Merge changes Ic355174a,I8b04ba9e,I3fefb13c
* changes:
libhardware_legacy: Android.mk -> Android.bp
remove legacy audio policy
libhardware_legacy doesn't need libmedia
diff --git a/Android.bp b/Android.bp
index 087342a..d3a3a9c 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,10 +1,51 @@
// Copyright 2006 The Android Open Source Project
+subdirs = [
+ "audio",
+]
+
+cc_library_headers {
+ name: "libhardware_legacy_headers",
+ export_include_dirs: ["include"],
+
+ header_libs: ["libcutils_headers"],
+ export_header_lib_headers: ["libcutils_headers"],
+}
+
cc_library {
name: "libpower",
- srcs: ["power/power.c"],
+ srcs: ["power.c"],
export_include_dirs: ["include"],
shared_libs: ["libcutils", "liblog"],
vendor_available: true,
}
+
+cc_library_shared {
+ name: "libhardware_legacy",
+
+ shared_libs: [
+ "libbase",
+ "libdl",
+ "libcutils",
+ "liblog",
+ ],
+
+ header_libs: [
+ "libhardware_legacy_headers",
+ ],
+ export_header_lib_headers: ["libhardware_legacy_headers"],
+
+ export_include_dirs: ["include"],
+
+ cflags: [
+ "-DQEMU_HARDWARE",
+ "-Wno-unused-parameter",
+ "-Wno-gnu-designator",
+ ],
+
+ srcs: [
+ "power.c",
+ "uevent.c",
+ ],
+}
diff --git a/Android.mk b/Android.mk
deleted file mode 100644
index d17ba64..0000000
--- a/Android.mk
+++ /dev/null
@@ -1,41 +0,0 @@
-# Copyright 2006 The Android Open Source Project
-
-# Setting LOCAL_PATH will mess up all-subdir-makefiles, so do it beforehand.
-legacy_modules := power uevent
-
-SAVE_MAKEFILES := $(call all-named-subdir-makefiles,$(legacy_modules))
-LEGACY_AUDIO_MAKEFILES := $(call all-named-subdir-makefiles,audio)
-
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := libbase libcutils liblog libmedia
-LOCAL_EXPORT_SHARED_LIBRARY_HEADERS := libmedia
-
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-
-LOCAL_CFLAGS += -DQEMU_HARDWARE -Wno-unused-parameter -Wno-gnu-designator
-QEMU_HARDWARE := true
-
-LOCAL_SHARED_LIBRARIES += libdl
-
-include $(SAVE_MAKEFILES)
-
-# TODO: Remove this line b/29915755
-ifndef BRILLO
-LOCAL_WHOLE_STATIC_LIBRARIES := libwifi-hal-common
-endif
-
-LOCAL_MODULE:= libhardware_legacy
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := libhardware_legacy_headers
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-include $(BUILD_HEADER_LIBRARY)
-
-# legacy_audio builds it's own set of libraries that aren't linked into
-# hardware_legacy
-include $(LEGACY_AUDIO_MAKEFILES)
diff --git a/audio/Android.bp b/audio/Android.bp
new file mode 100644
index 0000000..5141dee
--- /dev/null
+++ b/audio/Android.bp
@@ -0,0 +1,25 @@
+// Copyright 2011 The Android Open Source Project
+
+//AUDIO_POLICY_TEST := true
+//ENABLE_AUDIO_DUMP := true
+
+cc_library_static {
+
+ srcs: [
+ "AudioHardwareInterface.cpp",
+ "audio_hw_hal.cpp",
+ ],
+
+ name: "libaudiohw_legacy",
+ static_libs: ["libmedia_helper"],
+ cflags: [
+ "-Wno-unused-parameter",
+ "-Wno-gnu-designator",
+ ],
+
+ header_libs: [
+ "libbase_headers",
+ "libhardware_legacy_headers",
+ ],
+ export_header_lib_headers: ["libhardware_legacy_headers"],
+}
diff --git a/audio/Android.mk b/audio/Android.mk
deleted file mode 100644
index a64c6b8..0000000
--- a/audio/Android.mk
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright 2011 The Android Open Source Project
-
-#AUDIO_POLICY_TEST := true
-#ENABLE_AUDIO_DUMP := true
-
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- AudioHardwareInterface.cpp \
- audio_hw_hal.cpp
-
-LOCAL_MODULE := libaudiohw_legacy
-LOCAL_SHARED_LIBRARIES := libmedia
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_CFLAGS := -Wno-unused-parameter -Wno-gnu-designator
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/../include
-
-include $(BUILD_STATIC_LIBRARY)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- AudioPolicyManagerBase.cpp \
- AudioPolicyCompatClient.cpp \
- audio_policy_hal.cpp
-
-ifeq ($(AUDIO_POLICY_TEST),true)
- LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
-LOCAL_SHARED_LIBRARIES := libmedia
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_MODULE := libaudiopolicy_legacy
-LOCAL_CFLAGS += -Wno-unused-parameter
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/../include
-
-include $(BUILD_STATIC_LIBRARY)
-
-# The default audio policy, for now still implemented on top of legacy
-# policy code
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- AudioPolicyManagerDefault.cpp
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libmedia \
- libutils \
- liblog
-
-LOCAL_STATIC_LIBRARIES := \
- libmedia_helper
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- libaudiopolicy_legacy
-
-LOCAL_MODULE := audio_policy.default
-LOCAL_MODULE_RELATIVE_PATH := hw
-LOCAL_CFLAGS := -Wno-unused-parameter
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-
-include $(BUILD_SHARED_LIBRARY)
-
diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp
deleted file mode 100644
index e2ee222..0000000
--- a/audio/AudioPolicyCompatClient.cpp
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyCompatClient"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-audio_module_handle_t AudioPolicyCompatClient::loadHwModule(const char *moduleName)
-{
- return mServiceOps->load_hw_module(mService, moduleName);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openOutput(audio_module_handle_t module,
- audio_devices_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- audio_channel_mask_t *pChannelMask,
- uint32_t *pLatencyMs,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
-{
- return mServiceOps->open_output_on_module(mService, module, pDevices, pSamplingRate,
- pFormat, pChannelMask, pLatencyMs,
- flags, offloadInfo);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openDuplicateOutput(audio_io_handle_t output1,
- audio_io_handle_t output2)
-{
- return mServiceOps->open_duplicate_output(mService, output1, output2);
-}
-
-status_t AudioPolicyCompatClient::closeOutput(audio_io_handle_t output)
-{
- return mServiceOps->close_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::suspendOutput(audio_io_handle_t output)
-{
- return mServiceOps->suspend_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::restoreOutput(audio_io_handle_t output)
-{
- return mServiceOps->restore_output(mService, output);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openInput(audio_module_handle_t module,
- audio_devices_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- audio_channel_mask_t *pChannelMask)
-{
- return mServiceOps->open_input_on_module(mService, module, pDevices,
- pSamplingRate, pFormat, pChannelMask);
-}
-
-status_t AudioPolicyCompatClient::closeInput(audio_io_handle_t input)
-{
- return mServiceOps->close_input(mService, input);
-}
-
-status_t AudioPolicyCompatClient::invalidateStream(AudioSystem::stream_type stream)
-{
- return mServiceOps->invalidate_stream(mService, (audio_stream_type_t)stream);
-}
-
-status_t AudioPolicyCompatClient::moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
- audio_io_handle_t dstOutput)
-{
- return mServiceOps->move_effects(mService, session, srcOutput, dstOutput);
-}
-
-String8 AudioPolicyCompatClient::getParameters(audio_io_handle_t ioHandle, const String8& keys)
-{
- char *str;
- String8 out_str8;
-
- str = mServiceOps->get_parameters(mService, ioHandle, keys.string());
- out_str8 = String8(str);
- free(str);
-
- return out_str8;
-}
-
-void AudioPolicyCompatClient::setParameters(audio_io_handle_t ioHandle,
- const String8& keyValuePairs,
- int delayMs)
-{
- mServiceOps->set_parameters(mService, ioHandle, keyValuePairs.string(),
- delayMs);
-}
-
-status_t AudioPolicyCompatClient::setStreamVolume(
- AudioSystem::stream_type stream,
- float volume,
- audio_io_handle_t output,
- int delayMs)
-{
- return mServiceOps->set_stream_volume(mService, (audio_stream_type_t)stream,
- volume, output, delayMs);
-}
-
-status_t AudioPolicyCompatClient::startTone(ToneGenerator::tone_type tone,
- AudioSystem::stream_type stream)
-{
- return mServiceOps->start_tone(mService,
- AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
- (audio_stream_type_t)stream);
-}
-
-status_t AudioPolicyCompatClient::stopTone()
-{
- return mServiceOps->stop_tone(mService);
-}
-
-status_t AudioPolicyCompatClient::setVoiceVolume(float volume, int delayMs)
-{
- return mServiceOps->set_voice_volume(mService, volume, delayMs);
-}
-
-}; // namespace android_audio_legacy
diff --git a/audio/AudioPolicyCompatClient.h b/audio/AudioPolicyCompatClient.h
deleted file mode 100644
index 32804be..0000000
--- a/audio/AudioPolicyCompatClient.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-#define ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-
-/************************************/
-/* FOR BACKWARDS COMPATIBILITY ONLY */
-/************************************/
-namespace android_audio_legacy {
-
-class AudioPolicyCompatClient : public AudioPolicyClientInterface {
-public:
- AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps,
- void *service) :
- mServiceOps(serviceOps) , mService(service) {}
-
- virtual audio_module_handle_t loadHwModule(const char *moduleName);
-
- virtual audio_io_handle_t openOutput(audio_module_handle_t module,
- audio_devices_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- audio_channel_mask_t *pChannelMask,
- uint32_t *pLatencyMs,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
- audio_io_handle_t output2);
- virtual status_t closeOutput(audio_io_handle_t output);
- virtual status_t suspendOutput(audio_io_handle_t output);
- virtual status_t restoreOutput(audio_io_handle_t output);
- virtual audio_io_handle_t openInput(audio_module_handle_t module,
- audio_devices_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- audio_channel_mask_t *pChannelMask);
- virtual status_t closeInput(audio_io_handle_t input);
- virtual status_t invalidateStream(AudioSystem::stream_type stream);
- virtual status_t moveEffects(audio_session_t session,
- audio_io_handle_t srcOutput,
- audio_io_handle_t dstOutput);
-
- virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
- virtual void setParameters(audio_io_handle_t ioHandle,
- const String8& keyValuePairs,
- int delayMs = 0);
- virtual status_t setStreamVolume(AudioSystem::stream_type stream,
- float volume,
- audio_io_handle_t output,
- int delayMs = 0);
- virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
- virtual status_t stopTone();
- virtual status_t setVoiceVolume(float volume, int delayMs = 0);
-
-private:
- struct audio_policy_service_ops* mServiceOps;
- void* mService;
-};
-
-}; // namespace android_audio_legacy
-
-#endif // ANDROID_AUDIOPOLICYCLIENTLEGACY_H
diff --git a/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp
deleted file mode 100644
index 74ee22a..0000000
--- a/audio/AudioPolicyManagerBase.cpp
+++ /dev/null
@@ -1,4368 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyManagerBase"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-// A device mask for all audio input devices that are considered "virtual" when evaluating
-// active inputs in getActiveInput()
-#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX
-// A device mask for all audio output devices that are considered "remote" when evaluating
-// active output devices in isStreamActiveRemotely()
-#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
-
-#include <inttypes.h>
-#include <math.h>
-
-#include <cutils/properties.h>
-#include <utils/Log.h>
-#include <utils/Timers.h>
-
-#include <hardware/audio.h>
-#include <hardware/audio_effect.h>
-#include <hardware_legacy/audio_policy_conf.h>
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-
-namespace android_audio_legacy {
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device,
- AudioSystem::device_connection_state state,
- const char *device_address)
-{
- // device_address can be NULL and should be handled as an empty string in this case,
- // and it is not checked by AudioPolicyInterfaceImpl.cpp
- if (device_address == NULL) {
- device_address = "";
- }
- ALOGV("setDeviceConnectionState() device: 0x%X, state %d, address %s", device, state, device_address);
-
- // connect/disconnect only 1 device at a time
- if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
-
- if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
- ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
- return BAD_VALUE;
- }
-
- // handle output devices
- if (audio_is_output_device(device)) {
- SortedVector <audio_io_handle_t> outputs;
-
- if (!mHasA2dp && audio_is_a2dp_out_device(device)) {
- ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
- return BAD_VALUE;
- }
- if (!mHasUsb && audio_is_usb_out_device(device)) {
- ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
- return BAD_VALUE;
- }
- if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
- ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
- return BAD_VALUE;
- }
-
- // save a copy of the opened output descriptors before any output is opened or closed
- // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
- mPreviousOutputs = mOutputs;
- String8 paramStr;
- switch (state)
- {
- // handle output device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE:
- if (mAvailableOutputDevices & device) {
- ALOGW("setDeviceConnectionState() device already connected: %x", device);
- return INVALID_OPERATION;
- }
- ALOGV("setDeviceConnectionState() connecting device %x", device);
-
- if (mHasA2dp && audio_is_a2dp_out_device(device)) {
- // handle A2DP device connection
- AudioParameter param;
- param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
- paramStr = param.toString();
- } else if (mHasUsb && audio_is_usb_out_device(device)) {
- // handle USB device connection
- paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- }
-
- if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) {
- return INVALID_OPERATION;
- }
- ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
- outputs.size());
- // register new device as available
- mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
-
- if (mHasA2dp && audio_is_a2dp_out_device(device)) {
- // handle A2DP device connection
- mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- mA2dpSuspended = false;
- } else if (audio_is_bluetooth_sco_device(device)) {
- // handle SCO device connection
- mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- } else if (mHasUsb && audio_is_usb_out_device(device)) {
- // handle USB device connection
- mUsbOutCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- }
-
- break;
- // handle output device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
- if (!(mAvailableOutputDevices & device)) {
- ALOGW("setDeviceConnectionState() device not connected: %x", device);
- return INVALID_OPERATION;
- }
-
- ALOGV("setDeviceConnectionState() disconnecting device %x", device);
- // remove device from available output devices
- mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
- checkOutputsForDevice(device, state, outputs, paramStr);
-
- if (mHasA2dp && audio_is_a2dp_out_device(device)) {
- // handle A2DP device disconnection
- mA2dpDeviceAddress = "";
- mA2dpSuspended = false;
- } else if (audio_is_bluetooth_sco_device(device)) {
- // handle SCO device disconnection
- mScoDeviceAddress = "";
- } else if (mHasUsb && audio_is_usb_out_device(device)) {
- // handle USB device disconnection
- mUsbOutCardAndDevice = "";
- }
- // not currently handling multiple simultaneous submixes: ignoring remote submix
- // case and address
- } break;
-
- default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- // outputs must be closed after checkOutputForAllStrategies() is executed
- if (!outputs.isEmpty()) {
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
- // close unused outputs after device disconnection or direct outputs that have been
- // opened by checkOutputsForDevice() to query dynamic parameters
- if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
- (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
- (desc->mDirectOpenCount == 0))) {
- closeOutput(outputs[i]);
- }
- }
- }
-
- updateDevicesAndOutputs();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- // do not force device change on duplicated output because if device is 0, it will
- // also force a device 0 for the two outputs it is duplicated to which may override
- // a valid device selection on those outputs.
- setOutputDevice(mOutputs.keyAt(i),
- getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
- !mOutputs.valueAt(i)->isDuplicated(),
- 0);
- }
-
- return NO_ERROR;
- } // end if is output device
-
- // handle input devices
- if (audio_is_input_device(device)) {
- SortedVector <audio_io_handle_t> inputs;
-
- String8 paramStr;
- switch (state)
- {
- // handle input device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE: {
- if (mAvailableInputDevices & device) {
- ALOGW("setDeviceConnectionState() device already connected: %d", device);
- return INVALID_OPERATION;
- }
-
- if (mHasUsb && audio_is_usb_in_device(device)) {
- // handle USB device connection
- paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- } else if (mHasA2dp && audio_is_a2dp_in_device(device)) {
- // handle A2DP device connection
- AudioParameter param;
- param.add(String8(AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS), String8(device_address));
- paramStr = param.toString();
- }
-
- if (checkInputsForDevice(device, state, inputs, paramStr) != NO_ERROR) {
- return INVALID_OPERATION;
- }
- mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
- }
- break;
-
- // handle input device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
- if (!(mAvailableInputDevices & device)) {
- ALOGW("setDeviceConnectionState() device not connected: %d", device);
- return INVALID_OPERATION;
- }
- checkInputsForDevice(device, state, inputs, paramStr);
- mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
- } break;
-
- default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- closeAllInputs();
-
- return NO_ERROR;
- } // end if is input device
-
- ALOGW("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device,
- const char *device_address)
-{
- // similar to setDeviceConnectionState
- if (device_address == NULL) {
- device_address = "";
- }
- AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
- String8 address = String8(device_address);
- if (audio_is_output_device(device)) {
- if (device & mAvailableOutputDevices) {
- if (audio_is_a2dp_out_device(device) &&
- (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
- return state;
- }
- if (audio_is_bluetooth_sco_device(device) &&
- address != "" && mScoDeviceAddress != address) {
- return state;
- }
- if (audio_is_usb_out_device(device) &&
- (!mHasUsb || (address != "" && mUsbOutCardAndDevice != address))) {
- ALOGE("getDeviceConnectionState() invalid device: %x", device);
- return state;
- }
- if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
- return state;
- }
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- } else if (audio_is_input_device(device)) {
- if (device & mAvailableInputDevices) {
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- }
-
- return state;
-}
-
-void AudioPolicyManagerBase::setPhoneState(int state)
-{
- ALOGV("setPhoneState() state %d", state);
- audio_devices_t newDevice = AUDIO_DEVICE_NONE;
- if (state < 0 || state >= AudioSystem::NUM_MODES) {
- ALOGW("setPhoneState() invalid state %d", state);
- return;
- }
-
- if (state == mPhoneState ) {
- ALOGW("setPhoneState() setting same state %d", state);
- return;
- }
-
- // if leaving call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isInCall()) {
- ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- handleIncallSonification(stream, false, true);
- }
- }
-
- // store previous phone state for management of sonification strategy below
- int oldState = mPhoneState;
- mPhoneState = state;
- bool force = false;
-
- // are we entering or starting a call
- if (!isStateInCall(oldState) && isStateInCall(state)) {
- ALOGV(" Entering call in setPhoneState()");
- // force routing command to audio hardware when starting a call
- // even if no device change is needed
- force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
- }
- } else if (isStateInCall(oldState) && !isStateInCall(state)) {
- ALOGV(" Exiting call in setPhoneState()");
- // force routing command to audio hardware when exiting a call
- // even if no device change is needed
- force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_DTMF][j];
- }
- } else if (isStateInCall(state) && (state != oldState)) {
- ALOGV(" Switching between telephony and VoIP in setPhoneState()");
- // force routing command to audio hardware when switching between telephony and VoIP
- // even if no device change is needed
- force = true;
- }
-
- // check for device and output changes triggered by new phone state
- newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
-
- AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
-
- // force routing command to audio hardware when ending call
- // even if no device change is needed
- if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
- newDevice = hwOutputDesc->device();
- }
-
- int delayMs = 0;
- if (isStateInCall(state)) {
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- // mute media and sonification strategies and delay device switch by the largest
- // latency of any output where either strategy is active.
- // This avoid sending the ring tone or music tail into the earpiece or headset.
- if ((desc->isStrategyActive(STRATEGY_MEDIA,
- SONIFICATION_HEADSET_MUSIC_DELAY,
- sysTime) ||
- desc->isStrategyActive(STRATEGY_SONIFICATION,
- SONIFICATION_HEADSET_MUSIC_DELAY,
- sysTime)) &&
- (delayMs < (int)desc->mLatency*2)) {
- delayMs = desc->mLatency*2;
- }
- setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
- setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
- getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
- setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
- setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
- getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
- }
- }
-
- // change routing is necessary
- setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
-
- // if entering in call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isStateInCall(state)) {
- ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- handleIncallSonification(stream, true, true);
- }
- }
-
- // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
- if (state == AudioSystem::MODE_RINGTONE &&
- isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
- mLimitRingtoneVolume = true;
- } else {
- mLimitRingtoneVolume = false;
- }
-}
-
-void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
- ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
- bool forceVolumeReeval = false;
- switch(usage) {
- case AudioSystem::FOR_COMMUNICATION:
- if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
- config != AudioSystem::FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
- return;
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_MEDIA:
- if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
- config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_ANALOG_DOCK &&
- config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
- config != AudioSystem::FORCE_NO_BT_A2DP) {
- ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_RECORD:
- if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_DOCK:
- if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
- config != AudioSystem::FORCE_BT_DESK_DOCK &&
- config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_ANALOG_DOCK &&
- config != AudioSystem::FORCE_DIGITAL_DOCK) {
- ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_SYSTEM:
- if (config != AudioSystem::FORCE_NONE &&
- config != AudioSystem::FORCE_SYSTEM_ENFORCED) {
- ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- default:
- ALOGW("setForceUse() invalid usage %d", usage);
- break;
- }
-
- // check for device and output changes triggered by new force usage
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t output = mOutputs.keyAt(i);
- audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
- setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
- if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
- applyStreamVolumes(output, newDevice, 0, true);
- }
- }
-
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0) {
- AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
- audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
- if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
- ALOGV("setForceUse() changing device from %x to %x for input %d",
- inputDesc->mDevice, newDevice, activeInput);
- inputDesc->mDevice = newDevice;
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
- mpClientInterface->setParameters(activeInput, param.toString());
- }
- }
-
-}
-
-AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
-{
- return mForceUse[usage];
-}
-
-void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
-{
- ALOGV("setSystemProperty() property %s, value %s", property, value);
-}
-
-// Find a direct output profile compatible with the parameters passed, even if the input flags do
-// not explicitly request a direct output
-AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getProfileForDirectOutput(
- audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
-{
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
- IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask,
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
- if (mAvailableOutputDevices & profile->mSupportedDevices) {
- return mHwModules[i]->mOutputProfiles[j];
- }
- }
- } else {
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask,
- AUDIO_OUTPUT_FLAG_DIRECT)) {
- if (mAvailableOutputDevices & profile->mSupportedDevices) {
- return mHwModules[i]->mOutputProfiles[j];
- }
- }
- }
- }
- }
- return 0;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- AudioSystem::output_flags flags,
- const audio_offload_info_t *offloadInfo)
-{
- audio_io_handle_t output = 0;
- uint32_t latency = 0;
- routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
- device, stream, samplingRate, format, channelMask, flags);
-
-#ifdef AUDIO_POLICY_TEST
- if (mCurOutput != 0) {
- ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
- mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
- if (mTestOutputs[mCurOutput] == 0) {
- ALOGV("getOutput() opening test output");
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = mTestDevice;
- outputDesc->mSamplingRate = mTestSamplingRate;
- outputDesc->mFormat = mTestFormat;
- outputDesc->mChannelMask = mTestChannels;
- outputDesc->mLatency = mTestLatencyMs;
- outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
- outputDesc->mRefCount[stream] = 0;
- mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags,
- offloadInfo);
- if (mTestOutputs[mCurOutput]) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"),mCurOutput);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
- addOutput(mTestOutputs[mCurOutput], outputDesc);
- }
- }
- return mTestOutputs[mCurOutput];
- }
-#endif //AUDIO_POLICY_TEST
-
- // open a direct output if required by specified parameters
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
- }
-
- // Do not allow offloading if one non offloadable effect is enabled. This prevents from
- // creating an offloaded track and tearing it down immediately after start when audioflinger
- // detects there is an active non offloadable effect.
- // FIXME: We should check the audio session here but we do not have it in this context.
- // This may prevent offloading in rare situations where effects are left active by apps
- // in the background.
- IOProfile *profile = NULL;
- if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
- !isNonOffloadableEffectEnabled()) {
- profile = getProfileForDirectOutput(device,
- samplingRate,
- format,
- channelMask,
- (audio_output_flags_t)flags);
- }
-
- if (profile != NULL) {
- AudioOutputDescriptor *outputDesc = NULL;
-
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (profile == desc->mProfile)) {
- outputDesc = desc;
- // reuse direct output if currently open and configured with same parameters
- if ((samplingRate == outputDesc->mSamplingRate) &&
- (format == outputDesc->mFormat) &&
- (channelMask == outputDesc->mChannelMask)) {
- outputDesc->mDirectOpenCount++;
- ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
- return mOutputs.keyAt(i);
- }
- }
- }
- // close direct output if currently open and configured with different parameters
- if (outputDesc != NULL) {
- closeOutput(outputDesc->mId);
- }
- outputDesc = new AudioOutputDescriptor(profile);
- outputDesc->mDevice = device;
- outputDesc->mSamplingRate = samplingRate;
- outputDesc->mFormat = format;
- outputDesc->mChannelMask = channelMask;
- outputDesc->mLatency = 0;
- outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
- outputDesc->mRefCount[stream] = 0;
- outputDesc->mStopTime[stream] = 0;
- outputDesc->mDirectOpenCount = 1;
- output = mpClientInterface->openOutput(profile->mModule->mHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags,
- offloadInfo);
-
- // only accept an output with the requested parameters
- if (output == 0 ||
- (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
- (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
- (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
- ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
- "format %d %d, channelMask %04x %04x", output, samplingRate,
- outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
- outputDesc->mChannelMask);
- if (output != 0) {
- mpClientInterface->closeOutput(output);
- }
- delete outputDesc;
- return 0;
- }
- audio_io_handle_t srcOutput = getOutputForEffect();
- addOutput(output, outputDesc);
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput == output) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
- }
- mPreviousOutputs = mOutputs;
- ALOGV("getOutput() returns new direct output %d", output);
- return output;
- }
-
- // ignoring channel mask due to downmix capability in mixer
-
- // open a non direct output
-
- // for non direct outputs, only PCM is supported
- if (audio_is_linear_pcm(format)) {
- // get which output is suitable for the specified stream. The actual
- // routing change will happen when startOutput() will be called
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
-
- output = selectOutput(outputs, flags);
- }
- ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
- "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
-
- ALOGV("getOutput() returns output %d", output);
-
- return output;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- AudioSystem::output_flags flags)
-{
- // select one output among several that provide a path to a particular device or set of
- // devices (the list was previously build by getOutputsForDevice()).
- // The priority is as follows:
- // 1: the output with the highest number of requested policy flags
- // 2: the primary output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
- if (outputs.size() == 1) {
- return outputs[0];
- }
-
- int maxCommonFlags = 0;
- audio_io_handle_t outputFlags = 0;
- audio_io_handle_t outputPrimary = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
- if (!outputDesc->isDuplicated()) {
- int commonFlags = (int)AudioSystem::popCount(outputDesc->mProfile->mFlags & flags);
- if (commonFlags > maxCommonFlags) {
- outputFlags = outputs[i];
- maxCommonFlags = commonFlags;
- ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
- }
- if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- outputPrimary = outputs[i];
- }
- }
- }
-
- if (outputFlags != 0) {
- return outputFlags;
- }
- if (outputPrimary != 0) {
- return outputPrimary;
- }
-
- return outputs[0];
-}
-
-status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
- audio_session_t session)
-{
- ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("startOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // increment usage count for this stream on the requested output:
- // NOTE that the usage count is the same for duplicated output and hardware output which is
- // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
- outputDesc->changeRefCount(stream, 1);
-
- if (outputDesc->mRefCount[stream] == 1) {
- audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
- routing_strategy strategy = getStrategy(stream);
- bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
- (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
- uint32_t waitMs = 0;
- bool force = false;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (desc != outputDesc) {
- // force a device change if any other output is managed by the same hw
- // module and has a current device selection that differs from selected device.
- // In this case, the audio HAL must receive the new device selection so that it can
- // change the device currently selected by the other active output.
- if (outputDesc->sharesHwModuleWith(desc) &&
- desc->device() != newDevice) {
- force = true;
- }
- // wait for audio on other active outputs to be presented when starting
- // a notification so that audio focus effect can propagate.
- uint32_t latency = desc->latency();
- if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
- waitMs = latency;
- }
- }
- }
- uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, true, false);
- }
-
- // apply volume rules for current stream and device if necessary
- checkAndSetVolume(stream,
- mStreams[stream].getVolumeIndex(newDevice),
- output,
- newDevice);
-
- // update the outputs if starting an output with a stream that can affect notification
- // routing
- handleNotificationRoutingForStream(stream);
- if (waitMs > muteWaitMs) {
- usleep((waitMs - muteWaitMs) * 2 * 1000);
- }
- }
- return NO_ERROR;
-}
-
-
-status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
- audio_session_t session)
-{
- ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("stopOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, false, false);
- }
-
- if (outputDesc->mRefCount[stream] > 0) {
- // decrement usage count of this stream on the output
- outputDesc->changeRefCount(stream, -1);
- // store time at which the stream was stopped - see isStreamActive()
- if (outputDesc->mRefCount[stream] == 0) {
- outputDesc->mStopTime[stream] = systemTime();
- audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
- // delay the device switch by twice the latency because stopOutput() is executed when
- // the track stop() command is received and at that time the audio track buffer can
- // still contain data that needs to be drained. The latency only covers the audio HAL
- // and kernel buffers. Also the latency does not always include additional delay in the
- // audio path (audio DSP, CODEC ...)
- setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
-
- // force restoring the device selection on other active outputs if it differs from the
- // one being selected for this output
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t curOutput = mOutputs.keyAt(i);
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (curOutput != output &&
- desc->isActive() &&
- outputDesc->sharesHwModuleWith(desc) &&
- (newDevice != desc->device())) {
- setOutputDevice(curOutput,
- getNewDevice(curOutput, false /*fromCache*/),
- true,
- outputDesc->mLatency*2);
- }
- }
- // update the outputs if stopping one with a stream that can affect notification routing
- handleNotificationRoutingForStream(stream);
- }
- return NO_ERROR;
- } else {
- ALOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
- }
-}
-
-void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
-{
- ALOGV("releaseOutput() %d", output);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("releaseOutput() releasing unknown output %d", output);
- return;
- }
-
-#ifdef AUDIO_POLICY_TEST
- int testIndex = testOutputIndex(output);
- if (testIndex != 0) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
- if (outputDesc->isActive()) {
- mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
- mOutputs.removeItem(output);
- mTestOutputs[testIndex] = 0;
- }
- return;
- }
-#endif //AUDIO_POLICY_TEST
-
- AudioOutputDescriptor *desc = mOutputs.valueAt(index);
- if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
- if (desc->mDirectOpenCount <= 0) {
- ALOGW("releaseOutput() invalid open count %d for output %d",
- desc->mDirectOpenCount, output);
- return;
- }
- if (--desc->mDirectOpenCount == 0) {
- closeOutput(output);
- // If effects where present on the output, audioflinger moved them to the primary
- // output by default: move them back to the appropriate output.
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput != mPrimaryOutput) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
- }
- }
- }
-}
-
-
-audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- AudioSystem::audio_in_acoustics acoustics)
-{
- audio_io_handle_t input = 0;
- audio_devices_t device = getDeviceForInputSource(inputSource);
-
- ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
- inputSource, samplingRate, format, channelMask, acoustics);
-
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInput() could not find device for inputSource %d", inputSource);
- return 0;
- }
-
- // adapt channel selection to input source
- switch(inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- default:
- break;
- }
-
- IOProfile *profile = getInputProfile(device,
- samplingRate,
- format,
- channelMask);
- if (profile == NULL) {
- ALOGW("getInput() could not find profile for device 0x%X, samplingRate %d, format %d, "
- "channelMask 0x%X",
- device, samplingRate, format, channelMask);
- return 0;
- }
-
- if (profile->mModule->mHandle == 0) {
- ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
- return 0;
- }
-
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
-
- inputDesc->mInputSource = inputSource;
- inputDesc->mDevice = device;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = format;
- inputDesc->mChannelMask = channelMask;
- inputDesc->mRefCount = 0;
-
- input = mpClientInterface->openInput(profile->mModule->mHandle,
- &inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannelMask);
-
- // only accept input with the exact requested set of parameters
- if (input == 0 ||
- (samplingRate != inputDesc->mSamplingRate) ||
- (format != inputDesc->mFormat) ||
- (channelMask != inputDesc->mChannelMask)) {
- ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask 0x%X",
- samplingRate, format, channelMask);
- if (input != 0) {
- mpClientInterface->closeInput(input);
- }
- delete inputDesc;
- return 0;
- }
- addInput(input, inputDesc);
-
- return input;
-}
-
-status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
-{
- ALOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("startInput() unknown input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
- if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
- {
- // refuse 2 active AudioRecord clients at the same time except if the active input
- // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
- audio_io_handle_t activeInput = getActiveInput();
- if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
- AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
- if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
- ALOGW("startInput() preempting already started low-priority input %d", activeInput);
- stopInput(activeInput);
- releaseInput(activeInput);
- } else {
- ALOGW("startInput() input %d failed: other input already started", input);
- return INVALID_OPERATION;
- }
- }
- }
-
- audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
- if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
- inputDesc->mDevice = newDevice;
- }
-
- // automatically enable the remote submix output when input is started
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
- }
-
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
-
- int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
- AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
-
- param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
- ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
- mpClientInterface->setParameters(input, param.toString());
-
- inputDesc->mRefCount = 1;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
-{
- ALOGV("stopInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("stopInput() unknown input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
- if (inputDesc->mRefCount == 0) {
- ALOGW("stopInput() input %d already stopped", input);
- return INVALID_OPERATION;
- } else {
- // automatically disable the remote submix output when input is stopped
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
- }
-
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), 0);
- mpClientInterface->setParameters(input, param.toString());
- inputDesc->mRefCount = 0;
- return NO_ERROR;
- }
-}
-
-void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
-{
- ALOGV("releaseInput() %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("releaseInput() releasing unknown input %d", input);
- return;
- }
- mpClientInterface->closeInput(input);
- delete mInputs.valueAt(index);
- mInputs.removeItem(input);
-
- ALOGV("releaseInput() exit");
-}
-
-void AudioPolicyManagerBase::closeAllInputs() {
- for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- mpClientInterface->closeInput(mInputs.keyAt(input_index));
- }
- mInputs.clear();
-}
-
-void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
- int indexMin,
- int indexMax)
-{
- ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
- if (indexMin < 0 || indexMin >= indexMax) {
- ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
- return;
- }
- mStreams[stream].mIndexMin = indexMin;
- mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream,
- int index,
- audio_devices_t device)
-{
-
- if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
-
- // Force max volume if stream cannot be muted
- if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
-
- ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
- stream, device, index);
-
- // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
- // clear all device specific values
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- mStreams[stream].mIndexCur.clear();
- }
- mStreams[stream].mIndexCur.add(device, index);
-
- // compute and apply stream volume on all outputs according to connected device
- status_t status = NO_ERROR;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_devices_t curDevice =
- getDeviceForVolume(mOutputs.valueAt(i)->device());
- if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
- status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
- }
- }
- return status;
-}
-
-status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream,
- int *index,
- audio_devices_t device)
-{
- if (index == NULL) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
- // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
- // the strategy the stream belongs to.
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
- }
- device = getDeviceForVolume(device);
-
- *index = mStreams[stream].getVolumeIndex(device);
- ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
- return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::selectOutputForEffects(
- const SortedVector<audio_io_handle_t>& outputs)
-{
- // select one output among several suitable for global effects.
- // The priority is as follows:
- // 1: An offloaded output. If the effect ends up not being offloadable,
- // AudioFlinger will invalidate the track and the offloaded output
- // will be closed causing the effect to be moved to a PCM output.
- // 2: A deep buffer output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
-
- audio_io_handle_t outputOffloaded = 0;
- audio_io_handle_t outputDeepBuffer = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
- ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- outputOffloaded = outputs[i];
- }
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
- outputDeepBuffer = outputs[i];
- }
- }
-
- ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
- outputOffloaded, outputDeepBuffer);
- if (outputOffloaded != 0) {
- return outputOffloaded;
- }
- if (outputDeepBuffer != 0) {
- return outputDeepBuffer;
- }
-
- return outputs[0];
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(const effect_descriptor_t *desc)
-{
- // apply simple rule where global effects are attached to the same output as MUSIC streams
-
- routing_strategy strategy = getStrategy(AudioSystem::MUSIC);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
-
- audio_io_handle_t output = selectOutputForEffects(dstOutputs);
- ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
- output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
-
- return output;
-}
-
-status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- audio_session_t session,
- int id)
-{
- ssize_t index = mOutputs.indexOfKey(io);
- if (index < 0) {
- index = mInputs.indexOfKey(io);
- if (index < 0) {
- ALOGW("registerEffect() unknown io %d", io);
- return INVALID_OPERATION;
- }
- }
-
- if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
- ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
- desc->name, desc->memoryUsage);
- return INVALID_OPERATION;
- }
- mTotalEffectsMemory += desc->memoryUsage;
- ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
- desc->name, io, strategy, session, id);
- ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
-
- EffectDescriptor *pDesc = new EffectDescriptor();
- memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
- pDesc->mIo = io;
- pDesc->mStrategy = (routing_strategy)strategy;
- pDesc->mSession = session;
- pDesc->mEnabled = false;
-
- mEffects.add(id, pDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::unregisterEffect(int id)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- EffectDescriptor *pDesc = mEffects.valueAt(index);
-
- setEffectEnabled(pDesc, false);
-
- if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
- ALOGW("unregisterEffect() memory %d too big for total %d",
- pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
- pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
- }
- mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
- ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
- pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-
- mEffects.removeItem(id);
- delete pDesc;
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- return setEffectEnabled(mEffects.valueAt(index), enabled);
-}
-
-status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
-{
- if (enabled == pDesc->mEnabled) {
- ALOGV("setEffectEnabled(%s) effect already %s",
- enabled?"true":"false", enabled?"enabled":"disabled");
- return INVALID_OPERATION;
- }
-
- if (enabled) {
- if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
- ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
- pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
- return INVALID_OPERATION;
- }
- mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
- } else {
- if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
- ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
- pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
- pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
- }
- mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
- }
- pDesc->mEnabled = enabled;
- return NO_ERROR;
-}
-
-bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled()
-{
- for (size_t i = 0; i < mEffects.size(); i++) {
- const EffectDescriptor * const pDesc = mEffects.valueAt(i);
- if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
- ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
- ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
- pDesc->mDesc.name, pDesc->mSession);
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
- outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
- if ((inputDescriptor->mInputSource == (int)source ||
- (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION &&
- inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
- && (inputDescriptor->mRefCount > 0)) {
- return true;
- }
- }
- return false;
-}
-
-
-status_t AudioPolicyManagerBase::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
- result.append(buffer);
-
- snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
- result.append(buffer);
- snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
- result.append(buffer);
- snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbOutCardAndDevice.string());
- result.append(buffer);
- snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
-
- snprintf(buffer, SIZE, "\nHW Modules dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mHwModules.size(); i++) {
- snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
- write(fd, buffer, strlen(buffer));
- mHwModules[i]->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nOutputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mOutputs.size(); i++) {
- snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mOutputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nInputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mInputs.size(); i++) {
- snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mInputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nStreams dump:\n");
- write(fd, buffer, strlen(buffer));
- snprintf(buffer, SIZE,
- " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02zu ", i);
- write(fd, buffer, strlen(buffer));
- mStreams[i].dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
- (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
- write(fd, buffer, strlen(buffer));
-
- snprintf(buffer, SIZE, "Registered effects:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mEffects.size(); i++) {
- snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mEffects.valueAt(i)->dump(fd);
- }
-
-
- return NO_ERROR;
-}
-
-// This function checks for the parameters which can be offloaded.
-// This can be enhanced depending on the capability of the DSP and policy
-// of the system.
-bool AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo)
-{
- ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
- " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
- offloadInfo.sample_rate, offloadInfo.channel_mask,
- offloadInfo.format,
- offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
- offloadInfo.has_video);
-
- // Check if offload has been disabled
- char propValue[PROPERTY_VALUE_MAX];
- if (property_get("audio.offload.disable", propValue, "0")) {
- if (atoi(propValue) != 0) {
- ALOGV("offload disabled by audio.offload.disable=%s", propValue );
- return false;
- }
- }
-
- // Check if stream type is music, then only allow offload as of now.
- if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
- {
- ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
- return false;
- }
-
- //TODO: enable audio offloading with video when ready
- if (offloadInfo.has_video)
- {
- ALOGV("isOffloadSupported: has_video == true, returning false");
- return false;
- }
-
- //If duration is less than minimum value defined in property, return false
- if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
- if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
- ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
- return false;
- }
- } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
- ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
- return false;
- }
-
- // Do not allow offloading if one non offloadable effect is enabled. This prevents from
- // creating an offloaded track and tearing it down immediately after start when audioflinger
- // detects there is an active non offloadable effect.
- // FIXME: We should check the audio session here but we do not have it in this context.
- // This may prevent offloading in rare situations where effects are left active by apps
- // in the background.
- if (isNonOffloadableEffectEnabled()) {
- return false;
- }
-
- // See if there is a profile to support this.
- // AUDIO_DEVICE_NONE
- IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
- offloadInfo.sample_rate,
- offloadInfo.format,
- offloadInfo.channel_mask,
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
- return (profile != NULL);
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerBase
-// ----------------------------------------------------------------------------
-
-AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
- :
-#ifdef AUDIO_POLICY_TEST
- Thread(false),
-#endif //AUDIO_POLICY_TEST
- mPrimaryOutput((audio_io_handle_t)0),
- mAvailableOutputDevices(AUDIO_DEVICE_NONE),
- mPhoneState(AudioSystem::MODE_NORMAL),
- mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
- mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
- mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false),
- mSpeakerDrcEnabled(false)
-{
- mpClientInterface = clientInterface;
-
- for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
- mForceUse[i] = AudioSystem::FORCE_NONE;
- }
-
- mA2dpDeviceAddress = String8("");
- mScoDeviceAddress = String8("");
- mUsbOutCardAndDevice = String8("");
-
- if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
- if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
- ALOGE("could not load audio policy configuration file, setting defaults");
- defaultAudioPolicyConfig();
- }
- }
-
- // must be done after reading the policy
- initializeVolumeCurves();
-
- // open all output streams needed to access attached devices
- for (size_t i = 0; i < mHwModules.size(); i++) {
- mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
- if (mHwModules[i]->mHandle == 0) {
- ALOGW("could not open HW module %s", mHwModules[i]->mName);
- continue;
- }
- // open all output streams needed to access attached devices
- // except for direct output streams that are only opened when they are actually
- // required by an app.
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
-
- if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
- ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
- outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
- outProfile->mSupportedDevices);
- audio_io_handle_t output = mpClientInterface->openOutput(
- outProfile->mModule->mHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (output == 0) {
- delete outputDesc;
- } else {
- mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
- (outProfile->mSupportedDevices & mAttachedOutputDevices));
- if (mPrimaryOutput == 0 &&
- outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- mPrimaryOutput = output;
- }
- addOutput(output, outputDesc);
- setOutputDevice(output,
- (audio_devices_t)(mDefaultOutputDevice &
- outProfile->mSupportedDevices),
- true);
- }
- }
- }
- }
-
- ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
- "Not output found for attached devices %08x",
- (mAttachedOutputDevices & ~mAvailableOutputDevices));
-
- ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
-
- updateDevicesAndOutputs();
-
-#ifdef AUDIO_POLICY_TEST
- if (mPrimaryOutput != 0) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
-
- mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mTestSamplingRate = 44100;
- mTestFormat = AudioSystem::PCM_16_BIT;
- mTestChannels = AudioSystem::CHANNEL_OUT_STEREO;
- mTestLatencyMs = 0;
- mCurOutput = 0;
- mDirectOutput = false;
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- mTestOutputs[i] = 0;
- }
-
- const size_t SIZE = 256;
- char buffer[SIZE];
- snprintf(buffer, SIZE, "AudioPolicyManagerTest");
- run(buffer, ANDROID_PRIORITY_AUDIO);
- }
-#endif //AUDIO_POLICY_TEST
-}
-
-AudioPolicyManagerBase::~AudioPolicyManagerBase()
-{
-#ifdef AUDIO_POLICY_TEST
- exit();
-#endif //AUDIO_POLICY_TEST
- for (size_t i = 0; i < mOutputs.size(); i++) {
- mpClientInterface->closeOutput(mOutputs.keyAt(i));
- delete mOutputs.valueAt(i);
- }
- for (size_t i = 0; i < mInputs.size(); i++) {
- mpClientInterface->closeInput(mInputs.keyAt(i));
- delete mInputs.valueAt(i);
- }
- for (size_t i = 0; i < mHwModules.size(); i++) {
- delete mHwModules[i];
- }
-}
-
-status_t AudioPolicyManagerBase::initCheck()
-{
- return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerBase::threadLoop()
-{
- ALOGV("entering threadLoop()");
- while (!exitPending())
- {
- String8 command;
- int valueInt;
- String8 value;
-
- Mutex::Autolock _l(mLock);
- mWaitWorkCV.waitRelative(mLock, milliseconds(50));
-
- command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
- AudioParameter param = AudioParameter(command);
-
- if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
- valueInt != 0) {
- ALOGV("Test command %s received", command.string());
- String8 target;
- if (param.get(String8("target"), target) != NO_ERROR) {
- target = "Manager";
- }
- if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_output"));
- mCurOutput = valueInt;
- }
- if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_direct"));
- if (value == "false") {
- mDirectOutput = false;
- } else if (value == "true") {
- mDirectOutput = true;
- }
- }
- if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_input"));
- mTestInput = valueInt;
- }
-
- if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_format"));
- int format = AudioSystem::INVALID_FORMAT;
- if (value == "PCM 16 bits") {
- format = AudioSystem::PCM_16_BIT;
- } else if (value == "PCM 8 bits") {
- format = AudioSystem::PCM_8_BIT;
- } else if (value == "Compressed MP3") {
- format = AudioSystem::MP3;
- }
- if (format != AudioSystem::INVALID_FORMAT) {
- if (target == "Manager") {
- mTestFormat = format;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("format"), format);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_channels"));
- int channels = 0;
-
- if (value == "Channels Stereo") {
- channels = AudioSystem::CHANNEL_OUT_STEREO;
- } else if (value == "Channels Mono") {
- channels = AudioSystem::CHANNEL_OUT_MONO;
- }
- if (channels != 0) {
- if (target == "Manager") {
- mTestChannels = channels;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("channels"), channels);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_sampleRate"));
- if (valueInt >= 0 && valueInt <= 96000) {
- int samplingRate = valueInt;
- if (target == "Manager") {
- mTestSamplingRate = samplingRate;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("sampling_rate"), samplingRate);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
-
- if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_reopen"));
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
- mpClientInterface->closeOutput(mPrimaryOutput);
-
- audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
-
- delete mOutputs.valueFor(mPrimaryOutput);
- mOutputs.removeItem(mPrimaryOutput);
-
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mPrimaryOutput == 0) {
- ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
- } else {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
- addOutput(mPrimaryOutput, outputDesc);
- }
- }
-
-
- mpClientInterface->setParameters(0, String8("test_cmd_policy="));
- }
- }
- return false;
-}
-
-void AudioPolicyManagerBase::exit()
-{
- {
- AutoMutex _l(mLock);
- requestExit();
- mWaitWorkCV.signal();
- }
- requestExitAndWait();
-}
-
-int AudioPolicyManagerBase::testOutputIndex(audio_io_handle_t output)
-{
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- if (output == mTestOutputs[i]) return i;
- }
- return 0;
-}
-#endif //AUDIO_POLICY_TEST
-
-// ---
-
-void AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
-{
- outputDesc->mId = id;
- mOutputs.add(id, outputDesc);
-}
-
-void AudioPolicyManagerBase::addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc)
-{
- inputDesc->mId = id;
- mInputs.add(id, inputDesc);
-}
-
-status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device,
- AudioSystem::device_connection_state state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 paramStr)
-{
- AudioOutputDescriptor *desc;
-
- if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
- // first list already open outputs that can be routed to this device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) {
- ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- }
- }
- // then look for output profiles that can be routed to this device
- SortedVector<IOProfile *> profiles;
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
- profiles.add(mHwModules[i]->mOutputProfiles[j]);
- }
- }
- }
-
- if (profiles.isEmpty() && outputs.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
-
- // open outputs for matching profiles if needed. Direct outputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
- IOProfile *profile = profiles[profile_index];
-
- // nothing to do if one output is already opened for this profile
- size_t j;
- for (j = 0; j < mOutputs.size(); j++) {
- desc = mOutputs.valueAt(j);
- if (!desc->isDuplicated() && desc->mProfile == profile) {
- break;
- }
- }
- if (j != mOutputs.size()) {
- continue;
- }
-
- ALOGV("opening output for device %08x with params %s", device, paramStr.string());
- desc = new AudioOutputDescriptor(profile);
- desc->mDevice = device;
- audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
- offloadInfo.sample_rate = desc->mSamplingRate;
- offloadInfo.format = desc->mFormat;
- offloadInfo.channel_mask = desc->mChannelMask;
-
- audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask,
- &desc->mLatency,
- desc->mFlags,
- &offloadInfo);
- if (output != 0) {
- if (!paramStr.isEmpty()) {
- // Here is where the out_set_parameters() for card & device gets called
- mpClientInterface->setParameters(output, paramStr);
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadSamplingRates(value + 1, profile);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkOutputsForDevice() direct output sup formats %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadFormats(value + 1, profile);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadOutChannels(value + 1, profile);
- }
- }
- if (((profile->mSamplingRates[0] == 0) &&
- (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == 0) &&
- (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) &&
- (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkOutputsForDevice() direct output missing param");
- mpClientInterface->closeOutput(output);
- output = 0;
- } else if (profile->mSamplingRates[0] == 0) {
- mpClientInterface->closeOutput(output);
- desc->mSamplingRate = profile->mSamplingRates[1];
- offloadInfo.sample_rate = desc->mSamplingRate;
- output = mpClientInterface->openOutput(
- profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask,
- &desc->mLatency,
- desc->mFlags,
- &offloadInfo);
- }
-
- if (output != 0) {
- addOutput(output, desc);
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
- audio_io_handle_t duplicatedOutput = 0;
-
- // set initial stream volume for device
- applyStreamVolumes(output, device, 0, true);
-
- //TODO: configure audio effect output stage here
-
- // open a duplicating output thread for the new output and the primary output
- duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
- mPrimaryOutput);
- if (duplicatedOutput != 0) {
- // add duplicated output descriptor
- AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
- dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
- dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
- dupOutputDesc->mSamplingRate = desc->mSamplingRate;
- dupOutputDesc->mFormat = desc->mFormat;
- dupOutputDesc->mChannelMask = desc->mChannelMask;
- dupOutputDesc->mLatency = desc->mLatency;
- addOutput(duplicatedOutput, dupOutputDesc);
- applyStreamVolumes(duplicatedOutput, device, 0, true);
- } else {
- ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
- mPrimaryOutput, output);
- mpClientInterface->closeOutput(output);
- mOutputs.removeItem(output);
- output = 0;
- }
- }
- }
- }
- if (output == 0) {
- ALOGW("checkOutputsForDevice() could not open output for device %x", device);
- delete desc;
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- outputs.add(output);
- ALOGV("checkOutputsForDevice(): adding output %d", output);
- }
- }
-
- if (profiles.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
- } else { // Disconnect
- // check if one opened output is not needed any more after disconnecting one device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() &&
- !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
- ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
- if (profile->mSupportedDevices & device) {
- ALOGV("checkOutputsForDevice(): clearing direct output profile %zu on module %zu",
- j, i);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::checkInputsForDevice(audio_devices_t device,
- AudioSystem::device_connection_state state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 paramStr)
-{
- AudioInputDescriptor *desc;
- if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
- // first list already open inputs that can be routed to this device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile->mSupportedDevices & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
-
- // then look for input profiles that can be routed to this device
- SortedVector<IOProfile *> profiles;
- for (size_t module_index = 0; module_index < mHwModules.size(); module_index++)
- {
- if (mHwModules[module_index]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_index]->mInputProfiles.size();
- profile_index++)
- {
- if (mHwModules[module_index]->mInputProfiles[profile_index]->mSupportedDevices
- & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
- profile_index, module_index);
- profiles.add(mHwModules[module_index]->mInputProfiles[profile_index]);
- }
- }
- }
-
- if (profiles.isEmpty() && inputs.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
-
- // open inputs for matching profiles if needed. Direct inputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
-
- IOProfile *profile = profiles[profile_index];
- // nothing to do if one input is already opened for this profile
- size_t input_index;
- for (input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile == profile) {
- break;
- }
- }
- if (input_index != mInputs.size()) {
- continue;
- }
-
- ALOGV("opening input for device 0x%X with params %s", device, paramStr.string());
- desc = new AudioInputDescriptor(profile);
- desc->mDevice = device;
-
- audio_io_handle_t input = mpClientInterface->openInput(profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask);
-
- if (input != 0) {
- if (!paramStr.isEmpty()) {
- mpClientInterface->setParameters(input, paramStr);
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadSamplingRates(value + 1, profile);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadFormats(value + 1, profile);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkInputsForDevice() direct input sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- loadInChannels(value + 1, profile);
- }
- }
- if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkInputsForDevice() direct input missing param");
- mpClientInterface->closeInput(input);
- input = 0;
- }
-
- if (input != 0) {
- addInput(input, desc);
- }
- } // endif input != 0
-
- if (input == 0) {
- ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
- delete desc;
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- inputs.add(input);
- ALOGV("checkInputsForDevice(): adding input %d", input);
- }
- } // end scan profiles
-
- if (profiles.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
- } else {
- // Disconnect
- // check if one opened input is not needed any more after disconnecting one device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (!(desc->mProfile->mSupportedDevices & mAvailableInputDevices)) {
- ALOGV("checkInputsForDevice(): disconnecting adding input %d",
- mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t module_index = 0; module_index < mHwModules.size(); module_index++)
- {
- if (mHwModules[module_index]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_index]->mInputProfiles.size();
- profile_index++)
- {
- IOProfile *profile = mHwModules[module_index]->mInputProfiles[profile_index];
- if (profile->mSupportedDevices & device) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
- profile_index, module_index);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- } // end disconnect
-
- return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output)
-{
- ALOGV("closeOutput(%d)", output);
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- if (outputDesc == NULL) {
- ALOGW("closeOutput() unknown output %d", output);
- return;
- }
-
- // look for duplicated outputs connected to the output being removed.
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
- if (dupOutputDesc->isDuplicated() &&
- (dupOutputDesc->mOutput1 == outputDesc ||
- dupOutputDesc->mOutput2 == outputDesc)) {
- AudioOutputDescriptor *outputDesc2;
- if (dupOutputDesc->mOutput1 == outputDesc) {
- outputDesc2 = dupOutputDesc->mOutput2;
- } else {
- outputDesc2 = dupOutputDesc->mOutput1;
- }
- // As all active tracks on duplicated output will be deleted,
- // and as they were also referenced on the other output, the reference
- // count for their stream type must be adjusted accordingly on
- // the other output.
- for (int j = 0; j < (int)AudioSystem::NUM_STREAM_TYPES; j++) {
- int refCount = dupOutputDesc->mRefCount[j];
- outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount);
- }
- audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
- ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
-
- mpClientInterface->closeOutput(duplicatedOutput);
- delete mOutputs.valueFor(duplicatedOutput);
- mOutputs.removeItem(duplicatedOutput);
- }
- }
-
- AudioParameter param;
- param.add(String8("closing"), String8("true"));
- mpClientInterface->setParameters(output, param.toString());
-
- mpClientInterface->closeOutput(output);
- delete outputDesc;
- mOutputs.removeItem(output);
- mPreviousOutputs = mOutputs;
-}
-
-SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
-{
- SortedVector<audio_io_handle_t> outputs;
-
- ALOGVV("getOutputsForDevice() device %04x", device);
- for (size_t i = 0; i < openOutputs.size(); i++) {
- ALOGVV("output %d isDuplicated=%d device=%04x",
- i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
- if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
- ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
- outputs.add(openOutputs.keyAt(i));
- }
- }
- return outputs;
-}
-
-bool AudioPolicyManagerBase::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2)
-{
- if (outputs1.size() != outputs2.size()) {
- return false;
- }
- for (size_t i = 0; i < outputs1.size(); i++) {
- if (outputs1[i] != outputs2[i]) {
- return false;
- }
- }
- return true;
-}
-
-void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy)
-{
- audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
- audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
-
- if (!vectorsEqual(srcOutputs,dstOutputs)) {
- ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
- strategy, srcOutputs[0], dstOutputs[0]);
- // mute strategy while moving tracks from one output to another
- for (size_t i = 0; i < srcOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
- if (desc->isStrategyActive(strategy)) {
- setStrategyMute(strategy, true, srcOutputs[i]);
- setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
- }
- }
-
- // Move effects associated to this strategy from previous output to new output
- if (strategy == STRATEGY_MEDIA) {
- audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
- SortedVector<audio_io_handle_t> moved;
- for (size_t i = 0; i < mEffects.size(); i++) {
- EffectDescriptor *desc = mEffects.valueAt(i);
- if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
- desc->mIo != fxOutput) {
- if (moved.indexOf(desc->mIo) < 0) {
- ALOGV("checkOutputForStrategy() moving effect %d to output %d",
- mEffects.keyAt(i), fxOutput);
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
- fxOutput);
- moved.add(desc->mIo);
- }
- desc->mIo = fxOutput;
- }
- }
- }
- // Move tracks associated to this strategy from previous output to new output
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- if (getStrategy((AudioSystem::stream_type)i) == strategy) {
- mpClientInterface->invalidateStream((AudioSystem::stream_type)i);
- }
- }
- }
-}
-
-void AudioPolicyManagerBase::checkOutputForAllStrategies()
-{
- checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
- checkOutputForStrategy(STRATEGY_PHONE);
- checkOutputForStrategy(STRATEGY_SONIFICATION);
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- checkOutputForStrategy(STRATEGY_MEDIA);
- checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput()
-{
- if (!mHasA2dp) {
- return 0;
- }
-
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
- return mOutputs.keyAt(i);
- }
- }
-
- return 0;
-}
-
-void AudioPolicyManagerBase::checkA2dpSuspend()
-{
- if (!mHasA2dp) {
- return;
- }
- audio_io_handle_t a2dpOutput = getA2dpOutput();
- if (a2dpOutput == 0) {
- return;
- }
-
- // suspend A2DP output if:
- // (NOT already suspended) &&
- // ((SCO device is connected &&
- // (forced usage for communication || for record is SCO))) ||
- // (phone state is ringing || in call)
- //
- // restore A2DP output if:
- // (Already suspended) &&
- // ((SCO device is NOT connected ||
- // (forced usage NOT for communication && NOT for record is SCO))) &&
- // (phone state is NOT ringing && NOT in call)
- //
- if (mA2dpSuspended) {
- if (((mScoDeviceAddress == "") ||
- ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
- (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
- ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
- (mPhoneState != AudioSystem::MODE_RINGTONE))) {
-
- mpClientInterface->restoreOutput(a2dpOutput);
- mA2dpSuspended = false;
- }
- } else {
- if (((mScoDeviceAddress != "") &&
- ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
- (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
- ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
- (mPhoneState == AudioSystem::MODE_RINGTONE))) {
-
- mpClientInterface->suspendOutput(a2dpOutput);
- mA2dpSuspended = true;
- }
- }
-}
-
-audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
-{
- audio_devices_t device = AUDIO_DEVICE_NONE;
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- // check the following by order of priority to request a routing change if necessary:
- // 1: the strategy enforced audible is active on the output:
- // use device for strategy enforced audible
- // 2: we are in call or the strategy phone is active on the output:
- // use device for strategy phone
- // 3: the strategy sonification is active on the output:
- // use device for strategy sonification
- // 4: the strategy "respectful" sonification is active on the output:
- // use device for strategy "respectful" sonification
- // 5: the strategy media is active on the output:
- // use device for strategy media
- // 6: the strategy DTMF is active on the output:
- // use device for strategy DTMF
- if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
- } else if (isInCall() ||
- outputDesc->isStrategyActive(STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
- device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
- device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
- }
-
- ALOGV("getNewDevice() selected device %x", device);
- return device;
-}
-
-uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
- return (uint32_t)getStrategy(stream);
-}
-
-audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
- audio_devices_t devices;
- // By checking the range of stream before calling getStrategy, we avoid
- // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
- // and then return STRATEGY_MEDIA, but we want to return the empty set.
- if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
- devices = AUDIO_DEVICE_NONE;
- } else {
- AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
- devices = getDeviceForStrategy(strategy, true /*fromCache*/);
- }
- return devices;
-}
-
-AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
- AudioSystem::stream_type stream) {
- // stream to strategy mapping
- switch (stream) {
- case AudioSystem::VOICE_CALL:
- case AudioSystem::BLUETOOTH_SCO:
- return STRATEGY_PHONE;
- case AudioSystem::RING:
- case AudioSystem::ALARM:
- return STRATEGY_SONIFICATION;
- case AudioSystem::NOTIFICATION:
- return STRATEGY_SONIFICATION_RESPECTFUL;
- case AudioSystem::DTMF:
- return STRATEGY_DTMF;
- default:
- ALOGE("unknown stream type");
- case AudioSystem::SYSTEM:
- // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
- // while key clicks are played produces a poor result
- case AudioSystem::TTS:
- case AudioSystem::MUSIC:
- return STRATEGY_MEDIA;
- case AudioSystem::ENFORCED_AUDIBLE:
- return STRATEGY_ENFORCED_AUDIBLE;
- }
-}
-
-void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
- switch(stream) {
- case AudioSystem::MUSIC:
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- updateDevicesAndOutputs();
- break;
- default:
- break;
- }
-}
-
-audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
- bool fromCache)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
-
- if (fromCache) {
- ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
- strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
- }
-
- switch (strategy) {
-
- case STRATEGY_SONIFICATION_RESPECTFUL:
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActiveRemotely(AudioSystem::MUSIC,
- SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing on a remote device, use the the sonification behavior.
- // Note that we test this usecase before testing if media is playing because
- // the isStreamActive() method only informs about the activity of a stream, not
- // if it's for local playback. Note also that we use the same delay between both tests
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing (or has recently played), use the same device
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- } else {
- // when media is not playing anymore, fall back on the sonification behavior
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- }
-
- break;
-
- case STRATEGY_DTMF:
- if (!isInCall()) {
- // when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- break;
- }
- // when in call, DTMF and PHONE strategies follow the same rules
- // FALL THROUGH
-
- case STRATEGY_PHONE:
- // for phone strategy, we first consider the forced use and then the available devices by order
- // of priority
- switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
- case AudioSystem::FORCE_BT_SCO:
- if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
- // if SCO device is requested but no SCO device is available, fall back to default case
- // FALL THROUGH
-
- default: // FORCE_NONE
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (mHasA2dp && !isInCall() &&
- (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
- if (mPhoneState != AudioSystem::MODE_IN_CALL) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
- if (device) break;
- device = mDefaultOutputDevice;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
- }
- break;
-
- case AudioSystem::FORCE_SPEAKER:
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
- // A2DP speaker when forcing to speaker output
- if (mHasA2dp && !isInCall() &&
- (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
- }
- if (mPhoneState != AudioSystem::MODE_IN_CALL) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- if (device) break;
- device = mDefaultOutputDevice;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
- }
- break;
- }
- break;
-
- case STRATEGY_SONIFICATION:
-
- // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
- // handleIncallSonification().
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
- break;
- }
- // FALL THROUGH
-
- case STRATEGY_ENFORCED_AUDIBLE:
- // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
- // except:
- // - when in call where it doesn't default to STRATEGY_PHONE behavior
- // - in countries where not enforced in which case it follows STRATEGY_MEDIA
-
- if ((strategy == STRATEGY_SONIFICATION) ||
- (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
- }
- }
- // The second device used for sonification is the same as the device used by media strategy
- // FALL THROUGH
-
- case STRATEGY_MEDIA: {
- uint32_t device2 = AUDIO_DEVICE_NONE;
- if (strategy != STRATEGY_SONIFICATION) {
- // no sonification on remote submix (e.g. WFD)
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
- // no sonification on aux digital (e.g. HDMI)
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- }
-
- // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
- // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
- device |= device2;
- if (device) break;
- device = mDefaultOutputDevice;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
- }
- } break;
-
- default:
- ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
- break;
- }
-
- ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
- return device;
-}
-
-void AudioPolicyManagerBase::updateDevicesAndOutputs()
-{
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- }
- mPreviousOutputs = mOutputs;
-}
-
-uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs)
-{
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- if (outputDesc->isDuplicated()) {
- return 0;
- }
-
- uint32_t muteWaitMs = 0;
- audio_devices_t device = outputDesc->device();
- bool shouldMute = outputDesc->isActive() && (AudioSystem::popCount(device) >= 2);
- // temporary mute output if device selection changes to avoid volume bursts due to
- // different per device volumes
- bool tempMute = outputDesc->isActive() && (device != prevDevice);
-
- for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- bool mute = shouldMute && (curDevice & device) && (curDevice != device);
- bool doMute = false;
-
- if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = true;
- } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = false;
- }
- if (doMute || tempMute) {
- for (size_t j = 0; j < mOutputs.size(); j++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(j);
- // skip output if it does not share any device with current output
- if ((desc->supportedDevices() & outputDesc->supportedDevices())
- == AUDIO_DEVICE_NONE) {
- continue;
- }
- audio_io_handle_t curOutput = mOutputs.keyAt(j);
- ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
- mute ? "muting" : "unmuting", i, curDevice, curOutput);
- setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
- if (desc->isStrategyActive((routing_strategy)i)) {
- // do tempMute only for current output
- if (tempMute && (desc == outputDesc)) {
- setStrategyMute((routing_strategy)i, true, curOutput);
- setStrategyMute((routing_strategy)i, false, curOutput,
- desc->latency() * 2, device);
- }
- if ((tempMute && (desc == outputDesc)) || mute) {
- if (muteWaitMs < desc->latency()) {
- muteWaitMs = desc->latency();
- }
- }
- }
- }
- }
- }
-
- // FIXME: should not need to double latency if volume could be applied immediately by the
- // audioflinger mixer. We must account for the delay between now and the next time
- // the audioflinger thread for this output will process a buffer (which corresponds to
- // one buffer size, usually 1/2 or 1/4 of the latency).
- muteWaitMs *= 2;
- // wait for the PCM output buffers to empty before proceeding with the rest of the command
- if (muteWaitMs > delayMs) {
- muteWaitMs -= delayMs;
- usleep(muteWaitMs * 1000);
- return muteWaitMs;
- }
- return 0;
-}
-
-uint32_t AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force,
- int delayMs)
-{
- ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- AudioParameter param;
- uint32_t muteWaitMs;
-
- if (outputDesc->isDuplicated()) {
- muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
- muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
- return muteWaitMs;
- }
- // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
- // output profile
- if ((device != AUDIO_DEVICE_NONE) &&
- ((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
- return 0;
- }
-
- // filter devices according to output selected
- device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
-
- audio_devices_t prevDevice = outputDesc->mDevice;
-
- ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
-
- if (device != AUDIO_DEVICE_NONE) {
- outputDesc->mDevice = device;
-
- // Force routing if previously asked for this output
- if (outputDesc->mForceRouting) {
- ALOGV("Force routing to current device as previous device was null for this output");
- force = true;
-
- // Request consumed. Reset mForceRouting to false
- outputDesc->mForceRouting = false;
- }
- }
- else {
- // Device is null and does not reflect the routing. Save the necessity to force
- // re-routing upon next attempt to select a non-null device for this output
- outputDesc->mForceRouting = true;
- }
-
- muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
-
- // Do not change the routing if:
- // - the requested device is AUDIO_DEVICE_NONE
- // - the requested device is the same as current device and force is not specified.
- // Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
- ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
- return muteWaitMs;
- }
-
- ALOGV("setOutputDevice() changing device");
- // do the routing
- param.addInt(String8(AudioParameter::keyRouting), (int)device);
- mpClientInterface->setParameters(output, param.toString(), delayMs);
-
- // update stream volumes according to new device
- applyStreamVolumes(output, device, delayMs);
-
- return muteWaitMs;
-}
-
-AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask)
-{
- // Choose an input profile based on the requested capture parameters: select the first available
- // profile supporting all requested parameters.
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
- {
- IOProfile *profile = mHwModules[i]->mInputProfiles[j];
- // profile->log();
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
- return profile;
- }
- }
- }
- return NULL;
-}
-
-audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
-
- switch (inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- break;
- }
- // FALL THROUGH
-
- case AUDIO_SOURCE_DEFAULT:
- case AUDIO_SOURCE_MIC:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
- break;
- }
- // FALL THROUGH
-
- case AUDIO_SOURCE_VOICE_RECOGNITION:
- case AUDIO_SOURCE_HOTWORD:
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
- mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_CAMCORDER:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- case AUDIO_SOURCE_VOICE_CALL:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- }
- break;
- case AUDIO_SOURCE_REMOTE_SUBMIX:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
- break;
- default:
- ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
- break;
- }
- ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
- return device;
-}
-
-bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device)
-{
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
- return true;
- }
- return false;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs)
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
- if ((input_descriptor->mRefCount > 0)
- && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
- return mInputs.keyAt(i);
- }
- }
- return 0;
-}
-
-
-audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device)
-{
- if (device == AUDIO_DEVICE_NONE) {
- // this happens when forcing a route update and no track is active on an output.
- // In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (AudioSystem::popCount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- ALOGW_IF(AudioSystem::popCount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- device_category deviceCategory = getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[VOLMAX].mIndex -
- curve[VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
- *AudioPolicyManagerBase::sVolumeProfiles[AudioSystem::NUM_STREAM_TYPES]
- [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_SYSTEM
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_RING
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_MUSIC
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_ALARM
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_NOTIFICATION
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_DTMF
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_TTS
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
-};
-
-void AudioPolicyManagerBase::initializeVolumeCurves()
-{
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[i].mVolumeCurve[j] =
- sVolumeProfiles[i][j];
- }
- }
-
- // Check availability of DRC on speaker path: if available, override some of the speaker curves
- if (mSpeakerDrcEnabled) {
- mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sDefaultSystemVolumeCurveDrc;
- mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- }
-}
-
-float AudioPolicyManagerBase::computeVolume(int stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device)
-{
- float volume = 1.0;
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- StreamDescriptor &streamDesc = mStreams[stream];
-
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- // if volume is not 0 (not muted), force media volume to max on digital output
- if (stream == AudioSystem::MUSIC &&
- index != mStreams[stream].mIndexMin &&
- (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
- device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) {
- return 1.0;
- }
-
- volume = volIndexToAmpl(device, streamDesc, index);
-
- // if a headset is connected, apply the following rules to ring tones and notifications
- // to avoid sound level bursts in user's ears:
- // - always attenuate ring tones and notifications volume by 6dB
- // - if music is playing, always limit the volume to current music volume,
- // with a minimum threshold at -36dB so that notification is always perceived.
- const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream);
- if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AUDIO_DEVICE_OUT_WIRED_HEADSET |
- AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
- ((stream_strategy == STRATEGY_SONIFICATION)
- || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
- || (stream == AudioSystem::SYSTEM)
- || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
- (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) &&
- streamDesc.mCanBeMuted) {
- volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
- // when the phone is ringing we must consider that music could have been paused just before
- // by the music application and behave as if music was active if the last music track was
- // just stopped
- if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
- mLimitRingtoneVolume) {
- audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
- float musicVol = computeVolume(AudioSystem::MUSIC,
- mStreams[AudioSystem::MUSIC].getVolumeIndex(musicDevice),
- output,
- musicDevice);
- float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
- musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
- if (volume > minVol) {
- volume = minVol;
- ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
- }
- }
- }
-
- return volume;
-}
-
-status_t AudioPolicyManagerBase::checkAndSetVolume(int stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
-
- // do not change actual stream volume if the stream is muted
- if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
- ALOGVV("checkAndSetVolume() stream %d muted count %d",
- stream, mOutputs.valueFor(output)->mMuteCount[stream]);
- return NO_ERROR;
- }
-
- // do not change in call volume if bluetooth is connected and vice versa
- if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
- (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
- ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
- stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
- return INVALID_OPERATION;
- }
-
- float volume = computeVolume(stream, index, output, device);
- // We actually change the volume if:
- // - the float value returned by computeVolume() changed
- // - the force flag is set
- if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
- force) {
- mOutputs.valueFor(output)->mCurVolume[stream] = volume;
- ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
- // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
- // enabled
- if (stream == AudioSystem::BLUETOOTH_SCO) {
- mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
- }
- mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
- }
-
- if (stream == AudioSystem::VOICE_CALL ||
- stream == AudioSystem::BLUETOOTH_SCO) {
- float voiceVolume;
- // Force voice volume to max for bluetooth SCO as volume is managed by the headset
- if (stream == AudioSystem::VOICE_CALL) {
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
- } else {
- voiceVolume = 1.0;
- }
-
- if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- mLastVoiceVolume = voiceVolume;
- }
- }
-
- return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
- ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
-
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- checkAndSetVolume(stream,
- mStreams[stream].getVolumeIndex(device),
- output,
- device,
- delayMs,
- force);
- }
-}
-
-void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
- setStreamMute(stream, on, output, delayMs, device);
- }
- }
-}
-
-void AudioPolicyManagerBase::setStreamMute(int stream,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- StreamDescriptor &streamDesc = mStreams[stream];
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
- stream, on, output, outputDesc->mMuteCount[stream], device);
-
- if (on) {
- if (outputDesc->mMuteCount[stream] == 0) {
- if (streamDesc.mCanBeMuted &&
- ((stream != AudioSystem::ENFORCED_AUDIBLE) ||
- (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) {
- checkAndSetVolume(stream, 0, output, device, delayMs);
- }
- }
- // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
- outputDesc->mMuteCount[stream]++;
- } else {
- if (outputDesc->mMuteCount[stream] == 0) {
- ALOGV("setStreamMute() unmuting non muted stream!");
- return;
- }
- if (--outputDesc->mMuteCount[stream] == 0) {
- checkAndSetVolume(stream,
- streamDesc.getVolumeIndex(device),
- output,
- device,
- delayMs);
- }
- }
-}
-
-void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange)
-{
- // if the stream pertains to sonification strategy and we are in call we must
- // mute the stream if it is low visibility. If it is high visibility, we must play a tone
- // in the device used for phone strategy and play the tone if the selected device does not
- // interfere with the device used for phone strategy
- // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
- // many times as there are active tracks on the output
- const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream);
- if ((stream_strategy == STRATEGY_SONIFICATION) ||
- ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
- ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
- stream, starting, outputDesc->mDevice, stateChange);
- if (outputDesc->mRefCount[stream]) {
- int muteCount = 1;
- if (stateChange) {
- muteCount = outputDesc->mRefCount[stream];
- }
- if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
- ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- } else {
- ALOGV("handleIncallSonification() high visibility");
- if (outputDesc->device() &
- getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
- ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- }
- if (starting) {
- mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
- } else {
- mpClientInterface->stopTone();
- }
- }
- }
- }
-}
-
-bool AudioPolicyManagerBase::isInCall()
-{
- return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManagerBase::isStateInCall(int state) {
- return ((state == AudioSystem::MODE_IN_CALL) ||
- (state == AudioSystem::MODE_IN_COMMUNICATION));
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
-{
- return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
-{
- return MAX_EFFECTS_MEMORY;
-}
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor(
- const IOProfile *profile)
- : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
- mChannelMask(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0),
- mForceRouting(false)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
- mFlags = profile->mFlags;
- }
-}
-
-audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith(
- const AudioOutputDescriptor *outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices ;
- }
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if ((sysTime == 0) && (inPastMs != 0)) {
- sysTime = systemTime();
- }
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- if (((getStrategy((AudioSystem::stream_type)i) == strategy) ||
- (NUM_STRATEGIES == strategy)) &&
- isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-
-status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
- : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
- mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
- mInputSource(0), mProfile(profile)
-{
- if (profile != NULL) {
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
- }
-}
-
-status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = AudioPolicyManagerBase::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManagerBase::StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " I/O: %d\n", mIo);
- result.append(buffer);
- snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
- result.append(buffer);
- snprintf(buffer, SIZE, " Session: %d\n", mSession);
- result.append(buffer);
- snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
- result.append(buffer);
- snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManagerBase::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(AUDIO_MODULE_HANDLE_NONE)
-{
-}
-
-AudioPolicyManagerBase::HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- delete mOutputProfiles[i];
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- delete mInputProfiles[i];
- }
- free((void *)mName);
-}
-
-void AudioPolicyManagerBase::HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
-}
-
-AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module)
- : mFlags((audio_output_flags_t)0), mModule(module)
-{
-}
-
-AudioPolicyManagerBase::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags) const
-{
- if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
- return false;
- }
-
- if ((mSupportedDevices & device) != device) {
- return false;
- }
- if ((mFlags & flags) != flags) {
- return false;
- }
- size_t i;
- for (i = 0; i < mSamplingRates.size(); i++)
- {
- if (mSamplingRates[i] == samplingRate) {
- break;
- }
- }
- if (i == mSamplingRates.size()) {
- return false;
- }
- for (i = 0; i < mFormats.size(); i++)
- {
- if (mFormats[i] == format) {
- break;
- }
- }
- if (i == mFormats.size()) {
- return false;
- }
- for (i = 0; i < mChannelMasks.size(); i++)
- {
- if (mChannelMasks[i] == channelMask) {
- break;
- }
- }
- if (i == mChannelMasks.size()) {
- return false;
- }
- return true;
-}
-
-void AudioPolicyManagerBase::IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - sampling rates: ");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
- }
-
- snprintf(buffer, SIZE, " - channel masks: ");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
- }
-
- snprintf(buffer, SIZE, " - formats: ");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
- }
-
- snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-void AudioPolicyManagerBase::IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices);
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-// --- audio_policy.conf file parsing
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const struct StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
-};
-
-const struct StringToEnum sFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
-};
-
-const struct StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
- STRING_TO_ENUM(AUDIO_FORMAT_AC3),
- STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const struct StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const struct StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-
-uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-bool AudioPolicyManagerBase::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sFlagNameToEnumTable,
- ARRAY_SIZE(sFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return (audio_output_flags_t)flag;
-}
-
-audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
-void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- profile->mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- profile->mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- profile->mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- profile->mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- profile->mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- profile->mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module)
-{
- cnode *node = root->first_child;
-
- IOProfile *profile = new IOProfile(module);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- loadSamplingRates((char *)node->value, profile);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- loadFormats((char *)node->value, profile);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- loadInChannels((char *)node->value, profile);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices = parseDeviceNames((char *)node->value);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input mSupportedDevices 0x%X", profile->mSupportedDevices);
-
- module->mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- delete profile;
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module)
-{
- cnode *node = root->first_child;
-
- IOProfile *profile = new IOProfile(module);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- loadSamplingRates((char *)node->value, profile);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- loadFormats((char *)node->value, profile);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- loadOutChannels((char *)node->value, profile);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices = parseDeviceNames((char *)node->value);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseFlagNames((char *)node->value);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
- profile->mSupportedDevices, profile->mFlags);
-
- module->mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- delete profile;
- return BAD_VALUE;
- }
-}
-
-void AudioPolicyManagerBase::loadHwModule(cnode *root)
-{
- cnode *node = config_find(root, OUTPUTS_TAG);
- status_t status = NAME_NOT_FOUND;
-
- HwModule *module = new HwModule(root->name);
-
- if (node != NULL) {
- if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
- mHasA2dp = true;
- } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
- mHasUsb = true;
- } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
- mHasRemoteSubmix = true;
- }
-
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading output %s", node->name);
- status_t tmpStatus = loadOutput(node, module);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- node = config_find(root, INPUTS_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading input %s", node->name);
- status_t tmpStatus = loadInput(node, module);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- if (status == NO_ERROR) {
- mHwModules.add(module);
- } else {
- delete module;
- }
-}
-
-void AudioPolicyManagerBase::loadHwModules(cnode *root)
-{
- cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
- if (node == NULL) {
- return;
- }
-
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModules() loading module %s", node->name);
- loadHwModule(node);
- node = node->next;
- }
-}
-
-void AudioPolicyManagerBase::loadGlobalConfig(cnode *root)
-{
- cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
- if (node == NULL) {
- return;
- }
- node = node->first_child;
- while (node) {
- if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
- mAttachedOutputDevices = parseDeviceNames((char *)node->value);
- ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
- "loadGlobalConfig() no attached output devices");
- ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
- } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
- mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- (char *)node->value);
- ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
- "loadGlobalConfig() default device not specified");
- ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
- } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
- mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
- ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
- } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
- mSpeakerDrcEnabled = stringToBool((char *)node->value);
- ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
- }
- node = node->next;
- }
-}
-
-status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path)
-{
- cnode *root;
- char *data;
-
- data = (char *)load_file(path, NULL);
- if (data == NULL) {
- return -ENODEV;
- }
- root = config_node("", "");
- config_load(root, data);
-
- loadGlobalConfig(root);
- loadHwModules(root);
-
- config_free(root);
- free(root);
- free(data);
-
- ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
-
- return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::defaultAudioPolicyConfig(void)
-{
- HwModule *module;
- IOProfile *profile;
-
- mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
- mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
-
- module = new HwModule("primary");
-
- profile = new IOProfile(module);
- profile->mSamplingRates.add(44100);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
- profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
- profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
- module->mOutputProfiles.add(profile);
-
- profile = new IOProfile(module);
- profile->mSamplingRates.add(8000);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
- profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
- module->mInputProfiles.add(profile);
-
- mHwModules.add(module);
-}
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerDefault.cpp b/audio/AudioPolicyManagerDefault.cpp
deleted file mode 100644
index 9083638..0000000
--- a/audio/AudioPolicyManagerDefault.cpp
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyManagerDefault"
-//#define LOG_NDEBUG 0
-
-#include "AudioPolicyManagerDefault.h"
-
-namespace android_audio_legacy {
-
-extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
-{
- return new AudioPolicyManagerDefault(clientInterface);
-}
-
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerDefault.h b/audio/AudioPolicyManagerDefault.h
deleted file mode 100644
index 987fdf0..0000000
--- a/audio/AudioPolicyManagerDefault.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#include <stdint.h>
-#include <stdbool.h>
-
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-
-namespace android_audio_legacy {
-
-class AudioPolicyManagerDefault: public AudioPolicyManagerBase
-{
-
-public:
- explicit AudioPolicyManagerDefault(AudioPolicyClientInterface *clientInterface)
- : AudioPolicyManagerBase(clientInterface) {}
-
- virtual ~AudioPolicyManagerDefault() {}
-
-};
-};
diff --git a/audio/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
deleted file mode 100644
index b7fe245..0000000
--- a/audio/audio_policy_hal.cpp
+++ /dev/null
@@ -1,477 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "legacy_audio_policy_hal"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioPolicyInterface.h>
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-extern "C" {
-
-struct legacy_ap_module {
- struct audio_policy_module module;
-};
-
-struct legacy_ap_device {
- struct audio_policy_device device;
-};
-
-struct legacy_audio_policy {
- struct audio_policy policy;
-
- void *service;
- struct audio_policy_service_ops *aps_ops;
- AudioPolicyCompatClient *service_client;
- AudioPolicyInterface *apm;
-};
-
-static inline struct legacy_audio_policy * to_lap(struct audio_policy *pol)
-{
- return reinterpret_cast<struct legacy_audio_policy *>(pol);
-}
-
-static inline const struct legacy_audio_policy * to_clap(const struct audio_policy *pol)
-{
- return reinterpret_cast<const struct legacy_audio_policy *>(pol);
-}
-
-
-static int ap_set_device_connection_state(struct audio_policy *pol,
- audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->setDeviceConnectionState(
- (AudioSystem::audio_devices)device,
- (AudioSystem::device_connection_state)state,
- device_address);
-}
-
-static audio_policy_dev_state_t ap_get_device_connection_state(
- const struct audio_policy *pol,
- audio_devices_t device,
- const char *device_address)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return (audio_policy_dev_state_t)lap->apm->getDeviceConnectionState(
- (AudioSystem::audio_devices)device,
- device_address);
-}
-
-static void ap_set_phone_state(struct audio_policy *pol, audio_mode_t state)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- // as this is the legacy API, don't change it to use audio_mode_t instead of int
- lap->apm->setPhoneState((int) state);
-}
-
- /* indicate a change in ringer mode */
-static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
- uint32_t mask)
-{
- // deprecated, never called
-}
-
- /* force using a specific device category for the specified usage */
-static void ap_set_force_use(struct audio_policy *pol,
- audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- lap->apm->setForceUse((AudioSystem::force_use)usage,
- (AudioSystem::forced_config)config);
-}
-
- /* retreive current device category forced for a given usage */
-static audio_policy_forced_cfg_t ap_get_force_use(
- const struct audio_policy *pol,
- audio_policy_force_use_t usage)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return (audio_policy_forced_cfg_t)lap->apm->getForceUse(
- (AudioSystem::force_use)usage);
-}
-
-/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
- * can still be muted. */
-static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
- bool can_mute)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- lap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
-}
-
-static int ap_init_check(const struct audio_policy *pol)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->initCheck();
-}
-
-static audio_io_handle_t ap_get_output(struct audio_policy *pol,
- audio_stream_type_t stream,
- uint32_t sampling_rate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
-
- ALOGV("%s: tid %d", __func__, gettid());
- return lap->apm->getOutput((AudioSystem::stream_type)stream,
- sampling_rate, format, channelMask,
- (AudioSystem::output_flags)flags,
- offloadInfo);
-}
-
-static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
- audio_stream_type_t stream, audio_session_t session)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->startOutput(output, (AudioSystem::stream_type)stream,
- session);
-}
-
-static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
- audio_stream_type_t stream, audio_session_t session)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->stopOutput(output, (AudioSystem::stream_type)stream,
- session);
-}
-
-static void ap_release_output(struct audio_policy *pol,
- audio_io_handle_t output)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- lap->apm->releaseOutput(output);
-}
-
-static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
- uint32_t sampling_rate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_in_acoustics_t acoustics)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->getInput((int) inputSource, sampling_rate, format, channelMask,
- (AudioSystem::audio_in_acoustics)acoustics);
-}
-
-static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->startInput(input);
-}
-
-static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->stopInput(input);
-}
-
-static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- lap->apm->releaseInput(input);
-}
-
-static void ap_init_stream_volume(struct audio_policy *pol,
- audio_stream_type_t stream, int index_min,
- int index_max)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- lap->apm->initStreamVolume((AudioSystem::stream_type)stream, index_min,
- index_max);
-}
-
-static int ap_set_stream_volume_index(struct audio_policy *pol,
- audio_stream_type_t stream,
- int index)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
- index,
- AUDIO_DEVICE_OUT_DEFAULT);
-}
-
-static int ap_get_stream_volume_index(const struct audio_policy *pol,
- audio_stream_type_t stream,
- int *index)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
- index,
- AUDIO_DEVICE_OUT_DEFAULT);
-}
-
-static int ap_set_stream_volume_index_for_device(struct audio_policy *pol,
- audio_stream_type_t stream,
- int index,
- audio_devices_t device)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
- index,
- device);
-}
-
-static int ap_get_stream_volume_index_for_device(const struct audio_policy *pol,
- audio_stream_type_t stream,
- int *index,
- audio_devices_t device)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
- index,
- device);
-}
-
-static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
- audio_stream_type_t stream)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->getStrategyForStream((AudioSystem::stream_type)stream);
-}
-
-static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol,
- audio_stream_type_t stream)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->getDevicesForStream((AudioSystem::stream_type)stream);
-}
-
-static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
- const struct effect_descriptor_s *desc)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->getOutputForEffect(desc);
-}
-
-static int ap_register_effect(struct audio_policy *pol,
- const struct effect_descriptor_s *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- audio_session_t session,
- int id)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->registerEffect(desc, io, strategy, session, id);
-}
-
-static int ap_unregister_effect(struct audio_policy *pol, int id)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->unregisterEffect(id);
-}
-
-static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled)
-{
- struct legacy_audio_policy *lap = to_lap(pol);
- return lap->apm->setEffectEnabled(id, enabled);
-}
-
-static bool ap_is_stream_active(const struct audio_policy *pol, audio_stream_type_t stream,
- uint32_t in_past_ms)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->isStreamActive((int) stream, in_past_ms);
-}
-
-static bool ap_is_stream_active_remotely(const struct audio_policy *pol, audio_stream_type_t stream,
- uint32_t in_past_ms)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->isStreamActiveRemotely((int) stream, in_past_ms);
-}
-
-static bool ap_is_source_active(const struct audio_policy *pol, audio_source_t source)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->isSourceActive(source);
-}
-
-static int ap_dump(const struct audio_policy *pol, int fd)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->dump(fd);
-}
-
-static bool ap_is_offload_supported(const struct audio_policy *pol,
- const audio_offload_info_t *info)
-{
- const struct legacy_audio_policy *lap = to_clap(pol);
- return lap->apm->isOffloadSupported(*info);
-}
-
-static int create_legacy_ap(const struct audio_policy_device *device,
- struct audio_policy_service_ops *aps_ops,
- void *service,
- struct audio_policy **ap)
-{
- struct legacy_audio_policy *lap;
- int ret;
-
- if (!service || !aps_ops)
- return -EINVAL;
-
- lap = (struct legacy_audio_policy *)calloc(1, sizeof(*lap));
- if (!lap)
- return -ENOMEM;
-
- lap->policy.set_device_connection_state = ap_set_device_connection_state;
- lap->policy.get_device_connection_state = ap_get_device_connection_state;
- lap->policy.set_phone_state = ap_set_phone_state;
- lap->policy.set_ringer_mode = ap_set_ringer_mode;
- lap->policy.set_force_use = ap_set_force_use;
- lap->policy.get_force_use = ap_get_force_use;
- lap->policy.set_can_mute_enforced_audible =
- ap_set_can_mute_enforced_audible;
- lap->policy.init_check = ap_init_check;
- lap->policy.get_output = ap_get_output;
- lap->policy.start_output = ap_start_output;
- lap->policy.stop_output = ap_stop_output;
- lap->policy.release_output = ap_release_output;
- lap->policy.get_input = ap_get_input;
- lap->policy.start_input = ap_start_input;
- lap->policy.stop_input = ap_stop_input;
- lap->policy.release_input = ap_release_input;
- lap->policy.init_stream_volume = ap_init_stream_volume;
- lap->policy.set_stream_volume_index = ap_set_stream_volume_index;
- lap->policy.get_stream_volume_index = ap_get_stream_volume_index;
- lap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device;
- lap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device;
- lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
- lap->policy.get_devices_for_stream = ap_get_devices_for_stream;
- lap->policy.get_output_for_effect = ap_get_output_for_effect;
- lap->policy.register_effect = ap_register_effect;
- lap->policy.unregister_effect = ap_unregister_effect;
- lap->policy.set_effect_enabled = ap_set_effect_enabled;
- lap->policy.is_stream_active = ap_is_stream_active;
- lap->policy.is_stream_active_remotely = ap_is_stream_active_remotely;
- lap->policy.is_source_active = ap_is_source_active;
- lap->policy.dump = ap_dump;
- lap->policy.is_offload_supported = ap_is_offload_supported;
-
- lap->service = service;
- lap->aps_ops = aps_ops;
- lap->service_client =
- new AudioPolicyCompatClient(aps_ops, service);
- if (!lap->service_client) {
- ret = -ENOMEM;
- goto err_new_compat_client;
- }
-
- lap->apm = createAudioPolicyManager(lap->service_client);
- if (!lap->apm) {
- ret = -ENOMEM;
- goto err_create_apm;
- }
-
- *ap = &lap->policy;
- return 0;
-
-err_create_apm:
- delete lap->service_client;
-err_new_compat_client:
- free(lap);
- *ap = NULL;
- return ret;
-}
-
-static int destroy_legacy_ap(const struct audio_policy_device *ap_dev,
- struct audio_policy *ap)
-{
- struct legacy_audio_policy *lap = to_lap(ap);
-
- if (!lap)
- return 0;
-
- if (lap->apm)
- destroyAudioPolicyManager(lap->apm);
- if (lap->service_client)
- delete lap->service_client;
- free(lap);
- return 0;
-}
-
-static int legacy_ap_dev_close(hw_device_t* device)
-{
- if (device)
- free(device);
- return 0;
-}
-
-static int legacy_ap_dev_open(const hw_module_t* module, const char* name,
- hw_device_t** device)
-{
- struct legacy_ap_device *dev;
-
- if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
- return -EINVAL;
-
- dev = (struct legacy_ap_device *)calloc(1, sizeof(*dev));
- if (!dev)
- return -ENOMEM;
-
- dev->device.common.tag = HARDWARE_DEVICE_TAG;
- dev->device.common.version = 0;
- dev->device.common.module = const_cast<hw_module_t*>(module);
- dev->device.common.close = legacy_ap_dev_close;
- dev->device.create_audio_policy = create_legacy_ap;
- dev->device.destroy_audio_policy = destroy_legacy_ap;
-
- *device = &dev->device.common;
-
- return 0;
-}
-
-static struct hw_module_methods_t legacy_ap_module_methods = {
- .open = legacy_ap_dev_open
-};
-
-struct legacy_ap_module HAL_MODULE_INFO_SYM = {
- .module = {
- .common = {
- .tag = HARDWARE_MODULE_TAG,
- .version_major = 1,
- .version_minor = 0,
- .id = AUDIO_POLICY_HARDWARE_MODULE_ID,
- .name = "LEGACY Audio Policy HAL",
- .author = "The Android Open Source Project",
- .methods = &legacy_ap_module_methods,
- .dso = NULL,
- .reserved = {0},
- },
- },
-};
-
-}; // extern "C"
-
-}; // namespace android_audio_legacy
diff --git a/power/power.c b/power.c
similarity index 100%
rename from power/power.c
rename to power.c
diff --git a/power/Android.mk b/power/Android.mk
deleted file mode 100644
index 3e3ff5d..0000000
--- a/power/Android.mk
+++ /dev/null
@@ -1,3 +0,0 @@
-# Copyright 2006 The Android Open Source Project
-
-LOCAL_SRC_FILES += power/power.c
diff --git a/uevent/uevent.c b/uevent.c
similarity index 100%
rename from uevent/uevent.c
rename to uevent.c
diff --git a/uevent/Android.mk b/uevent/Android.mk
deleted file mode 100644
index 2d8b524..0000000
--- a/uevent/Android.mk
+++ /dev/null
@@ -1,3 +0,0 @@
-# Copyright 2008 The Android Open Source Project
-
-LOCAL_SRC_FILES += uevent/uevent.c