float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
-(float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
-(float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
-(float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
-(float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
-(float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
-(float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
-(float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
-(float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
-(float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
-(float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
-(float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
-(float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
-(float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
-(float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
-(float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
-(float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
-(float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
-(float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
-(float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
-(float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
-(float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
-(float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
-(float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
-(float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
-(float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
-(float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
-(float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
-(float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
-(float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
-(float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
-(float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
-(float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
-(float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
-(float)0.95501995, (float)0.95694035, (float)0.95881975, (float)0.96065807, (float)0.96245527,
-(float)0.96421117, (float)0.96592581, (float)0.96759909, (float)0.96923089, (float)0.97082120,
-(float)0.97236991, (float)0.97387701, (float)0.97534233, (float)0.97676587, (float)0.97814763,
-(float)0.97948742, (float)0.98078531, (float)0.98204112, (float)0.98325491, (float)0.98442656,
-(float)0.98555607, (float)0.98664331, (float)0.98768836, (float)0.98869103, (float)0.98965138,
-(float)0.99056935, (float)0.99144489, (float)0.99227792, (float)0.99306846, (float)0.99381649,
-(float)0.99452192, (float)0.99518472, (float)0.99580491, (float)0.99638247, (float)0.99691731,
-(float)0.99740952, (float)0.99785894, (float)0.99826562, (float)0.99862951, (float)0.99895066,
-(float)0.99922901, (float)0.99946457, (float)0.99965733, (float)0.99980724, (float)0.99991435,
-(float)0.99997860, (float)1.00000000, (float)0.99997860, (float)0.99991435, (float)0.99980724,
-(float)0.99965733, (float)0.99946457, (float)0.99922901, (float)0.99895066, (float)0.99862951,
-(float)0.99826562, (float)0.99785894, (float)0.99740946, (float)0.99691731, (float)0.99638247,
-(float)0.99580491, (float)0.99518472, (float)0.99452192, (float)0.99381644, (float)0.99306846,
-(float)0.99227792, (float)0.99144489, (float)0.99056935, (float)0.98965138, (float)0.98869103,
-(float)0.98768836, (float)0.98664331, (float)0.98555607, (float)0.98442656, (float)0.98325491,
-(float)0.98204112, (float)0.98078525, (float)0.97948742, (float)0.97814757, (float)0.97676587,
-(float)0.97534227, (float)0.97387695, (float)0.97236991, (float)0.97082120, (float)0.96923089,
-(float)0.96759909, (float)0.96592581, (float)0.96421117, (float)0.96245521, (float)0.96065807,
-(float)0.95881969, (float)0.95694029, (float)0.95501995, (float)0.95305860, (float)0.95105648,
-(float)0.94901365, (float)0.94693011, (float)0.94480604, (float)0.94264150, (float)0.94043654,
-(float)0.93819129, (float)0.93590593, (float)0.93358046, (float)0.93121493, (float)0.92880952,
-(float)0.92636436, (float)0.92387950, (float)0.92135507, (float)0.91879123, (float)0.91618794,
-(float)0.91354543, (float)0.91086382, (float)0.90814310, (float)0.90538365, (float)0.90258527,
-(float)0.89974827, (float)0.89687276, (float)0.89395875, (float)0.89100647, (float)0.88801610,
-(float)0.88498759, (float)0.88192123, (float)0.87881714, (float)0.87567532, (float)0.87249595,
-(float)0.86927933, (float)0.86602539, (float)0.86273432, (float)0.85940641, (float)0.85604161,
-(float)0.85264009, (float)0.84920216, (float)0.84572780, (float)0.84221715, (float)0.83867055,
-(float)0.83508795, (float)0.83146954, (float)0.82781565, (float)0.82412612, (float)0.82040137,
-(float)0.81664157, (float)0.81284660, (float)0.80901700, (float)0.80515265, (float)0.80125374,
-(float)0.79732066, (float)0.79335332, (float)0.78935200, (float)0.78531694, (float)0.78124815,
-(float)0.77714586, (float)0.77301049, (float)0.76884180, (float)0.76464021, (float)0.76040596,
-(float)0.75613904, (float)0.75183970, (float)0.74750835, (float)0.74314481, (float)0.73874938,
-(float)0.73432249, (float)0.72986400, (float)0.72537428, (float)0.72085363, (float)0.71630186,
-(float)0.71171951, (float)0.70710677, (float)0.70246363, (float)0.69779032, (float)0.69308734,
-(float)0.68835449, (float)0.68359220, (float)0.67880070, (float)0.67398006, (float)0.66913044,
-(float)0.66425240, (float)0.65934575, (float)0.65441096, (float)0.64944804, (float)0.64445722,
-(float)0.63943905, (float)0.63439327, (float)0.62932026, (float)0.62422055, (float)0.61909389,
-(float)0.61394072, (float)0.60876143, (float)0.60355592, (float)0.59832448, (float)0.59306765,
-(float)0.58778518, (float)0.58247757, (float)0.57714522, (float)0.57178789, (float)0.56640613,
-(float)0.56100023, (float)0.55557019, (float)0.55011630, (float)0.54463905, (float)0.53913826,
-(float)0.53361434, (float)0.52806783, (float)0.52249849, (float)0.51690674, (float)0.51129305,
-(float)0.50565726, (float)0.50000006, (float)0.49432117, (float)0.48862115, (float)0.48290038,
-(float)0.47715873, (float)0.47139663, (float)0.46561456, (float)0.45981231, (float)0.45399037,
-(float)0.44814920, (float)0.44228864, (float)0.43640912, (float)0.43051112, (float)0.42459446,
-(float)0.41865960, (float)0.41270703, (float)0.40673658, (float)0.40074870, (float)0.39474386,
-(float)0.38872188, (float)0.38268328, (float)0.37662849, (float)0.37055734, (float)0.36447033,
-(float)0.35836792, (float)0.35224995, (float)0.34611690, (float)0.33996922, (float)0.33380675,
-(float)0.32763001, (float)0.32143945, (float)0.31523487, (float)0.30901679, (float)0.30278572,
-(float)0.29654145, (float)0.29028472, (float)0.28401530, (float)0.27773371, (float)0.27144048,
-(float)0.26513538, (float)0.25881892, (float)0.25249159, (float)0.24615324, (float)0.23980433,
-(float)0.23344538, (float)0.22707619, (float)0.22069728, (float)0.21430916, (float)0.20791161,
-(float)0.20150517, (float)0.19509031, (float)0.18866688, (float)0.18223536, (float)0.17579627,
-(float)0.16934940, (float)0.16289529, (float)0.15643445, (float)0.14996666, (float)0.14349243,
-(float)0.13701232, (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
-(float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220, (float)0.07845904,
-(float)0.07193252, (float)0.06540315, (float)0.05887074, (float)0.05233581, (float)0.04579888,
-(float)0.03925974, (float)0.03271893, (float)0.02617695, (float)0.01963361, (float)0.01308943,
-(float)0.00654493, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000
-};
-
-
-// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
-static const float kBlocks480w1024[1024] = {
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
-(float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
-(float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
-(float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
-(float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
-(float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
-(float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
-(float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
-(float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
-(float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
-(float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
-(float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
-(float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
-(float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
-(float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
-(float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
-(float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
-(float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
-(float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
-(float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
-(float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
-(float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
-(float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
-(float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
-(float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
-(float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
-(float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
-(float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
-(float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
-(float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
-(float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
-(float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
-(float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
-(float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
-(float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
-(float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
-(float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
-(float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
-(float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
-(float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
-(float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
-(float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
-(float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
-(float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
-(float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
-(float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
-(float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
-(float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
-(float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
-(float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
-(float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
-(float)0.73653996, (float)0.73874950, (float)0.74095118, (float)0.74314487, (float)0.74533057,
-(float)0.74750835, (float)0.74967808, (float)0.75183982, (float)0.75399351, (float)0.75613910,
-(float)0.75827658, (float)0.76040596, (float)0.76252723, (float)0.76464027, (float)0.76674515,
-(float)0.76884186, (float)0.77093029, (float)0.77301043, (float)0.77508241, (float)0.77714598,
-(float)0.77920127, (float)0.78124821, (float)0.78328675, (float)0.78531694, (float)0.78733873,
-(float)0.78935206, (float)0.79135692, (float)0.79335338, (float)0.79534125, (float)0.79732066,
-(float)0.79929149, (float)0.80125386, (float)0.80320752, (float)0.80515265, (float)0.80708915,
-(float)0.80901700, (float)0.81093621, (float)0.81284672, (float)0.81474853, (float)0.81664157,
-(float)0.81852591, (float)0.82040149, (float)0.82226825, (float)0.82412618, (float)0.82597536,
-(float)0.82781565, (float)0.82964706, (float)0.83146966, (float)0.83328325, (float)0.83508795,
-(float)0.83688378, (float)0.83867061, (float)0.84044838, (float)0.84221727, (float)0.84397703,
-(float)0.84572780, (float)0.84746957, (float)0.84920216, (float)0.85092574, (float)0.85264021,
-(float)0.85434544, (float)0.85604161, (float)0.85772866, (float)0.85940641, (float)0.86107504,
-(float)0.86273444, (float)0.86438453, (float)0.86602545, (float)0.86765707, (float)0.86927933,
-(float)0.87089235, (float)0.87249607, (float)0.87409031, (float)0.87567532, (float)0.87725097,
-(float)0.87881714, (float)0.88037390, (float)0.88192129, (float)0.88345921, (float)0.88498765,
-(float)0.88650668, (float)0.88801610, (float)0.88951612, (float)0.89100653, (float)0.89248741,
-(float)0.89395881, (float)0.89542055, (float)0.89687276, (float)0.89831537, (float)0.89974827,
-(float)0.90117162, (float)0.90258533, (float)0.90398932, (float)0.90538365, (float)0.90676826,
-(float)0.90814316, (float)0.90950841, (float)0.91086388, (float)0.91220951, (float)0.91354549,
-(float)0.91487163, (float)0.91618794, (float)0.91749454, (float)0.91879123, (float)0.92007810,
-(float)0.92135513, (float)0.92262226, (float)0.92387950, (float)0.92512691, (float)0.92636442,
-(float)0.92759192, (float)0.92880958, (float)0.93001723, (float)0.93121493, (float)0.93240267,
-(float)0.93358046, (float)0.93474817, (float)0.93590593, (float)0.93705362, (float)0.93819135,
-(float)0.93931901, (float)0.94043654, (float)0.94154406, (float)0.94264150, (float)0.94372880,
-(float)0.94480604, (float)0.94587320, (float)0.94693011, (float)0.94797695, (float)0.94901365,
-(float)0.95004016, (float)0.95105654, (float)0.95206273, (float)0.95305866, (float)0.95404440,
-(float)0.95501995, (float)0.95598525, (float)0.95694035, (float)0.95788521, (float)0.95881975,
-(float)0.95974404, (float)0.96065807, (float)0.96156180, (float)0.96245527, (float)0.96333838,
-(float)0.96421117, (float)0.96507370, (float)0.96592581, (float)0.96676767, (float)0.96759909,
-(float)0.96842021, (float)0.96923089, (float)0.97003126, (float)0.97082120, (float)0.97160077,
-(float)0.97236991, (float)0.97312868, (float)0.97387701, (float)0.97461486, (float)0.97534233,
-(float)0.97605932, (float)0.97676587, (float)0.97746199, (float)0.97814763, (float)0.97882277,
-(float)0.97948742, (float)0.98014158, (float)0.98078531, (float)0.98141843, (float)0.98204112,
-(float)0.98265332, (float)0.98325491, (float)0.98384601, (float)0.98442656, (float)0.98499662,
-(float)0.98555607, (float)0.98610497, (float)0.98664331, (float)0.98717111, (float)0.98768836,
-(float)0.98819500, (float)0.98869103, (float)0.98917651, (float)0.98965138, (float)0.99011570,
-(float)0.99056935, (float)0.99101239, (float)0.99144489, (float)0.99186671, (float)0.99227792,
-(float)0.99267852, (float)0.99306846, (float)0.99344778, (float)0.99381649, (float)0.99417448,
-(float)0.99452192, (float)0.99485862, (float)0.99518472, (float)0.99550015, (float)0.99580491,
-(float)0.99609905, (float)0.99638247, (float)0.99665523, (float)0.99691731, (float)0.99716878,
-(float)0.99740952, (float)0.99763954, (float)0.99785894, (float)0.99806762, (float)0.99826562,
-(float)0.99845290, (float)0.99862951, (float)0.99879545, (float)0.99895066, (float)0.99909520,
-(float)0.99922901, (float)0.99935216, (float)0.99946457, (float)0.99956632, (float)0.99965733,
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-(float)0.99011564, (float)0.98965138, (float)0.98917651, (float)0.98869103, (float)0.98819494,
-(float)0.98768836, (float)0.98717111, (float)0.98664331, (float)0.98610497, (float)0.98555607,
-(float)0.98499656, (float)0.98442656, (float)0.98384601, (float)0.98325491, (float)0.98265326,
-(float)0.98204112, (float)0.98141843, (float)0.98078525, (float)0.98014158, (float)0.97948742,
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-(float)0.95404440, (float)0.95305860, (float)0.95206267, (float)0.95105648, (float)0.95004016,
-(float)0.94901365, (float)0.94797695, (float)0.94693011, (float)0.94587314, (float)0.94480604,
-(float)0.94372880, (float)0.94264150, (float)0.94154406, (float)0.94043654, (float)0.93931895,
-(float)0.93819129, (float)0.93705362, (float)0.93590593, (float)0.93474817, (float)0.93358046,
-(float)0.93240267, (float)0.93121493, (float)0.93001723, (float)0.92880952, (float)0.92759192,
-(float)0.92636436, (float)0.92512691, (float)0.92387950, (float)0.92262226, (float)0.92135507,
-(float)0.92007804, (float)0.91879123, (float)0.91749448, (float)0.91618794, (float)0.91487157,
-(float)0.91354543, (float)0.91220951, (float)0.91086382, (float)0.90950835, (float)0.90814310,
-(float)0.90676820, (float)0.90538365, (float)0.90398932, (float)0.90258527, (float)0.90117157,
-(float)0.89974827, (float)0.89831525, (float)0.89687276, (float)0.89542055, (float)0.89395875,
-(float)0.89248741, (float)0.89100647, (float)0.88951600, (float)0.88801610, (float)0.88650662,
-(float)0.88498759, (float)0.88345915, (float)0.88192123, (float)0.88037384, (float)0.87881714,
-(float)0.87725091, (float)0.87567532, (float)0.87409031, (float)0.87249595, (float)0.87089223,
-(float)0.86927933, (float)0.86765701, (float)0.86602539, (float)0.86438447, (float)0.86273432,
-(float)0.86107504, (float)0.85940641, (float)0.85772860, (float)0.85604161, (float)0.85434544,
-(float)0.85264009, (float)0.85092574, (float)0.84920216, (float)0.84746951, (float)0.84572780,
-(float)0.84397697, (float)0.84221715, (float)0.84044844, (float)0.83867055, (float)0.83688372,
-(float)0.83508795, (float)0.83328319, (float)0.83146954, (float)0.82964706, (float)0.82781565,
-(float)0.82597530, (float)0.82412612, (float)0.82226813, (float)0.82040137, (float)0.81852591,
-(float)0.81664157, (float)0.81474847, (float)0.81284660, (float)0.81093609, (float)0.80901700,
-(float)0.80708915, (float)0.80515265, (float)0.80320752, (float)0.80125374, (float)0.79929143,
-(float)0.79732066, (float)0.79534125, (float)0.79335332, (float)0.79135686, (float)0.78935200,
-(float)0.78733861, (float)0.78531694, (float)0.78328675, (float)0.78124815, (float)0.77920121,
-(float)0.77714586, (float)0.77508223, (float)0.77301049, (float)0.77093029, (float)0.76884180,
-(float)0.76674509, (float)0.76464021, (float)0.76252711, (float)0.76040596, (float)0.75827658,
-(float)0.75613904, (float)0.75399339, (float)0.75183970, (float)0.74967796, (float)0.74750835,
-(float)0.74533057, (float)0.74314481, (float)0.74095106, (float)0.73874938, (float)0.73653996,
-(float)0.73432249, (float)0.73209721, (float)0.72986400, (float)0.72762305, (float)0.72537428,
-(float)0.72311789, (float)0.72085363, (float)0.71858162, (float)0.71630186, (float)0.71401453,
-(float)0.71171951, (float)0.70941705, (float)0.70710677, (float)0.70478898, (float)0.70246363,
-(float)0.70013070, (float)0.69779032, (float)0.69544268, (float)0.69308734, (float)0.69072461,
-(float)0.68835449, (float)0.68597704, (float)0.68359220, (float)0.68120021, (float)0.67880070,
-(float)0.67639399, (float)0.67398006, (float)0.67155886, (float)0.66913044, (float)0.66669512,
-(float)0.66425240, (float)0.66180259, (float)0.65934575, (float)0.65688181, (float)0.65441096,
-(float)0.65193301, (float)0.64944804, (float)0.64695609, (float)0.64445722, (float)0.64195150,
-(float)0.63943905, (float)0.63691956, (float)0.63439327, (float)0.63186014, (float)0.62932026,
-(float)0.62677372, (float)0.62422055, (float)0.62166059, (float)0.61909389, (float)0.61652064,
-(float)0.61394072, (float)0.61135429, (float)0.60876143, (float)0.60616189, (float)0.60355592,
-(float)0.60094339, (float)0.59832448, (float)0.59569913, (float)0.59306765, (float)0.59042960,
-(float)0.58778518, (float)0.58513451, (float)0.58247757, (float)0.57981461, (float)0.57714522,
-(float)0.57446963, (float)0.57178789, (float)0.56910002, (float)0.56640613, (float)0.56370628,
-(float)0.56100023, (float)0.55828822, (float)0.55557019, (float)0.55284619, (float)0.55011630,
-(float)0.54738069, (float)0.54463905, (float)0.54189152, (float)0.53913826, (float)0.53637916,
-(float)0.53361434, (float)0.53084403, (float)0.52806783, (float)0.52528596, (float)0.52249849,
-(float)0.51970541, (float)0.51690674, (float)0.51410276, (float)0.51129305, (float)0.50847787,
-(float)0.50565726, (float)0.50283122, (float)0.50000006, (float)0.49716327, (float)0.49432117,
-(float)0.49147379, (float)0.48862115, (float)0.48576325, (float)0.48290038, (float)0.48003212,
-(float)0.47715873, (float)0.47428021, (float)0.47139663, (float)0.46850798, (float)0.46561456,
-(float)0.46271589, (float)0.45981231, (float)0.45690379, (float)0.45399037, (float)0.45107210,
-(float)0.44814920, (float)0.44522130, (float)0.44228864, (float)0.43935123, (float)0.43640912,
-(float)0.43346232, (float)0.43051112, (float)0.42755505, (float)0.42459446, (float)0.42162928,
-(float)0.41865960, (float)0.41568545, (float)0.41270703, (float)0.40972400, (float)0.40673658,
-(float)0.40374479, (float)0.40074870, (float)0.39774850, (float)0.39474386, (float)0.39173496,
-(float)0.38872188, (float)0.38570464, (float)0.38268328, (float)0.37965804, (float)0.37662849,
-(float)0.37359491, (float)0.37055734, (float)0.36751580, (float)0.36447033, (float)0.36142117,
-(float)0.35836792, (float)0.35531086, (float)0.35224995, (float)0.34918529, (float)0.34611690,
-(float)0.34304500, (float)0.33996922, (float)0.33688980, (float)0.33380675, (float)0.33072016,
-(float)0.32763001, (float)0.32453656, (float)0.32143945, (float)0.31833887, (float)0.31523487,
-(float)0.31212750, (float)0.30901679, (float)0.30590302, (float)0.30278572, (float)0.29966521,
-(float)0.29654145, (float)0.29341453, (float)0.29028472, (float)0.28715155, (float)0.28401530,
-(float)0.28087601, (float)0.27773371, (float)0.27458847, (float)0.27144048, (float)0.26828936,
-(float)0.26513538, (float)0.26197854, (float)0.25881892, (float)0.25565651, (float)0.25249159,
-(float)0.24932374, (float)0.24615324, (float)0.24298008, (float)0.23980433, (float)0.23662600,
-(float)0.23344538, (float)0.23026201, (float)0.22707619, (float)0.22388794, (float)0.22069728,
-(float)0.21750426, (float)0.21430916, (float)0.21111152, (float)0.20791161, (float)0.20470949,
-(float)0.20150517, (float)0.19829892, (float)0.19509031, (float)0.19187963, (float)0.18866688,
-(float)0.18545210, (float)0.18223536, (float)0.17901689, (float)0.17579627, (float)0.17257376,
-(float)0.16934940, (float)0.16612324, (float)0.16289529, (float)0.15966584, (float)0.15643445,
-(float)0.15320137, (float)0.14996666, (float)0.14673033, (float)0.14349243, (float)0.14025325,
-(float)0.13701232, (float)0.13376991, (float)0.13052608, (float)0.12728085, (float)0.12403426,
-(float)0.12078657, (float)0.11753736, (float)0.11428688, (float)0.11103519, (float)0.10778230,
-(float)0.10452849, (float)0.10127334, (float)0.09801710, (float)0.09475980, (float)0.09150149,
-(float)0.08824220, (float)0.08498220, (float)0.08172106, (float)0.07845904, (float)0.07519618,
-(float)0.07193252, (float)0.06866808, (float)0.06540315, (float)0.06213728, (float)0.05887074,
-(float)0.05560357, (float)0.05233581, (float)0.04906749, (float)0.04579888, (float)0.04252954,
-(float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798, (float)0.02617695,
-(float)0.02290541, (float)0.01963361, (float)0.01636161, (float)0.01308943, (float)0.00981712,
-(float)0.00654493, (float)0.00327244, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000};
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
diff --git a/src/modules/audio_processing/ns/noise_suppression.c b/src/modules/audio_processing/ns/noise_suppression.c
new file mode 100644
index 0000000..d33caa9
--- /dev/null
+++ b/src/modules/audio_processing/ns/noise_suppression.c
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "noise_suppression.h"
+#include "ns_core.h"
+#include "defines.h"
+
+int WebRtcNs_get_version(char* versionStr, short length) {
+ const char version[] = "NS 2.2.0";
+ const short versionLen = (short)strlen(version) + 1; // +1: null-termination
+
+ if (versionStr == NULL) {
+ return -1;
+ }
+
+ if (versionLen > length) {
+ return -1;
+ }
+
+ strncpy(versionStr, version, versionLen);
+
+ return 0;
+}
+
+int WebRtcNs_Create(NsHandle** NS_inst) {
+ *NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
+ if (*NS_inst != NULL) {
+ (*(NSinst_t**)NS_inst)->initFlag = 0;
+ return 0;
+ } else {
+ return -1;
+ }
+
+}
+
+int WebRtcNs_Free(NsHandle* NS_inst) {
+ free(NS_inst);
+ return 0;
+}
+
+
+int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs) {
+ return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
+}
+
+int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
+ return WebRtcNs_set_policy_core((NSinst_t*) NS_inst, mode);
+}
+
+
+int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H,
+ short* outframe, short* outframe_H) {
+ return WebRtcNs_ProcessCore(
+ (NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
+}
diff --git a/src/modules/audio_processing/ns/noise_suppression_x.c b/src/modules/audio_processing/ns/noise_suppression_x.c
new file mode 100644
index 0000000..afdea7b
--- /dev/null
+++ b/src/modules/audio_processing/ns/noise_suppression_x.c
@@ -0,0 +1,65 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "noise_suppression_x.h"
+#include "nsx_core.h"
+#include "nsx_defines.h"
+
+int WebRtcNsx_get_version(char* versionStr, short length) {
+ const char version[] = "NS\t3.1.0";
+ const short versionLen = (short)strlen(version) + 1; // +1: null-termination
+
+ if (versionStr == NULL) {
+ return -1;
+ }
+
+ if (versionLen > length) {
+ return -1;
+ }
+
+ strncpy(versionStr, version, versionLen);
+
+ return 0;
+}
+
+int WebRtcNsx_Create(NsxHandle** nsxInst) {
+ *nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
+ if (*nsxInst != NULL) {
+ (*(NsxInst_t**)nsxInst)->initFlag = 0;
+ return 0;
+ } else {
+ return -1;
+ }
+
+}
+
+int WebRtcNsx_Free(NsxHandle* nsxInst) {
+ free(nsxInst);
+ return 0;
+}
+
+int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs) {
+ return WebRtcNsx_InitCore((NsxInst_t*)nsxInst, fs);
+}
+
+int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode) {
+ return WebRtcNsx_set_policy_core((NsxInst_t*)nsxInst, mode);
+}
+
+int WebRtcNsx_Process(NsxHandle* nsxInst, short* speechFrame,
+ short* speechFrameHB, short* outFrame,
+ short* outFrameHB) {
+ return WebRtcNsx_ProcessCore(
+ (NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame, outFrameHB);
+}
+
diff --git a/src/modules/audio_processing/ns/main/source/ns.gyp b/src/modules/audio_processing/ns/ns.gypi
similarity index 65%
rename from src/modules/audio_processing/ns/main/source/ns.gyp
rename to src/modules/audio_processing/ns/ns.gypi
index c8488b2..3e3d2e1 100644
--- a/src/modules/audio_processing/ns/main/source/ns.gyp
+++ b/src/modules/audio_processing/ns/ns.gypi
@@ -7,27 +7,24 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../../../common_settings.gypi',
- ],
'targets': [
{
'target_name': 'ns',
'type': '<(library)',
'dependencies': [
- '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
- '../../../utility/util.gyp:apm_util'
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ 'apm_util'
],
'include_dirs': [
- '../interface',
+ 'interface',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'interface',
],
},
'sources': [
- '../interface/noise_suppression.h',
+ 'interface/noise_suppression.h',
'noise_suppression.c',
'windows_private.h',
'defines.h',
@@ -39,18 +36,19 @@
'target_name': 'ns_fix',
'type': '<(library)',
'dependencies': [
- '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
- '../interface',
+ 'interface',
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ 'interface',
],
},
'sources': [
- '../interface/noise_suppression_x.h',
+ 'interface/noise_suppression_x.h',
'noise_suppression_x.c',
'nsx_defines.h',
'nsx_core.c',
@@ -59,9 +57,3 @@
},
],
}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/ns/ns_core.c b/src/modules/audio_processing/ns/ns_core.c
new file mode 100644
index 0000000..e80f699
--- /dev/null
+++ b/src/modules/audio_processing/ns/ns_core.c
@@ -0,0 +1,1305 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+#include <math.h>
+//#include <stdio.h>
+#include <stdlib.h>
+#include "noise_suppression.h"
+#include "ns_core.h"
+#include "windows_private.h"
+#include "fft4g.h"
+#include "signal_processing_library.h"
+
+// Set Feature Extraction Parameters
+void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) {
+ //bin size of histogram
+ inst->featureExtractionParams.binSizeLrt = (float)0.1;
+ inst->featureExtractionParams.binSizeSpecFlat = (float)0.05;
+ inst->featureExtractionParams.binSizeSpecDiff = (float)0.1;
+
+ //range of histogram over which lrt threshold is computed
+ inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0;
+
+ //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
+ // thresholds for prior model
+ inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff
+ inst->featureExtractionParams.factor2ModelPars = (float)0.9; //for spectral_flatness:
+ // used when noise is flatter than speech
+
+ //peak limit for spectral flatness (varies between 0 and 1)
+ inst->featureExtractionParams.thresPosSpecFlat = (float)0.6;
+
+ //limit on spacing of two highest peaks in histogram: spacing determined by bin size
+ inst->featureExtractionParams.limitPeakSpacingSpecFlat =
+ 2 * inst->featureExtractionParams.binSizeSpecFlat;
+ inst->featureExtractionParams.limitPeakSpacingSpecDiff =
+ 2 * inst->featureExtractionParams.binSizeSpecDiff;
+
+ //limit on relevance of second peak:
+ inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5;
+ inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5;
+
+ // fluctuation limit of lrt feature
+ inst->featureExtractionParams.thresFluctLrt = (float)0.05;
+
+ //limit on the max and min values for the feature thresholds
+ inst->featureExtractionParams.maxLrt = (float)1.0;
+ inst->featureExtractionParams.minLrt = (float)0.20;
+
+ inst->featureExtractionParams.maxSpecFlat = (float)0.95;
+ inst->featureExtractionParams.minSpecFlat = (float)0.10;
+
+ inst->featureExtractionParams.maxSpecDiff = (float)1.0;
+ inst->featureExtractionParams.minSpecDiff = (float)0.16;
+
+ //criteria of weight of histogram peak to accept/reject feature
+ inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3
+ * (inst->modelUpdatePars[1])); //for spectral flatness
+ inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3
+ * (inst->modelUpdatePars[1])); //for spectral difference
+}
+
+// Initialize state
+int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs) {
+ int i;
+ //We only support 10ms frames
+
+ //check for valid pointer
+ if (inst == NULL) {
+ return -1;
+ }
+
+ // Initialization of struct
+ if (fs == 8000 || fs == 16000 || fs == 32000) {
+ inst->fs = fs;
+ } else {
+ return -1;
+ }
+ inst->windShift = 0;
+ if (fs == 8000) {
+ // We only support 10ms frames
+ inst->blockLen = 80;
+ inst->blockLen10ms = 80;
+ inst->anaLen = 128;
+ inst->window = kBlocks80w128;
+ inst->outLen = 0;
+ } else if (fs == 16000) {
+ // We only support 10ms frames
+ inst->blockLen = 160;
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->window = kBlocks160w256;
+ inst->outLen = 0;
+ } else if (fs == 32000) {
+ // We only support 10ms frames
+ inst->blockLen = 160;
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->window = kBlocks160w256;
+ inst->outLen = 0;
+ }
+ inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
+
+ // Initialize fft work arrays.
+ inst->ip[0] = 0; // Setting this triggers initialization.
+ memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ WebRtc_rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft);
+
+ memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+
+ //for HB processing
+ memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+
+ //for quantile noise estimation
+ memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
+ inst->lquantile[i] = (float)8.0;
+ inst->density[i] = (float)0.3;
+ }
+
+ for (i = 0; i < SIMULT; i++) {
+ inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
+ }
+
+ inst->updates = 0;
+
+ // Wiener filter initialization
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->smooth[i] = (float)1.0;
+ }
+
+ // Set the aggressiveness: default
+ inst->aggrMode = 0;
+
+ //initialize variables for new method
+ inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->magnPrev[i] = (float)0.0; //previous mag spectrum
+ inst->noisePrev[i] = (float)0.0; //previous noise-spectrum
+ inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold)
+ inst->magnAvgPause[i] = (float)0.0; //conservative noise spectrum estimate
+ inst->speechProbHB[i] = (float)0.0; //for estimation of HB in second pass
+ inst->initMagnEst[i] = (float)0.0; //initial average mag spectrum
+ }
+
+ //feature quantities
+ inst->featureData[0] = SF_FEATURE_THR; //spectral flatness (start on threshold)
+ inst->featureData[1] = (float)0.0; //spectral entropy: not used in this version
+ inst->featureData[2] = (float)0.0; //spectral variance: not used in this version
+ inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold)
+ inst->featureData[4] = SF_FEATURE_THR; //spectral template diff (start on threshold)
+ inst->featureData[5] = (float)0.0; //normalization for spectral-diff
+ inst->featureData[6] = (float)0.0; //window time-average of input magnitude spectrum
+
+ //histogram quantities: used to estimate/update thresholds for features
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ inst->histLrt[i] = 0;
+ inst->histSpecFlat[i] = 0;
+ inst->histSpecDiff[i] = 0;
+ }
+
+ inst->blockInd = -1; //frame counter
+ inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature
+ inst->priorModelPars[1] = (float)0.5; //threshold for spectral flatness:
+ // determined on-line
+ inst->priorModelPars[2] = (float)1.0; //sgn_map par for spectral measure:
+ // 1 for flatness measure
+ inst->priorModelPars[3] = (float)0.5; //threshold for template-difference feature:
+ // determined on-line
+ inst->priorModelPars[4] = (float)1.0; //default weighting parameter for lrt feature
+ inst->priorModelPars[5] = (float)0.0; //default weighting parameter for
+ // spectral flatness feature
+ inst->priorModelPars[6] = (float)0.0; //default weighting parameter for
+ // spectral difference feature
+
+ inst->modelUpdatePars[0] = 2; //update flag for parameters:
+ // 0 no update, 1=update once, 2=update every window
+ inst->modelUpdatePars[1] = 500; //window for update
+ inst->modelUpdatePars[2] = 0; //counter for update of conservative noise spectrum
+ //counter if the feature thresholds are updated during the sequence
+ inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
+
+ inst->signalEnergy = 0.0;
+ inst->sumMagn = 0.0;
+ inst->whiteNoiseLevel = 0.0;
+ inst->pinkNoiseNumerator = 0.0;
+ inst->pinkNoiseExp = 0.0;
+
+ WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration
+
+ //default mode
+ WebRtcNs_set_policy_core(inst, 0);
+
+
+ memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
+
+ inst->initFlag = 1;
+ return 0;
+}
+
+int WebRtcNs_set_policy_core(NSinst_t* inst, int mode) {
+ // allow for modes:0,1,2,3
+ if (mode < 0 || mode > 3) {
+ return (-1);
+ }
+
+ inst->aggrMode = mode;
+ if (mode == 0) {
+ inst->overdrive = (float)1.0;
+ inst->denoiseBound = (float)0.5;
+ inst->gainmap = 0;
+ } else if (mode == 1) {
+ //inst->overdrive = (float)1.25;
+ inst->overdrive = (float)1.0;
+ inst->denoiseBound = (float)0.25;
+ inst->gainmap = 1;
+ } else if (mode == 2) {
+ //inst->overdrive = (float)1.25;
+ inst->overdrive = (float)1.1;
+ inst->denoiseBound = (float)0.125;
+ inst->gainmap = 1;
+ } else if (mode == 3) {
+ //inst->overdrive = (float)1.30;
+ inst->overdrive = (float)1.25;
+ inst->denoiseBound = (float)0.09;
+ inst->gainmap = 1;
+ }
+ return 0;
+}
+
+// Estimate noise
+void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) {
+ int i, s, offset;
+ float lmagn[HALF_ANAL_BLOCKL], delta;
+
+ if (inst->updates < END_STARTUP_LONG) {
+ inst->updates++;
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ lmagn[i] = (float)log(magn[i]);
+ }
+
+ // loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // newquantest(...)
+ for (i = 0; i < inst->magnLen; i++) {
+ // compute delta
+ if (inst->density[offset + i] > 1.0) {
+ delta = FACTOR * (float)1.0 / inst->density[offset + i];
+ } else {
+ delta = FACTOR;
+ }
+
+ // update log quantile estimate
+ if (lmagn[i] > inst->lquantile[offset + i]) {
+ inst->lquantile[offset + i] += QUANTILE * delta
+ / (float)(inst->counter[s] + 1);
+ } else {
+ inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta
+ / (float)(inst->counter[s] + 1);
+ }
+
+ // update density estimate
+ if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH) {
+ inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
+ + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s] + 1);
+ }
+ } // end loop over magnitude spectrum
+
+ if (inst->counter[s] >= END_STARTUP_LONG) {
+ inst->counter[s] = 0;
+ if (inst->updates >= END_STARTUP_LONG) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
+ }
+ }
+ }
+
+ inst->counter[s]++;
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->updates < END_STARTUP_LONG) {
+ // Use the last "s" to get noise during startup that differ from zero.
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
+ }
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = inst->quantile[i];
+ }
+}
+
+// Extract thresholds for feature parameters
+// histograms are computed over some window_size (given by inst->modelUpdatePars[1])
+// thresholds and weights are extracted every window
+// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
+// threshold and weights are returned in: inst->priorModelPars
+void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) {
+ int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
+ int maxPeak1, maxPeak2;
+ int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff;
+
+ float binMid, featureSum;
+ float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
+ float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
+
+ //3 features: lrt, flatness, difference
+ //lrt_feature = inst->featureData[3];
+ //flat_feature = inst->featureData[0];
+ //diff_feature = inst->featureData[4];
+
+ //update histograms
+ if (flag == 0) {
+ // LRT
+ if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt)
+ && (inst->featureData[3] >= 0.0)) {
+ i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt);
+ inst->histLrt[i]++;
+ }
+ // Spectral flatness
+ if ((inst->featureData[0] < HIST_PAR_EST
+ * inst->featureExtractionParams.binSizeSpecFlat)
+ && (inst->featureData[0] >= 0.0)) {
+ i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat);
+ inst->histSpecFlat[i]++;
+ }
+ // Spectral difference
+ if ((inst->featureData[4] < HIST_PAR_EST
+ * inst->featureExtractionParams.binSizeSpecDiff)
+ && (inst->featureData[4] >= 0.0)) {
+ i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff);
+ inst->histSpecDiff[i]++;
+ }
+ }
+
+ // extract parameters for speech/noise probability
+ if (flag == 1) {
+ //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt
+ avgHistLrt = 0.0;
+ avgHistLrtCompl = 0.0;
+ avgSquareHistLrt = 0.0;
+ numHistLrt = 0;
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt;
+ if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt) {
+ avgHistLrt += inst->histLrt[i] * binMid;
+ numHistLrt += inst->histLrt[i];
+ }
+ avgSquareHistLrt += inst->histLrt[i] * binMid * binMid;
+ avgHistLrtCompl += inst->histLrt[i] * binMid;
+ }
+ if (numHistLrt > 0) {
+ avgHistLrt = avgHistLrt / ((float)numHistLrt);
+ }
+ avgHistLrtCompl = avgHistLrtCompl / ((float)inst->modelUpdatePars[1]);
+ avgSquareHistLrt = avgSquareHistLrt / ((float)inst->modelUpdatePars[1]);
+ fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
+ // get threshold for lrt feature:
+ if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
+ //very low fluct, so likely noise
+ inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
+ } else {
+ inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars
+ * avgHistLrt;
+ // check if value is within min/max range
+ if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt) {
+ inst->priorModelPars[0] = inst->featureExtractionParams.minLrt;
+ }
+ if (inst->priorModelPars[0] > inst->featureExtractionParams.maxLrt) {
+ inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
+ }
+ }
+ // done with lrt feature
+
+ //
+ // for spectral flatness and spectral difference: compute the main peaks of histogram
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecFlat = 0.0;
+ posPeak2SpecFlat = 0.0;
+ weightPeak1SpecFlat = 0;
+ weightPeak2SpecFlat = 0;
+
+ // peaks for flatness
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat;
+ if (inst->histSpecFlat[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecFlat = weightPeak1SpecFlat;
+ posPeak2SpecFlat = posPeak1SpecFlat;
+
+ maxPeak1 = inst->histSpecFlat[i];
+ weightPeak1SpecFlat = inst->histSpecFlat[i];
+ posPeak1SpecFlat = binMid;
+ } else if (inst->histSpecFlat[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecFlat[i];
+ weightPeak2SpecFlat = inst->histSpecFlat[i];
+ posPeak2SpecFlat = binMid;
+ }
+ }
+
+ //compute two peaks for spectral difference
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecDiff = 0.0;
+ posPeak2SpecDiff = 0.0;
+ weightPeak1SpecDiff = 0;
+ weightPeak2SpecDiff = 0;
+ // peaks for spectral difference
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff;
+ if (inst->histSpecDiff[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecDiff = weightPeak1SpecDiff;
+ posPeak2SpecDiff = posPeak1SpecDiff;
+
+ maxPeak1 = inst->histSpecDiff[i];
+ weightPeak1SpecDiff = inst->histSpecDiff[i];
+ posPeak1SpecDiff = binMid;
+ } else if (inst->histSpecDiff[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecDiff[i];
+ weightPeak2SpecDiff = inst->histSpecDiff[i];
+ posPeak2SpecDiff = binMid;
+ }
+ }
+
+ // for spectrum flatness feature
+ useFeatureSpecFlat = 1;
+ // merge the two peaks if they are close
+ if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat)
+ < inst->featureExtractionParams.limitPeakSpacingSpecFlat)
+ && (weightPeak2SpecFlat
+ > inst->featureExtractionParams.limitPeakWeightsSpecFlat
+ * weightPeak1SpecFlat)) {
+ weightPeak1SpecFlat += weightPeak2SpecFlat;
+ posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat);
+ }
+ //reject if weight of peaks is not large enough, or peak value too small
+ if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat
+ || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat) {
+ useFeatureSpecFlat = 0;
+ }
+ // if selected, get the threshold
+ if (useFeatureSpecFlat == 1) {
+ // compute the threshold
+ inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars
+ * posPeak1SpecFlat;
+ //check if value is within min/max range
+ if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat) {
+ inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat;
+ }
+ if (inst->priorModelPars[1] > inst->featureExtractionParams.maxSpecFlat) {
+ inst->priorModelPars[1] = inst->featureExtractionParams.maxSpecFlat;
+ }
+ }
+ // done with flatness feature
+
+ // for template feature
+ useFeatureSpecDiff = 1;
+ // merge the two peaks if they are close
+ if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff)
+ < inst->featureExtractionParams.limitPeakSpacingSpecDiff)
+ && (weightPeak2SpecDiff
+ > inst->featureExtractionParams.limitPeakWeightsSpecDiff
+ * weightPeak1SpecDiff)) {
+ weightPeak1SpecDiff += weightPeak2SpecDiff;
+ posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff);
+ }
+ // get the threshold value
+ inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars
+ * posPeak1SpecDiff;
+ //reject if weight of peaks is not large enough
+ if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff) {
+ useFeatureSpecDiff = 0;
+ }
+ //check if value is within min/max range
+ if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff) {
+ inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff;
+ }
+ if (inst->priorModelPars[3] > inst->featureExtractionParams.maxSpecDiff) {
+ inst->priorModelPars[3] = inst->featureExtractionParams.maxSpecDiff;
+ }
+ // done with spectral difference feature
+
+ // don't use template feature if fluctuation of lrt feature is very low:
+ // most likely just noise state
+ if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
+ useFeatureSpecDiff = 0;
+ }
+
+ // select the weights between the features
+ // inst->priorModelPars[4] is weight for lrt: always selected
+ // inst->priorModelPars[5] is weight for spectral flatness
+ // inst->priorModelPars[6] is weight for spectral difference
+ featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
+ inst->priorModelPars[4] = (float)1.0 / featureSum;
+ inst->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
+ inst->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
+
+ // set hists to zero for next update
+ if (inst->modelUpdatePars[0] >= 1) {
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ inst->histLrt[i] = 0;
+ inst->histSpecFlat[i] = 0;
+ inst->histSpecDiff[i] = 0;
+ }
+ }
+ } // end of flag == 1
+}
+
+// Compute spectral flatness on input spectrum
+// magnIn is the magnitude spectrum
+// spectral flatness is returned in inst->featureData[0]
+void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) {
+ int i;
+ int shiftLP = 1; //option to remove first bin(s) from spectral measures
+ float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
+
+ // comute spectral measures
+ // for flatness
+ avgSpectralFlatnessNum = 0.0;
+ avgSpectralFlatnessDen = inst->sumMagn;
+ for (i = 0; i < shiftLP; i++) {
+ avgSpectralFlatnessDen -= magnIn[i];
+ }
+ // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
+ for (i = shiftLP; i < inst->magnLen; i++) {
+ if (magnIn[i] > 0.0) {
+ avgSpectralFlatnessNum += (float)log(magnIn[i]);
+ } else {
+ inst->featureData[0] -= SPECT_FL_TAVG * inst->featureData[0];
+ return;
+ }
+ }
+ //normalize
+ avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen;
+ avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen;
+
+ //ratio and inverse log: check for case of log(0)
+ spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
+
+ //time-avg update of spectral flatness feature
+ inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]);
+ // done with flatness feature
+}
+
+// Compute the difference measure between input spectrum and a template/learned noise spectrum
+// magnIn is the input spectrum
+// the reference/template spectrum is inst->magnAvgPause[i]
+// returns (normalized) spectral difference in inst->featureData[4]
+void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) {
+ // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
+ int i;
+ float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
+
+ avgPause = 0.0;
+ avgMagn = inst->sumMagn;
+ // compute average quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ //conservative smooth noise spectrum from pause frames
+ avgPause += inst->magnAvgPause[i];
+ }
+ avgPause = avgPause / ((float)inst->magnLen);
+ avgMagn = avgMagn / ((float)inst->magnLen);
+
+ covMagnPause = 0.0;
+ varPause = 0.0;
+ varMagn = 0.0;
+ // compute variance and covariance quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause);
+ varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause);
+ varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
+ }
+ covMagnPause = covMagnPause / ((float)inst->magnLen);
+ varPause = varPause / ((float)inst->magnLen);
+ varMagn = varMagn / ((float)inst->magnLen);
+ // update of average magnitude spectrum
+ inst->featureData[6] += inst->signalEnergy;
+
+ avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001);
+ // normalize and compute time-avg update of difference feature
+ avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001));
+ inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]);
+}
+
+// Compute speech/noise probability
+// speech/noise probability is returned in: probSpeechFinal
+//magn is the input magnitude spectrum
+//noise is the noise spectrum
+//snrLocPrior is the prior snr for each freq.
+//snr loc_post is the post snr for each freq.
+void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snrLocPrior,
+ float* snrLocPost) {
+ int i, sgnMap;
+ float invLrt, gainPrior, indPrior;
+ float logLrtTimeAvgKsum, besselTmp;
+ float indicator0, indicator1, indicator2;
+ float tmpFloat1, tmpFloat2;
+ float weightIndPrior0, weightIndPrior1, weightIndPrior2;
+ float threshPrior0, threshPrior1, threshPrior2;
+ float widthPrior, widthPrior0, widthPrior1, widthPrior2;
+
+ widthPrior0 = WIDTH_PR_MAP;
+ widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region:
+ // lower range, so increase width in tanh map
+ widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure
+
+ //threshold parameters for features
+ threshPrior0 = inst->priorModelPars[0];
+ threshPrior1 = inst->priorModelPars[1];
+ threshPrior2 = inst->priorModelPars[3];
+
+ //sign for flatness feature
+ sgnMap = (int)(inst->priorModelPars[2]);
+
+ //weight parameters for features
+ weightIndPrior0 = inst->priorModelPars[4];
+ weightIndPrior1 = inst->priorModelPars[5];
+ weightIndPrior2 = inst->priorModelPars[6];
+
+ // compute feature based on average LR factor
+ // this is the average over all frequencies of the smooth log lrt
+ logLrtTimeAvgKsum = 0.0;
+ for (i = 0; i < inst->magnLen; i++) {
+ tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i];
+ tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001);
+ besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2;
+ inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1)
+ - inst->logLrtTimeAvg[i]);
+ logLrtTimeAvgKsum += inst->logLrtTimeAvg[i];
+ }
+ logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen);
+ inst->featureData[3] = logLrtTimeAvgKsum;
+ // done with computation of LR factor
+
+ //
+ //compute the indicator functions
+ //
+
+ // average lrt feature
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (logLrtTimeAvgKsum < threshPrior0) {
+ widthPrior = widthPrior1;
+ }
+ // compute indicator function: sigmoid map
+ indicator0 = (float)0.5 * ((float)tanh(widthPrior *
+ (logLrtTimeAvgKsum - threshPrior0)) + (float)1.0);
+
+ //spectral flatness feature
+ tmpFloat1 = inst->featureData[0];
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (sgnMap == 1 && (tmpFloat1 > threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ if (sgnMap == -1 && (tmpFloat1 < threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ // compute indicator function: sigmoid map
+ indicator1 = (float)0.5 * ((float)tanh((float)sgnMap *
+ widthPrior * (threshPrior1 - tmpFloat1)) + (float)1.0);
+
+ //for template spectrum-difference
+ tmpFloat1 = inst->featureData[4];
+ widthPrior = widthPrior0;
+ //use larger width in tanh map for pause regions
+ if (tmpFloat1 < threshPrior2) {
+ widthPrior = widthPrior2;
+ }
+ // compute indicator function: sigmoid map
+ indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2))
+ + (float)1.0);
+
+ //combine the indicator function with the feature weights
+ indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2
+ * indicator2;
+ // done with computing indicator function
+
+ //compute the prior probability
+ inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb);
+ // make sure probabilities are within range: keep floor to 0.01
+ if (inst->priorSpeechProb > 1.0) {
+ inst->priorSpeechProb = (float)1.0;
+ }
+ if (inst->priorSpeechProb < 0.01) {
+ inst->priorSpeechProb = (float)0.01;
+ }
+
+ //final speech probability: combine prior model with LR factor:
+ gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001);
+ for (i = 0; i < inst->magnLen; i++) {
+ invLrt = (float)exp(-inst->logLrtTimeAvg[i]);
+ invLrt = (float)gainPrior * invLrt;
+ probSpeechFinal[i] = (float)1.0 / ((float)1.0 + invLrt);
+ }
+}
+
+int WebRtcNs_ProcessCore(NSinst_t* inst,
+ short* speechFrame,
+ short* speechFrameHB,
+ short* outFrame,
+ short* outFrameHB) {
+ // main routine for noise reduction
+
+ int flagHB = 0;
+ int i;
+ const int kStartBand = 5; // Skip first frequency bins during estimation.
+ int updateParsFlag;
+
+ float energy1, energy2, gain, factor, factor1, factor2;
+ float signalEnergy, sumMagn;
+ float snrPrior, currentEstimateStsa;
+ float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
+ float gammaNoiseTmp, gammaNoiseOld;
+ float noiseUpdateTmp, fTmp, dTmp;
+ float fin[BLOCKL_MAX], fout[BLOCKL_MAX];
+ float winData[ANAL_BLOCKL_MAX];
+ float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
+ float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
+ float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
+ float probSpeechFinal[HALF_ANAL_BLOCKL], previousEstimateStsa[HALF_ANAL_BLOCKL];
+ float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
+ // Variables during startup
+ float sum_log_i = 0.0;
+ float sum_log_i_square = 0.0;
+ float sum_log_magn = 0.0;
+ float sum_log_i_log_magn = 0.0;
+ float parametric_noise = 0.0;
+ float parametric_exp = 0.0;
+ float parametric_num = 0.0;
+
+ // SWB variables
+ int deltaBweHB = 1;
+ int deltaGainHB = 1;
+ float decayBweHB = 1.0;
+ float gainMapParHB = 1.0;
+ float gainTimeDomainHB = 1.0;
+ float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
+
+ // Check that initiation has been done
+ if (inst->initFlag != 1) {
+ return (-1);
+ }
+ // Check for valid pointers based on sampling rate
+ if (inst->fs == 32000) {
+ if (speechFrameHB == NULL) {
+ return -1;
+ }
+ flagHB = 1;
+ // range for averaging low band quantities for H band gain
+ deltaBweHB = (int)inst->magnLen / 4;
+ deltaGainHB = deltaBweHB;
+ }
+ //
+ updateParsFlag = inst->modelUpdatePars[0];
+ //
+
+ //for LB do all processing
+ // convert to float
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fin[i] = (float)speechFrame[i];
+ }
+ // update analysis buffer for L band
+ memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
+ sizeof(float) * (inst->anaLen - inst->blockLen10ms));
+ memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
+ sizeof(float) * inst->blockLen10ms);
+
+ if (flagHB == 1) {
+ // convert to float
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fin[i] = (float)speechFrameHB[i];
+ }
+ // update analysis buffer for H band
+ memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
+ sizeof(float) * (inst->anaLen - inst->blockLen10ms));
+ memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
+ sizeof(float) * inst->blockLen10ms);
+ }
+
+ // check if processing needed
+ if (inst->outLen == 0) {
+ // windowing
+ energy1 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ winData[i] = inst->window[i] * inst->dataBuf[i];
+ energy1 += winData[i] * winData[i];
+ }
+ if (energy1 == 0.0) {
+ // synthesize the special case of zero input
+ // we want to avoid updating statistics in this case:
+ // Updating feature statistics when we have zeros only will cause thresholds to
+ // move towards zero signal situations. This in turn has the effect that once the
+ // signal is "turned on" (non-zero values) everything will be treated as speech
+ // and there is no noise suppression effect. Depending on the duration of the
+ // inactive signal it takes a considerable amount of time for the system to learn
+ // what is noise and what is speech.
+
+ // read out fully processed segment
+ for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
+ fout[i - inst->windShift] = inst->syntBuf[i];
+ }
+ // update synthesis buffer
+ memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
+ sizeof(float) * inst->blockLen);
+
+ // out buffer
+ inst->outLen = inst->blockLen - inst->blockLen10ms;
+ if (inst->blockLen > inst->blockLen10ms) {
+ for (i = 0; i < inst->outLen; i++) {
+ inst->outBuf[i] = fout[i + inst->blockLen10ms];
+ }
+ }
+ // convert to short
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = fout[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrame[i] = (short)dTmp;
+ }
+
+ // for time-domain gain of HB
+ if (flagHB == 1) {
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = inst->dataBufHB[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrameHB[i] = (short)dTmp;
+ }
+ } // end of H band gain computation
+ //
+ return 0;
+ }
+
+ //
+ inst->blockInd++; // Update the block index only when we process a block.
+ // FFT
+ WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
+
+ imag[0] = 0;
+ real[0] = winData[0];
+ magn[0] = (float)(fabs(real[0]) + 1.0f);
+ imag[inst->magnLen - 1] = 0;
+ real[inst->magnLen - 1] = winData[1];
+ magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
+ signalEnergy = (float)(real[0] * real[0]) +
+ (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]);
+ sumMagn = magn[0] + magn[inst->magnLen - 1];
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[0] += magn[0];
+ inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
+ tmpFloat2 = log((float)(inst->magnLen - 1));
+ sum_log_i = tmpFloat2;
+ sum_log_i_square = tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[inst->magnLen - 1]);
+ sum_log_magn = tmpFloat1;
+ sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
+ }
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ real[i] = winData[2 * i];
+ imag[i] = winData[2 * i + 1];
+ // magnitude spectrum
+ fTmp = real[i] * real[i];
+ fTmp += imag[i] * imag[i];
+ signalEnergy += fTmp;
+ magn[i] = ((float)sqrt(fTmp)) + 1.0f;
+ sumMagn += magn[i];
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[i] += magn[i];
+ if (i >= kStartBand) {
+ tmpFloat2 = log((float)i);
+ sum_log_i += tmpFloat2;
+ sum_log_i_square += tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[i]);
+ sum_log_magn += tmpFloat1;
+ sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
+ }
+ }
+ }
+ signalEnergy = signalEnergy / ((float)inst->magnLen);
+ inst->signalEnergy = signalEnergy;
+ inst->sumMagn = sumMagn;
+
+ //compute spectral flatness on input spectrum
+ WebRtcNs_ComputeSpectralFlatness(inst, magn);
+ // quantile noise estimate
+ WebRtcNs_NoiseEstimation(inst, magn, noise);
+ //compute simplified noise model during startup
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ // Estimate White noise
+ inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
+ // Estimate Pink noise parameters
+ tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
+ tmpFloat1 -= (sum_log_i * sum_log_i);
+ tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the estimated spectrum to be positive
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ inst->pinkNoiseNumerator += tmpFloat3;
+ tmpFloat2 = (sum_log_i * sum_log_magn);
+ tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the pink noise power to be in the interval [0, 1];
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ if (tmpFloat3 > 1.0f) {
+ tmpFloat3 = 1.0f;
+ }
+ inst->pinkNoiseExp += tmpFloat3;
+
+ // Calculate frequency independent parts of parametric noise estimate.
+ if (inst->pinkNoiseExp == 0.0f) {
+ // Use white noise estimate
+ parametric_noise = inst->whiteNoiseLevel;
+ } else {
+ // Use pink noise estimate
+ parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
+ parametric_num *= (float)(inst->blockInd + 1);
+ parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
+ parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp);
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Estimate the background noise using the white and pink noise parameters
+ if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand)) {
+ // Use pink noise estimate
+ parametric_noise = parametric_num / pow((float)i, parametric_exp);
+ }
+ theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise);
+ theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
+ // Weight quantile noise with modeled noise
+ noise[i] *= (inst->blockInd);
+ tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd);
+ noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
+ noise[i] /= END_STARTUP_SHORT;
+ }
+ }
+ //compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure
+ if (inst->blockInd < END_STARTUP_LONG) {
+ inst->featureData[5] *= inst->blockInd;
+ inst->featureData[5] += signalEnergy;
+ inst->featureData[5] /= (inst->blockInd + 1);
+ }
+
+#ifdef PROCESS_FLOW_0
+ if (inst->blockInd > END_STARTUP_LONG) {
+ //option: average the quantile noise: for check with AEC2
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (float)0.6 * inst->noisePrev[i] + (float)0.4 * noise[i];
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Wiener with over sub-substraction:
+ theFilter[i] = (magn[i] - inst->overdrive * noise[i]) / (magn[i] + (float)0.0001);
+ }
+ }
+#else
+ //start processing at frames == converged+1
+ //
+ // STEP 1: compute prior and post snr based on quantile noise est
+ //
+
+ // compute DD estimate of prior SNR: needed for new method
+ for (i = 0; i < inst->magnLen; i++) {
+ // post snr
+ snrLocPost[i] = (float)0.0;
+ if (magn[i] > noise[i]) {
+ snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
+ }
+ // previous post snr
+ // previous estimate: based on previous frame with gain filter
+ previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001)
+ * (inst->smooth[i]);
+ // DD estimate is sum of two terms: current estimate and previous estimate
+ // directed decision update of snrPrior
+ snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
+ * snrLocPost[i];
+ // post and prior snr needed for step 2
+ } // end of loop over freqs
+#ifdef PROCESS_FLOW_1
+ for (i = 0; i < inst->magnLen; i++) {
+ // gain filter
+ tmpFloat1 = inst->overdrive + snrLocPrior[i];
+ tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
+ theFilter[i] = (float)tmpFloat2;
+ } // end of loop over freqs
+#endif
+ // done with step 1: dd computation of prior and post snr
+
+ //
+ //STEP 2: compute speech/noise likelihood
+ //
+#ifdef PROCESS_FLOW_2
+ // compute difference of input spectrum with learned/estimated noise spectrum
+ WebRtcNs_ComputeSpectralDifference(inst, magn);
+ // compute histograms for parameter decisions (thresholds and weights for features)
+ // parameters are extracted once every window time (=inst->modelUpdatePars[1])
+ if (updateParsFlag >= 1) {
+ // counter update
+ inst->modelUpdatePars[3]--;
+ // update histogram
+ if (inst->modelUpdatePars[3] > 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 0);
+ }
+ // compute model parameters
+ if (inst->modelUpdatePars[3] == 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 1);
+ inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
+ // if wish to update only once, set flag to zero
+ if (updateParsFlag == 1) {
+ inst->modelUpdatePars[0] = 0;
+ } else {
+ // update every window:
+ // get normalization for spectral difference for next window estimate
+ inst->featureData[6] = inst->featureData[6]
+ / ((float)inst->modelUpdatePars[1]);
+ inst->featureData[5] = (float)0.5 * (inst->featureData[6]
+ + inst->featureData[5]);
+ inst->featureData[6] = (float)0.0;
+ }
+ }
+ }
+ // compute speech/noise probability
+ WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost);
+ // time-avg parameter for noise update
+ gammaNoiseTmp = NOISE_UPDATE;
+ for (i = 0; i < inst->magnLen; i++) {
+ probSpeech = probSpeechFinal[i];
+ probNonSpeech = (float)1.0 - probSpeech;
+ // temporary noise update:
+ // use it for speech frames if update value is less than previous
+ noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
+ * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
+ //
+ // time-constant based on speech/noise state
+ gammaNoiseOld = gammaNoiseTmp;
+ gammaNoiseTmp = NOISE_UPDATE;
+ // increase gamma (i.e., less noise update) for frame likely to be speech
+ if (probSpeech > PROB_RANGE) {
+ gammaNoiseTmp = SPEECH_UPDATE;
+ }
+ // conservative noise update
+ if (probSpeech < PROB_RANGE) {
+ inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
+ }
+ // noise update
+ if (gammaNoiseTmp == gammaNoiseOld) {
+ noise[i] = noiseUpdateTmp;
+ } else {
+ noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
+ * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
+ // allow for noise update downwards:
+ // if noise update decreases the noise, it is safe, so allow it to happen
+ if (noiseUpdateTmp < noise[i]) {
+ noise[i] = noiseUpdateTmp;
+ }
+ }
+ } // end of freq loop
+ // done with step 2: noise update
+
+ //
+ // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
+ //
+ for (i = 0; i < inst->magnLen; i++) {
+ // post and prior snr
+ currentEstimateStsa = (float)0.0;
+ if (magn[i] > noise[i]) {
+ currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
+ }
+ // DD estimate is sume of two terms: current estimate and previous estimate
+ // directed decision update of snrPrior
+ snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
+ * currentEstimateStsa;
+ // gain filter
+ tmpFloat1 = inst->overdrive + snrPrior;
+ tmpFloat2 = (float)snrPrior / tmpFloat1;
+ theFilter[i] = (float)tmpFloat2;
+ } // end of loop over freqs
+ // done with step3
+#endif
+#endif
+
+ for (i = 0; i < inst->magnLen; i++) {
+ // flooring bottom
+ if (theFilter[i] < inst->denoiseBound) {
+ theFilter[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilter[i] > (float)1.0) {
+ theFilter[i] = 1.0;
+ }
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ // flooring bottom
+ if (theFilterTmp[i] < inst->denoiseBound) {
+ theFilterTmp[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilterTmp[i] > (float)1.0) {
+ theFilterTmp[i] = 1.0;
+ }
+ // Weight the two suppression filters
+ theFilter[i] *= (inst->blockInd);
+ theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
+ theFilter[i] += theFilterTmp[i];
+ theFilter[i] /= (END_STARTUP_SHORT);
+ }
+ // smoothing
+#ifdef PROCESS_FLOW_0
+ inst->smooth[i] *= SMOOTH; // value set to 0.7 in define.h file
+ inst->smooth[i] += ((float)1.0 - SMOOTH) * theFilter[i];
+#else
+ inst->smooth[i] = theFilter[i];
+#endif
+ real[i] *= inst->smooth[i];
+ imag[i] *= inst->smooth[i];
+ }
+ // keep track of noise and magn spectrum for next frame
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->noisePrev[i] = noise[i];
+ inst->magnPrev[i] = magn[i];
+ }
+ // back to time domain
+ winData[0] = real[0];
+ winData[1] = real[inst->magnLen - 1];
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ winData[2 * i] = real[i];
+ winData[2 * i + 1] = imag[i];
+ }
+ WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
+
+ for (i = 0; i < inst->anaLen; i++) {
+ real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
+ }
+
+ //scale factor: only do it after END_STARTUP_LONG time
+ factor = (float)1.0;
+ if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) {
+ factor1 = (float)1.0;
+ factor2 = (float)1.0;
+
+ energy2 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ energy2 += (float)real[i] * (float)real[i];
+ }
+ gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
+
+#ifdef PROCESS_FLOW_2
+ // scaling for new version
+ if (gain > B_LIM) {
+ factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
+ if (gain * factor1 > (float)1.0) {
+ factor1 = (float)1.0 / gain;
+ }
+ }
+ if (gain < B_LIM) {
+ //don't reduce scale too much for pause regions:
+ // attenuation here should be controlled by flooring
+ if (gain <= inst->denoiseBound) {
+ gain = inst->denoiseBound;
+ }
+ factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
+ }
+ //combine both scales with speech/noise prob:
+ // note prior (priorSpeechProb) is not frequency dependent
+ factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb)
+ * factor2;
+#else
+ if (gain > B_LIM) {
+ factor = (float)1.0 + (float)1.3 * (gain - B_LIM);
+ } else {
+ factor = (float)1.0 + (float)2.0 * (gain - B_LIM);
+ }
+ if (gain * factor > (float)1.0) {
+ factor = (float)1.0 / gain;
+ }
+#endif
+ } // out of inst->gainmap==1
+
+ // synthesis
+ for (i = 0; i < inst->anaLen; i++) {
+ inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
+ }
+ // read out fully processed segment
+ for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
+ fout[i - inst->windShift] = inst->syntBuf[i];
+ }
+ // update synthesis buffer
+ memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
+ sizeof(float) * inst->blockLen);
+
+ // out buffer
+ inst->outLen = inst->blockLen - inst->blockLen10ms;
+ if (inst->blockLen > inst->blockLen10ms) {
+ for (i = 0; i < inst->outLen; i++) {
+ inst->outBuf[i] = fout[i + inst->blockLen10ms];
+ }
+ }
+ } // end of if out.len==0
+ else {
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ fout[i] = inst->outBuf[i];
+ }
+ memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms,
+ sizeof(float) * (inst->outLen - inst->blockLen10ms));
+ memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0,
+ sizeof(float) * inst->blockLen10ms);
+ inst->outLen -= inst->blockLen10ms;
+ }
+
+ // convert to short
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = fout[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrame[i] = (short)dTmp;
+ }
+
+ // for time-domain gain of HB
+ if (flagHB == 1) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->speechProbHB[i] = probSpeechFinal[i];
+ }
+ if (inst->blockInd > END_STARTUP_LONG) {
+ // average speech prob from low band
+ // avg over second half (i.e., 4->8kHz) of freq. spectrum
+ avgProbSpeechHB = 0.0;
+ for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) {
+ avgProbSpeechHB += inst->speechProbHB[i];
+ }
+ avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
+ // average filter gain from low band
+ // average over second half (i.e., 4->8kHz) of freq. spectrum
+ avgFilterGainHB = 0.0;
+ for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++) {
+ avgFilterGainHB += inst->smooth[i];
+ }
+ avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
+ avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
+ // gain based on speech prob:
+ gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
+ //combine gain with low band gain
+ gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
+ if (avgProbSpeechHB >= (float)0.5) {
+ gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
+ }
+ gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
+ } // end of converged
+ //make sure gain is within flooring range
+ // flooring bottom
+ if (gainTimeDomainHB < inst->denoiseBound) {
+ gainTimeDomainHB = inst->denoiseBound;
+ }
+ // flooring top
+ if (gainTimeDomainHB > (float)1.0) {
+ gainTimeDomainHB = 1.0;
+ }
+ //apply gain
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ dTmp = gainTimeDomainHB * inst->dataBufHB[i];
+ if (dTmp < WEBRTC_SPL_WORD16_MIN) {
+ dTmp = WEBRTC_SPL_WORD16_MIN;
+ } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
+ dTmp = WEBRTC_SPL_WORD16_MAX;
+ }
+ outFrameHB[i] = (short)dTmp;
+ }
+ } // end of H band gain computation
+ //
+
+ return 0;
+}
diff --git a/src/modules/audio_processing/ns/ns_core.h b/src/modules/audio_processing/ns/ns_core.h
new file mode 100644
index 0000000..2f4c34f
--- /dev/null
+++ b/src/modules/audio_processing/ns/ns_core.h
@@ -0,0 +1,179 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
+
+#include "defines.h"
+
+typedef struct NSParaExtract_t_ {
+
+ //bin size of histogram
+ float binSizeLrt;
+ float binSizeSpecFlat;
+ float binSizeSpecDiff;
+ //range of histogram over which lrt threshold is computed
+ float rangeAvgHistLrt;
+ //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
+ //thresholds for prior model
+ float factor1ModelPars; //for lrt and spectral difference
+ float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech
+ //peak limit for spectral flatness (varies between 0 and 1)
+ float thresPosSpecFlat;
+ //limit on spacing of two highest peaks in histogram: spacing determined by bin size
+ float limitPeakSpacingSpecFlat;
+ float limitPeakSpacingSpecDiff;
+ //limit on relevance of second peak:
+ float limitPeakWeightsSpecFlat;
+ float limitPeakWeightsSpecDiff;
+ //limit on fluctuation of lrt feature
+ float thresFluctLrt;
+ //limit on the max and min values for the feature thresholds
+ float maxLrt;
+ float minLrt;
+ float maxSpecFlat;
+ float minSpecFlat;
+ float maxSpecDiff;
+ float minSpecDiff;
+ //criteria of weight of histogram peak to accept/reject feature
+ int thresWeightSpecFlat;
+ int thresWeightSpecDiff;
+
+} NSParaExtract_t;
+
+typedef struct NSinst_t_ {
+
+ WebRtc_UWord32 fs;
+ int blockLen;
+ int blockLen10ms;
+ int windShift;
+ int outLen;
+ int anaLen;
+ int magnLen;
+ int aggrMode;
+ const float* window;
+ float dataBuf[ANAL_BLOCKL_MAX];
+ float syntBuf[ANAL_BLOCKL_MAX];
+ float outBuf[3 * BLOCKL_MAX];
+
+ int initFlag;
+ // parameters for quantile noise estimation
+ float density[SIMULT* HALF_ANAL_BLOCKL];
+ float lquantile[SIMULT* HALF_ANAL_BLOCKL];
+ float quantile[HALF_ANAL_BLOCKL];
+ int counter[SIMULT];
+ int updates;
+ // parameters for Wiener filter
+ float smooth[HALF_ANAL_BLOCKL];
+ float overdrive;
+ float denoiseBound;
+ int gainmap;
+ // fft work arrays.
+ int ip[IP_LENGTH];
+ float wfft[W_LENGTH];
+
+ // parameters for new method: some not needed, will reduce/cleanup later
+ WebRtc_Word32 blockInd; //frame index counter
+ int modelUpdatePars[4]; //parameters for updating or estimating
+ // thresholds/weights for prior model
+ float priorModelPars[7]; //parameters for prior model
+ float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
+ float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame
+ float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing
+ float priorSpeechProb; //prior speech/noise probability
+ float featureData[7]; //data for features
+ float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate
+ float signalEnergy; //energy of magn
+ float sumMagn; //sum of magn
+ float whiteNoiseLevel; //initial noise estimate
+ float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate
+ float pinkNoiseNumerator; //pink noise parameter: numerator
+ float pinkNoiseExp; //pink noise parameter: power of freq
+ NSParaExtract_t featureExtractionParams; //parameters for feature extraction
+ //histograms for parameter estimation
+ int histLrt[HIST_PAR_EST];
+ int histSpecFlat[HIST_PAR_EST];
+ int histSpecDiff[HIST_PAR_EST];
+ //quantities for high band estimate
+ float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT
+ float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB
+
+} NSinst_t;
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcNs_InitCore(...)
+ *
+ * This function initializes a noise suppression instance
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - fs : Sampling frequency
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs);
+
+/****************************************************************************
+ * WebRtcNs_set_policy_core(...)
+ *
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
+ *
+ * Output:
+ * - NS_inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNs_set_policy_core(NSinst_t* inst, int mode);
+
+/****************************************************************************
+ * WebRtcNs_ProcessCore
+ *
+ * Do noise suppression.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - inFrameLow : Input speech frame for lower band
+ * - inFrameHigh : Input speech frame for higher band
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outFrameLow : Output speech frame for lower band
+ * - outFrameHigh : Output speech frame for higher band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+
+
+int WebRtcNs_ProcessCore(NSinst_t* inst,
+ short* inFrameLow,
+ short* inFrameHigh,
+ short* outFrameLow,
+ short* outFrameHigh);
+
+
+#ifdef __cplusplus
+}
+#endif
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
diff --git a/src/modules/audio_processing/ns/nsx_core.c b/src/modules/audio_processing/ns/nsx_core.c
new file mode 100644
index 0000000..51bde0c
--- /dev/null
+++ b/src/modules/audio_processing/ns/nsx_core.c
@@ -0,0 +1,2444 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "noise_suppression_x.h"
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+
+#include "cpu_features_wrapper.h"
+#include "nsx_core.h"
+
+// Skip first frequency bins during estimation. (0 <= value < 64)
+static const int kStartBand = 5;
+
+// Constants to compensate for shifting signal log(2^shifts).
+const WebRtc_Word16 WebRtcNsx_kLogTable[9] = {
+ 0, 177, 355, 532, 710, 887, 1065, 1242, 1420
+};
+
+const WebRtc_Word16 WebRtcNsx_kCounterDiv[201] = {
+ 32767, 16384, 10923, 8192, 6554, 5461, 4681,
+ 4096, 3641, 3277, 2979, 2731, 2521, 2341, 2185, 2048, 1928, 1820, 1725, 1638, 1560,
+ 1489, 1425, 1365, 1311, 1260, 1214, 1170, 1130, 1092, 1057, 1024, 993, 964, 936, 910,
+ 886, 862, 840, 819, 799, 780, 762, 745, 728, 712, 697, 683, 669, 655, 643, 630, 618,
+ 607, 596, 585, 575, 565, 555, 546, 537, 529, 520, 512, 504, 496, 489, 482, 475, 468,
+ 462, 455, 449, 443, 437, 431, 426, 420, 415, 410, 405, 400, 395, 390, 386, 381, 377,
+ 372, 368, 364, 360, 356, 352, 349, 345, 341, 338, 334, 331, 328, 324, 321, 318, 315,
+ 312, 309, 306, 303, 301, 298, 295, 293, 290, 287, 285, 282, 280, 278, 275, 273, 271,
+ 269, 266, 264, 262, 260, 258, 256, 254, 252, 250, 248, 246, 245, 243, 241, 239, 237,
+ 236, 234, 232, 231, 229, 228, 226, 224, 223, 221, 220, 218, 217, 216, 214, 213, 211,
+ 210, 209, 207, 206, 205, 204, 202, 201, 200, 199, 197, 196, 195, 194, 193, 192, 191,
+ 189, 188, 187, 186, 185, 184, 183, 182, 181, 180, 179, 178, 177, 176, 175, 174, 173,
+ 172, 172, 171, 170, 169, 168, 167, 166, 165, 165, 164, 163
+};
+
+const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256] = {
+ 0, 1, 3, 4, 6, 7, 9, 10, 11, 13, 14, 16, 17, 18, 20, 21,
+ 22, 24, 25, 26, 28, 29, 30, 32, 33, 34, 36, 37, 38, 40, 41, 42,
+ 44, 45, 46, 47, 49, 50, 51, 52, 54, 55, 56, 57, 59, 60, 61, 62,
+ 63, 65, 66, 67, 68, 69, 71, 72, 73, 74, 75, 77, 78, 79, 80, 81,
+ 82, 84, 85, 86, 87, 88, 89, 90, 92, 93, 94, 95, 96, 97, 98, 99,
+ 100, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 116, 117,
+ 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133,
+ 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
+ 150, 151, 152, 153, 154, 155, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
+ 165, 166, 167, 168, 169, 169, 170, 171, 172, 173, 174, 175, 176, 177, 178, 178,
+ 179, 180, 181, 182, 183, 184, 185, 185, 186, 187, 188, 189, 190, 191, 192, 192,
+ 193, 194, 195, 196, 197, 198, 198, 199, 200, 201, 202, 203, 203, 204, 205, 206,
+ 207, 208, 208, 209, 210, 211, 212, 212, 213, 214, 215, 216, 216, 217, 218, 219,
+ 220, 220, 221, 222, 223, 224, 224, 225, 226, 227, 228, 228, 229, 230, 231, 231,
+ 232, 233, 234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 244,
+ 244, 245, 246, 247, 247, 248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
+};
+
+static const WebRtc_Word16 kPowTableFrac[1024] = {
+ 0, 1, 1, 2, 3, 3, 4, 5,
+ 6, 6, 7, 8, 8, 9, 10, 10,
+ 11, 12, 13, 13, 14, 15, 15, 16,
+ 17, 17, 18, 19, 20, 20, 21, 22,
+ 22, 23, 24, 25, 25, 26, 27, 27,
+ 28, 29, 30, 30, 31, 32, 32, 33,
+ 34, 35, 35, 36, 37, 37, 38, 39,
+ 40, 40, 41, 42, 42, 43, 44, 45,
+ 45, 46, 47, 48, 48, 49, 50, 50,
+ 51, 52, 53, 53, 54, 55, 56, 56,
+ 57, 58, 58, 59, 60, 61, 61, 62,
+ 63, 64, 64, 65, 66, 67, 67, 68,
+ 69, 69, 70, 71, 72, 72, 73, 74,
+ 75, 75, 76, 77, 78, 78, 79, 80,
+ 81, 81, 82, 83, 84, 84, 85, 86,
+ 87, 87, 88, 89, 90, 90, 91, 92,
+ 93, 93, 94, 95, 96, 96, 97, 98,
+ 99, 100, 100, 101, 102, 103, 103, 104,
+ 105, 106, 106, 107, 108, 109, 109, 110,
+ 111, 112, 113, 113, 114, 115, 116, 116,
+ 117, 118, 119, 119, 120, 121, 122, 123,
+ 123, 124, 125, 126, 126, 127, 128, 129,
+ 130, 130, 131, 132, 133, 133, 134, 135,
+ 136, 137, 137, 138, 139, 140, 141, 141,
+ 142, 143, 144, 144, 145, 146, 147, 148,
+ 148, 149, 150, 151, 152, 152, 153, 154,
+ 155, 156, 156, 157, 158, 159, 160, 160,
+ 161, 162, 163, 164, 164, 165, 166, 167,
+ 168, 168, 169, 170, 171, 172, 173, 173,
+ 174, 175, 176, 177, 177, 178, 179, 180,
+ 181, 181, 182, 183, 184, 185, 186, 186,
+ 187, 188, 189, 190, 190, 191, 192, 193,
+ 194, 195, 195, 196, 197, 198, 199, 200,
+ 200, 201, 202, 203, 204, 205, 205, 206,
+ 207, 208, 209, 210, 210, 211, 212, 213,
+ 214, 215, 215, 216, 217, 218, 219, 220,
+ 220, 221, 222, 223, 224, 225, 225, 226,
+ 227, 228, 229, 230, 231, 231, 232, 233,
+ 234, 235, 236, 237, 237, 238, 239, 240,
+ 241, 242, 243, 243, 244, 245, 246, 247,
+ 248, 249, 249, 250, 251, 252, 253, 254,
+ 255, 255, 256, 257, 258, 259, 260, 261,
+ 262, 262, 263, 264, 265, 266, 267, 268,
+ 268, 269, 270, 271, 272, 273, 274, 275,
+ 276, 276, 277, 278, 279, 280, 281, 282,
+ 283, 283, 284, 285, 286, 287, 288, 289,
+ 290, 291, 291, 292, 293, 294, 295, 296,
+ 297, 298, 299, 299, 300, 301, 302, 303,
+ 304, 305, 306, 307, 308, 308, 309, 310,
+ 311, 312, 313, 314, 315, 316, 317, 318,
+ 318, 319, 320, 321, 322, 323, 324, 325,
+ 326, 327, 328, 328, 329, 330, 331, 332,
+ 333, 334, 335, 336, 337, 338, 339, 339,
+ 340, 341, 342, 343, 344, 345, 346, 347,
+ 348, 349, 350, 351, 352, 352, 353, 354,
+ 355, 356, 357, 358, 359, 360, 361, 362,
+ 363, 364, 365, 366, 367, 367, 368, 369,
+ 370, 371, 372, 373, 374, 375, 376, 377,
+ 378, 379, 380, 381, 382, 383, 384, 385,
+ 385, 386, 387, 388, 389, 390, 391, 392,
+ 393, 394, 395, 396, 397, 398, 399, 400,
+ 401, 402, 403, 404, 405, 406, 407, 408,
+ 409, 410, 410, 411, 412, 413, 414, 415,
+ 416, 417, 418, 419, 420, 421, 422, 423,
+ 424, 425, 426, 427, 428, 429, 430, 431,
+ 432, 433, 434, 435, 436, 437, 438, 439,
+ 440, 441, 442, 443, 444, 445, 446, 447,
+ 448, 449, 450, 451, 452, 453, 454, 455,
+ 456, 457, 458, 459, 460, 461, 462, 463,
+ 464, 465, 466, 467, 468, 469, 470, 471,
+ 472, 473, 474, 475, 476, 477, 478, 479,
+ 480, 481, 482, 483, 484, 485, 486, 487,
+ 488, 489, 490, 491, 492, 493, 494, 495,
+ 496, 498, 499, 500, 501, 502, 503, 504,
+ 505, 506, 507, 508, 509, 510, 511, 512,
+ 513, 514, 515, 516, 517, 518, 519, 520,
+ 521, 522, 523, 525, 526, 527, 528, 529,
+ 530, 531, 532, 533, 534, 535, 536, 537,
+ 538, 539, 540, 541, 542, 544, 545, 546,
+ 547, 548, 549, 550, 551, 552, 553, 554,
+ 555, 556, 557, 558, 560, 561, 562, 563,
+ 564, 565, 566, 567, 568, 569, 570, 571,
+ 572, 574, 575, 576, 577, 578, 579, 580,
+ 581, 582, 583, 584, 585, 587, 588, 589,
+ 590, 591, 592, 593, 594, 595, 596, 597,
+ 599, 600, 601, 602, 603, 604, 605, 606,
+ 607, 608, 610, 611, 612, 613, 614, 615,
+ 616, 617, 618, 620, 621, 622, 623, 624,
+ 625, 626, 627, 628, 630, 631, 632, 633,
+ 634, 635, 636, 637, 639, 640, 641, 642,
+ 643, 644, 645, 646, 648, 649, 650, 651,
+ 652, 653, 654, 656, 657, 658, 659, 660,
+ 661, 662, 664, 665, 666, 667, 668, 669,
+ 670, 672, 673, 674, 675, 676, 677, 678,
+ 680, 681, 682, 683, 684, 685, 687, 688,
+ 689, 690, 691, 692, 693, 695, 696, 697,
+ 698, 699, 700, 702, 703, 704, 705, 706,
+ 708, 709, 710, 711, 712, 713, 715, 716,
+ 717, 718, 719, 720, 722, 723, 724, 725,
+ 726, 728, 729, 730, 731, 732, 733, 735,
+ 736, 737, 738, 739, 741, 742, 743, 744,
+ 745, 747, 748, 749, 750, 751, 753, 754,
+ 755, 756, 757, 759, 760, 761, 762, 763,
+ 765, 766, 767, 768, 770, 771, 772, 773,
+ 774, 776, 777, 778, 779, 780, 782, 783,
+ 784, 785, 787, 788, 789, 790, 792, 793,
+ 794, 795, 796, 798, 799, 800, 801, 803,
+ 804, 805, 806, 808, 809, 810, 811, 813,
+ 814, 815, 816, 818, 819, 820, 821, 823,
+ 824, 825, 826, 828, 829, 830, 831, 833,
+ 834, 835, 836, 838, 839, 840, 841, 843,
+ 844, 845, 846, 848, 849, 850, 851, 853,
+ 854, 855, 857, 858, 859, 860, 862, 863,
+ 864, 866, 867, 868, 869, 871, 872, 873,
+ 874, 876, 877, 878, 880, 881, 882, 883,
+ 885, 886, 887, 889, 890, 891, 893, 894,
+ 895, 896, 898, 899, 900, 902, 903, 904,
+ 906, 907, 908, 909, 911, 912, 913, 915,
+ 916, 917, 919, 920, 921, 923, 924, 925,
+ 927, 928, 929, 931, 932, 933, 935, 936,
+ 937, 938, 940, 941, 942, 944, 945, 946,
+ 948, 949, 950, 952, 953, 955, 956, 957,
+ 959, 960, 961, 963, 964, 965, 967, 968,
+ 969, 971, 972, 973, 975, 976, 977, 979,
+ 980, 981, 983, 984, 986, 987, 988, 990,
+ 991, 992, 994, 995, 996, 998, 999, 1001,
+ 1002, 1003, 1005, 1006, 1007, 1009, 1010, 1012,
+ 1013, 1014, 1016, 1017, 1018, 1020, 1021, 1023
+};
+
+static const WebRtc_Word16 kIndicatorTable[17] = {
+ 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718,
+ 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187
+};
+
+// hybrib Hanning & flat window
+static const WebRtc_Word16 kBlocks80w128x[128] = {
+ 0, 536, 1072, 1606, 2139, 2669, 3196, 3720, 4240, 4756, 5266,
+ 5771, 6270, 6762, 7246, 7723, 8192, 8652, 9102, 9543, 9974, 10394,
+ 10803, 11200, 11585, 11958, 12318, 12665, 12998, 13318, 13623, 13913, 14189,
+ 14449, 14694, 14924, 15137, 15334, 15515, 15679, 15826, 15956, 16069, 16165,
+ 16244, 16305, 16349, 16375, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16375, 16349, 16305, 16244, 16165, 16069, 15956,
+ 15826, 15679, 15515, 15334, 15137, 14924, 14694, 14449, 14189, 13913, 13623,
+ 13318, 12998, 12665, 12318, 11958, 11585, 11200, 10803, 10394, 9974, 9543,
+ 9102, 8652, 8192, 7723, 7246, 6762, 6270, 5771, 5266, 4756, 4240,
+ 3720, 3196, 2669, 2139, 1606, 1072, 536
+};
+
+// hybrib Hanning & flat window
+static const WebRtc_Word16 kBlocks160w256x[256] = {
+ 0, 268, 536, 804, 1072, 1339, 1606, 1872,
+ 2139, 2404, 2669, 2933, 3196, 3459, 3720, 3981,
+ 4240, 4499, 4756, 5012, 5266, 5520, 5771, 6021,
+ 6270, 6517, 6762, 7005, 7246, 7486, 7723, 7959,
+ 8192, 8423, 8652, 8878, 9102, 9324, 9543, 9760,
+ 9974, 10185, 10394, 10600, 10803, 11003, 11200, 11394,
+ 11585, 11773, 11958, 12140, 12318, 12493, 12665, 12833,
+ 12998, 13160, 13318, 13472, 13623, 13770, 13913, 14053,
+ 14189, 14321, 14449, 14574, 14694, 14811, 14924, 15032,
+ 15137, 15237, 15334, 15426, 15515, 15599, 15679, 15754,
+ 15826, 15893, 15956, 16015, 16069, 16119, 16165, 16207,
+ 16244, 16277, 16305, 16329, 16349, 16364, 16375, 16382,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
+ 16384, 16382, 16375, 16364, 16349, 16329, 16305, 16277,
+ 16244, 16207, 16165, 16119, 16069, 16015, 15956, 15893,
+ 15826, 15754, 15679, 15599, 15515, 15426, 15334, 15237,
+ 15137, 15032, 14924, 14811, 14694, 14574, 14449, 14321,
+ 14189, 14053, 13913, 13770, 13623, 13472, 13318, 13160,
+ 12998, 12833, 12665, 12493, 12318, 12140, 11958, 11773,
+ 11585, 11394, 11200, 11003, 10803, 10600, 10394, 10185,
+ 9974, 9760, 9543, 9324, 9102, 8878, 8652, 8423,
+ 8192, 7959, 7723, 7486, 7246, 7005, 6762, 6517,
+ 6270, 6021, 5771, 5520, 5266, 5012, 4756, 4499,
+ 4240, 3981, 3720, 3459, 3196, 2933, 2669, 2404,
+ 2139, 1872, 1606, 1339, 1072, 804, 536, 268
+};
+
+// Gain factor1 table: Input value in Q8 and output value in Q13
+// original floating point code
+// if (gain > blim) {
+// factor1 = 1.0 + 1.3 * (gain - blim);
+// if (gain * factor1 > 1.0) {
+// factor1 = 1.0 / gain;
+// }
+// } else {
+// factor1 = 1.0;
+// }
+static const WebRtc_Word16 kFactor1Table[257] = {
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8233, 8274, 8315, 8355, 8396, 8436, 8475, 8515, 8554, 8592, 8631, 8669,
+ 8707, 8745, 8783, 8820, 8857, 8894, 8931, 8967, 9003, 9039, 9075, 9111, 9146, 9181,
+ 9216, 9251, 9286, 9320, 9354, 9388, 9422, 9456, 9489, 9523, 9556, 9589, 9622, 9655,
+ 9687, 9719, 9752, 9784, 9816, 9848, 9879, 9911, 9942, 9973, 10004, 10035, 10066,
+ 10097, 10128, 10158, 10188, 10218, 10249, 10279, 10308, 10338, 10368, 10397, 10426,
+ 10456, 10485, 10514, 10543, 10572, 10600, 10629, 10657, 10686, 10714, 10742, 10770,
+ 10798, 10826, 10854, 10882, 10847, 10810, 10774, 10737, 10701, 10666, 10631, 10596,
+ 10562, 10527, 10494, 10460, 10427, 10394, 10362, 10329, 10297, 10266, 10235, 10203,
+ 10173, 10142, 10112, 10082, 10052, 10023, 9994, 9965, 9936, 9908, 9879, 9851, 9824,
+ 9796, 9769, 9742, 9715, 9689, 9662, 9636, 9610, 9584, 9559, 9534, 9508, 9484, 9459,
+ 9434, 9410, 9386, 9362, 9338, 9314, 9291, 9268, 9245, 9222, 9199, 9176, 9154, 9132,
+ 9110, 9088, 9066, 9044, 9023, 9002, 8980, 8959, 8939, 8918, 8897, 8877, 8857, 8836,
+ 8816, 8796, 8777, 8757, 8738, 8718, 8699, 8680, 8661, 8642, 8623, 8605, 8586, 8568,
+ 8550, 8532, 8514, 8496, 8478, 8460, 8443, 8425, 8408, 8391, 8373, 8356, 8339, 8323,
+ 8306, 8289, 8273, 8256, 8240, 8224, 8208, 8192
+};
+
+// For Factor2 tables
+// original floating point code
+// if (gain > blim) {
+// factor2 = 1.0;
+// } else {
+// factor2 = 1.0 - 0.3 * (blim - gain);
+// if (gain <= inst->denoiseBound) {
+// factor2 = 1.0 - 0.3 * (blim - inst->denoiseBound);
+// }
+// }
+//
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness1[257] = {
+ 7577, 7577, 7577, 7577, 7577, 7577,
+ 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness2[257] = {
+ 7270, 7270, 7270, 7270, 7270, 7306,
+ 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// Gain factor table: Input value in Q8 and output value in Q13
+static const WebRtc_Word16 kFactor2Aggressiveness3[257] = {
+ 7184, 7184, 7184, 7229, 7270, 7306,
+ 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
+ 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
+ 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
+ 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
+ 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
+ 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
+};
+
+// sum of log2(i) from table index to inst->anaLen2 in Q5
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kSumLogIndex[66] = {
+ 0, 22917, 22917, 22885, 22834, 22770, 22696, 22613,
+ 22524, 22428, 22326, 22220, 22109, 21994, 21876, 21754,
+ 21629, 21501, 21370, 21237, 21101, 20963, 20822, 20679,
+ 20535, 20388, 20239, 20089, 19937, 19783, 19628, 19470,
+ 19312, 19152, 18991, 18828, 18664, 18498, 18331, 18164,
+ 17994, 17824, 17653, 17480, 17306, 17132, 16956, 16779,
+ 16602, 16423, 16243, 16063, 15881, 15699, 15515, 15331,
+ 15146, 14960, 14774, 14586, 14398, 14209, 14019, 13829,
+ 13637, 13445
+};
+
+// sum of log2(i)^2 from table index to inst->anaLen2 in Q2
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kSumSquareLogIndex[66] = {
+ 0, 16959, 16959, 16955, 16945, 16929, 16908, 16881,
+ 16850, 16814, 16773, 16729, 16681, 16630, 16575, 16517,
+ 16456, 16392, 16325, 16256, 16184, 16109, 16032, 15952,
+ 15870, 15786, 15700, 15612, 15521, 15429, 15334, 15238,
+ 15140, 15040, 14938, 14834, 14729, 14622, 14514, 14404,
+ 14292, 14179, 14064, 13947, 13830, 13710, 13590, 13468,
+ 13344, 13220, 13094, 12966, 12837, 12707, 12576, 12444,
+ 12310, 12175, 12039, 11902, 11763, 11624, 11483, 11341,
+ 11198, 11054
+};
+
+// log2(table index) in Q12
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kLogIndex[129] = {
+ 0, 0, 4096, 6492, 8192, 9511, 10588, 11499,
+ 12288, 12984, 13607, 14170, 14684, 15157, 15595, 16003,
+ 16384, 16742, 17080, 17400, 17703, 17991, 18266, 18529,
+ 18780, 19021, 19253, 19476, 19691, 19898, 20099, 20292,
+ 20480, 20662, 20838, 21010, 21176, 21338, 21496, 21649,
+ 21799, 21945, 22087, 22226, 22362, 22495, 22625, 22752,
+ 22876, 22998, 23117, 23234, 23349, 23462, 23572, 23680,
+ 23787, 23892, 23994, 24095, 24195, 24292, 24388, 24483,
+ 24576, 24668, 24758, 24847, 24934, 25021, 25106, 25189,
+ 25272, 25354, 25434, 25513, 25592, 25669, 25745, 25820,
+ 25895, 25968, 26041, 26112, 26183, 26253, 26322, 26390,
+ 26458, 26525, 26591, 26656, 26721, 26784, 26848, 26910,
+ 26972, 27033, 27094, 27154, 27213, 27272, 27330, 27388,
+ 27445, 27502, 27558, 27613, 27668, 27722, 27776, 27830,
+ 27883, 27935, 27988, 28039, 28090, 28141, 28191, 28241,
+ 28291, 28340, 28388, 28437, 28484, 28532, 28579, 28626,
+ 28672
+};
+
+// determinant of estimation matrix in Q0 corresponding to the log2 tables above
+// Note that the first table value is invalid, since log2(0) = -infinity
+static const WebRtc_Word16 kDeterminantEstMatrix[66] = {
+ 0, 29814, 25574, 22640, 20351, 18469, 16873, 15491,
+ 14277, 13199, 12233, 11362, 10571, 9851, 9192, 8587,
+ 8030, 7515, 7038, 6596, 6186, 5804, 5448, 5115,
+ 4805, 4514, 4242, 3988, 3749, 3524, 3314, 3116,
+ 2930, 2755, 2590, 2435, 2289, 2152, 2022, 1900,
+ 1785, 1677, 1575, 1478, 1388, 1302, 1221, 1145,
+ 1073, 1005, 942, 881, 825, 771, 721, 674,
+ 629, 587, 547, 510, 475, 442, 411, 382,
+ 355, 330
+};
+
+// Declare function pointers.
+NoiseEstimation WebRtcNsx_NoiseEstimation;
+PrepareSpectrum WebRtcNsx_PrepareSpectrum;
+SynthesisUpdate WebRtcNsx_SynthesisUpdate;
+AnalysisUpdate WebRtcNsx_AnalysisUpdate;
+Denormalize WebRtcNsx_Denormalize;
+CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
+
+// Update the noise estimation information.
+static void UpdateNoiseEstimate(NsxInst_t* inst, int offset) {
+ WebRtc_Word32 tmp32no1 = 0;
+ WebRtc_Word32 tmp32no2 = 0;
+ WebRtc_Word16 tmp16 = 0;
+ const WebRtc_Word16 kExp2Const = 11819; // Q13
+
+ int i = 0;
+
+ tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
+ inst->magnLen);
+ // Guarantee a Q-domain as high as possible and still fit in int16
+ inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ kExp2Const, tmp16, 21);
+ for (i = 0; i < inst->magnLen; i++) {
+ // inst->quantile[i]=exp(inst->lquantile[offset+i]);
+ // in Q21
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ inst->noiseEstLogQuantile[offset + i]);
+ tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+ tmp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ tmp16 -= 21;// shift 21 to get result in Q0
+ tmp16 += (WebRtc_Word16) inst->qNoise; //shift to get result in Q(qNoise)
+ if (tmp16 < 0) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ }
+ inst->noiseEstQuantile[i] = WebRtcSpl_SatW32ToW16(tmp32no1);
+ }
+}
+
+// Noise Estimation
+static void NoiseEstimationC(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise) {
+ WebRtc_Word16 lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
+ WebRtc_Word16 countProd, delta, zeros, frac;
+ WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
+ const int16_t log2_const = 22713; // Q15
+ const int16_t width_factor = 21845;
+
+ int i, s, offset;
+
+ tabind = inst->stages - inst->normData;
+ assert(tabind < 9);
+ assert(tabind > -9);
+ if (tabind < 0) {
+ logval = -WebRtcNsx_kLogTable[-tabind];
+ } else {
+ logval = WebRtcNsx_kLogTable[tabind];
+ }
+
+ // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
+ // magn is in Q(-stages), and the real lmagn values are:
+ // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
+ // lmagn in Q8
+ for (i = 0; i < inst->magnLen; i++) {
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magn[i] << zeros)
+ & 0x7FFFFFFF) >> 23);
+ // log2(magn(i))
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ // log2(magn(i))*log(2)
+ lmagn[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
+ // + log(2^stages)
+ lmagn[i] += logval;
+ } else {
+ lmagn[i] = logval;//0;
+ }
+ }
+
+ // loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // Get counter values from state
+ counter = inst->noiseEstCounter[s];
+ assert(counter < 201);
+ countDiv = WebRtcNsx_kCounterDiv[counter];
+ countProd = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(counter, countDiv);
+
+ // quant_est(...)
+ for (i = 0; i < inst->magnLen; i++) {
+ // compute delta
+ if (inst->noiseEstDensity[offset + i] > 512) {
+ // Get the value for delta by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
+ delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ } else {
+ delta = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ delta = FACTOR_Q7_STARTUP;
+ }
+ }
+
+ // update log quantile estimate
+ tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
+ // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
+ // CounterDiv=1/(inst->counter[s]+1) in Q15
+ tmp16 += 2;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
+ inst->noiseEstLogQuantile[offset + i] += tmp16no1;
+ } else {
+ tmp16 += 1;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
+ // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
+ inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
+ if (inst->noiseEstLogQuantile[offset + i] < logval) {
+ // This is the smallest fixed point representation we can
+ // have, hence we limit the output.
+ inst->noiseEstLogQuantile[offset + i] = logval;
+ }
+ }
+
+ // update density estimate
+ if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
+ < WIDTH_Q8) {
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->noiseEstDensity[offset + i], countProd, 15);
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ width_factor, countDiv, 15);
+ inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
+ }
+ } // end loop over magnitude spectrum
+
+ if (counter >= END_STARTUP_LONG) {
+ inst->noiseEstCounter[s] = 0;
+ if (inst->blockIndex >= END_STARTUP_LONG) {
+ UpdateNoiseEstimate(inst, offset);
+ }
+ }
+ inst->noiseEstCounter[s]++;
+
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ UpdateNoiseEstimate(inst, offset);
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (WebRtc_UWord32)(inst->noiseEstQuantile[i]); // Q(qNoise)
+ }
+ (*q_noise) = (WebRtc_Word16)inst->qNoise;
+}
+
+// Filter the data in the frequency domain, and create spectrum.
+static void PrepareSpectrumC(NsxInst_t* inst, int16_t* freq_buf) {
+ int i = 0, j = 0;
+ int16_t tmp16 = 0;
+
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
+ (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ inst->imag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
+ (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ }
+
+ freq_buf[0] = inst->real[0];
+ freq_buf[1] = -inst->imag[0];
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ tmp16 = (inst->anaLen << 1) - j;
+ freq_buf[j] = inst->real[i];
+ freq_buf[j + 1] = -inst->imag[i];
+ freq_buf[tmp16] = inst->real[i];
+ freq_buf[tmp16 + 1] = inst->imag[i];
+ }
+ freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+}
+
+// Denormalize the input buffer.
+static __inline void DenormalizeC(NsxInst_t* inst, int16_t* in, int factor) {
+ int i = 0, j = 0;
+ int32_t tmp32 = 0;
+ for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
+ tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)in[j],
+ factor - inst->normData);
+ inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ }
+}
+
+// For the noise supression process, synthesis, read out fully processed
+// segment, and update synthesis buffer.
+static void SynthesisUpdateC(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor) {
+ int i = 0;
+ int16_t tmp16a = 0;
+ int16_t tmp16b = 0;
+ int32_t tmp32 = 0;
+
+ // synthesis
+ for (i = 0; i < inst->anaLen; i++) {
+ tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13); // Q0
+ // Down shift with rounding
+ tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(inst->synthesisBuffer[i],
+ tmp16b); // Q0
+ }
+
+ // read out fully processed segment
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ out_frame[i] = inst->synthesisBuffer[i]; // Q0
+ }
+
+ // update synthesis buffer
+ WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ inst->synthesisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
+ + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
+}
+
+// Update analysis buffer for lower band, and window data before FFT.
+static void AnalysisUpdateC(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech) {
+ int i = 0;
+
+ // For lower band update analysis buffer.
+ WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
+ inst->analysisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
+ + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+
+ // Window data before FFT.
+ for (i = 0; i < inst->anaLen; i++) {
+ out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ }
+}
+
+// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
+// zeros, and normalize it.
+static __inline void CreateComplexBufferC(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out) {
+ int i = 0, j = 0;
+ for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
+ out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ out[j + 1] = 0; // Insert zeros in imaginary part
+ }
+}
+
+void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
+ WebRtc_Word16 pink_noise_exp_avg,
+ WebRtc_Word32 pink_noise_num_avg,
+ int freq_index,
+ WebRtc_UWord32* noise_estimate,
+ WebRtc_UWord32* noise_estimate_avg) {
+ WebRtc_Word32 tmp32no1 = 0;
+ WebRtc_Word32 tmp32no2 = 0;
+
+ WebRtc_Word16 int_part = 0;
+ WebRtc_Word16 frac_part = 0;
+
+ // Use pink noise estimate
+ // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
+ assert(freq_index >= 0);
+ assert(freq_index < 129);
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(pink_noise_exp_avg, kLogIndex[freq_index]); // Q26
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 15); // Q11
+ tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
+
+ // Calculate output: 2^tmp32no1
+ // Output in Q(minNorm-stages)
+ tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11);
+ if (tmp32no1 > 0) {
+ int_part = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
+ frac_part = (WebRtc_Word16)(tmp32no1 & 0x000007ff); // Q11
+ // Piecewise linear approximation of 'b' in
+ // 2^(int_part+frac_part) = 2^int_part * (1 + b)
+ // 'b' is given in Q11 and below stored in frac_part.
+ if (WEBRTC_SPL_RSHIFT_W16(frac_part, 10)) {
+ // Upper fractional part
+ tmp32no2 = WEBRTC_SPL_MUL_16_16(2048 - frac_part, 1244); // Q21
+ tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
+ } else {
+ // Lower fractional part
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(frac_part, 804), 10);
+ }
+ // Shift fractional part to Q(minNorm-stages)
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
+ *noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (WebRtc_UWord32)tmp32no2;
+ // Scale up to initMagnEst, which is not block averaged
+ *noise_estimate = (*noise_estimate_avg) * (WebRtc_UWord32)(inst->blockIndex + 1);
+ }
+}
+
+// Initialize state
+WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs) {
+ int i;
+
+ //check for valid pointer
+ if (inst == NULL) {
+ return -1;
+ }
+ //
+
+ // Initialization of struct
+ if (fs == 8000 || fs == 16000 || fs == 32000) {
+ inst->fs = fs;
+ } else {
+ return -1;
+ }
+
+ if (fs == 8000) {
+ inst->blockLen10ms = 80;
+ inst->anaLen = 128;
+ inst->stages = 7;
+ inst->window = kBlocks80w128x;
+ inst->thresholdLogLrt = 131072; //default threshold for LRT feature
+ inst->maxLrt = 0x0040000;
+ inst->minLrt = 52429;
+ } else if (fs == 16000) {
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->stages = 8;
+ inst->window = kBlocks160w256x;
+ inst->thresholdLogLrt = 212644; //default threshold for LRT feature
+ inst->maxLrt = 0x0080000;
+ inst->minLrt = 104858;
+ } else if (fs == 32000) {
+ inst->blockLen10ms = 160;
+ inst->anaLen = 256;
+ inst->stages = 8;
+ inst->window = kBlocks160w256x;
+ inst->thresholdLogLrt = 212644; //default threshold for LRT feature
+ inst->maxLrt = 0x0080000;
+ inst->minLrt = 104858;
+ }
+ inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
+ inst->magnLen = inst->anaLen2 + 1;
+
+ WebRtcSpl_ZerosArrayW16(inst->analysisBuffer, ANAL_BLOCKL_MAX);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer, ANAL_BLOCKL_MAX);
+
+ // for HB processing
+ WebRtcSpl_ZerosArrayW16(inst->dataBufHBFX, ANAL_BLOCKL_MAX);
+ // for quantile noise estimation
+ WebRtcSpl_ZerosArrayW16(inst->noiseEstQuantile, HALF_ANAL_BLOCKL);
+ for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
+ inst->noiseEstLogQuantile[i] = 2048; // Q8
+ inst->noiseEstDensity[i] = 153; // Q9
+ }
+ for (i = 0; i < SIMULT; i++) {
+ inst->noiseEstCounter[i] = (WebRtc_Word16)(END_STARTUP_LONG * (i + 1)) / SIMULT;
+ }
+
+ // Initialize suppression filter with ones
+ WebRtcSpl_MemSetW16((WebRtc_Word16*)inst->noiseSupFilter, 16384, HALF_ANAL_BLOCKL);
+
+ // Set the aggressiveness: default
+ inst->aggrMode = 0;
+
+ //initialize variables for new method
+ inst->priorNonSpeechProb = 8192; // Q14(0.5) prior probability for speech/noise
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ inst->prevMagnU16[i] = 0;
+ inst->prevNoiseU32[i] = 0; //previous noise-spectrum
+ inst->logLrtTimeAvgW32[i] = 0; //smooth LR ratio
+ inst->avgMagnPause[i] = 0; //conservative noise spectrum estimate
+ inst->initMagnEst[i] = 0; //initial average magnitude spectrum
+ }
+
+ //feature quantities
+ inst->thresholdSpecDiff = 50; //threshold for difference feature: determined on-line
+ inst->thresholdSpecFlat = 20480; //threshold for flatness: determined on-line
+ inst->featureLogLrt = inst->thresholdLogLrt; //average LRT factor (= threshold)
+ inst->featureSpecFlat = inst->thresholdSpecFlat; //spectral flatness (= threshold)
+ inst->featureSpecDiff = inst->thresholdSpecDiff; //spectral difference (= threshold)
+ inst->weightLogLrt = 6; //default weighting par for LRT feature
+ inst->weightSpecFlat = 0; //default weighting par for spectral flatness feature
+ inst->weightSpecDiff = 0; //default weighting par for spectral difference feature
+
+ inst->curAvgMagnEnergy = 0; //window time-average of input magnitude spectrum
+ inst->timeAvgMagnEnergy = 0; //normalization for spectral difference
+ inst->timeAvgMagnEnergyTmp = 0; //normalization for spectral difference
+
+ //histogram quantities: used to estimate/update thresholds for features
+ WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
+
+ inst->blockIndex = -1; //frame counter
+
+ //inst->modelUpdate = 500; //window for update
+ inst->modelUpdate = (1 << STAT_UPDATES); //window for update
+ inst->cntThresUpdate = 0; //counter feature thresholds updates
+
+ inst->sumMagn = 0;
+ inst->magnEnergy = 0;
+ inst->prevQMagn = 0;
+ inst->qNoise = 0;
+ inst->prevQNoise = 0;
+
+ inst->energyIn = 0;
+ inst->scaleEnergyIn = 0;
+
+ inst->whiteNoiseLevel = 0;
+ inst->pinkNoiseNumerator = 0;
+ inst->pinkNoiseExp = 0;
+ inst->minNorm = 15; // Start with full scale
+ inst->zeroInputSignal = 0;
+
+ //default mode
+ WebRtcNsx_set_policy_core(inst, 0);
+
+#ifdef NS_FILEDEBUG
+ inst->infile = fopen("indebug.pcm", "wb");
+ inst->outfile = fopen("outdebug.pcm", "wb");
+ inst->file1 = fopen("file1.pcm", "wb");
+ inst->file2 = fopen("file2.pcm", "wb");
+ inst->file3 = fopen("file3.pcm", "wb");
+ inst->file4 = fopen("file4.pcm", "wb");
+ inst->file5 = fopen("file5.pcm", "wb");
+#endif
+
+ // Initialize function pointers.
+ WebRtcNsx_NoiseEstimation = NoiseEstimationC;
+ WebRtcNsx_PrepareSpectrum = PrepareSpectrumC;
+ WebRtcNsx_SynthesisUpdate = SynthesisUpdateC;
+ WebRtcNsx_AnalysisUpdate = AnalysisUpdateC;
+ WebRtcNsx_Denormalize = DenormalizeC;
+ WebRtcNsx_CreateComplexBuffer = CreateComplexBufferC;
+
+#ifdef WEBRTC_DETECT_ARM_NEON
+ uint64_t features = WebRtc_GetCPUFeaturesARM();
+ if ((features & kCPUFeatureNEON) != 0)
+ {
+ WebRtcNsx_InitNeon();
+ }
+#elif defined(WEBRTC_ARCH_ARM_NEON)
+ WebRtcNsx_InitNeon();
+#endif
+
+ inst->initFlag = 1;
+
+ return 0;
+}
+
+int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode) {
+ // allow for modes:0,1,2,3
+ if (mode < 0 || mode > 3) {
+ return -1;
+ }
+
+ inst->aggrMode = mode;
+ if (mode == 0) {
+ inst->overdrive = 256; // Q8(1.0)
+ inst->denoiseBound = 8192; // Q14(0.5)
+ inst->gainMap = 0; // No gain compensation
+ } else if (mode == 1) {
+ inst->overdrive = 256; // Q8(1.0)
+ inst->denoiseBound = 4096; // Q14(0.25)
+ inst->factor2Table = kFactor2Aggressiveness1;
+ inst->gainMap = 1;
+ } else if (mode == 2) {
+ inst->overdrive = 282; // ~= Q8(1.1)
+ inst->denoiseBound = 2048; // Q14(0.125)
+ inst->factor2Table = kFactor2Aggressiveness2;
+ inst->gainMap = 1;
+ } else if (mode == 3) {
+ inst->overdrive = 320; // Q8(1.25)
+ inst->denoiseBound = 1475; // ~= Q14(0.09)
+ inst->factor2Table = kFactor2Aggressiveness3;
+ inst->gainMap = 1;
+ }
+ return 0;
+}
+
+// Extract thresholds for feature parameters
+// histograms are computed over some window_size (given by window_pars)
+// thresholds and weights are extracted every window
+// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
+// threshold and weights are returned in: inst->priorModelPars
+void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
+ WebRtc_UWord32 tmpU32;
+ WebRtc_UWord32 histIndex;
+ WebRtc_UWord32 posPeak1SpecFlatFX, posPeak2SpecFlatFX;
+ WebRtc_UWord32 posPeak1SpecDiffFX, posPeak2SpecDiffFX;
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word32 fluctLrtFX, thresFluctLrtFX;
+ WebRtc_Word32 avgHistLrtFX, avgSquareHistLrtFX, avgHistLrtComplFX;
+
+ WebRtc_Word16 j;
+ WebRtc_Word16 numHistLrt;
+
+ int i;
+ int useFeatureSpecFlat, useFeatureSpecDiff, featureSum;
+ int maxPeak1, maxPeak2;
+ int weightPeak1SpecFlat, weightPeak2SpecFlat;
+ int weightPeak1SpecDiff, weightPeak2SpecDiff;
+
+ //update histograms
+ if (!flag) {
+ // LRT
+ // Type casting to UWord32 is safe since negative values will not be wrapped to larger
+ // values than HIST_PAR_EST
+ histIndex = (WebRtc_UWord32)(inst->featureLogLrt);
+ if (histIndex < HIST_PAR_EST) {
+ inst->histLrt[histIndex]++;
+ }
+ // Spectral flatness
+ // (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
+ histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
+ if (histIndex < HIST_PAR_EST) {
+ inst->histSpecFlat[histIndex]++;
+ }
+ // Spectral difference
+ histIndex = HIST_PAR_EST;
+ if (inst->timeAvgMagnEnergy > 0) {
+ // Guard against division by zero
+ // If timeAvgMagnEnergy == 0 we have no normalizing statistics and
+ // therefore can't update the histogram
+ histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
+ inst->timeAvgMagnEnergy);
+ }
+ if (histIndex < HIST_PAR_EST) {
+ inst->histSpecDiff[histIndex]++;
+ }
+ }
+
+ // extract parameters for speech/noise probability
+ if (flag) {
+ useFeatureSpecDiff = 1;
+ //for LRT feature:
+ // compute the average over inst->featureExtractionParams.rangeAvgHistLrt
+ avgHistLrtFX = 0;
+ avgSquareHistLrtFX = 0;
+ numHistLrt = 0;
+ for (i = 0; i < BIN_SIZE_LRT; i++) {
+ j = (2 * i + 1);
+ tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
+ avgHistLrtFX += tmp32;
+ numHistLrt += inst->histLrt[i];
+ avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
+ }
+ avgHistLrtComplFX = avgHistLrtFX;
+ for (; i < HIST_PAR_EST; i++) {
+ j = (2 * i + 1);
+ tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
+ avgHistLrtComplFX += tmp32;
+ avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
+ }
+ fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
+ fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
+ thresFluctLrtFX = THRES_FLUCT_LRT * numHistLrt;
+ // get threshold for LRT feature:
+ tmpU32 = (FACTOR_1_LRT_DIFF * (WebRtc_UWord32)avgHistLrtFX);
+ if ((fluctLrtFX < thresFluctLrtFX) || (numHistLrt == 0) ||
+ (tmpU32 > (WebRtc_UWord32)(100 * numHistLrt))) {
+ //very low fluctuation, so likely noise
+ inst->thresholdLogLrt = inst->maxLrt;
+ } else {
+ tmp32 = (WebRtc_Word32)((tmpU32 << (9 + inst->stages)) / numHistLrt /
+ 25);
+ // check if value is within min/max range
+ inst->thresholdLogLrt = WEBRTC_SPL_SAT(inst->maxLrt,
+ tmp32,
+ inst->minLrt);
+ }
+ if (fluctLrtFX < thresFluctLrtFX) {
+ // Do not use difference feature if fluctuation of LRT feature is very low:
+ // most likely just noise state
+ useFeatureSpecDiff = 0;
+ }
+
+ // for spectral flatness and spectral difference: compute the main peaks of histogram
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecFlatFX = 0;
+ posPeak2SpecFlatFX = 0;
+ weightPeak1SpecFlat = 0;
+ weightPeak2SpecFlat = 0;
+
+ // peaks for flatness
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ if (inst->histSpecFlat[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecFlat = weightPeak1SpecFlat;
+ posPeak2SpecFlatFX = posPeak1SpecFlatFX;
+
+ maxPeak1 = inst->histSpecFlat[i];
+ weightPeak1SpecFlat = inst->histSpecFlat[i];
+ posPeak1SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
+ } else if (inst->histSpecFlat[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecFlat[i];
+ weightPeak2SpecFlat = inst->histSpecFlat[i];
+ posPeak2SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
+ }
+ }
+
+ // for spectral flatness feature
+ useFeatureSpecFlat = 1;
+ // merge the two peaks if they are close
+ if ((posPeak1SpecFlatFX - posPeak2SpecFlatFX < LIM_PEAK_SPACE_FLAT_DIFF)
+ && (weightPeak2SpecFlat * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecFlat)) {
+ weightPeak1SpecFlat += weightPeak2SpecFlat;
+ posPeak1SpecFlatFX = (posPeak1SpecFlatFX + posPeak2SpecFlatFX) >> 1;
+ }
+ //reject if weight of peaks is not large enough, or peak value too small
+ if (weightPeak1SpecFlat < THRES_WEIGHT_FLAT_DIFF || posPeak1SpecFlatFX
+ < THRES_PEAK_FLAT) {
+ useFeatureSpecFlat = 0;
+ } else { // if selected, get the threshold
+ // compute the threshold and check if value is within min/max range
+ inst->thresholdSpecFlat = WEBRTC_SPL_SAT(MAX_FLAT_Q10, FACTOR_2_FLAT_Q10
+ * posPeak1SpecFlatFX, MIN_FLAT_Q10); //Q10
+ }
+ // done with flatness feature
+
+ if (useFeatureSpecDiff) {
+ //compute two peaks for spectral difference
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecDiffFX = 0;
+ posPeak2SpecDiffFX = 0;
+ weightPeak1SpecDiff = 0;
+ weightPeak2SpecDiff = 0;
+ // peaks for spectral difference
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ if (inst->histSpecDiff[i] > maxPeak1) {
+ // Found new "first" peak
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecDiff = weightPeak1SpecDiff;
+ posPeak2SpecDiffFX = posPeak1SpecDiffFX;
+
+ maxPeak1 = inst->histSpecDiff[i];
+ weightPeak1SpecDiff = inst->histSpecDiff[i];
+ posPeak1SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
+ } else if (inst->histSpecDiff[i] > maxPeak2) {
+ // Found new "second" peak
+ maxPeak2 = inst->histSpecDiff[i];
+ weightPeak2SpecDiff = inst->histSpecDiff[i];
+ posPeak2SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
+ }
+ }
+
+ // merge the two peaks if they are close
+ if ((posPeak1SpecDiffFX - posPeak2SpecDiffFX < LIM_PEAK_SPACE_FLAT_DIFF)
+ && (weightPeak2SpecDiff * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecDiff)) {
+ weightPeak1SpecDiff += weightPeak2SpecDiff;
+ posPeak1SpecDiffFX = (posPeak1SpecDiffFX + posPeak2SpecDiffFX) >> 1;
+ }
+ // get the threshold value and check if value is within min/max range
+ inst->thresholdSpecDiff = WEBRTC_SPL_SAT(MAX_DIFF, FACTOR_1_LRT_DIFF
+ * posPeak1SpecDiffFX, MIN_DIFF); //5x bigger
+ //reject if weight of peaks is not large enough
+ if (weightPeak1SpecDiff < THRES_WEIGHT_FLAT_DIFF) {
+ useFeatureSpecDiff = 0;
+ }
+ // done with spectral difference feature
+ }
+
+ // select the weights between the features
+ // inst->priorModelPars[4] is weight for LRT: always selected
+ featureSum = 6 / (1 + useFeatureSpecFlat + useFeatureSpecDiff);
+ inst->weightLogLrt = featureSum;
+ inst->weightSpecFlat = useFeatureSpecFlat * featureSum;
+ inst->weightSpecDiff = useFeatureSpecDiff * featureSum;
+
+ // set histograms to zero for next update
+ WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
+ WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
+ } // end of flag == 1
+}
+
+
+// Compute spectral flatness on input spectrum
+// magn is the magnitude spectrum
+// spectral flatness is returned in inst->featureSpecFlat
+void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, WebRtc_UWord16* magn) {
+ WebRtc_UWord32 tmpU32;
+ WebRtc_UWord32 avgSpectralFlatnessNum, avgSpectralFlatnessDen;
+
+ WebRtc_Word32 tmp32;
+ WebRtc_Word32 currentSpectralFlatness, logCurSpectralFlatness;
+
+ WebRtc_Word16 zeros, frac, intPart;
+
+ int i;
+
+ // for flatness
+ avgSpectralFlatnessNum = 0;
+ avgSpectralFlatnessDen = inst->sumMagn - (WebRtc_UWord32)magn[0]; // Q(normData-stages)
+
+ // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
+ // flatness = exp( sum(log(magn[i]))/N - log(sum(magn[i])/N) )
+ // = exp( sum(log(magn[i]))/N ) * N / sum(magn[i])
+ // = 2^( sum(log2(magn[i]))/N - (log2(sum(magn[i])) - log2(N)) ) [This is used]
+ for (i = 1; i < inst->magnLen; i++) {
+ // First bin is excluded from spectrum measures. Number of bins is now a power of 2
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
+ frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_UWord32)(magn[i]) << zeros)
+ & 0x7FFFFFFF) >> 23);
+ // log2(magn(i))
+ assert(frac < 256);
+ tmpU32 = (WebRtc_UWord32)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]); // Q8
+ avgSpectralFlatnessNum += tmpU32; // Q8
+ } else {
+ //if at least one frequency component is zero, treat separately
+ tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
+ inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
+ return;
+ }
+ }
+ //ratio and inverse log: check for case of log(0)
+ zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
+ frac = (WebRtc_Word16)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
+ // log2(avgSpectralFlatnessDen)
+ assert(frac < 256);
+ tmp32 = (WebRtc_Word32)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]); // Q8
+ logCurSpectralFlatness = (WebRtc_Word32)avgSpectralFlatnessNum;
+ logCurSpectralFlatness += ((WebRtc_Word32)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
+ logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
+ logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
+ tmp32 = (WebRtc_Word32)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
+ & 0x0001FFFF)); //Q17
+ intPart = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
+ intPart += 7; // Shift 7 to get the output in Q10 (from Q17 = -17+10)
+ if (intPart > 0) {
+ currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
+ } else {
+ currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
+ }
+
+ //time average update of spectral flatness feature
+ tmp32 = currentSpectralFlatness - (WebRtc_Word32)inst->featureSpecFlat; // Q10
+ tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
+ inst->featureSpecFlat = (WebRtc_UWord32)((WebRtc_Word32)inst->featureSpecFlat
+ + WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
+ // done with flatness feature
+}
+
+
+// Compute the difference measure between input spectrum and a template/learned noise spectrum
+// magn_tmp is the input spectrum
+// the reference/template spectrum is inst->magn_avg_pause[i]
+// returns (normalized) spectral difference in inst->featureSpecDiff
+void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, WebRtc_UWord16* magnIn) {
+ // This is to be calculated:
+ // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+ WebRtc_UWord32 varMagnUFX, varPauseUFX, avgDiffNormMagnUFX;
+
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 avgPauseFX, avgMagnFX, covMagnPauseFX;
+ WebRtc_Word32 maxPause, minPause;
+
+ WebRtc_Word16 tmp16no1;
+
+ int i, norm32, nShifts;
+
+ avgPauseFX = 0;
+ maxPause = 0;
+ minPause = inst->avgMagnPause[0]; // Q(prevQMagn)
+ // compute average quantities
+ for (i = 0; i < inst->magnLen; i++) {
+ // Compute mean of magn_pause
+ avgPauseFX += inst->avgMagnPause[i]; // in Q(prevQMagn)
+ maxPause = WEBRTC_SPL_MAX(maxPause, inst->avgMagnPause[i]);
+ minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]);
+ }
+ // normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
+ avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
+ avgMagnFX = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
+ // Largest possible deviation in magnPause for (co)var calculations
+ tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
+ // Get number of shifts to make sure we don't get wrap around in varPause
+ nShifts = WEBRTC_SPL_MAX(0, 10 + inst->stages - WebRtcSpl_NormW32(tmp32no1));
+
+ varMagnUFX = 0;
+ varPauseUFX = 0;
+ covMagnPauseFX = 0;
+ for (i = 0; i < inst->magnLen; i++) {
+ // Compute var and cov of magn and magn_pause
+ tmp16no1 = (WebRtc_Word16)((WebRtc_Word32)magnIn[i] - avgMagnFX);
+ tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
+ varMagnUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
+ covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
+ varPauseUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
+ }
+ //update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
+ inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
+ + inst->stages - 1);
+
+ avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
+ if ((varPauseUFX) && (covMagnPauseFX)) {
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
+ norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
+ if (norm32 > 0) {
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
+ } else {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
+ }
+ tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
+
+ nShifts += norm32;
+ nShifts <<= 1;
+ if (nShifts < 0) {
+ varPauseUFX >>= (-nShifts); // Q(2*(qMagn+norm32+minPause))
+ nShifts = 0;
+ }
+ if (varPauseUFX > 0) {
+ // Q(2*(qMagn+norm32-16+minPause))
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX);
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
+
+ // Q(2*qMagn)
+ avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1);
+ } else {
+ avgDiffNormMagnUFX = 0;
+ }
+ }
+ //normalize and compute time average update of difference feature
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
+ if (inst->featureSpecDiff > tmpU32no1) {
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
+ SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
+ inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
+ SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
+ inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
+ }
+}
+
+// Compute speech/noise probability
+// speech/noise probability is returned in: probSpeechFinal
+//snrLocPrior is the prior SNR for each frequency (in Q11)
+//snrLocPost is the post SNR for each frequency (in Q11)
+void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, WebRtc_UWord16* nonSpeechProbFinal,
+ WebRtc_UWord32* priorLocSnr, WebRtc_UWord32* postLocSnr) {
+ WebRtc_UWord32 zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
+
+ WebRtc_Word32 invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
+ WebRtc_Word32 frac32, logTmp;
+ WebRtc_Word32 logLrtTimeAvgKsumFX;
+
+ WebRtc_Word16 indPriorFX16;
+ WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
+
+ int i, normTmp, normTmp2, nShifts;
+
+ // compute feature based on average LR factor
+ // this is the average over all frequencies of the smooth log LRT
+ logLrtTimeAvgKsumFX = 0;
+ for (i = 0; i < inst->magnLen; i++) {
+ besselTmpFX32 = (WebRtc_Word32)postLocSnr[i]; // Q11
+ normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
+ num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp)
+ if (normTmp > 10) {
+ den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
+ } else {
+ den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
+ }
+ if (den > 0) {
+ besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
+ } else {
+ besselTmpFX32 -= num; // Q11
+ }
+
+ // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]);
+ // Here, LRT_TAVG = 0.5
+ zeros = WebRtcSpl_NormU32(priorLocSnr[i]);
+ frac32 = (WebRtc_Word32)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19);
+ tmp32 = WEBRTC_SPL_MUL(frac32, frac32);
+ tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19);
+ tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)frac32, 5412, 12);
+ frac32 = tmp32 + 37;
+ // tmp32 = log2(priorLocSnr[i])
+ tmp32 = (WebRtc_Word32)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
+ logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12
+ inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
+
+ logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12
+ }
+ inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2
+ // done with computation of LR factor
+
+ //
+ //compute the indicator functions
+ //
+
+ // average LRT feature
+ // FLOAT code
+ // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0);
+ tmpIndFX = 16384; // Q14(1.0)
+ tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12
+ nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5;
+ //use larger width in tanh map for pause regions
+ if (tmp32no1 < 0) {
+ tmpIndFX = 0;
+ tmp32no1 = -tmp32no1;
+ //widthPrior = widthPrior * 2.0;
+ nShifts++;
+ }
+ tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
+ // compute indicator function: sigmoid map
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14);
+ if ((tableIndex < 16) && (tableIndex >= 0)) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmp32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
+ if (tmpIndFX == 0) {
+ tmpIndFX = 8192 - tmp16no2; // Q14
+ } else {
+ tmpIndFX = 8192 + tmp16no2; // Q14
+ }
+ }
+ indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14
+
+ //spectral flatness feature
+ if (inst->weightSpecFlat) {
+ tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10
+ tmpIndFX = 16384; // Q14(1.0)
+ //use larger width in tanh map for pause regions
+ tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10
+ nShifts = 4;
+ if (inst->thresholdSpecFlat < tmpU32no1) {
+ tmpIndFX = 0;
+ tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat;
+ //widthPrior = widthPrior * 2.0;
+ nShifts++;
+ }
+ tmp32no1 = (WebRtc_Word32)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
+ nShifts), 25); //Q14
+ tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14
+ // compute indicator function: sigmoid map
+ // FLOAT code
+ // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0);
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
+ if (tableIndex < 16) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
+ if (tmpIndFX) {
+ tmpIndFX = 8192 + tmp16no2; // Q14
+ } else {
+ tmpIndFX = 8192 - tmp16no2; // Q14
+ }
+ }
+ indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14
+ }
+
+ //for template spectral-difference
+ if (inst->weightSpecDiff) {
+ tmpU32no1 = 0;
+ if (inst->featureSpecDiff) {
+ normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
+ WebRtcSpl_NormU32(inst->featureSpecDiff));
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages
+ - normTmp);
+ if (tmpU32no2 > 0) {
+ // Q(20 - inst->stages)
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2);
+ } else {
+ tmpU32no1 = (WebRtc_UWord32)(0x7fffffff);
+ }
+ }
+ tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25);
+ tmpU32no2 = tmpU32no1 - tmpU32no3;
+ nShifts = 1;
+ tmpIndFX = 16384; // Q14(1.0)
+ //use larger width in tanh map for pause regions
+ if (tmpU32no2 & 0x80000000) {
+ tmpIndFX = 0;
+ tmpU32no2 = tmpU32no3 - tmpU32no1;
+ //widthPrior = widthPrior * 2.0;
+ nShifts--;
+ }
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
+ // compute indicator function: sigmoid map
+ /* FLOAT code
+ indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
+ */
+ tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
+ if (tableIndex < 16) {
+ tmp16no2 = kIndicatorTable[tableIndex];
+ tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
+ frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
+ tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ tmp16no1, frac, 14);
+ if (tmpIndFX) {
+ tmpIndFX = 8192 + tmp16no2;
+ } else {
+ tmpIndFX = 8192 - tmp16no2;
+ }
+ }
+ indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14
+ }
+
+ //combine the indicator function with the feature weights
+ // FLOAT code
+ // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2);
+ indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14
+ // done with computing indicator function
+
+ //compute the prior probability
+ // FLOAT code
+ // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb);
+ tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14
+ inst->priorNonSpeechProb += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
+ PRIOR_UPDATE_Q14, tmp16, 14); // Q14
+
+ //final speech probability: combine prior model with LR factor:
+
+ memset(nonSpeechProbFinal, 0, sizeof(WebRtc_UWord16) * inst->magnLen);
+
+ if (inst->priorNonSpeechProb > 0) {
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT code
+ // invLrt = exp(inst->logLrtTimeAvg[i]);
+ // invLrt = inst->priorSpeechProb * invLrt;
+ // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt);
+ // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt;
+ // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt);
+ if (inst->logLrtTimeAvgW32[i] < 65300) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637),
+ 14); // Q12
+ intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12);
+ if (intPart < -8) {
+ intPart = -8;
+ }
+ frac = (WebRtc_Word16)(tmp32no1 & 0x00000fff); // Q12
+
+ // Quadratic approximation of 2^frac
+ tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
+ tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
+ invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
+ + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
+
+ normTmp = WebRtcSpl_NormW32(invLrtFX);
+ normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
+ if (normTmp + normTmp2 >= 7) {
+ if (normTmp + normTmp2 < 15) {
+ invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp);
+ // Q(normTmp+normTmp2-7)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb));
+ // Q(normTmp+normTmp2+7)
+ invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14
+ } else {
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22
+ invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
+ }
+
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22
+
+ nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1,
+ (WebRtc_Word32)inst->priorNonSpeechProb + invLrtFX); // Q8
+ }
+ }
+ }
+ }
+}
+
+// Transform input (speechFrame) to frequency domain magnitude (magnU16)
+void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, WebRtc_UWord16* magnU16) {
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2;
+
+ WebRtc_Word32 tmp_1_w32 = 0;
+ WebRtc_Word32 tmp_2_w32 = 0;
+ WebRtc_Word32 sum_log_magn = 0;
+ WebRtc_Word32 sum_log_i_log_magn = 0;
+
+ WebRtc_UWord16 sum_log_magn_u16 = 0;
+ WebRtc_UWord16 tmp_u16 = 0;
+
+ WebRtc_Word16 sum_log_i = 0;
+ WebRtc_Word16 sum_log_i_square = 0;
+ WebRtc_Word16 frac = 0;
+ WebRtc_Word16 log2 = 0;
+ WebRtc_Word16 matrix_determinant = 0;
+ WebRtc_Word16 winData[ANAL_BLOCKL_MAX], maxWinData;
+ WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
+
+ int i, j;
+ int zeros;
+ int net_norm = 0;
+ int right_shifts_in_magnU16 = 0;
+ int right_shifts_in_initMagnEst = 0;
+
+ // Update analysis buffer for lower band, and window data before FFT.
+ WebRtcNsx_AnalysisUpdate(inst, winData, speechFrame);
+
+ // Get input energy
+ inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
+
+ // Reset zero input flag
+ inst->zeroInputSignal = 0;
+ // Acquire norm for winData
+ maxWinData = WebRtcSpl_MaxAbsValueW16(winData, inst->anaLen);
+ inst->normData = WebRtcSpl_NormW16(maxWinData);
+ if (maxWinData == 0) {
+ // Treat zero input separately.
+ inst->zeroInputSignal = 1;
+ return;
+ }
+
+ // Determine the net normalization in the frequency domain
+ net_norm = inst->stages - inst->normData;
+ // Track lowest normalization factor and use it to prevent wrap around in shifting
+ right_shifts_in_magnU16 = inst->normData - inst->minNorm;
+ right_shifts_in_initMagnEst = WEBRTC_SPL_MAX(-right_shifts_in_magnU16, 0);
+ inst->minNorm -= right_shifts_in_initMagnEst;
+ right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0);
+
+ // create realImag as winData interleaved with zeros (= imag. part), normalize it
+ WebRtcNsx_CreateComplexBuffer(inst, winData, realImag);
+
+ // bit-reverse position of elements in array and FFT the array
+ WebRtcSpl_ComplexBitReverse(realImag, inst->stages); // Q(normData-stages)
+ WebRtcSpl_ComplexFFT(realImag, inst->stages, 1);
+
+ inst->imag[0] = 0; // Q(normData-stages)
+ inst->imag[inst->anaLen2] = 0;
+ inst->real[0] = realImag[0]; // Q(normData-stages)
+ inst->real[inst->anaLen2] = realImag[inst->anaLen];
+ // Q(2*(normData-stages))
+ inst->magnEnergy = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[0], inst->real[0]);
+ inst->magnEnergy += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[inst->anaLen2],
+ inst->real[inst->anaLen2]);
+ magnU16[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[0]); // Q(normData-stages)
+ magnU16[inst->anaLen2] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[inst->anaLen2]);
+ inst->sumMagn = (WebRtc_UWord32)magnU16[0]; // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[inst->anaLen2];
+
+ if (inst->blockIndex >= END_STARTUP_SHORT) {
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ inst->real[i] = realImag[j];
+ inst->imag[i] = -realImag[j + 1];
+ // magnitude spectrum
+ // energy in Q(2*(normData-stages))
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
+ tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
+ inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
+
+ magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
+ }
+ } else {
+ //
+ // Gather information during startup for noise parameter estimation
+ //
+
+ // Switch initMagnEst to Q(minNorm-stages)
+ inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
+ right_shifts_in_initMagnEst);
+ inst->initMagnEst[inst->anaLen2] =
+ WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
+ right_shifts_in_initMagnEst); // Q(minNorm-stages)
+
+ // Shift magnU16 to same domain as initMagnEst
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[0],
+ right_shifts_in_magnU16); // Q(minNorm-stages)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[inst->anaLen2],
+ right_shifts_in_magnU16); // Q(minNorm-stages)
+
+ // Update initMagnEst
+ inst->initMagnEst[0] += tmpU32no1; // Q(minNorm-stages)
+ inst->initMagnEst[inst->anaLen2] += tmpU32no2; // Q(minNorm-stages)
+
+ log2 = 0;
+ if (magnU16[inst->anaLen2]) {
+ // Calculate log2(magnU16[inst->anaLen2])
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[inst->anaLen2]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[inst->anaLen2] << zeros) &
+ 0x7FFFFFFF) >> 23); // Q8
+ // log2(magnU16(i)) in Q8
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]);
+ }
+
+ sum_log_magn = (WebRtc_Word32)log2; // Q8
+ // sum_log_i_log_magn in Q17
+ sum_log_i_log_magn = (WEBRTC_SPL_MUL_16_16(kLogIndex[inst->anaLen2], log2) >> 3);
+
+ for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ inst->real[i] = realImag[j];
+ inst->imag[i] = -realImag[j + 1];
+ // magnitude spectrum
+ // energy in Q(2*(normData-stages))
+ tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
+ tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
+ inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
+
+ magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
+ inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
+
+ // Switch initMagnEst to Q(minNorm-stages)
+ inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
+ right_shifts_in_initMagnEst);
+
+ // Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[i],
+ right_shifts_in_magnU16);
+ // Update initMagnEst
+ inst->initMagnEst[i] += tmpU32no1; // Q(minNorm-stages)
+
+ if (i >= kStartBand) {
+ // For pink noise estimation. Collect data neglecting lower frequency band
+ log2 = 0;
+ if (magnU16[i]) {
+ zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[i]);
+ frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[i] << zeros) &
+ 0x7FFFFFFF) >> 23);
+ // log2(magnU16(i)) in Q8
+ assert(frac < 256);
+ log2 = (WebRtc_Word16)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ }
+ sum_log_magn += (WebRtc_Word32)log2; // Q8
+ // sum_log_i_log_magn in Q17
+ sum_log_i_log_magn += (WEBRTC_SPL_MUL_16_16(kLogIndex[i], log2) >> 3);
+ }
+ }
+
+ //
+ //compute simplified noise model during startup
+ //
+
+ // Estimate White noise
+
+ // Switch whiteNoiseLevel to Q(minNorm-stages)
+ inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
+ right_shifts_in_initMagnEst);
+
+ // Update the average magnitude spectrum, used as noise estimate.
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
+
+ // Replacing division above with 'stages' shifts
+ // Shift to same Q-domain as whiteNoiseLevel
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
+ // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
+ assert(END_STARTUP_SHORT < 128);
+ inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
+
+ // Estimate Pink noise parameters
+ // Denominator used in both parameter estimates.
+ // The value is only dependent on the size of the frequency band (kStartBand)
+ // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
+ assert(kStartBand < 66);
+ matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
+ sum_log_i = kSumLogIndex[kStartBand]; // Q5
+ sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
+ if (inst->fs == 8000) {
+ // Adjust values to shorter blocks in narrow band.
+ tmp_1_w32 = (WebRtc_Word32)matrix_determinant;
+ tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
+ tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
+ tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4);
+ tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)
+ (inst->magnLen - kStartBand), kSumSquareLogIndex[65], 2);
+ matrix_determinant = (WebRtc_Word16)tmp_1_w32;
+ sum_log_i -= kSumLogIndex[65]; // Q5
+ sum_log_i_square -= kSumSquareLogIndex[65]; // Q2
+ }
+
+ // Necessary number of shifts to fit sum_log_magn in a word16
+ zeros = 16 - WebRtcSpl_NormW32(sum_log_magn);
+ if (zeros < 0) {
+ zeros = 0;
+ }
+ tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
+ sum_log_magn_u16 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
+
+ // Calculate and update pinkNoiseNumerator. Result in Q11.
+ tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((WebRtc_UWord32)sum_log_i_log_magn, 12); // Q5
+
+ // Shift the largest value of sum_log_i and tmp32no3 before multiplication
+ tmp_u16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)sum_log_i, 1); // Q6
+ if ((WebRtc_UWord32)sum_log_i > tmpU32no1) {
+ tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros);
+ } else {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
+ }
+ tmp_2_w32 -= (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
+ matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
+ tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
+ tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11
+ if (tmp_2_w32 < 0) {
+ tmp_2_w32 = 0;
+ }
+ inst->pinkNoiseNumerator += tmp_2_w32; // Q11
+
+ // Calculate and update pinkNoiseExp. Result in Q14.
+ tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i, sum_log_magn_u16); // Q(14-zeros)
+ tmp_1_w32 = WEBRTC_SPL_RSHIFT_W32(sum_log_i_log_magn, 3 + zeros);
+ tmp_1_w32 = WEBRTC_SPL_MUL((WebRtc_Word32)(inst->magnLen - kStartBand),
+ tmp_1_w32);
+ tmp_2_w32 -= tmp_1_w32; // Q(14-zeros)
+ if (tmp_2_w32 > 0) {
+ // If the exponential parameter is negative force it to zero, which means a
+ // flat spectrum.
+ tmp_1_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q14
+ inst->pinkNoiseExp += WEBRTC_SPL_SAT(16384, tmp_1_w32, 0); // Q14
+ }
+ }
+}
+
+void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) {
+ WebRtc_Word32 energyOut;
+
+ WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
+ WebRtc_Word16 tmp16no1, tmp16no2;
+ WebRtc_Word16 energyRatio;
+ WebRtc_Word16 gainFactor, gainFactor1, gainFactor2;
+
+ int i;
+ int outCIFFT;
+ int scaleEnergyOut = 0;
+
+ if (inst->zeroInputSignal) {
+ // synthesize the special case of zero input
+ // read out fully processed segment
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrame[i] = inst->synthesisBuffer[i]; // Q0
+ }
+ // update synthesis buffer
+ WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ inst->synthesisBuffer + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
+ inst->blockLen10ms);
+ return;
+ }
+
+ // Filter the data in the frequency domain, and create spectrum.
+ WebRtcNsx_PrepareSpectrum(inst, realImag);
+
+ // bit-reverse position of elements in array and IFFT it
+ WebRtcSpl_ComplexBitReverse(realImag, inst->stages);
+ outCIFFT = WebRtcSpl_ComplexIFFT(realImag, inst->stages, 1);
+
+ // Denormalize.
+ WebRtcNsx_Denormalize(inst, realImag, outCIFFT);
+
+ //scale factor: only do it after END_STARTUP_LONG time
+ gainFactor = 8192; // 8192 = Q13(1.0)
+ if (inst->gainMap == 1 &&
+ inst->blockIndex > END_STARTUP_LONG &&
+ inst->energyIn > 0) {
+ energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
+ if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000)) {
+ energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
+ - inst->scaleEnergyIn);
+ } else {
+ inst->energyIn = WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 8 + scaleEnergyOut
+ - inst->scaleEnergyIn); // Q(-8-scaleEnergyOut)
+ }
+
+ assert(inst->energyIn > 0);
+ energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut
+ + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
+ // Limit the ratio to [0, 1] in Q8, i.e., [0, 256]
+ energyRatio = WEBRTC_SPL_SAT(256, energyRatio, 0);
+
+ // all done in lookup tables now
+ assert(energyRatio < 257);
+ gainFactor1 = kFactor1Table[energyRatio]; // Q8
+ gainFactor2 = inst->factor2Table[energyRatio]; // Q8
+
+ //combine both scales with speech/noise prob: note prior (priorSpeechProb) is not frequency dependent
+
+ // factor = inst->priorSpeechProb*factor1 + (1.0-inst->priorSpeechProb)*factor2; // original code
+ tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(16384 - inst->priorNonSpeechProb,
+ gainFactor1, 14); // Q13 16384 = Q14(1.0)
+ tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
+ gainFactor2, 14); // Q13;
+ gainFactor = tmp16no1 + tmp16no2; // Q13
+ } // out of flag_gain_map==1
+
+ // Synthesis, read out fully processed segment, and update synthesis buffer.
+ WebRtcNsx_SynthesisUpdate(inst, outFrame, gainFactor);
+}
+
+int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFrameHB,
+ short* outFrame, short* outFrameHB) {
+ // main routine for noise suppression
+
+ WebRtc_UWord32 tmpU32no1, tmpU32no2, tmpU32no3;
+ WebRtc_UWord32 satMax, maxNoiseU32;
+ WebRtc_UWord32 tmpMagnU32, tmpNoiseU32;
+ WebRtc_UWord32 nearMagnEst;
+ WebRtc_UWord32 noiseUpdateU32;
+ WebRtc_UWord32 noiseU32[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 postLocSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 priorLocSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 prevNearSnr[HALF_ANAL_BLOCKL];
+ WebRtc_UWord32 curNearSnr;
+ WebRtc_UWord32 priorSnr;
+ WebRtc_UWord32 noise_estimate = 0;
+ WebRtc_UWord32 noise_estimate_avg = 0;
+ WebRtc_UWord32 numerator = 0;
+
+ WebRtc_Word32 tmp32no1, tmp32no2;
+ WebRtc_Word32 pink_noise_num_avg = 0;
+
+ WebRtc_UWord16 tmpU16no1;
+ WebRtc_UWord16 magnU16[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 prevNoiseU16[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 nonSpeechProbFinal[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 gammaNoise, prevGammaNoise;
+ WebRtc_UWord16 noiseSupFilterTmp[HALF_ANAL_BLOCKL];
+
+ WebRtc_Word16 qMagn, qNoise;
+ WebRtc_Word16 avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
+ WebRtc_Word16 pink_noise_exp_avg = 0;
+
+ int i;
+ int nShifts, postShifts;
+ int norm32no1, norm32no2;
+ int flag, sign;
+ int q_domain_to_use = 0;
+
+ // Code for ARMv7-Neon platform assumes the following:
+ assert(inst->anaLen % 16 == 0);
+ assert(inst->anaLen2 % 8 == 0);
+ assert(inst->blockLen10ms % 16 == 0);
+ assert(inst->magnLen == inst->anaLen2 + 1);
+
+#ifdef NS_FILEDEBUG
+ fwrite(spframe, sizeof(short), inst->blockLen10ms, inst->infile);
+#endif
+
+ // Check that initialization has been done
+ if (inst->initFlag != 1) {
+ return -1;
+ }
+ // Check for valid pointers based on sampling rate
+ if ((inst->fs == 32000) && (speechFrameHB == NULL)) {
+ return -1;
+ }
+
+ // Store speechFrame and transform to frequency domain
+ WebRtcNsx_DataAnalysis(inst, speechFrame, magnU16);
+
+ if (inst->zeroInputSignal) {
+ WebRtcNsx_DataSynthesis(inst, outFrame);
+
+ if (inst->fs == 32000) {
+ // update analysis buffer for H band
+ // append new data to buffer FX
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms,
+ inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms,
+ speechFrameHB, inst->blockLen10ms);
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
+ }
+ } // end of H band gain computation
+ return 0;
+ }
+
+ // Update block index when we have something to process
+ inst->blockIndex++;
+ //
+
+ // Norm of magn
+ qMagn = inst->normData - inst->stages;
+
+ // Compute spectral flatness on input spectrum
+ WebRtcNsx_ComputeSpectralFlatness(inst, magnU16);
+
+ // quantile noise estimate
+ WebRtcNsx_NoiseEstimation(inst, magnU16, noiseU32, &qNoise);
+
+ //noise estimate from previous frame
+ for (i = 0; i < inst->magnLen; i++) {
+ prevNoiseU16[i] = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
+ }
+
+ if (inst->blockIndex < END_STARTUP_SHORT) {
+ // Noise Q-domain to be used later; see description at end of section.
+ q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages);
+
+ // Calculate frequency independent parts in parametric noise estimate and calculate
+ // the estimate for the lower frequency band (same values for all frequency bins)
+ if (inst->pinkNoiseExp) {
+ pink_noise_exp_avg = (WebRtc_Word16)WebRtcSpl_DivW32W16(inst->pinkNoiseExp,
+ (WebRtc_Word16)(inst->blockIndex + 1)); // Q14
+ pink_noise_num_avg = WebRtcSpl_DivW32W16(inst->pinkNoiseNumerator,
+ (WebRtc_Word16)(inst->blockIndex + 1)); // Q11
+ WebRtcNsx_CalcParametricNoiseEstimate(inst,
+ pink_noise_exp_avg,
+ pink_noise_num_avg,
+ kStartBand,
+ &noise_estimate,
+ &noise_estimate_avg);
+ } else {
+ // Use white noise estimate if we have poor pink noise parameter estimates
+ noise_estimate = inst->whiteNoiseLevel; // Q(minNorm-stages)
+ noise_estimate_avg = noise_estimate / (inst->blockIndex + 1); // Q(minNorm-stages)
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Estimate the background noise using the pink noise parameters if permitted
+ if ((inst->pinkNoiseExp) && (i >= kStartBand)) {
+ // Reset noise_estimate
+ noise_estimate = 0;
+ noise_estimate_avg = 0;
+ // Calculate the parametric noise estimate for current frequency bin
+ WebRtcNsx_CalcParametricNoiseEstimate(inst,
+ pink_noise_exp_avg,
+ pink_noise_num_avg,
+ i,
+ &noise_estimate,
+ &noise_estimate_avg);
+ }
+ // Calculate parametric Wiener filter
+ noiseSupFilterTmp[i] = inst->denoiseBound;
+ if (inst->initMagnEst[i]) {
+ // numerator = (initMagnEst - noise_estimate * overdrive)
+ // Result in Q(8+minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
+ numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8);
+ if (numerator > tmpU32no1) {
+ // Suppression filter coefficient larger than zero, so calculate.
+ numerator -= tmpU32no1;
+
+ // Determine number of left shifts in numerator for best accuracy after
+ // division
+ nShifts = WebRtcSpl_NormU32(numerator);
+ nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
+
+ // Shift numerator to Q(nShifts+8+minNorm-stages)
+ numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts);
+
+ // Shift denominator to Q(nShifts-6+minNorm-stages)
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
+ if (tmpU32no1 == 0) {
+ // This is only possible if numerator = 0, in which case
+ // we don't need any division.
+ tmpU32no1 = 1;
+ }
+ tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
+ noiseSupFilterTmp[i] = (WebRtc_UWord16)WEBRTC_SPL_SAT(16384, tmpU32no2,
+ (WebRtc_UWord32)(inst->denoiseBound)); // Q14
+ }
+ }
+ // Weight quantile noise 'noiseU32' with modeled noise 'noise_estimate_avg'
+ // 'noiseU32 is in Q(qNoise) and 'noise_estimate' in Q(minNorm-stages)
+ // To guarantee that we do not get wrap around when shifting to the same domain
+ // we use the lowest one. Furthermore, we need to save 6 bits for the weighting.
+ // 'noise_estimate_avg' can handle this operation by construction, but 'noiseU32'
+ // may not.
+
+ // Shift 'noiseU32' to 'q_domain_to_use'
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
+ // Shift 'noise_estimate_avg' to 'q_domain_to_use'
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
+ - q_domain_to_use);
+ // Make a simple check to see if we have enough room for weighting 'tmpU32no1'
+ // without wrap around
+ nShifts = 0;
+ if (tmpU32no1 & 0xfc000000) {
+ tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
+ nShifts = 6;
+ }
+ tmpU32no1 *= inst->blockIndex;
+ tmpU32no2 *= (END_STARTUP_SHORT - inst->blockIndex);
+ // Add them together and divide by startup length
+ noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
+ // Shift back if necessary
+ noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
+ }
+ // Update new Q-domain for 'noiseU32'
+ qNoise = q_domain_to_use;
+ }
+ // compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // substituting division with shift ending up in Q(-2*stages)
+ inst->timeAvgMagnEnergyTmp
+ += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
+ 2 * inst->normData + inst->stages - 1);
+ inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
+ inst->blockIndex + 1);
+ }
+
+ //start processing at frames == converged+1
+ // STEP 1: compute prior and post SNR based on quantile noise estimates
+
+ // compute direct decision (DD) estimate of prior SNR: needed for new method
+ satMax = (WebRtc_UWord32)1048575;// Largest possible value without getting overflow despite shifting 12 steps
+ postShifts = 6 + qMagn - qNoise;
+ nShifts = 5 - inst->prevQMagn + inst->prevQNoise;
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT:
+ // post SNR
+ // postLocSnr[i] = 0.0;
+ // if (magn[i] > noise[i])
+ // {
+ // postLocSnr[i] = magn[i] / (noise[i] + 0.0001);
+ // }
+ // // previous post SNR
+ // // previous estimate: based on previous frame with gain filter (smooth is previous filter)
+ //
+ // prevNearSnr[i] = inst->prevMagnU16[i] / (inst->noisePrev[i] + 0.0001) * (inst->smooth[i]);
+ //
+ // // DD estimate is sum of two terms: current estimate and previous estimate
+ // // directed decision update of priorSnr (or we actually store [2*priorSnr+1])
+ //
+ // priorLocSnr[i] = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * (postLocSnr[i] - 1.0);
+
+ // calculate post SNR: output in Q11
+ postLocSnr[i] = 2048; // 1.0 in Q11
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], 6); // Q(6+qMagn)
+ if (postShifts < 0) {
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
+ }
+ if (tmpU32no1 > tmpU32no2) {
+ // Current magnitude larger than noise
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ } else {
+ postLocSnr[i] = satMax;
+ }
+ }
+
+ // calculate prevNearSnr[i] and save for later instead of recalculating it later
+ nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
+
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ } else {
+ tmpU32no1 = satMax; // Q11
+ }
+ prevNearSnr[i] = tmpU32no1; // Q11
+
+ //directed decision update of priorSnr
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
+ priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
+ // priorLocSnr = 1 + 2*priorSnr
+ priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
+ } // end of loop over frequencies
+ // done with step 1: DD computation of prior and post SNR
+
+ // STEP 2: compute speech/noise likelihood
+
+ //compute difference of input spectrum with learned/estimated noise spectrum
+ WebRtcNsx_ComputeSpectralDifference(inst, magnU16);
+ //compute histograms for determination of parameters (thresholds and weights for features)
+ //parameters are extracted once every window time (=inst->modelUpdate)
+ //counter update
+ inst->cntThresUpdate++;
+ flag = (int)(inst->cntThresUpdate == inst->modelUpdate);
+ //update histogram
+ WebRtcNsx_FeatureParameterExtraction(inst, flag);
+ //compute model parameters
+ if (flag) {
+ inst->cntThresUpdate = 0; // Reset counter
+ //update every window:
+ // get normalization for spectral difference for next window estimate
+
+ // Shift to Q(-2*stages)
+ inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
+
+ tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
+ // Update featureSpecDiff
+ if ((tmpU32no1 != inst->timeAvgMagnEnergy) && (inst->featureSpecDiff) &&
+ (inst->timeAvgMagnEnergy > 0)) {
+ norm32no1 = 0;
+ tmpU32no3 = tmpU32no1;
+ while (0xFFFF0000 & tmpU32no3) {
+ tmpU32no3 >>= 1;
+ norm32no1++;
+ }
+ tmpU32no2 = inst->featureSpecDiff;
+ while (0xFFFF0000 & tmpU32no2) {
+ tmpU32no2 >>= 1;
+ norm32no1++;
+ }
+ tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
+ tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
+ if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1) {
+ inst->featureSpecDiff = 0x007FFFFF;
+ } else {
+ inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
+ WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
+ }
+ }
+
+ inst->timeAvgMagnEnergy = tmpU32no1; // Q(-2*stages)
+ inst->curAvgMagnEnergy = 0;
+ }
+
+ //compute speech/noise probability
+ WebRtcNsx_SpeechNoiseProb(inst, nonSpeechProbFinal, priorLocSnr, postLocSnr);
+
+ //time-avg parameter for noise update
+ gammaNoise = NOISE_UPDATE_Q8; // Q8
+
+ maxNoiseU32 = 0;
+ postShifts = inst->prevQNoise - qMagn;
+ nShifts = inst->prevQMagn - qMagn;
+ for (i = 0; i < inst->magnLen; i++) {
+ // temporary noise update: use it for speech frames if update value is less than previous
+ // the formula has been rewritten into:
+ // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
+
+ if (postShifts < 0) {
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
+ } else {
+ tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
+ }
+ if (prevNoiseU16[i] > tmpU32no2) {
+ sign = -1;
+ tmpU32no1 = prevNoiseU16[i] - tmpU32no2;
+ } else {
+ sign = 1;
+ tmpU32no1 = tmpU32no2 - prevNoiseU16[i];
+ }
+ noiseUpdateU32 = inst->prevNoiseU32[i]; // Q(prevQNoise+11)
+ tmpU32no3 = 0;
+ if ((tmpU32no1) && (nonSpeechProbFinal[i])) {
+ // This value will be used later, if gammaNoise changes
+ tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
+ if (0x7c000000 & tmpU32no3) {
+ // Shifting required before multiplication
+ tmpU32no2
+ = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
+ } else {
+ // We can do shifting after multiplication
+ tmpU32no2
+ = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
+ }
+ if (sign > 0) {
+ noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
+ } else {
+ // This operation is safe. We can never get wrap around, since worst
+ // case scenario means magnU16 = 0
+ noiseUpdateU32 -= tmpU32no2; // Q(prevQNoise+11)
+ }
+ }
+
+ //increase gamma (i.e., less noise update) for frame likely to be speech
+ prevGammaNoise = gammaNoise;
+ gammaNoise = NOISE_UPDATE_Q8;
+ //time-constant based on speech/noise state
+ //increase gamma (i.e., less noise update) for frames likely to be speech
+ if (nonSpeechProbFinal[i] < ONE_MINUS_PROB_RANGE_Q8) {
+ gammaNoise = GAMMA_NOISE_TRANS_AND_SPEECH_Q8;
+ }
+
+ if (prevGammaNoise != gammaNoise) {
+ // new noise update
+ // this line is the same as above, only that the result is stored in a different variable and the gammaNoise
+ // has changed
+ //
+ // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
+
+ if (0x7c000000 & tmpU32no3) {
+ // Shifting required before multiplication
+ tmpU32no2
+ = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
+ } else {
+ // We can do shifting after multiplication
+ tmpU32no2
+ = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
+ }
+ if (sign > 0) {
+ tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
+ } else {
+ tmpU32no1 = inst->prevNoiseU32[i] - tmpU32no2; // Q(prevQNoise+11)
+ }
+ if (noiseUpdateU32 > tmpU32no1) {
+ noiseUpdateU32 = tmpU32no1; // Q(prevQNoise+11)
+ }
+ }
+ noiseU32[i] = noiseUpdateU32; // Q(prevQNoise+11)
+ if (noiseUpdateU32 > maxNoiseU32) {
+ maxNoiseU32 = noiseUpdateU32;
+ }
+
+ // conservative noise update
+ // // original FLOAT code
+ // if (prob_speech < PROB_RANGE) {
+ // inst->avgMagnPause[i] = inst->avgMagnPause[i] + (1.0 - gamma_pause)*(magn[i] - inst->avgMagnPause[i]);
+ // }
+
+ tmp32no2 = WEBRTC_SPL_SHIFT_W32(inst->avgMagnPause[i], -nShifts);
+ if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8) {
+ if (nShifts < 0) {
+ tmp32no1 = (WebRtc_Word32)magnU16[i] - tmp32no2; // Q(qMagn)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts)
+ - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
+ }
+ tmp32no2 += tmp32no1; // Q(qMagn)
+ }
+ inst->avgMagnPause[i] = tmp32no2;
+ } // end of frequency loop
+
+ norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
+ qNoise = inst->prevQNoise + norm32no1 - 5;
+ // done with step 2: noise update
+
+ // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
+ nShifts = inst->prevQNoise + 11 - qMagn;
+ for (i = 0; i < inst->magnLen; i++) {
+ // FLOAT code
+ // // post and prior SNR
+ // curNearSnr = 0.0;
+ // if (magn[i] > noise[i])
+ // {
+ // curNearSnr = magn[i] / (noise[i] + 0.0001) - 1.0;
+ // }
+ // // DD estimate is sum of two terms: current estimate and previous estimate
+ // // directed decision update of snrPrior
+ // snrPrior = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * curNearSnr;
+ // // gain filter
+ // tmpFloat1 = inst->overdrive + snrPrior;
+ // tmpFloat2 = snrPrior / tmpFloat1;
+ // theFilter[i] = tmpFloat2;
+
+ // calculate curNearSnr again, this is necessary because a new noise estimate has been made since then. for the original
+ curNearSnr = 0; // Q11
+ if (nShifts < 0) {
+ // This case is equivalent with magn < noise which implies curNearSnr = 0;
+ tmpMagnU32 = (WebRtc_UWord32)magnU16[i]; // Q(qMagn)
+ tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn)
+ } else if (nShifts > 17) {
+ tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17)
+ tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
+ } else {
+ tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], nShifts); // Q(qNoise_prev+11)
+ tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
+ }
+ if (tmpMagnU32 > tmpNoiseU32) {
+ tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
+ norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
+ tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
+ tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
+ if (tmpU32no2 > 0) {
+ tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
+ }
+ curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
+ }
+
+ //directed decision update of priorSnr
+ // FLOAT
+ // priorSnr = DD_PR_SNR * prevNearSnr + (1.0-DD_PR_SNR) * curNearSnr;
+
+ tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
+ tmpU32no2 = WEBRTC_SPL_UMUL_32_16(curNearSnr, ONE_MINUS_DD_PR_SNR_Q11); // Q22
+ priorSnr = tmpU32no1 + tmpU32no2; // Q22
+
+ //gain filter
+ tmpU32no1 = (WebRtc_UWord32)(inst->overdrive)
+ + WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
+ assert(inst->overdrive > 0);
+ tmpU16no1 = (WebRtc_UWord16)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
+ inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
+
+ // Weight in the parametric Wiener filter during startup
+ if (inst->blockIndex < END_STARTUP_SHORT) {
+ // Weight the two suppression filters
+ tmpU32no1 = WEBRTC_SPL_UMUL_16_16(inst->noiseSupFilter[i],
+ (WebRtc_UWord16)inst->blockIndex);
+ tmpU32no2 = WEBRTC_SPL_UMUL_16_16(noiseSupFilterTmp[i],
+ (WebRtc_UWord16)(END_STARTUP_SHORT
+ - inst->blockIndex));
+ tmpU32no1 += tmpU32no2;
+ inst->noiseSupFilter[i] = (WebRtc_UWord16)WebRtcSpl_DivU32U16(tmpU32no1,
+ END_STARTUP_SHORT);
+ }
+ } // end of loop over frequencies
+ //done with step3
+
+ // save noise and magnitude spectrum for next frame
+ inst->prevQNoise = qNoise;
+ inst->prevQMagn = qMagn;
+ if (norm32no1 > 5) {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11)
+ inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
+ }
+ } else {
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
+ inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
+ }
+ }
+
+ WebRtcNsx_DataSynthesis(inst, outFrame);
+#ifdef NS_FILEDEBUG
+ fwrite(outframe, sizeof(short), inst->blockLen10ms, inst->outfile);
+#endif
+
+ //for H band:
+ // only update data buffer, then apply time-domain gain is applied derived from L band
+ if (inst->fs == 32000) {
+ // update analysis buffer for H band
+ // append new data to buffer FX
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms, inst->anaLen - inst->blockLen10ms);
+ WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms, speechFrameHB, inst->blockLen10ms);
+ // range for averaging low band quantities for H band gain
+
+ gainTimeDomainHB = 16384; // 16384 = Q14(1.0)
+ //average speech prob from low band
+ //average filter gain from low band
+ //avg over second half (i.e., 4->8kHz) of freq. spectrum
+ tmpU32no1 = 0; // Q12
+ tmpU16no1 = 0; // Q8
+ for (i = inst->anaLen2 - (inst->anaLen2 >> 2); i < inst->anaLen2; i++) {
+ tmpU16no1 += nonSpeechProbFinal[i]; // Q8
+ tmpU32no1 += (WebRtc_UWord32)(inst->noiseSupFilter[i]); // Q14
+ }
+ avgProbSpeechHB = (WebRtc_Word16)(4096
+ - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12
+ avgFilterGainHB = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
+ tmpU32no1, inst->stages - 3); // Q14
+
+ // // original FLOAT code
+ // // gain based on speech probability:
+ // avg_prob_speech_tt=(float)2.0*avg_prob_speech-(float)1.0;
+ // gain_mod=(float)0.5*((float)1.0+(float)tanh(avg_prob_speech_tt)); // between 0 and 1
+
+ // gain based on speech probability:
+ // original expression: "0.5 * (1 + tanh(2x-1))"
+ // avgProbSpeechHB has been anyway saturated to a value between 0 and 1 so the other cases don't have to be dealt with
+ // avgProbSpeechHB and gainModHB are in Q12, 3607 = Q12(0.880615234375) which is a zero point of
+ // |0.5 * (1 + tanh(2x-1)) - x| - |0.5 * (1 + tanh(2x-1)) - 0.880615234375| meaning that from that point the error of approximating
+ // the expression with f(x) = x would be greater than the error of approximating the expression with f(x) = 0.880615234375
+ // error: "|0.5 * (1 + tanh(2x-1)) - x| from x=0 to 0.880615234375" -> http://www.wolframalpha.com/input/?i=|0.5+*+(1+%2B+tanh(2x-1))+-+x|+from+x%3D0+to+0.880615234375
+ // and: "|0.5 * (1 + tanh(2x-1)) - 0.880615234375| from x=0.880615234375 to 1" -> http://www.wolframalpha.com/input/?i=+|0.5+*+(1+%2B+tanh(2x-1))+-+0.880615234375|+from+x%3D0.880615234375+to+1
+ gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607);
+
+ // // original FLOAT code
+ // //combine gain with low band gain
+ // if (avg_prob_speech < (float)0.5) {
+ // gain_time_domain_HB=(float)0.5*gain_mod+(float)0.5*avg_filter_gain;
+ // }
+ // else {
+ // gain_time_domain_HB=(float)0.25*gain_mod+(float)0.75*avg_filter_gain;
+ // }
+
+
+ //combine gain with low band gain
+ if (avgProbSpeechHB < 2048) {
+ // 2048 = Q12(0.5)
+ // the next two lines in float are "gain_time_domain = 0.5 * gain_mod + 0.5 * avg_filter_gain"; Q2(0.5) = 2 equals one left shift
+ gainTimeDomainHB = (gainModHB << 1) + (avgFilterGainHB >> 1); // Q14
+ } else {
+ // "gain_time_domain = 0.25 * gain_mod + 0.75 * agv_filter_gain;"
+ gainTimeDomainHB = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, avgFilterGainHB, 2); // 3 = Q2(0.75); Q14
+ gainTimeDomainHB += gainModHB; // Q14
+ }
+ //make sure gain is within flooring range
+ gainTimeDomainHB
+ = WEBRTC_SPL_SAT(16384, gainTimeDomainHB, (WebRtc_Word16)(inst->denoiseBound)); // 16384 = Q14(1.0)
+
+
+ //apply gain
+ for (i = 0; i < inst->blockLen10ms; i++) {
+ outFrameHB[i]
+ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
+ }
+ } // end of H band gain computation
+
+ return 0;
+}
+
+
diff --git a/src/modules/audio_processing/ns/nsx_core.h b/src/modules/audio_processing/ns/nsx_core.h
new file mode 100644
index 0000000..0a0faf9
--- /dev/null
+++ b/src/modules/audio_processing/ns/nsx_core.h
@@ -0,0 +1,222 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
+
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+#include "nsx_defines.h"
+
+#ifdef NS_FILEDEBUG
+#include <stdio.h>
+#endif
+
+typedef struct NsxInst_t_ {
+ WebRtc_UWord32 fs;
+
+ const WebRtc_Word16* window;
+ WebRtc_Word16 analysisBuffer[ANAL_BLOCKL_MAX];
+ WebRtc_Word16 synthesisBuffer[ANAL_BLOCKL_MAX];
+ WebRtc_UWord16 noiseSupFilter[HALF_ANAL_BLOCKL];
+ WebRtc_UWord16 overdrive; /* Q8 */
+ WebRtc_UWord16 denoiseBound; /* Q14 */
+ const WebRtc_Word16* factor2Table;
+ WebRtc_Word16 noiseEstLogQuantile[SIMULT* HALF_ANAL_BLOCKL];
+ WebRtc_Word16 noiseEstDensity[SIMULT* HALF_ANAL_BLOCKL];
+ WebRtc_Word16 noiseEstCounter[SIMULT];
+ WebRtc_Word16 noiseEstQuantile[HALF_ANAL_BLOCKL];
+
+ WebRtc_Word16 anaLen;
+ int anaLen2;
+ int magnLen;
+ int aggrMode;
+ int stages;
+ int initFlag;
+ int gainMap;
+
+ WebRtc_Word32 maxLrt;
+ WebRtc_Word32 minLrt;
+ WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
+ WebRtc_Word32 featureLogLrt;
+ WebRtc_Word32 thresholdLogLrt;
+ WebRtc_Word16 weightLogLrt;
+
+ WebRtc_UWord32 featureSpecDiff;
+ WebRtc_UWord32 thresholdSpecDiff;
+ WebRtc_Word16 weightSpecDiff;
+
+ WebRtc_UWord32 featureSpecFlat;
+ WebRtc_UWord32 thresholdSpecFlat;
+ WebRtc_Word16 weightSpecFlat;
+
+ WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
+ WebRtc_UWord32 magnEnergy;
+ WebRtc_UWord32 sumMagn;
+ WebRtc_UWord32 curAvgMagnEnergy;
+ WebRtc_UWord32 timeAvgMagnEnergy;
+ WebRtc_UWord32 timeAvgMagnEnergyTmp;
+
+ WebRtc_UWord32 whiteNoiseLevel; //initial noise estimate
+ WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
+ WebRtc_Word32 pinkNoiseNumerator; //pink noise parameter: numerator
+ WebRtc_Word32 pinkNoiseExp; //pink noise parameter: power of freq
+ int minNorm; //smallest normalization factor
+ int zeroInputSignal; //zero input signal flag
+
+ WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
+ WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
+ WebRtc_Word16 priorNonSpeechProb; //prior speech/noise probability // Q14
+
+ int blockIndex; //frame index counter
+ int modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
+ int cntThresUpdate;
+
+ //histograms for parameter estimation
+ WebRtc_Word16 histLrt[HIST_PAR_EST];
+ WebRtc_Word16 histSpecFlat[HIST_PAR_EST];
+ WebRtc_Word16 histSpecDiff[HIST_PAR_EST];
+
+ //quantities for high band estimate
+ WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
+
+ int qNoise;
+ int prevQNoise;
+ int prevQMagn;
+ int blockLen10ms;
+
+ WebRtc_Word16 real[ANAL_BLOCKL_MAX];
+ WebRtc_Word16 imag[ANAL_BLOCKL_MAX];
+ WebRtc_Word32 energyIn;
+ int scaleEnergyIn;
+ int normData;
+
+} NsxInst_t;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/****************************************************************************
+ * WebRtcNsx_InitCore(...)
+ *
+ * This function initializes a noise suppression instance
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - fs : Sampling frequency
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs);
+
+/****************************************************************************
+ * WebRtcNsx_set_policy_core(...)
+ *
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
+ *
+ * Output:
+ * - inst : Initialized instance
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode);
+
+/****************************************************************************
+ * WebRtcNsx_ProcessCore
+ *
+ * Do noise suppression.
+ *
+ * Input:
+ * - inst : Instance that should be initialized
+ * - inFrameLow : Input speech frame for lower band
+ * - inFrameHigh : Input speech frame for higher band
+ *
+ * Output:
+ * - inst : Updated instance
+ * - outFrameLow : Output speech frame for lower band
+ * - outFrameHigh : Output speech frame for higher band
+ *
+ * Return value : 0 - OK
+ * -1 - Error
+ */
+int WebRtcNsx_ProcessCore(NsxInst_t* inst,
+ short* inFrameLow,
+ short* inFrameHigh,
+ short* outFrameLow,
+ short* outFrameHigh);
+
+/****************************************************************************
+ * Some function pointers, for internal functions shared by ARM NEON and
+ * generic C code.
+ */
+// Noise Estimation.
+typedef void (*NoiseEstimation)(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise);
+extern NoiseEstimation WebRtcNsx_NoiseEstimation;
+
+// Filter the data in the frequency domain, and create spectrum.
+typedef void (*PrepareSpectrum)(NsxInst_t* inst,
+ int16_t* freq_buff);
+extern PrepareSpectrum WebRtcNsx_PrepareSpectrum;
+
+// For the noise supression process, synthesis, read out fully processed
+// segment, and update synthesis buffer.
+typedef void (*SynthesisUpdate)(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor);
+extern SynthesisUpdate WebRtcNsx_SynthesisUpdate;
+
+// Update analysis buffer for lower band, and window data before FFT.
+typedef void (*AnalysisUpdate)(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech);
+extern AnalysisUpdate WebRtcNsx_AnalysisUpdate;
+
+// Denormalize the input buffer.
+typedef void (*Denormalize)(NsxInst_t* inst,
+ int16_t* in,
+ int factor);
+extern Denormalize WebRtcNsx_Denormalize;
+
+// Create a complex number buffer, as the intput interleaved with zeros,
+// and normalize it.
+typedef void (*CreateComplexBuffer)(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out);
+extern CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
+
+/****************************************************************************
+ * Initialization of the above function pointers for ARM Neon.
+ */
+void WebRtcNsx_InitNeon(void);
+
+extern const WebRtc_Word16 WebRtcNsx_kLogTable[9];
+extern const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256];
+extern const WebRtc_Word16 WebRtcNsx_kCounterDiv[201];
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
diff --git a/src/modules/audio_processing/ns/nsx_core_neon.c b/src/modules/audio_processing/ns/nsx_core_neon.c
new file mode 100644
index 0000000..2f85abd
--- /dev/null
+++ b/src/modules/audio_processing/ns/nsx_core_neon.c
@@ -0,0 +1,734 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "nsx_core.h"
+
+#include <arm_neon.h>
+#include <assert.h>
+
+// Update the noise estimation information.
+static void UpdateNoiseEstimateNeon(NsxInst_t* inst, int offset) {
+ int i = 0;
+ const int16_t kExp2Const = 11819; // Q13
+ int16_t* ptr_noiseEstLogQuantile = NULL;
+ int16_t* ptr_noiseEstQuantile = NULL;
+ int16x4_t kExp2Const16x4 = vdup_n_s16(kExp2Const);
+ int32x4_t twentyOne32x4 = vdupq_n_s32(21);
+ int32x4_t constA32x4 = vdupq_n_s32(0x1fffff);
+ int32x4_t constB32x4 = vdupq_n_s32(0x200000);
+
+ int16_t tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
+ inst->magnLen);
+
+ // Guarantee a Q-domain as high as possible and still fit in int16
+ inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2Const,
+ tmp16,
+ 21);
+
+ int32x4_t qNoise32x4 = vdupq_n_s32(inst->qNoise);
+
+ for (ptr_noiseEstLogQuantile = &inst->noiseEstLogQuantile[offset],
+ ptr_noiseEstQuantile = &inst->noiseEstQuantile[0];
+ ptr_noiseEstQuantile < &inst->noiseEstQuantile[inst->magnLen - 3];
+ ptr_noiseEstQuantile += 4, ptr_noiseEstLogQuantile += 4) {
+
+ // tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ // inst->noiseEstLogQuantile[offset + i]);
+ int16x4_t v16x4 = vld1_s16(ptr_noiseEstLogQuantile);
+ int32x4_t v32x4B = vmull_s16(v16x4, kExp2Const16x4);
+
+ // tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+ int32x4_t v32x4A = vandq_s32(v32x4B, constA32x4);
+ v32x4A = vorrq_s32(v32x4A, constB32x4);
+
+ // tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ v32x4B = vshrq_n_s32(v32x4B, 21);
+
+ // tmp16 -= 21;// shift 21 to get result in Q0
+ v32x4B = vsubq_s32(v32x4B, twentyOne32x4);
+
+ // tmp16 += (int16_t) inst->qNoise;
+ // shift to get result in Q(qNoise)
+ v32x4B = vaddq_s32(v32x4B, qNoise32x4);
+
+ // if (tmp16 < 0) {
+ // tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ // } else {
+ // tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ // }
+ v32x4B = vshlq_s32(v32x4A, v32x4B);
+
+ // tmp16 = WebRtcSpl_SatW32ToW16(tmp32no1);
+ v16x4 = vqmovn_s32(v32x4B);
+
+ //inst->noiseEstQuantile[i] = tmp16;
+ vst1_s16(ptr_noiseEstQuantile, v16x4);
+ }
+
+ // Last iteration:
+
+ // inst->quantile[i]=exp(inst->lquantile[offset+i]);
+ // in Q21
+ int32_t tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
+ *ptr_noiseEstLogQuantile);
+ int32_t tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
+
+ tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
+ tmp16 -= 21;// shift 21 to get result in Q0
+ tmp16 += (int16_t) inst->qNoise; //shift to get result in Q(qNoise)
+ if (tmp16 < 0) {
+ tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
+ } else {
+ tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+ }
+ *ptr_noiseEstQuantile = WebRtcSpl_SatW32ToW16(tmp32no1);
+}
+
+// Noise Estimation
+static void NoiseEstimationNeon(NsxInst_t* inst,
+ uint16_t* magn,
+ uint32_t* noise,
+ int16_t* q_noise) {
+ int16_t lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
+ int16_t countProd, delta, zeros, frac;
+ int16_t log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
+ const int16_t log2_const = 22713;
+ const int16_t width_factor = 21845;
+
+ int i, s, offset;
+
+ tabind = inst->stages - inst->normData;
+ assert(tabind < 9);
+ assert(tabind > -9);
+ if (tabind < 0) {
+ logval = -WebRtcNsx_kLogTable[-tabind];
+ } else {
+ logval = WebRtcNsx_kLogTable[tabind];
+ }
+
+ int16x8_t logval_16x8 = vdupq_n_s16(logval);
+
+ // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
+ // magn is in Q(-stages), and the real lmagn values are:
+ // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
+ // lmagn in Q8
+ for (i = 0; i < inst->magnLen; i++) {
+ if (magn[i]) {
+ zeros = WebRtcSpl_NormU32((uint32_t)magn[i]);
+ frac = (int16_t)((((uint32_t)magn[i] << zeros)
+ & 0x7FFFFFFF) >> 23);
+ assert(frac < 256);
+ // log2(magn(i))
+ log2 = (int16_t)(((31 - zeros) << 8)
+ + WebRtcNsx_kLogTableFrac[frac]);
+ // log2(magn(i))*log(2)
+ lmagn[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
+ // + log(2^stages)
+ lmagn[i] += logval;
+ } else {
+ lmagn[i] = logval;
+ }
+ }
+
+ int16x4_t Q3_16x4 = vdup_n_s16(3);
+ int16x8_t WIDTHQ8_16x8 = vdupq_n_s16(WIDTH_Q8);
+ int16x8_t WIDTHFACTOR_16x8 = vdupq_n_s16(width_factor);
+
+ int16_t factor = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG)
+ factor = FACTOR_Q7_STARTUP;
+
+ // Loop over simultaneous estimates
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * inst->magnLen;
+
+ // Get counter values from state
+ counter = inst->noiseEstCounter[s];
+ assert(counter < 201);
+ countDiv = WebRtcNsx_kCounterDiv[counter];
+ countProd = (int16_t)WEBRTC_SPL_MUL_16_16(counter, countDiv);
+
+ // quant_est(...)
+ int16_t deltaBuff[8];
+ int16x4_t tmp16x4_0;
+ int16x4_t tmp16x4_1;
+ int16x4_t countDiv_16x4 = vdup_n_s16(countDiv);
+ int16x8_t countProd_16x8 = vdupq_n_s16(countProd);
+ int16x8_t tmp16x8_0 = vdupq_n_s16(countDiv);
+ int16x8_t prod16x8 = vqrdmulhq_s16(WIDTHFACTOR_16x8, tmp16x8_0);
+ int16x8_t tmp16x8_1;
+ int16x8_t tmp16x8_2;
+ int16x8_t tmp16x8_3;
+ int16x8_t tmp16x8_4;
+ int16x8_t tmp16x8_5;
+ int32x4_t tmp32x4;
+
+ for (i = 0; i < inst->magnLen - 7; i += 8) {
+ // Compute delta.
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ tmp16x8_0 = vdupq_n_s16(factor);
+ vst1q_s16(deltaBuff, tmp16x8_0);
+
+ int j;
+ for (j = 0; j < 8; j++) {
+ if (inst->noiseEstDensity[offset + i + j] > 512) {
+ // Get values for deltaBuff by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i + j]);
+ deltaBuff[j] = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ }
+ }
+
+ // Update log quantile estimate
+
+ // tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[0]), countDiv_16x4);
+ tmp16x4_1 = vshrn_n_s32(tmp32x4, 14);
+ tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[4]), countDiv_16x4);
+ tmp16x4_0 = vshrn_n_s32(tmp32x4, 14);
+ tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // Keep for several lines.
+
+ // prepare for the "if" branch
+ // tmp16 += 2;
+ // tmp16_1 = (Word16)(tmp16>>2);
+ tmp16x8_1 = vrshrq_n_s16(tmp16x8_0, 2);
+
+ // inst->noiseEstLogQuantile[offset+i] + tmp16_1;
+ tmp16x8_2 = vld1q_s16(&inst->noiseEstLogQuantile[offset + i]); // Keep
+ tmp16x8_1 = vaddq_s16(tmp16x8_2, tmp16x8_1); // Keep for several lines
+
+ // Prepare for the "else" branch
+ // tmp16 += 1;
+ // tmp16_1 = (Word16)(tmp16>>1);
+ tmp16x8_0 = vrshrq_n_s16(tmp16x8_0, 1);
+
+ // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
+ tmp32x4 = vmull_s16(vget_low_s16(tmp16x8_0), Q3_16x4);
+ tmp16x4_1 = vshrn_n_s32(tmp32x4, 1);
+
+ // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
+ tmp32x4 = vmull_s16(vget_high_s16(tmp16x8_0), Q3_16x4);
+ tmp16x4_0 = vshrn_n_s32(tmp32x4, 1);
+
+ // inst->noiseEstLogQuantile[offset + i] - tmp16_2;
+ tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // keep
+ tmp16x8_0 = vsubq_s16(tmp16x8_2, tmp16x8_0);
+
+ // logval is the smallest fixed point representation we can have. Values
+ // below that will correspond to values in the interval [0, 1], which
+ // can't possibly occur.
+ tmp16x8_0 = vmaxq_s16(tmp16x8_0, logval_16x8);
+
+ // Do the if-else branches:
+ tmp16x8_3 = vld1q_s16(&lmagn[i]); // keep for several lines
+ tmp16x8_5 = vsubq_s16(tmp16x8_3, tmp16x8_2);
+ __asm__("vcgt.s16 %q0, %q1, #0"::"w"(tmp16x8_4), "w"(tmp16x8_5));
+ __asm__("vbit %q0, %q1, %q2"::
+ "w"(tmp16x8_2), "w"(tmp16x8_1), "w"(tmp16x8_4));
+ __asm__("vbif %q0, %q1, %q2"::
+ "w"(tmp16x8_2), "w"(tmp16x8_0), "w"(tmp16x8_4));
+ vst1q_s16(&inst->noiseEstLogQuantile[offset + i], tmp16x8_2);
+
+ // Update density estimate
+ // tmp16_1 + tmp16_2
+ tmp16x8_1 = vld1q_s16(&inst->noiseEstDensity[offset + i]);
+ tmp16x8_0 = vqrdmulhq_s16(tmp16x8_1, countProd_16x8);
+ tmp16x8_0 = vaddq_s16(tmp16x8_0, prod16x8);
+
+ // lmagn[i] - inst->noiseEstLogQuantile[offset + i]
+ tmp16x8_3 = vsubq_s16(tmp16x8_3, tmp16x8_2);
+ tmp16x8_3 = vabsq_s16(tmp16x8_3);
+ tmp16x8_4 = vcgtq_s16(WIDTHQ8_16x8, tmp16x8_3);
+ __asm__("vbit %q0, %q1, %q2"::
+ "w"(tmp16x8_1), "w"(tmp16x8_0), "w"(tmp16x8_4));
+ vst1q_s16(&inst->noiseEstDensity[offset + i], tmp16x8_1);
+ } // End loop over magnitude spectrum
+
+ // Last iteration over magnitude spectrum:
+ // compute delta
+ if (inst->noiseEstDensity[offset + i] > 512) {
+ // Get values for deltaBuff by shifting intead of dividing.
+ int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
+ delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
+ } else {
+ delta = FACTOR_Q7;
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ // Smaller step size during startup. This prevents from using
+ // unrealistic values causing overflow.
+ delta = FACTOR_Q7_STARTUP;
+ }
+ }
+ // update log quantile estimate
+ tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
+ if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
+ // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
+ // CounterDiv=1/(inst->counter[s]+1) in Q15
+ tmp16 += 2;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
+ inst->noiseEstLogQuantile[offset + i] += tmp16no1;
+ } else {
+ tmp16 += 1;
+ tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
+ // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
+ tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
+ inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
+ if (inst->noiseEstLogQuantile[offset + i] < logval) {
+ // logval is the smallest fixed point representation we can have.
+ // Values below that will correspond to values in the interval
+ // [0, 1], which can't possibly occur.
+ inst->noiseEstLogQuantile[offset + i] = logval;
+ }
+ }
+
+ // update density estimate
+ if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
+ < WIDTH_Q8) {
+ tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ inst->noiseEstDensity[offset + i], countProd, 15);
+ tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ width_factor, countDiv, 15);
+ inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
+ }
+
+
+ if (counter >= END_STARTUP_LONG) {
+ inst->noiseEstCounter[s] = 0;
+ if (inst->blockIndex >= END_STARTUP_LONG) {
+ UpdateNoiseEstimateNeon(inst, offset);
+ }
+ }
+ inst->noiseEstCounter[s]++;
+
+ } // end loop over simultaneous estimates
+
+ // Sequentially update the noise during startup
+ if (inst->blockIndex < END_STARTUP_LONG) {
+ UpdateNoiseEstimateNeon(inst, offset);
+ }
+
+ for (i = 0; i < inst->magnLen; i++) {
+ noise[i] = (uint32_t)(inst->noiseEstQuantile[i]); // Q(qNoise)
+ }
+ (*q_noise) = (int16_t)inst->qNoise;
+}
+
+// Filter the data in the frequency domain, and create spectrum.
+static void PrepareSpectrumNeon(NsxInst_t* inst, int16_t* freq_buf) {
+
+ // (1) Filtering.
+
+ // Fixed point C code for the next block is as follows:
+ // for (i = 0; i < inst->magnLen; i++) {
+ // inst->real[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
+ // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ // inst->imag[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
+ // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
+ // }
+
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_imag = &inst->imag[0];
+ uint16_t* ptr_noiseSupFilter = &inst->noiseSupFilter[0];
+
+ // Filter the rest in the frequency domain.
+ for (; ptr_real < &inst->real[inst->magnLen - 1];) {
+ // Loop unrolled once. Both pointers are incremented by 4 twice.
+ __asm__ __volatile__(
+ "vld1.16 d20, [%[ptr_real]]\n\t"
+ "vld1.16 d22, [%[ptr_imag]]\n\t"
+ "vld1.16 d23, [%[ptr_noiseSupFilter]]!\n\t"
+ "vmull.s16 q10, d20, d23\n\t"
+ "vmull.s16 q11, d22, d23\n\t"
+ "vshrn.s32 d20, q10, #14\n\t"
+ "vshrn.s32 d22, q11, #14\n\t"
+ "vst1.16 d20, [%[ptr_real]]!\n\t"
+ "vst1.16 d22, [%[ptr_imag]]!\n\t"
+
+ "vld1.16 d18, [%[ptr_real]]\n\t"
+ "vld1.16 d24, [%[ptr_imag]]\n\t"
+ "vld1.16 d25, [%[ptr_noiseSupFilter]]!\n\t"
+ "vmull.s16 q9, d18, d25\n\t"
+ "vmull.s16 q12, d24, d25\n\t"
+ "vshrn.s32 d18, q9, #14\n\t"
+ "vshrn.s32 d24, q12, #14\n\t"
+ "vst1.16 d18, [%[ptr_real]]!\n\t"
+ "vst1.16 d24, [%[ptr_imag]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_imag]"+r"(ptr_imag),
+ [ptr_real]"+r"(ptr_real),
+ [ptr_noiseSupFilter]"+r"(ptr_noiseSupFilter)
+ :
+ :"d18", "d19", "d20", "d21", "d22", "d23", "d24", "d25",
+ "q9", "q10", "q11", "q12"
+ );
+ }
+
+ // Filter the last pair of elements in the frequency domain.
+ *ptr_real = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_real,
+ (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
+ *ptr_imag = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_imag,
+ (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
+
+ // (2) Create spectrum.
+
+ // Fixed point C code for the rest of the function is as follows:
+ // freq_buf[0] = inst->real[0];
+ // freq_buf[1] = -inst->imag[0];
+ // for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
+ // tmp16 = (inst->anaLen << 1) - j;
+ // freq_buf[j] = inst->real[i];
+ // freq_buf[j + 1] = -inst->imag[i];
+ // freq_buf[tmp16] = inst->real[i];
+ // freq_buf[tmp16 + 1] = inst->imag[i];
+ // }
+ // freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ // freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+
+ freq_buf[0] = inst->real[0];
+ freq_buf[1] = -inst->imag[0];
+
+ int offset = -16;
+ int16_t* ptr_realImag1 = &freq_buf[2];
+ int16_t* ptr_realImag2 = ptr_realImag2 = &freq_buf[(inst->anaLen << 1) - 8];
+ ptr_real = &inst->real[1];
+ ptr_imag = &inst->imag[1];
+ for (; ptr_real < &inst->real[inst->anaLen2 - 11];) {
+ // Loop unrolled once. All pointers are incremented twice.
+ __asm__ __volatile__(
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_imag]]!\n\t"
+ // Negate and interleave:
+ "vmov.s16 d20, d22\n\t"
+ "vneg.s16 d21, d23\n\t"
+ "vzip.16 d20, d21\n\t"
+ // Write 8 elements to &freq_buf[j]
+ "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
+ // Interleave and reverse elements:
+ "vzip.16 d22, d23\n\t"
+ "vrev64.32 d18, d23\n\t"
+ "vrev64.32 d19, d22\n\t"
+ // Write 8 elements to &freq_buf[tmp16]
+ "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
+
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_imag]]!\n\t"
+ // Negate and interleave:
+ "vmov.s16 d20, d22\n\t"
+ "vneg.s16 d21, d23\n\t"
+ "vzip.16 d20, d21\n\t"
+ // Write 8 elements to &freq_buf[j]
+ "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
+ // Interleave and reverse elements:
+ "vzip.16 d22, d23\n\t"
+ "vrev64.32 d18, d23\n\t"
+ "vrev64.32 d19, d22\n\t"
+ // Write 8 elements to &freq_buf[tmp16]
+ "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
+
+ // Specify constraints.
+ :[ptr_imag]"+r"(ptr_imag),
+ [ptr_real]"+r"(ptr_real),
+ [ptr_realImag1]"+r"(ptr_realImag1),
+ [ptr_realImag2]"+r"(ptr_realImag2)
+ :[offset]"r"(offset)
+ :"d18", "d19", "d20", "d21", "d22", "d23"
+ );
+ }
+ for (ptr_realImag2 += 6;
+ ptr_real <= &inst->real[inst->anaLen2];
+ ptr_real += 1, ptr_imag += 1, ptr_realImag1 += 2, ptr_realImag2 -= 2) {
+ *ptr_realImag1 = *ptr_real;
+ *(ptr_realImag1 + 1) = -(*ptr_imag);
+ *ptr_realImag2 = *ptr_real;
+ *(ptr_realImag2 + 1) = *ptr_imag;
+ }
+
+ freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
+ freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
+}
+
+// Denormalize the input buffer.
+static __inline void DenormalizeNeon(NsxInst_t* inst, int16_t* in, int factor) {
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_in = &in[0];
+
+ __asm__ __volatile__("vdup.32 q10, %0" ::
+ "r"((int32_t)(factor - inst->normData)) : "q10");
+ for (; ptr_real < &inst->real[inst->anaLen];) {
+
+ // Loop unrolled once. Both pointers are incremented.
+ __asm__ __volatile__(
+ // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
+ // factor - inst->normData);
+ "vld2.16 {d24, d25}, [%[ptr_in]]!\n\t"
+ "vmovl.s16 q12, d24\n\t"
+ "vshl.s32 q12, q10\n\t"
+ // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqmovn.s32 d24, q12\n\t"
+ "vst1.16 d24, [%[ptr_real]]!\n\t"
+
+ // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
+ // factor - inst->normData);
+ "vld2.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vmovl.s16 q11, d22\n\t"
+ "vshl.s32 q11, q10\n\t"
+ // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqmovn.s32 d22, q11\n\t"
+ "vst1.16 d22, [%[ptr_real]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_in]"+r"(ptr_in),
+ [ptr_real]"+r"(ptr_real)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+}
+
+// For the noise supress process, synthesis, read out fully processed segment,
+// and update synthesis buffer.
+static void SynthesisUpdateNeon(NsxInst_t* inst,
+ int16_t* out_frame,
+ int16_t gain_factor) {
+ int16_t* ptr_real = &inst->real[0];
+ int16_t* ptr_syn = &inst->synthesisBuffer[0];
+ int16_t* ptr_window = &inst->window[0];
+
+ // synthesis
+ __asm__ __volatile__("vdup.16 d24, %0" : : "r"(gain_factor) : "d24");
+ // Loop unrolled once. All pointers are incremented in the assembly code.
+ for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
+ __asm__ __volatile__(
+ // Load variables.
+ "vld1.16 d22, [%[ptr_real]]!\n\t"
+ "vld1.16 d23, [%[ptr_window]]!\n\t"
+ "vld1.16 d25, [%[ptr_syn]]\n\t"
+ // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ "vmull.s16 q11, d22, d23\n\t"
+ "vrshrn.i32 d22, q11, #14\n\t"
+ // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
+ "vmull.s16 q11, d24, d22\n\t"
+ // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqrshrn.s32 d22, q11, #13\n\t"
+ // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
+ // inst->synthesisBuffer[i], tmp16b); // Q0
+ "vqadd.s16 d25, d22\n\t"
+ "vst1.16 d25, [%[ptr_syn]]!\n\t"
+
+ // Load variables.
+ "vld1.16 d26, [%[ptr_real]]!\n\t"
+ "vld1.16 d27, [%[ptr_window]]!\n\t"
+ "vld1.16 d28, [%[ptr_syn]]\n\t"
+ // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->real[i], 14); // Q0, window in Q14
+ "vmull.s16 q13, d26, d27\n\t"
+ "vrshrn.i32 d26, q13, #14\n\t"
+ // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
+ "vmull.s16 q13, d24, d26\n\t"
+ // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
+ "vqrshrn.s32 d26, q13, #13\n\t"
+ // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
+ // inst->synthesisBuffer[i], tmp16b); // Q0
+ "vqadd.s16 d28, d26\n\t"
+ "vst1.16 d28, [%[ptr_syn]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_real]"+r"(ptr_real),
+ [ptr_window]"+r"(ptr_window),
+ [ptr_syn]"+r"(ptr_syn)
+ :
+ :"d22", "d23", "d24", "d25", "d26", "d27", "d28", "q11", "q12", "q13"
+ );
+ }
+
+ int16_t* ptr_out = &out_frame[0];
+ ptr_syn = &inst->synthesisBuffer[0];
+ // read out fully processed segment
+ for (; ptr_syn < &inst->synthesisBuffer[inst->blockLen10ms];) {
+ // Loop unrolled once. Both pointers are incremented in the assembly code.
+ __asm__ __volatile__(
+ // out_frame[i] = inst->synthesisBuffer[i]; // Q0
+ "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
+ "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
+ :[ptr_syn]"+r"(ptr_syn),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+
+ // Update synthesis buffer.
+ // C code:
+ // WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
+ // inst->synthesisBuffer + inst->blockLen10ms,
+ // inst->anaLen - inst->blockLen10ms);
+ ptr_out = &inst->synthesisBuffer[0],
+ ptr_syn = &inst->synthesisBuffer[inst->blockLen10ms];
+ for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
+ // Loop unrolled once. Both pointers are incremented in the assembly code.
+ __asm__ __volatile__(
+ "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
+ "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
+ :[ptr_syn]"+r"(ptr_syn),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25"
+ );
+ }
+
+ // C code:
+ // WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
+ // + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
+ __asm__ __volatile__("vdup.16 q10, %0" : : "r"(0) : "q10");
+ for (; ptr_out < &inst->synthesisBuffer[inst->anaLen];) {
+ // Loop unrolled once. Pointer is incremented in the assembly code.
+ __asm__ __volatile__(
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ :[ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21"
+ );
+ }
+}
+
+// Update analysis buffer for lower band, and window data before FFT.
+static void AnalysisUpdateNeon(NsxInst_t* inst,
+ int16_t* out,
+ int16_t* new_speech) {
+
+ int16_t* ptr_ana = &inst->analysisBuffer[inst->blockLen10ms];
+ int16_t* ptr_out = &inst->analysisBuffer[0];
+
+ // For lower band update analysis buffer.
+ // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
+ // inst->analysisBuffer + inst->blockLen10ms,
+ // inst->anaLen - inst->blockLen10ms);
+ for (; ptr_out < &inst->analysisBuffer[inst->anaLen - inst->blockLen10ms];) {
+ // Loop unrolled once, so both pointers are incremented by 8 twice.
+ __asm__ __volatile__(
+ "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23"
+ );
+ }
+
+ // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
+ // + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+ for (ptr_ana = new_speech; ptr_out < &inst->analysisBuffer[inst->anaLen];) {
+ // Loop unrolled once, so both pointers are incremented by 8 twice.
+ __asm__ __volatile__(
+ "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
+ "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
+ "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23"
+ );
+ }
+
+ // Window data before FFT
+ int16_t* ptr_window = &inst->window[0];
+ ptr_out = &out[0];
+ ptr_ana = &inst->analysisBuffer[0];
+ for (; ptr_out < &out[inst->anaLen];) {
+
+ // Loop unrolled once, so all pointers are incremented by 4 twice.
+ __asm__ __volatile__(
+ "vld1.16 d20, [%[ptr_ana]]!\n\t"
+ "vld1.16 d21, [%[ptr_window]]!\n\t"
+ // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ "vmull.s16 q10, d20, d21\n\t"
+ "vrshrn.i32 d20, q10, #14\n\t"
+ "vst1.16 d20, [%[ptr_out]]!\n\t"
+
+ "vld1.16 d22, [%[ptr_ana]]!\n\t"
+ "vld1.16 d23, [%[ptr_window]]!\n\t"
+ // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+ // inst->window[i], inst->analysisBuffer[i], 14); // Q0
+ "vmull.s16 q11, d22, d23\n\t"
+ "vrshrn.i32 d22, q11, #14\n\t"
+ "vst1.16 d22, [%[ptr_out]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_ana]"+r"(ptr_ana),
+ [ptr_window]"+r"(ptr_window),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d20", "d21", "d22", "d23", "q10", "q11"
+ );
+ }
+}
+
+// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
+// zeros, and normalize it.
+static __inline void CreateComplexBufferNeon(NsxInst_t* inst,
+ int16_t* in,
+ int16_t* out) {
+ int16_t* ptr_out = &out[0];
+ int16_t* ptr_in = &in[0];
+
+ __asm__ __volatile__("vdup.16 d25, %0" : : "r"(0) : "d25");
+ __asm__ __volatile__("vdup.16 q10, %0" : : "r"(inst->normData) : "q10");
+ for (; ptr_in < &in[inst->anaLen];) {
+
+ // Loop unrolled once, so ptr_in is incremented by 8 twice,
+ // and ptr_out is incremented by 8 four times.
+ __asm__ __volatile__(
+ // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vshl.s16 q11, q10\n\t"
+ "vmov d24, d23\n\t"
+
+ // out[j + 1] = 0; // Insert zeros in imaginary part
+ "vmov d23, d25\n\t"
+ "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
+
+ // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
+ "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
+ "vshl.s16 q11, q10\n\t"
+ "vmov d24, d23\n\t"
+
+ // out[j + 1] = 0; // Insert zeros in imaginary part
+ "vmov d23, d25\n\t"
+ "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
+ "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
+
+ // Specify constraints.
+ :[ptr_in]"+r"(ptr_in),
+ [ptr_out]"+r"(ptr_out)
+ :
+ :"d22", "d23", "d24", "d25", "q10", "q11"
+ );
+ }
+}
+
+void WebRtcNsx_InitNeon(void) {
+ WebRtcNsx_NoiseEstimation = NoiseEstimationNeon;
+ WebRtcNsx_PrepareSpectrum = PrepareSpectrumNeon;
+ WebRtcNsx_SynthesisUpdate = SynthesisUpdateNeon;
+ WebRtcNsx_AnalysisUpdate = AnalysisUpdateNeon;
+ WebRtcNsx_Denormalize = DenormalizeNeon;
+ WebRtcNsx_CreateComplexBuffer = CreateComplexBufferNeon;
+}
diff --git a/src/modules/audio_processing/ns/main/source/nsx_defines.h b/src/modules/audio_processing/ns/nsx_defines.h
similarity index 97%
rename from src/modules/audio_processing/ns/main/source/nsx_defines.h
rename to src/modules/audio_processing/ns/nsx_defines.h
index 58796b9..cd1e3bf 100644
--- a/src/modules/audio_processing/ns/main/source/nsx_defines.h
+++ b/src/modules/audio_processing/ns/nsx_defines.h
@@ -18,6 +18,7 @@
#define END_STARTUP_SHORT 50
#define FACTOR_Q16 (WebRtc_Word32)2621440 // 40 in Q16
#define FACTOR_Q7 (WebRtc_Word16)5120 // 40 in Q7
+#define FACTOR_Q7_STARTUP (WebRtc_Word16)1024 // 8 in Q7
#define WIDTH_Q8 3 // 0.01 in Q8 (or 25 )
//PARAMETERS FOR NEW METHOD
#define DD_PR_SNR_Q11 2007 // ~= Q11(0.98) DD update of prior SNR
diff --git a/src/modules/audio_processing/ns/windows_private.h b/src/modules/audio_processing/ns/windows_private.h
new file mode 100644
index 0000000..44c2e84
--- /dev/null
+++ b/src/modules/audio_processing/ns/windows_private.h
@@ -0,0 +1,574 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
+
+// Hanning window for 4ms 16kHz
+static const float kHanning64w128[128] = {
+ 0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
+ 0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
+ 0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
+ 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+ 0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
+ 0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
+ 0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
+ 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+ 0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
+ 0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
+ 0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
+ 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+ 0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
+ 0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
+ 0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
+ 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+ 0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
+ 0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
+ 0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
+ 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+ 0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
+ 0.99969881869620f, 1.00000000000000f,
+ 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
+ 0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
+ 0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
+ 0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
+ 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
+ 0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
+ 0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
+ 0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
+ 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
+ 0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
+ 0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
+ 0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
+ 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
+ 0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
+ 0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
+ 0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
+ 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
+ 0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
+ 0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
+ 0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
+ 0.07356456359967f, 0.04906767432742f, 0.02454122852291f
+};
+
+
+
+// hybrib Hanning & flat window
+static const float kBlocks80w128[128] = {
+ (float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
+ (float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
+ (float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
+ (float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
+ (float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
+ (float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
+ (float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
+ (float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
+ (float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
+ (float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
+ (float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
+ (float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
+ (float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
+ (float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
+ (float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
+ (float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
+ (float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
+ (float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
+ (float)0.09801714, (float)0.06540313, (float)0.03271908
+};
+
+// hybrib Hanning & flat window
+static const float kBlocks160w256[256] = {
+ (float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
+ (float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
+ (float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
+ (float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
+ (float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
+ (float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
+ (float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
+ (float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
+ (float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
+ (float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
+ (float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
+ (float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
+ (float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
+ (float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
+ (float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
+ (float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
+ (float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
+ (float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
+ (float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
+ (float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
+ (float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
+ (float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
+ (float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
+ (float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
+ (float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
+ (float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
+ (float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
+ (float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
+ (float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
+ (float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
+ (float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
+ (float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
+ (float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
+ (float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
+ (float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
+ (float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
+ (float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
+ (float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
+ (float)0.01636173
+};
+
+// hybrib Hanning & flat window: for 20ms
+static const float kBlocks320w512[512] = {
+ (float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
+ (float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
+ (float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
+ (float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
+ (float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
+ (float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
+ (float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
+ (float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
+ (float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
+ (float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
+ (float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
+ (float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
+ (float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
+ (float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
+ (float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
+ (float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
+ (float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
+ (float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
+ (float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
+ (float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
+ (float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
+ (float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
+ (float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
+ (float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
+ (float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
+ (float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
+ (float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
+ (float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
+ (float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
+ (float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
+ (float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
+ (float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
+ (float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
+ (float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
+ (float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
+ (float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
+ (float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
+ (float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
+ (float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
+ (float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
+ (float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
+ (float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
+ (float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
+ (float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
+ (float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
+ (float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
+ (float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
+ (float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
+ (float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
+ (float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
+ (float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
+ (float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
+ (float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
+ (float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
+ (float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
+ (float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
+ (float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
+ (float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
+ (float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
+ (float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
+ (float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
+ (float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
+ (float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
+ (float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
+ (float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
+ (float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
+ (float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
+ (float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
+ (float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
+ (float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
+ (float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
+ (float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
+ (float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
+ (float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
+ (float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
+ (float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
+ (float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
+ (float)0.01636173, (float)0.00818114
+};
+
+
+// Hanning window: for 15ms at 16kHz with symmetric zeros
+static const float kBlocks240w512[512] = {
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
+ (float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
+ (float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
+ (float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
+ (float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
+ (float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
+ (float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
+ (float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
+ (float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
+ (float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
+ (float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
+ (float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
+ (float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
+ (float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
+ (float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
+ (float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
+ (float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
+ (float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
+ (float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
+ (float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
+ (float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
+ (float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
+ (float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
+ (float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
+ (float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
+ (float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
+ (float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
+ (float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
+ (float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
+ (float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
+ (float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
+ (float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
+ (float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
+ (float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
+ (float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
+ (float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
+ (float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
+ (float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
+ (float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
+ (float)0.95501995, (float)0.95694035, (float)0.95881975, (float)0.96065807, (float)0.96245527,
+ (float)0.96421117, (float)0.96592581, (float)0.96759909, (float)0.96923089, (float)0.97082120,
+ (float)0.97236991, (float)0.97387701, (float)0.97534233, (float)0.97676587, (float)0.97814763,
+ (float)0.97948742, (float)0.98078531, (float)0.98204112, (float)0.98325491, (float)0.98442656,
+ (float)0.98555607, (float)0.98664331, (float)0.98768836, (float)0.98869103, (float)0.98965138,
+ (float)0.99056935, (float)0.99144489, (float)0.99227792, (float)0.99306846, (float)0.99381649,
+ (float)0.99452192, (float)0.99518472, (float)0.99580491, (float)0.99638247, (float)0.99691731,
+ (float)0.99740952, (float)0.99785894, (float)0.99826562, (float)0.99862951, (float)0.99895066,
+ (float)0.99922901, (float)0.99946457, (float)0.99965733, (float)0.99980724, (float)0.99991435,
+ (float)0.99997860, (float)1.00000000, (float)0.99997860, (float)0.99991435, (float)0.99980724,
+ (float)0.99965733, (float)0.99946457, (float)0.99922901, (float)0.99895066, (float)0.99862951,
+ (float)0.99826562, (float)0.99785894, (float)0.99740946, (float)0.99691731, (float)0.99638247,
+ (float)0.99580491, (float)0.99518472, (float)0.99452192, (float)0.99381644, (float)0.99306846,
+ (float)0.99227792, (float)0.99144489, (float)0.99056935, (float)0.98965138, (float)0.98869103,
+ (float)0.98768836, (float)0.98664331, (float)0.98555607, (float)0.98442656, (float)0.98325491,
+ (float)0.98204112, (float)0.98078525, (float)0.97948742, (float)0.97814757, (float)0.97676587,
+ (float)0.97534227, (float)0.97387695, (float)0.97236991, (float)0.97082120, (float)0.96923089,
+ (float)0.96759909, (float)0.96592581, (float)0.96421117, (float)0.96245521, (float)0.96065807,
+ (float)0.95881969, (float)0.95694029, (float)0.95501995, (float)0.95305860, (float)0.95105648,
+ (float)0.94901365, (float)0.94693011, (float)0.94480604, (float)0.94264150, (float)0.94043654,
+ (float)0.93819129, (float)0.93590593, (float)0.93358046, (float)0.93121493, (float)0.92880952,
+ (float)0.92636436, (float)0.92387950, (float)0.92135507, (float)0.91879123, (float)0.91618794,
+ (float)0.91354543, (float)0.91086382, (float)0.90814310, (float)0.90538365, (float)0.90258527,
+ (float)0.89974827, (float)0.89687276, (float)0.89395875, (float)0.89100647, (float)0.88801610,
+ (float)0.88498759, (float)0.88192123, (float)0.87881714, (float)0.87567532, (float)0.87249595,
+ (float)0.86927933, (float)0.86602539, (float)0.86273432, (float)0.85940641, (float)0.85604161,
+ (float)0.85264009, (float)0.84920216, (float)0.84572780, (float)0.84221715, (float)0.83867055,
+ (float)0.83508795, (float)0.83146954, (float)0.82781565, (float)0.82412612, (float)0.82040137,
+ (float)0.81664157, (float)0.81284660, (float)0.80901700, (float)0.80515265, (float)0.80125374,
+ (float)0.79732066, (float)0.79335332, (float)0.78935200, (float)0.78531694, (float)0.78124815,
+ (float)0.77714586, (float)0.77301049, (float)0.76884180, (float)0.76464021, (float)0.76040596,
+ (float)0.75613904, (float)0.75183970, (float)0.74750835, (float)0.74314481, (float)0.73874938,
+ (float)0.73432249, (float)0.72986400, (float)0.72537428, (float)0.72085363, (float)0.71630186,
+ (float)0.71171951, (float)0.70710677, (float)0.70246363, (float)0.69779032, (float)0.69308734,
+ (float)0.68835449, (float)0.68359220, (float)0.67880070, (float)0.67398006, (float)0.66913044,
+ (float)0.66425240, (float)0.65934575, (float)0.65441096, (float)0.64944804, (float)0.64445722,
+ (float)0.63943905, (float)0.63439327, (float)0.62932026, (float)0.62422055, (float)0.61909389,
+ (float)0.61394072, (float)0.60876143, (float)0.60355592, (float)0.59832448, (float)0.59306765,
+ (float)0.58778518, (float)0.58247757, (float)0.57714522, (float)0.57178789, (float)0.56640613,
+ (float)0.56100023, (float)0.55557019, (float)0.55011630, (float)0.54463905, (float)0.53913826,
+ (float)0.53361434, (float)0.52806783, (float)0.52249849, (float)0.51690674, (float)0.51129305,
+ (float)0.50565726, (float)0.50000006, (float)0.49432117, (float)0.48862115, (float)0.48290038,
+ (float)0.47715873, (float)0.47139663, (float)0.46561456, (float)0.45981231, (float)0.45399037,
+ (float)0.44814920, (float)0.44228864, (float)0.43640912, (float)0.43051112, (float)0.42459446,
+ (float)0.41865960, (float)0.41270703, (float)0.40673658, (float)0.40074870, (float)0.39474386,
+ (float)0.38872188, (float)0.38268328, (float)0.37662849, (float)0.37055734, (float)0.36447033,
+ (float)0.35836792, (float)0.35224995, (float)0.34611690, (float)0.33996922, (float)0.33380675,
+ (float)0.32763001, (float)0.32143945, (float)0.31523487, (float)0.30901679, (float)0.30278572,
+ (float)0.29654145, (float)0.29028472, (float)0.28401530, (float)0.27773371, (float)0.27144048,
+ (float)0.26513538, (float)0.25881892, (float)0.25249159, (float)0.24615324, (float)0.23980433,
+ (float)0.23344538, (float)0.22707619, (float)0.22069728, (float)0.21430916, (float)0.20791161,
+ (float)0.20150517, (float)0.19509031, (float)0.18866688, (float)0.18223536, (float)0.17579627,
+ (float)0.16934940, (float)0.16289529, (float)0.15643445, (float)0.14996666, (float)0.14349243,
+ (float)0.13701232, (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
+ (float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220, (float)0.07845904,
+ (float)0.07193252, (float)0.06540315, (float)0.05887074, (float)0.05233581, (float)0.04579888,
+ (float)0.03925974, (float)0.03271893, (float)0.02617695, (float)0.01963361, (float)0.01308943,
+ (float)0.00654493, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000
+};
+
+
+// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
+static const float kBlocks480w1024[1024] = {
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
+ (float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
+ (float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
+ (float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
+ (float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
+ (float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
+ (float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
+ (float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
+ (float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
+ (float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
+ (float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
+ (float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
+ (float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
+ (float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
+ (float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
+ (float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
+ (float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
+ (float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
+ (float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
+ (float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
+ (float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
+ (float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
+ (float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
+ (float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
+ (float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
+ (float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
+ (float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
+ (float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
+ (float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
+ (float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
+ (float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
+ (float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
+ (float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
+ (float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
+ (float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
+ (float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
+ (float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
+ (float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
+ (float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
+ (float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
+ (float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
+ (float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
+ (float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
+ (float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
+ (float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
+ (float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
+ (float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
+ (float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
+ (float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
+ (float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
+ (float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
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+ (float)0.10452849, (float)0.10127334, (float)0.09801710, (float)0.09475980, (float)0.09150149,
+ (float)0.08824220, (float)0.08498220, (float)0.08172106, (float)0.07845904, (float)0.07519618,
+ (float)0.07193252, (float)0.06866808, (float)0.06540315, (float)0.06213728, (float)0.05887074,
+ (float)0.05560357, (float)0.05233581, (float)0.04906749, (float)0.04579888, (float)0.04252954,
+ (float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798, (float)0.02617695,
+ (float)0.02290541, (float)0.01963361, (float)0.01636161, (float)0.01308943, (float)0.00981712,
+ (float)0.00654493, (float)0.00327244, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
+ (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000
+};
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
diff --git a/src/modules/audio_processing/main/source/processing_component.cc b/src/modules/audio_processing/processing_component.cc
similarity index 100%
rename from src/modules/audio_processing/main/source/processing_component.cc
rename to src/modules/audio_processing/processing_component.cc
diff --git a/src/modules/audio_processing/main/source/processing_component.h b/src/modules/audio_processing/processing_component.h
similarity index 90%
rename from src/modules/audio_processing/main/source/processing_component.h
rename to src/modules/audio_processing/processing_component.h
index 3d8a02b..3af0c4d 100644
--- a/src/modules/audio_processing/main/source/processing_component.h
+++ b/src/modules/audio_processing/processing_component.h
@@ -18,16 +18,6 @@
namespace webrtc {
class AudioProcessingImpl;
-/*template <class T>
-class ComponentHandle {
- public:
- ComponentHandle();
- virtual ~ComponentHandle();
-
- virtual int Create() = 0;
- virtual T* ptr() const = 0;
-};*/
-
class ProcessingComponent {
public:
explicit ProcessingComponent(const AudioProcessingImpl* apm);
@@ -37,10 +27,11 @@
virtual int Destroy();
virtual int get_version(char* version, int version_len_bytes) const = 0;
+ bool is_component_enabled() const;
+
protected:
virtual int Configure();
int EnableComponent(bool enable);
- bool is_component_enabled() const;
void* handle(int index) const;
int num_handles() const;
diff --git a/src/modules/audio_processing/main/source/splitting_filter.cc b/src/modules/audio_processing/splitting_filter.cc
similarity index 100%
rename from src/modules/audio_processing/main/source/splitting_filter.cc
rename to src/modules/audio_processing/splitting_filter.cc
diff --git a/src/modules/audio_processing/main/source/splitting_filter.h b/src/modules/audio_processing/splitting_filter.h
similarity index 100%
rename from src/modules/audio_processing/main/source/splitting_filter.h
rename to src/modules/audio_processing/splitting_filter.h
diff --git a/src/modules/audio_processing/main/test/android/apmtest/AndroidManifest.xml b/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/AndroidManifest.xml
rename to src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
diff --git a/src/modules/audio_processing/main/test/android/apmtest/default.properties b/src/modules/audio_processing/test/android/apmtest/default.properties
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/default.properties
rename to src/modules/audio_processing/test/android/apmtest/default.properties
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/Android.mk b/src/modules/audio_processing/test/android/apmtest/jni/Android.mk
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/jni/Android.mk
rename to src/modules/audio_processing/test/android/apmtest/jni/Android.mk
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/Application.mk b/src/modules/audio_processing/test/android/apmtest/jni/Application.mk
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/jni/Application.mk
rename to src/modules/audio_processing/test/android/apmtest/jni/Application.mk
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/main.c b/src/modules/audio_processing/test/android/apmtest/jni/main.c
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/jni/main.c
rename to src/modules/audio_processing/test/android/apmtest/jni/main.c
diff --git a/src/modules/audio_processing/main/test/android/apmtest/res/values/strings.xml b/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
similarity index 100%
rename from src/modules/audio_processing/main/test/android/apmtest/res/values/strings.xml
rename to src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
diff --git a/src/modules/audio_processing/main/test/process_test/apmtest.m b/src/modules/audio_processing/test/apmtest.m
similarity index 90%
rename from src/modules/audio_processing/main/test/process_test/apmtest.m
rename to src/modules/audio_processing/test/apmtest.m
index 6152bb5..3172cd1 100644
--- a/src/modules/audio_processing/main/test/process_test/apmtest.m
+++ b/src/modules/audio_processing/test/apmtest.m
@@ -1,4 +1,4 @@
-function apmtest(task, testname, casenumber, legacy)
+function apmtest(task, testname, filepath, casenumber, legacy)
%APMTEST is a tool to process APM file sets and easily display the output.
% APMTEST(TASK, TESTNAME, CASENUMBER) performs one of several TASKs:
% 'test' Processes the files to produce test output.
@@ -17,27 +17,33 @@
% 'ns' The NS test set.
% 'vad' The VAD test set.
%
+% FILEPATH specifies the path to the test data files.
+%
% CASENUMBER can be used to select a single test case. Omit CASENUMBER,
% or set to zero, to use all test cases.
%
-if nargin < 4
+if nargin < 5 || isempty(legacy)
% Set to true to run old VQE recordings.
legacy = false;
end
-if nargin < 3
+if nargin < 4 || isempty(casenumber)
casenumber = 0;
end
-if nargin < 2
- task = 'test';
+if nargin < 3 || isempty(filepath)
+ filepath = 'data/';
end
-if nargin < 1
+if nargin < 2 || isempty(testname)
testname = 'all';
end
+if nargin < 1 || isempty(task)
+ task = 'test';
+end
+
if ~strcmp(task, 'test') && ~strcmp(task, 'list') && ~strcmp(task, 'show')
error(['TASK ' task ' is not recognized']);
end
@@ -46,17 +52,10 @@
error(['CASENUMBER must be specified for TASK ' task]);
end
-filepath = 'data/';
inpath = [filepath 'input/'];
outpath = [filepath 'output/'];
refpath = [filepath 'reference/'];
-% Temporary
-if legacy
- refpath = [filepath 'output/'];
- outpath = [filepath 'reference/'];
-end
-
if strcmp(testname, 'all')
tests = {'apm','apmm','aec','aecm','agc','ns','vad'};
else
@@ -64,7 +63,7 @@
end
if legacy
- progname = '/usr/local/google/p4/dev/depot/test';
+ progname = './test';
else
progname = './process_test';
end
@@ -127,24 +126,24 @@
error(['TESTNAME ' tests{i} ' is not recognized']);
end
- inpath = [inpath testdir];
- outpath = [outpath testdir];
- refpath = [refpath testdir];
+ inpathtest = [inpath testdir];
+ outpathtest = [outpath testdir];
+ refpathtest = [refpath testdir];
- if ~exist(inpath,'dir')
- error(['Input directory ' inpath ' does not exist']);
+ if ~exist(inpathtest,'dir')
+ error(['Input directory ' inpathtest ' does not exist']);
end
- if ~exist(refpath,'dir')
- warning(['Reference directory ' refpath ' does not exist']);
+ if ~exist(refpathtest,'dir')
+ warning(['Reference directory ' refpathtest ' does not exist']);
end
- [status, errMsg] = mkdir(outpath);
+ [status, errMsg] = mkdir(outpathtest);
if (status == 0)
error(errMsg);
end
- [nErr, nCases] = recurseDir(inpath, outpath, refpath, outfile, ...
+ [nErr, nCases] = recurseDir(inpathtest, outpathtest, refpathtest, outfile, ...
progname, opt, simulateMode, nErr, nCases, task, casenumber, legacy);
if strcmp(task, 'test') || strcmp(task, 'show')
@@ -221,13 +220,13 @@
if exist([inpath 'vqeEvent.dat'])
system(['ln -s -f ' inpath 'vqeEvent.dat ' eventFile]);
- elseif exist([inpath 'apm_event.day'])
+ elseif exist([inpath 'apm_event.dat'])
system(['ln -s -f ' inpath 'apm_event.dat ' eventFile]);
end
if exist([inpath 'vqeBuf.dat'])
system(['ln -s -f ' inpath 'vqeBuf.dat ' delayFile]);
- elseif exist([inpath 'apm_delay.day'])
+ elseif exist([inpath 'apm_delay.dat'])
system(['ln -s -f ' inpath 'apm_delay.dat ' delayFile]);
end
@@ -296,10 +295,6 @@
diffvector);
%spclab(fs, diffvector);
end
-
- if vadTest == 1
- spclab([refpath vadoutfile], [outpath vadoutfile]);
- end
end
end
end
diff --git a/src/modules/audio_processing/test/process_test.cc b/src/modules/audio_processing/test/process_test.cc
new file mode 100644
index 0000000..2023ddb
--- /dev/null
+++ b/src/modules/audio_processing/test/process_test.cc
@@ -0,0 +1,964 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <sys/stat.h>
+#endif
+
+#include "gtest/gtest.h"
+
+#include "audio_processing.h"
+#include "cpu_features_wrapper.h"
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "tick_util.h"
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+
+using webrtc::AudioFrame;
+using webrtc::AudioProcessing;
+using webrtc::EchoCancellation;
+using webrtc::GainControl;
+using webrtc::NoiseSuppression;
+using webrtc::scoped_array;
+using webrtc::TickInterval;
+using webrtc::TickTime;
+
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace {
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file,
+ ::google::protobuf::MessageLite* msg) {
+ // The "wire format" for the size is little-endian.
+ // Assume process_test is running on a little-endian machine.
+ int32_t size = 0;
+ if (fread(&size, sizeof(int32_t), 1, file) != 1) {
+ return false;
+ }
+ if (size <= 0) {
+ return false;
+ }
+ const size_t usize = static_cast<size_t>(size);
+
+ scoped_array<char> array(new char[usize]);
+ if (fread(array.get(), sizeof(char), usize, file) != usize) {
+ return false;
+ }
+
+ msg->Clear();
+ return msg->ParseFromArray(array.get(), usize);
+}
+
+void PrintStat(const AudioProcessing::Statistic& stat) {
+ printf("%d, %d, %d\n", stat.average,
+ stat.maximum,
+ stat.minimum);
+}
+
+void usage() {
+ printf(
+ "Usage: process_test [options] [-pb PROTOBUF_FILE]\n"
+ " [-ir REVERSE_FILE] [-i PRIMARY_FILE] [-o OUT_FILE]\n");
+ printf(
+ "process_test is a test application for AudioProcessing.\n\n"
+ "When a protobuf debug file is available, specify it with -pb.\n"
+ "Alternately, when -ir or -i is used, the specified files will be\n"
+ "processed directly in a simulation mode. Otherwise the full set of\n"
+ "legacy test files is expected to be present in the working directory.\n");
+ printf("\n");
+ printf("Options\n");
+ printf("General configuration (only used for the simulation mode):\n");
+ printf(" -fs SAMPLE_RATE_HZ\n");
+ printf(" -ch CHANNELS_IN CHANNELS_OUT\n");
+ printf(" -rch REVERSE_CHANNELS\n");
+ printf("\n");
+ printf("Component configuration:\n");
+ printf(
+ "All components are disabled by default. Each block below begins with a\n"
+ "flag to enable the component with default settings. The subsequent flags\n"
+ "in the block are used to provide configuration settings.\n");
+ printf("\n -aec Echo cancellation\n");
+ printf(" --drift_compensation\n");
+ printf(" --no_drift_compensation\n");
+ printf(" --no_echo_metrics\n");
+ printf(" --no_delay_logging\n");
+ printf("\n -aecm Echo control mobile\n");
+ printf(" --aecm_echo_path_in_file FILE\n");
+ printf(" --aecm_echo_path_out_file FILE\n");
+ printf("\n -agc Gain control\n");
+ printf(" --analog\n");
+ printf(" --adaptive_digital\n");
+ printf(" --fixed_digital\n");
+ printf(" --target_level LEVEL\n");
+ printf(" --compression_gain GAIN\n");
+ printf(" --limiter\n");
+ printf(" --no_limiter\n");
+ printf("\n -hpf High pass filter\n");
+ printf("\n -ns Noise suppression\n");
+ printf(" --ns_low\n");
+ printf(" --ns_moderate\n");
+ printf(" --ns_high\n");
+ printf(" --ns_very_high\n");
+ printf("\n -vad Voice activity detection\n");
+ printf(" --vad_out_file FILE\n");
+ printf("\n Level metrics (enabled by default)\n");
+ printf(" --no_level_metrics\n");
+ printf("\n");
+ printf("Modifiers:\n");
+ printf(" --noasm Disable SSE optimization.\n");
+ printf(" --delay DELAY Add DELAY ms to input value.\n");
+ printf(" --perf Measure performance.\n");
+ printf(" --quiet Suppress text output.\n");
+ printf(" --no_progress Suppress progress.\n");
+ printf(" --debug_file FILE Dump a debug recording.\n");
+}
+
+// void function for gtest.
+void void_main(int argc, char* argv[]) {
+ if (argc > 1 && strcmp(argv[1], "--help") == 0) {
+ usage();
+ return;
+ }
+
+ if (argc < 2) {
+ printf("Did you mean to run without arguments?\n");
+ printf("Try `process_test --help' for more information.\n\n");
+ }
+
+ AudioProcessing* apm = AudioProcessing::Create(0);
+ ASSERT_TRUE(apm != NULL);
+
+ const char* pb_filename = NULL;
+ const char* far_filename = NULL;
+ const char* near_filename = NULL;
+ const char* out_filename = NULL;
+ const char* vad_out_filename = NULL;
+ const char* aecm_echo_path_in_filename = NULL;
+ const char* aecm_echo_path_out_filename = NULL;
+
+ int32_t sample_rate_hz = 16000;
+ int32_t device_sample_rate_hz = 16000;
+
+ int num_capture_input_channels = 1;
+ int num_capture_output_channels = 1;
+ int num_render_channels = 1;
+
+ int samples_per_channel = sample_rate_hz / 100;
+
+ bool simulating = false;
+ bool perf_testing = false;
+ bool verbose = true;
+ bool progress = true;
+ int extra_delay_ms = 0;
+ //bool interleaved = true;
+
+ ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+ for (int i = 1; i < argc; i++) {
+ if (strcmp(argv[i], "-pb") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify protobuf filename after -pb";
+ pb_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-ir") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -ir";
+ far_filename = argv[i];
+ simulating = true;
+
+ } else if (strcmp(argv[i], "-i") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -i";
+ near_filename = argv[i];
+ simulating = true;
+
+ } else if (strcmp(argv[i], "-o") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after -o";
+ out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-fs") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify sample rate after -fs";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
+ samples_per_channel = sample_rate_hz / 100;
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(sample_rate_hz));
+
+ } else if (strcmp(argv[i], "-ch") == 0) {
+ i++;
+ ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
+ i++;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_channels(num_capture_input_channels,
+ num_capture_output_channels));
+
+ } else if (strcmp(argv[i], "-rch") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify number of channels after -rch";
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_reverse_channels(num_render_channels));
+
+ } else if (strcmp(argv[i], "-aec") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_metrics(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_delay_logging(true));
+
+ } else if (strcmp(argv[i], "--drift_compensation") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ // TODO(ajm): this is enabled in the VQE test app by default. Investigate
+ // why it can give better performance despite passing zeros.
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_drift_compensation(true));
+ } else if (strcmp(argv[i], "--no_drift_compensation") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_drift_compensation(false));
+
+ } else if (strcmp(argv[i], "--no_echo_metrics") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_metrics(false));
+
+ } else if (strcmp(argv[i], "--no_delay_logging") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->enable_delay_logging(false));
+
+ } else if (strcmp(argv[i], "--no_level_metrics") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
+
+ } else if (strcmp(argv[i], "-aecm") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
+
+ } else if (strcmp(argv[i], "--aecm_echo_path_in_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_in_file";
+ aecm_echo_path_in_filename = argv[i];
+
+ } else if (strcmp(argv[i], "--aecm_echo_path_out_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_out_file";
+ aecm_echo_path_out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "-agc") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+
+ } else if (strcmp(argv[i], "--analog") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
+
+ } else if (strcmp(argv[i], "--adaptive_digital") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+
+ } else if (strcmp(argv[i], "--fixed_digital") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_mode(GainControl::kFixedDigital));
+
+ } else if (strcmp(argv[i], "--target_level") == 0) {
+ i++;
+ int level;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &level));
+
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_target_level_dbfs(level));
+
+ } else if (strcmp(argv[i], "--compression_gain") == 0) {
+ i++;
+ int gain;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &gain));
+
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_compression_gain_db(gain));
+
+ } else if (strcmp(argv[i], "--limiter") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->enable_limiter(true));
+
+ } else if (strcmp(argv[i], "--no_limiter") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->enable_limiter(false));
+
+ } else if (strcmp(argv[i], "-hpf") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->high_pass_filter()->Enable(true));
+
+ } else if (strcmp(argv[i], "-ns") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+
+ } else if (strcmp(argv[i], "--ns_low") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kLow));
+
+ } else if (strcmp(argv[i], "--ns_moderate") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kModerate));
+
+ } else if (strcmp(argv[i], "--ns_high") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kHigh));
+
+ } else if (strcmp(argv[i], "--ns_very_high") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+ ASSERT_EQ(apm->kNoError,
+ apm->noise_suppression()->set_level(NoiseSuppression::kVeryHigh));
+
+ } else if (strcmp(argv[i], "-vad") == 0) {
+ ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+
+ } else if (strcmp(argv[i], "--vad_out_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --vad_out_file";
+ vad_out_filename = argv[i];
+
+ } else if (strcmp(argv[i], "--noasm") == 0) {
+ WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
+ // We need to reinitialize here if components have already been enabled.
+ ASSERT_EQ(apm->kNoError, apm->Initialize());
+
+ } else if (strcmp(argv[i], "--delay") == 0) {
+ i++;
+ ASSERT_EQ(1, sscanf(argv[i], "%d", &extra_delay_ms));
+
+ } else if (strcmp(argv[i], "--perf") == 0) {
+ perf_testing = true;
+
+ } else if (strcmp(argv[i], "--quiet") == 0) {
+ verbose = false;
+ progress = false;
+
+ } else if (strcmp(argv[i], "--no_progress") == 0) {
+ progress = false;
+
+ } else if (strcmp(argv[i], "--debug_file") == 0) {
+ i++;
+ ASSERT_LT(i, argc) << "Specify filename after --debug_file";
+ ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i]));
+ } else {
+ FAIL() << "Unrecognized argument " << argv[i];
+ }
+ }
+ // If we're reading a protobuf file, ensure a simulation hasn't also
+ // been requested (which makes no sense...)
+ ASSERT_FALSE(pb_filename && simulating);
+
+ if (verbose) {
+ printf("Sample rate: %d Hz\n", sample_rate_hz);
+ printf("Primary channels: %d (in), %d (out)\n",
+ num_capture_input_channels,
+ num_capture_output_channels);
+ printf("Reverse channels: %d \n", num_render_channels);
+ }
+
+ const char far_file_default[] = "apm_far.pcm";
+ const char near_file_default[] = "apm_near.pcm";
+ const char out_file_default[] = "out.pcm";
+ const char event_filename[] = "apm_event.dat";
+ const char delay_filename[] = "apm_delay.dat";
+ const char drift_filename[] = "apm_drift.dat";
+ const char vad_file_default[] = "vad_out.dat";
+
+ if (!simulating) {
+ far_filename = far_file_default;
+ near_filename = near_file_default;
+ }
+
+ if (!out_filename) {
+ out_filename = out_file_default;
+ }
+
+ if (!vad_out_filename) {
+ vad_out_filename = vad_file_default;
+ }
+
+ FILE* pb_file = NULL;
+ FILE* far_file = NULL;
+ FILE* near_file = NULL;
+ FILE* out_file = NULL;
+ FILE* event_file = NULL;
+ FILE* delay_file = NULL;
+ FILE* drift_file = NULL;
+ FILE* vad_out_file = NULL;
+ FILE* aecm_echo_path_in_file = NULL;
+ FILE* aecm_echo_path_out_file = NULL;
+
+ if (pb_filename) {
+ pb_file = fopen(pb_filename, "rb");
+ ASSERT_TRUE(NULL != pb_file) << "Unable to open protobuf file "
+ << pb_filename;
+ } else {
+ if (far_filename) {
+ far_file = fopen(far_filename, "rb");
+ ASSERT_TRUE(NULL != far_file) << "Unable to open far-end audio file "
+ << far_filename;
+ }
+
+ near_file = fopen(near_filename, "rb");
+ ASSERT_TRUE(NULL != near_file) << "Unable to open near-end audio file "
+ << near_filename;
+ if (!simulating) {
+ event_file = fopen(event_filename, "rb");
+ ASSERT_TRUE(NULL != event_file) << "Unable to open event file "
+ << event_filename;
+
+ delay_file = fopen(delay_filename, "rb");
+ ASSERT_TRUE(NULL != delay_file) << "Unable to open buffer file "
+ << delay_filename;
+
+ drift_file = fopen(drift_filename, "rb");
+ ASSERT_TRUE(NULL != drift_file) << "Unable to open drift file "
+ << drift_filename;
+ }
+ }
+
+ out_file = fopen(out_filename, "wb");
+ ASSERT_TRUE(NULL != out_file) << "Unable to open output audio file "
+ << out_filename;
+
+ int near_size_bytes = 0;
+ if (pb_file) {
+ struct stat st;
+ stat(pb_filename, &st);
+ // Crude estimate, but should be good enough.
+ near_size_bytes = st.st_size / 3;
+ } else {
+ struct stat st;
+ stat(near_filename, &st);
+ near_size_bytes = st.st_size;
+ }
+
+ if (apm->voice_detection()->is_enabled()) {
+ vad_out_file = fopen(vad_out_filename, "wb");
+ ASSERT_TRUE(NULL != vad_out_file) << "Unable to open VAD output file "
+ << vad_out_file;
+ }
+
+ if (aecm_echo_path_in_filename != NULL) {
+ aecm_echo_path_in_file = fopen(aecm_echo_path_in_filename, "rb");
+ ASSERT_TRUE(NULL != aecm_echo_path_in_file) << "Unable to open file "
+ << aecm_echo_path_in_filename;
+
+ const size_t path_size =
+ apm->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path(new char[path_size]);
+ ASSERT_EQ(path_size, fread(echo_path.get(),
+ sizeof(char),
+ path_size,
+ aecm_echo_path_in_file));
+ EXPECT_EQ(apm->kNoError,
+ apm->echo_control_mobile()->SetEchoPath(echo_path.get(),
+ path_size));
+ fclose(aecm_echo_path_in_file);
+ aecm_echo_path_in_file = NULL;
+ }
+
+ if (aecm_echo_path_out_filename != NULL) {
+ aecm_echo_path_out_file = fopen(aecm_echo_path_out_filename, "wb");
+ ASSERT_TRUE(NULL != aecm_echo_path_out_file) << "Unable to open file "
+ << aecm_echo_path_out_filename;
+ }
+
+ size_t read_count = 0;
+ int reverse_count = 0;
+ int primary_count = 0;
+ int near_read_bytes = 0;
+ TickInterval acc_ticks;
+
+ AudioFrame far_frame;
+ AudioFrame near_frame;
+
+ int delay_ms = 0;
+ int drift_samples = 0;
+ int capture_level = 127;
+ int8_t stream_has_voice = 0;
+
+ TickTime t0 = TickTime::Now();
+ TickTime t1 = t0;
+ WebRtc_Word64 max_time_us = 0;
+ WebRtc_Word64 max_time_reverse_us = 0;
+ WebRtc_Word64 min_time_us = 1e6;
+ WebRtc_Word64 min_time_reverse_us = 1e6;
+
+ // TODO(ajm): Ideally we would refactor this block into separate functions,
+ // but for now we want to share the variables.
+ if (pb_file) {
+ Event event_msg;
+ while (ReadMessageFromFile(pb_file, &event_msg)) {
+ std::ostringstream trace_stream;
+ trace_stream << "Processed frames: " << reverse_count << " (reverse), "
+ << primary_count << " (primary)";
+ SCOPED_TRACE(trace_stream.str());
+
+ if (event_msg.type() == Event::INIT) {
+ ASSERT_TRUE(event_msg.has_init());
+ const Init msg = event_msg.init();
+
+ ASSERT_TRUE(msg.has_sample_rate());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(msg.sample_rate()));
+
+ ASSERT_TRUE(msg.has_device_sample_rate());
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_device_sample_rate_hz(
+ msg.device_sample_rate()));
+
+ ASSERT_TRUE(msg.has_num_input_channels());
+ ASSERT_TRUE(msg.has_num_output_channels());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_channels(msg.num_input_channels(),
+ msg.num_output_channels()));
+
+ ASSERT_TRUE(msg.has_num_reverse_channels());
+ ASSERT_EQ(apm->kNoError,
+ apm->set_num_reverse_channels(msg.num_reverse_channels()));
+
+ samples_per_channel = msg.sample_rate() / 100;
+ far_frame._frequencyInHz = msg.sample_rate();
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = msg.num_reverse_channels();
+ near_frame._frequencyInHz = msg.sample_rate();
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (verbose) {
+ printf("Init at frame: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+ printf(" Sample rate: %d Hz\n", msg.sample_rate());
+ printf(" Primary channels: %d (in), %d (out)\n",
+ msg.num_input_channels(),
+ msg.num_output_channels());
+ printf(" Reverse channels: %d \n", msg.num_reverse_channels());
+ }
+
+ } else if (event_msg.type() == Event::REVERSE_STREAM) {
+ ASSERT_TRUE(event_msg.has_reverse_stream());
+ const ReverseStream msg = event_msg.reverse_stream();
+ reverse_count++;
+
+ ASSERT_TRUE(msg.has_data());
+ ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
+ far_frame._audioChannel, msg.data().size());
+ memcpy(far_frame._payloadData, msg.data().data(), msg.data().size());
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->AnalyzeReverseStream(&far_frame));
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_reverse_us) {
+ max_time_reverse_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_reverse_us) {
+ min_time_reverse_us = tick_diff.Microseconds();
+ }
+ }
+
+ } else if (event_msg.type() == Event::STREAM) {
+ ASSERT_TRUE(event_msg.has_stream());
+ const Stream msg = event_msg.stream();
+ primary_count++;
+
+ // ProcessStream could have changed this for the output frame.
+ near_frame._audioChannel = apm->num_input_channels();
+
+ ASSERT_TRUE(msg.has_input_data());
+ ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
+ near_frame._audioChannel, msg.input_data().size());
+ memcpy(near_frame._payloadData,
+ msg.input_data().data(),
+ msg.input_data().size());
+
+ near_read_bytes += msg.input_data().size();
+ if (progress && primary_count % 100 == 0) {
+ printf("%.0f%% complete\r",
+ (near_read_bytes * 100.0) / near_size_bytes);
+ fflush(stdout);
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_stream_analog_level(msg.level()));
+ ASSERT_EQ(apm->kNoError,
+ apm->set_stream_delay_ms(msg.delay() + extra_delay_ms));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_stream_drift_samples(msg.drift()));
+
+ int err = apm->ProcessStream(&near_frame);
+ if (err == apm->kBadStreamParameterWarning) {
+ printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
+ }
+ ASSERT_TRUE(err == apm->kNoError ||
+ err == apm->kBadStreamParameterWarning);
+ ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
+
+ capture_level = apm->gain_control()->stream_analog_level();
+
+ stream_has_voice =
+ static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
+ if (vad_out_file != NULL) {
+ ASSERT_EQ(1u, fwrite(&stream_has_voice,
+ sizeof(stream_has_voice),
+ 1,
+ vad_out_file));
+ }
+
+ if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
+ ASSERT_EQ(msg.level(), capture_level);
+ }
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_us) {
+ max_time_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_us) {
+ min_time_us = tick_diff.Microseconds();
+ }
+ }
+
+ size_t size = samples_per_channel * near_frame._audioChannel;
+ ASSERT_EQ(size, fwrite(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ out_file));
+ }
+ }
+
+ ASSERT_TRUE(feof(pb_file));
+
+ } else {
+ enum Events {
+ kInitializeEvent,
+ kRenderEvent,
+ kCaptureEvent,
+ kResetEventDeprecated
+ };
+ int16_t event = 0;
+ while (simulating || feof(event_file) == 0) {
+ std::ostringstream trace_stream;
+ trace_stream << "Processed frames: " << reverse_count << " (reverse), "
+ << primary_count << " (primary)";
+ SCOPED_TRACE(trace_stream.str());
+
+ if (simulating) {
+ if (far_file == NULL) {
+ event = kCaptureEvent;
+ } else {
+ if (event == kRenderEvent) {
+ event = kCaptureEvent;
+ } else {
+ event = kRenderEvent;
+ }
+ }
+ } else {
+ read_count = fread(&event, sizeof(event), 1, event_file);
+ if (read_count != 1) {
+ break;
+ }
+ }
+
+ far_frame._frequencyInHz = sample_rate_hz;
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = num_render_channels;
+ near_frame._frequencyInHz = sample_rate_hz;
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (event == kInitializeEvent || event == kResetEventDeprecated) {
+ ASSERT_EQ(1u,
+ fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file));
+ samples_per_channel = sample_rate_hz / 100;
+
+ ASSERT_EQ(1u,
+ fread(&device_sample_rate_hz,
+ sizeof(device_sample_rate_hz),
+ 1,
+ event_file));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->set_sample_rate_hz(sample_rate_hz));
+
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_device_sample_rate_hz(
+ device_sample_rate_hz));
+
+ far_frame._frequencyInHz = sample_rate_hz;
+ far_frame._payloadDataLengthInSamples = samples_per_channel;
+ far_frame._audioChannel = num_render_channels;
+ near_frame._frequencyInHz = sample_rate_hz;
+ near_frame._payloadDataLengthInSamples = samples_per_channel;
+
+ if (verbose) {
+ printf("Init at frame: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+ printf(" Sample rate: %d Hz\n", sample_rate_hz);
+ }
+
+ } else if (event == kRenderEvent) {
+ reverse_count++;
+
+ size_t size = samples_per_channel * num_render_channels;
+ read_count = fread(far_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ far_file);
+
+ if (simulating) {
+ if (read_count != size) {
+ // Read an equal amount from the near file to avoid errors due to
+ // not reaching end-of-file.
+ EXPECT_EQ(0, fseek(near_file, read_count * sizeof(int16_t),
+ SEEK_CUR));
+ break; // This is expected.
+ }
+ } else {
+ ASSERT_EQ(size, read_count);
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ ASSERT_EQ(apm->kNoError,
+ apm->AnalyzeReverseStream(&far_frame));
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_reverse_us) {
+ max_time_reverse_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_reverse_us) {
+ min_time_reverse_us = tick_diff.Microseconds();
+ }
+ }
+
+ } else if (event == kCaptureEvent) {
+ primary_count++;
+ near_frame._audioChannel = num_capture_input_channels;
+
+ size_t size = samples_per_channel * num_capture_input_channels;
+ read_count = fread(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ near_file);
+
+ near_read_bytes += read_count * sizeof(int16_t);
+ if (progress && primary_count % 100 == 0) {
+ printf("%.0f%% complete\r",
+ (near_read_bytes * 100.0) / near_size_bytes);
+ fflush(stdout);
+ }
+ if (simulating) {
+ if (read_count != size) {
+ break; // This is expected.
+ }
+
+ delay_ms = 0;
+ drift_samples = 0;
+ } else {
+ ASSERT_EQ(size, read_count);
+
+ // TODO(ajm): sizeof(delay_ms) for current files?
+ ASSERT_EQ(1u,
+ fread(&delay_ms, 2, 1, delay_file));
+ ASSERT_EQ(1u,
+ fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
+ }
+
+ if (perf_testing) {
+ t0 = TickTime::Now();
+ }
+
+ // TODO(ajm): fake an analog gain while simulating.
+
+ int capture_level_in = capture_level;
+ ASSERT_EQ(apm->kNoError,
+ apm->gain_control()->set_stream_analog_level(capture_level));
+ ASSERT_EQ(apm->kNoError,
+ apm->set_stream_delay_ms(delay_ms + extra_delay_ms));
+ ASSERT_EQ(apm->kNoError,
+ apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
+
+ int err = apm->ProcessStream(&near_frame);
+ if (err == apm->kBadStreamParameterWarning) {
+ printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
+ }
+ ASSERT_TRUE(err == apm->kNoError ||
+ err == apm->kBadStreamParameterWarning);
+ ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
+
+ capture_level = apm->gain_control()->stream_analog_level();
+
+ stream_has_voice =
+ static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
+ if (vad_out_file != NULL) {
+ ASSERT_EQ(1u, fwrite(&stream_has_voice,
+ sizeof(stream_has_voice),
+ 1,
+ vad_out_file));
+ }
+
+ if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
+ ASSERT_EQ(capture_level_in, capture_level);
+ }
+
+ if (perf_testing) {
+ t1 = TickTime::Now();
+ TickInterval tick_diff = t1 - t0;
+ acc_ticks += tick_diff;
+ if (tick_diff.Microseconds() > max_time_us) {
+ max_time_us = tick_diff.Microseconds();
+ }
+ if (tick_diff.Microseconds() < min_time_us) {
+ min_time_us = tick_diff.Microseconds();
+ }
+ }
+
+ size = samples_per_channel * near_frame._audioChannel;
+ ASSERT_EQ(size, fwrite(near_frame._payloadData,
+ sizeof(int16_t),
+ size,
+ out_file));
+ }
+ else {
+ FAIL() << "Event " << event << " is unrecognized";
+ }
+ }
+ }
+ printf("100%% complete\r");
+
+ if (aecm_echo_path_out_file != NULL) {
+ const size_t path_size =
+ apm->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path(new char[path_size]);
+ apm->echo_control_mobile()->GetEchoPath(echo_path.get(), path_size);
+ ASSERT_EQ(path_size, fwrite(echo_path.get(),
+ sizeof(char),
+ path_size,
+ aecm_echo_path_out_file));
+ fclose(aecm_echo_path_out_file);
+ aecm_echo_path_out_file = NULL;
+ }
+
+ if (verbose) {
+ printf("\nProcessed frames: %d (primary), %d (reverse)\n",
+ primary_count, reverse_count);
+
+ if (apm->level_estimator()->is_enabled()) {
+ printf("\n--Level metrics--\n");
+ printf("RMS: %d dBFS\n", -apm->level_estimator()->RMS());
+ }
+ if (apm->echo_cancellation()->are_metrics_enabled()) {
+ EchoCancellation::Metrics metrics;
+ apm->echo_cancellation()->GetMetrics(&metrics);
+ printf("\n--Echo metrics--\n");
+ printf("(avg, max, min)\n");
+ printf("ERL: ");
+ PrintStat(metrics.echo_return_loss);
+ printf("ERLE: ");
+ PrintStat(metrics.echo_return_loss_enhancement);
+ printf("ANLP: ");
+ PrintStat(metrics.a_nlp);
+ }
+ if (apm->echo_cancellation()->is_delay_logging_enabled()) {
+ int median = 0;
+ int std = 0;
+ apm->echo_cancellation()->GetDelayMetrics(&median, &std);
+ printf("\n--Delay metrics--\n");
+ printf("Median: %3d\n", median);
+ printf("Standard deviation: %3d\n", std);
+ }
+ }
+
+ if (!pb_file) {
+ int8_t temp_int8;
+ if (far_file) {
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, far_file);
+ EXPECT_NE(0, feof(far_file)) << "Far-end file not fully processed";
+ }
+
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, near_file);
+ EXPECT_NE(0, feof(near_file)) << "Near-end file not fully processed";
+
+ if (!simulating) {
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, event_file);
+ EXPECT_NE(0, feof(event_file)) << "Event file not fully processed";
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, delay_file);
+ EXPECT_NE(0, feof(delay_file)) << "Delay file not fully processed";
+ read_count = fread(&temp_int8, sizeof(temp_int8), 1, drift_file);
+ EXPECT_NE(0, feof(drift_file)) << "Drift file not fully processed";
+ }
+ }
+
+ if (perf_testing) {
+ if (primary_count > 0) {
+ WebRtc_Word64 exec_time = acc_ticks.Milliseconds();
+ printf("\nTotal time: %.3f s, file time: %.2f s\n",
+ exec_time * 0.001, primary_count * 0.01);
+ printf("Time per frame: %.3f ms (average), %.3f ms (max),"
+ " %.3f ms (min)\n",
+ (exec_time * 1.0) / primary_count,
+ (max_time_us + max_time_reverse_us) / 1000.0,
+ (min_time_us + min_time_reverse_us) / 1000.0);
+ } else {
+ printf("Warning: no capture frames\n");
+ }
+ }
+
+ AudioProcessing::Destroy(apm);
+ apm = NULL;
+}
+} // namespace
+
+int main(int argc, char* argv[])
+{
+ void_main(argc, argv);
+
+ // Optional, but removes memory leak noise from Valgrind.
+ google::protobuf::ShutdownProtobufLibrary();
+ return 0;
+}
diff --git a/src/modules/audio_processing/test/unit_test.cc b/src/modules/audio_processing/test/unit_test.cc
new file mode 100644
index 0000000..6fe5905
--- /dev/null
+++ b/src/modules/audio_processing/test/unit_test.cc
@@ -0,0 +1,1256 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "gtest/gtest.h"
+
+#include "audio_processing.h"
+#include "event_wrapper.h"
+#include "module_common_types.h"
+#include "scoped_ptr.h"
+#include "signal_processing_library.h"
+#include "testsupport/fileutils.h"
+#include "thread_wrapper.h"
+#include "trace.h"
+#ifdef WEBRTC_ANDROID
+#include "external/webrtc/src/modules/audio_processing/test/unittest.pb.h"
+#else
+#include "webrtc/audio_processing/unittest.pb.h"
+#endif
+
+using webrtc::AudioProcessing;
+using webrtc::AudioFrame;
+using webrtc::GainControl;
+using webrtc::NoiseSuppression;
+using webrtc::EchoCancellation;
+using webrtc::EventWrapper;
+using webrtc::scoped_array;
+using webrtc::Trace;
+using webrtc::LevelEstimator;
+using webrtc::EchoCancellation;
+using webrtc::EchoControlMobile;
+using webrtc::VoiceDetection;
+
+namespace {
+// When false, this will compare the output data with the results stored to
+// file. This is the typical case. When the file should be updated, it can
+// be set to true with the command-line switch --write_output_data.
+bool write_output_data = false;
+
+class ApmTest : public ::testing::Test {
+ protected:
+ ApmTest();
+ virtual void SetUp();
+ virtual void TearDown();
+
+ static void SetUpTestCase() {
+ Trace::CreateTrace();
+ std::string trace_filename = webrtc::test::OutputPath() +
+ "audioproc_trace.txt";
+ ASSERT_EQ(0, Trace::SetTraceFile(trace_filename.c_str()));
+ }
+
+ static void TearDownTestCase() {
+ Trace::ReturnTrace();
+ }
+ // Path to where the resource files to be used for this test are located.
+ const std::string resource_path;
+ const std::string output_filename;
+ webrtc::AudioProcessing* apm_;
+ webrtc::AudioFrame* frame_;
+ webrtc::AudioFrame* revframe_;
+ FILE* far_file_;
+ FILE* near_file_;
+};
+
+ApmTest::ApmTest()
+ : resource_path(webrtc::test::ProjectRootPath() +
+ "test/data/audio_processing/"),
+#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
+ output_filename(resource_path + "output_data_fixed.pb"),
+#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ output_filename(resource_path + "output_data_float.pb"),
+#endif
+ apm_(NULL),
+ frame_(NULL),
+ revframe_(NULL),
+ far_file_(NULL),
+ near_file_(NULL) {}
+
+void ApmTest::SetUp() {
+ apm_ = AudioProcessing::Create(0);
+ ASSERT_TRUE(apm_ != NULL);
+
+ frame_ = new AudioFrame();
+ revframe_ = new AudioFrame();
+
+ ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
+
+ frame_->_payloadDataLengthInSamples = 320;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 32000;
+ revframe_->_payloadDataLengthInSamples = 320;
+ revframe_->_audioChannel = 2;
+ revframe_->_frequencyInHz = 32000;
+
+ std::string input_filename = resource_path + "aec_far.pcm";
+ far_file_ = fopen(input_filename.c_str(), "rb");
+ ASSERT_TRUE(far_file_ != NULL) << "Could not open input file " <<
+ input_filename << "\n";
+ input_filename = resource_path + "aec_near.pcm";
+ near_file_ = fopen(input_filename.c_str(), "rb");
+ ASSERT_TRUE(near_file_ != NULL) << "Could not open input file " <<
+ input_filename << "\n";
+}
+
+void ApmTest::TearDown() {
+ if (frame_) {
+ delete frame_;
+ }
+ frame_ = NULL;
+
+ if (revframe_) {
+ delete revframe_;
+ }
+ revframe_ = NULL;
+
+ if (far_file_) {
+ ASSERT_EQ(0, fclose(far_file_));
+ }
+ far_file_ = NULL;
+
+ if (near_file_) {
+ ASSERT_EQ(0, fclose(near_file_));
+ }
+ near_file_ = NULL;
+
+ if (apm_ != NULL) {
+ AudioProcessing::Destroy(apm_);
+ }
+ apm_ = NULL;
+}
+
+void MixStereoToMono(const int16_t* stereo,
+ int16_t* mono,
+ int samples_per_channel) {
+ for (int i = 0; i < samples_per_channel; i++) {
+ int32_t int32 = (static_cast<int32_t>(stereo[i * 2]) +
+ static_cast<int32_t>(stereo[i * 2 + 1])) >> 1;
+ mono[i] = static_cast<int16_t>(int32);
+ }
+}
+
+template <class T>
+T MaxValue(T a, T b) {
+ return a > b ? a : b;
+}
+
+template <class T>
+T AbsValue(T a) {
+ return a > 0 ? a : -a;
+}
+
+void SetFrameTo(AudioFrame* frame, int16_t value) {
+ for (int i = 0; i < frame->_payloadDataLengthInSamples * frame->_audioChannel;
+ ++i) {
+ frame->_payloadData[i] = value;
+ }
+}
+
+int16_t MaxAudioFrame(const AudioFrame& frame) {
+ const int length = frame._payloadDataLengthInSamples * frame._audioChannel;
+ int16_t max = AbsValue(frame._payloadData[0]);
+ for (int i = 1; i < length; i++) {
+ max = MaxValue(max, AbsValue(frame._payloadData[i]));
+ }
+
+ return max;
+}
+
+bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+ if (frame1._payloadDataLengthInSamples !=
+ frame2._payloadDataLengthInSamples) {
+ return false;
+ }
+ if (frame1._audioChannel !=
+ frame2._audioChannel) {
+ return false;
+ }
+ if (memcmp(frame1._payloadData, frame2._payloadData,
+ frame1._payloadDataLengthInSamples * frame1._audioChannel *
+ sizeof(int16_t))) {
+ return false;
+ }
+ return true;
+}
+
+void TestStats(const AudioProcessing::Statistic& test,
+ const webrtc::audioproc::Test::Statistic& reference) {
+ EXPECT_EQ(reference.instant(), test.instant);
+ EXPECT_EQ(reference.average(), test.average);
+ EXPECT_EQ(reference.maximum(), test.maximum);
+ EXPECT_EQ(reference.minimum(), test.minimum);
+}
+
+void WriteStatsMessage(const AudioProcessing::Statistic& output,
+ webrtc::audioproc::Test::Statistic* message) {
+ message->set_instant(output.instant);
+ message->set_average(output.average);
+ message->set_maximum(output.maximum);
+ message->set_minimum(output.minimum);
+}
+
+void WriteMessageLiteToFile(const std::string filename,
+ const ::google::protobuf::MessageLite& message) {
+ FILE* file = fopen(filename.c_str(), "wb");
+ ASSERT_TRUE(file != NULL) << "Could not open " << filename;
+ int size = message.ByteSize();
+ ASSERT_GT(size, 0);
+ unsigned char* array = new unsigned char[size];
+ ASSERT_TRUE(message.SerializeToArray(array, size));
+
+ ASSERT_EQ(1u, fwrite(&size, sizeof(int), 1, file));
+ ASSERT_EQ(static_cast<size_t>(size),
+ fwrite(array, sizeof(unsigned char), size, file));
+
+ delete [] array;
+ fclose(file);
+}
+
+void ReadMessageLiteFromFile(const std::string filename,
+ ::google::protobuf::MessageLite* message) {
+ assert(message != NULL);
+
+ FILE* file = fopen(filename.c_str(), "rb");
+ ASSERT_TRUE(file != NULL) << "Could not open " << filename;
+ int size = 0;
+ ASSERT_EQ(1u, fread(&size, sizeof(int), 1, file));
+ ASSERT_GT(size, 0);
+ unsigned char* array = new unsigned char[size];
+ ASSERT_EQ(static_cast<size_t>(size),
+ fread(array, sizeof(unsigned char), size, file));
+
+ ASSERT_TRUE(message->ParseFromArray(array, size));
+
+ delete [] array;
+ fclose(file);
+}
+
+struct ThreadData {
+ ThreadData(int thread_num_, AudioProcessing* ap_)
+ : thread_num(thread_num_),
+ error(false),
+ ap(ap_) {}
+ int thread_num;
+ bool error;
+ AudioProcessing* ap;
+};
+
+// Don't use GTest here; non-thread-safe on Windows (as of 1.5.0).
+bool DeadlockProc(void* thread_object) {
+ ThreadData* thread_data = static_cast<ThreadData*>(thread_object);
+ AudioProcessing* ap = thread_data->ap;
+ int err = ap->kNoError;
+
+ AudioFrame primary_frame;
+ AudioFrame reverse_frame;
+ primary_frame._payloadDataLengthInSamples = 320;
+ primary_frame._audioChannel = 2;
+ primary_frame._frequencyInHz = 32000;
+ reverse_frame._payloadDataLengthInSamples = 320;
+ reverse_frame._audioChannel = 2;
+ reverse_frame._frequencyInHz = 32000;
+
+ ap->echo_cancellation()->Enable(true);
+ ap->gain_control()->Enable(true);
+ ap->high_pass_filter()->Enable(true);
+ ap->level_estimator()->Enable(true);
+ ap->noise_suppression()->Enable(true);
+ ap->voice_detection()->Enable(true);
+
+ if (thread_data->thread_num % 2 == 0) {
+ err = ap->AnalyzeReverseStream(&reverse_frame);
+ if (err != ap->kNoError) {
+ printf("Error in AnalyzeReverseStream(): %d\n", err);
+ thread_data->error = true;
+ return false;
+ }
+ }
+
+ if (thread_data->thread_num % 2 == 1) {
+ ap->set_stream_delay_ms(0);
+ ap->echo_cancellation()->set_stream_drift_samples(0);
+ ap->gain_control()->set_stream_analog_level(0);
+ err = ap->ProcessStream(&primary_frame);
+ if (err == ap->kStreamParameterNotSetError) {
+ printf("Expected kStreamParameterNotSetError in ProcessStream(): %d\n",
+ err);
+ } else if (err != ap->kNoError) {
+ printf("Error in ProcessStream(): %d\n", err);
+ thread_data->error = true;
+ return false;
+ }
+ ap->gain_control()->stream_analog_level();
+ }
+
+ EventWrapper* event = EventWrapper::Create();
+ event->Wait(1);
+ delete event;
+ event = NULL;
+
+ return true;
+}
+
+/*TEST_F(ApmTest, Deadlock) {
+ const int num_threads = 16;
+ std::vector<ThreadWrapper*> threads(num_threads);
+ std::vector<ThreadData*> thread_data(num_threads);
+
+ ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
+
+ for (int i = 0; i < num_threads; i++) {
+ thread_data[i] = new ThreadData(i, apm_);
+ threads[i] = ThreadWrapper::CreateThread(DeadlockProc,
+ thread_data[i],
+ kNormalPriority,
+ 0);
+ ASSERT_TRUE(threads[i] != NULL);
+ unsigned int thread_id = 0;
+ threads[i]->Start(thread_id);
+ }
+
+ EventWrapper* event = EventWrapper::Create();
+ ASSERT_EQ(kEventTimeout, event->Wait(5000));
+ delete event;
+ event = NULL;
+
+ for (int i = 0; i < num_threads; i++) {
+ // This will return false if the thread has deadlocked.
+ ASSERT_TRUE(threads[i]->Stop());
+ ASSERT_FALSE(thread_data[i]->error);
+ delete threads[i];
+ threads[i] = NULL;
+ delete thread_data[i];
+ thread_data[i] = NULL;
+ }
+}*/
+
+TEST_F(ApmTest, StreamParameters) {
+ // No errors when the components are disabled.
+ EXPECT_EQ(apm_->kNoError,
+ apm_->ProcessStream(frame_));
+
+ // -- Missing AGC level --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError,
+ apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(false));
+
+ // -- Missing delay --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+
+ // -- Missing drift --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Resets after successful ProcessStream().
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // Other stream parameters set correctly.
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
+
+ // -- No stream parameters --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->AnalyzeReverseStream(revframe_));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError,
+ apm_->ProcessStream(frame_));
+
+ // -- All there --
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+}
+
+TEST_F(ApmTest, Channels) {
+ // Testing number of invalid channels
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
+ // Testing number of valid channels
+ for (int i = 1; i < 3; i++) {
+ for (int j = 1; j < 3; j++) {
+ if (j > i) {
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
+ } else {
+ EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
+ EXPECT_EQ(j, apm_->num_output_channels());
+ }
+ }
+ EXPECT_EQ(i, apm_->num_input_channels());
+ EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
+ EXPECT_EQ(i, apm_->num_reverse_channels());
+ }
+}
+
+TEST_F(ApmTest, SampleRates) {
+ // Testing invalid sample rates
+ EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
+ // Testing valid sample rates
+ int fs[] = {8000, 16000, 32000};
+ for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
+ EXPECT_EQ(fs[i], apm_->sample_rate_hz());
+ }
+}
+
+
+TEST_F(ApmTest, EchoCancellation) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(4000));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(100000));
+
+ int rate[] = {16000, 44100, 48000};
+ for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
+ EXPECT_EQ(rate[i],
+ apm_->echo_cancellation()->device_sample_rate_hz());
+ }
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_suppression_level(
+ static_cast<EchoCancellation::SuppressionLevel>(-1)));
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_cancellation()->set_suppression_level(
+ static_cast<EchoCancellation::SuppressionLevel>(4)));
+
+ EchoCancellation::SuppressionLevel level[] = {
+ EchoCancellation::kLowSuppression,
+ EchoCancellation::kModerateSuppression,
+ EchoCancellation::kHighSuppression,
+ };
+ for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_suppression_level(level[i]));
+ EXPECT_EQ(level[i],
+ apm_->echo_cancellation()->suppression_level());
+ }
+
+ EchoCancellation::Metrics metrics;
+ EXPECT_EQ(apm_->kNotEnabledError,
+ apm_->echo_cancellation()->GetMetrics(&metrics));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
+
+ int median = 0;
+ int std = 0;
+ EXPECT_EQ(apm_->kNotEnabledError,
+ apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_delay_logging_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_delay_logging_enabled());
+
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+ EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
+}
+
+TEST_F(ApmTest, EchoControlMobile) {
+ // AECM won't use super-wideband.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ // Turn AECM on (and AEC off)
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+ EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->set_routing_mode(
+ static_cast<EchoControlMobile::RoutingMode>(-1)));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->set_routing_mode(
+ static_cast<EchoControlMobile::RoutingMode>(5)));
+
+ // Toggle routing modes
+ EchoControlMobile::RoutingMode mode[] = {
+ EchoControlMobile::kQuietEarpieceOrHeadset,
+ EchoControlMobile::kEarpiece,
+ EchoControlMobile::kLoudEarpiece,
+ EchoControlMobile::kSpeakerphone,
+ EchoControlMobile::kLoudSpeakerphone,
+ };
+ for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->set_routing_mode(mode[i]));
+ EXPECT_EQ(mode[i],
+ apm_->echo_control_mobile()->routing_mode());
+ }
+ // Turn comfort noise off/on
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->enable_comfort_noise(false));
+ EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->enable_comfort_noise(true));
+ EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
+ // Set and get echo path
+ const size_t echo_path_size =
+ apm_->echo_control_mobile()->echo_path_size_bytes();
+ scoped_array<char> echo_path_in(new char[echo_path_size]);
+ scoped_array<char> echo_path_out(new char[echo_path_size]);
+ EXPECT_EQ(apm_->kNullPointerError,
+ apm_->echo_control_mobile()->SetEchoPath(NULL, echo_path_size));
+ EXPECT_EQ(apm_->kNullPointerError,
+ apm_->echo_control_mobile()->GetEchoPath(NULL, echo_path_size));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(), 1));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
+ echo_path_size));
+ for (size_t i = 0; i < echo_path_size; i++) {
+ echo_path_in[i] = echo_path_out[i] + 1;
+ }
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(), 1));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(),
+ echo_path_size));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
+ echo_path_size));
+ for (size_t i = 0; i < echo_path_size; i++) {
+ EXPECT_EQ(echo_path_in[i], echo_path_out[i]);
+ }
+ // Turn AECM off
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+ EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
+}
+
+TEST_F(ApmTest, GainControl) {
+ // Testing gain modes
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(-1)));
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(3)));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(
+ apm_->gain_control()->mode()));
+
+ GainControl::Mode mode[] = {
+ GainControl::kAdaptiveAnalog,
+ GainControl::kAdaptiveDigital,
+ GainControl::kFixedDigital
+ };
+ for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(mode[i]));
+ EXPECT_EQ(mode[i], apm_->gain_control()->mode());
+ }
+ // Testing invalid target levels
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_target_level_dbfs(-3));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_target_level_dbfs(-40));
+ // Testing valid target levels
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_target_level_dbfs(
+ apm_->gain_control()->target_level_dbfs()));
+
+ int level_dbfs[] = {0, 6, 31};
+ for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
+ EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
+ }
+
+ // Testing invalid compression gains
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_compression_gain_db(-1));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_compression_gain_db(100));
+
+ // Testing valid compression gains
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_compression_gain_db(
+ apm_->gain_control()->compression_gain_db()));
+
+ int gain_db[] = {0, 10, 90};
+ for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_compression_gain_db(gain_db[i]));
+ EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
+ }
+
+ // Testing limiter off/on
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
+ EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
+ EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
+
+ // Testing invalid level limits
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(-1, 512));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(100000, 512));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, -1));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, 100000));
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->gain_control()->set_analog_level_limits(512, 255));
+
+ // Testing valid level limits
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(
+ apm_->gain_control()->analog_level_minimum(),
+ apm_->gain_control()->analog_level_maximum()));
+
+ int min_level[] = {0, 255, 1024};
+ for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
+ EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
+ }
+
+ int max_level[] = {0, 1024, 65535};
+ for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
+ EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
+ }
+
+ // TODO(ajm): stream_is_saturated() and stream_analog_level()
+
+ // Turn AGC off
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
+ EXPECT_FALSE(apm_->gain_control()->is_enabled());
+}
+
+TEST_F(ApmTest, NoiseSuppression) {
+ // Tesing invalid suppression levels
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->noise_suppression()->set_level(
+ static_cast<NoiseSuppression::Level>(-1)));
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->noise_suppression()->set_level(
+ static_cast<NoiseSuppression::Level>(5)));
+
+ // Tesing valid suppression levels
+ NoiseSuppression::Level level[] = {
+ NoiseSuppression::kLow,
+ NoiseSuppression::kModerate,
+ NoiseSuppression::kHigh,
+ NoiseSuppression::kVeryHigh
+ };
+ for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->noise_suppression()->set_level(level[i]));
+ EXPECT_EQ(level[i], apm_->noise_suppression()->level());
+ }
+
+ // Turing NS on/off
+ EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
+ EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
+ EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
+}
+
+TEST_F(ApmTest, HighPassFilter) {
+ // Turing HP filter on/off
+ EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
+ EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
+ EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
+}
+
+TEST_F(ApmTest, LevelEstimator) {
+ // Turning level estimator on/off
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_FALSE(apm_->level_estimator()->is_enabled());
+
+ EXPECT_EQ(apm_->kNotEnabledError, apm_->level_estimator()->RMS());
+
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_TRUE(apm_->level_estimator()->is_enabled());
+
+ // Run this test in wideband; in super-wb, the splitting filter distorts the
+ // audio enough to cause deviation from the expectation for small values.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ frame_->_payloadDataLengthInSamples = 160;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 16000;
+
+ // Min value if no frames have been processed.
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+
+ // Min value on zero frames.
+ SetFrameTo(frame_, 0);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+
+ // Try a few RMS values.
+ // (These also test that the value resets after retrieving it.)
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(0, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 30000);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(1, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 10000);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(10, apm_->level_estimator()->RMS());
+
+ SetFrameTo(frame_, 10);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(70, apm_->level_estimator()->RMS());
+
+ // Min value if _energy == 0.
+ SetFrameTo(frame_, 10000);
+ uint32_t energy = frame_->_energy; // Save default to restore below.
+ frame_->_energy = 0;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(127, apm_->level_estimator()->RMS());
+ frame_->_energy = energy;
+
+ // Verify reset after enable/disable.
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ SetFrameTo(frame_, 1);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(90, apm_->level_estimator()->RMS());
+
+ // Verify reset after initialize.
+ SetFrameTo(frame_, 32767);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ SetFrameTo(frame_, 1);
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(90, apm_->level_estimator()->RMS());
+}
+
+TEST_F(ApmTest, VoiceDetection) {
+ // Test external VAD
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_stream_has_voice(true));
+ EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_stream_has_voice(false));
+ EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
+
+ // Tesing invalid likelihoods
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->voice_detection()->set_likelihood(
+ static_cast<VoiceDetection::Likelihood>(-1)));
+
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->voice_detection()->set_likelihood(
+ static_cast<VoiceDetection::Likelihood>(5)));
+
+ // Tesing valid likelihoods
+ VoiceDetection::Likelihood likelihood[] = {
+ VoiceDetection::kVeryLowLikelihood,
+ VoiceDetection::kLowLikelihood,
+ VoiceDetection::kModerateLikelihood,
+ VoiceDetection::kHighLikelihood
+ };
+ for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_likelihood(likelihood[i]));
+ EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
+ }
+
+ /* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
+ // Tesing invalid frame sizes
+ EXPECT_EQ(apm_->kBadParameterError,
+ apm_->voice_detection()->set_frame_size_ms(12));
+
+ // Tesing valid frame sizes
+ for (int i = 10; i <= 30; i += 10) {
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->set_frame_size_ms(i));
+ EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
+ }
+ */
+
+ // Turing VAD on/off
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_TRUE(apm_->voice_detection()->is_enabled());
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+ EXPECT_FALSE(apm_->voice_detection()->is_enabled());
+
+ // Test that AudioFrame activity is maintained when VAD is disabled.
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+ AudioFrame::VADActivity activity[] = {
+ AudioFrame::kVadActive,
+ AudioFrame::kVadPassive,
+ AudioFrame::kVadUnknown
+ };
+ for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) {
+ frame_->_vadActivity = activity[i];
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(activity[i], frame_->_vadActivity);
+ }
+
+ // Test that AudioFrame activity is set when VAD is enabled.
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ frame_->_vadActivity = AudioFrame::kVadUnknown;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_NE(AudioFrame::kVadUnknown, frame_->_vadActivity);
+
+ // TODO(bjornv): Add tests for streamed voice; stream_has_voice()
+}
+
+TEST_F(ApmTest, SplittingFilter) {
+ // Verify the filter is not active through undistorted audio when:
+ // 1. No components are enabled...
+ SetFrameTo(frame_, 1000);
+ AudioFrame frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+
+ // 2. Only the level estimator is enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+
+ // 3. Only VAD is enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+
+ // 4. Both VAD and the level estimator are enabled...
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+ EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
+ EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
+
+ // 5. Not using super-wb.
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ frame_->_payloadDataLengthInSamples = 160;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 16000;
+ // Enable AEC, which would require the filter in super-wb. We rely on the
+ // first few frames of data being unaffected by the AEC.
+ // TODO(andrew): This test, and the one below, rely rather tenuously on the
+ // behavior of the AEC. Think of something more robust.
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+
+ // Check the test is valid. We should have distortion from the filter
+ // when AEC is enabled (which won't affect the audio).
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
+ frame_->_payloadDataLengthInSamples = 320;
+ frame_->_audioChannel = 2;
+ frame_->_frequencyInHz = 32000;
+ SetFrameTo(frame_, 1000);
+ frame_copy = *frame_;
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
+}
+
+// TODO(andrew): expand test to verify output.
+TEST_F(ApmTest, DebugDump) {
+ const std::string filename = webrtc::test::OutputPath() + "debug.aec";
+ EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(NULL));
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ // Stopping without having started should be OK.
+ EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
+ EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+ EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+
+ // Verify the file has been written.
+ ASSERT_TRUE(fopen(filename.c_str(), "r") != NULL);
+ // Clean it up.
+ ASSERT_EQ(0, remove(filename.c_str()));
+#else
+ EXPECT_EQ(apm_->kUnsupportedFunctionError,
+ apm_->StartDebugRecording(filename.c_str()));
+ EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
+
+ // Verify the file has NOT been written.
+ ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+TEST_F(ApmTest, Process) {
+ GOOGLE_PROTOBUF_VERIFY_VERSION;
+ webrtc::audioproc::OutputData output_data;
+
+ if (!write_output_data) {
+ ReadMessageLiteFromFile(output_filename, &output_data);
+ } else {
+ // We don't have a file; add the required tests to the protobuf.
+ // TODO(ajm): vary the output channels as well?
+ const int channels[] = {1, 2};
+ const size_t channels_size = sizeof(channels) / sizeof(*channels);
+#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
+ // AECM doesn't support super-wb.
+ const int sample_rates[] = {8000, 16000};
+#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ const int sample_rates[] = {8000, 16000, 32000};
+#endif
+ const size_t sample_rates_size = sizeof(sample_rates) / sizeof(*sample_rates);
+ for (size_t i = 0; i < channels_size; i++) {
+ for (size_t j = 0; j < channels_size; j++) {
+ for (size_t k = 0; k < sample_rates_size; k++) {
+ webrtc::audioproc::Test* test = output_data.add_test();
+ test->set_num_reverse_channels(channels[i]);
+ test->set_num_input_channels(channels[j]);
+ test->set_num_output_channels(channels[j]);
+ test->set_sample_rate(sample_rates[k]);
+ }
+ }
+ }
+ }
+
+#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
+ EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
+ EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_drift_compensation(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_metrics(true));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->enable_delay_logging(true));
+ EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_analog_level_limits(0, 255));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+#endif
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->high_pass_filter()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->level_estimator()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->noise_suppression()->Enable(true));
+
+ EXPECT_EQ(apm_->kNoError,
+ apm_->voice_detection()->Enable(true));
+
+ for (int i = 0; i < output_data.test_size(); i++) {
+ printf("Running test %d of %d...\n", i + 1, output_data.test_size());
+
+ webrtc::audioproc::Test* test = output_data.mutable_test(i);
+ const int samples_per_channel = test->sample_rate() / 100;
+ revframe_->_payloadDataLengthInSamples = samples_per_channel;
+ revframe_->_audioChannel = test->num_reverse_channels();
+ revframe_->_frequencyInHz = test->sample_rate();
+ frame_->_payloadDataLengthInSamples = samples_per_channel;
+ frame_->_audioChannel = test->num_input_channels();
+ frame_->_frequencyInHz = test->sample_rate();
+
+ EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+ ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(test->sample_rate()));
+ ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(frame_->_audioChannel,
+ frame_->_audioChannel));
+ ASSERT_EQ(apm_->kNoError,
+ apm_->set_num_reverse_channels(revframe_->_audioChannel));
+
+ int frame_count = 0;
+ int has_echo_count = 0;
+ int has_voice_count = 0;
+ int is_saturated_count = 0;
+ int analog_level = 127;
+ int analog_level_average = 0;
+ int max_output_average = 0;
+
+ while (1) {
+ // Read far-end frame
+ const size_t frame_size = samples_per_channel * 2;
+ size_t read_count = fread(revframe_->_payloadData,
+ sizeof(int16_t),
+ frame_size,
+ far_file_);
+ if (read_count != frame_size) {
+ // Check that the file really ended.
+ ASSERT_NE(0, feof(far_file_));
+ break; // This is expected.
+ }
+
+ if (revframe_->_audioChannel == 1) {
+ MixStereoToMono(revframe_->_payloadData, revframe_->_payloadData,
+ samples_per_channel);
+ }
+
+ EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
+
+ EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->set_stream_drift_samples(0));
+ EXPECT_EQ(apm_->kNoError,
+ apm_->gain_control()->set_stream_analog_level(analog_level));
+
+ // Read near-end frame
+ read_count = fread(frame_->_payloadData,
+ sizeof(int16_t),
+ frame_size,
+ near_file_);
+ if (read_count != frame_size) {
+ // Check that the file really ended.
+ ASSERT_NE(0, feof(near_file_));
+ break; // This is expected.
+ }
+
+ if (frame_->_audioChannel == 1) {
+ MixStereoToMono(frame_->_payloadData, frame_->_payloadData,
+ samples_per_channel);
+ }
+ frame_->_vadActivity = AudioFrame::kVadUnknown;
+
+ EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
+
+ max_output_average += MaxAudioFrame(*frame_);
+
+ if (apm_->echo_cancellation()->stream_has_echo()) {
+ has_echo_count++;
+ }
+
+ analog_level = apm_->gain_control()->stream_analog_level();
+ analog_level_average += analog_level;
+ if (apm_->gain_control()->stream_is_saturated()) {
+ is_saturated_count++;
+ }
+ if (apm_->voice_detection()->stream_has_voice()) {
+ has_voice_count++;
+ EXPECT_EQ(AudioFrame::kVadActive, frame_->_vadActivity);
+ } else {
+ EXPECT_EQ(AudioFrame::kVadPassive, frame_->_vadActivity);
+ }
+
+ frame_count++;
+ }
+ max_output_average /= frame_count;
+ analog_level_average /= frame_count;
+
+#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ EchoCancellation::Metrics echo_metrics;
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->GetMetrics(&echo_metrics));
+ int median = 0;
+ int std = 0;
+ EXPECT_EQ(apm_->kNoError,
+ apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
+
+ int rms_level = apm_->level_estimator()->RMS();
+ EXPECT_LE(0, rms_level);
+ EXPECT_GE(127, rms_level);
+#endif
+
+ if (!write_output_data) {
+ EXPECT_EQ(test->has_echo_count(), has_echo_count);
+ EXPECT_EQ(test->has_voice_count(), has_voice_count);
+ EXPECT_EQ(test->is_saturated_count(), is_saturated_count);
+
+ EXPECT_EQ(test->analog_level_average(), analog_level_average);
+ EXPECT_EQ(test->max_output_average(), max_output_average);
+
+#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ webrtc::audioproc::Test::EchoMetrics reference =
+ test->echo_metrics();
+ TestStats(echo_metrics.residual_echo_return_loss,
+ reference.residual_echo_return_loss());
+ TestStats(echo_metrics.echo_return_loss,
+ reference.echo_return_loss());
+ TestStats(echo_metrics.echo_return_loss_enhancement,
+ reference.echo_return_loss_enhancement());
+ TestStats(echo_metrics.a_nlp,
+ reference.a_nlp());
+
+ webrtc::audioproc::Test::DelayMetrics reference_delay =
+ test->delay_metrics();
+ EXPECT_EQ(reference_delay.median(), median);
+ EXPECT_EQ(reference_delay.std(), std);
+
+ EXPECT_EQ(test->rms_level(), rms_level);
+#endif
+ } else {
+ test->set_has_echo_count(has_echo_count);
+ test->set_has_voice_count(has_voice_count);
+ test->set_is_saturated_count(is_saturated_count);
+
+ test->set_analog_level_average(analog_level_average);
+ test->set_max_output_average(max_output_average);
+
+#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
+ webrtc::audioproc::Test::EchoMetrics* message =
+ test->mutable_echo_metrics();
+ WriteStatsMessage(echo_metrics.residual_echo_return_loss,
+ message->mutable_residual_echo_return_loss());
+ WriteStatsMessage(echo_metrics.echo_return_loss,
+ message->mutable_echo_return_loss());
+ WriteStatsMessage(echo_metrics.echo_return_loss_enhancement,
+ message->mutable_echo_return_loss_enhancement());
+ WriteStatsMessage(echo_metrics.a_nlp,
+ message->mutable_a_nlp());
+
+ webrtc::audioproc::Test::DelayMetrics* message_delay =
+ test->mutable_delay_metrics();
+ message_delay->set_median(median);
+ message_delay->set_std(std);
+
+ test->set_rms_level(rms_level);
+#endif
+ }
+
+ rewind(far_file_);
+ rewind(near_file_);
+ }
+
+ if (write_output_data) {
+ WriteMessageLiteToFile(output_filename, output_data);
+ }
+}
+} // namespace
+
+int main(int argc, char** argv) {
+ ::testing::InitGoogleTest(&argc, argv);
+
+ for (int i = 1; i < argc; i++) {
+ if (strcmp(argv[i], "--write_output_data") == 0) {
+ write_output_data = true;
+ }
+ }
+
+ int err = RUN_ALL_TESTS();
+
+ // Optional, but removes memory leak noise from Valgrind.
+ google::protobuf::ShutdownProtobufLibrary();
+ return err;
+}
diff --git a/src/modules/audio_processing/test/unittest.proto b/src/modules/audio_processing/test/unittest.proto
new file mode 100644
index 0000000..67ba722
--- /dev/null
+++ b/src/modules/audio_processing/test/unittest.proto
@@ -0,0 +1,52 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+message Test {
+ optional int32 num_reverse_channels = 1;
+ optional int32 num_input_channels = 2;
+ optional int32 num_output_channels = 3;
+ optional int32 sample_rate = 4;
+
+ message Frame {
+ }
+
+ repeated Frame frame = 5;
+
+ optional int32 analog_level_average = 6;
+ optional int32 max_output_average = 7;
+
+ optional int32 has_echo_count = 8;
+ optional int32 has_voice_count = 9;
+ optional int32 is_saturated_count = 10;
+
+ message Statistic {
+ optional int32 instant = 1;
+ optional int32 average = 2;
+ optional int32 maximum = 3;
+ optional int32 minimum = 4;
+ }
+
+ message EchoMetrics {
+ optional Statistic residual_echo_return_loss = 1;
+ optional Statistic echo_return_loss = 2;
+ optional Statistic echo_return_loss_enhancement = 3;
+ optional Statistic a_nlp = 4;
+ }
+
+ optional EchoMetrics echo_metrics = 11;
+
+ message DelayMetrics {
+ optional int32 median = 1;
+ optional int32 std = 2;
+ }
+
+ optional DelayMetrics delay_metrics = 12;
+
+ optional int32 rms_level = 13;
+}
+
+message OutputData {
+ repeated Test test = 1;
+}
+
diff --git a/src/modules/audio_processing/test/unpack.cc b/src/modules/audio_processing/test/unpack.cc
new file mode 100644
index 0000000..2337131
--- /dev/null
+++ b/src/modules/audio_processing/test/unpack.cc
@@ -0,0 +1,216 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Commandline tool to unpack audioproc debug files.
+//
+// The debug files are dumped as protobuf blobs. For analysis, it's necessary
+// to unpack the file into its component parts: audio and other data.
+
+#include <stdio.h>
+
+#include "google/gflags.h"
+#include "scoped_ptr.h"
+#include "typedefs.h"
+#include "webrtc/audio_processing/debug.pb.h"
+
+using webrtc::scoped_array;
+
+using webrtc::audioproc::Event;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+using webrtc::audioproc::Init;
+
+// TODO(andrew): unpack more of the data.
+DEFINE_string(input_file, "input.pcm", "The name of the input stream file.");
+DEFINE_string(output_file, "ref_out.pcm",
+ "The name of the reference output stream file.");
+DEFINE_string(reverse_file, "reverse.pcm",
+ "The name of the reverse input stream file.");
+DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
+DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
+DEFINE_string(level_file, "level.int32", "The name of the level file.");
+DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
+DEFINE_bool(full, false,
+ "Unpack the full set of files (normally not needed).");
+
+// TODO(andrew): move this to a helper class to share with process_test.cc?
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file,
+ ::google::protobuf::MessageLite* msg) {
+ // The "wire format" for the size is little-endian.
+ // Assume process_test is running on a little-endian machine.
+ int32_t size = 0;
+ if (fread(&size, sizeof(int32_t), 1, file) != 1) {
+ return false;
+ }
+ if (size <= 0) {
+ return false;
+ }
+ const size_t usize = static_cast<size_t>(size);
+
+ scoped_array<char> array(new char[usize]);
+ if (fread(array.get(), sizeof(char), usize, file) != usize) {
+ return false;
+ }
+
+ msg->Clear();
+ return msg->ParseFromArray(array.get(), usize);
+}
+
+int main(int argc, char* argv[]) {
+ std::string program_name = argv[0];
+ std::string usage = "Commandline tool to unpack audioproc debug files.\n"
+ "Example usage:\n" + program_name + " debug_dump.pb\n";
+ google::SetUsageMessage(usage);
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ if (argc < 2) {
+ printf("%s", google::ProgramUsage());
+ return 1;
+ }
+
+ FILE* debug_file = fopen(argv[1], "rb");
+ if (debug_file == NULL) {
+ printf("Unable to open %s\n", argv[1]);
+ return 1;
+ }
+ FILE* input_file = fopen(FLAGS_input_file.c_str(), "wb");
+ if (input_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_input_file.c_str());
+ return 1;
+ }
+ FILE* output_file = fopen(FLAGS_output_file.c_str(), "wb");
+ if (output_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_output_file.c_str());
+ return 1;
+ }
+ FILE* reverse_file = fopen(FLAGS_reverse_file.c_str(), "wb");
+ if (reverse_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_reverse_file.c_str());
+ return 1;
+ }
+ FILE* settings_file = fopen(FLAGS_settings_file.c_str(), "wb");
+ if (settings_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_settings_file.c_str());
+ return 1;
+ }
+
+ FILE* delay_file = NULL;
+ FILE* drift_file = NULL;
+ FILE* level_file = NULL;
+ if (FLAGS_full) {
+ delay_file = fopen(FLAGS_delay_file.c_str(), "wb");
+ if (delay_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_delay_file.c_str());
+ return 1;
+ }
+ drift_file = fopen(FLAGS_drift_file.c_str(), "wb");
+ if (drift_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_drift_file.c_str());
+ return 1;
+ }
+ level_file = fopen(FLAGS_level_file.c_str(), "wb");
+ if (level_file == NULL) {
+ printf("Unable to open %s\n", FLAGS_level_file.c_str());
+ return 1;
+ }
+ }
+
+ Event event_msg;
+ int frame_count = 0;
+ while (ReadMessageFromFile(debug_file, &event_msg)) {
+ if (event_msg.type() == Event::REVERSE_STREAM) {
+ if (!event_msg.has_reverse_stream()) {
+ printf("Corrupted input file: ReverseStream missing.\n");
+ return 1;
+ }
+
+ const ReverseStream msg = event_msg.reverse_stream();
+ if (msg.has_data()) {
+ if (fwrite(msg.data().data(), msg.data().size(), 1, reverse_file) !=
+ 1) {
+ printf("Error when writing to %s\n", FLAGS_reverse_file.c_str());
+ return 1;
+ }
+ }
+ } else if (event_msg.type() == Event::STREAM) {
+ frame_count++;
+ if (!event_msg.has_stream()) {
+ printf("Corrupted input file: Stream missing.\n");
+ return 1;
+ }
+
+ const Stream msg = event_msg.stream();
+ if (msg.has_input_data()) {
+ if (fwrite(msg.input_data().data(), msg.input_data().size(), 1,
+ input_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_input_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_output_data()) {
+ if (fwrite(msg.output_data().data(), msg.output_data().size(), 1,
+ output_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_output_file.c_str());
+ return 1;
+ }
+ }
+
+ if (FLAGS_full) {
+ if (msg.has_delay()) {
+ int32_t delay = msg.delay();
+ if (fwrite(&delay, sizeof(int32_t), 1, delay_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_delay_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_drift()) {
+ int32_t drift = msg.drift();
+ if (fwrite(&drift, sizeof(int32_t), 1, drift_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_drift_file.c_str());
+ return 1;
+ }
+ }
+
+ if (msg.has_level()) {
+ int32_t level = msg.level();
+ if (fwrite(&level, sizeof(int32_t), 1, level_file) != 1) {
+ printf("Error when writing to %s\n", FLAGS_level_file.c_str());
+ return 1;
+ }
+ }
+ }
+ } else if (event_msg.type() == Event::INIT) {
+ if (!event_msg.has_init()) {
+ printf("Corrupted input file: Init missing.\n");
+ return 1;
+ }
+
+ const Init msg = event_msg.init();
+ // These should print out zeros if they're missing.
+ fprintf(settings_file, "Init at frame: %d\n", frame_count);
+ fprintf(settings_file, " Sample rate: %d\n", msg.sample_rate());
+ fprintf(settings_file, " Device sample rate: %d\n",
+ msg.device_sample_rate());
+ fprintf(settings_file, " Input channels: %d\n",
+ msg.num_input_channels());
+ fprintf(settings_file, " Output channels: %d\n",
+ msg.num_output_channels());
+ fprintf(settings_file, " Reverse channels: %d\n",
+ msg.num_reverse_channels());
+
+ fprintf(settings_file, "\n");
+ }
+ }
+
+ return 0;
+}
diff --git a/src/modules/audio_processing/utility/Android.mk b/src/modules/audio_processing/utility/Android.mk
index 7e758ce..bd3d039 100644
--- a/src/modules/audio_processing/utility/Android.mk
+++ b/src/modules/audio_processing/utility/Android.mk
@@ -10,40 +10,34 @@
include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../../../android-webrtc.mk
+
LOCAL_ARM_MODE := arm
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
LOCAL_MODULE := libwebrtc_apm_utility
LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := fft4g.c \
- ring_buffer.c
+LOCAL_SRC_FILES := \
+ fft4g.c \
+ ring_buffer.c \
+ delay_estimator.c \
+ delay_estimator_wrapper.c
# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
# Include paths placed before CFLAGS/CPPFLAGS
LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/../../.. \
+ $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
libdl \
libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
+ifndef NDK_ROOT
include external/stlport/libstlport.mk
+endif
include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/utility/delay_estimator.c b/src/modules/audio_processing/utility/delay_estimator.c
new file mode 100644
index 0000000..24ee74d
--- /dev/null
+++ b/src/modules/audio_processing/utility/delay_estimator.c
@@ -0,0 +1,319 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "delay_estimator.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+// Number of right shifts for scaling is linearly depending on number of bits in
+// the far-end binary spectrum.
+static const int kShiftsAtZero = 13; // Right shifts at zero binary spectrum.
+static const int kShiftsLinearSlope = 3;
+
+static const int32_t kProbabilityOffset = 1024; // 2 in Q9.
+static const int32_t kProbabilityLowerLimit = 8704; // 17 in Q9.
+static const int32_t kProbabilityMinSpread = 2816; // 5.5 in Q9.
+
+// Counts and returns number of bits of a 32-bit word.
+static int BitCount(uint32_t u32) {
+ uint32_t tmp = u32 - ((u32 >> 1) & 033333333333) -
+ ((u32 >> 2) & 011111111111);
+ tmp = ((tmp + (tmp >> 3)) & 030707070707);
+ tmp = (tmp + (tmp >> 6));
+ tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
+
+ return ((int) tmp);
+}
+
+// Compares the |binary_vector| with all rows of the |binary_matrix| and counts
+// per row the number of times they have the same value.
+//
+// Inputs:
+// - binary_vector : binary "vector" stored in a long
+// - binary_matrix : binary "matrix" stored as a vector of long
+// - matrix_size : size of binary "matrix"
+//
+// Output:
+// - bit_counts : "Vector" stored as a long, containing for each
+// row the number of times the matrix row and the
+// input vector have the same value
+//
+static void BitCountComparison(uint32_t binary_vector,
+ const uint32_t* binary_matrix,
+ int matrix_size,
+ int32_t* bit_counts) {
+ int n = 0;
+
+ // Compare |binary_vector| with all rows of the |binary_matrix|
+ for (; n < matrix_size; n++) {
+ bit_counts[n] = (int32_t) BitCount(binary_vector ^ binary_matrix[n]);
+ }
+}
+
+int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+
+ if (handle->mean_bit_counts != NULL) {
+ free(handle->mean_bit_counts);
+ handle->mean_bit_counts = NULL;
+ }
+ if (handle->bit_counts != NULL) {
+ free(handle->bit_counts);
+ handle->bit_counts = NULL;
+ }
+ if (handle->binary_far_history != NULL) {
+ free(handle->binary_far_history);
+ handle->binary_far_history = NULL;
+ }
+ if (handle->binary_near_history != NULL) {
+ free(handle->binary_near_history);
+ handle->binary_near_history = NULL;
+ }
+ if (handle->far_bit_counts != NULL) {
+ free(handle->far_bit_counts);
+ handle->far_bit_counts = NULL;
+ }
+
+ free(handle);
+
+ return 0;
+}
+
+int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
+ int max_delay,
+ int lookahead) {
+ BinaryDelayEstimator* self = NULL;
+ int history_size = max_delay + lookahead;
+
+ if (handle == NULL) {
+ return -1;
+ }
+ if (max_delay < 0) {
+ return -1;
+ }
+ if (lookahead < 0) {
+ return -1;
+ }
+ if (history_size < 2) {
+ // Must be this large for buffer shifting.
+ return -1;
+ }
+
+ self = malloc(sizeof(BinaryDelayEstimator));
+ *handle = self;
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->mean_bit_counts = NULL;
+ self->bit_counts = NULL;
+ self->binary_far_history = NULL;
+ self->far_bit_counts = NULL;
+
+ self->history_size = history_size;
+ self->near_history_size = lookahead + 1;
+
+ // Allocate memory for spectrum buffers.
+ self->mean_bit_counts = malloc(history_size * sizeof(int32_t));
+ if (self->mean_bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->bit_counts = malloc(history_size * sizeof(int32_t));
+ if (self->bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ // Allocate memory for history buffers.
+ self->binary_far_history = malloc(history_size * sizeof(uint32_t));
+ if (self->binary_far_history == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->binary_near_history = malloc(self->near_history_size *
+ sizeof(uint32_t));
+ if (self->binary_near_history == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->far_bit_counts = malloc(history_size * sizeof(int));
+ if (self->far_bit_counts == NULL) {
+ WebRtc_FreeBinaryDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+
+ return 0;
+}
+
+int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle) {
+ int i = 0;
+ assert(handle != NULL);
+
+ memset(handle->bit_counts, 0, sizeof(int32_t) * handle->history_size);
+ memset(handle->binary_far_history, 0,
+ sizeof(uint32_t) * handle->history_size);
+ memset(handle->binary_near_history, 0,
+ sizeof(uint32_t) * handle->near_history_size);
+ memset(handle->far_bit_counts, 0, sizeof(int) * handle->history_size);
+ for (i = 0; i < handle->history_size; ++i) {
+ handle->mean_bit_counts[i] = (20 << 9); // 20 in Q9.
+ }
+ handle->minimum_probability = (32 << 9); // 32 in Q9.
+ handle->last_delay_probability = (32 << 9); // 32 in Q9.
+
+ // Default return value if we're unable to estimate. -1 is used for errors.
+ handle->last_delay = -2;
+
+ return 0;
+}
+
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
+ uint32_t binary_far_spectrum,
+ uint32_t binary_near_spectrum) {
+ int i = 0;
+ int candidate_delay = -1;
+
+ int32_t value_best_candidate = 16384; // 32 in Q9, (max |mean_bit_counts|).
+ int32_t value_worst_candidate = 0;
+
+ assert(handle != NULL);
+ // Shift binary spectrum history and insert current |binary_far_spectrum|.
+ memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
+ (handle->history_size - 1) * sizeof(uint32_t));
+ handle->binary_far_history[0] = binary_far_spectrum;
+
+ // Shift history of far-end binary spectrum bit counts and insert bit count
+ // of current |binary_far_spectrum|.
+ memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
+ (handle->history_size - 1) * sizeof(int));
+ handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
+
+ if (handle->near_history_size > 1) {
+ // If we apply lookahead, shift near-end binary spectrum history. Insert
+ // current |binary_near_spectrum| and pull out the delayed one.
+ memmove(&(handle->binary_near_history[1]),
+ &(handle->binary_near_history[0]),
+ (handle->near_history_size - 1) * sizeof(uint32_t));
+ handle->binary_near_history[0] = binary_near_spectrum;
+ binary_near_spectrum =
+ handle->binary_near_history[handle->near_history_size - 1];
+ }
+
+ // Compare with delayed spectra and store the |bit_counts| for each delay.
+ BitCountComparison(binary_near_spectrum,
+ handle->binary_far_history,
+ handle->history_size,
+ handle->bit_counts);
+
+ // Update |mean_bit_counts|, which is the smoothed version of |bit_counts|.
+ for (i = 0; i < handle->history_size; i++) {
+ // |bit_counts| is constrained to [0, 32], meaning we can smooth with a
+ // factor up to 2^26. We use Q9.
+ int32_t bit_count = (handle->bit_counts[i] << 9); // Q9.
+
+ // Update |mean_bit_counts| only when far-end signal has something to
+ // contribute. If |far_bit_counts| is zero the far-end signal is weak and
+ // we likely have a poor echo condition, hence don't update.
+ if (handle->far_bit_counts[i] > 0) {
+ // Make number of right shifts piecewise linear w.r.t. |far_bit_counts|.
+ int shifts = kShiftsAtZero;
+ shifts -= (kShiftsLinearSlope * handle->far_bit_counts[i]) >> 4;
+ WebRtc_MeanEstimatorFix(bit_count, shifts, &(handle->mean_bit_counts[i]));
+ }
+ }
+
+ // Find |candidate_delay|, |value_best_candidate| and |value_worst_candidate|
+ // of |mean_bit_counts|.
+ for (i = 0; i < handle->history_size; i++) {
+ if (handle->mean_bit_counts[i] < value_best_candidate) {
+ value_best_candidate = handle->mean_bit_counts[i];
+ candidate_delay = i;
+ }
+ if (handle->mean_bit_counts[i] > value_worst_candidate) {
+ value_worst_candidate = handle->mean_bit_counts[i];
+ }
+ }
+
+ // The |value_best_candidate| is a good indicator on the probability of
+ // |candidate_delay| being an accurate delay (a small |value_best_candidate|
+ // means a good binary match). In the following sections we make a decision
+ // whether to update |last_delay| or not.
+ // 1) If the difference bit counts between the best and the worst delay
+ // candidates is too small we consider the situation to be unreliable and
+ // don't update |last_delay|.
+ // 2) If the situation is reliable we update |last_delay| if the value of the
+ // best candidate delay has a value less than
+ // i) an adaptive threshold |minimum_probability|, or
+ // ii) this corresponding value |last_delay_probability|, but updated at
+ // this time instant.
+
+ // Update |minimum_probability|.
+ if ((handle->minimum_probability > kProbabilityLowerLimit) &&
+ (value_worst_candidate - value_best_candidate > kProbabilityMinSpread)) {
+ // The "hard" threshold can't be lower than 17 (in Q9).
+ // The valley in the curve also has to be distinct, i.e., the
+ // difference between |value_worst_candidate| and |value_best_candidate| has
+ // to be large enough.
+ int32_t threshold = value_best_candidate + kProbabilityOffset;
+ if (threshold < kProbabilityLowerLimit) {
+ threshold = kProbabilityLowerLimit;
+ }
+ if (handle->minimum_probability > threshold) {
+ handle->minimum_probability = threshold;
+ }
+ }
+ // Update |last_delay_probability|.
+ // We use a Markov type model, i.e., a slowly increasing level over time.
+ handle->last_delay_probability++;
+ if (value_worst_candidate > value_best_candidate + kProbabilityOffset) {
+ // Reliable delay value for usage.
+ if (value_best_candidate < handle->minimum_probability) {
+ handle->last_delay = candidate_delay;
+ }
+ if (value_best_candidate < handle->last_delay_probability) {
+ handle->last_delay = candidate_delay;
+ // Reset |last_delay_probability|.
+ handle->last_delay_probability = value_best_candidate;
+ }
+ }
+
+ return handle->last_delay;
+}
+
+int WebRtc_binary_last_delay(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+ return handle->last_delay;
+}
+
+int WebRtc_history_size(BinaryDelayEstimator* handle) {
+ assert(handle != NULL);
+ return handle->history_size;
+}
+
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+ int factor,
+ int32_t* mean_value) {
+ int32_t diff = new_value - *mean_value;
+
+ // mean_new = mean_value + ((new_value - mean_value) >> factor);
+ if (diff < 0) {
+ diff = -((-diff) >> factor);
+ } else {
+ diff = (diff >> factor);
+ }
+ *mean_value += diff;
+}
diff --git a/src/modules/audio_processing/utility/delay_estimator.h b/src/modules/audio_processing/utility/delay_estimator.h
new file mode 100644
index 0000000..a376dfe
--- /dev/null
+++ b/src/modules/audio_processing/utility/delay_estimator.h
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on binary converted spectra.
+// The return value is 0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+
+#include "typedefs.h"
+
+typedef struct {
+ // Pointer to bit counts.
+ int32_t* mean_bit_counts;
+ int* far_bit_counts;
+
+ // Array only used locally in ProcessBinarySpectrum() but whose size is
+ // determined at run-time.
+ int32_t* bit_counts;
+
+ // Binary history variables.
+ uint32_t* binary_far_history;
+ uint32_t* binary_near_history;
+
+ // Delay estimation variables.
+ int32_t minimum_probability;
+ int last_delay_probability;
+
+ // Delay memory.
+ int last_delay;
+
+ // Buffer size.
+ int history_size;
+
+ // Near-end buffer size.
+ int near_history_size;
+} BinaryDelayEstimator;
+
+// Releases the memory allocated by WebRtc_CreateBinaryDelayEstimator(...).
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle);
+
+// Refer to WebRtc_CreateDelayEstimator() in delay_estimator_wrapper.h.
+int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
+ int max_delay,
+ int lookahead);
+
+// Initializes the delay estimation instance created with
+// WebRtc_CreateBinaryDelayEstimator(...).
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Output:
+// - handle : Initialized instance.
+//
+int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle);
+
+// Estimates and returns the delay between the binary far-end and binary near-
+// end spectra. The value will be offset by the lookahead (i.e. the lookahead
+// should be subtracted from the returned value).
+// Inputs:
+// - handle : Pointer to the delay estimation instance.
+// - binary_far_spectrum : Far-end binary spectrum.
+// - binary_near_spectrum : Near-end binary spectrum of the current block.
+//
+// Output:
+// - handle : Updated instance.
+//
+// Return value:
+// - delay : >= 0 - Calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
+ uint32_t binary_far_spectrum,
+ uint32_t binary_near_spectrum);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_ProcessBinarySpectrum(...).
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - delay : >= 0 - Last calculated delay value
+// -1 - Error
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_binary_last_delay(BinaryDelayEstimator* handle);
+
+// Returns the history size used in the far-end buffers to calculate the delay
+// over.
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - history_size : > 0 - Far-end history size.
+// -1 - Error.
+//
+int WebRtc_history_size(BinaryDelayEstimator* handle);
+
+// Updates the |mean_value| recursively with a step size of 2^-|factor|. This
+// function is used internally in the Binary Delay Estimator as well as the
+// Fixed point wrapper.
+//
+// Inputs:
+// - new_value : The new value the mean should be updated with.
+// - factor : The step size, in number of right shifts.
+//
+// Input/Output:
+// - mean_value : Pointer to the mean value.
+//
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+ int factor,
+ int32_t* mean_value);
+
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
diff --git a/src/modules/audio_processing/utility/delay_estimator_wrapper.c b/src/modules/audio_processing/utility/delay_estimator_wrapper.c
new file mode 100644
index 0000000..438c95f
--- /dev/null
+++ b/src/modules/audio_processing/utility/delay_estimator_wrapper.c
@@ -0,0 +1,336 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "delay_estimator_wrapper.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "delay_estimator.h"
+
+typedef union {
+ float float_;
+ int32_t int32_;
+} SpectrumType;
+
+typedef struct {
+ // Pointers to mean values of spectrum.
+ SpectrumType* mean_far_spectrum;
+ SpectrumType* mean_near_spectrum;
+ // |mean_*_spectrum| initialization indicator.
+ int far_spectrum_initialized;
+ int near_spectrum_initialized;
+
+ int spectrum_size;
+
+ // Binary spectrum based delay estimator
+ BinaryDelayEstimator* binary_handle;
+} DelayEstimator;
+
+// Only bit |kBandFirst| through bit |kBandLast| are processed and
+// |kBandFirst| - |kBandLast| must be < 32.
+static const int kBandFirst = 12;
+static const int kBandLast = 43;
+
+static __inline uint32_t SetBit(uint32_t in, int pos) {
+ uint32_t mask = (1 << pos);
+ uint32_t out = (in | mask);
+
+ return out;
+}
+
+// Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(),
+// but for float.
+//
+// Inputs:
+// - new_value : New additional value.
+// - scale : Scale for smoothing (should be less than 1.0).
+//
+// Input/Output:
+// - mean_value : Pointer to the mean value for updating.
+//
+static void MeanEstimatorFloat(float new_value,
+ float scale,
+ float* mean_value) {
+ assert(scale < 1.0f);
+ *mean_value += (new_value - *mean_value) * scale;
+}
+
+// Computes the binary spectrum by comparing the input |spectrum| with a
+// |threshold_spectrum|. Float and fixed point versions.
+//
+// Inputs:
+// - spectrum : Spectrum of which the binary spectrum should be
+// calculated.
+// - threshold_spectrum : Threshold spectrum with which the input
+// spectrum is compared.
+// Return:
+// - out : Binary spectrum.
+//
+static uint32_t BinarySpectrumFix(uint16_t* spectrum,
+ SpectrumType* threshold_spectrum,
+ int q_domain,
+ int* threshold_initialized) {
+ int i = kBandFirst;
+ uint32_t out = 0;
+
+ assert(q_domain < 16);
+
+ if (!(*threshold_initialized)) {
+ // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // value. This speeds up the convergence.
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ if (spectrum[i] > 0) {
+ // Convert input spectrum from Q(|q_domain|) to Q15.
+ int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
+ threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
+ *threshold_initialized = 1;
+ }
+ }
+ }
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ // Convert input spectrum from Q(|q_domain|) to Q15.
+ int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
+ // Update the |threshold_spectrum|.
+ WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
+ // Convert |spectrum| at current frequency bin to a binary value.
+ if (spectrum_q15 > threshold_spectrum[i].int32_) {
+ out = SetBit(out, i - kBandFirst);
+ }
+ }
+
+ return out;
+}
+
+static uint32_t BinarySpectrumFloat(float* spectrum,
+ SpectrumType* threshold_spectrum,
+ int* threshold_initialized) {
+ int i = kBandFirst;
+ uint32_t out = 0;
+ const float kScale = 1 / 64.0;
+
+ if (!(*threshold_initialized)) {
+ // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // value. This speeds up the convergence.
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ if (spectrum[i] > 0.0f) {
+ threshold_spectrum[i].float_ = (spectrum[i] / 2);
+ *threshold_initialized = 1;
+ }
+ }
+ }
+
+ for (i = kBandFirst; i <= kBandLast; i++) {
+ // Update the |threshold_spectrum|.
+ MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
+ // Convert |spectrum| at current frequency bin to a binary value.
+ if (spectrum[i] > threshold_spectrum[i].float_) {
+ out = SetBit(out, i - kBandFirst);
+ }
+ }
+
+ return out;
+}
+
+int WebRtc_FreeDelayEstimator(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ if (self->mean_far_spectrum != NULL) {
+ free(self->mean_far_spectrum);
+ self->mean_far_spectrum = NULL;
+ }
+ if (self->mean_near_spectrum != NULL) {
+ free(self->mean_near_spectrum);
+ self->mean_near_spectrum = NULL;
+ }
+
+ WebRtc_FreeBinaryDelayEstimator(self->binary_handle);
+
+ free(self);
+
+ return 0;
+}
+
+int WebRtc_CreateDelayEstimator(void** handle,
+ int spectrum_size,
+ int max_delay,
+ int lookahead) {
+ DelayEstimator* self = NULL;
+
+ // Check if the sub band used in the delay estimation is small enough to fit
+ // the binary spectra in a uint32_t.
+ assert(kBandLast - kBandFirst < 32);
+
+ if (handle == NULL) {
+ return -1;
+ }
+ if (spectrum_size < kBandLast) {
+ return -1;
+ }
+
+ self = malloc(sizeof(DelayEstimator));
+ *handle = self;
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->mean_far_spectrum = NULL;
+ self->mean_near_spectrum = NULL;
+
+ // Create binary delay estimator.
+ if (WebRtc_CreateBinaryDelayEstimator(&self->binary_handle,
+ max_delay,
+ lookahead) != 0) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ // Allocate memory for spectrum buffers.
+ self->mean_far_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
+ if (self->mean_far_spectrum == NULL) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+ self->mean_near_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
+ if (self->mean_near_spectrum == NULL) {
+ WebRtc_FreeDelayEstimator(self);
+ self = NULL;
+ return -1;
+ }
+
+ self->spectrum_size = spectrum_size;
+
+ return 0;
+}
+
+int WebRtc_InitDelayEstimator(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ // Initialize binary delay estimator.
+ if (WebRtc_InitBinaryDelayEstimator(self->binary_handle) != 0) {
+ return -1;
+ }
+ // Set averaged far and near end spectra to zero.
+ memset(self->mean_far_spectrum, 0,
+ sizeof(SpectrumType) * self->spectrum_size);
+ memset(self->mean_near_spectrum, 0,
+ sizeof(SpectrumType) * self->spectrum_size);
+ // Reset initialization indicators.
+ self->far_spectrum_initialized = 0;
+ self->near_spectrum_initialized = 0;
+
+ return 0;
+}
+
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+ uint16_t* far_spectrum,
+ uint16_t* near_spectrum,
+ int spectrum_size,
+ int far_q,
+ int near_q) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+ uint32_t binary_far_spectrum = 0;
+ uint32_t binary_near_spectrum = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (far_spectrum == NULL) {
+ // Empty far end spectrum.
+ return -1;
+ }
+ if (near_spectrum == NULL) {
+ // Empty near end spectrum.
+ return -1;
+ }
+ if (spectrum_size != self->spectrum_size) {
+ // Data sizes don't match.
+ return -1;
+ }
+ if (far_q > 15) {
+ // If |far_q| is larger than 15 we cannot guarantee no wrap around.
+ return -1;
+ }
+ if (near_q > 15) {
+ // If |near_q| is larger than 15 we cannot guarantee no wrap around.
+ return -1;
+ }
+
+ // Get binary spectra.
+ binary_far_spectrum = BinarySpectrumFix(far_spectrum,
+ self->mean_far_spectrum,
+ far_q,
+ &(self->far_spectrum_initialized));
+ binary_near_spectrum = BinarySpectrumFix(near_spectrum,
+ self->mean_near_spectrum,
+ near_q,
+ &(self->near_spectrum_initialized));
+
+ return WebRtc_ProcessBinarySpectrum(self->binary_handle,
+ binary_far_spectrum,
+ binary_near_spectrum);
+}
+
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+ float* far_spectrum,
+ float* near_spectrum,
+ int spectrum_size) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+ uint32_t binary_far_spectrum = 0;
+ uint32_t binary_near_spectrum = 0;
+
+ if (self == NULL) {
+ return -1;
+ }
+ if (far_spectrum == NULL) {
+ // Empty far end spectrum.
+ return -1;
+ }
+ if (near_spectrum == NULL) {
+ // Empty near end spectrum.
+ return -1;
+ }
+ if (spectrum_size != self->spectrum_size) {
+ // Data sizes don't match.
+ return -1;
+ }
+
+ // Get binary spectra.
+ binary_far_spectrum = BinarySpectrumFloat(far_spectrum,
+ self->mean_far_spectrum,
+ &(self->far_spectrum_initialized));
+ binary_near_spectrum = BinarySpectrumFloat(near_spectrum,
+ self->mean_near_spectrum,
+ &(self->near_spectrum_initialized));
+
+ return WebRtc_ProcessBinarySpectrum(self->binary_handle,
+ binary_far_spectrum,
+ binary_near_spectrum);
+}
+
+int WebRtc_last_delay(void* handle) {
+ DelayEstimator* self = (DelayEstimator*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ return WebRtc_binary_last_delay(self->binary_handle);
+}
diff --git a/src/modules/audio_processing/utility/delay_estimator_wrapper.h b/src/modules/audio_processing/utility/delay_estimator_wrapper.h
new file mode 100644
index 0000000..2a47b5d
--- /dev/null
+++ b/src/modules/audio_processing/utility/delay_estimator_wrapper.h
@@ -0,0 +1,110 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on block by block basis.
+// The return value is 0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+
+#include "typedefs.h"
+
+// Releases the memory allocated by WebRtc_CreateDelayEstimator(...)
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+int WebRtc_FreeDelayEstimator(void* handle);
+
+// Allocates the memory needed by the delay estimation. The memory needs to be
+// initialized separately through WebRtc_InitDelayEstimator(...).
+//
+// Inputs:
+// - handle : Instance that should be created.
+// - spectrum_size : Size of the spectrum used both in far-end and
+// near-end. Used to allocate memory for spectrum
+// specific buffers.
+// - max_delay : The maximum delay which can be estimated. Needed to
+// allocate memory for history buffers.
+// - lookahead : Amount of non-causal lookahead to use. This can
+// detect cases in which a near-end signal occurs before
+// the corresponding far-end signal. It will delay the
+// estimate for the current block by an equal amount,
+// and the returned values will be offset by it.
+//
+// A value of zero is the typical no-lookahead case.
+// This also represents the minimum delay which can be
+// estimated.
+//
+// Output:
+// - handle : Created instance.
+//
+int WebRtc_CreateDelayEstimator(void** handle,
+ int spectrum_size,
+ int max_delay,
+ int lookahead);
+
+// Initializes the delay estimation instance created with
+// WebRtc_CreateDelayEstimator(...)
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Output:
+// - handle : Initialized instance.
+//
+int WebRtc_InitDelayEstimator(void* handle);
+
+// Estimates and returns the delay between the far-end and near-end blocks. The
+// value will be offset by the lookahead (i.e. the lookahead should be
+// subtracted from the returned value).
+// Inputs:
+// - handle : Pointer to the delay estimation instance.
+// - far_spectrum : Pointer to the far-end spectrum data.
+// - near_spectrum : Pointer to the near-end spectrum data of the current
+// block.
+// - spectrum_size : The size of the data arrays (same for both far- and
+// near-end).
+// - far_q : The Q-domain of the far-end data.
+// - near_q : The Q-domain of the near-end data.
+//
+// Output:
+// - handle : Updated instance.
+//
+// Return value:
+// - delay : >= 0 - Calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+ uint16_t* far_spectrum,
+ uint16_t* near_spectrum,
+ int spectrum_size,
+ int far_q,
+ int near_q);
+
+// See WebRtc_DelayEstimatorProcessFix() for description.
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+ float* far_spectrum,
+ float* near_spectrum,
+ int spectrum_size);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_DelayEstimatorProcess(...).
+//
+// Input:
+// - handle : Pointer to the delay estimation instance.
+//
+// Return value:
+// - delay : >= 0 - Last calculated delay value.
+// -1 - Error.
+// -2 - Insufficient data for estimation.
+//
+int WebRtc_last_delay(void* handle);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
diff --git a/src/modules/audio_processing/utility/fft4g.c b/src/modules/audio_processing/utility/fft4g.c
index 9a84368..cbc4dc3 100644
--- a/src/modules/audio_processing/utility/fft4g.c
+++ b/src/modules/audio_processing/utility/fft4g.c
@@ -286,14 +286,24 @@
w[] and ip[] are compatible with all routines.
*/
-void cdft(int n, int isgn, float *a, int *ip, float *w)
-{
- void makewt(int nw, int *ip, float *w);
- void bitrv2(int n, int *ip, float *a);
- void bitrv2conj(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void cftbsub(int n, float *a, float *w);
+static void makewt(int nw, int *ip, float *w);
+static void makect(int nc, int *ip, float *c);
+static void bitrv2(int n, int *ip, float *a);
+static void bitrv2conj(int n, int *ip, float *a);
+static void cftfsub(int n, float *a, float *w);
+static void cftbsub(int n, float *a, float *w);
+static void cft1st(int n, float *a, float *w);
+static void cftmdl(int n, int l, float *a, float *w);
+static void rftfsub(int n, float *a, int nc, float *c);
+static void rftbsub(int n, float *a, int nc, float *c);
+#if 0 // Not used.
+static void dctsub(int n, float *a, int nc, float *c)
+static void dstsub(int n, float *a, int nc, float *c)
+#endif
+
+void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w)
+{
if (n > (ip[0] << 2)) {
makewt(n >> 2, ip, w);
}
@@ -311,15 +321,8 @@
}
-void rdft(int n, int isgn, float *a, int *ip, float *w)
+void WebRtc_rdft(int n, int isgn, float *a, int *ip, float *w)
{
- void makewt(int nw, int *ip, float *w);
- void makect(int nc, int *ip, float *c);
- void bitrv2(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void cftbsub(int n, float *a, float *w);
- void rftfsub(int n, float *a, int nc, float *c);
- void rftbsub(int n, float *a, int nc, float *c);
int nw, nc;
float xi;
@@ -357,17 +360,9 @@
}
}
-
-void ddct(int n, int isgn, float *a, int *ip, float *w)
+#if 0 // Not used.
+static void ddct(int n, int isgn, float *a, int *ip, float *w)
{
- void makewt(int nw, int *ip, float *w);
- void makect(int nc, int *ip, float *c);
- void bitrv2(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void cftbsub(int n, float *a, float *w);
- void rftfsub(int n, float *a, int nc, float *c);
- void rftbsub(int n, float *a, int nc, float *c);
- void dctsub(int n, float *a, int nc, float *c);
int j, nw, nc;
float xr;
@@ -417,16 +412,8 @@
}
-void ddst(int n, int isgn, float *a, int *ip, float *w)
+static void ddst(int n, int isgn, float *a, int *ip, float *w)
{
- void makewt(int nw, int *ip, float *w);
- void makect(int nc, int *ip, float *c);
- void bitrv2(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void cftbsub(int n, float *a, float *w);
- void rftfsub(int n, float *a, int nc, float *c);
- void rftbsub(int n, float *a, int nc, float *c);
- void dstsub(int n, float *a, int nc, float *c);
int j, nw, nc;
float xr;
@@ -476,14 +463,8 @@
}
-void dfct(int n, float *a, float *t, int *ip, float *w)
+static void dfct(int n, float *a, float *t, int *ip, float *w)
{
- void makewt(int nw, int *ip, float *w);
- void makect(int nc, int *ip, float *c);
- void bitrv2(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void rftfsub(int n, float *a, int nc, float *c);
- void dctsub(int n, float *a, int nc, float *c);
int j, k, l, m, mh, nw, nc;
float xr, xi, yr, yi;
@@ -571,15 +552,8 @@
}
}
-
-void dfst(int n, float *a, float *t, int *ip, float *w)
+static void dfst(int n, float *a, float *t, int *ip, float *w)
{
- void makewt(int nw, int *ip, float *w);
- void makect(int nc, int *ip, float *c);
- void bitrv2(int n, int *ip, float *a);
- void cftfsub(int n, float *a, float *w);
- void rftfsub(int n, float *a, int nc, float *c);
- void dstsub(int n, float *a, int nc, float *c);
int j, k, l, m, mh, nw, nc;
float xr, xi, yr, yi;
@@ -657,6 +631,7 @@
}
a[0] = 0;
}
+#endif // Not used.
/* -------- initializing routines -------- */
@@ -664,9 +639,8 @@
#include <math.h>
-void makewt(int nw, int *ip, float *w)
+static void makewt(int nw, int *ip, float *w)
{
- void bitrv2(int n, int *ip, float *a);
int j, nwh;
float delta, x, y;
@@ -694,7 +668,7 @@
}
-void makect(int nc, int *ip, float *c)
+static void makect(int nc, int *ip, float *c)
{
int j, nch;
float delta;
@@ -716,7 +690,7 @@
/* -------- child routines -------- */
-void bitrv2(int n, int *ip, float *a)
+static void bitrv2(int n, int *ip, float *a)
{
int j, j1, k, k1, l, m, m2;
float xr, xi, yr, yi;
@@ -816,7 +790,7 @@
}
-void bitrv2conj(int n, int *ip, float *a)
+static void bitrv2conj(int n, int *ip, float *a)
{
int j, j1, k, k1, l, m, m2;
float xr, xi, yr, yi;
@@ -925,10 +899,8 @@
}
-void cftfsub(int n, float *a, float *w)
+static void cftfsub(int n, float *a, float *w)
{
- void cft1st(int n, float *a, float *w);
- void cftmdl(int n, int l, float *a, float *w);
int j, j1, j2, j3, l;
float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
@@ -977,10 +949,8 @@
}
-void cftbsub(int n, float *a, float *w)
+static void cftbsub(int n, float *a, float *w)
{
- void cft1st(int n, float *a, float *w);
- void cftmdl(int n, int l, float *a, float *w);
int j, j1, j2, j3, l;
float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
@@ -1029,7 +999,7 @@
}
-void cft1st(int n, float *a, float *w)
+static void cft1st(int n, float *a, float *w)
{
int j, k1, k2;
float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
@@ -1134,7 +1104,7 @@
}
-void cftmdl(int n, int l, float *a, float *w)
+static void cftmdl(int n, int l, float *a, float *w)
{
int j, j1, j2, j3, k, k1, k2, m, m2;
float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
@@ -1261,7 +1231,7 @@
}
-void rftfsub(int n, float *a, int nc, float *c)
+static void rftfsub(int n, float *a, int nc, float *c)
{
int j, k, kk, ks, m;
float wkr, wki, xr, xi, yr, yi;
@@ -1286,7 +1256,7 @@
}
-void rftbsub(int n, float *a, int nc, float *c)
+static void rftbsub(int n, float *a, int nc, float *c)
{
int j, k, kk, ks, m;
float wkr, wki, xr, xi, yr, yi;
@@ -1312,8 +1282,8 @@
a[m + 1] = -a[m + 1];
}
-
-void dctsub(int n, float *a, int nc, float *c)
+#if 0 // Not used.
+static void dctsub(int n, float *a, int nc, float *c)
{
int j, k, kk, ks, m;
float wkr, wki, xr;
@@ -1334,7 +1304,7 @@
}
-void dstsub(int n, float *a, int nc, float *c)
+static void dstsub(int n, float *a, int nc, float *c)
{
int j, k, kk, ks, m;
float wkr, wki, xr;
@@ -1353,4 +1323,4 @@
}
a[m] *= c[0];
}
-
+#endif // Not used.
diff --git a/src/modules/audio_processing/utility/fft4g.h b/src/modules/audio_processing/utility/fft4g.h
index 373ff14..14a52a1 100644
--- a/src/modules/audio_processing/utility/fft4g.h
+++ b/src/modules/audio_processing/utility/fft4g.h
@@ -11,8 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
-void rdft(int, int, float *, int *, float *);
-void cdft(int, int, float *, int *, float *);
+void WebRtc_rdft(int, int, float *, int *, float *);
+void WebRtc_cdft(int, int, float *, int *, float *);
#endif
-
diff --git a/src/modules/audio_processing/utility/ring_buffer.c b/src/modules/audio_processing/utility/ring_buffer.c
index ea2e354..8b2b436 100644
--- a/src/modules/audio_processing/utility/ring_buffer.c
+++ b/src/modules/audio_processing/utility/ring_buffer.c
@@ -8,232 +8,264 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/*
- * Provides a generic ring buffer that can be written to and read from with
- * arbitrarily sized blocks. The AEC uses this for several different tasks.
- */
+// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
+// otherwise specified, functions return 0 on success and -1 on error.
-#include <stdlib.h>
-#include <string.h>
#include "ring_buffer.h"
+#include <stddef.h> // size_t
+#include <stdlib.h>
+#include <string.h>
+
+enum Wrap {
+ SAME_WRAP,
+ DIFF_WRAP
+};
+
typedef struct {
- int readPos;
- int writePos;
- int size;
- char rwWrap;
- bufdata_t *data;
+ size_t read_pos;
+ size_t write_pos;
+ size_t element_count;
+ size_t element_size;
+ enum Wrap rw_wrap;
+ char* data;
} buf_t;
-enum {SAME_WRAP, DIFF_WRAP};
+// Get address of region(s) from which we can read data.
+// If the region is contiguous, |data_ptr_bytes_2| will be zero.
+// If non-contiguous, |data_ptr_bytes_2| will be the size in bytes of the second
+// region. Returns room available to be read or |element_count|, whichever is
+// smaller.
+static size_t GetBufferReadRegions(buf_t* buf,
+ size_t element_count,
+ void** data_ptr_1,
+ size_t* data_ptr_bytes_1,
+ void** data_ptr_2,
+ size_t* data_ptr_bytes_2) {
-int WebRtcApm_CreateBuffer(void **bufInst, int size)
-{
- buf_t *buf = NULL;
+ const size_t readable_elements = WebRtc_available_read(buf);
+ const size_t read_elements = (readable_elements < element_count ?
+ readable_elements : element_count);
+ const size_t margin = buf->element_count - buf->read_pos;
- if (size < 0) {
- return -1;
- }
+ // Check to see if read is not contiguous.
+ if (read_elements > margin) {
+ // Write data in two blocks that wrap the buffer.
+ *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
+ *data_ptr_bytes_1 = margin * buf->element_size;
+ *data_ptr_2 = buf->data;
+ *data_ptr_bytes_2 = (read_elements - margin) * buf->element_size;
+ } else {
+ *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
+ *data_ptr_bytes_1 = read_elements * buf->element_size;
+ *data_ptr_2 = NULL;
+ *data_ptr_bytes_2 = 0;
+ }
- buf = malloc(sizeof(buf_t));
- *bufInst = buf;
- if (buf == NULL) {
- return -1;
- }
+ return read_elements;
+}
- buf->data = malloc(size*sizeof(bufdata_t));
- if (buf->data == NULL) {
- free(buf);
- buf = NULL;
- return -1;
- }
+int WebRtc_CreateBuffer(void** handle,
+ size_t element_count,
+ size_t element_size) {
+ buf_t* self = NULL;
- buf->size = size;
+ if (handle == NULL) {
+ return -1;
+ }
+
+ self = malloc(sizeof(buf_t));
+ if (self == NULL) {
+ return -1;
+ }
+ *handle = self;
+
+ self->data = malloc(element_count * element_size);
+ if (self->data == NULL) {
+ free(self);
+ self = NULL;
+ return -1;
+ }
+
+ self->element_count = element_count;
+ self->element_size = element_size;
+
+ return 0;
+}
+
+int WebRtc_InitBuffer(void* handle) {
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ self->read_pos = 0;
+ self->write_pos = 0;
+ self->rw_wrap = SAME_WRAP;
+
+ // Initialize buffer to zeros
+ memset(self->data, 0, self->element_count * self->element_size);
+
+ return 0;
+}
+
+int WebRtc_FreeBuffer(void* handle) {
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return -1;
+ }
+
+ free(self->data);
+ free(self);
+
+ return 0;
+}
+
+size_t WebRtc_ReadBuffer(void* handle,
+ void** data_ptr,
+ void* data,
+ size_t element_count) {
+
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
return 0;
-}
-
-int WebRtcApm_InitBuffer(void *bufInst)
-{
- buf_t *buf = (buf_t*)bufInst;
-
- buf->readPos = 0;
- buf->writePos = 0;
- buf->rwWrap = SAME_WRAP;
-
- // Initialize buffer to zeros
- memset(buf->data, 0, sizeof(bufdata_t)*buf->size);
-
+ }
+ if (data == NULL) {
return 0;
-}
-
-int WebRtcApm_FreeBuffer(void *bufInst)
-{
- buf_t *buf = (buf_t*)bufInst;
-
- if (buf == NULL) {
- return -1;
- }
-
- free(buf->data);
- free(buf);
-
+ }
+ if (data_ptr == NULL) {
return 0;
+ }
+
+ {
+ void* buf_ptr_1 = NULL;
+ void* buf_ptr_2 = NULL;
+ size_t buf_ptr_bytes_1 = 0;
+ size_t buf_ptr_bytes_2 = 0;
+ const size_t read_count = GetBufferReadRegions(self,
+ element_count,
+ &buf_ptr_1,
+ &buf_ptr_bytes_1,
+ &buf_ptr_2,
+ &buf_ptr_bytes_2);
+
+ if (buf_ptr_bytes_2 > 0) {
+ // We have a wrap around when reading the buffer. Copy the buffer data to
+ // |data| and point to it.
+ memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
+ memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
+ *data_ptr = data;
+ } else {
+ *data_ptr = buf_ptr_1;
+ }
+
+ // Update read position
+ WebRtc_MoveReadPtr(handle, (int) read_count);
+
+ return read_count;
+ }
}
-int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size)
-{
- buf_t *buf = (buf_t*)bufInst;
- int n = 0, margin = 0;
+size_t WebRtc_WriteBuffer(void* handle,
+ const void* data,
+ size_t element_count) {
- if (size <= 0 || size > buf->size) {
- return -1;
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+ if (data == NULL) {
+ return 0;
+ }
+
+ {
+ const size_t free_elements = WebRtc_available_write(handle);
+ const size_t write_elements = (free_elements < element_count ? free_elements
+ : element_count);
+ size_t n = write_elements;
+ const size_t margin = self->element_count - self->write_pos;
+
+ if (write_elements > margin) {
+ // Buffer wrap around when writing.
+ memcpy(self->data + self->write_pos * self->element_size,
+ data, margin * self->element_size);
+ self->write_pos = 0;
+ n -= margin;
+ self->rw_wrap = DIFF_WRAP;
}
+ memcpy(self->data + self->write_pos * self->element_size,
+ ((const char*) data) + ((write_elements - n) * self->element_size),
+ n * self->element_size);
+ self->write_pos += n;
- n = size;
- if (buf->rwWrap == DIFF_WRAP) {
- margin = buf->size - buf->readPos;
- if (n > margin) {
- buf->rwWrap = SAME_WRAP;
- memcpy(data, buf->data + buf->readPos,
- sizeof(bufdata_t)*margin);
- buf->readPos = 0;
- n = size - margin;
- }
- else {
- memcpy(data, buf->data + buf->readPos,
- sizeof(bufdata_t)*n);
- buf->readPos += n;
- return n;
- }
- }
-
- if (buf->rwWrap == SAME_WRAP) {
- margin = buf->writePos - buf->readPos;
- if (margin > n)
- margin = n;
- memcpy(data + size - n, buf->data + buf->readPos,
- sizeof(bufdata_t)*margin);
- buf->readPos += margin;
- n -= margin;
- }
-
- return size - n;
+ return write_elements;
+ }
}
-int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size)
-{
- buf_t *buf = (buf_t*)bufInst;
- int n = 0, margin = 0;
+int WebRtc_MoveReadPtr(void* handle, int element_count) {
- if (size < 0 || size > buf->size) {
- return -1;
+ buf_t* self = (buf_t*) handle;
+
+ if (self == NULL) {
+ return 0;
+ }
+
+ {
+ // We need to be able to take care of negative changes, hence use "int"
+ // instead of "size_t".
+ const int free_elements = (int) WebRtc_available_write(handle);
+ const int readable_elements = (int) WebRtc_available_read(handle);
+ int read_pos = (int) self->read_pos;
+
+ if (element_count > readable_elements) {
+ element_count = readable_elements;
+ }
+ if (element_count < -free_elements) {
+ element_count = -free_elements;
}
- n = size;
- if (buf->rwWrap == SAME_WRAP) {
- margin = buf->size - buf->writePos;
- if (n > margin) {
- buf->rwWrap = DIFF_WRAP;
- memcpy(buf->data + buf->writePos, data,
- sizeof(bufdata_t)*margin);
- buf->writePos = 0;
- n = size - margin;
- }
- else {
- memcpy(buf->data + buf->writePos, data,
- sizeof(bufdata_t)*n);
- buf->writePos += n;
- return n;
- }
+ read_pos += element_count;
+ if (read_pos > (int) self->element_count) {
+ // Buffer wrap around. Restart read position and wrap indicator.
+ read_pos -= (int) self->element_count;
+ self->rw_wrap = SAME_WRAP;
+ }
+ if (read_pos < 0) {
+ // Buffer wrap around. Restart read position and wrap indicator.
+ read_pos += (int) self->element_count;
+ self->rw_wrap = DIFF_WRAP;
}
- if (buf->rwWrap == DIFF_WRAP) {
- margin = buf->readPos - buf->writePos;
- if (margin > n)
- margin = n;
- memcpy(buf->data + buf->writePos, data + size - n,
- sizeof(bufdata_t)*margin);
- buf->writePos += margin;
- n -= margin;
- }
+ self->read_pos = (size_t) read_pos;
- return size - n;
+ return element_count;
+ }
}
-int WebRtcApm_FlushBuffer(void *bufInst, int size)
-{
- buf_t *buf = (buf_t*)bufInst;
- int n = 0, margin = 0;
+size_t WebRtc_available_read(const void* handle) {
+ const buf_t* self = (buf_t*) handle;
- if (size <= 0 || size > buf->size) {
- return -1;
- }
+ if (self == NULL) {
+ return 0;
+ }
- n = size;
- if (buf->rwWrap == DIFF_WRAP) {
- margin = buf->size - buf->readPos;
- if (n > margin) {
- buf->rwWrap = SAME_WRAP;
- buf->readPos = 0;
- n = size - margin;
- }
- else {
- buf->readPos += n;
- return n;
- }
- }
-
- if (buf->rwWrap == SAME_WRAP) {
- margin = buf->writePos - buf->readPos;
- if (margin > n)
- margin = n;
- buf->readPos += margin;
- n -= margin;
- }
-
- return size - n;
+ if (self->rw_wrap == SAME_WRAP) {
+ return self->write_pos - self->read_pos;
+ } else {
+ return self->element_count - self->read_pos + self->write_pos;
+ }
}
-int WebRtcApm_StuffBuffer(void *bufInst, int size)
-{
- buf_t *buf = (buf_t*)bufInst;
- int n = 0, margin = 0;
+size_t WebRtc_available_write(const void* handle) {
+ const buf_t* self = (buf_t*) handle;
- if (size <= 0 || size > buf->size) {
- return -1;
- }
+ if (self == NULL) {
+ return 0;
+ }
- n = size;
- if (buf->rwWrap == SAME_WRAP) {
- margin = buf->readPos;
- if (n > margin) {
- buf->rwWrap = DIFF_WRAP;
- buf->readPos = buf->size - 1;
- n -= margin + 1;
- }
- else {
- buf->readPos -= n;
- return n;
- }
- }
-
- if (buf->rwWrap == DIFF_WRAP) {
- margin = buf->readPos - buf->writePos;
- if (margin > n)
- margin = n;
- buf->readPos -= margin;
- n -= margin;
- }
-
- return size - n;
-}
-
-int WebRtcApm_get_buffer_size(const void *bufInst)
-{
- const buf_t *buf = (buf_t*)bufInst;
-
- if (buf->rwWrap == SAME_WRAP)
- return buf->writePos - buf->readPos;
- else
- return buf->size - buf->readPos + buf->writePos;
+ return self->element_count - WebRtc_available_read(handle);
}
diff --git a/src/modules/audio_processing/utility/ring_buffer.h b/src/modules/audio_processing/utility/ring_buffer.h
index 0fd261d..3c44029 100644
--- a/src/modules/audio_processing/utility/ring_buffer.h
+++ b/src/modules/audio_processing/utility/ring_buffer.h
@@ -8,34 +8,46 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/*
- * Specifies the interface for the AEC generic buffer.
- */
+// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
+// otherwise specified, functions return 0 on success and -1 on error.
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-// Determines buffer datatype
-typedef short bufdata_t;
+#include <stddef.h> // size_t
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcApm_CreateBuffer(void **bufInst, int size);
-int WebRtcApm_InitBuffer(void *bufInst);
-int WebRtcApm_FreeBuffer(void *bufInst);
+int WebRtc_CreateBuffer(void** handle,
+ size_t element_count,
+ size_t element_size);
+int WebRtc_InitBuffer(void* handle);
+int WebRtc_FreeBuffer(void* handle);
-// Returns number of samples read
-int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size);
+// Reads data from the buffer. The |data_ptr| will point to the address where
+// it is located. If all |element_count| data are feasible to read without
+// buffer wrap around |data_ptr| will point to the location in the buffer.
+// Otherwise, the data will be copied to |data| (memory allocation done by the
+// user) and |data_ptr| points to the address of |data|. |data_ptr| is only
+// guaranteed to be valid until the next call to WebRtc_WriteBuffer().
+// Returns number of elements read.
+size_t WebRtc_ReadBuffer(void* handle,
+ void** data_ptr,
+ void* data,
+ size_t element_count);
-// Returns number of samples written
-int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size);
+// Writes |data| to buffer and returns the number of elements written.
+size_t WebRtc_WriteBuffer(void* handle, const void* data, size_t element_count);
-// Returns number of samples flushed
-int WebRtcApm_FlushBuffer(void *bufInst, int size);
+// Moves the buffer read position and returns the number of elements moved.
+// Positive |element_count| moves the read position towards the write position,
+// that is, flushing the buffer. Negative |element_count| moves the read
+// position away from the the write position, that is, stuffing the buffer.
+// Returns number of elements moved.
+int WebRtc_MoveReadPtr(void* handle, int element_count);
-// Returns number of samples stuffed
-int WebRtcApm_StuffBuffer(void *bufInst, int size);
+// Returns number of available elements to read.
+size_t WebRtc_available_read(const void* handle);
-// Returns number of samples in buffer
-int WebRtcApm_get_buffer_size(const void *bufInst);
+// Returns number of available elements for write.
+size_t WebRtc_available_write(const void* handle);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
diff --git a/src/modules/audio_processing/utility/util.gyp b/src/modules/audio_processing/utility/util.gypi
similarity index 77%
rename from src/modules/audio_processing/utility/util.gyp
rename to src/modules/audio_processing/utility/util.gypi
index 3348da8..3c3024a 100644
--- a/src/modules/audio_processing/utility/util.gyp
+++ b/src/modules/audio_processing/utility/util.gypi
@@ -7,23 +7,27 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../../common_settings.gypi',
- ],
'targets': [
{
'target_name': 'apm_util',
'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+ ],
'direct_dependent_settings': {
'include_dirs': [
'.',
],
},
'sources': [
- 'ring_buffer.c',
- 'ring_buffer.h',
+ 'delay_estimator.c',
+ 'delay_estimator.h',
+ 'delay_estimator_wrapper.c',
+ 'delay_estimator_wrapper.h',
'fft4g.c',
'fft4g.h',
+ 'ring_buffer.c',
+ 'ring_buffer.h',
],
},
],
diff --git a/src/modules/audio_processing/main/source/voice_detection_impl.cc b/src/modules/audio_processing/voice_detection_impl.cc
similarity index 92%
rename from src/modules/audio_processing/main/source/voice_detection_impl.cc
rename to src/modules/audio_processing/voice_detection_impl.cc
index 3eb446e..49aac2e 100644
--- a/src/modules/audio_processing/main/source/voice_detection_impl.cc
+++ b/src/modules/audio_processing/voice_detection_impl.cc
@@ -74,16 +74,16 @@
// TODO(ajm): concatenate data in frame buffer here.
- int vad_ret_val;
- vad_ret_val = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
- apm_->split_sample_rate_hz(),
- mixed_data,
- frame_size_samples_);
-
- if (vad_ret_val == 0) {
+ int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
+ apm_->split_sample_rate_hz(),
+ mixed_data,
+ frame_size_samples_);
+ if (vad_ret == 0) {
stream_has_voice_ = false;
- } else if (vad_ret_val == 1) {
+ audio->set_activity(AudioFrame::kVadPassive);
+ } else if (vad_ret == 1) {
stream_has_voice_ = true;
+ audio->set_activity(AudioFrame::kVadActive);
} else {
return apm_->kUnspecifiedError;
}
diff --git a/src/modules/audio_processing/main/source/voice_detection_impl.h b/src/modules/audio_processing/voice_detection_impl.h
similarity index 100%
rename from src/modules/audio_processing/main/source/voice_detection_impl.h
rename to src/modules/audio_processing/voice_detection_impl.h
diff --git a/src/modules/interface/module.h b/src/modules/interface/module.h
index f270978..a274d95 100644
--- a/src/modules/interface/module.h
+++ b/src/modules/interface/module.h
@@ -1,33 +1,65 @@
-#ifndef MODULE_H
-#define MODULE_H
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_INTERFACE_MODULE_H_
+#define MODULES_INTERFACE_MODULE_H_
+
+#include <assert.h>
#include "typedefs.h"
-namespace webrtc
-{
+namespace webrtc {
-class Module
-{
-public:
- // Returns version of the module and its components.
- virtual int32_t Version(char* version,
- uint32_t& remainingBufferInBytes,
- uint32_t& position) const = 0;
+class Module {
+ public:
+ // Change the unique identifier of this object.
+ virtual int32_t ChangeUniqueId(const int32_t id) = 0;
- // Change the unique identifier of this object.
- virtual int32_t ChangeUniqueId(const int32_t id) = 0;
+ // Returns the number of milliseconds until the module want a worker
+ // thread to call Process.
+ virtual int32_t TimeUntilNextProcess() = 0;
- // Returns the number of milliseconds until the module want a worker
- // thread to call Process.
- virtual int32_t TimeUntilNextProcess() = 0 ;
+ // Process any pending tasks such as timeouts.
+ virtual int32_t Process() = 0;
- // Process any pending tasks such as timeouts.
- virtual int32_t Process() = 0 ;
-
-protected:
- virtual ~Module() {}
+ protected:
+ virtual ~Module() {}
};
-} // namespace webrtc
+// Reference counted version of the module interface.
+class RefCountedModule : public Module {
+ public:
+ // Increase the reference count by one.
+ // Returns the incremented reference count.
+ // TODO(perkj): Make this pure virtual when Chromium have implemented
+ // reference counting ADM and Video capture module.
+ virtual int32_t AddRef() {
+ assert(!"Not implemented.");
+ return 1;
+ }
-#endif // MODULE_H
+ // Decrease the reference count by one.
+ // Returns the decreased reference count.
+ // Returns 0 if the last reference was just released.
+ // When the reference count reach 0 the object will self-destruct.
+ // TODO(perkj): Make this pure virtual when Chromium have implemented
+ // reference counting ADM and Video capture module.
+ virtual int32_t Release() {
+ assert(!"Not implemented.");
+ return 1;
+ }
+
+ protected:
+ virtual ~RefCountedModule() {}
+};
+
+} // namespace webrtc
+
+#endif // MODULES_INTERFACE_MODULE_H_
diff --git a/src/modules/interface/module_common_types.h b/src/modules/interface/module_common_types.h
index 1865b0f..0319dfe 100644
--- a/src/modules/interface/module_common_types.h
+++ b/src/modules/interface/module_common_types.h
@@ -27,6 +27,11 @@
WebRtc_UWord16 headerLength;
};
+struct RTPHeaderExtension
+{
+ WebRtc_Word32 transmissionTimeOffset;
+};
+
struct RTPAudioHeader
{
WebRtc_UWord8 numEnergy; // number of valid entries in arrOfEnergy
@@ -37,18 +42,48 @@
struct RTPVideoHeaderH263
{
+ void InitRTPVideoHeaderH263() {};
bool independentlyDecodable; // H.263-1998 if no P bit it's not independently decodable
bool bits; // H.263 mode B, Xor the lasy byte of previus packet with the
// first byte of this packet
};
+
enum {kNoPictureId = -1};
+enum {kNoTl0PicIdx = -1};
+enum {kNoTemporalIdx = -1};
+enum {kNoKeyIdx = -1};
+enum {kNoSimulcastIdx = 0};
+
struct RTPVideoHeaderVP8
{
- bool startBit; // Start of partition.
- bool stopBit; // Stop of partition.
- WebRtc_Word16 pictureId; // Picture ID index, 15 bits;
- // kNoPictureId if PictureID does not exist.
- bool nonReference; // Frame is discardable.
+ void InitRTPVideoHeaderVP8()
+ {
+ nonReference = false;
+ pictureId = kNoPictureId;
+ tl0PicIdx = kNoTl0PicIdx;
+ temporalIdx = kNoTemporalIdx;
+ layerSync = false;
+ keyIdx = kNoKeyIdx;
+ partitionId = 0;
+ beginningOfPartition = false;
+ frameWidth = 0;
+ frameHeight = 0;
+ }
+
+ bool nonReference; // Frame is discardable.
+ WebRtc_Word16 pictureId; // Picture ID index, 15 bits;
+ // kNoPictureId if PictureID does not exist.
+ WebRtc_Word16 tl0PicIdx; // TL0PIC_IDX, 8 bits;
+ // kNoTl0PicIdx means no value provided.
+ WebRtc_Word8 temporalIdx; // Temporal layer index, or kNoTemporalIdx.
+ bool layerSync; // This frame is a layer sync frame.
+ // Disabled if temporalIdx == kNoTemporalIdx.
+ int keyIdx; // 5 bits; kNoKeyIdx means not used.
+ int partitionId; // VP8 partition ID
+ bool beginningOfPartition; // True if this packet is the first
+ // in a VP8 partition. Otherwise false
+ int frameWidth; // Exists for key frames.
+ int frameHeight; // Exists for key frames.
};
union RTPVideoTypeHeader
{
@@ -72,6 +107,8 @@
WebRtc_UWord16 height;
bool isFirstPacket; // first packet in frame
+ WebRtc_UWord8 simulcastIdx; // Index if the simulcast encoder creating
+ // this frame, 0 if not using simulcast.
RTPVideoCodecTypes codec;
RTPVideoTypeHeader codecHeader;
};
@@ -86,6 +123,7 @@
RTPHeader header;
FrameType frameType;
RTPTypeHeader type;
+ RTPHeaderExtension extension;
};
class RTPFragmentationHeader
@@ -360,11 +398,11 @@
struct VideoContentMetrics
{
VideoContentMetrics(): motionMagnitudeNZ(0), sizeZeroMotion(0), spatialPredErr(0),
- spatialPredErrH(0), spatialPredErrV(0), motionPredErr(0),
+ spatialPredErrH(0), spatialPredErrV(0), motionPredErr(0),
motionHorizontalness(0), motionClusterDistortion(0),
nativeWidth(0), nativeHeight(0), contentChange(false) { }
void Reset(){ motionMagnitudeNZ = 0; sizeZeroMotion = 0; spatialPredErr = 0;
- spatialPredErrH = 0; spatialPredErrV = 0; motionPredErr = 0;
+ spatialPredErrH = 0; spatialPredErrV = 0; motionPredErr = 0;
motionHorizontalness = 0; motionClusterDistortion = 0;
nativeWidth = 0; nativeHeight = 0; contentChange = false; }
@@ -697,7 +735,7 @@
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
- const WebRtc_UWord32 frequencyInHz,
+ const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel = 1,
@@ -719,7 +757,7 @@
// Supporting Stereo, stereo samples are interleaved
mutable WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
WebRtc_UWord16 _payloadDataLengthInSamples;
- WebRtc_UWord32 _frequencyInHz;
+ int _frequencyInHz;
WebRtc_UWord8 _audioChannel;
SpeechType _speechType;
VADActivity _vadActivity;
@@ -756,7 +794,7 @@
const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* payloadData,
const WebRtc_UWord16 payloadDataLengthInSamples,
- const WebRtc_UWord32 frequencyInHz,
+ const int frequencyInHz,
const SpeechType speechType,
const VADActivity vadActivity,
const WebRtc_UWord8 audioChannel,
diff --git a/src/system_wrappers/OWNERS b/src/system_wrappers/OWNERS
index 32dcbbc..4091a93 100644
--- a/src/system_wrappers/OWNERS
+++ b/src/system_wrappers/OWNERS
@@ -1,7 +1,7 @@
-hellner@google.com
-pwestin@google.com
-perkj@google.com
-henrika@google.com
-grunell@google.com
-mflodman@google.com
-niklase@google.com
\ No newline at end of file
+henrike@webrtc.org
+pwestin@webrtc.org
+perkj@webrtc.org
+henrika@webrtc.org
+henrikg@webrtc.org
+mflodman@webrtc.org
+niklas.enbom@webrtc.org
\ No newline at end of file
diff --git a/src/system_wrappers/interface/cpu_features_wrapper.h b/src/system_wrappers/interface/cpu_features_wrapper.h
index 5d8a828..d949592 100644
--- a/src/system_wrappers/interface/cpu_features_wrapper.h
+++ b/src/system_wrappers/interface/cpu_features_wrapper.h
@@ -15,18 +15,33 @@
extern "C" {
#endif
-// list of features.
+#include <typedefs.h>
+
+// List of features in x86.
typedef enum {
kSSE2,
kSSE3
} CPUFeature;
+// List of features in ARM.
+enum {
+ kCPUFeatureARMv7 = (1 << 0),
+ kCPUFeatureVFPv3 = (1 << 1),
+ kCPUFeatureNEON = (1 << 2),
+ kCPUFeatureLDREXSTREX = (1 << 3)
+};
+
typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
// returns true if the CPU supports the feature.
extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
// No CPU feature is available => straight C path.
extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
+// Return the features in an ARM device.
+// It detects the features in the hardware platform, and returns supported
+// values in the above enum definition as a bitmask.
+extern uint64_t WebRtc_GetCPUFeaturesARM(void);
+
#if defined(__cplusplus) || defined(c_plusplus)
} // extern "C"
#endif
diff --git a/src/system_wrappers/interface/cpu_info.h b/src/system_wrappers/interface/cpu_info.h
new file mode 100644
index 0000000..a6da29f
--- /dev/null
+++ b/src/system_wrappers/interface/cpu_info.h
@@ -0,0 +1,27 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+class CpuInfo
+{
+public:
+ static WebRtc_UWord32 DetectNumberOfCores();
+
+private:
+ CpuInfo() {}
+ static WebRtc_UWord32 _numberOfCores;
+};
+} // namespace webrtc
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
diff --git a/src/system_wrappers/interface/cpu_wrapper.h b/src/system_wrappers/interface/cpu_wrapper.h
index b72c20c..d938741 100644
--- a/src/system_wrappers/interface/cpu_wrapper.h
+++ b/src/system_wrappers/interface/cpu_wrapper.h
@@ -17,8 +17,6 @@
class CpuWrapper
{
public:
- static WebRtc_UWord32 DetectNumberOfCores();
-
static CpuWrapper* CreateCpu();
virtual ~CpuWrapper() {}
@@ -34,6 +32,14 @@
// Note that the pointer passed as cpu_usage is redirected to a local member
// of the CPU wrapper.
// numCores is the number of cores in the cpu_usage array.
+ // The return value is -1 for failure or 0-100, indicating the average
+ // CPU usage across all cores.
+ // Note: on some OSs this class is initialized lazy. This means that it
+ // might not yet be possible to retrieve any CPU metrics. When this happens
+ // the return value will be zero (indicating that there is not a failure),
+ // numCores will be 0 and cpu_usage will be set to NULL (indicating that
+ // no metrics are available yet). Once the initialization is completed,
+ // which can take in the order of seconds, CPU metrics can be retrieved.
virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
WebRtc_UWord32*& cpu_usage) = 0;
@@ -42,10 +48,6 @@
protected:
CpuWrapper() {}
-
-private:
- static WebRtc_UWord32 _numberOfCores;
-
};
} // namespace webrtc
#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
diff --git a/src/system_wrappers/interface/critical_section_wrapper.h b/src/system_wrappers/interface/critical_section_wrapper.h
index ad31497..cfec9ae 100644
--- a/src/system_wrappers/interface/critical_section_wrapper.h
+++ b/src/system_wrappers/interface/critical_section_wrapper.h
@@ -33,18 +33,26 @@
virtual void Leave() = 0;
};
-// RAII extension of the critical section. Prevents Enter/Leave missmatches and
+// RAII extension of the critical section. Prevents Enter/Leave mismatches and
// provides more compact critical section syntax.
class CriticalSectionScoped
{
public:
- CriticalSectionScoped(CriticalSectionWrapper& critsec)
- :
- _ptrCritSec(&critsec)
+ // Deprecated, don't add more users of this constructor.
+ // TODO(mflodman) Remove this version of the constructor when no one is
+ // using it any longer.
+ explicit CriticalSectionScoped(CriticalSectionWrapper& critsec)
+ : _ptrCritSec(&critsec)
{
_ptrCritSec->Enter();
}
+ explicit CriticalSectionScoped(CriticalSectionWrapper* critsec)
+ : _ptrCritSec(critsec)
+ {
+ _ptrCritSec->Enter();
+ }
+
~CriticalSectionScoped()
{
if (_ptrCritSec)
diff --git a/src/system_wrappers/interface/data_log.h b/src/system_wrappers/interface/data_log.h
new file mode 100644
index 0000000..6fc1d64
--- /dev/null
+++ b/src/system_wrappers/interface/data_log.h
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This singleton can be used for logging data for offline processing. Data
+ * logged with it can conveniently be parsed and processed with e.g. Matlab.
+ *
+ * Following is an example of the log file format, starting with the header
+ * row at line 1, and the data rows following.
+ * col1,col2,col3,multi-value-col4[3],,,col5
+ * 123,10.2,-243,1,2,3,100
+ * 241,12.3,233,1,2,3,200
+ * 13,16.4,-13,1,2,3,300
+ *
+ * As can be seen in the example, a multi-value-column is specified with the
+ * name followed the number of elements it contains. This followed by
+ * number of elements - 1 empty columns.
+ *
+ * Without multi-value-columns this format can be natively by Matlab. With
+ * multi-value-columns a small Matlab script is needed, available at
+ * trunk/tools/matlab/parseLog.m.
+ *
+ * Table names and column names are case sensitive.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+
+#include <string>
+
+#include "data_log_impl.h"
+
+namespace webrtc {
+
+class DataLog {
+ public:
+ // Creates a log which uses a separate thread (referred to as the file
+ // writer thread) for writing log rows to file.
+ //
+ // Calls to this function after the log object has been created will only
+ // increment the reference counter.
+ static int CreateLog();
+
+ // Decrements the reference counter and deletes the log when the counter
+ // reaches 0. Should be called equal number of times as successful calls to
+ // CreateLog or memory leak will occur.
+ static void ReturnLog();
+
+ // Combines the string table_name and the integer table_id into a new string
+ // table_name + _ + table_id. The new string will be lower-case.
+ static std::string Combine(const std::string& table_name, int table_id);
+
+ // Adds a new table, with the name table_name, and creates the file, with the
+ // name table_name + ".txt", to which the table will be written.
+ // table_name is treated in a case sensitive way.
+ static int AddTable(const std::string& table_name);
+
+ // Adds a new column to a table. The column will be a multi-value-column
+ // if multi_value_length is greater than 1.
+ // table_name and column_name are treated in a case sensitive way.
+ static int AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length);
+
+ // Inserts a single value into a table with name table_name at the column with
+ // name column_name.
+ // Note that the ValueContainer makes use of the copy constructor,
+ // operator= and operator<< of the type T, and that the template type must
+ // implement a deep copy copy constructor and operator=.
+ // Copy constructor and operator= must not be disabled for the type T.
+ // table_name and column_name are treated in a case sensitive way.
+ template<class T>
+ static int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ T value) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->InsertCell(
+ table_name,
+ column_name,
+ new ValueContainer<T>(value));
+ }
+
+ // Inserts an array of values into a table with name table_name at the
+ // column specified by column_name, which must be a multi-value-column.
+ // Note that the MultiValueContainer makes use of the copy constructor,
+ // operator= and operator<< of the type T, and that the template type
+ // must implement a deep copy copy constructor and operator=.
+ // Copy constructor and operator= must not be disabled for the type T.
+ // table_name and column_name are treated in a case sensitive way.
+ template<class T>
+ static int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const T* array,
+ int length) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->InsertCell(
+ table_name,
+ column_name,
+ new MultiValueContainer<T>(array, length));
+ }
+
+ // For the table with name table_name: Writes the current row to file.
+ // Starts a new empty row.
+ // table_name is treated in a case-sensitive way.
+ static int NextRow(const std::string& table_name);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
diff --git a/src/system_wrappers/interface/data_log_c.h b/src/system_wrappers/interface/data_log_c.h
new file mode 100644
index 0000000..fffbb4f
--- /dev/null
+++ b/src/system_wrappers/interface/data_log_c.h
@@ -0,0 +1,89 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is a pure C wrapper of the DataLog class. The functions are directly
+ * mapped here except for InsertCell as C does not support templates.
+ * See data_log.h for a description of the functions.
+ */
+
+#ifndef SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
+#define SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
+
+#include <stddef.h> /* size_t */
+
+#include "typedefs.h" /* NOLINT(build/include) */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * All char* parameters in this file are expected to be null-terminated
+ * character sequences.
+ */
+int WebRtcDataLog_CreateLog();
+void WebRtcDataLog_ReturnLog();
+char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
+ const char* table_name, int table_id);
+int WebRtcDataLog_AddTable(const char* table_name);
+int WebRtcDataLog_AddColumn(const char* table_name, const char* column_name,
+ int multi_value_length);
+
+int WebRtcDataLog_InsertCell_int(const char* table_name,
+ const char* column_name,
+ int value);
+int WebRtcDataLog_InsertArray_int(const char* table_name,
+ const char* column_name,
+ const int* values,
+ int length);
+int WebRtcDataLog_InsertCell_float(const char* table_name,
+ const char* column_name,
+ float value);
+int WebRtcDataLog_InsertArray_float(const char* table_name,
+ const char* column_name,
+ const float* values,
+ int length);
+int WebRtcDataLog_InsertCell_double(const char* table_name,
+ const char* column_name,
+ double value);
+int WebRtcDataLog_InsertArray_double(const char* table_name,
+ const char* column_name,
+ const double* values,
+ int length);
+int WebRtcDataLog_InsertCell_int32(const char* table_name,
+ const char* column_name,
+ int32_t value);
+int WebRtcDataLog_InsertArray_int32(const char* table_name,
+ const char* column_name,
+ const int32_t* values,
+ int length);
+int WebRtcDataLog_InsertCell_uint32(const char* table_name,
+ const char* column_name,
+ uint32_t value);
+int WebRtcDataLog_InsertArray_uint32(const char* table_name,
+ const char* column_name,
+ const uint32_t* values,
+ int length);
+int WebRtcDataLog_InsertCell_int64(const char* table_name,
+ const char* column_name,
+ int64_t value);
+int WebRtcDataLog_InsertArray_int64(const char* table_name,
+ const char* column_name,
+ const int64_t* values,
+ int length);
+
+int WebRtcDataLog_NextRow(const char* table_name);
+
+#ifdef __cplusplus
+} /* end of extern "C" */
+#endif
+
+#endif /* SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_ */ /* NOLINT */
diff --git a/src/system_wrappers/interface/data_log_impl.h b/src/system_wrappers/interface/data_log_impl.h
new file mode 100644
index 0000000..cef4964
--- /dev/null
+++ b/src/system_wrappers/interface/data_log_impl.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the helper classes for the DataLog APIs. See data_log.h
+ * for the APIs.
+ *
+ * These classes are helper classes used for logging data for offline
+ * processing. Data logged with these classes can conveniently be parsed and
+ * processed with e.g. Matlab.
+ */
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+
+#include <map>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "scoped_ptr.h"
+#include "typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class EventWrapper;
+class LogTable;
+class RWLockWrapper;
+class ThreadWrapper;
+
+// All container classes need to implement a ToString-function to be
+// writable to file. Enforce this via the Container interface.
+class Container {
+ public:
+ virtual ~Container() {}
+
+ virtual void ToString(std::string* container_string) const = 0;
+};
+
+template<class T>
+class ValueContainer : public Container {
+ public:
+ explicit ValueContainer(T data) : data_(data) {}
+
+ virtual void ToString(std::string* container_string) const {
+ *container_string = "";
+ std::stringstream ss;
+ ss << data_ << ",";
+ ss >> *container_string;
+ }
+
+ private:
+ T data_;
+};
+
+template<class T>
+class MultiValueContainer : public Container {
+ public:
+ MultiValueContainer(const T* data, int length)
+ : data_(data, data + length) {
+ }
+
+ virtual void ToString(std::string* container_string) const {
+ *container_string = "";
+ std::stringstream ss;
+ for (size_t i = 0; i < data_.size(); ++i)
+ ss << data_[i] << ",";
+ *container_string += ss.str();
+ }
+
+ private:
+ std::vector<T> data_;
+};
+
+class DataLogImpl {
+ public:
+ ~DataLogImpl();
+
+ // The implementation of the CreateLog() method declared in data_log.h.
+ // See data_log.h for a description.
+ static int CreateLog();
+
+ // The implementation of the StaticInstance() method declared in data_log.h.
+ // See data_log.h for a description.
+ static DataLogImpl* StaticInstance();
+
+ // The implementation of the ReturnLog() method declared in data_log.h. See
+ // data_log.h for a description.
+ static void ReturnLog();
+
+ // The implementation of the AddTable() method declared in data_log.h. See
+ // data_log.h for a description.
+ int AddTable(const std::string& table_name);
+
+ // The implementation of the AddColumn() method declared in data_log.h. See
+ // data_log.h for a description.
+ int AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length);
+
+ // Inserts a Container into a table with name table_name at the column
+ // with name column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const Container* value_container);
+
+ // The implementation of the NextRow() method declared in data_log.h. See
+ // data_log.h for a description.
+ int NextRow(const std::string& table_name);
+
+ private:
+ DataLogImpl();
+
+ // Initializes the DataLogImpl object, allocates and starts the
+ // thread file_writer_thread_.
+ int Init();
+
+ // Write all complete rows in every table to file.
+ // This function should only be called by the file_writer_thread_ if that
+ // thread is running to avoid race conditions.
+ void Flush();
+
+ // Run() is called by the thread file_writer_thread_.
+ static bool Run(void* obj);
+
+ // This function writes data to file. Note, it blocks if there is no data
+ // that should be written to file availble. Flush is the non-blocking
+ // version of this function.
+ void Process();
+
+ // Stops the continuous calling of Process().
+ void StopThread();
+
+ // Collection of tables indexed by the table name as std::string.
+ typedef std::map<std::string, LogTable*> TableMap;
+ typedef webrtc::scoped_ptr<CriticalSectionWrapper> CritSectScopedPtr;
+
+ static CritSectScopedPtr crit_sect_;
+ static DataLogImpl* instance_;
+ int counter_;
+ TableMap tables_;
+ EventWrapper* flush_event_;
+ ThreadWrapper* file_writer_thread_;
+ RWLockWrapper* tables_lock_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
diff --git a/src/system_wrappers/interface/file_wrapper.h b/src/system_wrappers/interface/file_wrapper.h
index 8f0cd8c..4d17438 100644
--- a/src/system_wrappers/interface/file_wrapper.h
+++ b/src/system_wrappers/interface/file_wrapper.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
+#include <stddef.h>
+
#include "common_types.h"
#include "typedefs.h"
@@ -18,11 +20,11 @@
// write from/to a file.
namespace webrtc {
+
class FileWrapper : public InStream, public OutStream
{
public:
- enum { kMaxFileNameSize = 1024};
- enum { kFileMaxTextMessageSize = 1024};
+ static const size_t kMaxFileNameSize = 1024;
// Factory method. Constructor disabled.
static FileWrapper* Create();
@@ -31,42 +33,47 @@
virtual bool Open() const = 0;
// Opens a file in read or write mode, decided by the readOnly parameter.
- virtual WebRtc_Word32 OpenFile(const WebRtc_Word8* fileNameUTF8,
- const bool readOnly,
- const bool loop = false,
- const bool text = false) = 0;
+ virtual int OpenFile(const char* fileNameUTF8,
+ bool readOnly,
+ bool loop = false,
+ bool text = false) = 0;
- virtual WebRtc_Word32 CloseFile() = 0;
+ virtual int CloseFile() = 0;
- // Limits the file size.
- virtual WebRtc_Word32 SetMaxFileSize(WebRtc_Word32 bytes) = 0;
+ // Limits the file size to |bytes|. Writing will fail after the cap
+ // is hit. Pass zero to use an unlimited size.
+ virtual int SetMaxFileSize(size_t bytes) = 0;
// Flush any pending writes.
- virtual WebRtc_Word32 Flush() = 0;
+ virtual int Flush() = 0;
- // Returns the opened file's name in fileNameUTF8. size is the allocated
- // size of fileNameUTF8. The name will be truncated if the size of
- // fileNameUTF8 is to small.
- virtual WebRtc_Word32 FileName(WebRtc_Word8* fileNameUTF8,
- WebRtc_UWord32 size) const = 0;
+ // Returns the opened file's name in |fileNameUTF8|. Provide the size of
+ // the buffer in bytes in |size|. The name will be truncated if |size| is
+ // too small.
+ virtual int FileName(char* fileNameUTF8,
+ size_t size) const = 0;
- // Write text to the opened file. The written text can contain plain text
- // and text with type specifiers in the same way as sprintf works.
- virtual WebRtc_Word32 WriteText(const WebRtc_Word8* text, ...) = 0;
+ // Write |format| to the opened file. Arguments are taken in the same manner
+ // as printf. That is, supply a format string containing text and
+ // specifiers. Returns the number of characters written or -1 on error.
+ virtual int WriteText(const char* format, ...) = 0;
- // Reads len number of bytes from buf to file.
- virtual int Read(void* buf, int len) = 0;
+ // Inherited from Instream.
+ // Reads |length| bytes from file to |buf|. Returns the number of bytes read
+ // or -1 on error.
+ virtual int Read(void* buf, int length) = 0;
- // Writes len number of bytes to buf from file. Please note that the actual
- // writing to file may happen some time later. Call flush to force a write
- // to take affect
- virtual bool Write(const void *buf,int len) = 0;
+ // Inherited from OutStream.
+ // Writes |length| bytes from |buf| to file. The actual writing may happen
+ // some time later. Call Flush() to force a write.
+ virtual bool Write(const void *buf, int length) = 0;
+ // Inherited from both Instream and OutStream.
// Rewinds the file to the start. Only available when OpenFile() has been
- // called with loop argument set to true. Or readOnly argument has been set
- // to false.
+ // called with |loop| == true or |readOnly| == true.
virtual int Rewind() = 0;
};
+
} // namespace webrtc
#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
diff --git a/src/system_wrappers/interface/list_wrapper.h b/src/system_wrappers/interface/list_wrapper.h
index bc10ad4..3608ada 100644
--- a/src/system_wrappers/interface/list_wrapper.h
+++ b/src/system_wrappers/interface/list_wrapper.h
@@ -34,7 +34,6 @@
private:
const void* item_ptr_;
const unsigned int item_;
- DISALLOW_COPY_AND_ASSIGN(ListItem);
};
class ListWrapper
@@ -102,7 +101,6 @@
ListItem* first_;
ListItem* last_;
unsigned int size_;
- DISALLOW_COPY_AND_ASSIGN(ListWrapper);
};
} //namespace webrtc
diff --git a/src/system_wrappers/interface/map_wrapper.h b/src/system_wrappers/interface/map_wrapper.h
index 9297382..7d4e733 100644
--- a/src/system_wrappers/interface/map_wrapper.h
+++ b/src/system_wrappers/interface/map_wrapper.h
@@ -31,7 +31,6 @@
private:
int item_id_;
void* item_pointer_;
- DISALLOW_COPY_AND_ASSIGN(MapItem);
};
class MapWrapper
@@ -70,7 +69,6 @@
private:
std::map<int, MapItem*> map_;
- DISALLOW_COPY_AND_ASSIGN(MapWrapper);
};
} // namespace webrtc
diff --git a/src/system_wrappers/interface/ref_count.h b/src/system_wrappers/interface/ref_count.h
new file mode 100644
index 0000000..f90b0b3
--- /dev/null
+++ b/src/system_wrappers/interface/ref_count.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+#define SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+
+#include "system_wrappers/interface/atomic32_wrapper.h"
+
+namespace webrtc {
+
+// This class can be used for instantiating
+// reference counted objects.
+// int32_t AddRef() and int32_t Release().
+// Usage:
+// RefCountImpl<T>* implementation = new RefCountImpl<T>(p);
+//
+// Example:
+// class MyInterface {
+// public:
+// virtual void DoSomething() = 0;
+// virtual int32_t AddRef() = 0;
+// virtual int32_t Release() = 0:
+// private:
+// virtual ~MyInterface(){};
+// }
+// class MyImplementation : public MyInterface {
+// public:
+// virtual DoSomething() { printf("hello"); };
+// };
+// MyImplementation* CreateMyImplementation() {
+// RefCountImpl<MyImplementation>* implementation =
+// new RefCountImpl<MyImplementation>();
+// return implementation;
+// }
+
+template <class T>
+class RefCountImpl : public T {
+ public:
+ RefCountImpl() : ref_count_(0) {}
+
+ template<typename P>
+ explicit RefCountImpl(P p) : T(p), ref_count_(0) {}
+
+ template<typename P1, typename P2>
+ RefCountImpl(P1 p1, P2 p2) : T(p1, p2), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3>
+ RefCountImpl(P1 p1, P2 p2, P3 p3) : T(p1, p2, p3), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3, typename P4>
+ RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4) : T(p1, p2, p3, p4), ref_count_(0) {}
+
+ template<typename P1, typename P2, typename P3, typename P4, typename P5>
+ RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5)
+ : T(p1, p2, p3, p4, p5), ref_count_(0) {}
+
+ virtual int32_t AddRef() {
+ return ++ref_count_;
+ }
+
+ virtual int32_t Release() {
+ int32_t ref_count;
+ ref_count = --ref_count_;
+ if (ref_count == 0)
+ delete this;
+ return ref_count;
+ }
+
+ protected:
+ Atomic32Wrapper ref_count_;
+};
+
+} // namespace webrtc
+
+#endif // SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
diff --git a/src/system_wrappers/interface/scoped_ptr.h b/src/system_wrappers/interface/scoped_ptr.h
new file mode 100644
index 0000000..74b6ad3
--- /dev/null
+++ b/src/system_wrappers/interface/scoped_ptr.h
@@ -0,0 +1,258 @@
+// (C) Copyright Greg Colvin and Beman Dawes 1998, 1999.
+// Copyright (c) 2001, 2002 Peter Dimov
+//
+// Permission to copy, use, modify, sell and distribute this software
+// is granted provided this copyright notice appears in all copies.
+// This software is provided "as is" without express or implied
+// warranty, and with no claim as to its suitability for any purpose.
+//
+// See http://www.boost.org/libs/smart_ptr/scoped_ptr.htm for documentation.
+//
+
+// scoped_ptr mimics a built-in pointer except that it guarantees deletion
+// of the object pointed to, either on destruction of the scoped_ptr or via
+// an explicit reset(). scoped_ptr is a simple solution for simple needs;
+// use shared_ptr or std::auto_ptr if your needs are more complex.
+
+// scoped_ptr_malloc added in by Google. When one of
+// these goes out of scope, instead of doing a delete or delete[], it
+// calls free(). scoped_ptr_malloc<char> is likely to see much more
+// use than any other specializations.
+
+// release() added in by Google. Use this to conditionally
+// transfer ownership of a heap-allocated object to the caller, usually on
+// method success.
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
+
+#include <assert.h> // for assert
+#include <stdlib.h> // for free() decl
+
+#include <cstddef> // for std::ptrdiff_t
+
+#ifdef _WIN32
+namespace std { using ::ptrdiff_t; };
+#endif // _WIN32
+
+namespace webrtc {
+
+template <typename T>
+class scoped_ptr {
+ private:
+
+ T* ptr;
+
+ scoped_ptr(scoped_ptr const &);
+ scoped_ptr & operator=(scoped_ptr const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_ptr(T* p = NULL): ptr(p) {}
+
+ ~scoped_ptr() {
+ typedef char type_must_be_complete[sizeof(T)];
+ delete ptr;
+ }
+
+ void reset(T* p = NULL) {
+ typedef char type_must_be_complete[sizeof(T)];
+
+ if (ptr != p) {
+ T* obj = ptr;
+ ptr = p;
+ // Delete last, in case obj destructor indirectly results in ~scoped_ptr
+ delete obj;
+ }
+ }
+
+ T& operator*() const {
+ assert(ptr != NULL);
+ return *ptr;
+ }
+
+ T* operator->() const {
+ assert(ptr != NULL);
+ return ptr;
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_ptr & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = NULL;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ delete ptr;
+ ptr = NULL;
+ }
+ return &ptr;
+ }
+
+ T** use() {
+ return &ptr;
+ }
+};
+
+template<typename T> inline
+void swap(scoped_ptr<T>& a, scoped_ptr<T>& b) {
+ a.swap(b);
+}
+
+
+
+
+// scoped_array extends scoped_ptr to arrays. Deletion of the array pointed to
+// is guaranteed, either on destruction of the scoped_array or via an explicit
+// reset(). Use shared_array or std::vector if your needs are more complex.
+
+template<typename T>
+class scoped_array {
+ private:
+
+ T* ptr;
+
+ scoped_array(scoped_array const &);
+ scoped_array & operator=(scoped_array const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_array(T* p = NULL) : ptr(p) {}
+
+ ~scoped_array() {
+ typedef char type_must_be_complete[sizeof(T)];
+ delete[] ptr;
+ }
+
+ void reset(T* p = NULL) {
+ typedef char type_must_be_complete[sizeof(T)];
+
+ if (ptr != p) {
+ T* arr = ptr;
+ ptr = p;
+ // Delete last, in case arr destructor indirectly results in ~scoped_array
+ delete [] arr;
+ }
+ }
+
+ T& operator[](std::ptrdiff_t i) const {
+ assert(ptr != NULL);
+ assert(i >= 0);
+ return ptr[i];
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_array & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = NULL;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ delete [] ptr;
+ ptr = NULL;
+ }
+ return &ptr;
+ }
+};
+
+template<class T> inline
+void swap(scoped_array<T>& a, scoped_array<T>& b) {
+ a.swap(b);
+}
+
+// scoped_ptr_malloc<> is similar to scoped_ptr<>, but it accepts a
+// second template argument, the function used to free the object.
+
+template<typename T, void (*FF)(void*) = free> class scoped_ptr_malloc {
+ private:
+
+ T* ptr;
+
+ scoped_ptr_malloc(scoped_ptr_malloc const &);
+ scoped_ptr_malloc & operator=(scoped_ptr_malloc const &);
+
+ public:
+
+ typedef T element_type;
+
+ explicit scoped_ptr_malloc(T* p = 0): ptr(p) {}
+
+ ~scoped_ptr_malloc() {
+ FF(static_cast<void*>(ptr));
+ }
+
+ void reset(T* p = 0) {
+ if (ptr != p) {
+ FF(static_cast<void*>(ptr));
+ ptr = p;
+ }
+ }
+
+ T& operator*() const {
+ assert(ptr != 0);
+ return *ptr;
+ }
+
+ T* operator->() const {
+ assert(ptr != 0);
+ return ptr;
+ }
+
+ T* get() const {
+ return ptr;
+ }
+
+ void swap(scoped_ptr_malloc & b) {
+ T* tmp = b.ptr;
+ b.ptr = ptr;
+ ptr = tmp;
+ }
+
+ T* release() {
+ T* tmp = ptr;
+ ptr = 0;
+ return tmp;
+ }
+
+ T** accept() {
+ if (ptr) {
+ FF(static_cast<void*>(ptr));
+ ptr = 0;
+ }
+ return &ptr;
+ }
+};
+
+template<typename T, void (*FF)(void*)> inline
+void swap(scoped_ptr_malloc<T,FF>& a, scoped_ptr_malloc<T,FF>& b) {
+ a.swap(b);
+}
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
diff --git a/src/system_wrappers/interface/scoped_refptr.h b/src/system_wrappers/interface/scoped_refptr.h
new file mode 100644
index 0000000..0df15be
--- /dev/null
+++ b/src/system_wrappers/interface/scoped_refptr.h
@@ -0,0 +1,137 @@
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file under third_party_mods/chromium or at:
+// http://src.chromium.org/svn/trunk/src/LICENSE
+
+#ifndef SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
+#define SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
+
+namespace webrtc {
+
+// Extracted from Chromium's src/base/memory/ref_counted.h.
+
+//
+// A smart pointer class for reference counted objects. Use this class instead
+// of calling AddRef and Release manually on a reference counted object to
+// avoid common memory leaks caused by forgetting to Release an object
+// reference. Sample usage:
+//
+// class MyFoo : public RefCounted<MyFoo> {
+// ...
+// };
+//
+// void some_function() {
+// scoped_refptr<MyFoo> foo = new MyFoo();
+// foo->Method(param);
+// // |foo| is released when this function returns
+// }
+//
+// void some_other_function() {
+// scoped_refptr<MyFoo> foo = new MyFoo();
+// ...
+// foo = NULL; // explicitly releases |foo|
+// ...
+// if (foo)
+// foo->Method(param);
+// }
+//
+// The above examples show how scoped_refptr<T> acts like a pointer to T.
+// Given two scoped_refptr<T> classes, it is also possible to exchange
+// references between the two objects, like so:
+//
+// {
+// scoped_refptr<MyFoo> a = new MyFoo();
+// scoped_refptr<MyFoo> b;
+//
+// b.swap(a);
+// // now, |b| references the MyFoo object, and |a| references NULL.
+// }
+//
+// To make both |a| and |b| in the above example reference the same MyFoo
+// object, simply use the assignment operator:
+//
+// {
+// scoped_refptr<MyFoo> a = new MyFoo();
+// scoped_refptr<MyFoo> b;
+//
+// b = a;
+// // now, |a| and |b| each own a reference to the same MyFoo object.
+// }
+//
+template <class T>
+class scoped_refptr {
+ public:
+ scoped_refptr() : ptr_(NULL) {
+ }
+
+ scoped_refptr(T* p) : ptr_(p) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ template <typename U>
+ scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+
+ ~scoped_refptr() {
+ if (ptr_)
+ ptr_->Release();
+ }
+
+ T* get() const { return ptr_; }
+ operator T*() const { return ptr_; }
+ T* operator->() const { return ptr_; }
+
+ // Release a pointer.
+ // The return value is the current pointer held by this object.
+ // If this object holds a NULL pointer, the return value is NULL.
+ // After this operation, this object will hold a NULL pointer,
+ // and will not own the object any more.
+ T* release() {
+ T* retVal = ptr_;
+ ptr_ = NULL;
+ return retVal;
+ }
+
+ scoped_refptr<T>& operator=(T* p) {
+ // AddRef first so that self assignment should work
+ if (p)
+ p->AddRef();
+ if (ptr_ )
+ ptr_->Release();
+ ptr_ = p;
+ return *this;
+ }
+
+ scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
+ return *this = r.ptr_;
+ }
+
+ template <typename U>
+ scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
+ return *this = r.get();
+ }
+
+ void swap(T** pp) {
+ T* p = ptr_;
+ ptr_ = *pp;
+ *pp = p;
+ }
+
+ void swap(scoped_refptr<T>& r) {
+ swap(&r.ptr_);
+ }
+
+ protected:
+ T* ptr_;
+};
+} // namespace webrtc
+
+#endif // SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
diff --git a/src/system_wrappers/interface/static_instance.h b/src/system_wrappers/interface/static_instance.h
new file mode 100644
index 0000000..8fe91cc
--- /dev/null
+++ b/src/system_wrappers/interface/static_instance.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
+
+#include <assert.h>
+
+#include "critical_section_wrapper.h"
+#ifdef _WIN32
+#include "fix_interlocked_exchange_pointer_win.h"
+#endif
+
+namespace webrtc {
+
+enum CountOperation {
+ kRelease,
+ kAddRef,
+ kAddRefNoCreate
+};
+enum CreateOperation {
+ kInstanceExists,
+ kCreate,
+ kDestroy
+};
+
+template <class T>
+// Construct On First Use idiom. Avoids
+// "static initialization order fiasco".
+static T* GetStaticInstance(CountOperation count_operation) {
+ // TODO (hellner): use atomic wrapper instead.
+ static volatile long instance_count = 0;
+ static T* volatile instance = NULL;
+ CreateOperation state = kInstanceExists;
+#ifndef _WIN32
+ // This memory is staticly allocated once. The application does not try to
+ // free this memory. This approach is taken to avoid issues with
+ // destruction order for statically allocated memory. The memory will be
+ // reclaimed by the OS and memory leak tools will not recognize memory
+ // reachable from statics leaked so no noise is added by doing this.
+ static CriticalSectionWrapper* crit_sect(
+ CriticalSectionWrapper::CreateCriticalSection());
+ CriticalSectionScoped lock(crit_sect);
+
+ if (count_operation ==
+ kAddRefNoCreate && instance_count == 0) {
+ return NULL;
+ }
+ if (count_operation ==
+ kAddRef ||
+ count_operation == kAddRefNoCreate) {
+ instance_count++;
+ if (instance_count == 1) {
+ state = kCreate;
+ }
+ } else {
+ instance_count--;
+ if (instance_count == 0) {
+ state = kDestroy;
+ }
+ }
+ if (state == kCreate) {
+ instance = T::CreateInstance();
+ } else if (state == kDestroy) {
+ T* old_instance = instance;
+ instance = NULL;
+ // The state will not change past this point. Release the critical
+ // section while deleting the object in case it would be blocking on
+ // access back to this object. (This is the case for the tracing class
+ // since the thread owned by the tracing class also traces).
+ // TODO(hellner): this is a bit out of place but here goes, de-couple
+ // thread implementation with trace implementation.
+ crit_sect->Leave();
+ if (old_instance) {
+ delete old_instance;
+ }
+ // Re-acquire the lock since the scoped critical section will release
+ // it.
+ crit_sect->Enter();
+ return NULL;
+ }
+#else // _WIN32
+ if (count_operation ==
+ kAddRefNoCreate && instance_count == 0) {
+ return NULL;
+ }
+ if (count_operation == kAddRefNoCreate) {
+ if (1 == InterlockedIncrement(&instance_count)) {
+ // The instance has been destroyed by some other thread. Rollback.
+ InterlockedDecrement(&instance_count);
+ assert(false);
+ return NULL;
+ }
+ // Sanity to catch corrupt state.
+ if (instance == NULL) {
+ assert(false);
+ InterlockedDecrement(&instance_count);
+ return NULL;
+ }
+ } else if (count_operation == kAddRef) {
+ if (instance_count == 0) {
+ state = kCreate;
+ } else {
+ if (1 == InterlockedIncrement(&instance_count)) {
+ // InterlockedDecrement because reference count should not be
+ // updated just yet (that's done when the instance is created).
+ InterlockedDecrement(&instance_count);
+ state = kCreate;
+ }
+ }
+ } else {
+ int newValue = InterlockedDecrement(&instance_count);
+ if (newValue == 0) {
+ state = kDestroy;
+ }
+ }
+
+ if (state == kCreate) {
+ // Create instance and let whichever thread finishes first assign its
+ // local copy to the global instance. All other threads reclaim their
+ // local copy.
+ T* new_instance = T::CreateInstance();
+ if (1 == InterlockedIncrement(&instance_count)) {
+ T* old_value = static_cast<T*> (InterlockedExchangePointer(
+ reinterpret_cast<void* volatile*>(&instance), new_instance));
+ assert(old_value == NULL);
+ assert(instance);
+ } else {
+ InterlockedDecrement(&instance_count);
+ if (new_instance) {
+ delete static_cast<T*>(new_instance);
+ }
+ }
+ } else if (state == kDestroy) {
+ T* old_value = static_cast<T*> (InterlockedExchangePointer(
+ reinterpret_cast<void* volatile*>(&instance), NULL));
+ if (old_value) {
+ delete static_cast<T*>(old_value);
+ }
+ return NULL;
+ }
+#endif // #ifndef _WIN32
+ return instance;
+}
+
+} // namspace webrtc
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
diff --git a/src/system_wrappers/interface/thread_wrapper.h b/src/system_wrappers/interface/thread_wrapper.h
index eccf3c2..72a06e8 100644
--- a/src/system_wrappers/interface/thread_wrapper.h
+++ b/src/system_wrappers/interface/thread_wrapper.h
@@ -21,7 +21,9 @@
// function.
#define ThreadObj void*
-// Callback function that the spawned thread will enter once spawned
+// Callback function that the spawned thread will enter once spawned.
+// A return value of false is interpreted as that the function has no
+// more work to do and that the thread can be released.
typedef bool(*ThreadRunFunction)(ThreadObj);
enum ThreadPriority
diff --git a/src/system_wrappers/interface/tick_util.h b/src/system_wrappers/interface/tick_util.h
index 4c28067..e78e53d 100644
--- a/src/system_wrappers/interface/tick_util.h
+++ b/src/system_wrappers/interface/tick_util.h
@@ -76,6 +76,11 @@
const TickInterval& rhs);
TickInterval& operator+=(const TickInterval& rhs);
+ friend bool operator>(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator<=(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator<(const TickInterval& lhs, const TickInterval& rhs);
+ friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
+
private:
TickInterval(WebRtc_Word64 interval);
@@ -107,6 +112,22 @@
time._ticks += ticks;
return time;
}
+inline bool operator>(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval > rhs._interval;
+}
+inline bool operator<=(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval <= rhs._interval;
+}
+inline bool operator<(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval <= rhs._interval;
+}
+inline bool operator>=(const TickInterval& lhs, const TickInterval& rhs)
+{
+ return lhs._interval >= rhs._interval;
+}
inline TickTime TickTime::Now()
{
diff --git a/src/system_wrappers/interface/trace.h b/src/system_wrappers/interface/trace.h
index 0f7df4d..8330f7c 100644
--- a/src/system_wrappers/interface/trace.h
+++ b/src/system_wrappers/interface/trace.h
@@ -18,14 +18,7 @@
#include "common_types.h"
#include "typedefs.h"
-#ifdef WEBRTC_NO_TRACE
- #define WEBRTC_TRACE
-#else
- // Ideally we would use __VA_ARGS__ but it's not supported by all compilers
- // such as VS2003 (it's supported in VS2005). TODO (hellner) why
- // would this be better than current implementation (not convinced)?
- #define WEBRTC_TRACE Trace::Add
-#endif
+#define WEBRTC_TRACE Trace::Add
namespace webrtc {
class Trace
diff --git a/src/system_wrappers/source/Android.mk b/src/system_wrappers/source/Android.mk
index f8e406f..575580a 100644
--- a/src/system_wrappers/source/Android.mk
+++ b/src/system_wrappers/source/Android.mk
@@ -1,14 +1,21 @@
-# This file is generated by gyp; do not edit. This means you!
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
LOCAL_ARM_MODE := arm
LOCAL_MODULE := libwebrtc_system_wrappers
LOCAL_MODULE_TAGS := optional
LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
LOCAL_SRC_FILES := \
map.cc \
rw_lock_generic.cc \
@@ -16,8 +23,10 @@
aligned_malloc.cc \
atomic32.cc \
condition_variable.cc \
- cpu.cc \
+ cpu_no_op.cc \
cpu_features.cc \
+ cpu_features_arm.c \
+ cpu_info.cc \
critical_section.cc \
event.cc \
file_impl.cc \
@@ -25,43 +34,28 @@
rw_lock.cc \
thread.cc \
trace_impl.cc \
- condition_variable_linux.cc \
+ condition_variable_posix.cc \
cpu_linux.cc \
- critical_section_linux.cc \
- event_linux.cc \
- thread_linux.cc \
- trace_linux.cc \
- rw_lock_linux.cc
+ critical_section_posix.cc \
+ event_posix.cc \
+ thread_posix.cc \
+ trace_posix.cc \
+ rw_lock_posix.cc
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
- '-DNO_HEAPCHECKER' \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_CLOCK_TYPE_REALTIME' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_ANDROID' \
- '-DANDROID'
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/spreadsortlib \
- $(LOCAL_PATH)/../interface
+LOCAL_C_INCLUDES := \
+ $(LOCAL_PATH)/../.. \
+ $(LOCAL_PATH)/../interface \
+ $(LOCAL_PATH)/spreadsortlib
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS :=
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
libdl \
libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
+ifndef NDK_ROOT
include external/stlport/libstlport.mk
+endif
include $(BUILD_STATIC_LIBRARY)
diff --git a/src/system_wrappers/source/aligned_malloc.cc b/src/system_wrappers/source/aligned_malloc.cc
index 6225753..bb10c6b 100644
--- a/src/system_wrappers/source/aligned_malloc.cc
+++ b/src/system_wrappers/source/aligned_malloc.cc
@@ -13,7 +13,7 @@
#include <assert.h>
#include <memory.h>
-#ifdef ANDROID
+#ifdef WEBRTC_ANDROID
#include <stdlib.h>
#endif
diff --git a/src/system_wrappers/source/atomic32.cc b/src/system_wrappers/source/atomic32.cc
index 3d6849e..588dd3e 100644
--- a/src/system_wrappers/source/atomic32.cc
+++ b/src/system_wrappers/source/atomic32.cc
@@ -11,7 +11,7 @@
#include "atomic32_wrapper.h"
#if defined(_WIN32)
- #include "atomic32_windows.h"
+ #include "atomic32_win.h"
#elif defined(WEBRTC_LINUX)
#include "atomic32_linux.h"
#elif defined(WEBRTC_MAC)
diff --git a/src/system_wrappers/source/condition_variable.cc b/src/system_wrappers/source/condition_variable.cc
index 7ca1b56..b37d037 100644
--- a/src/system_wrappers/source/condition_variable.cc
+++ b/src/system_wrappers/source/condition_variable.cc
@@ -11,16 +11,16 @@
#if defined(_WIN32)
#include <windows.h>
#include "condition_variable_wrapper.h"
- #include "condition_variable_windows.h"
+ #include "condition_variable_win.h"
#elif defined(WEBRTC_LINUX)
#include <pthread.h>
#include "condition_variable_wrapper.h"
- #include "condition_variable_linux.h"
+ #include "condition_variable_posix.h"
#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
#include <pthread.h>
#include "condition_variable_wrapper.h"
- #include "condition_variable_linux.h"
- #endif
+ #include "condition_variable_posix.h"
+#endif
namespace webrtc {
ConditionVariableWrapper*
@@ -29,7 +29,7 @@
#if defined(_WIN32)
return new ConditionVariableWindows;
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
- return ConditionVariableLinux::Create();
+ return ConditionVariablePosix::Create();
#else
return NULL;
#endif
diff --git a/src/system_wrappers/source/condition_variable_linux.cc b/src/system_wrappers/source/condition_variable_posix.cc
similarity index 81%
rename from src/system_wrappers/source/condition_variable_linux.cc
rename to src/system_wrappers/source/condition_variable_posix.cc
index 778c2cf..48835ab 100644
--- a/src/system_wrappers/source/condition_variable_linux.cc
+++ b/src/system_wrappers/source/condition_variable_posix.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "condition_variable_linux.h"
+#include "condition_variable_posix.h"
#if defined(WEBRTC_LINUX)
#include <ctime>
@@ -18,12 +18,12 @@
#include <errno.h>
-#include "critical_section_linux.h"
+#include "critical_section_posix.h"
namespace webrtc {
-ConditionVariableWrapper* ConditionVariableLinux::Create()
+ConditionVariableWrapper* ConditionVariablePosix::Create()
{
- ConditionVariableLinux* ptr = new ConditionVariableLinux;
+ ConditionVariablePosix* ptr = new ConditionVariablePosix;
if (!ptr)
{
return NULL;
@@ -39,11 +39,11 @@
return ptr;
}
-ConditionVariableLinux::ConditionVariableLinux()
+ConditionVariablePosix::ConditionVariablePosix()
{
}
-int ConditionVariableLinux::Construct()
+int ConditionVariablePosix::Construct()
{
int result = 0;
#ifdef WEBRTC_CLOCK_TYPE_REALTIME
@@ -74,21 +74,21 @@
return 0;
}
-ConditionVariableLinux::~ConditionVariableLinux()
+ConditionVariablePosix::~ConditionVariablePosix()
{
pthread_cond_destroy(&_cond);
}
-void ConditionVariableLinux::SleepCS(CriticalSectionWrapper& critSect)
+void ConditionVariablePosix::SleepCS(CriticalSectionWrapper& critSect)
{
- CriticalSectionLinux* cs = reinterpret_cast<CriticalSectionLinux*>(
+ CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
&critSect);
pthread_cond_wait(&_cond, &cs->_mutex);
}
bool
-ConditionVariableLinux::SleepCS(
+ConditionVariablePosix::SleepCS(
CriticalSectionWrapper& critSect,
unsigned long maxTimeInMS)
{
@@ -101,7 +101,7 @@
const int NANOSECONDS_PER_SECOND = 1000000000;
const int NANOSECONDS_PER_MILLISECOND = 1000000;
- CriticalSectionLinux* cs = reinterpret_cast<CriticalSectionLinux*>(
+ CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
&critSect);
if (maxTimeInMS != INFINITE)
@@ -139,12 +139,12 @@
}
}
-void ConditionVariableLinux::Wake()
+void ConditionVariablePosix::Wake()
{
pthread_cond_signal(&_cond);
}
-void ConditionVariableLinux::WakeAll()
+void ConditionVariablePosix::WakeAll()
{
pthread_cond_broadcast(&_cond);
}
diff --git a/src/system_wrappers/source/condition_variable_linux.h b/src/system_wrappers/source/condition_variable_posix.h
similarity index 72%
rename from src/system_wrappers/source/condition_variable_linux.h
rename to src/system_wrappers/source/condition_variable_posix.h
index 0300c5b..c239a47 100644
--- a/src/system_wrappers/source/condition_variable_linux.h
+++ b/src/system_wrappers/source/condition_variable_posix.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
#include "condition_variable_wrapper.h"
#include <pthread.h>
namespace webrtc {
-class ConditionVariableLinux : public ConditionVariableWrapper
+class ConditionVariablePosix : public ConditionVariableWrapper
{
public:
static ConditionVariableWrapper* Create();
- ~ConditionVariableLinux();
+ ~ConditionVariablePosix();
void SleepCS(CriticalSectionWrapper& critSect);
bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
@@ -28,7 +28,7 @@
void WakeAll();
private:
- ConditionVariableLinux();
+ ConditionVariablePosix();
int Construct();
private:
@@ -36,4 +36,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
diff --git a/src/system_wrappers/source/cpu.cc b/src/system_wrappers/source/cpu.cc
index 2285872..3df5d18 100644
--- a/src/system_wrappers/source/cpu.cc
+++ b/src/system_wrappers/source/cpu.cc
@@ -11,74 +11,25 @@
#include "cpu_wrapper.h"
#if defined(_WIN32)
- #include <Windows.h>
- #include "engine_configurations.h"
- #include "cpu_windows.h"
+ #include "cpu_win.h"
#elif defined(WEBRTC_MAC)
- #include <sys/types.h>
- #include <sys/sysctl.h>
#include "cpu_mac.h"
#elif defined(WEBRTC_MAC_INTEL)
#include "cpu_mac.h"
-#elif defined(ANDROID)
+#elif defined(WEBRTC_ANDROID)
// Not implemented yet, might be possible to use Linux implementation
#else // defined(WEBRTC_LINUX)
- #include <sys/sysinfo.h>
#include "cpu_linux.h"
#endif
-#include "trace.h"
-
namespace webrtc {
-WebRtc_UWord32 CpuWrapper::_numberOfCores = 0;
-
-WebRtc_UWord32 CpuWrapper::DetectNumberOfCores()
-{
- if (!_numberOfCores)
- {
-#if defined(_WIN32)
- SYSTEM_INFO si;
- GetSystemInfo(&si);
- _numberOfCores = static_cast<WebRtc_UWord32>(si.dwNumberOfProcessors);
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
-
-#elif defined(WEBRTC_LINUX) && !defined(ANDROID)
- _numberOfCores = get_nprocs();
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
-
-#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
- int name[] = {CTL_HW, HW_AVAILCPU};
- int ncpu;
- size_t size = sizeof(ncpu);
- if(0 == sysctl(name, 2, &ncpu, &size, NULL, 0))
- {
- _numberOfCores = static_cast<WebRtc_UWord32>(ncpu);
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "Failed to get number of cores");
- _numberOfCores = 1;
- }
-#else
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
- "No function to get number of cores");
- _numberOfCores = 1;
-#endif
- }
- return _numberOfCores;
-}
-
CpuWrapper* CpuWrapper::CreateCpu()
{
#if defined(_WIN32)
return new CpuWindows();
#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
return new CpuWrapperMac();
-#elif defined(ANDROID)
+#elif defined(WEBRTC_ANDROID)
return 0;
#else
return new CpuLinux();
diff --git a/src/system_wrappers/source/cpu_features.cc b/src/system_wrappers/source/cpu_features.cc
index 850dc9b..41a86e3 100644
--- a/src/system_wrappers/source/cpu_features.cc
+++ b/src/system_wrappers/source/cpu_features.cc
@@ -8,17 +8,29 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+// Parts of this file derived from Chromium's base/cpu.cc.
+
#include "cpu_features_wrapper.h"
+#include "typedefs.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(_MSC_VER)
+#include <intrin.h>
+#endif
+#endif
+
// No CPU feature is available => straight C path.
int GetCPUInfoNoASM(CPUFeature feature) {
(void)feature;
return 0;
}
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#ifndef _MSC_VER
// Intrinsic for "cpuid".
#if defined(__pic__) && defined(__i386__)
-static inline void cpuid(int cpu_info[4], int info_type) {
+static inline void __cpuid(int cpu_info[4], int info_type) {
__asm__ volatile (
"mov %%ebx, %%edi\n"
"cpuid\n"
@@ -26,20 +38,22 @@
: "=a"(cpu_info[0]), "=D"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
: "a"(info_type));
}
-#elif defined(__i386__) || defined(__x86_64__)
-static inline void cpuid(int cpu_info[4], int info_type) {
+#else
+static inline void __cpuid(int cpu_info[4], int info_type) {
__asm__ volatile (
"cpuid\n"
: "=a"(cpu_info[0]), "=b"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
: "a"(info_type));
}
#endif
+#endif // _MSC_VER
+#endif // WEBRTC_ARCH_X86_FAMILY
-#if defined(__i386__) || defined(__x86_64__)
+#if defined(WEBRTC_ARCH_X86_FAMILY)
// Actual feature detection for x86.
static int GetCPUInfo(CPUFeature feature) {
int cpu_info[4];
- cpuid(cpu_info, 1);
+ __cpuid(cpu_info, 1);
if (feature == kSSE2) {
return 0 != (cpu_info[3] & 0x04000000);
}
diff --git a/src/system_wrappers/source/cpu_features_arm.c b/src/system_wrappers/source/cpu_features_arm.c
new file mode 100644
index 0000000..1065118
--- /dev/null
+++ b/src/system_wrappers/source/cpu_features_arm.c
@@ -0,0 +1,333 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is derived from Android's NDK package r7, located at
+// <ndk>/sources/android/cpufeatures/ (downloadable from
+// http://developer.android.com/sdk/ndk/index.html).
+
+#include "cpu_features_wrapper.h"
+
+#include <fcntl.h>
+#include <errno.h>
+#include <pthread.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+// Define CPU family.
+typedef enum {
+ CPU_FAMILY_UNKNOWN = 0,
+ CPU_FAMILY_ARM,
+ CPU_FAMILY_X86,
+ CPU_FAMILY_MAX // Do not remove.
+} CpuFamily;
+
+static pthread_once_t g_once;
+static CpuFamily g_cpuFamily;
+static uint64_t g_cpuFeatures;
+static int g_cpuCount;
+
+static const int cpufeatures_debug = 0;
+
+#ifdef __arm__
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_ARM
+#elif defined __i386__
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_X86
+#else
+# define DEFAULT_CPU_FAMILY CPU_FAMILY_UNKNOWN
+#endif
+
+#define D(...) \
+ do { \
+ if (cpufeatures_debug) { \
+ printf(__VA_ARGS__); fflush(stdout); \
+ } \
+ } while (0)
+
+/* Read the content of /proc/cpuinfo into a user-provided buffer.
+ * Return the length of the data, or -1 on error. Does *not*
+ * zero-terminate the content. Will not read more
+ * than 'buffsize' bytes.
+ */
+static int read_file(const char* pathname, char* buffer, size_t buffsize) {
+ int fd, len;
+
+ fd = open(pathname, O_RDONLY);
+ if (fd < 0)
+ return -1;
+
+ do {
+ len = read(fd, buffer, buffsize);
+ } while (len < 0 && errno == EINTR);
+
+ close(fd);
+
+ return len;
+}
+
+/* Extract the content of a the first occurence of a given field in
+ * the content of /proc/cpuinfo and return it as a heap-allocated
+ * string that must be freed by the caller.
+ *
+ * Return NULL if not found
+ */
+static char* extract_cpuinfo_field(char* buffer, int buflen, const char* field) {
+ int fieldlen = strlen(field);
+ char* bufend = buffer + buflen;
+ char* result = NULL;
+ int len, ignore;
+ const char* p, *q;
+
+ /* Look for first field occurence, and ensures it starts the line.
+ */
+ p = buffer;
+ bufend = buffer + buflen;
+ for (;;) {
+ p = memmem(p, bufend - p, field, fieldlen);
+ if (p == NULL)
+ goto EXIT;
+
+ if (p == buffer || p[-1] == '\n')
+ break;
+
+ p += fieldlen;
+ }
+
+ /* Skip to the first column followed by a space */
+ p += fieldlen;
+ p = memchr(p, ':', bufend - p);
+ if (p == NULL || p[1] != ' ')
+ goto EXIT;
+
+ /* Find the end of the line */
+ p += 2;
+ q = memchr(p, '\n', bufend - p);
+ if (q == NULL)
+ q = bufend;
+
+ /* Copy the line into a heap-allocated buffer */
+ len = q - p;
+ result = malloc(len + 1);
+ if (result == NULL)
+ goto EXIT;
+
+ memcpy(result, p, len);
+ result[len] = '\0';
+
+EXIT:
+ return result;
+}
+
+/* Count the number of occurences of a given field prefix in /proc/cpuinfo.
+ */
+static int count_cpuinfo_field(char* buffer, int buflen, const char* field) {
+ int fieldlen = strlen(field);
+ const char* p = buffer;
+ const char* bufend = buffer + buflen;
+ const char* q;
+ int count = 0;
+
+ for (;;) {
+ const char* q;
+
+ p = memmem(p, bufend - p, field, fieldlen);
+ if (p == NULL)
+ break;
+
+ /* Ensure that the field is at the start of a line */
+ if (p > buffer && p[-1] != '\n') {
+ p += fieldlen;
+ continue;
+ }
+
+
+ /* skip any whitespace */
+ q = p + fieldlen;
+ while (q < bufend && (*q == ' ' || *q == '\t'))
+ q++;
+
+ /* we must have a colon now */
+ if (q < bufend && *q == ':') {
+ count += 1;
+ q ++;
+ }
+ p = q;
+ }
+
+ return count;
+}
+
+/* Like strlen(), but for constant string literals */
+#define STRLEN_CONST(x) ((sizeof(x)-1)
+
+
+/* Checks that a space-separated list of items contains one given 'item'.
+ * Returns 1 if found, 0 otherwise.
+ */
+static int has_list_item(const char* list, const char* item) {
+ const char* p = list;
+ int itemlen = strlen(item);
+
+ if (list == NULL)
+ return 0;
+
+ while (*p) {
+ const char* q;
+
+ /* skip spaces */
+ while (*p == ' ' || *p == '\t')
+ p++;
+
+ /* find end of current list item */
+ q = p;
+ while (*q && *q != ' ' && *q != '\t')
+ q++;
+
+ if (itemlen == q - p && !memcmp(p, item, itemlen))
+ return 1;
+
+ /* skip to next item */
+ p = q;
+ }
+ return 0;
+}
+
+
+static void cpuInit(void) {
+ char cpuinfo[4096];
+ int cpuinfo_len;
+
+ g_cpuFamily = DEFAULT_CPU_FAMILY;
+ g_cpuFeatures = 0;
+ g_cpuCount = 1;
+
+ cpuinfo_len = read_file("/proc/cpuinfo", cpuinfo, sizeof cpuinfo);
+ D("cpuinfo_len is (%d):\n%.*s\n", cpuinfo_len,
+ cpuinfo_len >= 0 ? cpuinfo_len : 0, cpuinfo);
+
+ if (cpuinfo_len < 0) { /* should not happen */
+ return;
+ }
+
+ /* Count the CPU cores, the value may be 0 for single-core CPUs */
+ g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "processor");
+ if (g_cpuCount == 0) {
+ g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "Processor");
+ if (g_cpuCount == 0) {
+ g_cpuCount = 1;
+ }
+ }
+
+ D("found cpuCount = %d\n", g_cpuCount);
+
+#ifdef __arm__
+ {
+ char* features = NULL;
+ char* architecture = NULL;
+
+ /* Extract architecture from the "CPU Architecture" field.
+ * The list is well-known, unlike the the output of
+ * the 'Processor' field which can vary greatly.
+ *
+ * See the definition of the 'proc_arch' array in
+ * $KERNEL/arch/arm/kernel/setup.c and the 'c_show' function in
+ * same file.
+ */
+ char* cpuArch = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "CPU architecture");
+
+ if (cpuArch != NULL) {
+ char* end;
+ long archNumber;
+ int hasARMv7 = 0;
+
+ D("found cpuArch = '%s'\n", cpuArch);
+
+ /* read the initial decimal number, ignore the rest */
+ archNumber = strtol(cpuArch, &end, 10);
+
+ /* Here we assume that ARMv8 will be upwards compatible with v7
+ * in the future. Unfortunately, there is no 'Features' field to
+ * indicate that Thumb-2 is supported.
+ */
+ if (end > cpuArch && archNumber >= 7) {
+ hasARMv7 = 1;
+ }
+
+ /* Unfortunately, it seems that certain ARMv6-based CPUs
+ * report an incorrect architecture number of 7!
+ *
+ * We try to correct this by looking at the 'elf_format'
+ * field reported by the 'Processor' field, which is of the
+ * form of "(v7l)" for an ARMv7-based CPU, and "(v6l)" for
+ * an ARMv6-one.
+ */
+ if (hasARMv7) {
+ char* cpuProc = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "Processor");
+ if (cpuProc != NULL) {
+ D("found cpuProc = '%s'\n", cpuProc);
+ if (has_list_item(cpuProc, "(v6l)")) {
+ D("CPU processor and architecture mismatch!!\n");
+ hasARMv7 = 0;
+ }
+ free(cpuProc);
+ }
+ }
+
+ if (hasARMv7) {
+ g_cpuFeatures |= kCPUFeatureARMv7;
+ }
+
+ /* The LDREX / STREX instructions are available from ARMv6 */
+ if (archNumber >= 6) {
+ g_cpuFeatures |= kCPUFeatureLDREXSTREX;
+ }
+
+ free(cpuArch);
+ }
+
+ /* Extract the list of CPU features from 'Features' field */
+ char* cpuFeatures = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
+ "Features");
+
+ if (cpuFeatures != NULL) {
+
+ D("found cpuFeatures = '%s'\n", cpuFeatures);
+
+ if (has_list_item(cpuFeatures, "vfpv3"))
+ g_cpuFeatures |= kCPUFeatureVFPv3;
+
+ else if (has_list_item(cpuFeatures, "vfpv3d16"))
+ g_cpuFeatures |= kCPUFeatureVFPv3;
+
+ if (has_list_item(cpuFeatures, "neon")) {
+ /* Note: Certain kernels only report neon but not vfpv3
+ * in their features list. However, ARM mandates
+ * that if Neon is implemented, so must be VFPv3
+ * so always set the flag.
+ */
+ g_cpuFeatures |= kCPUFeatureNEON |
+ kCPUFeatureVFPv3;
+ }
+ free(cpuFeatures);
+ }
+ }
+#endif // __arm__
+
+#ifdef __i386__
+ g_cpuFamily = CPU_FAMILY_X86;
+#endif
+}
+
+
+uint64_t WebRtc_GetCPUFeaturesARM(void) {
+ pthread_once(&g_once, cpuInit);
+ return g_cpuFeatures;
+}
diff --git a/src/system_wrappers/source/cpu_info.cc b/src/system_wrappers/source/cpu_info.cc
new file mode 100644
index 0000000..e367abf
--- /dev/null
+++ b/src/system_wrappers/source/cpu_info.cc
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "cpu_info.h"
+
+#if defined(_WIN32)
+#include <Windows.h>
+#elif defined(WEBRTC_MAC)
+#include <sys/types.h>
+#include <sys/sysctl.h>
+#elif defined(WEBRTC_MAC_INTEL)
+// Intentionally empty
+#elif defined(WEBRTC_ANDROID)
+// Not implemented yet, might be possible to use Linux implementation
+#else // defined(WEBRTC_LINUX)
+#include <sys/sysinfo.h>
+#endif
+
+#include "trace.h"
+
+namespace webrtc {
+
+WebRtc_UWord32 CpuInfo::_numberOfCores = 0;
+
+WebRtc_UWord32 CpuInfo::DetectNumberOfCores()
+{
+ if (!_numberOfCores)
+ {
+#if defined(_WIN32)
+ SYSTEM_INFO si;
+ GetSystemInfo(&si);
+ _numberOfCores = static_cast<WebRtc_UWord32>(si.dwNumberOfProcessors);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+
+#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+ _numberOfCores = get_nprocs();
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+
+#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
+ int name[] = {CTL_HW, HW_AVAILCPU};
+ int ncpu;
+ size_t size = sizeof(ncpu);
+ if(0 == sysctl(name, 2, &ncpu, &size, NULL, 0))
+ {
+ _numberOfCores = static_cast<WebRtc_UWord32>(ncpu);
+ WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
+ "Available number of cores:%d", _numberOfCores);
+ } else
+ {
+ WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
+ "Failed to get number of cores");
+ _numberOfCores = 1;
+ }
+#else
+ WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
+ "No function to get number of cores");
+ _numberOfCores = 1;
+#endif
+ }
+ return _numberOfCores;
+}
+
+} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_linux.cc b/src/system_wrappers/source/cpu_linux.cc
index eff9704..9d7d89d 100644
--- a/src/system_wrappers/source/cpu_linux.cc
+++ b/src/system_wrappers/source/cpu_linux.cc
@@ -17,22 +17,29 @@
namespace webrtc {
CpuLinux::CpuLinux()
-{
- m_oldBusyTime = 0;
- m_oldIdleTime = 0;
- m_numCores = 0;
- m_numCores = GetNumCores();
- m_oldBusyTimeMulti = new long long[m_numCores];
- memset(m_oldBusyTimeMulti, 0, sizeof(long long) * m_numCores);
- m_oldIdleTimeMulti = new long long[m_numCores];
- memset(m_oldIdleTimeMulti, 0, sizeof(long long) * m_numCores);
- m_idleArray = new long long[m_numCores];
- memset(m_idleArray, 0, sizeof(long long) * m_numCores);
- m_busyArray = new long long[m_numCores];
- memset(m_busyArray, 0, sizeof(long long) * m_numCores);
- m_resultArray = new WebRtc_UWord32[m_numCores];
+ : m_oldBusyTime(0),
+ m_oldIdleTime(0),
+ m_oldBusyTimeMulti(NULL),
+ m_oldIdleTimeMulti(NULL),
+ m_idleArray(NULL),
+ m_busyArray(NULL),
+ m_resultArray(NULL),
+ m_numCores(0) {
+ const int result = GetNumCores();
+ if (result != -1) {
+ m_numCores = result;
+ m_oldBusyTimeMulti = new long long[m_numCores];
+ memset(m_oldBusyTimeMulti, 0, sizeof(long long) * m_numCores);
+ m_oldIdleTimeMulti = new long long[m_numCores];
+ memset(m_oldIdleTimeMulti, 0, sizeof(long long) * m_numCores);
+ m_idleArray = new long long[m_numCores];
+ memset(m_idleArray, 0, sizeof(long long) * m_numCores);
+ m_busyArray = new long long[m_numCores];
+ memset(m_busyArray, 0, sizeof(long long) * m_numCores);
+ m_resultArray = new WebRtc_UWord32[m_numCores];
- GetData(m_oldBusyTime, m_oldIdleTime, m_busyArray, m_idleArray);
+ GetData(m_oldBusyTime, m_oldIdleTime, m_busyArray, m_idleArray);
+ }
}
CpuLinux::~CpuLinux()
@@ -58,7 +65,8 @@
numCores = m_numCores;
long long busy = 0;
long long idle = 0;
- GetData(busy, idle, m_busyArray, m_idleArray);
+ if (GetData(busy, idle, m_busyArray, m_idleArray) != 0)
+ return -1;
long long deltaBusy = busy - m_oldBusyTime;
long long deltaIdle = idle - m_oldIdleTime;
@@ -109,18 +117,28 @@
}
char line[100];
- char* dummy = fgets(line, 100, fp);
+ if (fgets(line, 100, fp) == NULL) {
+ fclose(fp);
+ return -1;
+ }
char firstWord[100];
- sscanf(line, "%s ", firstWord);
- if(strncmp(firstWord, "cpu", 3)!=0)
- {
+ if (sscanf(line, "%s ", firstWord) != 1) {
+ fclose(fp);
+ return -1;
+ }
+ if (strncmp(firstWord, "cpu", 3) != 0) {
+ fclose(fp);
return -1;
}
char sUser[100];
char sNice[100];
char sSystem[100];
char sIdle[100];
- sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem, sIdle);
+ if (sscanf(line, "%s %s %s %s %s ",
+ firstWord, sUser, sNice, sSystem, sIdle) != 5) {
+ fclose(fp);
+ return -1;
+ }
long long luser = atoll(sUser);
long long lnice = atoll(sNice);
long long lsystem = atoll(sSystem);
@@ -130,9 +148,15 @@
idle = lidle;
for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
{
- dummy = fgets(line, 100, fp);
- sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem,
- sIdle);
+ if (fgets(line, 100, fp) == NULL) {
+ fclose(fp);
+ return -1;
+ }
+ if (sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem,
+ sIdle) != 5) {
+ fclose(fp);
+ return -1;
+ }
luser = atoll(sUser);
lnice = atoll(sNice);
lsystem = atoll(sSystem);
@@ -153,7 +177,10 @@
}
// Skip first line
char line[100];
- char* dummy = fgets(line, 100, fp);
+ if (!fgets(line, 100, fp))
+ {
+ return -1;
+ }
int numCores = -1;
char firstWord[100];
do
@@ -161,7 +188,9 @@
numCores++;
if (fgets(line, 100, fp))
{
- sscanf(line, "%s ", firstWord);
+ if (sscanf(line, "%s ", firstWord) != 1) {
+ firstWord[0] = '\0';
+ }
} else {
break;
}
diff --git a/src/system_wrappers/source/cpu_mac.cc b/src/system_wrappers/source/cpu_mac.cc
index c2a11e1..d82bf07 100644
--- a/src/system_wrappers/source/cpu_mac.cc
+++ b/src/system_wrappers/source/cpu_mac.cc
@@ -17,7 +17,12 @@
#include "tick_util.h"
namespace webrtc {
-CpuWrapperMac::CpuWrapperMac() : _cpuUsage(NULL)
+CpuWrapperMac::CpuWrapperMac()
+ : _cpuCount(0),
+ _cpuUsage(NULL),
+ _totalCpuUsage(0),
+ _lastTickCount(NULL),
+ _lastTime(0)
{
natural_t cpuCount;
processor_info_array_t infoArray;
@@ -33,6 +38,7 @@
return;
}
+ _cpuCount = cpuCount;
_cpuUsage = new WebRtc_UWord32[cpuCount];
_lastTickCount = new WebRtc_Word64[cpuCount];
_lastTime = TickTime::MillisecondTimestamp();
@@ -47,14 +53,15 @@
ticks += cpuLoadInfo[cpu].cpu_ticks[state];
}
_lastTickCount[cpu] = ticks;
+ _cpuUsage[cpu] = 0;
}
vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
}
CpuWrapperMac::~CpuWrapperMac()
{
- delete _cpuUsage;
- delete _lastTickCount;
+ delete[] _cpuUsage;
+ delete[] _lastTickCount;
}
WebRtc_Word32 CpuWrapperMac::CpuUsage()
@@ -68,29 +75,35 @@
CpuWrapperMac::CpuUsageMultiCore(WebRtc_UWord32& numCores,
WebRtc_UWord32*& array)
{
- natural_t cpuCount;
- processor_info_array_t infoArray;
- mach_msg_type_number_t infoCount;
-
// sanity check
if(_cpuUsage == NULL)
{
return -1;
}
+
WebRtc_Word64 now = TickTime::MillisecondTimestamp();
WebRtc_Word64 timeDiffMS = now - _lastTime;
- // TODO(hellner) why block here? Why not just return the old
- // value? Is this behavior consistent across all
- // platforms?
- // Make sure that at least 500 ms pass between calls.
- if(timeDiffMS < 500)
+ if(timeDiffMS >= 500)
{
- usleep((500-timeDiffMS)*1000);
- return CpuUsageMultiCore(numCores, array);
+ if(Update(timeDiffMS) != 0)
+ {
+ return -1;
+ }
+ _lastTime = now;
}
- _lastTime = now;
+
+ numCores = _cpuCount;
+ array = _cpuUsage;
+ return _totalCpuUsage / _cpuCount;
+}
- kern_return_t error = host_processor_info(mach_host_self(),
+WebRtc_Word32 CpuWrapperMac::Update(WebRtc_Word64 timeDiffMS)
+{
+ natural_t cpuCount;
+ processor_info_array_t infoArray;
+ mach_msg_type_number_t infoCount;
+
+ kern_return_t error = host_processor_info(mach_host_self(),
PROCESSOR_CPU_LOAD_INFO,
&cpuCount,
&infoArray,
@@ -103,7 +116,7 @@
processor_cpu_load_info_data_t* cpuLoadInfo =
(processor_cpu_load_info_data_t*) infoArray;
- WebRtc_Word32 totalCpuUsage = 0;
+ _totalCpuUsage = 0;
for (unsigned int cpu = 0; cpu < cpuCount; cpu++)
{
WebRtc_Word64 ticks = 0;
@@ -120,13 +133,11 @@
timeDiffMS);
}
_lastTickCount[cpu] = ticks;
- totalCpuUsage += _cpuUsage[cpu];
+ _totalCpuUsage += _cpuUsage[cpu];
}
vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
- numCores = cpuCount;
- array = _cpuUsage;
- return totalCpuUsage/cpuCount;
+ return 0;
}
} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_mac.h b/src/system_wrappers/source/cpu_mac.h
index 04cd097..f9f8207 100644
--- a/src/system_wrappers/source/cpu_mac.h
+++ b/src/system_wrappers/source/cpu_mac.h
@@ -35,7 +35,11 @@
virtual void Stop() {}
private:
+ WebRtc_Word32 Update(WebRtc_Word64 timeDiffMS);
+
+ WebRtc_UWord32 _cpuCount;
WebRtc_UWord32* _cpuUsage;
+ WebRtc_Word32 _totalCpuUsage;
WebRtc_Word64* _lastTickCount;
WebRtc_Word64 _lastTime;
};
diff --git a/src/system_wrappers/source/cpu_no_op.cc b/src/system_wrappers/source/cpu_no_op.cc
new file mode 100644
index 0000000..e42ef91
--- /dev/null
+++ b/src/system_wrappers/source/cpu_no_op.cc
@@ -0,0 +1,22 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include "cpu_wrapper.h"
+
+namespace webrtc {
+
+CpuWrapper* CpuWrapper::CreateCpu()
+{
+ return NULL;
+}
+
+} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_wrapper_unittest.cc b/src/system_wrappers/source/cpu_wrapper_unittest.cc
new file mode 100644
index 0000000..dd49c3a
--- /dev/null
+++ b/src/system_wrappers/source/cpu_wrapper_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/cpu_wrapper.h"
+
+#include "gtest/gtest.h"
+#include "system_wrappers/interface/cpu_info.h"
+#include "system_wrappers/interface/event_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+#include "system_wrappers/interface/trace.h"
+#include "testsupport/fileutils.h"
+
+using webrtc::CpuInfo;
+using webrtc::CpuWrapper;
+using webrtc::EventWrapper;
+using webrtc::scoped_ptr;
+using webrtc::Trace;
+
+TEST(CpuWrapperTest, Usage) {
+ Trace::CreateTrace();
+ std::string trace_file = webrtc::test::OutputPath() +
+ "cpu_wrapper_unittest.txt";
+ Trace::SetTraceFile(trace_file.c_str());
+ Trace::SetLevelFilter(webrtc::kTraceAll);
+ printf("Number of cores detected:%u\n", CpuInfo::DetectNumberOfCores());
+ scoped_ptr<CpuWrapper> cpu(CpuWrapper::CreateCpu());
+ ASSERT_TRUE(cpu.get() != NULL);
+ scoped_ptr<EventWrapper> sleep_event(EventWrapper::Create());
+ ASSERT_TRUE(sleep_event.get() != NULL);
+
+ int num_iterations = 0;
+ WebRtc_UWord32 num_cores = 0;
+ WebRtc_UWord32* cores = NULL;
+ bool cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
+ // Initializing the CPU measurements may take a couple of seconds on Windows.
+ // Since the initialization is lazy we need to wait until it is completed.
+ // Should not take more than 10000 ms.
+ while (cpu_usage_available && (++num_iterations < 10000)) {
+ if (cores != NULL) {
+ ASSERT_GT(num_cores, 0u);
+ break;
+ }
+ sleep_event->Wait(1);
+ cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
+ }
+ ASSERT_TRUE(cpu_usage_available);
+
+ const WebRtc_Word32 average = cpu->CpuUsageMultiCore(num_cores, cores);
+ ASSERT_TRUE(cores != NULL);
+ EXPECT_GT(num_cores, 0u);
+ EXPECT_GE(average, 0);
+ EXPECT_LE(average, 100);
+
+ printf("\nNumber of cores:%d\n", num_cores);
+ printf("Average cpu:%d\n", average);
+ for (WebRtc_UWord32 i = 0; i < num_cores; i++) {
+ printf("Core:%u CPU:%u \n", i, cores[i]);
+ EXPECT_GE(cores[i], 0u);
+ EXPECT_LE(cores[i], 100u);
+ }
+
+ Trace::ReturnTrace();
+};
diff --git a/src/system_wrappers/source/critical_section.cc b/src/system_wrappers/source/critical_section.cc
index 213c352..d3f3f01 100644
--- a/src/system_wrappers/source/critical_section.cc
+++ b/src/system_wrappers/source/critical_section.cc
@@ -10,9 +10,9 @@
#if defined(_WIN32)
#include <windows.h>
- #include "critical_section_windows.h"
+ #include "critical_section_win.h"
#else
- #include "critical_section_linux.h"
+ #include "critical_section_posix.h"
#endif
namespace webrtc {
@@ -21,7 +21,7 @@
#ifdef _WIN32
return new CriticalSectionWindows();
#else
- return new CriticalSectionLinux();
+ return new CriticalSectionPosix();
#endif
}
} // namespace webrtc
diff --git a/src/system_wrappers/source/critical_section_linux.cc b/src/system_wrappers/source/critical_section_posix.cc
similarity index 79%
rename from src/system_wrappers/source/critical_section_linux.cc
rename to src/system_wrappers/source/critical_section_posix.cc
index 35e81ae..b499b9f 100644
--- a/src/system_wrappers/source/critical_section_linux.cc
+++ b/src/system_wrappers/source/critical_section_posix.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "critical_section_linux.h"
+#include "critical_section_posix.h"
namespace webrtc {
-CriticalSectionLinux::CriticalSectionLinux()
+CriticalSectionPosix::CriticalSectionPosix()
{
pthread_mutexattr_t attr;
pthread_mutexattr_init(&attr);
@@ -19,19 +19,19 @@
pthread_mutex_init(&_mutex, &attr);
}
-CriticalSectionLinux::~CriticalSectionLinux()
+CriticalSectionPosix::~CriticalSectionPosix()
{
pthread_mutex_destroy(&_mutex);
}
void
-CriticalSectionLinux::Enter()
+CriticalSectionPosix::Enter()
{
pthread_mutex_lock(&_mutex);
}
void
-CriticalSectionLinux::Leave()
+CriticalSectionPosix::Leave()
{
pthread_mutex_unlock(&_mutex);
}
diff --git a/src/system_wrappers/source/critical_section_linux.h b/src/system_wrappers/source/critical_section_posix.h
similarity index 63%
rename from src/system_wrappers/source/critical_section_linux.h
rename to src/system_wrappers/source/critical_section_posix.h
index 5ada1cb..40b7dc9 100644
--- a/src/system_wrappers/source/critical_section_linux.h
+++ b/src/system_wrappers/source/critical_section_posix.h
@@ -8,28 +8,28 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
#include "critical_section_wrapper.h"
#include <pthread.h>
namespace webrtc {
-class CriticalSectionLinux : public CriticalSectionWrapper
+class CriticalSectionPosix : public CriticalSectionWrapper
{
public:
- CriticalSectionLinux();
+ CriticalSectionPosix();
- virtual ~CriticalSectionLinux();
+ virtual ~CriticalSectionPosix();
virtual void Enter();
virtual void Leave();
private:
pthread_mutex_t _mutex;
- friend class ConditionVariableLinux;
+ friend class ConditionVariablePosix;
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
diff --git a/src/system_wrappers/source/data_log.cc b/src/system_wrappers/source/data_log.cc
new file mode 100644
index 0000000..f123896
--- /dev/null
+++ b/src/system_wrappers/source/data_log.cc
@@ -0,0 +1,455 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "data_log.h"
+
+#include <assert.h>
+
+#include <algorithm>
+#include <list>
+
+#include "critical_section_wrapper.h"
+#include "event_wrapper.h"
+#include "file_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "thread_wrapper.h"
+
+namespace webrtc {
+
+DataLogImpl::CritSectScopedPtr DataLogImpl::crit_sect_(
+ CriticalSectionWrapper::CreateCriticalSection());
+
+DataLogImpl* DataLogImpl::instance_ = NULL;
+
+// A Row contains cells, which are indexed by the column names as std::string.
+// The string index is treated in a case sensitive way.
+class Row {
+ public:
+ Row();
+ ~Row();
+
+ // Inserts a Container into the cell of the column specified with
+ // column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& column_name,
+ const Container* value_container);
+
+ // Converts the value at the column specified by column_name to a string
+ // stored in value_string.
+ // column_name is treated in a case sensitive way.
+ void ToString(const std::string& column_name, std::string* value_string);
+
+ private:
+ // Collection of containers indexed by column name as std::string
+ typedef std::map<std::string, const Container*> CellMap;
+
+ CellMap cells_;
+ CriticalSectionWrapper* cells_lock_;
+};
+
+// A LogTable contains multiple rows, where only the latest row is active for
+// editing. The rows are defined by the ColumnMap, which contains the name of
+// each column and the length of the column (1 for one-value-columns and greater
+// than 1 for multi-value-columns).
+class LogTable {
+ public:
+ LogTable();
+ ~LogTable();
+
+ // Adds the column with name column_name to the table. The column will be a
+ // multi-value-column if multi_value_length is greater than 1.
+ // column_name is treated in a case sensitive way.
+ int AddColumn(const std::string& column_name, int multi_value_length);
+
+ // Buffers the current row while it is waiting to be written to file,
+ // which is done by a call to Flush(). A new row is available when the
+ // function returns
+ void NextRow();
+
+ // Inserts a Container into the cell of the column specified with
+ // column_name.
+ // column_name is treated in a case sensitive way.
+ int InsertCell(const std::string& column_name,
+ const Container* value_container);
+
+ // Creates a log file, named as specified in the string file_name, to
+ // where the table will be written when calling Flush().
+ int CreateLogFile(const std::string& file_name);
+
+ // Write all complete rows to file.
+ // May not be called by two threads simultaneously (doing so may result in
+ // a race condition). Will be called by the file_writer_thread_ when that
+ // thread is running.
+ void Flush();
+
+ private:
+ // Collection of multi_value_lengths indexed by column name as std::string
+ typedef std::map<std::string, int> ColumnMap;
+ typedef std::list<Row*> RowList;
+
+ ColumnMap columns_;
+ RowList rows_[2];
+ RowList* rows_history_;
+ RowList* rows_flush_;
+ Row* current_row_;
+ FileWrapper* file_;
+ bool write_header_;
+ CriticalSectionWrapper* table_lock_;
+};
+
+Row::Row()
+ : cells_(),
+ cells_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
+
+Row::~Row() {
+ for (CellMap::iterator it = cells_.begin(); it != cells_.end();) {
+ delete it->second;
+ // For maps all iterators (except the erased) are valid after an erase
+ cells_.erase(it++);
+ }
+ delete cells_lock_;
+}
+
+int Row::InsertCell(const std::string& column_name,
+ const Container* value_container) {
+ CriticalSectionScoped synchronize(cells_lock_);
+ assert(cells_.count(column_name) == 0);
+ if (cells_.count(column_name) > 0)
+ return -1;
+ cells_[column_name] = value_container;
+ return 0;
+}
+
+void Row::ToString(const std::string& column_name,
+ std::string* value_string) {
+ CriticalSectionScoped synchronize(cells_lock_);
+ const Container* container = cells_[column_name];
+ if (container == NULL) {
+ *value_string = "NaN,";
+ return;
+ }
+ container->ToString(value_string);
+}
+
+LogTable::LogTable()
+ : columns_(),
+ rows_(),
+ rows_history_(&rows_[0]),
+ rows_flush_(&rows_[1]),
+ current_row_(new Row),
+ file_(FileWrapper::Create()),
+ write_header_(true),
+ table_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
+
+LogTable::~LogTable() {
+ for (RowList::iterator row_it = rows_history_->begin();
+ row_it != rows_history_->end();) {
+ delete *row_it;
+ row_it = rows_history_->erase(row_it);
+ }
+ for (ColumnMap::iterator col_it = columns_.begin();
+ col_it != columns_.end();) {
+ // For maps all iterators (except the erased) are valid after an erase
+ columns_.erase(col_it++);
+ }
+ if (file_ != NULL) {
+ file_->Flush();
+ file_->CloseFile();
+ delete file_;
+ }
+ delete current_row_;
+ delete table_lock_;
+}
+
+int LogTable::AddColumn(const std::string& column_name,
+ int multi_value_length) {
+ assert(multi_value_length > 0);
+ if (!write_header_) {
+ // It's not allowed to add new columns after the header
+ // has been written.
+ assert(false);
+ return -1;
+ } else {
+ CriticalSectionScoped synchronize(table_lock_);
+ if (write_header_)
+ columns_[column_name] = multi_value_length;
+ else
+ return -1;
+ }
+ return 0;
+}
+
+void LogTable::NextRow() {
+ CriticalSectionScoped sync_rows(table_lock_);
+ rows_history_->push_back(current_row_);
+ current_row_ = new Row;
+}
+
+int LogTable::InsertCell(const std::string& column_name,
+ const Container* value_container) {
+ CriticalSectionScoped synchronize(table_lock_);
+ assert(columns_.count(column_name) > 0);
+ if (columns_.count(column_name) == 0)
+ return -1;
+ return current_row_->InsertCell(column_name, value_container);
+}
+
+int LogTable::CreateLogFile(const std::string& file_name) {
+ if (file_name.length() == 0)
+ return -1;
+ if (file_->Open())
+ return -1;
+ file_->OpenFile(file_name.c_str(),
+ false, // Open with read/write permissions
+ false, // Don't wraparound and write at the beginning when
+ // the file is full
+ true); // Open as a text file
+ if (file_ == NULL)
+ return -1;
+ return 0;
+}
+
+void LogTable::Flush() {
+ ColumnMap::iterator column_it;
+ bool commit_header = false;
+ if (write_header_) {
+ CriticalSectionScoped synchronize(table_lock_);
+ if (write_header_) {
+ commit_header = true;
+ write_header_ = false;
+ }
+ }
+ if (commit_header) {
+ for (column_it = columns_.begin();
+ column_it != columns_.end(); ++column_it) {
+ if (column_it->second > 1) {
+ file_->WriteText("%s[%u],", column_it->first.c_str(),
+ column_it->second);
+ for (int i = 1; i < column_it->second; ++i)
+ file_->WriteText(",");
+ } else {
+ file_->WriteText("%s,", column_it->first.c_str());
+ }
+ }
+ if (columns_.size() > 0)
+ file_->WriteText("\n");
+ }
+
+ // Swap the list used for flushing with the list containing the row history
+ // and clear the history. We also create a local pointer to the new
+ // list used for flushing to avoid race conditions if another thread
+ // calls this function while we are writing.
+ // We don't want to block the list while we're writing to file.
+ {
+ CriticalSectionScoped synchronize(table_lock_);
+ RowList* tmp = rows_flush_;
+ rows_flush_ = rows_history_;
+ rows_history_ = tmp;
+ rows_history_->clear();
+ }
+
+ // Write all complete rows to file and delete them
+ for (RowList::iterator row_it = rows_flush_->begin();
+ row_it != rows_flush_->end();) {
+ for (column_it = columns_.begin();
+ column_it != columns_.end(); ++column_it) {
+ std::string row_string;
+ (*row_it)->ToString(column_it->first, &row_string);
+ file_->WriteText("%s", row_string.c_str());
+ }
+ if (columns_.size() > 0)
+ file_->WriteText("\n");
+ delete *row_it;
+ row_it = rows_flush_->erase(row_it);
+ }
+}
+
+int DataLog::CreateLog() {
+ return DataLogImpl::CreateLog();
+}
+
+void DataLog::ReturnLog() {
+ return DataLogImpl::ReturnLog();
+}
+
+std::string DataLog::Combine(const std::string& table_name, int table_id) {
+ std::stringstream ss;
+ std::string combined_id = table_name;
+ std::string number_suffix;
+ ss << "_" << table_id;
+ ss >> number_suffix;
+ combined_id += number_suffix;
+ std::transform(combined_id.begin(), combined_id.end(), combined_id.begin(),
+ ::tolower);
+ return combined_id;
+}
+
+int DataLog::AddTable(const std::string& table_name) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->AddTable(table_name);
+}
+
+int DataLog::AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->DataLogImpl::StaticInstance()->AddColumn(table_name,
+ column_name,
+ multi_value_length);
+}
+
+int DataLog::NextRow(const std::string& table_name) {
+ DataLogImpl* data_log = DataLogImpl::StaticInstance();
+ if (data_log == NULL)
+ return -1;
+ return data_log->DataLogImpl::StaticInstance()->NextRow(table_name);
+}
+
+DataLogImpl::DataLogImpl()
+ : counter_(1),
+ tables_(),
+ flush_event_(EventWrapper::Create()),
+ file_writer_thread_(NULL),
+ tables_lock_(RWLockWrapper::CreateRWLock()) {
+}
+
+DataLogImpl::~DataLogImpl() {
+ StopThread();
+ Flush(); // Write any remaining rows
+ delete file_writer_thread_;
+ delete flush_event_;
+ for (TableMap::iterator it = tables_.begin(); it != tables_.end();) {
+ delete static_cast<LogTable*>(it->second);
+ // For maps all iterators (except the erased) are valid after an erase
+ tables_.erase(it++);
+ }
+ delete tables_lock_;
+}
+
+int DataLogImpl::CreateLog() {
+ CriticalSectionScoped synchronize(crit_sect_.get());
+ if (instance_ == NULL) {
+ instance_ = new DataLogImpl();
+ return instance_->Init();
+ } else {
+ ++instance_->counter_;
+ }
+ return 0;
+}
+
+int DataLogImpl::Init() {
+ file_writer_thread_ = ThreadWrapper::CreateThread(
+ DataLogImpl::Run,
+ instance_,
+ kHighestPriority,
+ "DataLog");
+ if (file_writer_thread_ == NULL)
+ return -1;
+ unsigned int thread_id = 0;
+ bool success = file_writer_thread_->Start(thread_id);
+ if (!success)
+ return -1;
+ return 0;
+}
+
+DataLogImpl* DataLogImpl::StaticInstance() {
+ return instance_;
+}
+
+void DataLogImpl::ReturnLog() {
+ CriticalSectionScoped synchronize(crit_sect_.get());
+ if (instance_ && instance_->counter_ > 1) {
+ --instance_->counter_;
+ return;
+ }
+ delete instance_;
+ instance_ = NULL;
+}
+
+int DataLogImpl::AddTable(const std::string& table_name) {
+ WriteLockScoped synchronize(*tables_lock_);
+ // Make sure we don't add a table which already exists
+ if (tables_.count(table_name) > 0)
+ return -1;
+ tables_[table_name] = new LogTable();
+ if (tables_[table_name]->CreateLogFile(table_name + ".txt") == -1)
+ return -1;
+ return 0;
+}
+
+int DataLogImpl::AddColumn(const std::string& table_name,
+ const std::string& column_name,
+ int multi_value_length) {
+ ReadLockScoped synchronize(*tables_lock_);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ return tables_[table_name]->AddColumn(column_name, multi_value_length);
+}
+
+int DataLogImpl::InsertCell(const std::string& table_name,
+ const std::string& column_name,
+ const Container* value_container) {
+ ReadLockScoped synchronize(*tables_lock_);
+ assert(tables_.count(table_name) > 0);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ return tables_[table_name]->InsertCell(column_name, value_container);
+}
+
+int DataLogImpl::NextRow(const std::string& table_name) {
+ ReadLockScoped synchronize(*tables_lock_);
+ if (tables_.count(table_name) == 0)
+ return -1;
+ tables_[table_name]->NextRow();
+ if (file_writer_thread_ == NULL) {
+ // Write every row to file as they get complete.
+ tables_[table_name]->Flush();
+ } else {
+ // Signal a complete row
+ flush_event_->Set();
+ }
+ return 0;
+}
+
+void DataLogImpl::Flush() {
+ ReadLockScoped synchronize(*tables_lock_);
+ for (TableMap::iterator it = tables_.begin(); it != tables_.end(); ++it) {
+ it->second->Flush();
+ }
+}
+
+bool DataLogImpl::Run(void* obj) {
+ static_cast<DataLogImpl*>(obj)->Process();
+ return true;
+}
+
+void DataLogImpl::Process() {
+ // Wait for a row to be complete
+ flush_event_->Wait(WEBRTC_EVENT_INFINITE);
+ Flush();
+}
+
+void DataLogImpl::StopThread() {
+ if (file_writer_thread_ != NULL) {
+ file_writer_thread_->SetNotAlive();
+ flush_event_->Set();
+ // Call Stop() repeatedly, waiting for the Flush() call in Process() to
+ // finish.
+ while (!file_writer_thread_->Stop()) continue;
+ }
+}
+
+} // namespace webrtc
diff --git a/src/system_wrappers/source/data_log_c.cc b/src/system_wrappers/source/data_log_c.cc
new file mode 100644
index 0000000..f8d7efd
--- /dev/null
+++ b/src/system_wrappers/source/data_log_c.cc
@@ -0,0 +1,145 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This is the pure C wrapper of the DataLog class.
+ */
+
+#include "system_wrappers/interface/data_log_c.h"
+
+#include <string>
+
+#include "system_wrappers/interface/data_log.h"
+
+extern "C" int WebRtcDataLog_CreateLog() {
+ return webrtc::DataLog::CreateLog();
+}
+
+extern "C" void WebRtcDataLog_ReturnLog() {
+ return webrtc::DataLog::ReturnLog();
+}
+
+extern "C" char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
+ const char* table_name, int table_id) {
+ if (!table_name) return NULL;
+ std::string combined = webrtc::DataLog::Combine(table_name, table_id);
+ if (combined.size() >= combined_len) return NULL;
+ std::copy(combined.begin(), combined.end(), combined_name);
+ combined_name[combined.size()] = '\0';
+ return combined_name;
+}
+
+extern "C" int WebRtcDataLog_AddTable(const char* table_name) {
+ if (!table_name) return -1;
+ return webrtc::DataLog::AddTable(table_name);
+}
+
+extern "C" int WebRtcDataLog_AddColumn(const char* table_name,
+ const char* column_name,
+ int multi_value_length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::AddColumn(table_name, column_name,
+ multi_value_length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int(const char* table_name,
+ const char* column_name,
+ int value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int(const char* table_name,
+ const char* column_name,
+ const int* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_float(const char* table_name,
+ const char* column_name,
+ float value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_float(const char* table_name,
+ const char* column_name,
+ const float* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_double(const char* table_name,
+ const char* column_name,
+ double value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_double(const char* table_name,
+ const char* column_name,
+ const double* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int32(const char* table_name,
+ const char* column_name,
+ int32_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int32(const char* table_name,
+ const char* column_name,
+ const int32_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_uint32(const char* table_name,
+ const char* column_name,
+ uint32_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_uint32(const char* table_name,
+ const char* column_name,
+ const uint32_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_InsertCell_int64(const char* table_name,
+ const char* column_name,
+ int64_t value) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, value);
+}
+
+extern "C" int WebRtcDataLog_InsertArray_int64(const char* table_name,
+ const char* column_name,
+ const int64_t* values,
+ int length) {
+ if (!table_name || !column_name) return -1;
+ return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
+}
+
+extern "C" int WebRtcDataLog_NextRow(const char* table_name) {
+ if (!table_name) return -1;
+ return webrtc::DataLog::NextRow(table_name);
+}
diff --git a/src/system_wrappers/source/data_log_c_helpers_unittest.c b/src/system_wrappers/source/data_log_c_helpers_unittest.c
new file mode 100644
index 0000000..e78a0e3
--- /dev/null
+++ b/src/system_wrappers/source/data_log_c_helpers_unittest.c
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/source/data_log_c_helpers_unittest.h"
+
+#include <assert.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "system_wrappers/interface/data_log_c.h"
+
+enum { kTestArrayLen = 4 };
+static const char kTableName[] = "c_wrapper_table";
+static const char kColumnName1[] = "Scalar";
+static const char kColumnName2[] = "Vector";
+
+int WebRtcDataLogCHelper_TestCreateLog() {
+ return WebRtcDataLog_CreateLog();
+}
+
+int WebRtcDataLogCHelper_TestReturnLog() {
+ WebRtcDataLog_ReturnLog();
+ return 0;
+}
+
+int WebRtcDataLogCHelper_TestCombine() {
+ const int kOutLen = strlen(kTableName) + 4; /* Room for "_17" + '\0' */
+ char* combined_name = malloc(kOutLen * sizeof(char));
+ char* out_ptr = WebRtcDataLog_Combine(combined_name, kOutLen, kTableName, 17);
+ int return_code = 0;
+ if (!out_ptr) {
+ return_code = -1;
+ }
+ if (strcmp(combined_name, "c_wrapper_table_17") != 0) {
+ return_code = -2;
+ }
+ free(combined_name);
+ return return_code;
+}
+
+int WebRtcDataLogCHelper_TestAddTable() {
+ return WebRtcDataLog_AddTable(kTableName);
+}
+
+int WebRtcDataLogCHelper_TestAddColumn() {
+ if (WebRtcDataLog_AddColumn(kTableName, kColumnName1, 1) != 0) {
+ return -1;
+ }
+ if (WebRtcDataLog_AddColumn(kTableName, kColumnName2, kTestArrayLen) != 0) {
+ return -2;
+ }
+ return 0;
+}
+
+int WebRtcDataLogCHelper_TestNextRow() {
+ return WebRtcDataLog_NextRow(kTableName);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int() {
+ return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int() {
+ int values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_float() {
+ return WebRtcDataLog_InsertCell_float(kTableName, kColumnName1, 17.0f);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_float() {
+ float values[kTestArrayLen] = {1.0f, 2.0f, 3.0f, 4.0f};
+ return WebRtcDataLog_InsertArray_float(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_double() {
+ return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17.0);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_double() {
+ double values[kTestArrayLen] = {1.0, 2.0, 3.0, 4.0};
+ return WebRtcDataLog_InsertArray_double(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int32() {
+ return WebRtcDataLog_InsertCell_int32(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int32() {
+ int32_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int32(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_uint32() {
+ return WebRtcDataLog_InsertCell_uint32(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_uint32() {
+ uint32_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_uint32(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
+
+int WebRtcDataLogCHelper_TestInsertCell_int64() {
+ return WebRtcDataLog_InsertCell_int64(kTableName, kColumnName1, 17);
+}
+
+int WebRtcDataLogCHelper_TestInsertArray_int64() {
+ int64_t values[kTestArrayLen] = {1, 2, 3, 4};
+ return WebRtcDataLog_InsertArray_int64(kTableName, kColumnName2, values,
+ kTestArrayLen);
+}
diff --git a/src/system_wrappers/source/data_log_c_helpers_unittest.h b/src/system_wrappers/source/data_log_c_helpers_unittest.h
new file mode 100644
index 0000000..ef86eae
--- /dev/null
+++ b/src/system_wrappers/source/data_log_c_helpers_unittest.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
+#define SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+int WebRtcDataLogCHelper_TestCreateLog();
+
+int WebRtcDataLogCHelper_TestReturnLog();
+
+int WebRtcDataLogCHelper_TestCombine();
+
+int WebRtcDataLogCHelper_TestAddTable();
+
+int WebRtcDataLogCHelper_TestAddColumn();
+
+int WebRtcDataLogCHelper_TestNextRow();
+
+int WebRtcDataLogCHelper_TestInsertCell_int();
+
+int WebRtcDataLogCHelper_TestInsertArray_int();
+
+int WebRtcDataLogCHelper_TestInsertCell_float();
+
+int WebRtcDataLogCHelper_TestInsertArray_float();
+
+int WebRtcDataLogCHelper_TestInsertCell_double();
+
+int WebRtcDataLogCHelper_TestInsertArray_double();
+
+int WebRtcDataLogCHelper_TestInsertCell_int32();
+
+int WebRtcDataLogCHelper_TestInsertArray_int32();
+
+int WebRtcDataLogCHelper_TestInsertCell_uint32();
+
+int WebRtcDataLogCHelper_TestInsertArray_uint32();
+
+int WebRtcDataLogCHelper_TestInsertCell_int64();
+
+int WebRtcDataLogCHelper_TestInsertArray_int64();
+
+#ifdef __cplusplus
+} // end of extern "C"
+#endif
+
+#endif // SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
diff --git a/src/system_wrappers/source/data_log_helpers_unittest.cc b/src/system_wrappers/source/data_log_helpers_unittest.cc
new file mode 100644
index 0000000..94b4d6e
--- /dev/null
+++ b/src/system_wrappers/source/data_log_helpers_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+
+#include "data_log.h"
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+TEST(TestDataLog, IntContainers) {
+ int c = 5;
+ webrtc::ValueContainer<int> v1(c);
+ c = 10;
+ webrtc::ValueContainer<int> v2(c);
+ std::string s1, s2;
+ v1.ToString(&s1);
+ v2.ToString(&s2);
+ ASSERT_EQ(s1, "5,");
+ ASSERT_EQ(s2, "10,");
+ v1 = v2;
+ v1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+}
+
+TEST(TestDataLog, DoubleContainers) {
+ double c = 3.5;
+ webrtc::ValueContainer<double> v1(c);
+ c = 10.3;
+ webrtc::ValueContainer<double> v2(c);
+ std::string s1, s2;
+ v1.ToString(&s1);
+ v2.ToString(&s2);
+ ASSERT_EQ(s1, "3.5,");
+ ASSERT_EQ(s2, "10.3,");
+ v1 = v2;
+ v1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+}
+
+TEST(TestDataLog, MultiValueContainers) {
+ int a[3] = {1, 2, 3};
+ int b[3] = {4, 5, 6};
+ webrtc::MultiValueContainer<int> m1(a, 3);
+ webrtc::MultiValueContainer<int> m2(b, 3);
+ webrtc::MultiValueContainer<int> m3(a, 3);
+ std::string s1, s2, s3;
+ m1.ToString(&s1);
+ m2.ToString(&s2);
+ ASSERT_EQ(s1, "1,2,3,");
+ ASSERT_EQ(s2, "4,5,6,");
+ m1 = m2;
+ m1.ToString(&s1);
+ ASSERT_EQ(s1, s2);
+ m3.ToString(&s3);
+ ASSERT_EQ(s3, "1,2,3,");
+}
diff --git a/src/system_wrappers/source/data_log_no_op.cc b/src/system_wrappers/source/data_log_no_op.cc
new file mode 100644
index 0000000..bedc82a
--- /dev/null
+++ b/src/system_wrappers/source/data_log_no_op.cc
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "data_log.h"
+
+#include <string>
+
+namespace webrtc {
+
+int DataLog::CreateLog() {
+ return 0;
+}
+
+void DataLog::ReturnLog() {
+}
+
+std::string DataLog::Combine(const std::string& table_name, int table_id) {
+ return std::string();
+}
+
+int DataLog::AddTable(const std::string& /*table_name*/) {
+ return 0;
+}
+
+int DataLog::AddColumn(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ int /*multi_value_length*/) {
+ return 0;
+}
+
+int DataLog::NextRow(const std::string& /*table_name*/) {
+ return 0;
+}
+
+DataLogImpl::DataLogImpl() {
+}
+
+DataLogImpl::~DataLogImpl() {
+}
+
+DataLogImpl* DataLogImpl::StaticInstance() {
+ return NULL;
+}
+
+void DataLogImpl::ReturnLog() {
+}
+
+int DataLogImpl::AddTable(const std::string& /*table_name*/) {
+ return 0;
+}
+
+int DataLogImpl::AddColumn(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ int /*multi_value_length*/) {
+ return 0;
+}
+
+int DataLogImpl::InsertCell(const std::string& /*table_name*/,
+ const std::string& /*column_name*/,
+ const Container* /*value_container*/) {
+ return 0;
+}
+
+int DataLogImpl::NextRow(const std::string& /*table_name*/) {
+ return 0;
+}
+
+void DataLogImpl::Flush() {
+}
+
+bool DataLogImpl::Run(void* /*obj*/) {
+ return true;
+}
+
+void DataLogImpl::Process() {
+}
+
+void DataLogImpl::StopThread() {
+}
+
+} // namespace webrtc
diff --git a/src/system_wrappers/source/data_log_unittest.cc b/src/system_wrappers/source/data_log_unittest.cc
new file mode 100644
index 0000000..c64ed94
--- /dev/null
+++ b/src/system_wrappers/source/data_log_unittest.cc
@@ -0,0 +1,310 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <map>
+#include <string>
+
+#include "system_wrappers/interface/data_log.h"
+#include "system_wrappers/interface/data_log_c.h"
+#include "system_wrappers/source/data_log_c_helpers_unittest.h"
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+// A class for storing the values expected from a log table column when
+// verifying a log table file.
+struct ExpectedValues {
+ public:
+ ExpectedValues()
+ : values(NULL),
+ multi_value_length(1) {
+ }
+
+ ExpectedValues(std::vector<std::string> expected_values,
+ int expected_multi_value_length)
+ : values(expected_values),
+ multi_value_length(expected_multi_value_length) {
+ }
+
+ std::vector<std::string> values;
+ int multi_value_length;
+};
+
+typedef std::map<std::string, ExpectedValues> ExpectedValuesMap;
+
+// A static class used for parsing and verifying data log files.
+class DataLogParser {
+ public:
+ // Verifies that the log table stored in the file "log_file" corresponds to
+ // the cells and columns specified in "columns".
+ static int VerifyTable(FILE* log_file, const ExpectedValuesMap& columns) {
+ int row = 0;
+ char line_buffer[kMaxLineLength];
+ char* ret = fgets(line_buffer, kMaxLineLength, log_file);
+ EXPECT_FALSE(ret == NULL);
+ if (ret == NULL)
+ return -1;
+
+ std::string line(line_buffer, kMaxLineLength);
+ VerifyHeader(line, columns);
+ while (fgets(line_buffer, kMaxLineLength, log_file) != NULL) {
+ line = std::string(line_buffer, kMaxLineLength);
+ size_t line_position = 0;
+
+ for (ExpectedValuesMap::const_iterator it = columns.begin();
+ it != columns.end(); ++it) {
+ std::string str = ParseElement(line, &line_position,
+ it->second.multi_value_length);
+ EXPECT_EQ(str, it->second.values[row]);
+ if (str != it->second.values[row])
+ return -1;
+ }
+ ++row;
+ }
+ return 0;
+ }
+
+ // Verifies the table header stored in "line" to correspond with the header
+ // specified in "columns".
+ static int VerifyHeader(const std::string& line,
+ const ExpectedValuesMap& columns) {
+ size_t line_position = 0;
+ for (ExpectedValuesMap::const_iterator it = columns.begin();
+ it != columns.end(); ++it) {
+ std::string str = ParseElement(line, &line_position,
+ it->second.multi_value_length);
+ EXPECT_EQ(str, it->first);
+ if (str != it->first)
+ return -1;
+ }
+ return 0;
+ }
+
+ // Parses out and returns one element from the string "line", which contains
+ // one line read from a log table file. An element can either be a column
+ // header or a cell of a row.
+ static std::string ParseElement(const std::string& line,
+ size_t* line_position,
+ int multi_value_length) {
+ std::string parsed_cell;
+ parsed_cell = "";
+ for (int i = 0; i < multi_value_length; ++i) {
+ size_t next_separator = line.find(',', *line_position);
+ EXPECT_NE(next_separator, std::string::npos);
+ if (next_separator == std::string::npos)
+ break;
+ parsed_cell += line.substr(*line_position,
+ next_separator - *line_position + 1);
+ *line_position = next_separator + 1;
+ }
+ return parsed_cell;
+ }
+
+ // This constant defines the maximum line length the DataLogParser can
+ // parse.
+ enum { kMaxLineLength = 100 };
+};
+
+TEST(TestDataLog, CreateReturnTest) {
+ for (int i = 0; i < 10; ++i)
+ ASSERT_EQ(DataLog::CreateLog(), 0);
+ ASSERT_EQ(DataLog::AddTable(DataLog::Combine("a proper table", 1)), 0);
+ for (int i = 0; i < 10; ++i)
+ DataLog::ReturnLog();
+ ASSERT_LT(DataLog::AddTable(DataLog::Combine("table failure", 1)), 0);
+}
+
+TEST(TestDataLog, VerifyCombineMethod) {
+ EXPECT_EQ(std::string("a proper table_1"),
+ DataLog::Combine("a proper table", 1));
+}
+
+TEST(TestDataLog, VerifySingleTable) {
+ DataLog::CreateLog();
+ DataLog::AddTable(DataLog::Combine("table", 1));
+ DataLog::AddColumn(DataLog::Combine("table", 1), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 1), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 1), "size", 5);
+ WebRtc_UWord32 sizes[5] = {1400, 1500, 1600, 1700, 1800};
+ for (int i = 0; i < 10; ++i) {
+ DataLog::InsertCell(DataLog::Combine("table", 1), "arrival",
+ static_cast<double>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 1), "timestamp",
+ static_cast<WebRtc_Word64>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 1), "size", sizes, 5);
+ DataLog::NextRow(DataLog::Combine("table", 1));
+ }
+ DataLog::ReturnLog();
+ // Verify file
+ FILE* table = fopen("table_1.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ // Read the column names and verify with the expected columns.
+ // Note that the columns are written to file in alphabetical order.
+ // Data expected from parsing the file
+ const int kNumberOfRows = 10;
+ std::string string_arrival[kNumberOfRows] = {
+ "0,", "1,", "2,", "3,", "4,",
+ "5,", "6,", "7,", "8,", "9,"
+ };
+ std::string string_timestamp[kNumberOfRows] = {
+ "4354,", "4355,", "4356,", "4357,",
+ "4358,", "4359,", "4360,", "4361,",
+ "4362,", "4363,"
+ };
+ std::string string_sizes = "1400,1500,1600,1700,1800,";
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["size[5],,,,,"] = ExpectedValues(
+ std::vector<std::string>(10, string_sizes), 5);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+}
+
+TEST(TestDataLog, VerifyMultipleTables) {
+ DataLog::CreateLog();
+ DataLog::AddTable(DataLog::Combine("table", 2));
+ DataLog::AddTable(DataLog::Combine("table", 3));
+ DataLog::AddColumn(DataLog::Combine("table", 2), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 2), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 2), "size", 1);
+ DataLog::AddTable(DataLog::Combine("table", 4));
+ DataLog::AddColumn(DataLog::Combine("table", 3), "timestamp", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 3), "arrival", 1);
+ DataLog::AddColumn(DataLog::Combine("table", 4), "size", 1);
+ for (WebRtc_Word32 i = 0; i < 10; ++i) {
+ DataLog::InsertCell(DataLog::Combine("table", 2), "arrival",
+ static_cast<WebRtc_Word32>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 2), "timestamp",
+ static_cast<WebRtc_Word32>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 2), "size",
+ static_cast<WebRtc_Word32>(1200 + 10 * i));
+ DataLog::InsertCell(DataLog::Combine("table", 3), "timestamp",
+ static_cast<WebRtc_Word32>(4354 + i));
+ DataLog::InsertCell(DataLog::Combine("table", 3), "arrival",
+ static_cast<WebRtc_Word32>(i));
+ DataLog::InsertCell(DataLog::Combine("table", 4), "size",
+ static_cast<WebRtc_Word32>(1200 + 10 * i));
+ DataLog::NextRow(DataLog::Combine("table", 4));
+ DataLog::NextRow(DataLog::Combine("table", 2));
+ DataLog::NextRow(DataLog::Combine("table", 3));
+ }
+ DataLog::ReturnLog();
+
+ // Data expected from parsing the file
+ const int kNumberOfRows = 10;
+ std::string string_arrival[kNumberOfRows] = {
+ "0,", "1,", "2,", "3,", "4,",
+ "5,", "6,", "7,", "8,", "9,"
+ };
+ std::string string_timestamp[kNumberOfRows] = {
+ "4354,", "4355,", "4356,", "4357,",
+ "4358,", "4359,", "4360,", "4361,",
+ "4362,", "4363,"
+ };
+ std::string string_size[kNumberOfRows] = {
+ "1200,", "1210,", "1220,", "1230,",
+ "1240,", "1250,", "1260,", "1270,",
+ "1280,", "1290,"
+ };
+
+ // Verify table 2
+ {
+ FILE* table = fopen("table_2.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["size,"] = ExpectedValues(
+ std::vector<std::string>(string_size,
+ string_size + kNumberOfRows),
+ 1);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+
+ // Verify table 3
+ {
+ FILE* table = fopen("table_3.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["arrival,"] = ExpectedValues(
+ std::vector<std::string>(string_arrival,
+ string_arrival +
+ kNumberOfRows),
+ 1);
+ expected["timestamp,"] = ExpectedValues(
+ std::vector<std::string>(string_timestamp,
+ string_timestamp +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+
+ // Verify table 4
+ {
+ FILE* table = fopen("table_4.txt", "r");
+ ASSERT_FALSE(table == NULL);
+ ExpectedValuesMap expected;
+ expected["size,"] = ExpectedValues(
+ std::vector<std::string>(string_size,
+ string_size +
+ kNumberOfRows),
+ 1);
+ ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
+ fclose(table);
+ }
+}
+
+TEST(TestDataLogCWrapper, VerifyCWrapper) {
+ // Simply call all C wrapper log functions through the C helper unittests.
+ // Main purpose is to make sure that the linkage is correct.
+
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestCreateLog());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestCombine());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddTable());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddColumn());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_float());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_float());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_double());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_double());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_uint32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_uint32());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int64());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int64());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
+ EXPECT_EQ(0, WebRtcDataLogCHelper_TestReturnLog());
+}
diff --git a/src/system_wrappers/source/data_log_unittest_disabled.cc b/src/system_wrappers/source/data_log_unittest_disabled.cc
new file mode 100644
index 0000000..9d630b6
--- /dev/null
+++ b/src/system_wrappers/source/data_log_unittest_disabled.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "system_wrappers/interface/data_log.h"
+
+#include <cstdio>
+
+#include "gtest/gtest.h"
+
+using ::webrtc::DataLog;
+
+const char* kDataLogFileName = "table_1.txt";
+
+void PerformLogging(std::string table_name) {
+ // Simulate normal DataTable logging behavior using this table name.
+ ASSERT_EQ(0, DataLog::AddTable(table_name));
+ ASSERT_EQ(0, DataLog::AddColumn(table_name, "test", 1));
+ for (int i = 0; i < 10; ++i) {
+ // TODO(kjellander): Check InsertCell result when the DataLog dummy is
+ // fixed.
+ DataLog::InsertCell(table_name, "test", static_cast<double>(i));
+ ASSERT_EQ(0, DataLog::NextRow(table_name));
+ }
+}
+
+// Simple test to verify DataLog is still working when the GYP variable
+// enable_data_logging==0 (the default case).
+TEST(TestDataLogDisabled, VerifyLoggingWorks) {
+ ASSERT_EQ(0, DataLog::CreateLog());
+ // Generate a table_name name and assure it's an empty string
+ // (dummy behavior).
+ std::string table_name = DataLog::Combine("table", 1);
+ ASSERT_EQ("", table_name);
+ PerformLogging(table_name);
+ DataLog::ReturnLog();
+}
+
+TEST(TestDataLogDisabled, EnsureNoFileIsWritten) {
+ // Remove any previous data files on disk:
+ std::remove(kDataLogFileName);
+ ASSERT_EQ(0, DataLog::CreateLog());
+ // Don't use the table name we would get from Combine on a disabled DataLog.
+ // Use "table_1" instead (which is what an enabled DataLog would give us).
+ PerformLogging("table_1");
+ DataLog::ReturnLog();
+ // Verify no data log file have been written:
+ ASSERT_EQ(NULL, fopen(kDataLogFileName, "r"));
+}
diff --git a/src/system_wrappers/source/event.cc b/src/system_wrappers/source/event.cc
index 384b961..608cd53 100644
--- a/src/system_wrappers/source/event.cc
+++ b/src/system_wrappers/source/event.cc
@@ -12,10 +12,14 @@
#if defined(_WIN32)
#include <windows.h>
- #include "event_windows.h"
+ #include "event_win.h"
+#elif defined(WEBRTC_MAC_INTEL)
+ #include <ApplicationServices/ApplicationServices.h>
+ #include <pthread.h>
+ #include "event_posix.h"
#else
#include <pthread.h>
- #include "event_linux.h"
+ #include "event_posix.h"
#endif
namespace webrtc {
@@ -24,7 +28,7 @@
#if defined(_WIN32)
return new EventWindows();
#else
- return EventLinux::Create();
+ return EventPosix::Create();
#endif
}
@@ -45,6 +49,21 @@
{
return 0;
}
+#elif defined(WEBRTC_MAC_INTEL)
+ bool keyDown = false;
+ // loop through all Mac virtual key constant values
+ for(int keyIndex = 0; keyIndex <= 0x5C; keyIndex++)
+ {
+ keyDown |= CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, keyIndex);
+ }
+ if(keyDown)
+ {
+ return 1;
+ }
+ else
+ {
+ return 0;
+ }
#else
return -1;
#endif
diff --git a/src/system_wrappers/source/event_linux.cc b/src/system_wrappers/source/event_posix.cc
similarity index 90%
rename from src/system_wrappers/source/event_linux.cc
rename to src/system_wrappers/source/event_posix.cc
index dddd31c..b77b902 100644
--- a/src/system_wrappers/source/event_linux.cc
+++ b/src/system_wrappers/source/event_posix.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "event_linux.h"
+#include "event_posix.h"
#include <errno.h>
#include <pthread.h>
@@ -22,9 +22,9 @@
const long int E6 = 1000000;
const long int E9 = 1000 * E6;
-EventWrapper* EventLinux::Create()
+EventWrapper* EventPosix::Create()
{
- EventLinux* ptr = new EventLinux;
+ EventPosix* ptr = new EventPosix;
if (!ptr)
{
return NULL;
@@ -40,7 +40,7 @@
}
-EventLinux::EventLinux()
+EventPosix::EventPosix()
: _timerThread(0),
_timerEvent(0),
_periodic(false),
@@ -50,7 +50,7 @@
{
}
-int EventLinux::Construct()
+int EventPosix::Construct()
{
// Set start time to zero
memset(&_tCreate, 0, sizeof(_tCreate));
@@ -92,14 +92,14 @@
return 0;
}
-EventLinux::~EventLinux()
+EventPosix::~EventPosix()
{
StopTimer();
pthread_cond_destroy(&cond);
pthread_mutex_destroy(&mutex);
}
-bool EventLinux::Reset()
+bool EventPosix::Reset()
{
if (0 != pthread_mutex_lock(&mutex))
{
@@ -110,7 +110,7 @@
return true;
}
-bool EventLinux::Set()
+bool EventPosix::Set()
{
if (0 != pthread_mutex_lock(&mutex))
{
@@ -123,7 +123,7 @@
return true;
}
-EventTypeWrapper EventLinux::Wait(unsigned long timeout)
+EventTypeWrapper EventPosix::Wait(unsigned long timeout)
{
int retVal = 0;
if (0 != pthread_mutex_lock(&mutex))
@@ -178,7 +178,7 @@
}
}
-EventTypeWrapper EventLinux::Wait(timespec& tPulse)
+EventTypeWrapper EventPosix::Wait(timespec& tPulse)
{
int retVal = 0;
if (0 != pthread_mutex_lock(&mutex))
@@ -205,7 +205,7 @@
}
}
-bool EventLinux::StartTimer(bool periodic, unsigned long time)
+bool EventPosix::StartTimer(bool periodic, unsigned long time)
{
if (_timerThread)
{
@@ -223,7 +223,7 @@
}
// Start the timer thread
- _timerEvent = static_cast<EventLinux*>(EventWrapper::Create());
+ _timerEvent = static_cast<EventPosix*>(EventWrapper::Create());
const char* threadName = "WebRtc_event_timer_thread";
_timerThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
threadName);
@@ -237,12 +237,12 @@
return false;
}
-bool EventLinux::Run(ThreadObj obj)
+bool EventPosix::Run(ThreadObj obj)
{
- return static_cast<EventLinux*>(obj)->Process();
+ return static_cast<EventPosix*>(obj)->Process();
}
-bool EventLinux::Process()
+bool EventPosix::Process()
{
if (_tCreate.tv_sec == 0)
{
@@ -290,7 +290,7 @@
return true;
}
-bool EventLinux::StopTimer()
+bool EventPosix::StopTimer()
{
if(_timerThread)
{
diff --git a/src/system_wrappers/source/event_linux.h b/src/system_wrappers/source/event_posix.h
similarity index 81%
rename from src/system_wrappers/source/event_linux.h
rename to src/system_wrappers/source/event_posix.h
index 17d193f..0e5893b 100644
--- a/src/system_wrappers/source/event_linux.h
+++ b/src/system_wrappers/source/event_posix.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
#include "event_wrapper.h"
@@ -25,12 +25,12 @@
kDown = 2
};
-class EventLinux : public EventWrapper
+class EventPosix : public EventWrapper
{
public:
static EventWrapper* Create();
- virtual ~EventLinux();
+ virtual ~EventPosix();
virtual EventTypeWrapper Wait(unsigned long maxTime);
virtual bool Set();
@@ -40,7 +40,7 @@
virtual bool StopTimer();
private:
- EventLinux();
+ EventPosix();
int Construct();
static bool Run(ThreadObj obj);
@@ -53,7 +53,7 @@
pthread_mutex_t mutex;
ThreadWrapper* _timerThread;
- EventLinux* _timerEvent;
+ EventPosix* _timerEvent;
timespec _tCreate;
bool _periodic;
@@ -63,4 +63,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
diff --git a/src/system_wrappers/source/file_impl.cc b/src/system_wrappers/source/file_impl.cc
index 6046c2c..d163bf6 100644
--- a/src/system_wrappers/source/file_impl.cc
+++ b/src/system_wrappers/source/file_impl.cc
@@ -10,16 +10,17 @@
#include "file_impl.h"
-#include <cassert>
+#include <assert.h>
#ifdef _WIN32
- #include <Windows.h>
+#include <Windows.h>
#else
- #include <stdarg.h>
- #include <string.h>
+#include <stdarg.h>
+#include <string.h>
#endif
namespace webrtc {
+
FileWrapper* FileWrapper::Create()
{
return new FileWrapperImpl();
@@ -30,8 +31,7 @@
_open(false),
_looping(false),
_readOnly(false),
- _text(false),
- _maxSizeInBytes(-1),
+ _maxSizeInBytes(0),
_sizeInBytes(0)
{
memset(_fileNameUTF8, 0, kMaxFileNameSize);
@@ -45,7 +45,7 @@
}
}
-WebRtc_Word32 FileWrapperImpl::CloseFile()
+int FileWrapperImpl::CloseFile()
{
if (_id != NULL)
{
@@ -70,13 +70,13 @@
return -1;
}
-WebRtc_Word32 FileWrapperImpl::SetMaxFileSize(WebRtc_Word32 bytes)
+int FileWrapperImpl::SetMaxFileSize(size_t bytes)
{
_maxSizeInBytes = bytes;
return 0;
}
-WebRtc_Word32 FileWrapperImpl::Flush()
+int FileWrapperImpl::Flush()
{
if (_id != NULL)
{
@@ -85,40 +85,39 @@
return -1;
}
-WebRtc_Word32 FileWrapperImpl::FileName(WebRtc_Word8* fileNameUTF8,
- WebRtc_UWord32 size) const
+int FileWrapperImpl::FileName(char* fileNameUTF8,
+ size_t size) const
{
- WebRtc_Word32 len = static_cast<WebRtc_Word32>(strlen(_fileNameUTF8));
- if(len > kMaxFileNameSize)
+ size_t length = strlen(_fileNameUTF8);
+ if(length > kMaxFileNameSize)
{
assert(false);
return -1;
}
- if(len < 1)
+ if(length < 1)
{
return -1;
}
+
// Make sure to NULL terminate
- if(size < (WebRtc_UWord32)len)
+ if(size < length)
{
- len = size - 1;
+ length = size - 1;
}
- memcpy(fileNameUTF8, _fileNameUTF8, len);
- fileNameUTF8[len] = 0;
+ memcpy(fileNameUTF8, _fileNameUTF8, length);
+ fileNameUTF8[length] = 0;
return 0;
}
-bool
-FileWrapperImpl::Open() const
+bool FileWrapperImpl::Open() const
{
return _open;
}
-WebRtc_Word32 FileWrapperImpl::OpenFile(const WebRtc_Word8 *fileNameUTF8,
- const bool readOnly, const bool loop,
- const bool text)
+int FileWrapperImpl::OpenFile(const char *fileNameUTF8, bool readOnly,
+ bool loop, bool text)
{
- WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
+ size_t length = strlen(fileNameUTF8);
if (length > kMaxFileNameSize)
{
return -1;
@@ -174,7 +173,7 @@
if (tmpId != NULL)
{
- // + 1 comes fro copying the NULL termination charachter too
+ // +1 comes from copying the NULL termination character.
memcpy(_fileNameUTF8, fileNameUTF8, length + 1);
if (_id != NULL)
{
@@ -188,80 +187,79 @@
return -1;
}
-int FileWrapperImpl::Read(void *buf, int len)
+int FileWrapperImpl::Read(void* buf, int length)
{
- if(len < 0)
- {
- return 0;
- }
- if (_id != NULL)
- {
- WebRtc_Word32 res = static_cast<WebRtc_Word32>(fread(buf, 1, len, _id));
- if (res != len)
- {
- if(!_looping)
- {
- CloseFile();
- }
- }
- return res;
- }
- return -1;
-}
-
-WebRtc_Word32 FileWrapperImpl::WriteText(const WebRtc_Word8* text, ...)
-{
- assert(!_readOnly);
- assert(!_text);
+ if (length < 0)
+ return -1;
if (_id == NULL)
+ return -1;
+
+ int bytes_read = static_cast<int>(fread(buf, 1, length, _id));
+ if (bytes_read != length && !_looping)
{
+ CloseFile();
+ }
+ return bytes_read;
+}
+
+int FileWrapperImpl::WriteText(const char* format, ...)
+{
+ if (format == NULL)
+ return -1;
+
+ if (_readOnly)
+ return -1;
+
+ if (_id == NULL)
+ return -1;
+
+ va_list args;
+ va_start(args, format);
+ int num_chars = vfprintf(_id, format, args);
+ va_end(args);
+
+ if (num_chars >= 0)
+ {
+ return num_chars;
+ }
+ else
+ {
+ CloseFile();
return -1;
}
-
- char tempBuff[kFileMaxTextMessageSize];
- if (text)
- {
- va_list args;
- va_start(args, text);
-#ifdef _WIN32
- _vsnprintf(tempBuff, kFileMaxTextMessageSize-1, text, args);
-#else
- vsnprintf(tempBuff, kFileMaxTextMessageSize-1, text, args);
-#endif
- va_end(args);
- WebRtc_Word32 nBytes;
- nBytes = fprintf(_id, "%s", tempBuff);
- if (nBytes > 0)
- {
- return 0;
- }
- CloseFile();
- }
- return -1;
}
-bool FileWrapperImpl::Write(const void* buf, int len)
+bool FileWrapperImpl::Write(const void* buf, int length)
{
- assert(!_readOnly);
- if (_id != NULL)
- {
- // Check if it's time to stop writing.
- if ((_maxSizeInBytes != -1) &&
- _sizeInBytes + len > (WebRtc_UWord32)_maxSizeInBytes)
- {
- Flush();
- return false;
- }
+ if (buf == NULL)
+ return false;
- size_t nBytes = fwrite((WebRtc_UWord8*)buf, 1, len, _id);
- if (nBytes > 0)
- {
- _sizeInBytes += static_cast<WebRtc_Word32>(nBytes);
- return true;
- }
- CloseFile();
+ if (length < 0)
+ return false;
+
+ if (_readOnly)
+ return false;
+
+ if (_id == NULL)
+ return false;
+
+ // Check if it's time to stop writing.
+ if (_maxSizeInBytes > 0 && (_sizeInBytes + length) > _maxSizeInBytes)
+ {
+ Flush();
+ return false;
}
+
+ size_t num_bytes = fwrite(buf, 1, length, _id);
+ if (num_bytes > 0)
+ {
+ _sizeInBytes += num_bytes;
+ return true;
+ }
+
+ CloseFile();
return false;
}
+
} // namespace webrtc
diff --git a/src/system_wrappers/source/file_impl.h b/src/system_wrappers/source/file_impl.h
index cf6b734..31ab31e 100644
--- a/src/system_wrappers/source/file_impl.h
+++ b/src/system_wrappers/source/file_impl.h
@@ -16,42 +16,42 @@
#include <stdio.h>
namespace webrtc {
+
class FileWrapperImpl : public FileWrapper
{
public:
FileWrapperImpl();
virtual ~FileWrapperImpl();
- virtual WebRtc_Word32 FileName(WebRtc_Word8* fileNameUTF8,
- WebRtc_UWord32 size) const;
+ virtual int FileName(char* fileNameUTF8,
+ size_t size) const;
virtual bool Open() const;
- virtual WebRtc_Word32 OpenFile(const WebRtc_Word8* fileNameUTF8,
- const bool readOnly,
- const bool loop = false,
- const bool text = false);
+ virtual int OpenFile(const char* fileNameUTF8,
+ bool readOnly,
+ bool loop = false,
+ bool text = false);
- virtual WebRtc_Word32 CloseFile();
- virtual WebRtc_Word32 SetMaxFileSize(WebRtc_Word32 bytes);
- virtual WebRtc_Word32 Flush();
+ virtual int CloseFile();
+ virtual int SetMaxFileSize(size_t bytes);
+ virtual int Flush();
- virtual int Read(void* buf, int len);
- virtual bool Write(const void *buf, int len);
+ virtual int Read(void* buf, int length);
+ virtual bool Write(const void *buf, int length);
+ virtual int WriteText(const char* format, ...);
virtual int Rewind();
- virtual WebRtc_Word32 WriteText(const WebRtc_Word8* text, ...);
-
private:
- FILE* _id;
- bool _open;
- bool _looping;
- bool _readOnly;
- bool _text;
- WebRtc_Word32 _maxSizeInBytes; // -1 indicates file size limitation is off
- WebRtc_UWord32 _sizeInBytes;
- WebRtc_Word8 _fileNameUTF8[kMaxFileNameSize];
+ FILE* _id;
+ bool _open;
+ bool _looping;
+ bool _readOnly;
+ size_t _maxSizeInBytes; // -1 indicates file size limitation is off
+ size_t _sizeInBytes;
+ char _fileNameUTF8[kMaxFileNameSize];
};
+
} // namespace webrtc
#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
diff --git a/src/system_wrappers/source/list_no_stl.cc b/src/system_wrappers/source/list_no_stl.cc
index d45f27b..dbba571 100644
--- a/src/system_wrappers/source/list_no_stl.cc
+++ b/src/system_wrappers/source/list_no_stl.cc
@@ -79,7 +79,7 @@
int ListWrapper::PushBack(const void* ptr)
{
ListItem* item = new ListItem(ptr);
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
PushBackImpl(item);
return 0;
}
@@ -87,7 +87,7 @@
int ListWrapper::PushBack(const unsigned int item_id)
{
ListItem* item = new ListItem(item_id);
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
PushBackImpl(item);
return 0;
}
@@ -95,7 +95,7 @@
int ListWrapper::PushFront(const unsigned int item_id)
{
ListItem* item = new ListItem(item_id);
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
PushFrontImpl(item);
return 0;
}
@@ -103,7 +103,7 @@
int ListWrapper::PushFront(const void* ptr)
{
ListItem* item = new ListItem(ptr);
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
PushFrontImpl(item);
return 0;
}
@@ -159,7 +159,7 @@
{
return -1;
}
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
if (!existing_previous_item)
{
PushBackImpl(new_item);
@@ -195,7 +195,7 @@
{
return -1;
}
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
if (!existing_next_item)
{
PushBackImpl(new_item);
diff --git a/src/system_wrappers/source/list_unittest.cc b/src/system_wrappers/source/list_unittest.cc
index 3f3c88f..4d32f59 100644
--- a/src/system_wrappers/source/list_unittest.cc
+++ b/src/system_wrappers/source/list_unittest.cc
@@ -10,10 +10,12 @@
#include "gtest/gtest.h"
-#include "list_wrapper.h"
+#include "system_wrappers/interface/list_wrapper.h"
+#include "system_wrappers/interface/scoped_ptr.h"
using ::webrtc::ListWrapper;
using ::webrtc::ListItem;
+using ::webrtc::scoped_ptr;
// Note: kNumberOfElements needs to be even.
const unsigned int kNumberOfElements = 10;
@@ -38,8 +40,6 @@
virtual unsigned int GetUnsignedItem(
const ListItem* item) const = 0;
virtual ListItem* CreateListItem(unsigned int item_id) = 0;
- virtual bool DestroyListItem(ListItem* item) = 0;
-
unsigned int GetSize() const {
return list_.GetSize();
}
@@ -64,24 +64,52 @@
}
virtual int Erase(ListItem* item) = 0;
int Insert(ListItem* existing_previous_item,
- ListItem* new_item) {
- return list_.Insert(existing_previous_item, new_item);
+ ListItem* new_item) {
+ const int retval = list_.Insert(existing_previous_item, new_item);
+ if (retval != 0) {
+ EXPECT_TRUE(DestroyListItem(new_item));
+ }
+ return retval;
}
int InsertBefore(ListItem* existing_next_item,
- ListItem* new_item) {
- return list_.InsertBefore(existing_next_item, new_item);
+ ListItem* new_item) {
+ const int retval = list_.InsertBefore(existing_next_item, new_item);
+ if (retval != 0) {
+ EXPECT_TRUE(DestroyListItem(new_item));
+ }
+ return retval;
}
protected:
ListWrapperSimple() {}
+ virtual bool DestroyListItemContent(ListItem* item) = 0;
+ bool DestroyListItem(ListItem* item) {
+ const bool retval = DestroyListItemContent(item);
+ delete item;
+ return retval;
+ }
+
ListWrapper list_;
};
+void ClearList(ListWrapperSimple* list_wrapper) {
+ if (list_wrapper == NULL) {
+ return;
+ }
+ ListItem* list_item = list_wrapper->First();
+ while (list_item != NULL) {
+ EXPECT_EQ(list_wrapper->Erase(list_item), 0);
+ list_item = list_wrapper->First();
+ }
+}
+
class ListWrapperStatic : public ListWrapperSimple {
public:
ListWrapperStatic() {}
- virtual ~ListWrapperStatic() {}
+ virtual ~ListWrapperStatic() {
+ ClearList(this);
+ }
virtual unsigned int GetUnsignedItem(const ListItem* item) const {
return item->GetUnsignedItem();
@@ -89,11 +117,7 @@
virtual ListItem* CreateListItem(unsigned int item_id) {
return new ListItem(item_id);
}
- virtual bool DestroyListItem(ListItem* item) {
- if (item == NULL) {
- return false;
- }
- delete item;
+ virtual bool DestroyListItemContent(ListItem* item) {
return true;
}
virtual int PushBack(const unsigned int item_id) {
@@ -116,7 +140,9 @@
class ListWrapperDynamic : public ListWrapperSimple {
public:
ListWrapperDynamic() {}
- virtual ~ListWrapperDynamic() {}
+ virtual ~ListWrapperDynamic() {
+ ClearList(this);
+ }
virtual unsigned int GetUnsignedItem(const ListItem* item) const {
const unsigned int* return_value_pointer =
@@ -140,7 +166,7 @@
}
return return_value;
}
- virtual bool DestroyListItem(ListItem* item) {
+ virtual bool DestroyListItemContent(ListItem* item) {
if (item == NULL) {
return false;
}
@@ -151,7 +177,6 @@
return_value = true;
delete item_id_ptr;
}
- delete item;
return return_value;
}
virtual int PushBack(const unsigned int item_id) {
@@ -190,17 +215,15 @@
if (item == NULL) {
return -1;
}
- unsigned int* item_id_ptr = reinterpret_cast<unsigned int*> (
- item->GetItem());
- int return_value = -1;
- if (item_id_ptr != NULL) {
- delete item_id_ptr;
- return_value = 0;
+ int retval = 0;
+ if (!DestroyListItemContent(item)) {
+ retval = -1;
+ ADD_FAILURE();
}
if (list_.Erase(item) != 0) {
- return -1;
+ retval = -1;
}
- return return_value;
+ return retval;
}
};
@@ -212,15 +235,6 @@
return new ListWrapperDynamic();
}
-void ClearList(ListWrapperSimple* list) {
- if (list == NULL)
- {
- return;
- }
- while (list->Erase(list->First()) == 0) {
- }
-}
-
ListWrapperSimple* CreateAscendingList(bool static_allocation) {
ListWrapperSimple* return_value = ListWrapperSimple::Create(
static_allocation);
@@ -237,7 +251,7 @@
return return_value;
}
-ListWrapperSimple* CreateDecendingList(bool static_allocation) {
+ListWrapperSimple* CreateDescendingList(bool static_allocation) {
ListWrapperSimple* return_value = ListWrapperSimple::Create(
static_allocation);
if (return_value == NULL) {
@@ -323,17 +337,20 @@
TEST(ListWrapperTest,ReverseNewIntList) {
// Create a new temporary list with elements reversed those of
// new_int_list_
- const ListWrapperSimple* decending_list = CreateDecendingList(rand()%2);
- ASSERT_FALSE(decending_list == NULL);
- ASSERT_FALSE(decending_list->Empty());
- ASSERT_EQ(kNumberOfElements,decending_list->GetSize());
+ const scoped_ptr<ListWrapperSimple> descending_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(descending_list.get() == NULL);
+ ASSERT_FALSE(descending_list->Empty());
+ ASSERT_EQ(kNumberOfElements,descending_list->GetSize());
- const ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
- ASSERT_FALSE(ascending_list == NULL);
+ const scoped_ptr<ListWrapperSimple> ascending_list(
+ CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
ASSERT_FALSE(ascending_list->Empty());
ASSERT_EQ(kNumberOfElements,ascending_list->GetSize());
- ListWrapperSimple* list_to_reverse = ListWrapperSimple::Create(rand()%2);
+ scoped_ptr<ListWrapperSimple> list_to_reverse(
+ ListWrapperSimple::Create(rand()%2));
// Reverse the list using PushBack and Previous.
for (ListItem* item = ascending_list->Last(); item != NULL;
@@ -341,98 +358,97 @@
list_to_reverse->PushBack(ascending_list->GetUnsignedItem(item));
}
- ASSERT_TRUE(CompareLists(decending_list,list_to_reverse));
+ ASSERT_TRUE(CompareLists(descending_list.get(), list_to_reverse.get()));
- ListWrapperSimple* list_to_un_reverse =
- ListWrapperSimple::Create(rand()%2);
- ASSERT_FALSE(list_to_un_reverse == NULL);
+ scoped_ptr<ListWrapperSimple> list_to_un_reverse(
+ ListWrapperSimple::Create(rand()%2));
+ ASSERT_FALSE(list_to_un_reverse.get() == NULL);
// Reverse the reversed list using PushFront and Next.
for (ListItem* item = list_to_reverse->First(); item != NULL;
item = list_to_reverse->Next(item)) {
list_to_un_reverse->PushFront(list_to_reverse->GetUnsignedItem(item));
}
-
- ASSERT_TRUE(CompareLists(ascending_list,list_to_un_reverse));
+ ASSERT_TRUE(CompareLists(ascending_list.get(), list_to_un_reverse.get()));
}
TEST(ListWrapperTest,PopTest) {
- ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
- ASSERT_FALSE(ascending_list == NULL);
+ scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
ASSERT_FALSE(ascending_list->Empty());
- EXPECT_EQ(0,ascending_list->PopFront());
- EXPECT_EQ(1,ascending_list->GetUnsignedItem(ascending_list->First()));
-
- EXPECT_EQ(0,ascending_list->PopBack());
- EXPECT_EQ(kNumberOfElements - 2,ascending_list->GetUnsignedItem(
+ EXPECT_EQ(0, ascending_list->PopFront());
+ EXPECT_EQ(1U, ascending_list->GetUnsignedItem(ascending_list->First()));
+
+ EXPECT_EQ(0, ascending_list->PopBack());
+ EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetUnsignedItem(
ascending_list->Last()));
EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetSize());
}
// Use Insert to interleave two lists.
TEST(ListWrapperTest,InterLeaveTest) {
- ListWrapperSimple* interleave_list = CreateAscendingList(rand()%2);
- ASSERT_FALSE(interleave_list == NULL);
+ scoped_ptr<ListWrapperSimple> interleave_list(
+ CreateAscendingList(rand()%2));
+ ASSERT_FALSE(interleave_list.get() == NULL);
ASSERT_FALSE(interleave_list->Empty());
- ListWrapperSimple* decending_list = CreateDecendingList(rand()%2);
- ASSERT_FALSE(decending_list == NULL);
+ scoped_ptr<ListWrapperSimple> descending_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(descending_list.get() == NULL);
- for (int i = 0; i < kNumberOfElements/2; ++i) {
- ASSERT_EQ(0,interleave_list->PopBack());
- ASSERT_EQ(0,decending_list->PopBack());
+ for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
+ ASSERT_EQ(0, interleave_list->PopBack());
+ ASSERT_EQ(0, descending_list->PopBack());
}
- ASSERT_EQ(kNumberOfElements/2,interleave_list->GetSize());
- ASSERT_EQ(kNumberOfElements/2,decending_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, descending_list->GetSize());
- int insert_position = kNumberOfElements/2;
+ unsigned int insert_position = kNumberOfElements/2;
ASSERT_EQ(insert_position * 2, kNumberOfElements);
- while (!decending_list->Empty())
+ while (!descending_list->Empty())
{
- ListItem* item = decending_list->Last();
+ ListItem* item = descending_list->Last();
ASSERT_FALSE(item == NULL);
- const unsigned int item_id = decending_list->GetUnsignedItem(item);
- ASSERT_EQ(0,decending_list->Erase(item));
+ const unsigned int item_id = descending_list->GetUnsignedItem(item);
+ ASSERT_EQ(0, descending_list->Erase(item));
ListItem* insert_item = interleave_list->CreateListItem(item_id);
ASSERT_FALSE(insert_item == NULL);
item = interleave_list->First();
ASSERT_FALSE(item == NULL);
- for (int j = 0; j < insert_position - 1; ++j) {
+ for (unsigned int j = 0; j < insert_position - 1; ++j) {
item = interleave_list->Next(item);
ASSERT_FALSE(item == NULL);
}
- if (0 != interleave_list->Insert(item,insert_item)) {
- interleave_list->DestroyListItem(insert_item);
- FAIL();
- }
+ EXPECT_EQ(0, interleave_list->Insert(item, insert_item));
--insert_position;
}
-
- ListWrapperSimple* interleaved_list = CreateInterleavedList(rand()%2);
- ASSERT_FALSE(interleaved_list == NULL);
- ASSERT_FALSE(interleaved_list->Empty());
- ASSERT_TRUE(CompareLists(interleaved_list,interleave_list));
+ scoped_ptr<ListWrapperSimple> interleaved_list(
+ CreateInterleavedList(rand()%2));
+ ASSERT_FALSE(interleaved_list.get() == NULL);
+ ASSERT_FALSE(interleaved_list->Empty());
+ ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
}
// Use InsertBefore to interleave two lists.
TEST(ListWrapperTest,InterLeaveTestII) {
- ListWrapperSimple* interleave_list = CreateDecendingList(rand()%2);
- ASSERT_FALSE(interleave_list == NULL);
+ scoped_ptr<ListWrapperSimple> interleave_list(
+ CreateDescendingList(rand()%2));
+ ASSERT_FALSE(interleave_list.get() == NULL);
ASSERT_FALSE(interleave_list->Empty());
- ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
- ASSERT_FALSE(ascending_list == NULL);
+ scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
+ ASSERT_FALSE(ascending_list.get() == NULL);
- for (int i = 0; i < kNumberOfElements/2; ++i) {
- ASSERT_EQ(0,interleave_list->PopBack());
- ASSERT_EQ(0,ascending_list->PopBack());
+ for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
+ ASSERT_EQ(0, interleave_list->PopBack());
+ ASSERT_EQ(0, ascending_list->PopBack());
}
- ASSERT_EQ(kNumberOfElements/2,interleave_list->GetSize());
- ASSERT_EQ(kNumberOfElements/2,ascending_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
+ ASSERT_EQ(kNumberOfElements/2, ascending_list->GetSize());
- int insert_position = kNumberOfElements/2;
+ unsigned int insert_position = kNumberOfElements/2;
ASSERT_EQ(insert_position * 2, kNumberOfElements);
while (!ascending_list->Empty())
{
@@ -446,30 +462,18 @@
ASSERT_FALSE(insert_item == NULL);
item = interleave_list->First();
ASSERT_FALSE(item == NULL);
- for (int j = 0; j < insert_position - 1; ++j) {
+ for (unsigned int j = 0; j < insert_position - 1; ++j) {
item = interleave_list->Next(item);
ASSERT_FALSE(item == NULL);
}
- if (0 != interleave_list->InsertBefore(item,insert_item)) {
- interleave_list->DestroyListItem(insert_item);
- FAIL();
- }
+ EXPECT_EQ(interleave_list->InsertBefore(item, insert_item), 0);
--insert_position;
}
- ListWrapperSimple* interleaved_list = CreateInterleavedList(rand()%2);
- ASSERT_FALSE(interleaved_list == NULL);
+ scoped_ptr<ListWrapperSimple> interleaved_list(
+ CreateInterleavedList(rand()%2));
+ ASSERT_FALSE(interleaved_list.get() == NULL);
ASSERT_FALSE(interleaved_list->Empty());
- ASSERT_TRUE(CompareLists(interleaved_list,interleave_list));
-}
-
-int main(int argc, char **argv)
-{
- ::testing::InitGoogleTest(&argc, argv);
- // Added return_value so that it's convenient to put a breakpoint before
- // exiting please note that the return value from RUN_ALL_TESTS() must
- // be returned by the main function.
- const int return_value = RUN_ALL_TESTS();
- return return_value;
+ ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
}
diff --git a/src/system_wrappers/source/map.cc b/src/system_wrappers/source/map.cc
index 0bff155..331da32 100644
--- a/src/system_wrappers/source/map.cc
+++ b/src/system_wrappers/source/map.cc
@@ -13,7 +13,7 @@
#include "trace.h"
namespace webrtc {
-MapItem::MapItem(int id, void* item) : item_pointer_(item), item_id_(id)
+MapItem::MapItem(int id, void* item) : item_id_(id), item_pointer_(item)
{
}
diff --git a/src/system_wrappers/source/map_no_stl.cc b/src/system_wrappers/source/map_no_stl.cc
index cb0ac00..ef93a1f 100644
--- a/src/system_wrappers/source/map_no_stl.cc
+++ b/src/system_wrappers/source/map_no_stl.cc
@@ -75,7 +75,7 @@
{
MapNoStlItem* new_item = new MapNoStlItem(id, ptr);
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
MapNoStlItem* item = first_;
size_++;
if (!item)
@@ -144,7 +144,7 @@
MapNoStlItem* MapNoStl::Find(int id) const
{
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
MapNoStlItem* item = Locate(id);
return item;
}
@@ -155,13 +155,13 @@
{
return -1;
}
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
return Remove(item);
}
int MapNoStl::Erase(const int id)
{
- CriticalSectionScoped lock(*critical_section_);
+ CriticalSectionScoped lock(critical_section_);
MapNoStlItem* item = Locate(id);
if(!item)
{
diff --git a/src/system_wrappers/source/map_unittest.cc b/src/system_wrappers/source/map_unittest.cc
index 8e8ea07..1c85a92 100644
--- a/src/system_wrappers/source/map_unittest.cc
+++ b/src/system_wrappers/source/map_unittest.cc
@@ -15,7 +15,7 @@
using ::webrtc::MapWrapper;
using ::webrtc::MapItem;
-const unsigned int kNumberOfElements = 10;
+const int kNumberOfElements = 10;
int* ItemPointer(MapItem* item) {
if (item == NULL) {
@@ -48,7 +48,7 @@
if (supress) {
return;
}
- printf(message);
+ printf("%s", message);
}
bool CreateAscendingMap(MapWrapper* ascending_map) {
diff --git a/src/system_wrappers/source/rw_lock.cc b/src/system_wrappers/source/rw_lock.cc
index 47901d3..b308358 100644
--- a/src/system_wrappers/source/rw_lock.cc
+++ b/src/system_wrappers/source/rw_lock.cc
@@ -13,12 +13,12 @@
#include <assert.h>
#if defined(_WIN32)
- #include "rw_lock_windows.h"
-#elif defined(ANDROID)
+ #include "rw_lock_win.h"
+#elif defined(WEBRTC_ANDROID)
#include <stdlib.h>
#include "rw_lock_generic.h"
#else
- #include "rw_lock_linux.h"
+ #include "rw_lock_posix.h"
#endif
namespace webrtc {
@@ -26,10 +26,10 @@
{
#ifdef _WIN32
RWLockWrapper* lock = new RWLockWindows();
-#elif defined(ANDROID)
+#elif defined(WEBRTC_ANDROID)
RWLockWrapper* lock = new RWLockWrapperGeneric();
#else
- RWLockWrapper* lock = new RWLockLinux();
+ RWLockWrapper* lock = new RWLockPosix();
#endif
if(lock->Init() != 0)
{
diff --git a/src/system_wrappers/source/rw_lock_linux.cc b/src/system_wrappers/source/rw_lock_posix.cc
similarity index 72%
rename from src/system_wrappers/source/rw_lock_linux.cc
rename to src/system_wrappers/source/rw_lock_posix.cc
index 084dce8..81a161e 100644
--- a/src/system_wrappers/source/rw_lock_linux.cc
+++ b/src/system_wrappers/source/rw_lock_posix.cc
@@ -8,39 +8,39 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "rw_lock_linux.h"
+#include "rw_lock_posix.h"
namespace webrtc {
-RWLockLinux::RWLockLinux() : _lock()
+RWLockPosix::RWLockPosix() : _lock()
{
}
-RWLockLinux::~RWLockLinux()
+RWLockPosix::~RWLockPosix()
{
pthread_rwlock_destroy(&_lock);
}
-int RWLockLinux::Init()
+int RWLockPosix::Init()
{
return pthread_rwlock_init(&_lock, 0);
}
-void RWLockLinux::AcquireLockExclusive()
+void RWLockPosix::AcquireLockExclusive()
{
pthread_rwlock_wrlock(&_lock);
}
-void RWLockLinux::ReleaseLockExclusive()
+void RWLockPosix::ReleaseLockExclusive()
{
pthread_rwlock_unlock(&_lock);
}
-void RWLockLinux::AcquireLockShared()
+void RWLockPosix::AcquireLockShared()
{
pthread_rwlock_rdlock(&_lock);
}
-void RWLockLinux::ReleaseLockShared()
+void RWLockPosix::ReleaseLockShared()
{
pthread_rwlock_unlock(&_lock);
}
diff --git a/src/system_wrappers/source/rw_lock_linux.h b/src/system_wrappers/source/rw_lock_posix.h
similarity index 74%
rename from src/system_wrappers/source/rw_lock_linux.h
rename to src/system_wrappers/source/rw_lock_posix.h
index 391ee8f..929bbb8 100644
--- a/src/system_wrappers/source/rw_lock_linux.h
+++ b/src/system_wrappers/source/rw_lock_posix.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
#include "rw_lock_wrapper.h"
#include <pthread.h>
namespace webrtc {
-class RWLockLinux : public RWLockWrapper
+class RWLockPosix : public RWLockWrapper
{
public:
- RWLockLinux();
- virtual ~RWLockLinux();
+ RWLockPosix();
+ virtual ~RWLockPosix();
virtual void AcquireLockExclusive();
virtual void ReleaseLockExclusive();
@@ -36,4 +36,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
diff --git a/src/system_wrappers/source/system_wrappers.gyp b/src/system_wrappers/source/system_wrappers.gyp
index 0448941..ce2438f 100644
--- a/src/system_wrappers/source/system_wrappers.gyp
+++ b/src/system_wrappers/source/system_wrappers.gyp
@@ -1,13 +1,13 @@
-# Copyright (c) 2009 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
-
-# TODO: Rename files to use *_linux.cpp etc. names, to automatically include relevant files. Remove conditions section.
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [
- '../../common_settings.gypi', # Common settings
- ],
+ 'includes': [ '../../build/common.gypi', ],
'targets': [
{
'target_name': 'system_wrappers',
@@ -25,15 +25,24 @@
'../interface/aligned_malloc.h',
'../interface/atomic32_wrapper.h',
'../interface/condition_variable_wrapper.h',
+ '../interface/cpu_info.h',
'../interface/cpu_wrapper.h',
'../interface/cpu_features_wrapper.h',
'../interface/critical_section_wrapper.h',
+ '../interface/data_log.h',
+ '../interface/data_log_c.h',
+ '../interface/data_log_impl.h',
'../interface/event_wrapper.h',
'../interface/file_wrapper.h',
+ '../interface/fix_interlocked_exchange_pointer_win.h',
'../interface/list_wrapper.h',
'../interface/map_wrapper.h',
+ '../interface/ref_count.h',
'../interface/rw_lock_wrapper.h',
+ '../interface/scoped_ptr.h',
+ '../interface/scoped_refptr.h',
'../interface/sort.h',
+ '../interface/static_instance.h',
'../interface/thread_wrapper.h',
'../interface/tick_util.h',
'../interface/trace.h',
@@ -41,102 +50,134 @@
'atomic32.cc',
'atomic32_linux.h',
'atomic32_mac.h',
- 'atomic32_windows.h',
+ 'atomic32_win.h',
'condition_variable.cc',
- 'condition_variable_linux.h',
- 'condition_variable_windows.h',
+ 'condition_variable_posix.cc',
+ 'condition_variable_posix.h',
+ 'condition_variable_win.cc',
+ 'condition_variable_win.h',
'cpu.cc',
+ 'cpu_no_op.cc',
+ 'cpu_info.cc',
+ 'cpu_linux.cc',
'cpu_linux.h',
+ 'cpu_mac.cc',
'cpu_mac.h',
- 'cpu_windows.h',
+ 'cpu_win.cc',
+ 'cpu_win.h',
'cpu_features.cc',
'critical_section.cc',
- 'critical_section_linux.h',
- 'critical_section_windows.h',
+ 'critical_section_posix.cc',
+ 'critical_section_posix.h',
+ 'critical_section_win.cc',
+ 'critical_section_win.h',
+ 'data_log.cc',
+ 'data_log_c.cc',
+ 'data_log_no_op.cc',
'event.cc',
- 'event_linux.h',
- 'event_windows.h',
+ 'event_posix.cc',
+ 'event_posix.h',
+ 'event_win.cc',
+ 'event_win.h',
'file_impl.cc',
'file_impl.h',
'list_no_stl.cc',
'map.cc',
'rw_lock.cc',
- 'rw_lock_linux.h',
- 'rw_lock_windows.h',
+ 'rw_lock_posix.cc',
+ 'rw_lock_posix.h',
+ 'rw_lock_win.cc',
+ 'rw_lock_win.h',
'sort.cc',
'thread.cc',
- 'thread_linux.h',
- 'thread_windows.h',
+ 'thread_posix.cc',
+ 'thread_posix.h',
+ 'thread_win.cc',
+ 'thread_win.h',
+ 'set_thread_name_win.h',
'trace_impl.cc',
'trace_impl.h',
- 'trace_linux.h',
- 'trace_windows.h',
+ 'trace_impl_no_op.cc',
+ 'trace_posix.cc',
+ 'trace_posix.h',
+ 'trace_win.cc',
+ 'trace_win.h',
],
'conditions': [
+ ['enable_data_logging==1', {
+ 'sources!': [ 'data_log_no_op.cc', ],
+ },{
+ 'sources!': [ 'data_log.cc', ],
+ },],
['OS=="linux"', {
- 'sources': [
- 'condition_variable_linux.cc',
- 'cpu_linux.cc',
- 'critical_section_linux.cc',
- 'event_linux.cc',
- 'thread_linux.cc',
- 'trace_linux.cc',
- 'rw_lock_linux.cc',
- ],
'link_settings': {
- 'libraries': [
- '-lrt',
- ],
+ 'libraries': [ '-lrt', ],
},
}],
['OS=="mac"', {
- 'sources': [
- 'condition_variable_linux.cc',
- 'cpu_mac.cc',
- 'critical_section_linux.cc',
- 'event_linux.cc',
- 'rw_lock_linux.cc',
- 'thread_linux.cc',
- 'trace_linux.cc',
- ],
- }],
- ['OS=="win"', {
- 'sources': [
- 'atomic32_windows.h',
- 'condition_variable_windows.cc',
- 'condition_variable_windows.h',
- 'cpu_windows.cc',
- 'cpu_windows.h',
- 'critical_section_windows.cc',
- 'critical_section_windows.h',
- 'event_windows.cc',
- 'event_windows.h',
- 'rw_lock_windows.cc',
- 'rw_lock_windows.h',
- 'thread_windows.cc',
- 'thread_windows.h',
- 'trace_windows.cc',
- 'trace_windows.h',
- ],
'link_settings': {
- 'libraries': [
- '-lwinmm.lib',
- ],
+ 'libraries': [ '$(SDKROOT)/System/Library/Frameworks/ApplicationServices.framework', ],
},
}],
- ] # conditions
- },
- {
- 'target_name': 'system_wrappersTest',
- 'type': 'executable',
- 'dependencies': [
- 'system_wrappers'
- ],
- 'sources': [
- '../test/Test.cpp',
- ],
+ ['OS=="win"', {
+ 'link_settings': {
+ 'libraries': [ '-lwinmm.lib', ],
+ },
+ }],
+ ['build_with_chromium==1', {
+ 'sources!': [
+ 'cpu.cc',
+ 'cpu_linux.h',
+ 'cpu_mac.h',
+ 'cpu_win.h',
+ 'trace_impl.cc',
+ 'trace_impl.h',
+ 'trace_posix.cc',
+ 'trace_posix.h',
+ 'trace_win.cc',
+ 'trace_win.h',
+ ],
+ }, {
+ 'sources!': [
+ 'cpu_no_op.cc',
+ 'trace_impl_no_op.cc',
+ ],
+ }]
+ ] # conditions
},
], # targets
+ 'conditions': [
+ ['build_with_chromium==0', {
+ 'targets': [
+ {
+ 'target_name': 'system_wrappers_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'system_wrappers',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'cpu_wrapper_unittest.cc',
+ 'list_unittest.cc',
+ 'map_unittest.cc',
+ 'data_log_unittest.cc',
+ 'data_log_unittest_disabled.cc',
+ 'data_log_helpers_unittest.cc',
+ 'data_log_c_helpers_unittest.c',
+ 'data_log_c_helpers_unittest.h',
+ ],
+ 'conditions': [
+ ['enable_data_logging==1', {
+ 'sources!': [ 'data_log_unittest_disabled.cc', ],
+ }, {
+ 'sources!': [ 'data_log_unittest.cc', ],
+ }],
+ ],
+ },
+ ], # targets
+ }], # build_with_chromium
+ ], # conditions
}
# Local Variables:
diff --git a/src/system_wrappers/source/system_wrappers_tests.gyp b/src/system_wrappers/source/system_wrappers_tests.gyp
deleted file mode 100644
index 856f0c1..0000000
--- a/src/system_wrappers/source/system_wrappers_tests.gyp
+++ /dev/null
@@ -1,37 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../../common_settings.gypi',
- ],
- 'targets': [
- {
- 'target_name': 'unittest',
- 'type': 'executable',
- 'dependencies': [
- '../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '../../../testing/gtest.gyp:gtest',
- '../../../testing/gtest.gyp:gtest_main',
- ],
- 'include_dirs': [
- '../../../testing/gtest/include',
- ],
- 'sources': [
- 'list_unittest.cc',
- 'map_unittest.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/system_wrappers/source/thread.cc b/src/system_wrappers/source/thread.cc
index a136cba..32dcc63 100644
--- a/src/system_wrappers/source/thread.cc
+++ b/src/system_wrappers/source/thread.cc
@@ -11,9 +11,9 @@
#include "thread_wrapper.h"
#if defined(_WIN32)
- #include "thread_windows.h"
+ #include "thread_win.h"
#else
- #include "thread_linux.h"
+ #include "thread_posix.h"
#endif
namespace webrtc {
@@ -24,7 +24,7 @@
#if defined(_WIN32)
return new ThreadWindows(func, obj, prio, threadName);
#else
- return ThreadLinux::Create(func, obj, prio, threadName);
+ return ThreadPosix::Create(func, obj, prio, threadName);
#endif
}
} // namespace webrtc
diff --git a/src/system_wrappers/source/thread_linux.cc b/src/system_wrappers/source/thread_posix.cc
similarity index 89%
rename from src/system_wrappers/source/thread_linux.cc
rename to src/system_wrappers/source/thread_posix.cc
index 1281c1b..eb0e8f4 100644
--- a/src/system_wrappers/source/thread_linux.cc
+++ b/src/system_wrappers/source/thread_posix.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "thread_linux.h"
+#include "thread_posix.h"
#include <errno.h>
#include <string.h> // strncpy
@@ -30,12 +30,12 @@
{
static void* StartThread(void* lpParameter)
{
- static_cast<ThreadLinux*>(lpParameter)->Run();
+ static_cast<ThreadPosix*>(lpParameter)->Run();
return 0;
}
}
-#if (defined(WEBRTC_LINUX) && !defined(ANDROID))
+#if (defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID))
static pid_t gettid()
{
#if defined(__NR_gettid)
@@ -46,10 +46,10 @@
}
#endif
-ThreadWrapper* ThreadLinux::Create(ThreadRunFunction func, ThreadObj obj,
+ThreadWrapper* ThreadPosix::Create(ThreadRunFunction func, ThreadObj obj,
ThreadPriority prio, const char* threadName)
{
- ThreadLinux* ptr = new ThreadLinux(func, obj, prio, threadName);
+ ThreadPosix* ptr = new ThreadPosix(func, obj, prio, threadName);
if (!ptr)
{
return NULL;
@@ -63,7 +63,7 @@
return ptr;
}
-ThreadLinux::ThreadLinux(ThreadRunFunction func, ThreadObj obj,
+ThreadPosix::ThreadPosix(ThreadRunFunction func, ThreadObj obj,
ThreadPriority prio, const char* threadName)
: _runFunction(func),
_obj(obj),
@@ -83,10 +83,10 @@
}
}
-int ThreadLinux::Construct()
+int ThreadPosix::Construct()
{
int result = 0;
-#if !defined(ANDROID)
+#if !defined(WEBRTC_ANDROID)
// Enable immediate cancellation if requested, see Shutdown()
result = pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
if (result != 0)
@@ -108,7 +108,7 @@
return 0;
}
-ThreadLinux::~ThreadLinux()
+ThreadPosix::~ThreadPosix()
{
pthread_attr_destroy(&_attr);
delete _event;
@@ -118,9 +118,9 @@
!defined(WEBRTC_MAC) && !defined(WEBRTC_MAC_INTEL) && \
!defined(MAC_DYLIB) && !defined(MAC_INTEL_DYLIB)
#if HAS_THREAD_ID
-bool ThreadLinux::Start(unsigned int& threadID)
+bool ThreadPosix::Start(unsigned int& threadID)
#else
-bool ThreadLinux::Start(unsigned int& /*threadID*/)
+bool ThreadPosix::Start(unsigned int& /*threadID*/)
#endif
{
if (!_runFunction)
@@ -191,8 +191,8 @@
return true;
}
-#if (defined(WEBRTC_LINUX) && !defined(ANDROID))
-bool ThreadLinux::SetAffinity(const int* processorNumbers,
+#if (defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID))
+bool ThreadPosix::SetAffinity(const int* processorNumbers,
const unsigned int amountOfProcessors)
{
if (!processorNumbers || (amountOfProcessors == 0))
@@ -222,20 +222,20 @@
// NOTE: On Mac OS X, use the Thread affinity API in
// /usr/include/mach/thread_policy.h: thread_policy_set and mach_thread_self()
// instead of Linux gettid() syscall.
-bool ThreadLinux::SetAffinity(const int* , const unsigned int)
+bool ThreadPosix::SetAffinity(const int* , const unsigned int)
{
return false;
}
#endif
-void ThreadLinux::SetNotAlive()
+void ThreadPosix::SetNotAlive()
{
_alive = false;
}
-bool ThreadLinux::Shutdown()
+bool ThreadPosix::Shutdown()
{
-#if !defined(ANDROID)
+#if !defined(WEBRTC_ANDROID)
if (_thread && (0 != pthread_cancel(_thread)))
{
return false;
@@ -247,7 +247,7 @@
#endif
}
-bool ThreadLinux::Stop()
+bool ThreadPosix::Stop()
{
_alive = false;
@@ -270,7 +270,7 @@
}
}
-void ThreadLinux::Run()
+void ThreadPosix::Run()
{
_alive = true;
_dead = false;
diff --git a/src/system_wrappers/source/thread_linux.h b/src/system_wrappers/source/thread_posix.h
similarity index 85%
rename from src/system_wrappers/source/thread_linux.h
rename to src/system_wrappers/source/thread_posix.h
index 3e2b908..f664a52 100644
--- a/src/system_wrappers/source/thread_linux.h
+++ b/src/system_wrappers/source/thread_posix.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
#include "thread_wrapper.h"
#include <pthread.h>
@@ -17,15 +17,15 @@
namespace webrtc {
class EventWrapper;
-class ThreadLinux : public ThreadWrapper
+class ThreadPosix : public ThreadWrapper
{
public:
static ThreadWrapper* Create(ThreadRunFunction func, ThreadObj obj,
ThreadPriority prio, const char* threadName);
- ThreadLinux(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
+ ThreadPosix(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
const char* threadName);
- ~ThreadLinux();
+ ~ThreadPosix();
// From ThreadWrapper
virtual void SetNotAlive();
@@ -66,4 +66,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
diff --git a/src/system_wrappers/source/trace_impl.cc b/src/system_wrappers/source/trace_impl.cc
index 0a5f9db..1156519 100644
--- a/src/system_wrappers/source/trace_impl.cc
+++ b/src/system_wrappers/source/trace_impl.cc
@@ -14,13 +14,12 @@
#include <string.h> // memset
#ifdef _WIN32
-#include "trace_windows.h"
-#include "fix_interlocked_exchange_pointer_windows.h"
+#include "trace_win.h"
#else
#include <stdio.h>
#include <time.h>
#include <stdarg.h>
-#include "trace_linux.h"
+#include "trace_posix.h"
#endif // _WIN32
#define KEY_LEN_CHARS 31
@@ -35,176 +34,41 @@
static WebRtc_UWord32 levelFilter = kTraceDefault;
// Construct On First Use idiom. Avoids "static initialization order fiasco".
-Trace* TraceImpl::StaticInstance(TraceCount inc, const TraceLevel level)
+TraceImpl* TraceImpl::StaticInstance(CountOperation count_operation,
+ const TraceLevel level)
{
- // TODO (hellner): use atomic wrapper instead.
- static volatile long theTraceCount = 0;
- static Trace* volatile theTrace = NULL;
-
- TraceCreate state = WEBRTC_TRACE_EXIST;
-
- // Sanitys to avoid taking lock unless absolutely necessary (for
- // performance reasons). inc == WEBRTC_TRACE_INC_NO_CREATE) implies that
- // a message will be written to file.
- if(level != kTraceAll && inc == WEBRTC_TRACE_INC_NO_CREATE)
+ // Sanities to avoid taking lock unless absolutely necessary (for
+ // performance reasons).
+ // count_operation == kAddRefNoCreate implies that a message will be
+ // written to file.
+ if((level != kTraceAll) && (count_operation == kAddRefNoCreate))
{
if(!(level & levelFilter))
{
return NULL;
}
}
-
-#ifndef _WIN32
- // TODO (pwestin): crtiSect is never reclaimed. Fix memory leak.
- static CriticalSectionWrapper* crtiSect(
- CriticalSectionWrapper::CreateCriticalSection());
- CriticalSectionScoped lock(*crtiSect);
-
- if(inc == WEBRTC_TRACE_INC_NO_CREATE && theTraceCount == 0)
- {
- return NULL;
- }
-
- if(inc == WEBRTC_TRACE_INC || inc == WEBRTC_TRACE_INC_NO_CREATE)
- {
- theTraceCount++;
- if(theTraceCount == 1)
- {
- state = WEBRTC_TRACE_CREATE;
- }
- } else {
- theTraceCount--;
- if(theTraceCount == 0)
- {
- state = WEBRTC_TRACE_DESTROY;
- }
- }
- if(state == WEBRTC_TRACE_CREATE)
- {
- theTrace = TraceImpl::CreateTrace();
-
- } else if(state == WEBRTC_TRACE_DESTROY) {
- Trace* oldValue = theTrace;
- theTrace = NULL;
- // The lock is held by the scoped critical section. Release the lock
- // temporarily so that the trace can be safely deleted. If the lock
- // was kept during the delete, e.g. creating and destroying the trace
- // too quickly may lead to a deadlock.
- // This is due to the fact that starting and stopping a ThreadWrapper
- // thread will trigger writing of trace messages.
- // TODO (hellner): remove the tight coupling with the thread
- // implementation.
- crtiSect->Leave();
- if(oldValue)
- {
- delete static_cast<TraceImpl*>(oldValue);
- }
- // Re-aqcuire the lock.
- crtiSect->Enter();
- return NULL;
- }
-#else // _WIN32
- if(inc == WEBRTC_TRACE_INC_NO_CREATE && theTraceCount == 0)
- {
- return NULL;
- }
- if(inc == WEBRTC_TRACE_INC_NO_CREATE)
- {
- if(1 == InterlockedIncrement(&theTraceCount))
- {
- // The trace has been destroyed by some other thread. Rollback.
- InterlockedDecrement(&theTraceCount);
- assert(false);
- return NULL;
- }
- // Sanity to catch corrupt state.
- if(theTrace == NULL)
- {
- assert(false);
- InterlockedDecrement(&theTraceCount);
- return NULL;
- }
- } else if(inc == WEBRTC_TRACE_INC) {
- if(theTraceCount == 0)
- {
- state = WEBRTC_TRACE_CREATE;
- } else {
- if(1 == InterlockedIncrement(&theTraceCount))
- {
- // InterlockedDecrement because reference count should not be
- // updated just yet (that's done when the trace is created).
- InterlockedDecrement(&theTraceCount);
- state = WEBRTC_TRACE_CREATE;
- }
- }
- } else {
- int newValue = InterlockedDecrement(&theTraceCount);
- if(newValue == 0)
- {
- state = WEBRTC_TRACE_DESTROY;
- }
- }
-
- if(state == WEBRTC_TRACE_CREATE)
- {
- // Create trace and let whichever thread finishes first assign its local
- // copy to the global instance. All other threads reclaim their local
- // copy.
- Trace* newTrace = TraceImpl::CreateTrace();
- if(1 == InterlockedIncrement(&theTraceCount))
- {
- Trace* oldValue = (Trace*)InterlockedExchangePointer(
- reinterpret_cast<void* volatile*>(&theTrace), newTrace);
- assert(oldValue == NULL);
- assert(theTrace);
- } else {
- InterlockedDecrement(&theTraceCount);
- if(newTrace)
- {
- delete static_cast<TraceImpl*>(newTrace);
- }
- }
- return NULL;
- } else if(state == WEBRTC_TRACE_DESTROY)
- {
- Trace* oldValue = (Trace*)InterlockedExchangePointer(
- reinterpret_cast<void* volatile*>(&theTrace), NULL);
- if(oldValue)
- {
- delete static_cast<TraceImpl*>(oldValue);
- }
- return NULL;
- }
-#endif // #ifndef _WIN32
- return theTrace;
-}
-
-void Trace::CreateTrace()
-{
- TraceImpl::StaticInstance(WEBRTC_TRACE_INC);
-}
-
-void Trace::ReturnTrace()
-{
- TraceImpl::StaticInstance(WEBRTC_TRACE_DEC);
+ TraceImpl* impl =
+ GetStaticInstance<TraceImpl>(count_operation);
+ return impl;
}
TraceImpl* TraceImpl::GetTrace(const TraceLevel level)
{
- return (TraceImpl*)StaticInstance(WEBRTC_TRACE_INC_NO_CREATE, level);
+ return StaticInstance(kAddRefNoCreate, level);
}
-Trace* TraceImpl::CreateTrace()
+TraceImpl* TraceImpl::CreateInstance()
{
#if defined(_WIN32)
return new TraceWindows();
#else
- return new TraceLinux();
+ return new TracePosix();
#endif
}
TraceImpl::TraceImpl()
- : _critsectInterface(*CriticalSectionWrapper::CreateCriticalSection()),
+ : _critsectInterface(CriticalSectionWrapper::CreateCriticalSection()),
_callback(NULL),
_rowCountText(0),
_fileCountText(0),
@@ -212,7 +76,7 @@
_thread(*ThreadWrapper::CreateThread(TraceImpl::Run, this,
kHighestPriority, "Trace")),
_event(*EventWrapper::Create()),
- _critsectArray(*CriticalSectionWrapper::CreateCriticalSection()),
+ _critsectArray(CriticalSectionWrapper::CreateCriticalSection()),
_nextFreeIdx(),
_level(),
_length(),
@@ -271,8 +135,8 @@
delete &_event;
delete &_traceFile;
delete &_thread;
- delete &_critsectInterface;
- delete &_critsectArray;
+ delete _critsectInterface;
+ delete _critsectArray;
for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
{
@@ -607,9 +471,12 @@
if(_nextFreeIdx[_activeQueue] == WEBRTC_TRACE_MAX_QUEUE-1)
{
- // Loggin more messages than can be worked off. Log a warning.
+ // Logging more messages than can be worked off. Log a warning.
+ const char warning_msg[] = "WARNING MISSING TRACE MESSAGES\n";
+ _level[_activeQueue][_nextFreeIdx[_activeQueue]] = kTraceWarning;
+ _length[_activeQueue][_nextFreeIdx[_activeQueue]] = strlen(warning_msg);
memcpy(_messageQueue[_activeQueue][_nextFreeIdx[_activeQueue]],
- "WARNING MISSING TRACE MESSAGES\n", 32);
+ warning_msg, _length[_activeQueue][idx]);
_nextFreeIdx[_activeQueue]++;
}
}
@@ -832,7 +699,8 @@
}
memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthTo_);
- sprintf(fileNameWithCounterUTF8+lengthTo_, "_%lu%s", newCount,
+ sprintf(fileNameWithCounterUTF8+lengthTo_, "_%lu%s",
+ static_cast<long unsigned int> (newCount),
fileNameUTF8+lengthWithoutFileEnding);
return true;
}
@@ -865,21 +733,32 @@
}
memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthWithoutFileEnding);
sprintf(fileNameWithCounterUTF8+lengthWithoutFileEnding, "_%lu%s",
- newCount, fileNameUTF8+lengthWithoutFileEnding);
+ static_cast<long unsigned int> (newCount),
+ fileNameUTF8+lengthWithoutFileEnding);
return true;
}
+void Trace::CreateTrace()
+{
+ TraceImpl::StaticInstance(kAddRef);
+}
+
+void Trace::ReturnTrace()
+{
+ TraceImpl::StaticInstance(kRelease);
+}
+
WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 filter)
{
levelFilter = filter;
return 0;
-};
+}
WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& filter)
{
filter = levelFilter;
return 0;
-};
+}
WebRtc_Word32 Trace::TraceFile(WebRtc_Word8 fileName[FileWrapper::kMaxFileNameSize])
{
@@ -946,4 +825,5 @@
ReturnTrace();
}
}
+
} // namespace webrtc
diff --git a/src/system_wrappers/source/trace_impl.h b/src/system_wrappers/source/trace_impl.h
index 42e82fe..455a3d5 100644
--- a/src/system_wrappers/source/trace_impl.h
+++ b/src/system_wrappers/source/trace_impl.h
@@ -14,23 +14,11 @@
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/event_wrapper.h"
#include "system_wrappers/interface/file_wrapper.h"
+#include "system_wrappers/interface/static_instance.h"
#include "system_wrappers/interface/trace.h"
#include "system_wrappers/interface/thread_wrapper.h"
namespace webrtc {
-enum TraceCount
-{
- WEBRTC_TRACE_DEC = 0,
- WEBRTC_TRACE_INC = 1,
- WEBRTC_TRACE_INC_NO_CREATE = 2
-};
-
-enum TraceCreate
-{
- WEBRTC_TRACE_EXIST = 0,
- WEBRTC_TRACE_CREATE = 1,
- WEBRTC_TRACE_DESTROY = 2
-};
// TODO (pwestin) WEBRTC_TRACE_MAX_QUEUE needs to be tweaked
// TODO (hellner) the buffer should be close to how much the system can write to
@@ -58,12 +46,9 @@
public:
virtual ~TraceImpl();
- static Trace* CreateTrace();
+ static TraceImpl* CreateInstance();
static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
- static Trace* StaticInstance(TraceCount inc,
- const TraceLevel level = kTraceAll);
-
WebRtc_Word32 SetTraceFileImpl(const WebRtc_Word8* fileName,
const bool addFileCounter);
WebRtc_Word32 TraceFileImpl(
@@ -81,6 +66,9 @@
protected:
TraceImpl();
+ static TraceImpl* StaticInstance(CountOperation count_operation,
+ const TraceLevel level = kTraceAll);
+
// OS specific implementations
virtual WebRtc_Word32 AddThreadId(char* traceMessage) const = 0;
virtual WebRtc_Word32 AddTime(char* traceMessage,
@@ -93,6 +81,8 @@
bool Process();
private:
+ friend class Trace;
+
WebRtc_Word32 AddLevel(char* szMessage, const TraceLevel level) const;
WebRtc_Word32 AddModuleAndId(char* traceMessage, const TraceModule module,
@@ -119,7 +109,7 @@
void WriteToFile();
- CriticalSectionWrapper& _critsectInterface;
+ CriticalSectionWrapper* _critsectInterface;
TraceCallback* _callback;
WebRtc_UWord32 _rowCountText;
WebRtc_UWord32 _fileCountText;
@@ -129,7 +119,7 @@
EventWrapper& _event;
// _critsectArray protects _activeQueue
- CriticalSectionWrapper& _critsectArray;
+ CriticalSectionWrapper* _critsectArray;
WebRtc_UWord16 _nextFreeIdx[WEBRTC_TRACE_NUM_ARRAY];
TraceLevel _level[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
WebRtc_UWord16 _length[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
diff --git a/src/system_wrappers/source/trace_impl_no_op.cc b/src/system_wrappers/source/trace_impl_no_op.cc
new file mode 100644
index 0000000..1752871
--- /dev/null
+++ b/src/system_wrappers/source/trace_impl_no_op.cc
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+
+namespace webrtc {
+
+void Trace::CreateTrace()
+{
+}
+
+void Trace::ReturnTrace()
+{
+}
+
+WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 /*filter*/)
+{
+ return 0;
+}
+
+WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& /*filter*/)
+{
+ return 0;
+}
+
+WebRtc_Word32 Trace::TraceFile(
+ WebRtc_Word8 /*fileName*/[1024])
+{
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceFile(const WebRtc_Word8* /*fileName*/,
+ const bool /*addFileCounter*/)
+{
+ return -1;
+}
+
+WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* /*callback*/)
+{
+ return -1;
+}
+
+void Trace::Add(const TraceLevel /*level*/, const TraceModule /*module*/,
+ const WebRtc_Word32 /*id*/, const char* /*msg*/, ...)
+
+{
+}
+
+} // namespace webrtc
diff --git a/src/system_wrappers/source/trace_linux.cc b/src/system_wrappers/source/trace_posix.cc
similarity index 80%
rename from src/system_wrappers/source/trace_linux.cc
rename to src/system_wrappers/source/trace_posix.cc
index 8dba3be..198c434 100644
--- a/src/system_wrappers/source/trace_linux.cc
+++ b/src/system_wrappers/source/trace_posix.cc
@@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "trace_linux.h"
+#include "trace_posix.h"
#include <cassert>
#include <stdarg.h>
#include <stdio.h>
#include <string.h>
#include <time.h>
-
-#ifdef ANDROID
+#ifdef __linux__
+ #include <sys/syscall.h>
+#endif
+#ifdef WEBRTC_ANDROID
#include <pthread.h>
#else
#include <iostream>
@@ -37,26 +39,31 @@
#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
namespace webrtc {
-TraceLinux::TraceLinux()
+TracePosix::TracePosix()
{
_prevAPITickCount = time(NULL);
_prevTickCount = _prevAPITickCount;
}
-TraceLinux::~TraceLinux()
+TracePosix::~TracePosix()
{
StopThread();
}
-WebRtc_Word32 TraceLinux::AddThreadId(char* traceMessage) const
-{
- WebRtc_UWord64 threadId = (WebRtc_UWord64)pthread_self();
- sprintf(traceMessage, "%10llu; ", threadId);
- // 12 bytes are written.
- return 12;
+WebRtc_Word32 TracePosix::AddThreadId(char* traceMessage) const {
+#ifdef __linux__
+ pid_t threadId = (pid_t) syscall(__NR_gettid);
+ sprintf(traceMessage, "%10d; ", threadId);
+#else
+ WebRtc_UWord64 threadId = (WebRtc_UWord64)pthread_self();
+ sprintf(traceMessage, "%10llu; ",
+ static_cast<long long unsigned int>(threadId));
+#endif
+ // 12 bytes are written.
+ return 12;
}
-WebRtc_Word32 TraceLinux::AddTime(char* traceMessage,
+WebRtc_Word32 TracePosix::AddTime(char* traceMessage,
const TraceLevel level) const
{
time_t dwCurrentTimeInSeconds = time(NULL);
@@ -109,14 +116,14 @@
return 22;
}
-WebRtc_Word32 TraceLinux::AddBuildInfo(char* traceMessage) const
+WebRtc_Word32 TracePosix::AddBuildInfo(char* traceMessage) const
{
sprintf(traceMessage, "Build info: %s", BUILDINFO);
// Include NULL termination (hence + 1).
return strlen(traceMessage) + 1;
}
-WebRtc_Word32 TraceLinux::AddDateTimeInfo(char* traceMessage) const
+WebRtc_Word32 TracePosix::AddDateTimeInfo(char* traceMessage) const
{
time_t t;
time(&t);
diff --git a/src/system_wrappers/source/trace_linux.h b/src/system_wrappers/source/trace_posix.h
similarity index 79%
rename from src/system_wrappers/source/trace_linux.h
rename to src/system_wrappers/source/trace_posix.h
index 6e327a0..099bcc8 100644
--- a/src/system_wrappers/source/trace_linux.h
+++ b/src/system_wrappers/source/trace_posix.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
+#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
#include "critical_section_wrapper.h"
#include "trace_impl.h"
namespace webrtc {
-class TraceLinux : public TraceImpl
+class TracePosix : public TraceImpl
{
public:
- TraceLinux();
- virtual ~TraceLinux();
+ TracePosix();
+ virtual ~TracePosix();
virtual WebRtc_Word32 AddThreadId(char *traceMessage) const;
virtual WebRtc_Word32 AddTime(char* traceMessage,
@@ -34,4 +34,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
diff --git a/src/system_wrappers/test/Test.cpp b/src/system_wrappers/test/Test.cpp
deleted file mode 100644
index 7a34166..0000000
--- a/src/system_wrappers/test/Test.cpp
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <iostream>
-
-#ifdef _WIN32
- #include <windows.h>
- #include <tchar.h>
-#else
- #include <stdio.h>
- #define Sleep(x) usleep(x*1000)
-#endif
-
-#include "common_types.h"
-#include "trace.h"
-#include "cpu_wrapper.h"
-
-
-#ifdef _WIN32
-int _tmain(int argc, _TCHAR* argv[])
-#else
-int main(int argc, char* argv[])
-#endif
-{
- Trace::CreateTrace();
- Trace::SetTraceFile("testTrace.txt");
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- printf("Start system wrapper test\n");
-
- printf("Number of cores detected:%u\n", (unsigned int)CpuWrapper::DetectNumberOfCores());
-
- CpuWrapper* cpu = CpuWrapper::CreateCpu();
-
- WebRtc_UWord32 numCores;
- WebRtc_UWord32* cores;
-
- for(int i = 0; i< 10;i++)
- {
- WebRtc_Word32 total = cpu->CpuUsageMultiCore(numCores, cores);
-
- printf("\nNumCores:%d\n", (int)numCores);
- printf("Total cpu:%d\n", (int)total);
-
- for (WebRtc_UWord32 i = 0; i< numCores;i++)
- {
- printf("Core:%lu CPU:%lu \n", i, cores[i]);
- }
- Sleep(1000);
- }
-
- printf("Done system wrapper test\n");
-
- delete cpu;
-
- Trace::ReturnTrace();
-};
diff --git a/src/typedefs.h b/src/typedefs.h
index ae71690..ba87309 100644
--- a/src/typedefs.h
+++ b/src/typedefs.h
@@ -8,36 +8,40 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/*
- *
- * This file contains type definitions used in all WebRtc APIs.
- *
- */
+// This file contains platform-specific typedefs and defines.
-/* Reserved words definitions */
+#ifndef WEBRTC_TYPEDEFS_H_
+#define WEBRTC_TYPEDEFS_H_
+
+// Reserved words definitions
+// TODO(andrew): Look at removing these.
#define WEBRTC_EXTERN extern
#define G_CONST const
#define WEBRTC_INLINE extern __inline
-#ifndef WEBRTC_TYPEDEFS_H
-#define WEBRTC_TYPEDEFS_H
-
-/* Define WebRtc preprocessor identifiers based on the current build platform */
+// Define WebRTC preprocessor identifiers based on the current build platform.
+// TODO(andrew): Clean these up. We can probably remove everything in this
+// block.
+// - TARGET_MAC_INTEL and TARGET_MAC aren't used anywhere.
+// - In the few places where TARGET_PC is used, it should be replaced by
+// something more specific.
+// - Do we really support PowerPC? Probably not. Remove WEBRTC_MAC_INTEL
+// from build/common.gypi as well.
#if defined(WIN32)
- // Windows & Windows Mobile
+ // Windows & Windows Mobile.
#if !defined(WEBRTC_TARGET_PC)
#define WEBRTC_TARGET_PC
#endif
#elif defined(__APPLE__)
- // Mac OS X
- #if defined(__LITTLE_ENDIAN__ ) //TODO: is this used?
+ // Mac OS X.
+ #if defined(__LITTLE_ENDIAN__ )
#if !defined(WEBRTC_TARGET_MAC_INTEL)
#define WEBRTC_TARGET_MAC_INTEL
- #endif
+ #endif
#else
#if !defined(WEBRTC_TARGET_MAC)
#define WEBRTC_TARGET_MAC
- #endif
+ #endif
#endif
#else
// Linux etc.
@@ -46,6 +50,40 @@
#endif
#endif
+// Derived from Chromium's build/build_config.h
+// Processor architecture detection. For more info on what's defined, see:
+// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+// http://www.agner.org/optimize/calling_conventions.pdf
+// or with gcc, run: "echo | gcc -E -dM -"
+// TODO(andrew): replace WEBRTC_LITTLE_ENDIAN with WEBRTC_ARCH_LITTLE_ENDIAN?
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+// TODO(andrew): We'd prefer to control platform defines here, but this is
+// currently provided by the Android makefiles. Commented to avoid duplicate
+// definition warnings.
+//#define WEBRTC_ARCH_ARM
+// TODO(andrew): Chromium uses the following two defines. Should we switch?
+//#define WEBRTC_ARCH_ARM_FAMILY
+//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in typedefs.h
+#endif
+
+#if defined(__SSE2__) || defined(_MSC_VER)
+#define WEBRTC_USE_SSE2
+#endif
+
#if defined(WEBRTC_TARGET_PC)
#if !defined(_MSC_VER)
@@ -79,7 +117,7 @@
typedef char WebRtc_Word8;
typedef uint8_t WebRtc_UWord8;
- /* Define endian for the platform */
+ // Define endian for the platform
#define WEBRTC_LITTLE_ENDIAN
#elif defined(WEBRTC_TARGET_MAC_INTEL)
@@ -94,14 +132,11 @@
typedef uint16_t WebRtc_UWord16;
typedef uint8_t WebRtc_UWord8;
- /* Define endian for the platform */
+ // Define endian for the platform
#define WEBRTC_LITTLE_ENDIAN
#else
-
- #error "No platform defined for WebRtc type definitions (webrtc_typedefs.h)"
-
+ #error "No platform defined for WebRTC type definitions (typedefs.h)"
#endif
-
-#endif // WEBRTC_TYPEDEFS_H
+#endif // WEBRTC_TYPEDEFS_H_
diff --git a/test/OWNERS b/test/OWNERS
new file mode 100644
index 0000000..777963e
--- /dev/null
+++ b/test/OWNERS
@@ -0,0 +1,4 @@
+phoglund@webrtc.org
+kjellander@webrtc.org
+ivinnichenko@webrtc.org
+amyfong@webrtc.org
diff --git a/test/data/audio_processing/aec_far.pcm b/test/data/audio_processing/aec_far.pcm
new file mode 100644
index 0000000..fd6afc0
--- /dev/null
+++ b/test/data/audio_processing/aec_far.pcm
Binary files differ
diff --git a/test/data/audio_processing/aec_near.pcm b/test/data/audio_processing/aec_near.pcm
new file mode 100644
index 0000000..02c016c
--- /dev/null
+++ b/test/data/audio_processing/aec_near.pcm
Binary files differ
diff --git a/test/data/audio_processing/android/output_data_fixed.pb b/test/data/audio_processing/android/output_data_fixed.pb
new file mode 100644
index 0000000..2f45fd3
--- /dev/null
+++ b/test/data/audio_processing/android/output_data_fixed.pb
Binary files differ
diff --git a/test/data/audio_processing/android/output_data_float.pb b/test/data/audio_processing/android/output_data_float.pb
new file mode 100644
index 0000000..1bf18c2
--- /dev/null
+++ b/test/data/audio_processing/android/output_data_float.pb
Binary files differ
diff --git a/test/data/audio_processing/output_data_fixed.pb b/test/data/audio_processing/output_data_fixed.pb
new file mode 100644
index 0000000..81bc5af
--- /dev/null
+++ b/test/data/audio_processing/output_data_fixed.pb
Binary files differ
diff --git a/test/data/audio_processing/output_data_float.pb b/test/data/audio_processing/output_data_float.pb
new file mode 100644
index 0000000..ccd7509
--- /dev/null
+++ b/test/data/audio_processing/output_data_float.pb
Binary files differ
diff --git a/test/functional_test/README b/test/functional_test/README
new file mode 100644
index 0000000..a855135
--- /dev/null
+++ b/test/functional_test/README
@@ -0,0 +1,41 @@
+This test client is a simple functional test for WebRTC enabled Chrome build.
+
+The following is necessary to run the test:
+- A WebRTC Chrome binary.
+- A peerconnection_server binary (make peerconnection_server).
+
+It can be used in two scenarios:
+1. Single client calling itself with the server test page
+(peerconnection/samples/server/server_test.html) in loopback mode as a fake
+client.
+2. Call between two clients.
+
+To start the test for scenario (1):
+1. Start peerconnection_server.
+2. Start the WebRTC Chrome build: $ <path_to_chome_binary>/chrome
+--enable-media-stream --enable-p2papi --user-data-dir=<path_to_data>
+<path_to_data> is where Chrome looks for all its states, use for example
+"temp/chrome_webrtc_data". If you don't always start the browser from the same
+directory, use an absolute path instead.
+3. Open the server test page, ensure loopback is enabled, choose a name (for
+example "loopback") and connect to the server.
+4. Open the test page, connect to the server, select the loopback peer, click
+call.
+
+To start the test for scenario (2):
+1. Start peerconnection_server.
+2. Start the WebRTC Chrome build, see scenario (1).
+3. Open the test page, connect to the server.
+4. On another machine, start the WebRTC Chrome build.
+5. Open the test page, connect to the server, select the other peer, click call.
+
+Note 1: There is currently a limitation so that the camera device can only be
+accessed once, even if in the same browser instance. Hence the need to use two
+machines for scenario (2).
+
+Note 2: The web page must normally be on a web server to be able to access the
+camera for security reasons.
+See http://blog.chromium.org/2008/12/security-in-depth-local-web-pages.html
+for more details on this topic. This can be overridden with the flag
+--allow-file-access-from-files, in which case running it over the file://
+URI scheme works.
diff --git a/test/functional_test/webrtc_test.html b/test/functional_test/webrtc_test.html
new file mode 100644
index 0000000..e2d8939
--- /dev/null
+++ b/test/functional_test/webrtc_test.html
@@ -0,0 +1,594 @@
+<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
+
+<!--
+Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+
+Use of this source code is governed by a BSD-style license
+that can be found in the LICENSE file in the root of the source
+tree. An additional intellectual property rights grant can be found
+in the file PATENTS. All contributing project authors may
+be found in the AUTHORS file in the root of the source tree.
+-->
+
+<html>
+
+<head>
+<title>WebRTC Test</title>
+
+<style type="text/css">
+body, input, button, select, table {
+ font-family:"Lucida Grande", "Lucida Sans", Verdana, Arial, sans-serif;
+ font-size: 13 px;
+}
+body, input:enable, button:enable, select:enable, table {
+ color: rgb(51, 51, 51);
+}
+h1 {font-size: 40 px;}
+</style>
+
+<script type="text/javascript">
+
+// TODO: Catch more exceptions
+
+var server;
+var myId = -1;
+var myName;
+var remoteId = -1;
+var remoteName;
+var request = null;
+var hangingGet = null;
+var pc = null;
+var localStream = null;
+var disconnecting = false;
+var callState = 0; // 0 - Not started, 1 - Call ongoing
+
+
+// General
+
+function toggleExtraButtons() {
+ document.getElementById("createPcBtn").hidden =
+ !document.getElementById("createPcBtn").hidden;
+ document.getElementById("test1Btn").hidden =
+ !document.getElementById("test1Btn").hidden;
+}
+
+function trace(txt) {
+ var elem = document.getElementById("debug");
+ elem.innerHTML += txt + "<br>";
+}
+
+function trace_warning(txt) {
+ var wtxt = "<b>" + txt + "</b>";
+ trace(wtxt);
+}
+
+function trace_exception(e, txt) {
+ var etxt = "<b>" + txt + "</b> (" + e.name + " / " + e.message + ")";
+ trace(etxt);
+}
+
+function setCallState(state) {
+ trace("Changing call state: " + callState + " -> " + state);
+ callState = state;
+}
+
+function checkPeerConnection() {
+ if (!pc) {
+ trace_warning("No PeerConnection object exists");
+ return 0;
+ }
+ return 1;
+}
+
+
+// Local stream generation
+
+function gotStream(s) {
+ var url = webkitURL.createObjectURL(s);
+ document.getElementById("localView").src = url;
+ trace("User has granted access to local media. url = " + url);
+ localStream = s;
+}
+
+function gotStreamFailed(error) {
+ alert("Failed to get access to local media. Error code was " + error.code +
+ ".");
+ trace_warning("Failed to get access to local media. Error code was " +
+ error.code);
+}
+
+function getUserMedia() {
+ try {
+ navigator.webkitGetUserMedia("video,audio", gotStream, gotStreamFailed);
+ trace("Requested access to local media");
+ } catch (e) {
+ trace_exception(e, "getUserMedia error");
+ }
+}
+
+
+// Peer list and remote peer handling
+
+function peerExists(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id)
+ return true;
+ }
+ } catch (e) {
+ trace_exception(e, "Error searching for peer");
+ }
+ return false;
+}
+
+function addPeer(id, pname) {
+ var peerList = document.getElementById("peers");
+ var option = document.createElement("option");
+ option.text = pname;
+ option.value = id;
+ try {
+ // For IE earlier than version 8
+ peerList.add(option, x.options[null]);
+ } catch (e) {
+ peerList.add(option, null);
+ }
+}
+
+function removePeer(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ peerList.remove(i);
+ break;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error removing peer");
+ }
+}
+
+function clearPeerList() {
+ var peerList = document.getElementById("peers");
+ while (peerList.length > 0)
+ peerList.remove(0);
+}
+
+function setSelectedPeer(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ peerList.options[i].selected = true;
+ return true;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error setting selected peer");
+ }
+ return false;
+}
+
+function getPeerName(id) {
+ try {
+ var peerList = document.getElementById("peers");
+ for (var i = 0; i < peerList.length; i++) {
+ if (parseInt(peerList.options[i].value) == id) {
+ return peerList.options[i].text;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Error finding peer name");
+ return;
+ }
+ return;
+}
+
+function storeRemoteInfo() {
+ try {
+ var peerList = document.getElementById("peers");
+ if (peerList.selectedIndex < 0) {
+ alert("Please select a peer.");
+ return false;
+ } else
+ remoteId = parseInt(peerList.options[peerList.selectedIndex].value);
+ remoteName = peerList.options[peerList.selectedIndex].text;
+ } catch (e) {
+ trace_exception(e, "Error storing remote peer info");
+ return false;
+ }
+ return true;
+}
+
+
+// Call control
+
+function createPeerConnection() {
+ if (pc) {
+ trace_warning("PeerConnection object already exists");
+ }
+ trace("Creating PeerConnection object");
+ try {
+ pc = new webkitPeerConnection("STUN stun.l.google.com:19302",
+ onSignalingMessage);
+ pc.onaddstream = onAddStream;
+ pc.onremovestream = onRemoveStream;
+ } catch (e) {
+ trace_exception(e, "Create PeerConnection error");
+ }
+}
+
+function doCall() {
+ if (!storeRemoteInfo())
+ return;
+ document.getElementById("call").disabled = true;
+ document.getElementById("peers").disabled = true;
+ createPeerConnection();
+ trace("Adding stream");
+ pc.addStream(localStream);
+ document.getElementById("hangup").disabled = false;
+ setCallState(1);
+}
+
+function hangUp() {
+ document.getElementById("hangup").disabled = true;
+ trace("Sending BYE to " + remoteName + " (ID " + remoteId + ")");
+ sendToPeer(remoteId, "BYE");
+ closeCall();
+}
+
+function closeCall() {
+ trace("Stopping showing remote stream");
+ document.getElementById("remoteView").src = "dummy";
+ if (pc) {
+ trace("Stopping call [pc.close()]");
+ pc.close();
+ pc = null;
+ } else
+ trace("No pc object to close");
+ remoteId = -1;
+ document.getElementById("call").disabled = false;
+ document.getElementById("peers").disabled = false;
+ setCallState(0);
+}
+
+
+// PeerConnection callbacks
+
+function onAddStream(e) {
+ var stream = e.stream;
+ var url = webkitURL.createObjectURL(stream);
+ document.getElementById("remoteView").src = url;
+ trace("Started showing remote stream. url = " + url);
+}
+
+function onRemoveStream(e) {
+ // Currently if we get this callback, call has ended.
+ document.getElementById("remoteView").src = "";
+ trace("Stopped showing remote stream");
+}
+
+function onSignalingMessage(msg) {
+ trace("Sending message to " + remoteName + " (ID " + remoteId + "):\n" + msg);
+ sendToPeer(remoteId, msg);
+}
+
+// TODO: Add callbacks onconnecting, onopen and onstatechange.
+
+
+// Server interaction
+
+function handleServerNotification(data) {
+ trace("Server notification: " + data);
+ var parsed = data.split(",");
+ if (parseInt(parsed[2]) == 1) { // New peer
+ var peerId = parseInt(parsed[1]);
+ if (!peerExists(peerId)) {
+ var peerList = document.getElementById("peers");
+ if (peerList.length == 1 && peerList.options[0].value == -1)
+ clearPeerList();
+ addPeer(peerId, parsed[0]);
+ document.getElementById("peers").disabled = false;
+ document.getElementById("call").disabled = false;
+ }
+ } else if (parseInt(parsed[2]) == 0) { // Removed peer
+ removePeer(parseInt(parsed[1]));
+ if (document.getElementById("peers").length == 0) {
+ document.getElementById("peers").disabled = true;
+ addPeer(-1, "No other peer connected");
+ }
+ }
+}
+
+function handlePeerMessage(peer_id, msg) {
+ var peerName = getPeerName(peer_id);
+ if (peerName == undefined) {
+ trace_warning("Received message from unknown peer (ID " + peer_id +
+ "), ignoring message:");
+ trace(msg);
+ return;
+ }
+ trace("Received message from " + peerName + " (ID " + peer_id + "):\n" + msg);
+ // Assuming we receive the message from the peer we want to communicate with.
+ // TODO: Only accept messages from peer we communicate with with if call is
+ // ongoing.
+ if (msg.search("BYE") == 0) {
+ // Other side has hung up.
+ document.getElementById("hangup").disabled = true;
+ closeCall()
+ } else {
+ if (!pc) {
+ // Other side is calling us, startup
+ if (!setSelectedPeer(peer_id)) {
+ trace_warning("Recevied message from unknown peer, ignoring");
+ return;
+ }
+ if (!storeRemoteInfo())
+ return;
+ document.getElementById("call").disabled = true;
+ document.getElementById("peers").disabled = true;
+ createPeerConnection();
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_exception(e, "Process signaling message error");
+ }
+ trace("Adding stream");
+ pc.addStream(localStream);
+ document.getElementById("hangup").disabled = false;
+ } else {
+ try {
+ pc.processSignalingMessage(msg);
+ } catch (e) {
+ trace_exception(e, "Process signaling message error");
+ }
+ }
+ }
+}
+
+function getIntHeader(r, name) {
+ var val = r.getResponseHeader(name);
+ trace("header value: " + val);
+ return val != null && val.length ? parseInt(val) : -1;
+}
+
+function hangingGetCallback() {
+ try {
+ if (hangingGet.readyState != 4 || disconnecting)
+ return;
+ if (hangingGet.status != 200) {
+ trace_warning("server error, status: " + hangingGet.status + ", text: " +
+ hangingGet.statusText);
+ disconnect();
+ } else {
+ var peer_id = getIntHeader(hangingGet, "Pragma");
+ if (peer_id == myId) {
+ handleServerNotification(hangingGet.responseText);
+ } else {
+ handlePeerMessage(peer_id, hangingGet.responseText);
+ }
+ }
+
+ if (hangingGet) {
+ hangingGet.abort();
+ hangingGet = null;
+ }
+
+ if (myId != -1)
+ window.setTimeout(startHangingGet, 0);
+ } catch (e) {
+ trace_exception(e, "Hanging get error");
+ }
+}
+
+function onHangingGetTimeout() {
+ trace("hanging get timeout. issuing again");
+ hangingGet.abort();
+ hangingGet = null;
+ if (myId != -1)
+ window.setTimeout(startHangingGet, 0);
+}
+
+function startHangingGet() {
+ try {
+ hangingGet = new XMLHttpRequest();
+ hangingGet.onreadystatechange = hangingGetCallback;
+ hangingGet.ontimeout = onHangingGetTimeout;
+ hangingGet.open("GET", server + "/wait?peer_id=" + myId, true);
+ hangingGet.send();
+ } catch (e) {
+ trace_exception(e, "Start hanging get error");
+ }
+}
+
+function sendToPeer(peer_id, data) {
+ if (myId == -1) {
+ alert("Not connected.");
+ return;
+ }
+ if (peer_id == myId) {
+ alert("Can't send a message to oneself.");
+ return;
+ }
+ var r = new XMLHttpRequest();
+ r.open("POST", server + "/message?peer_id=" + myId + "&to=" + peer_id, false);
+ r.setRequestHeader("Content-Type", "text/plain");
+ r.send(data);
+ r = null;
+}
+
+function signInCallback() {
+ try {
+ if (request.readyState == 4) {
+ if (request.status == 200) {
+ var peers = request.responseText.split("\n");
+ myId = parseInt(peers[0].split(",")[1]);
+ trace("My id: " + myId);
+ clearPeerList();
+ var added = 0;
+ for (var i = 1; i < peers.length; ++i) {
+ if (peers[i].length > 0) {
+ trace("Peer " + i + ": " + peers[i]);
+ var parsed = peers[i].split(",");
+ addPeer(parseInt(parsed[1]), parsed[0]);
+ ++added;
+ }
+ }
+ if (added == 0)
+ addPeer(-1, "No other peer connected");
+ else {
+ document.getElementById("peers").disabled = false;
+ document.getElementById("call").disabled = false;
+ }
+ startHangingGet();
+ request = null;
+ document.getElementById("connect").disabled = true;
+ document.getElementById("disconnect").disabled = false;
+ }
+ }
+ } catch (e) {
+ trace_exception(e, "Sign in error");
+ document.getElementById("connect").disabled = false;
+ }
+}
+
+function signIn() {
+ try {
+ request = new XMLHttpRequest();
+ request.onreadystatechange = signInCallback;
+ request.open("GET", server + "/sign_in?" + myName, true);
+ request.send();
+ } catch (e) {
+ trace_exception(e, "Start sign in error");
+ document.getElementById("connect").disabled = false;
+ }
+}
+
+function connect() {
+ myName = document.getElementById("local").value.toLowerCase();
+ server = document.getElementById("server").value.toLowerCase();
+ if (myName.length == 0) {
+ alert("I need a name please.");
+ document.getElementById("local").focus();
+ } else {
+ // TODO: Disable connect button here, but we need a timeout and check if we
+ // have connected, if so enable it again.
+ signIn();
+ }
+}
+
+function disconnect() {
+ if (callState == 1)
+ hangUp();
+
+ disconnecting = true;
+
+ if (request) {
+ request.abort();
+ request = null;
+ }
+
+ if (hangingGet) {
+ hangingGet.abort();
+ hangingGet = null;
+ }
+
+ if (myId != -1) {
+ request = new XMLHttpRequest();
+ request.open("GET", server + "/sign_out?peer_id=" + myId, false);
+ request.send();
+ request = null;
+ myId = -1;
+ }
+
+ clearPeerList();
+ addPeer(-1, "Not connected");
+ document.getElementById("connect").disabled = false;
+ document.getElementById("disconnect").disabled = true;
+ document.getElementById("peers").disabled = true;
+ document.getElementById("call").disabled = true;
+
+ disconnecting = false;
+}
+
+
+// Window event handling
+
+window.onload = getUserMedia;
+window.onbeforeunload = disconnect;
+
+
+</script>
+</head>
+
+<body>
+<h1>WebRTC</h1>
+You must have a WebRTC capable browser in order to make calls using this test
+page.<br>
+
+<table border="0">
+<tr>
+ <td>Local Preview</td>
+ <td>Remote Video</td>
+</tr>
+<tr>
+ <td>
+ <video width="320" height="240" id="localView" autoplay="autoplay"></video>
+ </td>
+ <td>
+ <video width="640" height="480" id="remoteView" autoplay="autoplay"></video>
+ </td>
+</tr>
+</table>
+
+<table border="0">
+<tr>
+ <td valign="top">
+ <table border="0" cellpaddning="0" cellspacing="0">
+ <tr>
+ <td>Server:</td>
+ <td>
+ <input type="text" id="server" size="30" value="http://localhost:8888"/>
+ </td>
+ </tr>
+ <tr>
+ <td>Name:</td><td><input type="text" id="local" size="30" value="name"/></td>
+ </tr>
+ </table>
+ </td>
+ <td valign="top">
+ <button id="connect" onclick="connect();">Connect</button><br>
+ <button id="disconnect" onclick="disconnect();" disabled="true">Disconnect
+ </button>
+ </td>
+ <td> </td>
+ <td valign="top">
+ Connected peers:<br>
+ <select id="peers" size="5" disabled="true">
+ <option value="-1">Not connected</option>
+ </select>
+ </td>
+ <td valign="top">
+ <!--input type="text" id="peer_id" size="3" value="1"/><br-->
+ <button id="call" onclick="doCall();" disabled="true">Call</button><br>
+ <button id="hangup" onclick="hangUp();" disabled="true">Hang up</button><br>
+ </td>
+ <td> </td>
+ <td valign="top">
+ <button onclick="toggleExtraButtons();">Toggle extra buttons (debug)</button>
+ <br>
+ <button id="createPcBtn" onclick="createPeerConnection();" hidden="true">
+ Create peer connection</button>
+ </td>
+</tr>
+</table>
+
+<button onclick="document.getElementById('debug').innerHTML='';">Clear log
+</button>
+<pre id="debug"></pre>
+
+</body>
+
+</html>
+
diff --git a/test/metrics.gyp b/test/metrics.gyp
new file mode 100644
index 0000000..70483f9
--- /dev/null
+++ b/test/metrics.gyp
@@ -0,0 +1,46 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [
+ '../src/build/common.gypi',
+ ],
+ 'targets': [
+ {
+ # The metrics code must be kept in its own GYP file in order to
+ # avoid a circular dependency error due to the dependency on libyuv.
+ # If the code would be put in test.gyp a circular dependency error during
+ # GYP generation would occur, because the libyuv.gypi unittest target
+ # depends on test_support_main. See issue #160 for more info.
+ 'target_name': 'metrics',
+ 'type': '<(library)',
+ 'dependencies': [
+ '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
+ ],
+ 'include_dirs': [
+ '.',
+ ],
+ 'sources': [
+ 'testsupport/metrics/video_metrics.h',
+ 'testsupport/metrics/video_metrics.cc',
+ ],
+ },
+ {
+ 'target_name': 'metrics_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'metrics',
+ '<(webrtc_root)/../test/test.gyp:test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'testsupport/metrics/video_metrics_unittest.cc',
+ ],
+ },
+ ],
+}
diff --git a/test/run_all_unittests.cc b/test/run_all_unittests.cc
new file mode 100644
index 0000000..0cdf0cd
--- /dev/null
+++ b/test/run_all_unittests.cc
@@ -0,0 +1,16 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_suite.h"
+
+int main(int argc, char** argv) {
+ webrtc::test::TestSuite test_suite(argc, argv);
+ return test_suite.Run();
+}
diff --git a/test/test.gyp b/test/test.gyp
new file mode 100644
index 0000000..86a57ff
--- /dev/null
+++ b/test/test.gyp
@@ -0,0 +1,78 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# TODO(andrew): consider moving test_support to src/base/test.
+{
+ 'includes': [
+ '../src/build/common.gypi',
+ ],
+ 'targets': [
+ {
+ 'target_name': 'test_support',
+ 'type': 'static_library',
+ 'include_dirs': [
+ '.',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '.', # Some includes are hierarchical
+ ],
+ },
+ 'dependencies': [
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ '<(webrtc_root)/../testing/gmock.gyp:gmock',
+ ],
+ 'all_dependent_settings': {
+ 'include_dirs': [
+ '.',
+ ],
+ },
+ 'sources': [
+ 'test_suite.cc',
+ 'test_suite.h',
+ 'testsupport/fileutils.h',
+ 'testsupport/fileutils.cc',
+ 'testsupport/frame_reader.h',
+ 'testsupport/frame_reader.cc',
+ 'testsupport/frame_writer.h',
+ 'testsupport/frame_writer.cc',
+ 'testsupport/packet_reader.h',
+ 'testsupport/packet_reader.cc',
+ 'testsupport/mock/mock_frame_reader.h',
+ 'testsupport/mock/mock_frame_writer.h',
+ ],
+ },
+ {
+ # Depend on this target when you want to have test_support but also the
+ # main method needed for gtest to execute!
+ 'target_name': 'test_support_main',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'test_support',
+ ],
+ 'sources': [
+ 'run_all_unittests.cc',
+ ],
+ },
+ {
+ 'target_name': 'test_support_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'test_support_main',
+ '<(webrtc_root)/../testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'testsupport/unittest_utils.h',
+ 'testsupport/fileutils_unittest.cc',
+ 'testsupport/frame_reader_unittest.cc',
+ 'testsupport/frame_writer_unittest.cc',
+ 'testsupport/packet_reader_unittest.cc',
+ ],
+ },
+ ],
+}
diff --git a/test/test_suite.cc b/test/test_suite.cc
new file mode 100644
index 0000000..ac3f3a2
--- /dev/null
+++ b/test/test_suite.cc
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_suite.h"
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+
+namespace webrtc {
+namespace test {
+TestSuite::TestSuite(int argc, char** argv) {
+ testing::InitGoogleMock(&argc, argv); // Runs InitGoogleTest() internally.
+}
+
+TestSuite::~TestSuite() {
+}
+
+int TestSuite::Run() {
+ Initialize();
+ int result = RUN_ALL_TESTS();
+ Shutdown();
+ return result;
+}
+
+void TestSuite::Initialize() {
+ // TODO(andrew): initialize singletons here (e.g. Trace).
+}
+
+void TestSuite::Shutdown() {
+}
+} // namespace test
+} // namespace webrtc
diff --git a/test/test_suite.h b/test/test_suite.h
new file mode 100644
index 0000000..f500daa
--- /dev/null
+++ b/test/test_suite.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TEST_SUITE_H_
+#define TEST_TEST_SUITE_H_
+
+// Derived from Chromium's src/base/test/test_suite.h.
+
+// Defines a basic test suite framework for running gtest based tests. You can
+// instantiate this class in your main function and call its Run method to run
+// any gtest based tests that are linked into your executable.
+
+#include "src/system_wrappers/interface/constructor_magic.h"
+
+namespace webrtc {
+namespace test {
+class TestSuite {
+ public:
+ TestSuite(int argc, char** argv);
+ virtual ~TestSuite();
+
+ int Run();
+
+ protected:
+ // Override these for custom initialization and shutdown handling. Use these
+ // instead of putting complex code in your constructor/destructor.
+ virtual void Initialize();
+ virtual void Shutdown();
+
+ DISALLOW_COPY_AND_ASSIGN(TestSuite);
+};
+} // namespace test
+} // namespace webrtc
+
+#endif // TEST_TEST_SUITE_H_
diff --git a/test/testsupport/fileutils.cc b/test/testsupport/fileutils.cc
new file mode 100644
index 0000000..1e6bbca
--- /dev/null
+++ b/test/testsupport/fileutils.cc
@@ -0,0 +1,167 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/fileutils.h"
+
+#ifdef WIN32
+#include <direct.h>
+#define GET_CURRENT_DIR _getcwd
+#else
+#include <unistd.h>
+#define GET_CURRENT_DIR getcwd
+#endif
+
+#include <sys/stat.h> // To check for directory existence.
+#ifndef S_ISDIR // Not defined in stat.h on Windows.
+#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
+#endif
+
+#include <cstdio>
+
+#include "typedefs.h" // For architecture defines
+
+namespace webrtc {
+namespace test {
+
+#ifdef WIN32
+static const char* kPathDelimiter = "\\";
+#else
+static const char* kPathDelimiter = "/";
+#endif
+// The file we're looking for to identify the project root dir.
+static const char* kProjectRootFileName = "DEPS";
+static const char* kOutputDirName = "out";
+static const char* kFallbackPath = "./";
+static const char* kResourcesDirName = "resources";
+const char* kCannotFindProjectRootDir = "ERROR_CANNOT_FIND_PROJECT_ROOT_DIR";
+
+std::string ProjectRootPath() {
+ std::string working_dir = WorkingDir();
+ if (working_dir == kFallbackPath) {
+ return kCannotFindProjectRootDir;
+ }
+ // Check for our file that verifies the root dir.
+ std::string current_path(working_dir);
+ FILE* file = NULL;
+ int path_delimiter_index = current_path.find_last_of(kPathDelimiter);
+ while (path_delimiter_index > -1) {
+ std::string root_filename = current_path + kPathDelimiter +
+ kProjectRootFileName;
+ file = fopen(root_filename.c_str(), "r");
+ if (file != NULL) {
+ fclose(file);
+ return current_path + kPathDelimiter;
+ }
+ // Move up one directory in the directory tree.
+ current_path = current_path.substr(0, path_delimiter_index);
+ path_delimiter_index = current_path.find_last_of(kPathDelimiter);
+ }
+ // Reached the root directory.
+ fprintf(stderr, "Cannot find project root directory!\n");
+ return kCannotFindProjectRootDir;
+}
+
+std::string OutputPath() {
+ std::string path = ProjectRootPath();
+ if (path == kCannotFindProjectRootDir) {
+ return kFallbackPath;
+ }
+ path += kOutputDirName;
+ if (!CreateDirectory(path)) {
+ return kFallbackPath;
+ }
+ return path + kPathDelimiter;
+}
+
+std::string WorkingDir() {
+ char path_buffer[FILENAME_MAX];
+ if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
+ fprintf(stderr, "Cannot get current directory!\n");
+ return kFallbackPath;
+ } else {
+ return std::string(path_buffer);
+ }
+}
+
+bool CreateDirectory(std::string directory_name) {
+ struct stat path_info = {0};
+ // Check if the path exists already:
+ if (stat(directory_name.c_str(), &path_info) == 0) {
+ if (!S_ISDIR(path_info.st_mode)) {
+ fprintf(stderr, "Path %s exists but is not a directory! Remove this "
+ "file and re-run to create the directory.\n",
+ directory_name.c_str());
+ return false;
+ }
+ } else {
+#ifdef WIN32
+ return _mkdir(directory_name.c_str()) == 0;
+#else
+ return mkdir(directory_name.c_str(), S_IRWXU | S_IRWXG | S_IRWXO) == 0;
+#endif
+ }
+ return true;
+}
+
+bool FileExists(std::string file_name) {
+ struct stat file_info = {0};
+ return stat(file_name.c_str(), &file_info) == 0;
+}
+
+std::string ResourcePath(std::string name, std::string extension) {
+ std::string platform = "win";
+#ifdef WEBRTC_LINUX
+ platform = "linux";
+#endif // WEBRTC_LINUX
+#ifdef WEBRTC_MAC
+ platform = "mac";
+#endif // WEBRTC_MAC
+
+#ifdef WEBRTC_ARCH_64_BITS
+ std::string architecture = "64";
+#else
+ std::string architecture = "32";
+#endif // WEBRTC_ARCH_64_BITS
+
+ std::string resources_path = ProjectRootPath() + kResourcesDirName +
+ kPathDelimiter;
+ std::string resource_file = resources_path + name + "_" + platform + "_" +
+ architecture + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Try without architecture.
+ resource_file = resources_path + name + "_" + platform + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Try without platform.
+ resource_file = resources_path + name + "_" + architecture + "." + extension;
+ if (FileExists(resource_file)) {
+ return resource_file;
+ }
+ // Fall back on name without architecture or platform.
+ return resources_path + name + "." + extension;
+}
+
+size_t GetFileSize(std::string filename) {
+ FILE* f = fopen(filename.c_str(), "rb");
+ size_t size = 0;
+ if (f != NULL) {
+ if (fseek(f, 0, SEEK_END) == 0) {
+ size = ftell(f);
+ }
+ fclose(f);
+ }
+ return size;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/fileutils.h b/test/testsupport/fileutils.h
new file mode 100644
index 0000000..c89ac29
--- /dev/null
+++ b/test/testsupport/fileutils.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdio>
+
+// File utilities for testing purposes.
+//
+// The ProjectRootPath() method is a convenient way of getting an absolute
+// path to the project source tree root directory. Using this, it is easy to
+// refer to test resource files in a portable way.
+//
+// Notice that even if Windows platforms use backslash as path delimiter, it is
+// also supported to use slash, so there's no need for #ifdef checks in test
+// code for setting up the paths to the resource files.
+//
+// Example use:
+// Assume we have the following code being used in a test source file:
+// const std::string kInputFile = webrtc::test::ProjectRootPath() +
+// "test/data/voice_engine/audio_long16.wav";
+// // Use the kInputFile for the tests...
+//
+// Then here's some example outputs for different platforms:
+// Linux:
+// * Source tree located in /home/user/webrtc/trunk
+// * Test project located in /home/user/webrtc/trunk/src/testproject
+// * Test binary compiled as:
+// /home/user/webrtc/trunk/out/Debug/testproject_unittests
+// Then ProjectRootPath() will return /home/user/webrtc/trunk/ no matter if
+// the test binary is executed from standing in either of:
+// /home/user/webrtc/trunk
+// or
+// /home/user/webrtc/trunk/out/Debug
+// (or any other directory below the trunk for that matter).
+//
+// Windows:
+// * Source tree located in C:\Users\user\webrtc\trunk
+// * Test project located in C:\Users\user\webrtc\trunk\src\testproject
+// * Test binary compiled as:
+// C:\Users\user\webrtc\trunk\src\testproject\Debug\testproject_unittests.exe
+// Then ProjectRootPath() will return C:\Users\user\webrtc\trunk\ when the
+// test binary is executed from inside Visual Studio.
+// It will also return the same path if the test is executed from a command
+// prompt standing in C:\Users\user\webrtc\trunk\src\testproject\Debug
+//
+// Mac:
+// * Source tree located in /Users/user/webrtc/trunk
+// * Test project located in /Users/user/webrtc/trunk/src/testproject
+// * Test binary compiled as:
+// /Users/user/webrtc/trunk/xcodebuild/Debug/testproject_unittests
+// Then ProjectRootPath() will return /Users/user/webrtc/trunk/ no matter if
+// the test binary is executed from standing in either of:
+// /Users/user/webrtc/trunk
+// or
+// /Users/user/webrtc/trunk/out/Debug
+// (or any other directory below the trunk for that matter).
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
+#define WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
+
+#include <string>
+
+namespace webrtc {
+namespace test {
+
+// This is the "directory" returned if the ProjectPath() function fails
+// to find the project root.
+extern const char* kCannotFindProjectRootDir;
+
+// Finds the root dir of the project, to be able to set correct paths to
+// resource files used by tests.
+// The implementation is simple: it just looks for the file defined by
+// kProjectRootFileName, starting in the current directory (the working
+// directory) and then steps upward until it is found (or it is at the root of
+// the file system).
+// If the current working directory is above the project root dir, it will not
+// be found.
+//
+// If symbolic links occur in the path they will be resolved and the actual
+// directory will be returned.
+//
+// Returns the absolute path to the project root dir (usually the trunk dir)
+// WITH a trailing path delimiter.
+// If the project root is not found, the string specified by
+// kCannotFindProjectRootDir is returned.
+std::string ProjectRootPath();
+
+// Creates and returns the absolute path to the output directory where log files
+// and other test artifacts should be put. The output directory is always a
+// directory named "out" at the top-level of the project, i.e. a subfolder to
+// the path returned by ProjectRootPath().
+//
+// Details described for ProjectRootPath() apply here too.
+//
+// Returns the path WITH a trailing path delimiter. If the project root is not
+// found, the current working directory ("./") is returned as a fallback.
+std::string OutputPath();
+
+// Returns a path to a resource file for the currently executing platform.
+// Adapts to what filenames are currently present in the
+// [project-root]/resources/ dir.
+// Returns an absolute path according to this priority list (the directory
+// part of the path is left out for readability):
+// 1. [name]_[platform]_[architecture].[extension]
+// 2. [name]_[platform].[extension]
+// 3. [name]_[architecture].[extension]
+// 4. [name].[extension]
+// Where
+// * platform is either of "win", "mac" or "linux".
+// * architecture is either of "32" or "64".
+//
+// Arguments:
+// name - Name of the resource file. If a plain filename (no directory path)
+// is supplied, the file is assumed to be located in resources/
+// If a directory path is prepended to the filename, a subdirectory
+// hierarchy reflecting that path is assumed to be present.
+// extension - File extension, without the dot, i.e. "bmp" or "yuv".
+std::string ResourcePath(std::string name, std::string extension);
+
+// Gets the current working directory for the executing program.
+// Returns "./" if for some reason it is not possible to find the working
+// directory.
+std::string WorkingDir();
+
+// Creates a directory if it not already exists.
+// Returns true if successful. Will print an error message to stderr and return
+// false if a file with the same name already exists.
+bool CreateDirectory(std::string directory_name);
+
+// File size of the supplied file in bytes. Will return 0 if the file is
+// empty or if the file does not exist/is readable.
+size_t GetFileSize(std::string filename);
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
diff --git a/test/testsupport/fileutils_unittest.cc b/test/testsupport/fileutils_unittest.cc
new file mode 100644
index 0000000..a500a07
--- /dev/null
+++ b/test/testsupport/fileutils_unittest.cc
@@ -0,0 +1,191 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/fileutils.h"
+
+#include <cstdio>
+#include <list>
+#include <string>
+
+#include "gtest/gtest.h"
+
+#ifdef WIN32
+static const char* kPathDelimiter = "\\";
+#else
+static const char* kPathDelimiter = "/";
+#endif
+
+static const std::string kDummyDir = "file_utils_unittest_dummy_dir";
+static const std::string kResourcesDir = "resources";
+static const std::string kTestName = "fileutils_unittest";
+static const std::string kExtension = "tmp";
+
+typedef std::list<std::string> FileList;
+
+namespace webrtc {
+
+// Test fixture to restore the working directory between each test, since some
+// of them change it with chdir during execution (not restored by the
+// gtest framework).
+class FileUtilsTest : public testing::Test {
+ protected:
+ FileUtilsTest() {
+ }
+ virtual ~FileUtilsTest() {}
+ // Runs before the first test
+ static void SetUpTestCase() {
+ original_working_dir_ = webrtc::test::WorkingDir();
+ std::string resources_path = original_working_dir_ + kPathDelimiter +
+ kResourcesDir + kPathDelimiter;
+ webrtc::test::CreateDirectory(resources_path);
+
+ files_.push_back(resources_path + kTestName + "." + kExtension);
+ files_.push_back(resources_path + kTestName + "_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win_32." + kExtension);
+ files_.push_back(resources_path + kTestName + "_linux_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_mac_64." + kExtension);
+ files_.push_back(resources_path + kTestName + "_win_64." + kExtension);
+
+ // Now that the resources dir exists, write some empty test files into it.
+ for (FileList::iterator file_it = files_.begin();
+ file_it != files_.end(); ++file_it) {
+ FILE* file = fopen(file_it->c_str(), "wb");
+ ASSERT_TRUE(file != NULL) << "Failed to write file: " << file_it->c_str();
+ ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0);
+ fclose(file);
+ }
+ // Create a dummy subdir that can be chdir'ed into for testing purposes.
+ empty_dummy_dir_ = original_working_dir_ + kPathDelimiter + kDummyDir;
+ webrtc::test::CreateDirectory(empty_dummy_dir_);
+ }
+ static void TearDownTestCase() {
+ // Clean up all resource files written
+ for (FileList::iterator file_it = files_.begin();
+ file_it != files_.end(); ++file_it) {
+ remove(file_it->c_str());
+ }
+ std::remove(empty_dummy_dir_.c_str());
+ }
+ void SetUp() {
+ ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
+ }
+ void TearDown() {
+ ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
+ }
+ protected:
+ static FileList files_;
+ static std::string empty_dummy_dir_;
+ private:
+ static std::string original_working_dir_;
+};
+
+FileList FileUtilsTest::files_;
+std::string FileUtilsTest::original_working_dir_ = "";
+std::string FileUtilsTest::empty_dummy_dir_ = "";
+
+// Tests that the project root path is returned for the default working
+// directory that is automatically set when the test executable is launched.
+// The test is not fully testing the implementation, since we cannot be sure
+// of where the executable was launched from.
+// The test will fail if the top level directory is not named "trunk".
+TEST_F(FileUtilsTest, ProjectRootPathFromUnchangedWorkingDir) {
+ std::string path = webrtc::test::ProjectRootPath();
+ std::string expected_end = "trunk";
+ expected_end = kPathDelimiter + expected_end + kPathDelimiter;
+ ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) {
+ std::string path = webrtc::test::OutputPath();
+ std::string expected_end = "out";
+ expected_end = kPathDelimiter + expected_end + kPathDelimiter;
+ ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
+}
+
+// Tests setting the current working directory to a directory three levels
+// deeper from the current one. Then testing that the project path returned
+// is still the same, when the function under test is called again.
+TEST_F(FileUtilsTest, ProjectRootPathFromDeeperWorkingDir) {
+ std::string path = webrtc::test::ProjectRootPath();
+ std::string original_working_dir = path; // This is the correct project root
+ // Change to a subdirectory path.
+ ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
+ ASSERT_EQ(original_working_dir, webrtc::test::ProjectRootPath());
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromDeeperWorkingDir) {
+ std::string path = webrtc::test::OutputPath();
+ std::string original_working_dir = path;
+ ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
+ ASSERT_EQ(original_working_dir, webrtc::test::OutputPath());
+}
+
+// Tests with current working directory set to a directory higher up in the
+// directory tree than the project root dir. This case shall return a specified
+// error string as a directory (which will be an invalid path).
+TEST_F(FileUtilsTest, ProjectRootPathFromRootWorkingDir) {
+ // Change current working dir to the root of the current file system
+ // (this will always be "above" our project root dir).
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ(webrtc::test::kCannotFindProjectRootDir,
+ webrtc::test::ProjectRootPath());
+}
+
+// Similar to the above test, but for the output dir
+TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) {
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ("./", webrtc::test::OutputPath());
+}
+
+// Only tests that the code executes
+TEST_F(FileUtilsTest, CreateDirectory) {
+ std::string directory = "fileutils-unittest-empty-dir";
+ // Make sure it's removed if a previous test has failed:
+ std::remove(directory.c_str());
+ ASSERT_TRUE(webrtc::test::CreateDirectory(directory));
+ std::remove(directory.c_str());
+}
+
+TEST_F(FileUtilsTest, WorkingDirReturnsValue) {
+ // Hard to cover all platforms. Just test that it returns something without
+ // crashing:
+ std::string working_dir = webrtc::test::WorkingDir();
+ ASSERT_GT(working_dir.length(), 0u);
+}
+
+// Due to multiple platforms, it is hard to make a complete test for
+// ResourcePath. Manual testing has been performed by removing files and
+// verified the result confirms with the specified documentation for the
+// function.
+TEST_F(FileUtilsTest, ResourcePathReturnsValue) {
+ std::string resource = webrtc::test::ResourcePath(kTestName, kExtension);
+ ASSERT_GT(resource.find(kTestName), 0u);
+ ASSERT_GT(resource.find(kExtension), 0u);
+ ASSERT_EQ(0, chdir(kPathDelimiter));
+ ASSERT_EQ("./", webrtc::test::OutputPath());
+}
+
+TEST_F(FileUtilsTest, GetFileSizeExistingFile) {
+ ASSERT_GT(webrtc::test::GetFileSize(files_.front()), 0u);
+}
+
+TEST_F(FileUtilsTest, GetFileSizeNonExistingFile) {
+ ASSERT_EQ(0u, webrtc::test::GetFileSize("non-existing-file.tmp"));
+}
+
+} // namespace webrtc
diff --git a/test/testsupport/frame_reader.cc b/test/testsupport/frame_reader.cc
new file mode 100644
index 0000000..b05ea58
--- /dev/null
+++ b/test/testsupport/frame_reader.cc
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_reader.h"
+
+#include <cassert>
+
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+FrameReaderImpl::FrameReaderImpl(std::string input_filename,
+ int frame_length_in_bytes)
+ : input_filename_(input_filename),
+ frame_length_in_bytes_(frame_length_in_bytes),
+ input_file_(NULL) {
+}
+
+FrameReaderImpl::~FrameReaderImpl() {
+ Close();
+}
+
+bool FrameReaderImpl::Init() {
+ if (frame_length_in_bytes_ <= 0) {
+ fprintf(stderr, "Frame length must be >0, was %d\n",
+ frame_length_in_bytes_);
+ return false;
+ }
+ input_file_ = fopen(input_filename_.c_str(), "rb");
+ if (input_file_ == NULL) {
+ fprintf(stderr, "Couldn't open input file for reading: %s\n",
+ input_filename_.c_str());
+ return false;
+ }
+ // Calculate total number of frames.
+ size_t source_file_size = GetFileSize(input_filename_);
+ if (source_file_size <= 0u) {
+ fprintf(stderr, "Found empty file: %s\n", input_filename_.c_str());
+ return false;
+ }
+ number_of_frames_ = source_file_size / frame_length_in_bytes_;
+ return true;
+}
+
+void FrameReaderImpl::Close() {
+ if (input_file_ != NULL) {
+ fclose(input_file_);
+ input_file_ = NULL;
+ }
+}
+
+bool FrameReaderImpl::ReadFrame(WebRtc_UWord8* source_buffer) {
+ assert(source_buffer);
+ if (input_file_ == NULL) {
+ fprintf(stderr, "FrameReader is not initialized (input file is NULL)\n");
+ return false;
+ }
+ size_t nbr_read = fread(source_buffer, 1, frame_length_in_bytes_,
+ input_file_);
+ if (nbr_read != static_cast<unsigned int>(frame_length_in_bytes_) &&
+ ferror(input_file_)) {
+ fprintf(stderr, "Error reading from input file: %s\n",
+ input_filename_.c_str());
+ return false;
+ }
+ if (feof(input_file_) != 0) {
+ return false; // No more frames to process.
+ }
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/frame_reader.h b/test/testsupport/frame_reader.h
new file mode 100644
index 0000000..56d8fc4
--- /dev/null
+++ b/test/testsupport/frame_reader.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
+
+#include <cstdio>
+#include <string>
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles reading of frames from video files.
+class FrameReader {
+ public:
+ virtual ~FrameReader() {}
+
+ // Initializes the frame reader, i.e. opens the input file.
+ // This must be called before reading of frames has started.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Reads a frame into the supplied buffer, which must contain enough space
+ // for the frame size.
+ // Returns true if there are more frames to read, false if we've already
+ // read the last frame (in the previous call).
+ virtual bool ReadFrame(WebRtc_UWord8* source_buffer) = 0;
+
+ // Closes the input file if open. Essentially makes this class impossible
+ // to use anymore. Will also be invoked by the destructor.
+ virtual void Close() = 0;
+
+ // Frame length in bytes of a single frame image.
+ virtual int FrameLength() = 0;
+ // Total number of frames in the input video source.
+ virtual int NumberOfFrames() = 0;
+};
+
+class FrameReaderImpl : public FrameReader {
+ public:
+ // Creates a file handler. The input file is assumed to exist and be readable.
+ // Parameters:
+ // input_filename The file to read from.
+ // frame_length_in_bytes The size of each frame.
+ // For YUV this is 3 * width * height / 2
+ FrameReaderImpl(std::string input_filename, int frame_length_in_bytes);
+ virtual ~FrameReaderImpl();
+ bool Init();
+ bool ReadFrame(WebRtc_UWord8* source_buffer);
+ void Close();
+ int FrameLength() { return frame_length_in_bytes_; }
+ int NumberOfFrames() { return number_of_frames_; }
+
+ private:
+ std::string input_filename_;
+ int frame_length_in_bytes_;
+ int number_of_frames_;
+ FILE* input_file_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
diff --git a/test/testsupport/frame_reader_unittest.cc b/test/testsupport/frame_reader_unittest.cc
new file mode 100644
index 0000000..f1da5ce
--- /dev/null
+++ b/test/testsupport/frame_reader_unittest.cc
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_reader.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+const std::string kInputFilename = "temp_inputfile.tmp";
+const std::string kInputFileContents = "baz";
+// Setting the kFrameLength value to a value much larger than the
+// file to test causes the ReadFrame test to fail on Windows.
+const int kFrameLength = 1000;
+
+class FrameReaderTest: public testing::Test {
+ protected:
+ FrameReaderTest() {}
+ virtual ~FrameReaderTest() {}
+ void SetUp() {
+ // Cleanup any previous dummy input file.
+ std::remove(kInputFilename.c_str());
+
+ // Create a dummy input file.
+ FILE* dummy = fopen(kInputFilename.c_str(), "wb");
+ fprintf(dummy, "%s", kInputFileContents.c_str());
+ fclose(dummy);
+
+ frame_reader_ = new FrameReaderImpl(kInputFilename, kFrameLength);
+ ASSERT_TRUE(frame_reader_->Init());
+ }
+ void TearDown() {
+ delete frame_reader_;
+ // Cleanup the dummy input file.
+ std::remove(kInputFilename.c_str());
+ }
+ FrameReader* frame_reader_;
+};
+
+TEST_F(FrameReaderTest, InitSuccess) {
+ FrameReaderImpl frame_reader(kInputFilename, kFrameLength);
+ ASSERT_TRUE(frame_reader.Init());
+ ASSERT_EQ(kFrameLength, frame_reader.FrameLength());
+ ASSERT_EQ(0, frame_reader.NumberOfFrames());
+}
+
+TEST_F(FrameReaderTest, ReadFrame) {
+ WebRtc_UWord8 buffer[3];
+ bool result = frame_reader_->ReadFrame(buffer);
+ ASSERT_FALSE(result); // No more files to read.
+ ASSERT_EQ(kInputFileContents[0], buffer[0]);
+ ASSERT_EQ(kInputFileContents[1], buffer[1]);
+ ASSERT_EQ(kInputFileContents[2], buffer[2]);
+}
+
+TEST_F(FrameReaderTest, ReadFrameUninitialized) {
+ WebRtc_UWord8 buffer[3];
+ FrameReaderImpl file_reader(kInputFilename, kFrameLength);
+ ASSERT_FALSE(file_reader.ReadFrame(buffer));
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/frame_writer.cc b/test/testsupport/frame_writer.cc
new file mode 100644
index 0000000..5f32539
--- /dev/null
+++ b/test/testsupport/frame_writer.cc
@@ -0,0 +1,68 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_writer.h"
+
+#include <cassert>
+
+namespace webrtc {
+namespace test {
+
+FrameWriterImpl::FrameWriterImpl(std::string output_filename,
+ int frame_length_in_bytes)
+ : output_filename_(output_filename),
+ frame_length_in_bytes_(frame_length_in_bytes),
+ output_file_(NULL) {
+}
+
+FrameWriterImpl::~FrameWriterImpl() {
+ Close();
+}
+
+bool FrameWriterImpl::Init() {
+ if (frame_length_in_bytes_ <= 0) {
+ fprintf(stderr, "Frame length must be >0, was %d\n",
+ frame_length_in_bytes_);
+ return false;
+ }
+ output_file_ = fopen(output_filename_.c_str(), "wb");
+ if (output_file_ == NULL) {
+ fprintf(stderr, "Couldn't open output file for writing: %s\n",
+ output_filename_.c_str());
+ return false;
+ }
+ return true;
+}
+
+void FrameWriterImpl::Close() {
+ if (output_file_ != NULL) {
+ fclose(output_file_);
+ output_file_ = NULL;
+ }
+}
+
+bool FrameWriterImpl::WriteFrame(WebRtc_UWord8* frame_buffer) {
+ assert(frame_buffer);
+ if (output_file_ == NULL) {
+ fprintf(stderr, "FrameWriter is not initialized (output file is NULL)\n");
+ return false;
+ }
+ int bytes_written = fwrite(frame_buffer, 1, frame_length_in_bytes_,
+ output_file_);
+ if (bytes_written != frame_length_in_bytes_) {
+ fprintf(stderr, "Failed to write %d bytes to file %s\n",
+ frame_length_in_bytes_, output_filename_.c_str());
+ return false;
+ }
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/frame_writer.h b/test/testsupport/frame_writer.h
new file mode 100644
index 0000000..abc5d35
--- /dev/null
+++ b/test/testsupport/frame_writer.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
+#define WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
+
+#include <cstdio>
+#include <string>
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles writing of video files.
+class FrameWriter {
+ public:
+ virtual ~FrameWriter() {}
+
+ // Initializes the file handler, i.e. opens the input and output files etc.
+ // This must be called before reading or writing frames has started.
+ // Returns false if an error has occurred, in addition to printing to stderr.
+ virtual bool Init() = 0;
+
+ // Writes a frame of the configured frame length to the output file.
+ // Returns true if the write was successful, false otherwise.
+ virtual bool WriteFrame(WebRtc_UWord8* frame_buffer) = 0;
+
+ // Closes the output file if open. Essentially makes this class impossible
+ // to use anymore. Will also be invoked by the destructor.
+ virtual void Close() = 0;
+
+ // Frame length in bytes of a single frame image.
+ virtual int FrameLength() = 0;
+};
+
+class FrameWriterImpl : public FrameWriter {
+ public:
+ // Creates a file handler. The input file is assumed to exist and be readable
+ // and the output file must be writable.
+ // Parameters:
+ // output_filename The file to write. Will be overwritten if already
+ // existing.
+ // frame_length_in_bytes The size of each frame.
+ // For YUV: 3*width*height/2
+ FrameWriterImpl(std::string output_filename, int frame_length_in_bytes);
+ virtual ~FrameWriterImpl();
+ bool Init();
+ bool WriteFrame(WebRtc_UWord8* frame_buffer);
+ void Close();
+ int FrameLength() { return frame_length_in_bytes_; }
+
+ private:
+ std::string output_filename_;
+ int frame_length_in_bytes_;
+ int number_of_frames_;
+ FILE* output_file_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
diff --git a/test/testsupport/frame_writer_unittest.cc b/test/testsupport/frame_writer_unittest.cc
new file mode 100644
index 0000000..d25d1d2
--- /dev/null
+++ b/test/testsupport/frame_writer_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/frame_writer.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+const std::string kOutputFilename = "temp_outputfile.tmp";
+const int kFrameLength = 1000;
+
+class FrameWriterTest: public testing::Test {
+ protected:
+ FrameWriterTest() {}
+ virtual ~FrameWriterTest() {}
+ void SetUp() {
+ // Cleanup any previous output file.
+ std::remove(kOutputFilename.c_str());
+ frame_writer_ = new FrameWriterImpl(kOutputFilename, kFrameLength);
+ ASSERT_TRUE(frame_writer_->Init());
+ }
+ void TearDown() {
+ delete frame_writer_;
+ // Cleanup the temporary file.
+ std::remove(kOutputFilename.c_str());
+ }
+ FrameWriter* frame_writer_;
+};
+
+TEST_F(FrameWriterTest, InitSuccess) {
+ FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
+ ASSERT_TRUE(frame_writer.Init());
+ ASSERT_EQ(kFrameLength, frame_writer.FrameLength());
+}
+
+TEST_F(FrameWriterTest, WriteFrame) {
+ WebRtc_UWord8 buffer[kFrameLength];
+ memset(buffer, 9, kFrameLength); // Write lots of 9s to the buffer
+ bool result = frame_writer_->WriteFrame(buffer);
+ ASSERT_TRUE(result); // success
+ // Close the file and verify the size.
+ frame_writer_->Close();
+ ASSERT_EQ(kFrameLength,
+ static_cast<int>(GetFileSize(kOutputFilename)));
+}
+
+TEST_F(FrameWriterTest, WriteFrameUninitialized) {
+ WebRtc_UWord8 buffer[3];
+ FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
+ ASSERT_FALSE(frame_writer.WriteFrame(buffer));
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/metrics/video_metrics.cc b/test/testsupport/metrics/video_metrics.cc
new file mode 100644
index 0000000..9e61ec8
--- /dev/null
+++ b/test/testsupport/metrics/video_metrics.cc
@@ -0,0 +1,187 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/metrics/video_metrics.h"
+
+#include <algorithm> // min_element, max_element
+#include <cassert>
+#include <cstdio>
+
+#include "common_video/libyuv/include/libyuv.h"
+
+namespace webrtc {
+namespace test {
+
+// Used for calculating min and max values
+static bool LessForFrameResultValue (const FrameResult& s1,
+ const FrameResult& s2) {
+ return s1.value < s2.value;
+}
+
+enum VideoMetricsType { kPSNR, kSSIM, kBoth };
+
+// Calculates metrics for a frame and adds statistics to the result for it.
+void CalculateFrame(VideoMetricsType video_metrics_type,
+ uint8_t* ref,
+ uint8_t* test,
+ int width,
+ int height,
+ int frame_number,
+ QualityMetricsResult* result) {
+ FrameResult frame_result;
+ frame_result.frame_number = frame_number;
+ switch (video_metrics_type) {
+ case kPSNR:
+ frame_result.value = I420PSNR(ref, test, width, height);
+ break;
+ case kSSIM:
+ frame_result.value = I420SSIM(ref, test, width, height);
+ break;
+ default:
+ assert(false);
+ }
+ result->frames.push_back(frame_result);
+}
+
+// Calculates average, min and max values for the supplied struct, if non-NULL.
+void CalculateStats(QualityMetricsResult* result) {
+ if (result == NULL || result->frames.size() == 0) {
+ return;
+ }
+ // Calculate average
+ std::vector<FrameResult>::iterator iter;
+ double metrics_values_sum = 0.0;
+ for (iter = result->frames.begin(); iter != result->frames.end(); ++iter) {
+ metrics_values_sum += iter->value;
+ }
+ result->average = metrics_values_sum / result->frames.size();
+
+ // Calculate min/max statistics
+ iter = min_element(result->frames.begin(), result->frames.end(),
+ LessForFrameResultValue);
+ result->min = iter->value;
+ result->min_frame_number = iter->frame_number;
+ iter = max_element(result->frames.begin(), result->frames.end(),
+ LessForFrameResultValue);
+ result->max = iter->value;
+ result->max_frame_number = iter->frame_number;
+}
+
+// Single method that handles all combinations of video metrics calculation, to
+// minimize code duplication. Either psnr_result or ssim_result may be NULL,
+// depending on which VideoMetricsType is targeted.
+int CalculateMetrics(VideoMetricsType video_metrics_type,
+ const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result) {
+ assert(ref_filename != NULL);
+ assert(test_filename != NULL);
+ assert(width > 0);
+ assert(height > 0);
+
+ FILE* ref_fp = fopen(ref_filename, "rb");
+ if (ref_fp == NULL) {
+ // cannot open reference file
+ fprintf(stderr, "Cannot open file %s\n", ref_filename);
+ return -1;
+ }
+ FILE* test_fp = fopen(test_filename, "rb");
+ if (test_fp == NULL) {
+ // cannot open test file
+ fprintf(stderr, "Cannot open file %s\n", test_filename);
+ fclose(ref_fp);
+ return -2;
+ }
+ int frame_number = 0;
+
+ // Allocating size for one I420 frame.
+ const int frame_length = 3 * width * height >> 1;
+ uint8_t* ref = new uint8_t[frame_length];
+ uint8_t* test = new uint8_t[frame_length];
+
+ int ref_bytes = fread(ref, 1, frame_length, ref_fp);
+ int test_bytes = fread(test, 1, frame_length, test_fp);
+ while (ref_bytes == frame_length && test_bytes == frame_length) {
+ switch (video_metrics_type) {
+ case kPSNR:
+ CalculateFrame(kPSNR, ref, test, width, height, frame_number,
+ psnr_result);
+ break;
+ case kSSIM:
+ CalculateFrame(kSSIM, ref, test, width, height, frame_number,
+ ssim_result);
+ break;
+ case kBoth:
+ CalculateFrame(kPSNR, ref, test, width, height, frame_number,
+ psnr_result);
+ CalculateFrame(kSSIM, ref, test, width, height, frame_number,
+ ssim_result);
+ break;
+ default:
+ assert(false);
+ }
+ frame_number++;
+ ref_bytes = fread(ref, 1, frame_length, ref_fp);
+ test_bytes = fread(test, 1, frame_length, test_fp);
+ }
+ int return_code = 0;
+ if (frame_number == 0) {
+ fprintf(stderr, "Tried to measure video metrics from empty files "
+ "(reference file: %s test file: %s)\n", ref_filename,
+ test_filename);
+ return_code = -3;
+ } else {
+ CalculateStats(psnr_result);
+ CalculateStats(ssim_result);
+ }
+ delete [] ref;
+ delete [] test;
+ fclose(ref_fp);
+ fclose(test_fp);
+ return return_code;
+}
+
+int I420MetricsFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result) {
+ assert(psnr_result != NULL);
+ assert(ssim_result != NULL);
+ return CalculateMetrics(kBoth, ref_filename, test_filename, width, height,
+ psnr_result, ssim_result);
+}
+
+int I420PSNRFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result) {
+ assert(result != NULL);
+ return CalculateMetrics(kPSNR, ref_filename, test_filename, width, height,
+ result, NULL);
+}
+
+int I420SSIMFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result) {
+ assert(result != NULL);
+ return CalculateMetrics(kSSIM, ref_filename, test_filename, width, height,
+ NULL, result);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/metrics/video_metrics.h b/test/testsupport/metrics/video_metrics.h
new file mode 100644
index 0000000..df11a49
--- /dev/null
+++ b/test/testsupport/metrics/video_metrics.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
+#define WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
+
+#include <limits>
+#include <vector>
+
+namespace webrtc {
+namespace test {
+
+// Contains video quality metrics result for a single frame.
+struct FrameResult {
+ int frame_number;
+ double value;
+};
+
+// Result from a PSNR/SSIM calculation operation.
+// The frames in this data structure are 0-indexed.
+struct QualityMetricsResult {
+ QualityMetricsResult() :
+ average(0.0),
+ min(std::numeric_limits<double>::max()),
+ max(std::numeric_limits<double>::min()),
+ min_frame_number(-1),
+ max_frame_number(-1)
+ {};
+ double average;
+ double min;
+ double max;
+ int min_frame_number;
+ int max_frame_number;
+ std::vector<FrameResult> frames;
+};
+
+// Calculates PSNR and SSIM values for the reference and test video files
+// (must be in I420 format). All calculated values are filled into the
+// QualityMetricsResult stucts.
+// PSNR values have the unit decibel (dB) where a high value means the test file
+// is similar to the reference file. The higher value, the more similar.
+// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
+// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
+// identical. For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420MetricsFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* psnr_result,
+ QualityMetricsResult* ssim_result);
+
+// Calculates PSNR values for the reference and test video files (must be in
+// I420 format). All calculated values are filled into the QualityMetricsResult
+// struct.
+// PSNR values have the unit decibel (dB) where a high value means the test file
+// is similar to the reference file. The higher value, the more similar.
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
+//
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420PSNRFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result);
+
+// Calculates SSIM values for the reference and test video files (must be in
+// I420 format). All calculated values are filled into the QualityMetricsResult
+// struct.
+// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
+// identical.
+// This function only compares video frames up to the point when the shortest
+// video ends.
+// For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
+//
+// Return value:
+// 0 if successful, negative on errors:
+// -1 if the source file cannot be opened
+// -2 if the test file cannot be opened
+// -3 if any of the files are empty
+// -4 if any arguments are invalid.
+int I420SSIMFromFiles(const char* ref_filename,
+ const char* test_filename,
+ int width,
+ int height,
+ QualityMetricsResult* result);
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
diff --git a/test/testsupport/metrics/video_metrics_unittest.cc b/test/testsupport/metrics/video_metrics_unittest.cc
new file mode 100644
index 0000000..e77dbff
--- /dev/null
+++ b/test/testsupport/metrics/video_metrics_unittest.cc
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/metrics/video_metrics.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/fileutils.h"
+
+namespace webrtc {
+
+static const char* kEmptyFileName = "video_metrics_unittest_empty_file.tmp";
+static const char* kNonExistingFileName = "video_metrics_unittest_non_existing";
+static const int kWidth = 352;
+static const int kHeight = 288;
+
+static const int kMissingReferenceFileReturnCode = -1;
+static const int kMissingTestFileReturnCode = -2;
+static const int kEmptyFileReturnCode = -3;
+static const double kPsnrPerfectResult = 48.0;
+static const double kSsimPerfectResult = 1.0;
+
+class VideoMetricsTest: public testing::Test {
+ protected:
+ VideoMetricsTest() {
+ video_file_ = webrtc::test::ResourcePath("foreman_cif_short", "yuv");
+ }
+ virtual ~VideoMetricsTest() {}
+ void SetUp() {
+ // Create an empty file:
+ FILE* dummy = fopen(kEmptyFileName, "wb");
+ fclose(dummy);
+ }
+ void TearDown() {
+ std::remove(kEmptyFileName);
+ }
+ webrtc::test::QualityMetricsResult psnr_result_;
+ webrtc::test::QualityMetricsResult ssim_result_;
+ std::string video_file_;
+};
+
+// Tests that it is possible to run with the same reference as test file
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesPSNR) {
+ EXPECT_EQ(0, I420PSNRFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &psnr_result_));
+ EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
+}
+
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesSSIM) {
+ EXPECT_EQ(0, I420SSIMFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
+}
+
+TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesBothMetrics) {
+ EXPECT_EQ(0, I420MetricsFromFiles(video_file_.c_str(), video_file_.c_str(),
+ kWidth, kHeight, &psnr_result_,
+ &ssim_result_));
+ EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
+ EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
+}
+
+// Tests that the right return code is given when the reference file is missing.
+TEST_F(VideoMetricsTest, MissingReferenceFilePSNR) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420PSNRFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingReferenceFileSSIM) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420SSIMFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingReferenceFileBothMetrics) {
+ EXPECT_EQ(kMissingReferenceFileReturnCode,
+ I420MetricsFromFiles(kNonExistingFileName, video_file_.c_str(),
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+// Tests that the right return code is given when the test file is missing.
+TEST_F(VideoMetricsTest, MissingTestFilePSNR) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420PSNRFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingTestFileSSIM) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420SSIMFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, MissingTestFileBothMetrics) {
+ EXPECT_EQ(kMissingTestFileReturnCode,
+ I420MetricsFromFiles(video_file_.c_str(), kNonExistingFileName,
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+// Tests that the method can be executed with empty files.
+TEST_F(VideoMetricsTest, EmptyFilesPSNR) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420PSNRFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420PSNRFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, EmptyFilesSSIM) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420SSIMFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420SSIMFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight, &ssim_result_));
+}
+
+TEST_F(VideoMetricsTest, EmptyFilesBothMetrics) {
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420MetricsFromFiles(kEmptyFileName, video_file_.c_str(),
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+ EXPECT_EQ(kEmptyFileReturnCode,
+ I420MetricsFromFiles(video_file_.c_str(), kEmptyFileName,
+ kWidth, kHeight,
+ &psnr_result_, &ssim_result_));
+}
+
+} // namespace webrtc
diff --git a/test/testsupport/mock/mock_frame_reader.h b/test/testsupport/mock/mock_frame_reader.h
new file mode 100644
index 0000000..ecfc13c
--- /dev/null
+++ b/test/testsupport/mock/mock_frame_reader.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+
+#include "testsupport/frame_reader.h"
+
+#include "gmock/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockFrameReader : public FrameReader {
+ public:
+ MOCK_METHOD0(Init, bool());
+ MOCK_METHOD1(ReadFrame, bool(WebRtc_UWord8* source_buffer));
+ MOCK_METHOD0(Close, void());
+ MOCK_METHOD0(FrameLength, int());
+ MOCK_METHOD0(NumberOfFrames, int());
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
diff --git a/test/testsupport/mock/mock_frame_writer.h b/test/testsupport/mock/mock_frame_writer.h
new file mode 100644
index 0000000..ba79184
--- /dev/null
+++ b/test/testsupport/mock/mock_frame_writer.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
+#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
+
+#include "testsupport/frame_writer.h"
+
+#include "gmock/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockFrameWriter : public FrameWriter {
+ public:
+ MOCK_METHOD0(Init, bool());
+ MOCK_METHOD1(WriteFrame, bool(WebRtc_UWord8* frame_buffer));
+ MOCK_METHOD0(Close, void());
+ MOCK_METHOD0(FrameLength, int());
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
diff --git a/test/testsupport/packet_reader.cc b/test/testsupport/packet_reader.cc
new file mode 100644
index 0000000..e8859d1
--- /dev/null
+++ b/test/testsupport/packet_reader.cc
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/packet_reader.h"
+
+#include <cassert>
+#include <cstdio>
+
+namespace webrtc {
+namespace test {
+
+PacketReader::PacketReader()
+ : initialized_(false) {}
+
+PacketReader::~PacketReader() {}
+
+void PacketReader::InitializeReading(WebRtc_UWord8* data,
+ int data_length_in_bytes,
+ int packet_size_in_bytes) {
+ assert(data);
+ assert(data_length_in_bytes >= 0);
+ assert(packet_size_in_bytes > 0);
+ data_ = data;
+ data_length_ = data_length_in_bytes;
+ packet_size_ = packet_size_in_bytes;
+ currentIndex_ = 0;
+ initialized_ = true;
+}
+
+int PacketReader::NextPacket(WebRtc_UWord8** packet_pointer) {
+ if (!initialized_) {
+ fprintf(stderr, "Attempting to use uninitialized PacketReader!\n");
+ return -1;
+ }
+ *packet_pointer = data_ + currentIndex_;
+ // Check if we're about to read the last packet:
+ if (data_length_ - currentIndex_ <= packet_size_) {
+ int size = data_length_ - currentIndex_;
+ currentIndex_ = data_length_;
+ assert(size >= 0);
+ return size;
+ }
+ currentIndex_ += packet_size_;
+ assert(packet_size_ >= 0);
+ return packet_size_;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/packet_reader.h b/test/testsupport/packet_reader.h
new file mode 100644
index 0000000..4cb0bb1
--- /dev/null
+++ b/test/testsupport/packet_reader.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
+#define WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Reads chunks of data to simulate network packets from a byte array.
+class PacketReader {
+ public:
+ PacketReader();
+ virtual ~PacketReader();
+
+ // Inizializes a new reading operation. Must be done before invoking the
+ // NextPacket method.
+ // * data_length_in_bytes is the length of the data byte array. Must be >= 0.
+ // 0 length will result in no packets are read.
+ // * packet_size_in_bytes is the number of bytes to read in each NextPacket
+ // method call. Must be > 0
+ virtual void InitializeReading(WebRtc_UWord8* data, int data_length_in_bytes,
+ int packet_size_in_bytes);
+
+ // Moves the supplied pointer to the beginning of the next packet.
+ // Returns:
+ // * The size of the packet ready to read (lower than the packet size for
+ // the last packet)
+ // * 0 if there are no more packets to read
+ // * -1 if InitializeReading has not been called (also prints to stderr).
+ virtual int NextPacket(WebRtc_UWord8** packet_pointer);
+
+ private:
+ WebRtc_UWord8* data_;
+ int data_length_;
+ int packet_size_;
+ int currentIndex_;
+ bool initialized_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
diff --git a/test/testsupport/packet_reader_unittest.cc b/test/testsupport/packet_reader_unittest.cc
new file mode 100644
index 0000000..6719e4c
--- /dev/null
+++ b/test/testsupport/packet_reader_unittest.cc
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testsupport/packet_reader.h"
+
+#include "gtest/gtest.h"
+#include "testsupport/unittest_utils.h"
+
+namespace webrtc {
+namespace test {
+
+class PacketReaderTest: public PacketRelatedTest {
+ protected:
+ PacketReaderTest() {}
+ virtual ~PacketReaderTest() {}
+ void SetUp() {
+ reader_ = new PacketReader();
+ }
+ void TearDown() {
+ delete reader_;
+ }
+ void VerifyPacketData(int expected_length,
+ int actual_length,
+ WebRtc_UWord8* original_data_pointer,
+ WebRtc_UWord8* new_data_pointer) {
+ EXPECT_EQ(expected_length, actual_length);
+ EXPECT_EQ(*original_data_pointer, *new_data_pointer);
+ EXPECT_EQ(0, memcmp(original_data_pointer, new_data_pointer,
+ actual_length));
+ }
+ PacketReader* reader_;
+};
+
+// Test lack of initialization
+TEST_F(PacketReaderTest, Uninitialized) {
+ WebRtc_UWord8* data_pointer = NULL;
+ EXPECT_EQ(-1, reader_->NextPacket(&data_pointer));
+ EXPECT_EQ(NULL, data_pointer);
+}
+
+TEST_F(PacketReaderTest, InitializeZeroLengthArgument) {
+ reader_->InitializeReading(packet_data_, 0, kPacketSizeInBytes);
+ ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_));
+}
+
+// Test with something smaller than one packet
+TEST_F(PacketReaderTest, NormalSmallData) {
+ const int kDataLengthInBytes = 1499;
+ WebRtc_UWord8 data[kDataLengthInBytes];
+ WebRtc_UWord8* data_pointer = data;
+ memset(data, 1, kDataLengthInBytes);
+
+ reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
+ int length_to_read = reader_->NextPacket(&data_pointer);
+ VerifyPacketData(kDataLengthInBytes, length_to_read, data, data_pointer);
+ EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&data_pointer);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kDataLengthInBytes, data_pointer - data);
+}
+
+// Test with data length that exactly matches one packet
+TEST_F(PacketReaderTest, NormalOnePacketData) {
+ WebRtc_UWord8 data[kPacketSizeInBytes];
+ WebRtc_UWord8* data_pointer = data;
+ memset(data, 1, kPacketSizeInBytes);
+
+ reader_->InitializeReading(data, kPacketSizeInBytes, kPacketSizeInBytes);
+ int length_to_read = reader_->NextPacket(&data_pointer);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read, data, data_pointer);
+ EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&data_pointer);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kPacketSizeInBytes, data_pointer - data);
+}
+
+// Test with data length that will result in 3 packets
+TEST_F(PacketReaderTest, NormalLargeData) {
+ reader_->InitializeReading(packet_data_, kPacketDataLength,
+ kPacketSizeInBytes);
+
+ int length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read,
+ packet1_, packet_data_pointer_);
+
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(kPacketSizeInBytes, length_to_read,
+ packet2_, packet_data_pointer_);
+
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ VerifyPacketData(1u, length_to_read,
+ packet3_, packet_data_pointer_);
+
+ // Reading another one shall result in 0 bytes:
+ length_to_read = reader_->NextPacket(&packet_data_pointer_);
+ EXPECT_EQ(0, length_to_read);
+ EXPECT_EQ(kPacketDataLength, packet_data_pointer_ - packet_data_);
+}
+
+// Test with empty data.
+TEST_F(PacketReaderTest, EmptyData) {
+ const int kDataLengthInBytes = 0;
+ WebRtc_UWord8* data = new WebRtc_UWord8[kDataLengthInBytes];
+ reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
+ EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
+ // Do it again to make sure nothing changes
+ EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
+ delete[] data;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/testsupport/unittest_utils.h b/test/testsupport/unittest_utils.h
new file mode 100644
index 0000000..963a5d3
--- /dev/null
+++ b/test/testsupport/unittest_utils.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
+#define WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
+
+namespace webrtc {
+namespace test {
+
+const int kPacketSizeInBytes = 1500;
+const int kPacketDataLength = kPacketSizeInBytes * 2 + 1;
+const int kPacketDataNumberOfPackets = 3;
+
+// A base test fixture for packet related tests. Contains
+// two full prepared packets with 1s, 2s in their data and a third packet with
+// a single 3 in it (size=1).
+// A packet data structure is also available, that contains these three packets
+// in order.
+class PacketRelatedTest: public testing::Test {
+ protected:
+ // Tree packet byte arrays with data used for verification:
+ WebRtc_UWord8 packet1_[kPacketSizeInBytes];
+ WebRtc_UWord8 packet2_[kPacketSizeInBytes];
+ WebRtc_UWord8 packet3_[1];
+ // Construct a data structure containing these packets
+ WebRtc_UWord8 packet_data_[kPacketDataLength];
+ WebRtc_UWord8* packet_data_pointer_;
+
+ PacketRelatedTest() {
+ packet_data_pointer_ = packet_data_;
+
+ memset(packet1_, 1, kPacketSizeInBytes);
+ memset(packet2_, 2, kPacketSizeInBytes);
+ memset(packet3_, 3, 1);
+ // Fill the packet_data:
+ memcpy(packet_data_pointer_, packet1_, kPacketSizeInBytes);
+ memcpy(packet_data_pointer_ + kPacketSizeInBytes, packet2_,
+ kPacketSizeInBytes);
+ memcpy(packet_data_pointer_ + kPacketSizeInBytes * 2, packet3_, 1);
+ }
+ virtual ~PacketRelatedTest() {}
+ void SetUp() {
+ // Initialize the random generator with 0 to get deterministic behavior
+ srand(0);
+ }
+ void TearDown() {}
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
