概述
Android音频系统详解
参考好文:
Android 音频系统：从 AudioTrack 到 AudioFlinger
https://blog.csdn.net/zyuanyun/article/details/60890534
Android系统Audio框架介绍
http://blog.csdn.net/yangwen123/article/details/39502689
4.1 分析思路
a. Thread如何创建?
AudioPolicyService是策略的制定者,
AudioFlinger是策略的执行者,
所以: AudioPolicyService根据配置文件使唤AudioFlinger来创建Thread
b. Thread对应output, output对应哪些设备节点?
c. AudioTrack和Track的创建过程: AudioTrack对应哪一个Thread, 对应哪一个output?
d. AudioTrack如何传输数据给Thread？
AudioTrack如何播放、暂停、关闭?
使用HardWare Module 来操作硬件
hw mododule 的名字是什么? 哪些so文件
module支持哪些output?
output支持哪些device,参数是啥?
需要设置/system/etc/audio_policy.conf
4.2 以例子说明几个重要概念
stream type, strategy, device, output, profile, module : policy
out flag ： 比如对于某个专业APP, 它只从HDMI播放声音, 这时就可以指定out flag为AUDIO_OUTPUT_FLAG_DIRECT，这会导致最终的声音无需混音即直接输出到对应的device
Android系统里使用hardware module来访问硬件, 比如声卡
声卡上有喇叭、耳机等等，称为device
为了便于管理, 把一个设备上具有相同参数的一组device称为output,
一个module能支持哪些output，一个output能支持哪些device，使用配置文件/system/etc/audio_policy.conf来描述
app要播放声音, 要指定声音类型: stream type
有那么多的类型, 来来来, 先看它属于哪一类(策略): strategy
根据strategy确定要用什么设备播放: device, 喇叭、耳机还是蓝牙?
根据device确定output, 进而知道对应的playbackthread,
把声音数据传给这个thread
一个stream如何最终选择到一个device,
这些stream如何互相影响(一个高优先级的声音会使得其他声音静音)，
等等等, 统称为policy (政策)
输出、输入设备:
https://blog.csdn.net/zzqhost/article/details/7711935
概念:
module: 硬件操作库, 用来操作device
output: 一组有相同参数的,来自同一硬件的device
device: 喇叭,耳机,....
需要通过设置/system/etc/audio_policy.conf
profile : 配置,用来描述output
a. 本可以支持哪些设备
b. 参数: 采样率,通道
output:
a. 现在可以,实际上可以支持的device
b. 参数
profile: 可以支持喇叭,耳机
output: 接上耳机时,才可以支持耳机
APP: 播放音乐
问; 播放音乐时,有哪么多的路径, App怎么办?
答: App不用管,只用表明声音类型(stream type)
App要指定声音类型(stream type): 太多了,分个组 strategy(具有相同行为的stream)
播放的device相同,播放的优先级相同
4.3 所涉及文件形象讲解
系统服务APP:
frameworks/av/media/mediaserver/main_mediaserver.cpp
AudioFlinger :
AudioFlinger.cpp (frameworks/av/services/audioflinger/AudioFlinger.cpp)
Threads.cpp (frameworks/av/services/audioflinger/Threads.cpp)
Tracks.cpp (frameworks/av/services/audioflinger/Tracks.cpp)
audio_hw_hal.cpp (hardware/libhardware_legacy/audio/Audio_hw_hal.cpp)
AudioHardware.cpp (device/friendly-arm/common/libaudio/AudioHardware.cpp)
AudioPolicyService:
AudioPolicyService.cpp (frameworks/av/services/audiopolicy/AudioPolicyService.cpp)
AudioPolicyClientImpl.cpp (frameworks/av/services/audiopolicy/AudioPolicyClientImpl.cpp)
AudioPolicyInterfaceImpl.cpp(frameworks/av/services/audiopolicy/AudioPolicyInterfaceImpl.cpp)
AudioPolicyManager.cpp (device/friendly-arm/common/libaudio/AudioPolicyManager.cpp)
AudioPolicyManager.h (device/friendly-arm/common/libaudio/AudioPolicyManager.h)
AudioPolicyManagerBase.cpp (hardware/libhardware_legacy/audio/AudioPolicyManagerBase.cpp)
堪误: 上面3个文件被以下文件替代
AudioPolicyManager.cpp (frameworks/av/services/audiopolicy/AudioPolicyManager.cpp)
应用程序APP所用文件:
AudioTrack.java (frameworks/base/media/java/android/media/AudioTrack.java)
android_media_AudioTrack.cpp (frameworks/base/core/jni/android_media_AudioTrack.cpp)
AudioTrack.cpp (frameworks/av/media/libmedia/AudioTrack.cpp)
AudioSystem.cpp (frameworks/av/media/libmedia/AudioSystem.cpp)
音频文件框架图:
4.4 AudioPolicyService启动过程分析
a. 加载解析/vendor/etc/audio_policy.conf或/system/etc/audio_policy.conf
对于配置文件里的每一个module项, new HwModule(name), 放入mHwModules数组
对于module里的每一个output, new IOProfile, 放入module的mOutputProfiles
对于module里的每一个input, new IOProfile, 放入module的mInputProfiles
b. 根据module的name加载厂家提供的so文件 (通过AudioFlinger来加载)
c. 打开对应的output (通过AudioFlinger来open output)
问: 默认声卡是? 声卡/有耳机孔/喇叭,如何告知Andrdoi系统?
由厂家决定,用一个配置文件申明
AndroidPolicyService:
a. 读取,解析配置文件
b. AndroidPolicyService根据配置文件,调用AudioFlinger来打开output,创建线程
总结: 对于audio_policy,conf 里的每一个module:
使用loadHwModule来处理
a. new HwModule(名字"primary")
b. mOutputProfiles: 每一项对应于output的profile
c. mInputProfiles: 每一项对应一个Input的prifile
AudioPolicyManagerBase.cpp (z:android-5.0.2hardwarelibhardware_legacyaudio)
// ----------------------------------------------------------------------------
// 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++) {
// 找到frameworksavservicesaudioflingerAudioFlinger.cpploadHwModule
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
}
Audio_policy_conf.h (z:android-5.0.2hardwarelibhardware_legacyincludehardware_legacy)
#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
查找上面目录:
/vendor/etc/audio_policy.conf
#
# Audio policy configuration for generic device builds (goldfish audio HAL - emulator)
#
# Global configuration section: lists input and output devices always present on the device
# as well as the output device selected by default.
# Devices are designated by a string that corresponds to the enum in audio.h
global_configuration {
attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
default_output_device AUDIO_DEVICE_OUT_SPEAKER
attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC
}
# audio hardware module section: contains descriptors for all audio hw modules present on the
# device. Each hw module node is named after the corresponding hw module library base name.
# For instance, "primary" corresponds to audio.primary..so.
# The "primary" module is mandatory and must include at least one output with
# AUDIO_OUTPUT_FLAG_PRIMARY flag.
# Each module descriptor contains one or more output profile descriptors and zero or more
# input profile descriptors. Each profile lists all the parameters supported by a given output
# or input stream category.
# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding
# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "n".
audio_hw_modules {
primary { // 一个module对应厂家提供的一个so文件
outputs { // 一个module可以有多个output
primary { // 一个output里,表明它的参数
sampling_rates 44100
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL
flags AUDIO_OUTPUT_FLAG_PRIMARY // 默认设备
}
}
inputs { 一个module可以有多个input
primary {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET|AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_AUX_DIGITAL|AUDIO_DEVICE_IN_VOICE_CALL
}
}
}
}
AudioPolicyManagerBase.cpp (z:android-5.0.2hardwarelibhardware_legacyaudio)
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 %sn", path);
return NO_ERROR;
}
4.5 AudioFlinger启动过程分析
笔记分析:
a. 注册AudioFlinger服务
b. 被AudioPolicyService调用以打开厂家提供的so文件
b.1 加载哪个so文件? 文件名是什么? 文件名从何而来?
名字从/system/etc/audio_policy.conf得到 : primary
所以so文件就是 : audio.primary.XXX.so, eg. audio.primary.tiny4412.so
b.2 该so文件由什么源文件组成? 查看Android.mk
audio.primary.$(TARGET_DEVICE) : device/friendly-arm/common/libaudio/AudioHardware.cpp
libhardware_legacy
libhardware_legacy : hardware/libhardware_legacy/audio/audio_hw_hal.cpp
/work/android-5.0.2/device/friendly-arm/common/libaudio
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
LOCAL_SRC_FILES:=
AudioHardware.cpp
LOCAL_MODULE := audio.primary.$(TARGET_DEVICE)
LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
LOCAL_STATIC_LIBRARIES:= libmedia_helper
LOCAL_SHARED_LIBRARIES:=
libutils
liblog
libhardware_legacy
libtinyalsa
libaudioutils
LOCAL_WHOLE_STATIC_LIBRARIES := libaudiohw_legacy
LOCAL_MODULE_TAGS := optional
LOCAL_SHARED_LIBRARIES += libdl
LOCAL_C_INCLUDES +=
external/tinyalsa/include
system/media/audio_effects/include
system/media/audio_utils/include
device/friendly-arm/$(TARGET_DEVICE)/conf
ifeq ($(strip $(BOARD_USES_I2S_AUDIO)),true)
LOCAL_CFLAGS += -DUSES_I2S_AUDIO
endif
ifeq ($(strip $(BOARD_USES_PCM_AUDIO)),true)
LOCAL_CFLAGS += -DUSES_PCM_AUDIO
endif
ifeq ($(strip $(BOARD_USES_SPDIF_AUDIO)),true)
LOCAL_CFLAGS += -DUSES_SPDIF_AUDIO
endif
ifeq ($(strip $(USE_ULP_AUDIO)),true)
LOCAL_CFLAGS += -DUSE_ULP_AUDIO
endif
include $(BUILD_SHARED_LIBRARY)
include $(CLEAR_VARS)
LOCAL_SRC_FILES := AudioPolicyManager.cpp
LOCAL_SHARED_LIBRARIES := libcutils libutils
LOCAL_STATIC_LIBRARIES := libmedia_helper
LOCAL_WHOLE_STATIC_LIBRARIES := libaudiopolicy_legacy
LOCAL_MODULE := audio_policy.$(TARGET_DEVICE)
LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
LOCAL_MODULE_TAGS := optional
ifeq ($(BOARD_HAVE_BLUETOOTH),true)
LOCAL_CFLAGS += -DWITH_A2DP
endif
include $(BUILD_SHARED_LIBRARY)
/work/android-5.0.2/hardware/libhardware_legacy/audio
# 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_MODULE_TAGS := optional
LOCAL_STATIC_LIBRARIES := libmedia_helper
LOCAL_CFLAGS := -Wno-unused-parameter
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_STATIC_LIBRARIES := libmedia_helper
LOCAL_MODULE := libaudiopolicy_legacy
LOCAL_MODULE_TAGS := optional
LOCAL_CFLAGS += -Wno-unused-parameter
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
libutils
liblog
LOCAL_STATIC_LIBRARIES :=
libmedia_helper
LOCAL_WHOLE_STATIC_LIBRARIES :=
libaudiopolicy_legacy
LOCAL_MODULE := audio_policy.default
LOCAL_MODULE_RELATIVE_PATH := hw
LOCAL_MODULE_TAGS := optional
LOCAL_CFLAGS := -Wno-unused-parameter
include $(BUILD_SHARED_LIBRARY)
#ifeq ($(ENABLE_AUDIO_DUMP),true)
# LOCAL_SRC_FILES += AudioDumpInterface.cpp
# LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP
#endif
#
#ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
# LOCAL_CFLAGS += -D GENERIC_AUDIO
#endif
#ifeq ($(BOARD_HAVE_BLUETOOTH),true)
# LOCAL_SRC_FILES += A2dpAudioInterface.cpp
# LOCAL_SHARED_LIBRARIES += liba2dp
# LOCAL_C_INCLUDES += $(call include-path-for, bluez)
#
# LOCAL_CFLAGS +=
# -DWITH_BLUETOOTH
#endif
#
#include $(BUILD_SHARED_LIBRARY)
# AudioHardwareGeneric.cpp
# AudioHardwareStub.cpp
b.3 对硬件的封装:
AudioFlinger : AudioHwDevice (放入mAudioHwDevs数组中)
audio_hw_hal.cpp : audio_hw_device
厂家 : AudioHardware (派生自: AudioHardwareInterface)
AudioHwDevice是对audio_hw_device的封装,
audio_hw_device中函数的实现要通过AudioHardware类对象
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
if (name == NULL) {
return 0;
}
if (!settingsAllowed()) {
return 0;
}
Mutex::Autolock _l(mLock);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
audio_hw_device_t *dev;
int rc = load_audio_interface(name, &dev);
if (rc) {
ALOGI("loadHwModule() error %d loading module %s ", rc, name);
return 0;
}
......
return handle;
}
static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
int rc;
//if_name : primary
//AUDIO_HARDWARE_MODULE_ID : "audio"
// audio.promary.XXX.so
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
rc =audio_hw_device_open(mod, dev);
ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
rc = BAD_VALUE;
goto out;
}
return 0;
out:
*dev = NULL;
return rc;
}
hardware.c F:android_projectandroid_system_codehardwarelibhardware
127|shell@tiny4412:/ $ getprop "ro.hardware" // 查找属性值
tiny4412
static const char *variant_keys[] = {
"ro.hardware", /* This goes first so that it can pick up a different
file on the emulator. */
"ro.product.board",
"ro.board.platform",
"ro.arch"
};
int hw_get_module_by_class(const char *class_id, const char *inst,
const struct hw_module_t **module)
{
int i;
char prop[PATH_MAX];
char path[PATH_MAX];
char name[PATH_MAX];
char prop_name[PATH_MAX];
if (inst)
snprintf(name, PATH_MAX, "%s.%s", class_id, inst);
else
strlcpy(name, class_id, PATH_MAX);
/*
* Here we rely on the fact that calling dlopen multiple times on
* the same .so will simply increment a refcount (and not load
* a new copy of the library).
* We also assume that dlopen() is thread-safe.
*/
/* First try a property specific to the class and possibly instance */
snprintf(prop_name, sizeof(prop_name), "ro.hardware.%s", name);
if (property_get(prop_name, prop, NULL) > 0) {
if (hw_module_exists(path, sizeof(path), name, prop) == 0) {
goto found;
}
}
/* Loop through the configuration variants looking for a module */
for (i=0 ; i
if (property_get(variant_keys[i], prop, NULL) == 0) {
continue;
}
if (hw_module_exists(path, sizeof(path), name, prop) == 0) {
goto found;
}
}
/* Nothing found, try the default */
if (hw_module_exists(path, sizeof(path), name, "default") == 0) {
goto found;
}
return -ENOENT;
found:
/* load the module, if this fails, we're doomed, and we should not try
* to load a different variant. */
return load(class_id, path, module);
}
hardwarelibhardware_legacyaudioaudio_hw_hal.cpp
static int legacy_adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct legacy_audio_device *ladev;
int ret;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
ladev = (struct legacy_audio_device *)calloc(1, sizeof(*ladev));
if (!ladev)
return -ENOMEM;
ladev->device.common.tag= HARDWARE_DEVICE_TAG;
ladev->device.common.version= AUDIO_DEVICE_API_VERSION_2_0;
ladev->device.common.module= const_cast(module);
ladev->device.common.close= legacy_adev_close;
ladev->device.init_check = adev_init_check;
ladev->device.set_voice_volume = adev_set_voice_volume;
ladev->device.set_master_volume = adev_set_master_volume;
ladev->device.get_master_volume = adev_get_master_volume;
ladev->device.set_mode = adev_set_mode;
ladev->device.set_mic_mute = adev_set_mic_mute;
ladev->device.get_mic_mute = adev_get_mic_mute;
ladev->device.set_parameters = adev_set_parameters;
ladev->device.get_parameters = adev_get_parameters;
ladev->device.get_input_buffer_size = adev_get_input_buffer_size;
ladev->device.open_output_stream = adev_open_output_stream;
ladev->device.close_output_stream = adev_close_output_stream;
ladev->device.open_input_stream = adev_open_input_stream;
ladev->device.close_input_stream = adev_close_input_stream;
ladev->device.dump = adev_dump;
ladev->hwif = createAudioHardware();
if (!ladev->hwif) {
ret = -EIO;
goto err_create_audio_hw;
}
*device = &ladev->device.common;
return 0;
err_create_audio_hw:
free(ladev);
return ret;
}
c. 被AudioPolicyService调用来open output、创建playback thread
hardwarelibhardware_legacyaudioAudioPolicyManagerBase.cpp
AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
{
......
// 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);
}
}
}
}
......
}
frameworksavservicesaudioflingerAudioFlinger.cpp
status_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
module,
(devices != NULL) ? *devices : 0,
config->sample_rate,
config->format,
config->channel_mask,
flags);
if (*devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
// 创建一个播放线程
sp thread = openOutput_l(module, output, config, *devices, address, flags);
if (thread != 0) {
*latencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
// the first primary output opened designates the primary hw device
if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d has the primary audio interface", module);
mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
mHardwareStatus = AUDIO_HW_IDLE;
mPrimaryOutputSampleRate = config->sample_rate;
}
return NO_ERROR;
}
return NO_INIT;
}
// ----------------------------------------------------------------------------
sp AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
if (outHwDev == NULL) {
return 0;
}
audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId();
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
audio_stream_out_t *outStream = NULL;
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
//config->format = AUDIO_FORMAT_PCM_FLOAT;
//config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
//config->format = AUDIO_FORMAT_PCM_32_BIT;
//config->format = AUDIO_FORMAT_PCM_8_24_BIT;
// ALOGV("openOutput_l() upgrading format to %#08x", config->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
//config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
//config->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
status_t status = hwDevHal->open_output_stream(hwDevHal,
*output,
devices,
flags,
config,
&outStream,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, "
"channelMask %#x, status %d",
outStream,
config->sample_rate,
config->format,
config->channel_mask,
status);
if (status == NO_ERROR && outStream != NULL) {
AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags);
PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, devices);
ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, devices);
ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
} else {
// 创建线程
thread = new MixerThread(this, outputStream, *output, devices);
ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
}
// 将线程添加到数组里
mPlaybackThreads.add(*output, thread);
return thread;
}
return 0;
}
总结:
打开厂家提供的so文件
由AudioFlinger实现:
1> 从mHwModule取出每一个HwModule,根据name打开so文件
2> 构造硬件封装对象
AudioFlinger:
3. 打开module里的output,创建playbackThread:
由AudioFlinger实现: 对每一个Module中每一个output profile:
4 把outputDesc放入AudioPolicyManager
.mOutputs表示"已经打开的output"
以后可以根据一个整数(output)找到对应的outputDesc
硬件封装:
AudioFlinger.cpp : AudioHwDevice ==> 对应一个Module(so文件)所支持的设备
|
audio_hw_hal.cpp: audio_hw_device
|
AudioHardware.cpp : AudioHardWare
4.6 AudioTrack创建过程概述
a. 体验测试程序: frameworks/base/media/tests/audiotests/shared_mem_test.cpp
frameworks/base/media/tests/mediaframeworktest/src/com/android/mediaframeworktest/functional/audio/MediaAudioTrackTest.java
public void testSetStereoVolumeMax() throws Exception {
// constants for test
final String TEST_NAME = "testSetStereoVolumeMax";
final int TEST_SR = 22050;
final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
final int TEST_MODE = AudioTrack.MODE_STREAM;
final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
//-------- initialization --------------
int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);
AudioTrack track = new AudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF, TEST_FORMAT,
minBuffSize, TEST_MODE);
byte data[] = new byte[minBuffSize/2];
//-------- test --------------
track.write(data, 0, data.length);
track.write(data, 0, data.length);
track.play();
float maxVol = AudioTrack.getMaxVolume();
assertTrue(TEST_NAME, track.setStereoVolume(maxVol, maxVol) == AudioTrack.SUCCESS);
//-------- tear down --------------
track.release();
}
frameworksbasemediajavaandroidmediaAudioTrack.java
/**
* Class constructor with {@link AudioAttributes} and {@link AudioFormat}.
* @param attributes a non-null {@link AudioAttributes} instance.
* @param format a non-null {@link AudioFormat} instance describing the format of the data
* that will be played through this AudioTrack. See {@link AudioFormat.Builder} for
* configuring the audio format parameters such as encoding, channel mask and sample rate.
* @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read
* from for playback. If using the AudioTrack in streaming mode, you can write data into
* this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
* this is the maximum size of the sound that will be played for this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}.
* @param sessionId ID of audio session the AudioTrack must be attached to, or
* {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction
* time. See also {@link AudioManager#generateAudioSessionId()} to obtain a session ID before
* construction.
* @throws IllegalArgumentException
*/
public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId)
throws IllegalArgumentException {
// mState already == STATE_UNINITIALIZED
if (attributes == null) {
throw new IllegalArgumentException("Illegal null AudioAttributes");
}
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat");
}
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
int rate = 0;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE) != 0)
{
rate = format.getSampleRate();
} else {
rate = AudioSystem.getPrimaryOutputSamplingRate();
if (rate <= 0) {
rate = 44100;
}
}
int channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0)
{
channelMask = format.getChannelMask();
}
int encoding = AudioFormat.ENCODING_DEFAULT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
encoding = format.getEncoding();
}
audioParamCheck(rate, channelMask, encoding, mode);
mStreamType = AudioSystem.STREAM_DEFAULT;
audioBuffSizeCheck(bufferSizeInBytes);
mInitializationLooper = looper;
IBinder b = ServiceManager.getService(Context.APP_OPS_SERVICE);
mAppOps = IAppOpsService.Stub.asInterface(b);
mAttributes = (new AudioAttributes.Builder(attributes).build());
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference(this), mAttributes,
mSampleRate, mChannels, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session);
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing AudioTrack.");
return; // with mState == STATE_UNINITIALIZED
}
mSessionId = session[0];
if (mDataLoadMode == MODE_STATIC) {
mState = STATE_NO_STATIC_DATA;
} else {
mState = STATE_INITIALIZED;
}
}
frameworksavmedialibmediaAudioTrack.cpp
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
uint32_t notificationFrames,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes);
}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
uint32_t notificationFrames,
const sp& sharedBuffer,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %u, sessionId %d, transferType %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
sessionId, transferType);
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
transferType = TRANSFER_SHARED;
} else if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
return BAD_VALUE;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("Invalid stream type %d", streamType);
return BAD_VALUE;
}
setAttributesFromStreamType(streamType);
mStreamType = streamType;
} else {
if (!isValidAttributes(pAttributes)) {
ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
pAttributes->usage, pAttributes->content_type, pAttributes->flags,
pAttributes->tags);
}
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
setStreamTypeFromAttributes(mAttributes);
ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
}
status_t status;
if (sampleRate == 0) {
status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
if (status != NO_ERROR) {
ALOGE("Could not get output sample rate for stream type %d; status %d",
mStreamType, status);
return status;
}
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
ALOGE("8-bit data in shared memory is not supported");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|| !audio_is_linear_pcm(format)) {
ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
? "Offload request, forcing to Direct Output"
: "Not linear PCM, forcing to Direct Output");
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
if (mStreamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
mFrameSizeAF = mFrameSize;
} else {
ALOG_ASSERT(audio_is_linear_pcm(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
mFrameSizeAF = channelCount * audio_bytes_per_sample(
format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
if (offloadInfo != NULL) {
mOffloadInfoCopy = *offloadInfo;
mOffloadInfo = &mOffloadInfoCopy;
} else {
mOffloadInfo = NULL;
}
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
mSessionId = sessionId;
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingpid != mypid)) {
mClientPid = callingpid;
} else {
mClientPid = pid;
}
mAuxEffectId = 0;
mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
// create the IAudioTrack
status = createTrack_l();
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mState = STATE_STOPPED;
mUserData = user;
mLoopPeriod = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mServer = 0;
mPosition = 0;
mReleased = 0;
mStartUs = 0;
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
return NO_ERROR;
}
播放声音时都要创建AudioTrack对象,
java的AudioTrack对象创建时会导致c++的AudioTrack对象被创建;
所以分析的核心是c++的AudioTrack类,
创建AudioTrack时涉及一个重要函数: set
b. 猜测创建过程的主要工作
b.1 使用AudioTrack的属性, 根据AudioPolicy找到对应的output、playbackThread
b.2 在playbackThread中创建对应的track
b.3 APP的AudioTrack 和 playbackThread的mTracks中的track之间建立共享内存
c. 源码时序图
4.7 AudioPolicyManager堪误与回顾
frameworksavservicesaudiopolicyAudioPolicyService.cpp
void AudioPolicyService::onFirstRef()
{
char value[PROPERTY_VALUE_MAX];
const struct hw_module_t *module;
int forced_val;
int rc;
{
Mutex::Autolock _l(mLock);
// start tone playback thread
mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
// start audio commands thread
mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
// start output activity command thread
mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
#ifdef USE_LEGACY_AUDIO_POLICY // 使用旧的的策略
ALOGI("AudioPolicyService CSTOR in legacy mode");
/* instantiate the audio policy manager */
rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
if (rc) {
return;
}
rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc));
if (rc) {
return;
}
rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this,
&mpAudioPolicy);
ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc));
if (rc) {
return;
}
rc = mpAudioPolicy->init_check(mpAudioPolicy);
ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc));
if (rc) {
return;
}
ALOGI("Loaded audio policy from %s (%s)", module->name, module->id);
#else
ALOGI("AudioPolicyService CSTOR in new mode");
mAudioPolicyClient = new AudioPolicyClient(this);
mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient);
#endif
}
// load audio processing modules
spaudioPolicyEffects = new AudioPolicyEffects();
{
Mutex::Autolock _l(mLock);
mAudioPolicyEffects = audioPolicyEffects;
}
}
AudioPolicyFactory.cpp (z:android-5.0.2frameworksavservicesaudiopolicy)
extern "C" AudioPolicyInterface* createAudioPolicyManager(
AudioPolicyClientInterface *clientInterface)
{
returnnew AudioPolicyManager(clientInterface);
}
AudioPolicyManager.cpp (z:android-5.0.2frameworksavservicesaudiopolicy)
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
Thread(false),
#endif //AUDIO_POLICY_TEST
mPrimaryOutput((audio_io_handle_t)0),
mPhoneState(AUDIO_MODE_NORMAL),
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
mA2dpSuspended(false),
mSpeakerDrcEnabled(false), mNextUniqueId(1),
mAudioPortGeneration(1)
{
mUidCached = getuid();
mpClientInterface = clientInterface;
for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
}
mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
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();
}
}
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// must be done after reading the policy
initializeVolumeCurves();
// open all output streams needed to access attached devices
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
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.
// This also validates mAvailableOutputDevices list
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
const sp outProfile = mHwModules[i]->mOutputProfiles[j];
if (outProfile->mSupportedDevices.isEmpty()) {
ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
continue;
}
if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
audio_devices_t profileType = outProfile->mSupportedDevices.types();
if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) {
profileType = mDefaultOutputDevice->mDeviceType;
} else {
// chose first device present in mSupportedDevices also part of
// outputDeviceTypes
for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
profileType = outProfile->mSupportedDevices[k]->mDeviceType;
if ((profileType & outputDeviceTypes) != 0) {
break;
}
}
}
if ((profileType & outputDeviceTypes) == 0) {
continue;
}
sp outputDesc = new AudioOutputDescriptor(outProfile);
outputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
mHwModules[i]->mName);
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
ssize_t index =
mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
mAvailableOutputDevices[index]->mId = nextUniqueId();
mAvailableOutputDevices[index]->mModule = mHwModules[i];
}
}
if (mPrimaryOutput == 0 &&
outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = output;
}
addOutput(output, outputDesc);
setOutputDevice(output,
outputDesc->mDevice,
true);
}
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
const sp inProfile = mHwModules[i]->mInputProfiles[j];
if (inProfile->mSupportedDevices.isEmpty()) {
ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
continue;
}
// chose first device present in mSupportedDevices also part of
// inputDeviceTypes
audio_devices_t profileType = AUDIO_DEVICE_NONE;
for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
profileType = inProfile->mSupportedDevices[k]->mDeviceType;
if (profileType & inputDeviceTypes) {
break;
}
}
if ((profileType & inputDeviceTypes) == 0) {
continue;
}
sp inputDesc = new AudioInputDescriptor(inProfile);
inputDesc->mInputSource = AUDIO_SOURCE_MIC;
inputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = inputDesc->mSamplingRate;
config.channel_mask = inputDesc->mChannelMask;
config.format = inputDesc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
&input,
&config,
&inputDesc->mDevice,
String8(""),
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE);
if (status == NO_ERROR) {
for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
ssize_t index =
mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
mAvailableInputDevices[index]->mId = nextUniqueId();
mAvailableInputDevices[index]->mModule = mHwModules[i];
}
}
mpClientInterface->closeInput(input);
} else {
ALOGW("Cannot open input stream for device %08x on hw module %s",
inputDesc->mDevice,
mHwModules[i]->mName);
}
}
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
if (mAvailableOutputDevices[i]->mId == 0) {
ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
}
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
if (mAvailableInputDevices[i]->mId == 0) {
ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
}
i++;
}
// make sure default device is reachable
if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
}
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 = AUDIO_FORMAT_PCM_16_BIT;
mTestChannels = AUDIO_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
}
void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc)
{
outputDesc->mIoHandle = output;
outputDesc->mId = nextUniqueId();
mOutputs.add(output, outputDesc);
nextAudioPortGeneration();
}
4.8 AudioTrack创建过程_选择output
a. APP构造AudioTrack时指定了 stream type
b. AudioTrack::setAttributesFromStreamType
c. AudioPolicyManager::getStrategyForAttr
d. AudioPolicyManager::getDeviceForStrategy
e. AudioPolicyManager::getOutputForDevice
e.1 AudioPolicyManager::getOutputsForDevice
e.2 output = selectOutput(outputs, flags, format);
AudioTrack.cpp (z:android-5.0.2frameworksavmedialibmedia)
void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) {
mAttributes.flags = 0x0;
switch(streamType) {
case AUDIO_STREAM_DEFAULT:
case AUDIO_STREAM_MUSIC:
mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
mAttributes.usage = AUDIO_USAGE_MEDIA;
break;
case AUDIO_STREAM_VOICE_CALL:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
break;
case AUDIO_STREAM_ENFORCED_AUDIBLE:
mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
// intended fall through, attributes in common with STREAM_SYSTEM
case AUDIO_STREAM_SYSTEM:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
break;
case AUDIO_STREAM_RING:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
break;
case AUDIO_STREAM_ALARM:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
mAttributes.usage = AUDIO_USAGE_ALARM;
break;
case AUDIO_STREAM_NOTIFICATION:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
mAttributes.usage = AUDIO_USAGE_NOTIFICATION;
break;
case AUDIO_STREAM_BLUETOOTH_SCO:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
mAttributes.flags |= AUDIO_FLAG_SCO;
break;
case AUDIO_STREAM_DTMF:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
break;
case AUDIO_STREAM_TTS:
mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
break;
default:
ALOGE("invalid stream type %d when converting to attributes", streamType);
}
}
AudioPolicyManager.h (z:android-5.0.2frameworksavservicesaudiopolicy)
uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
// flags to strategy mapping
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
}
// usage to strategy mapping
switch (attr->usage) {
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
return (uint32_t) STRATEGY_MEDIA;
case AUDIO_USAGE_VOICE_COMMUNICATION:
return (uint32_t) STRATEGY_PHONE;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
return (uint32_t) STRATEGY_DTMF;
case AUDIO_USAGE_ALARM:
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
return (uint32_t) STRATEGY_SONIFICATION;
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
case AUDIO_USAGE_UNKNOWN:
default:
return (uint32_t) STRATEGY_MEDIA;
}
}
uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
// flags to strategy mapping
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
}
// usage to strategy mapping
switch (attr->usage) {
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
return (uint32_t) STRATEGY_MEDIA;
case AUDIO_USAGE_VOICE_COMMUNICATION:
return (uint32_t) STRATEGY_PHONE;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
return (uint32_t) STRATEGY_DTMF;
case AUDIO_USAGE_ALARM:
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
return (uint32_t) STRATEGY_SONIFICATION;
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
case AUDIO_USAGE_UNKNOWN:
default:
return (uint32_t) STRATEGY_MEDIA;
}
}
audio_devices_t AudioPolicyManager::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];
}
audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
switch (strategy) {
case STRATEGY_SONIFICATION_RESPECTFUL:
if (isInCall()) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
} else if (isStreamActiveRemotely(AUDIO_STREAM_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*/);
//user "safe" speaker if available instead of normal speaker to avoid triggering
//other acoustic safety mechanisms for notification
if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
} else if (isStreamActive(AUDIO_STREAM_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*/);
//user "safe" speaker if available instead of normal speaker to avoid triggering
//other acoustic safety mechanisms for notification
if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
}
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:
// Force use of only devices on primary output if:
// - in call AND
// - cannot route from voice call RX OR
// - audio HAL version is < 3.0 and TX device is on the primary HW module
if (mPhoneState == AUDIO_MODE_IN_CALL) {
audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
if (((mAvailableInputDevices.types() &
AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
(((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) &&
(hwOutputDesc->getAudioPort()->mModule->mHalVersion <
AUDIO_DEVICE_API_VERSION_3_0))) {
availableOutputDeviceTypes = availablePrimaryOutputDevices();
}
}
// for phone strategy, we first consider the forced use and then the available devices by order
// of priority
switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
case AUDIO_POLICY_FORCE_BT_SCO:
if (!isInCall() || strategy != STRATEGY_DTMF) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & 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 (!isInCall() &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
if (device) break;
if (mPhoneState != AUDIO_MODE_IN_CALL) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
if (device == AUDIO_DEVICE_NONE) {
ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
}
break;
case AUDIO_POLICY_FORCE_SPEAKER:
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
// A2DP speaker when forcing to speaker output
if (!isInCall() &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
if (device) break;
}
if (mPhoneState != AUDIO_MODE_IN_CALL) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
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[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
device = availableOutputDeviceTypes & 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 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
if ((device2 == AUDIO_DEVICE_NONE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
}
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
}
if ((device2 == AUDIO_DEVICE_NONE)) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
}
if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
// no sonification on aux digital (e.g. HDMI)
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
}
if ((device2 == AUDIO_DEVICE_NONE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
}
int device3 = AUDIO_DEVICE_NONE;
if (strategy == STRATEGY_MEDIA) {
// ARC, SPDIF and AUX_LINE can co-exist with others.
device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF);
device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE);
}
device2 |= device3;
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
// STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
device |= device2;
// If hdmi system audio mode is on, remove speaker out of output list.
if ((strategy == STRATEGY_MEDIA) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
device &= ~AUDIO_DEVICE_OUT_SPEAKER;
}
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
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;
}
AudioPolicyManager.h (z:android-5.0.2frameworksavservicesaudiopolicy)
audio_io_handle_t AudioPolicyManager::getOutputForDevice(
audio_devices_t device,
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
uint32_t latency = 0;
status_t status;
#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");
sp outputDesc = new AudioOutputDescriptor(NULL);
outputDesc->mDevice = mTestDevice;
outputDesc->mLatency = mTestLatencyMs;
outputDesc->mFlags =
(audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
outputDesc->mRefCount[stream] = 0;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = mTestSamplingRate;
config.channel_mask = mTestChannels;
config.format = mTestFormat;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(0,
&mTestOutputs[mCurOutput],
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status == NO_ERROR) {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mFormat = config.format;
outputDesc->mChannelMask = config.channel_mask;
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 = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
sp profile;
// skip direct output selection if the request can obviously be attached to a mixed output
// and not explicitly requested
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
audio_channel_count_from_out_mask(channelMask) <= 2) {
goto non_direct_output;
}
// 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 (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
!isNonOffloadableEffectEnabled()) {
profile = getProfileForDirectOutput(device,
samplingRate,
format,
channelMask,
(audio_output_flags_t)flags);
}
if (profile != 0) {
sp outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp 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->mIoHandle);
}
outputDesc = new AudioOutputDescriptor(profile);
outputDesc->mDevice = device;
outputDesc->mLatency = 0;
outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = samplingRate;
config.channel_mask = channelMask;
config.format = format;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(profile->mModule->mHandle,
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(samplingRate != 0 && samplingRate != config.sample_rate) ||
(format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
(channelMask != 0 && channelMask != config.channel_mask)) {
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 != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeOutput(output);
}
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
outputDesc->mRefCount[stream] = 0;
outputDesc->mStopTime[stream] = 0;
outputDesc->mDirectOpenCount = 1;
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);
mpClientInterface->onAudioPortListUpdate();
return output;
}
non_direct_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 outputs = getOutputsForDevice(device, mOutputs);
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
output = selectOutput(outputs, flags, format);
}
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;
}
SortedVector AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
DefaultKeyedVector > openOutputs)
{
SortedVector 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;
}
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