Re-land "Remove <(webrtc_root) from source file entries."
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.org
TBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index c8386f6..320b07a 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -127,7 +127,7 @@
# Disallow referencing source files with paths above the GYP file location.
source_pattern = input_api.re.compile(r'sources.*?\[(.*?)\]',
re.MULTILINE | re.DOTALL)
- file_pattern = input_api.re.compile(r"'(\.\./.*?)'")
+ file_pattern = input_api.re.compile(r"'((\.\./.*?)|(<\(webrtc_root\).*?))'")
violating_gyp_files = set()
violating_source_entries = []
for gyp_file in gyp_files:
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate b/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate
similarity index 100%
rename from webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate
rename to webrtc/modules/audio_coding/audio_codec_speed_tests.isolate
diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi
new file mode 100644
index 0000000..0a500d9
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_coding.gypi
@@ -0,0 +1,29 @@
+# Copyright (c) 2015 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',
+ 'codecs/interfaces.gypi',
+ 'codecs/cng/cng.gypi',
+ 'codecs/g711/g711.gypi',
+ 'codecs/g722/g722.gypi',
+ 'codecs/ilbc/ilbc.gypi',
+ 'codecs/isac/main/source/isac.gypi',
+ 'codecs/isac/fix/source/isacfix.gypi',
+ 'codecs/pcm16b/pcm16b.gypi',
+ 'codecs/red/red.gypi',
+ 'main/acm2/audio_coding_module.gypi',
+ 'neteq/neteq.gypi',
+ ],
+ 'conditions': [
+ ['include_opus==1', {
+ 'includes': ['codecs/opus/opus.gypi',],
+ }],
+ ],
+}
diff --git a/webrtc/modules/audio_coding/audio_coding_tests.gypi b/webrtc/modules/audio_coding/audio_coding_tests.gypi
new file mode 100644
index 0000000..86a92c5
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_coding_tests.gypi
@@ -0,0 +1,72 @@
+# Copyright (c) 2015 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',
+ 'codecs/isac/isac_test.gypi',
+ 'codecs/isac/isacfix_test.gypi',
+ ],
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests',
+ 'type': '<(gtest_target_type)',
+ 'dependencies': [
+ 'audio_processing',
+ 'iSACFix',
+ 'webrtc_opus',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'codecs/isac/fix/test/isac_speed_test.cc',
+ 'codecs/opus/opus_speed_test.cc',
+ 'codecs/tools/audio_codec_speed_test.h',
+ 'codecs/tools/audio_codec_speed_test.cc',
+ ],
+ 'conditions': [
+ ['OS=="android"', {
+ 'dependencies': [
+ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+ ],
+ }],
+ ],
+ },
+ ],
+ 'conditions': [
+ ['OS=="android"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests_apk_target',
+ 'type': 'none',
+ 'dependencies': [
+ '<(apk_tests_path):audio_codec_speed_tests_apk',
+ ],
+ },
+ ],
+ }],
+ ['test_isolation_mode != "noop"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests_run',
+ 'type': 'none',
+ 'dependencies': [
+ 'audio_codec_speed_tests',
+ ],
+ 'includes': [
+ '../../build/isolate.gypi',
+ ],
+ 'sources': [
+ 'audio_codec_speed_tests.isolate',
+ ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi
deleted file mode 100644
index e296634..0000000
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi
+++ /dev/null
@@ -1,66 +0,0 @@
-# Copyright (c) 2014 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_codec_speed_tests',
- 'type': '<(gtest_target_type)',
- 'dependencies': [
- 'audio_processing',
- 'iSACFix',
- 'webrtc_opus',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'audio_codec_speed_test.h',
- 'audio_codec_speed_test.cc',
- '<(webrtc_root)/modules/audio_coding/codecs/opus/opus_speed_test.cc',
- '<(webrtc_root)/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc',
- ],
- 'conditions': [
- ['OS=="android"', {
- 'dependencies': [
- '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
- ],
- }],
- ],
- }],
- 'conditions': [
- ['OS=="android"', {
- 'targets': [
- {
- 'target_name': 'audio_codec_speed_tests_apk_target',
- 'type': 'none',
- 'dependencies': [
- '<(apk_tests_path):audio_codec_speed_tests_apk',
- ],
- },
- ],
- }],
- ['test_isolation_mode != "noop"', {
- 'targets': [
- {
- 'target_name': 'audio_codec_speed_tests_run',
- 'type': 'none',
- 'dependencies': [
- 'audio_codec_speed_tests',
- ],
- 'includes': [
- '../../../../build/isolate.gypi',
- ],
- 'sources': [
- 'audio_codec_speed_tests.isolate',
- ],
- },
- ],
- }],
- ],
-}
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 691f308..23ee29f 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -9,17 +9,7 @@
{
'includes': [
'../build/common.gypi',
- 'audio_coding/codecs/interfaces.gypi',
- 'audio_coding/codecs/cng/cng.gypi',
- 'audio_coding/codecs/g711/g711.gypi',
- 'audio_coding/codecs/g722/g722.gypi',
- 'audio_coding/codecs/ilbc/ilbc.gypi',
- 'audio_coding/codecs/isac/main/source/isac.gypi',
- 'audio_coding/codecs/isac/fix/source/isacfix.gypi',
- 'audio_coding/codecs/pcm16b/pcm16b.gypi',
- 'audio_coding/codecs/red/red.gypi',
- 'audio_coding/main/acm2/audio_coding_module.gypi',
- 'audio_coding/neteq/neteq.gypi',
+ 'audio_coding/audio_coding.gypi',
'audio_conference_mixer/source/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
'audio_processing/audio_processing.gypi',
@@ -37,14 +27,9 @@
'video_render/video_render.gypi',
],
'conditions': [
- ['include_opus==1', {
- 'includes': ['audio_coding/codecs/opus/opus.gypi',],
- }],
['include_tests==1', {
'includes': [
- 'audio_coding/codecs/isac/isac_test.gypi',
- 'audio_coding/codecs/isac/isacfix_test.gypi',
- 'audio_coding/codecs/tools/audio_codec_speed_tests.gypi',
+ 'audio_coding/audio_coding_tests.gypi',
'audio_processing/audio_processing_tests.gypi',
'rtp_rtcp/test/testFec/test_fec.gypi',
'video_coding/main/source/video_coding_test.gypi',
@@ -101,6 +86,7 @@
'<(webrtc_root)/test/test.gyp:frame_generator',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
],
'sources': [
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
@@ -175,7 +161,6 @@
'audio_processing/agc/pitch_internal_unittest.cc',
'audio_processing/agc/pole_zero_filter_unittest.cc',
'audio_processing/agc/standalone_vad_unittest.cc',
- 'audio_processing/agc/test/test_utils.cc',
'audio_processing/beamformer/complex_matrix_unittest.cc',
'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
'audio_processing/beamformer/matrix_unittest.cc',
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 6580281..c3853f6 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -40,63 +40,67 @@
'test/bwe_test_logging.h',
], # source
},
- {
- 'target_name': 'bwe_tools_util',
- 'type': 'static_library',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- 'rtp_rtcp',
- ],
- 'sources': [
- 'tools/bwe_rtp.cc',
- 'tools/bwe_rtp.h',
- ],
- },
- {
- 'target_name': 'bwe_rtp_to_text',
- 'type': 'executable',
- 'includes': [
- '../rtp_rtcp/source/rtp_rtcp.gypi',
- ],
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- 'bwe_tools_util',
- 'rtp_rtcp',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'tools/rtp_to_text.cc',
- '<(webrtc_root)/test/rtp_file_reader.cc',
- '<(webrtc_root)/test/rtp_file_reader.h',
- ], # source
- },
- {
- 'target_name': 'bwe_rtp_play',
- 'type': 'executable',
- 'includes': [
- '../rtp_rtcp/source/rtp_rtcp.gypi',
- ],
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- 'bwe_tools_util',
- 'rtp_rtcp',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'tools/bwe_rtp_play.cc',
- '<(webrtc_root)/test/rtp_file_reader.cc',
- '<(webrtc_root)/test/rtp_file_reader.h',
- ], # source
- },
], # targets
+ 'conditions': [
+ ['include_tests==1', {
+ 'targets': [
+ {
+ 'target_name': 'bwe_tools_util',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ 'rtp_rtcp',
+ ],
+ 'sources': [
+ 'tools/bwe_rtp.cc',
+ 'tools/bwe_rtp.h',
+ ],
+ },
+ {
+ 'target_name': 'bwe_rtp_to_text',
+ 'type': 'executable',
+ 'includes': [
+ '../rtp_rtcp/source/rtp_rtcp.gypi',
+ ],
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+ 'bwe_tools_util',
+ 'rtp_rtcp',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'tools/rtp_to_text.cc',
+ ], # source
+ },
+ {
+ 'target_name': 'bwe_rtp_play',
+ 'type': 'executable',
+ 'includes': [
+ '../rtp_rtcp/source/rtp_rtcp.gypi',
+ ],
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+ 'bwe_tools_util',
+ 'rtp_rtcp',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'tools/bwe_rtp_play.cc',
+ ], # source
+ },
+ ],
+ }], # include_tests==1
+ ],
}
diff --git a/webrtc/modules/audio_processing/agc/test/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc
similarity index 100%
rename from webrtc/modules/audio_processing/agc/test/activity_metric.cc
rename to webrtc/tools/agc/activity_metric.cc
diff --git a/webrtc/modules/audio_processing/agc/test/agc_harness.cc b/webrtc/tools/agc/agc_harness.cc
similarity index 98%
rename from webrtc/modules/audio_processing/agc/test/agc_harness.cc
rename to webrtc/tools/agc/agc_harness.cc
index d7c32b0..02d0f65 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_harness.cc
+++ b/webrtc/tools/agc/agc_harness.cc
@@ -12,12 +12,12 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/channel_transport/include/channel_transport.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
+#include "webrtc/tools/agc/agc_manager.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.cc b/webrtc/tools/agc/agc_manager.cc
similarity index 98%
rename from webrtc/modules/audio_processing/agc/test/agc_manager.cc
rename to webrtc/tools/agc/agc_manager.cc
index a741e64..83c0d00 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_manager.cc
+++ b/webrtc/tools/agc/agc_manager.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
+#include "webrtc/tools/agc/agc_manager.h"
#include <assert.h>
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.h b/webrtc/tools/agc/agc_manager.h
similarity index 93%
rename from webrtc/modules/audio_processing/agc/test/agc_manager.h
rename to webrtc/tools/agc/agc_manager.h
index ec8161c..6b3e91d 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_manager.h
+++ b/webrtc/tools/agc/agc_manager.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
+#ifndef WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
+#define WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -78,4 +78,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
+#endif // WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc b/webrtc/tools/agc/agc_manager_integrationtest.cc
similarity index 97%
rename from webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc
rename to webrtc/tools/agc/agc_manager_integrationtest.cc
index 9dbbc22..d4b3632 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc
+++ b/webrtc/tools/agc/agc_manager_integrationtest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
+#include "webrtc/tools/agc/agc_manager.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc b/webrtc/tools/agc/agc_manager_unittest.cc
similarity index 99%
rename from webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc
rename to webrtc/tools/agc/agc_manager_unittest.cc
index 92464ef..fca8dec 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc
+++ b/webrtc/tools/agc/agc_manager_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
+#include "webrtc/tools/agc/agc_manager.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/webrtc/modules/audio_processing/agc/test/agc_test.cc b/webrtc/tools/agc/agc_test.cc
similarity index 97%
rename from webrtc/modules/audio_processing/agc/test/agc_test.cc
rename to webrtc/tools/agc/agc_test.cc
index 413b3b0..2976948 100644
--- a/webrtc/modules/audio_processing/agc/test/agc_test.cc
+++ b/webrtc/tools/agc/agc_test.cc
@@ -16,13 +16,13 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
-#include "webrtc/modules/audio_processing/agc/test/test_utils.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
+#include "webrtc/tools/agc/agc_manager.h"
+#include "webrtc/tools/agc/test_utils.h"
#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
diff --git a/webrtc/modules/audio_processing/agc/test/fake_agc.h b/webrtc/tools/agc/fake_agc.h
similarity index 83%
rename from webrtc/modules/audio_processing/agc/test/fake_agc.h
rename to webrtc/tools/agc/fake_agc.h
index e2aabd8..6b39cd7 100644
--- a/webrtc/modules/audio_processing/agc/test/fake_agc.h
+++ b/webrtc/tools/agc/fake_agc.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_
+#ifndef WEBRTC_TOOLS_AGC_FAKE_AGC_H_
+#define WEBRTC_TOOLS_AGC_FAKE_AGC_H_
#include "webrtc/modules/audio_processing/agc/agc.h"
@@ -43,4 +43,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_
+#endif // WEBRTC_TOOLS_AGC_FAKE_AGC_H_
diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.cc b/webrtc/tools/agc/test_utils.cc
similarity index 96%
rename from webrtc/modules/audio_processing/agc/test/test_utils.cc
rename to webrtc/tools/agc/test_utils.cc
index e7c884b..3a26cb9 100644
--- a/webrtc/modules/audio_processing/agc/test/test_utils.cc
+++ b/webrtc/tools/agc/test_utils.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/agc/test/test_utils.h"
+#include "webrtc/tools/agc/test_utils.h"
#include <cmath>
diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.h b/webrtc/tools/agc/test_utils.h
similarity index 83%
rename from webrtc/modules/audio_processing/agc/test/test_utils.h
rename to webrtc/tools/agc/test_utils.h
index 25dc496..2aca999 100644
--- a/webrtc/modules/audio_processing/agc/test/test_utils.h
+++ b/webrtc/tools/agc/test_utils.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+#ifndef WEBRTC_TOOLS_AGC_TEST_UTILS_H_
+#define WEBRTC_TOOLS_AGC_TEST_UTILS_H_
namespace webrtc {
class AudioFrame;
@@ -25,4 +25,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+#endif // WEBRTC_TOOLS_AGC_TEST_UTILS_H_
diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp
index 0a3d531..e2a5421 100644
--- a/webrtc/tools/tools.gyp
+++ b/webrtc/tools/tools.gyp
@@ -110,8 +110,16 @@
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
],
'sources': [
- '<(webrtc_root)/modules/audio_processing/agc/test/agc_manager.cc',
- '<(webrtc_root)/modules/audio_processing/agc/test/agc_manager.h',
+ 'agc/agc_manager.cc',
+ 'agc/agc_manager.h',
+ ],
+ },
+ {
+ 'target_name': 'agc_test_utils',
+ 'type': 'static_library',
+ 'sources': [
+ 'agc/test_utils.cc',
+ 'agc/test_utils.h',
],
},
{
@@ -126,7 +134,7 @@
'agc_manager',
],
'sources': [
- '<(webrtc_root)/modules/audio_processing/agc/test/agc_harness.cc',
+ 'agc/agc_harness.cc',
],
}, # agc_harness
{
@@ -139,10 +147,10 @@
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'agc_manager',
+ 'agc_test_utils',
],
'sources': [
- '<(webrtc_root)/modules/audio_processing/agc/test/agc_test.cc',
- '<(webrtc_root)/modules/audio_processing/agc/test/test_utils.cc',
+ 'agc/agc_test.cc',
],
}, # agc_proc
{
@@ -154,7 +162,7 @@
'agc_manager',
],
'sources': [
- '<(webrtc_root)/modules/audio_processing/agc/test/activity_metric.cc',
+ 'agc/activity_metric.cc',
],
}, # activity_metric
{
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index 3f772eb..543ffc4 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -96,13 +96,13 @@
'target_name': 'video_engine_tests',
'type': '<(gtest_target_type)',
'sources': [
- 'modules/audio_processing/agc/test/agc_manager_unittest.cc',
+ 'test/common_unittest.cc',
+ 'test/testsupport/metrics/video_metrics_unittest.cc',
+ 'tools/agc/agc_manager_unittest.cc',
'video/bitrate_estimator_tests.cc',
'video/end_to_end_tests.cc',
'video/send_statistics_proxy_unittest.cc',
'video/video_send_stream_tests.cc',
- 'test/common_unittest.cc',
- 'test/testsupport/metrics/video_metrics_unittest.cc',
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
@@ -131,7 +131,7 @@
'type': '<(gtest_target_type)',
'sources': [
'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
- 'modules/audio_processing/agc/test/agc_manager_integrationtest.cc',
+ 'tools/agc/agc_manager_integrationtest.cc',
'video/call_perf_tests.cc',
'video/full_stack.cc',
'video/rampup_tests.cc',
