- 1. 背景
- 2. init启动AudioServer进程
- 3. AudioFlinger::instantiate
- 4. AudioPolicyService::instantiate
- 5. 小结
1. 背景
为了系统学习Audio系统，首先从Audio的初始化流程开始学习，重点在分析通路的走向，如何加载音频策略，音频Hal库等方面展开,平台为Android P。 相关源码可以查看:https://android.googlesource.com/platform/frameworks/av/
整个通路如下图所示:
2. init启动AudioServer进程
当系统启动时，init进程会初始化Native的重要服务，包括java世界的入口zygote，显示方面的SurfaceFlinger，内存相关的管理lmkd，media相关的mediaserver，都是由init直接启动，而音频相关的AudioFlinger，AudioPolicyService也是由init启动的，只不过统一在AudioServer这个进程当中。
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//frameworks/av/media/audioserver/audioserver.rc
service audioserver /system/bin/audioserver
class core #audioserver为core类型的服务
user audioserver #user为audioserver,对应userid为1041
group audio camera drmrpc inet media mediadrm net_bt net_bt_admin net_bw_acct #group组
...
onrestart restart vendor.audio-hal-2-0 #重启audioserver时，需要重启vendor.audio-hal-2-0以及audio-hal-2-0
onrestart restart audio-hal-2-0
audioserver的入口在frameworks/av/media/audioserver/main_audioserver.cpp中，对比以往，audioFlinger/audioPolicyService被放入了mediaServer进程。
另外在看代码的过程，在启动AudioFlinger之前，还涉及到一个media.log的进程。当设置了ro.test_harness时，将会开启该进程。引入该进程的目的是由于在加入调试的android.util.Log(java),ALOGx(native)的调试打印可能会产生大量的垃圾打印，甚至会使关键日志因溢出而丢失。另一方面，还会影响依赖时效性(AudioFlinger中的FastMixer/FastCapture)。media.log则能够有效的解决该问题，并具有如下优势：
- 除非需要，否则它不会在主日志中产生垃圾内容。
- 即使在 mediaserver 崩溃或中断时，也可以对其进行检查。
- 在每个时间轴均是非阻塞的。
- 对性能的干扰较小（当然完全不会产生干扰的日志是不存在的）
以往的旧架构如下：
添加了media.log的新架构：
media.log会在共享内存中的环形缓存区保存日志，上层需要使用新的API(NBLOG)去代替以往的log。环形的缓存区能够保证任何内存损坏都不会导致media.log崩溃，并且media.log中的mediaLogService能够随时对其进行转储。
更多详情可以查看Android官网上的链接： https://source.android.google.cn/devices/audio/debugging#mediaLog
抛开media.log后，main_audioserver.cpp中会初始化音频的重要native服务:
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//frameworks/av/media/audioserver/main_audioserver.cpp
android::hardware::configureRpcThreadpool(4, false /*callerWillJoin*/);
sp<ProcessState> proc(ProcessState::self());
sp<IServiceManager> sm = defaultServiceManager();
//初始化AudioFlinger与AudioPolicyService
AudioFlinger::instantiate();
AudioPolicyService::instantiate();
//AAudioService是8.0推出的，通过mmap支持的HAL层和驱动结合，能够缩短延迟时间的服务。
aaudio_policy_t mmapPolicy = property_get_int32(AAUDIO_PROP_MMAP_POLICY,
AAUDIO_POLICY_NEVER);
if (mmapPolicy == AAUDIO_POLICY_AUTO || mmapPolicy == AAUDIO_POLICY_ALWAYS) {
AAudioService::instantiate();
}
//SoundTrigger是语音识别服务
SoundTriggerHwService::instantiate();
//监听Binder是否有服务与AudioServer通信
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
AAudio提供了一个低延迟数据路径。在“专有”模式下，该功能可让客户端应用代码直接写入到与 ALSA 驱动程序共享的内存映射缓冲区。在“共享”模式下，MMAP 缓冲区由在 AudioServer 中运行的混音器使用。在“专有”模式下，由于数据会绕过混音器，延迟时间会明显缩短。 关于AAudioService的内容可以查看:
https://source.android.google.cn/devices/audio/aaudio)
3. AudioFlinger::instantiate
AudioFlinger首先通过instantiate方法在audiosever中进行初始化。其继承了BinderService，可用于在Servicemanager中对服务AudioFlinger进行注册。
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//framworks/native/libs/binder/include/binder/BinderService.h
template<typename SERVICE>
class BinderService
{
public:
static status_t publish(bool allowIsolated = false,
int dumpFlags = IServiceManager::DUMP_FLAG_PRIORITY_DEFAULT) {
sp<IServiceManager> sm(defaultServiceManager());
//新建服务SERVICE，并将服务加到ServiceManager中。
return sm->addService(String16(SERVICE::getServiceName()), new SERVICE(), allowIsolated,
dumpFlags);
}
static void instantiate() { publish(); }
...
};
getServiceName需要在服务(AudioFlinger)中实现，AudioFlinger的服务名为”media.audio_flinger”。也可以在adb中通过service list查询当前系统的服务包括哪些。也可以通过dumpsys media.audio_flinger将AudioFlinger的关键信息打印出来。
当instantiate将服务注册后，可转到服务的实现分析，由于AudioFlinger继承了RefBase类，所以能够被智能指针引用，在publish中的addService时，参数二实质是被sp引用，所以AudioFlinger会在构造方法之后调用onFirstRef方法。
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//frameworks/av/services/audioflinger/AudioFlinger.cpp
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mMediaLogNotifier(new AudioFlinger::MediaLogNotifier()),
mPrimaryHardwareDev(NULL),
mAudioHwDevs(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mTotalMemory(0),
mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
mGlobalEffectEnableTime(0),
mSystemReady(false)
{
for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
}
getpid_cached = getpid();
//检查ro.test_harness属性，即之前说的media.log是否启用了
const bool doLog = property_get_bool("ro.test_harness", false);
if (doLog) {
//新建了大小为400K的MemoryDealer。本质是申请了匿名内存ashmem
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
//初始化sMediaLogInit，即获取media.log的引用sMediaLogService
(void) pthread_once(&sMediaLogOnce, sMediaLogInit);
}
//假如AudioFlinger崩溃，电池的状态可能会有问题，因此在初始化重设状态。
BatteryNotifier::getInstance().noteResetAudio();
//跟hidl通信相关的服务
mDevicesFactoryHal = DevicesFactoryHalInterface::create();
mEffectsFactoryHal = EffectsFactoryHalInterface::create();
//新建了一个MediaLogNotifier的线程，但是需要配合media.log服务运行，假如media.log没启动，该线程退出
mMediaLogNotifier->run("MediaLogNotifier");
//这部分代码和AudioFlinger调试相关,当打开了TEE_SINK时，系统播放音频时，会保留音频内容为pcm数据保存在/data/misc/audioserver目录里
#ifdef TEE_SINK
char value[PROPERTY_VALUE_MAX];
(void) property_get("ro.debuggable", value, "0");
int debuggable = atoi(value);
int teeEnabled = 0;
if (debuggable) {
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
if (teeEnabled & 1) {
mTeeSinkInputEnabled = true;
}
if (teeEnabled & 2) {
mTeeSinkOutputEnabled = true;
}
if (teeEnabled & 4) {
mTeeSinkTrackEnabled = true;
}
#endif
}
tee的调试功能还是相当有用，可以将播放的音频通过audioflinger转存为pcm数据,具体的操作也可以参照:
https://source.android.google.cn/devices/audio/debugging
此后，由于AudioFlinger被强引用指向，调用onFirstRef方法：
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//frameworks/av/services/audioflinger/AudioFlinger.cpp
void AudioFlinger::onFirstRef()
{
Mutex::Autolock _l(mLock);
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
//设置了ro.audio.flinger_standbytime_ms时间
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
} else {
//默认的standby时间为kDefaultStandbyTimeInNsecs，即3s。
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
}
}
mPatchPanel = new PatchPanel(this);
mMode = AUDIO_MODE_NORMAL;
//将自身放在gAudioFlinger的全局变量中,方便客户端调用AudioFlinger服务
gAudioFlinger = this;
}
系统默认在播放完音频后，经过mStandbyTimeInNsecs后会进入StandBy状态。
至于hidl相关，需要在方案里配置，当AudioFlinger创建hidl对象时:
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//frameworks/av/services/audioflinger/AudioFlinger.cpp
mDevicesFactoryHal = DevicesFactoryHalInterface::create();
mEffectsFactoryHal = EffectsFactoryHalInterface::create();
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//frameworks/av/media/libaudiohal/DevicesFactoryHalInterface.cpp
sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
//从高到低获取服务是否存在，来确定使用哪一套方案，本例使用的是2.0,需要在方案中配置。
if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
return new V4_0::DevicesFactoryHalHybrid();
}
if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
return new DevicesFactoryHalHybrid();
}
return nullptr;
}
//frameworks/av/media/libaudiohal/DevicesFactoryHalInterface.cpp
sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
return new V4_0::EffectsFactoryHalHidl();
}
if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
return new EffectsFactoryHalHidl();
}
return nullptr;
}
方案中可以通过如下方式选择的方案选择编译:
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PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
...
Hidl的工作将在后续继续分析。至此，AudioFlinger初始化完成。进入AudioPolicyService的初始化。
4. AudioPolicyService::instantiate
4.1 AudioPolicyService::onFirstRef
AudioPolicyService在服务中注册为“media.audio_policy”，其instantiate初始化与AudioFlinger如出一辙，因此可以直接到其构造函数进行分析:
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//frameworks/av/services/audiopolicy/service/AudioPolicyService.cpp
AudioPolicyService::AudioPolicyService()
: BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL),
mAudioPolicyManager(NULL), mAudioPolicyClient(NULL), mPhoneState(AUDIO_MODE_INVALID)
{
}
可以看出其构造方法仅仅通过初始化列表对成员变量进行了初始化，因此直接到onFirstRef进行分析:
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./services/audiopolicy/service/AudioPolicyService.cpp
void AudioPolicyService::onFirstRef()
{
{
Mutex::Autolock _l(mLock);
//AudioPolicyService创建了3个AudioCommandThread线程，
//用于铃声播放以及发送音频配置命令到AudioFlinger中(step 1)
mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
//新建AudioPolicyClient对象，并传入到createAudioPolicyManger接口(step 2)
mAudioPolicyClient = new AudioPolicyClient(this);
//新建出AduioPolicyManager对象。(step 3)
mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient);
}
sp<AudioPolicyEffects>audioPolicyEffects = new AudioPolicyEffects();
{
Mutex::Autolock _l(mLock);
mAudioPolicyEffects = audioPolicyEffects;
}
mUidPolicy = new UidPolicy(this);
mUidPolicy->registerSelf();
}
step 1: AudioCommandThread在threadLoop循环处理消息，处理的消息类型包括:
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START_TONE,
STOP_TONE,
SET_VOLUME,
SET_PARAMETERS,
SET_VOICE_VOLUME,
STOP_OUTPUT,
RELEASE_OUTPUT,
CREATE_AUDIO_PATCH,
RELEASE_AUDIO_PATCH,
UPDATE_AUDIOPORT_LIST,
UPDATE_AUDIOPATCH_LIST,
SET_AUDIOPORT_CONFIG,
DYN_POLICY_MIX_STATE_UPDATE,
RECORDING_CONFIGURATION_UPDATE
为了让AudioCommandThread处理消息，通过sendCommmand将command填充到mAudioCommands中，并随后通过mWaitWorkCV.signal()唤醒线程,线程首先通过mWaitWorkCV.wait(mLock)进行等待，当被唤醒后，就会开始处理，最终将给到AudioFlinger处理.
step 2: AudioPolicyClient继承了AudioPolicyClientInterface，基类AudioPolicyClientInterface定义在/hardware/libhardware_legacy/include/hardware_legacy/AudioPolicyInterface.h文件中，定义了hal层的重要接口,如下是部分接口展示:
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///hardware/libhardware_legacy/include/hardware_legacy/AudioPolicyInterface.h
//加载音频设备的描述文件audio_policy.conf
virtual audio_module_handle_t loadHwModule(const char *name) = 0;
//打开音频输出通道
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 = NULL) = 0;
//关闭音频输出通道
virtual status_t closeOutput(audio_io_handle_t output) = 0;
...
//打开音频输入通道
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) = 0;
virtual status_t closeInput(audio_io_handle_t input) = 0;
//设置音频音量
virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
...
AudioPolicyClient作为子类，需要实现这些方法，但实际的操作不是它来完成，而是AudioFlinger，如loadHwModule的实现:
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//frameworks/av/services/audiopolicy/service/AudioPolicyService.cpp
audio_module_handle_t AudioPolicyService::AudioPolicyClient::loadHwModule(const char *name)
{
//通过get_audio_flinger获取AudioFlinger
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
return AUDIO_MODULE_HANDLE_NONE;
}
//最终调用到AudioFlinger的loadHwModule完成
return af->loadHwModule(name);
}
step 3: 回到AudioPolicyService的流程，新建完AudioPolicyClient后作为参数，传到了方法createAudioPolicyManager中,其定义在frameworks/av/services/audiopolicy/manager/AudioPolicyFactory.cpp中。
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//frameworks/av/services/audiopolicy/manager/AudioPolicyFactory.cpp
extern "C" AudioPolicyInterface* createAudioPolicyManager(
AudioPolicyClientInterface *clientInterface)
{
//实质是新建了一个AudioPolicyManager对象。
return new AudioPolicyManager(clientInterface);
}
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//frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface, false /*forTesting*/)
{
loadConfig();
initialize();
}
4.2 AudioPolicyManager::loadConfig
于是问题变成了loadConfig以及initialize中做了什么操作, 先从loadConfig开始,实质通过ConfigParsingUtils的loadConfig分别读取了AUDIO_POLICY_VENDOR_CONFIG_FILE以及AUDIO_POLICY_CONFIG_FILE两个配置文件，即”/system/etc/audio_policy.conf” 以及”/vendor/etc/audio_policy.conf”。
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//frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
void AudioPolicyManager::loadConfig() {
if ((ConfigParsingUtils::loadConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, getConfig()) != NO_ERROR)
&& (ConfigParsingUtils::loadConfig(AUDIO_POLICY_CONFIG_FILE, getConfig()) != NO_ERROR)) {
getConfig().setDefault();
}
}
常规的audio_policy.conf格式如下, 其中配置了当前项目audio策略中支持的输入输出设备，采样率，采样精度，声道数等等:
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//audio_policy.conf
audio_hw_modules {
primary {
outputs {
primary {
sampling_rates 44100
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET
flags AUDIO_OUTPUT_FLAG_PRIMARY
}
}
inputs{
....
}
}
a2dp{
...
}
usb{
...
}
...
}
再来看ConfigParsingUtils的loadConfig的实现:
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//frameworks/av/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
status_t ConfigParsingUtils::loadConfig(const char *path, AudioPolicyConfig &config)
{
cnode *root;
char *data;
//step1. 读取audio_policy.conf的内容，load_file是libcutils的方法.
//将会申请一段内存并将文件内容读取到该内存中.
data = (char *)load_file(path, NULL);
if (data == NULL) {
return -ENODEV;
}
//step2. 创建了根节点cnode类型的root,name与value均设置为空
root = config_node("", "");
//step3. 解析data内容，config_utils实现了一套十分有趣的词法分析器解析配置文件,后续实现配置文件的时候也可以参照audio_policy这段逻辑.
//每个节点都有key/value键值对，每个节点都有指向孩子节点的指针,能够获取下一层的节点内容,也具有next指针可以获取同级的内容。
config_load(root, data);
HwModuleCollection hwModules;
//Step4.加载HwModules设备到HwModules集合中，如primary,a2dp,usb等
loadHwModules(root, hwModules, config);
//Step5.读取primary中的是否有global_configuration的部分
loadGlobalConfig(root, config, hwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
//Step6.设置hwModules
config.setHwModules(hwModules);
config_free(root);
free(root);
free(data);
return NO_ERROR;
}
AudioPolicyService维护着一个如下形式的集合(HwModuleCollection),也可以通过dumpsys media.audio_policy获取细节.同级的子节点之间可以通过next获取到下一个兄弟节点.
再深入到Step4中的loadHwModules看其实现:
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void ConfigParsingUtils::loadHwModules(cnode *root, HwModuleCollection &hwModules,
AudioPolicyConfig &config)
{
//AUDIO_HW_MODULE_TAG即"audio_hw_modules",找到根节点
cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
if (node == NULL) {
return;
}
node = node->first_child;
//为audio_hw_modules下的子节点各创建一个HwModule对象,并加入hwModules集合中
//node包括如primary,a2dp,usb等设备
while (node) {
sp<HwModule> module = new HwModule(node->name);
//调用loadHwModule处理
if (loadHwModule(node, module, config) == NO_ERROR) {
hwModules.add(module);
}
node = node->next;
}
}
status_t ConfigParsingUtils::loadHwModule(cnode *root, sp<HwModule> &module,
AudioPolicyConfig &config)
{
status_t status = NAME_NOT_FOUND;
//DEVICES_TAG即devices字段,遍历找到设备root下的devices
cnode *node = config_find(root, DEVICES_TAG);
if (node != NULL) {
node = node->first_child;
//新建一个DeviceVector类型的deivces,用于保存设备
DeviceVector devices;
while (node) {
//A. 使用loadHWModuleDevice加载设备
status_t tmpStatus = loadHwModuleDevice(node, devices);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
module->setDeclaredDevices(devices);
}
//OUTPUTS_TAG即outputs字段
node = config_find(root, OUTPUTS_TAG);
if (node != NULL) {
node = node->first_child;
while (node) {
//B. 调用loadHwModuleProfile
status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SOURCE);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
}
//OUTPUTS_TAG即intputs字段
node = config_find(root, INPUTS_TAG);
if (node != NULL) {
node = node->first_child;
while (node) {
//C. 调用loadHwModuleProfile
status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SINK);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
}
//D. 加载通用的属性
loadModuleGlobalConfig(root, module, config);
return status;
}
loadHwModule分别去找devices,outputs,inputs字段进行处理,分步进行分析:
A.devices
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status_t tmpStatus = loadHwModuleDevice(node, devices);
module->setDeclaredDevices(devices);
首先展示常规的devices字段下内容:
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//primary中devices配置
primary{
...
devices {
speaker {
type AUDIO_DEVICE_OUT_SPEAKER
gains {
gain_1 {
mode AUDIO_GAIN_MODE_JOINT
min_value_mB -8400
max_value_mB 4000
default_value_mB 0
step_value_mB 100
}
}
}
HDMI {
type AUDIO_DEVICE_OUT_AUX_DIGITAL
}
SPDIF {
type AUDIO_DEVICE_OUT_SPDIF
}
wired_headphone {
type AUDIO_DEVICE_OUT_WIRED_HEADPHONE
}
wired_headset {
type AUDIO_DEVICE_OUT_WIRED_HEADSET
}
BT_sco {
type AUDIO_DEVICE_OUT_BLUETOOTH_SCO
}
BT_sco_headset {
type AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET
}
}
...
}
可以看到devices中都是设备的名称,且都设置了type类型,speaker较为特殊,还设置了gains增益.来看代码是如何解析这段配置的:
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//static
status_t ConfigParsingUtils::loadHwModuleDevice(cnode *root, DeviceVector &devices)
{
cnode *node = root->first_child;
audio_devices_t type = AUDIO_DEVICE_NONE;
while (node) {
if (strcmp(node->name, APM_DEVICE_TYPE) == 0) {
//从这里获取type类型
deviceFromString(node->value, type);
break;
}
node = node->next;
}
if (type == AUDIO_DEVICE_NONE ||
(!audio_is_input_device(type) && !audio_is_output_device(type))) {
//type读取有误则退出
return BAD_VALUE;
}
//DeviceDescriptor用于描述设备的信息
sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(type, String8(root->name));
node = root->first_child;
while (node) {
if (strcmp(node->name, APM_DEVICE_ADDRESS) == 0) {
//处理address字段,设置设备地址
deviceDesc->mAddress = String8((char *)node->value);
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
//处理channel_masks字段
//输入设备
if (audio_is_input_device(type)) {
//新建了一个AudioProfile对象到deviceDesc中,且将channel_mask的值读取出来
deviceDesc->addAudioProfile(
new AudioProfile(gDynamicFormat,
inputChannelMasksFromString(node->value),
SampleRateVector()));
} else {//输出设备
deviceDesc->addAudioProfile(
new AudioProfile(gDynamicFormat,
outputChannelMasksFromString(node->value),
SampleRateVector()));
}
} else if (strcmp(node->name, GAINS_TAG) == 0) {
//处理gains字段
loadDeviceDescriptorGains(node, deviceDesc);
}
node = node->next;
}
//将当前deviceDesc加入devices Vector中
devices.add(deviceDesc);
return NO_ERROR;
}
//将上述处理的devices加入到HwModule类的mPorts中.mPorts类型为AudioPortVector.
void HwModule::setDeclaredDevices(const DeviceVector &devices)
{
mDeclaredDevices = devices;
for (size_t i = 0; i < devices.size(); i++) {
mPorts.add(devices[i]);
}
}
假如HwModules中配置了devices,会通过type以及设备名创建DeviceDescriptor(继承了AudioPort以及AudioPortConfig),并会继续解析,假如设置了如地址,通道,增益等,DeviceDescriptor就会设置相应的属性.最终将DeviceDescriptor加入到mPorts(AudioPort类)中进行管理.
B/C.Outputs and Inputs
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//Outputs
status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SOURCE);
//Inputs
status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SINK);
再来看下Outputs是怎么配置的:
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outputs {
primary {
sampling_rates 48000
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
devices speaker
flags AUDIO_OUTPUT_FLAG_PRIMARY
}
}
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status_t ConfigParsingUtils::loadHwModuleProfile(cnode *root, sp<HwModule> &module,
audio_port_role_t role)
{
cnode *node = root->first_child;
sp<IOProfile> profile = new IOProfile(String8(root->name), role);
AudioProfileVector audioProfiles;
SampleRateVector sampleRates;
ChannelsVector channels;
FormatVector formats;
while (node) {
//解析formats(采样精度)
if (strcmp(node->name, FORMATS_TAG) == 0 &&
strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
formats = formatsFromString(node->value);
//解析sampling_rates(采样率)
} else if (strcmp(node->name, SAMPLING_RATES_TAG) == 0 &&
strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
collectionFromString<SampleRateTraits>(node->value, sampleRates);
//解析channel_mask(通道数)
} else if (strcmp(node->name, CHANNELS_TAG) == 0 &&
strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
if (role == AUDIO_PORT_ROLE_SINK) {
channels = inputChannelMasksFromString(node->value);
} else {
channels = outputChannelMasksFromString(node->value);
}
//解析devices
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
DeviceVector devices;
loadDevicesFromTag(node->value, devices, module->getDeclaredDevices());
profile->setSupportedDevices(devices);
//解析flags
} else if (strcmp(node->name, FLAGS_TAG) == 0) {
if (role == AUDIO_PORT_ROLE_SINK) {
profile->setFlags(InputFlagConverter::maskFromString(node->value));
} else {
profile->setFlags(OutputFlagConverter::maskFromString(node->value));
}
//解析gains
} else if (strcmp(node->name, GAINS_TAG) == 0) {
loadAudioPortGains(node, *profile);
}
node = node->next;
}
//当精度为空时,会创建一个新的默认值为gDynamicFormat的AudioProfile加入到audioProfiles中
if (formats.isEmpty()) {
sp<AudioProfile> profileToAdd = new AudioProfile(gDynamicFormat, channels, sampleRates);
profileToAdd->setDynamicFormat(true);
profileToAdd->setDynamicChannels(channels.isEmpty());
profileToAdd->setDynamicRate(sampleRates.isEmpty());
audioProfiles.add(profileToAdd);
} else {
for (size_t i = 0; i < formats.size(); i++) {
对于每一个采样精度,都会创建一个AudioProfile
sp<AudioProfile> profileToAdd = new AudioProfile(formats[i], channels, sampleRates);
profileToAdd->setDynamicFormat(formats[i] == gDynamicFormat);
profileToAdd->setDynamicChannels(channels.isEmpty());
profileToAdd->setDynamicRate(sampleRates.isEmpty());
audioProfiles.add(profileToAdd);
}
}
//将audioProfiles设置为mProfiles
profile->setAudioProfiles(audioProfiles);
if (profile->hasSupportedDevices()) {
//将输入输出AudioProfile分别加入到mInputProfiles和mOutputProfiles中.同时mPorts会加入输入以及输出的Profile.
return module->addProfile(profile);
}
return BAD_VALUE;
}
B/C步骤分别对HwModules中的inputs,outputs进行解析,首先每个HwModules新建一个IOProfile,接着针对audio_policy.conf的设备树,解析出HwModules的outputs/inputs的采样频率,通道数,采样精度,支持设备等信息,并遍历精度,每个精度创建一个AudioProfile,最后将同一个HwModules中所有的AudioProfile加入到IOProfile中.
D.loadModuleGlobalConfig
除了上述的配置外,audio_policy.conf还可以配置通用类的属性,这类属性既可以在audio_hw_modules平行的地方设置,也可以在HwModules中设置,如:
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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_DEVICE_IN_REMOTE_SUBMIX
}
audio_hw_modules {
primary {
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_hal_version 3.0
}
...
}
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void ConfigParsingUtils::loadModuleGlobalConfig(cnode *root, const sp<HwModule> &module,
AudioPolicyConfig &config)
{
//GLOBAL_CONFIG_TAG表示global_configuration
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
if (node == NULL) {
return;
}
DeviceVector declaredDevices;
if (module != NULL) {
//获取A步骤的DeviceDescriptor
declaredDevices = module->getDeclaredDevices();
}
node = node->first_child;
while (node) {
//获取attached_output_devices
if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
DeviceVector availableOutputDevices;
//读取attached_output_devices的值
loadDevicesFromTag(node->value, availableOutputDevices, declaredDevices);
//增加到mAvailableOutputDevices中
config.addAvailableOutputDevices(availableOutputDevices);
//获取default_output_device
} else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
audio_devices_t device = AUDIO_DEVICE_NONE;
deviceFromString(node->value, device);
if (device != AUDIO_DEVICE_NONE) {
//新建DeviceDescriptor类,并将default_output_device的值设置为mDefaultOutputDevices的值
sp<DeviceDescriptor> defaultOutputDevice = new DeviceDescriptor(device);
config.setDefaultOutputDevice(defaultOutputDevice);
} else {
ALOGW("loadGlobalConfig() default device not specified");
}
//获取attached_input_devices,与Output类同
} else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
DeviceVector availableInputDevices;
loadDevicesFromTag(node->value, availableInputDevices, declaredDevices);
config.addAvailableInputDevices(availableInputDevices);
//获取audio_hal_version
} else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
uint32_t major, minor;
sscanf((char *)node->value, "%u.%u", &major, &minor);
//设置HAL版本
module->setHalVersion(major, minor);
ALOGV("loadGlobalConfig() mHalVersion = major %u minor %u", major, minor);
}
node = node->next;
}
}
通过通用的配置,可以看出,即使A中的devices中不设置,也可以通过通用配置去设置默认的DeviceDescriptor用以描述设备.如对audio_policy.conf感兴趣,可以参考如下链接:
4.3 AudioPolicyManager::initialize
4.3.1 initialize流程
initialize的初始化流程较长，后续将会分步分析,首先涉及到engine的概念，其类图如下：
- 第一阶段:
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//frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
status_t AudioPolicyManager::initialize() {
//初始化音量调节点
mVolumeCurves->initializeVolumeCurves(getConfig().isSpeakerDrcEnabled());
//获取engineInstance实例
audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
if (!engineInstance) {
return NO_INIT;
}
//engineInstance调用queryInterface，实际会调用到Engine类的queryInterface方法，并最终返回了ManagerInterfaceImpl类型(继承于AudioPolicyManagerInterface类)的mEngine.由于ManagerInterfaceImpl中传入了Engine对象，实际执行工作的还是Engine．
mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
if (mEngine == NULL) {
return NO_INIT;
}
//将AudioPolicyManager设置到Engine中的mApmObserver
mEngine->setObserver(this);
//检查mApmObserver是否为空
status_t status = mEngine->initCheck();
if (status != NO_ERROR) {
return status;
}
...
第一阶段engine首先进行了必要的初始化，还将AudioManager设为它的成员变量，最后检查是否为空。且engine实现了AudioPolicyInterface的必要接口，通过ManagerInterfaceImpl作为中转，但实际的实现仍然为engine。
- 第二阶段:
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//frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
//status_t AudioPolicyManager::initialize() {
...
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
//mHwModulesAll在初始化时被传入了AudioPolicyConfig中，所以当loadConfig完毕时，mHwModulesAll中就保存了audio_policy.conf中的音频设备结构了。
for (const auto& hwModule : mHwModulesAll) {
//Step 1. 遍历并调用loadHwModule加载。
//loadHwModule返回的audio_module_handle_t，设置到hwModule的mHandle中。
hwModule->setHandle(mpClientInterface->loadHwModule(hwModule->getName()));
if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
ALOGW("could not open HW module %s", hwModule->getName());
continue;
}
//AudioPolicyManager使用Vector类型的mHwModules重新将HwModule加入集合当中。
mHwModules.push_back(hwModule);
//OutputProfile即之前根据配置基于不同formats(精度)生成的AudioProfile
for (const auto& outProfile : hwModule->getOutputProfiles()) {
if (!outProfile->canOpenNewIo()) {
ALOGE("Invalid Output profile max open count %u for profile %s",
outProfile->maxOpenCount, outProfile->getTagName().c_str());
continue;
}
//Output中必须要有devices字段
if (!outProfile->hasSupportedDevices()) {
ALOGW("Output profile contains no device on module %s", hwModule->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
mTtsOutputAvailable = true;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
audio_devices_t profileType = outProfile->getSupportedDevicesType();
if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
profileType = mDefaultOutputDevice->type();
} else {
profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes);
}
if ((profileType & outputDeviceTypes) == 0) {
continue;
}
//Step 2.每一个AudioProfile会新建一个SwAudioOutputDescriptor
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
const DeviceVector &devicesForType = supportedDevices.getDevicesFromType(profileType);
String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress
: String8("");
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
//Step 3.调用open打开设备
status_t status = outputDesc->open(nullptr, profileType, address,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
hwModule->getName());
} else {
for (const auto& dev : supportedDevices) {
ssize_t index = mAvailableOutputDevices.indexOf(dev);
if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
//调用了AudioPort的attach方法(将成员变量mModule设置为hwModule),并调用getNextUniqueId分配Id
mAvailableOutputDevices[index]->attach(hwModule);
}
}
if (mPrimaryOutput == 0 &&
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
//Step 4. 将output(handle值)以及outputDesc作为一组键值对,加入到mOutputs中.
addOutput(output, outputDesc);
//Step 5. 设置输出设备
setOutputDevice(outputDesc,
profileType,
true,
0,
NULL,
address);
}
}
//输入设备的处理和output类同,暂不重复分析
....
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
if (!mAvailableOutputDevices[i]->isAttached()) {
ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type());
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
}
//调用setDeviceConnectionState
mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
if (!mAvailableInputDevices[i]->isAttached()) {
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
}
mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
// make sure default device is reachable
if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
status = NO_INIT;
}
// If microphones address is empty, set it according to device type
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->mAddress.isEmpty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
mAvailableInputDevices[i]->mAddress = String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS);
} else if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BACK_MIC) {
mAvailableInputDevices[i]->mAddress = String8(AUDIO_BACK_MICROPHONE_ADDRESS);
}
}
}
if (mPrimaryOutput == 0) {
ALOGE("Failed to open primary output");
status = NO_INIT;
}
updateDevicesAndOutputs();
return status;
}
4.3.2 loadHwModule_l
mpClientInterface的loadHwModule最终使用到AudioFlinger的loadHwModule:
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//frameworks/av/services/audioflinger/AudioFlinger.cpp
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
....
Mutex::Autolock _l(mLock);
return loadHwModule_l(name);
}
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
//检查mAudioHwDevs是否已经加载过这个设备了？有则从mAudioHwDevs这个DefaultKeyedVector
//中获取，其形式为key/value形式，key为audio_module_handle_t(handle),value为AudioHwDevice。
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
return mAudioHwDevs.keyAt(i);
}
}
sp<DeviceHalInterface> dev;
//mDevicesFactoryHal为DevicesFactoryHalHybrid,是之前调用DevicesFactoryHalInterface::create创建的.
//调用构造函数时会创建两个对象,分别为DevicesFactoryHalLocal以及DevicesFactoryHalHidl
//假如打开的设备不为a2dp或者hearing, 使用hidl经过binder形式去调用HAL,
//即使用DevicesFactoryHalHidl的openDevice接口;其余的以直通方式调用,调用的是DevicesFactoryHalLocal的openDevice接口。返回的dev为继承了DeviceHalInterface的对象。
int rc = mDevicesFactoryHal->openDevice(name, &dev);
if (rc) {
return AUDIO_MODULE_HANDLE_NONE;
}
//后面是一系列HAL初始化操作。
mHardwareStatus = AUDIO_HW_INIT;
//初始化audio Interface是否成功。
rc = dev->initCheck();
...
//获取下一个有效ID,保存为audio_module_handle_t类型
audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
//新建一个AudioHwDevices，加入到mAudioHwdevs集合中管理。
mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
return handle;
}
//frameworks/av/media/libaudiohal/DevicesFactoryHalInterface.cpp
DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
: mLocalFactory(new DevicesFactoryHalLocal()),
mHidlFactory(new DevicesFactoryHalHidl()) {
}
status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
return mHidlFactory->openDevice(name, device);
}
//a2dp,hearing使用直通,其余使用hidl
return mLocalFactory->openDevice(name, device);
}
mDevicesFactoryHal调用openDevice即最终是加载设备对应的库文件,而Android支持的音频设备包括(还有部分未展示):
| AUDIO_DEVICE_OUT_EARPIECE | 听筒 |
| AUDIO_DEVICE_OUT_SPEAKER | 喇叭 |
| AUDIO_DEVICE_OUT_WIRED_HEADSET | 带麦克风的耳机 |
| AUDIO_DEVICE_OUT_WIRED_HEADPHONE | 耳机 |
| AUDIO_DEVICE_OUT_BLUETOOTH_SCO | SCO蓝牙 |
| AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | SCO蓝牙带麦克风耳机 |
| AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT | SCO蓝牙车载套件 |
| AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | A2DP蓝牙 |
| AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | A2DP蓝牙耳机 |
| AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | A2DP喇叭 |
| AUDIO_DEVICE_OUT_AUX_DIGITAL | AUX |
| AUDIO_DEVICE_OUT_HDMI | HDMI |
| AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET | 模拟DOCK耳机 |
| AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | 数字DOCK耳机 |
| AUDIO_DEVICE_OUT_USB_ACCESSORY | USB配件 |
| AUDIO_DEVICE_OUT_USB_DEVICE | USB设备 |
| AUDIO_DEVICE_OUT_REMOTE_SUBMIX | SubMix设备 |
| AUDIO_DEVICE_OUT_TELEPHONY_TX | 电话 |
| … | … |
回到之前提及的loadHwModule_l,其中涉及到关键的hidl通信，从8.0开始，上层需要调用HAL层就需要借助hidl进行通信了。如下图所示, 在音频方面可见，使用了第一种Legacy Hal和第三种binderized模式。
如需要了解hidl相关内容，可以参考如下链接: Android O HIDL的实现对接
1.LegacyHal模式
以openDevice为例，当为primary,usb时采用的是LegacyHal。
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//frameworks/av/media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp
status_t DevicesFactoryHalLocal::openDevice(const char *name, sp<DeviceHalInterface> *device) {
audio_hw_device_t *dev;
//加载指定的audio interface，即加载所需要的库文件，如audio.primary.xxx.so
status_t rc = load_audio_interface(name, &dev);
if (rc == OK) {
//DeviceHalLocal封装了一层，是audio_hw_device_t的proxy端
*device = new DeviceHalLocal(dev);
}
return rc;
}
static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
int rc;
//hw_getmodule_by_class通过if_name参数在/system/lib/hw，/vendor/lib/hw，/odm/lib/hw找相关的库文件。
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
...
//对接到HAL层open接口
rc = audio_hw_device_open(mod, dev);
...
//失败时关闭设备
...
audio_hw_device_close(*dev);
return OK;
out:
*dev = NULL;
return rc;
}
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//hardware/libhardware/include/hardware/audio.h
static inline int audio_hw_device_open(const struct hw_module_t* module,
struct audio_hw_device** device)
{
//这里即调用到HAL层的真正实现,至此与HAL层正式对接上。
return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
TO_HW_DEVICE_T_OPEN(device));
}
//编译为audio.primary.$(TARGET_BOARD_PLATFORM).so
static struct hw_module_methods_t hal_module_methods = {
//最终调用到HAL层的adev_open接口，具体实现与厂商实现相关。
.open = adev_open,
};
2.Binderized模式
介绍binderized模式时，首先介绍相关的接口: IDevicesFactory在IDevicesFactory.hal文件定义了openDevice接口。编译时会通过hidl-gen生成对应的头文件IDevicesFactory.h和实现文件DevicesFactoryAll.cpp
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//hardware/interfaces/audio/2.0/IDevicesFactory.hal
interface IDevicesFactory {
typedef android.hardware.audio@2.0::Result Result;
enum Device : int32_t {
PRIMARY,
A2DP,
USB,
R_SUBMIX,
STUB
};
//定义的接口
openDevice(Device device) generates (Result retval, IDevice result);
};
经过hidl-gen生成的IDevicesFactory.h头文件
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//out/soong/.intermediates/hardware/interfaces/audio/2.0/android.hardware.audio@2.0_genc++_headers/gen/android/hardware/audio/2.0/IDevicesFactory.h
//hal文件经过hidl转换后，都会继承IBase类,IBase继承了RefBase，可以经过智能指针管理生命周期。
struct IDevicesFactory : public ::android::hidl::base::V1_0::IBase {
...
enum class Device : int32_t {
PRIMARY = 0,
A2DP = 1, // (::android::hardware::audio::V2_0::IDevicesFactory::Device.PRIMARY implicitly + 1)
USB = 2, // (::android::hardware::audio::V2_0::IDevicesFactory::Device.A2DP implicitly + 1)
R_SUBMIX = 3, // (::android::hardware::audio::V2_0::IDevicesFactory::Device.USB implicitly + 1)
STUB = 4, // (::android::hardware::audio::V2_0::IDevicesFactory::Device.R_SUBMIX implicitly + 1)
};
//定义了getservice
...
//声明openDevice为虚函数
virtual ::android::hardware::Return<void> openDevice(::android::hardware::audio::V2_0::IDevicesFactory::Device device, openDevice_cb _hidl_cb) = 0;
}
经过hidl-gen生成的DevicesFactoryAll.cpp实现文件，实现了getService，openDevice方法。回头看下AudioFlinger，当构造方法调用时，会调用DevicesFactoryHalInterface::create()时创建一个DevicesFactoryHalHidl对象.它的方式与Binder通信一致，先获取服务,返回对应的BpHwDevicesFactory对象，用以调用服务的接口(HAL接口),如openDevice。
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//frameworks/av/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp
//构造函数
DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
//mDevicesFactory即BpHwDevicesFactory,为proxy端
mDevicesFactory = IDevicesFactory::getService();
if (mDevicesFactory != 0) {
//注册死亡代理通知
mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
} else {
exit(1);
}
...
}
深入看getService的实现，最终会调用到ServiceManagement.cpp中的getRawServiceInternal方法，并判断是使用了hidl通信的binderized还是passthrough模式。
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//out/soong/.intermediates/hardware/interfaces/audio/2.0/android.hardware.audio@2.0_genc++/gen/android/hardware/audio/2.0/DevicesFactoryAll.cpp
::android::sp<IDevicesFactory> IDevicesFactory::getService(const std::string &serviceName, const bool getStub) {
//调用了getServiceInternal,泛型类别是BpHwDevicesFactory,应当是Proxy端
return ::android::hardware::details::getServiceInternal<BpHwDevicesFactory>(serviceName, true, getStub);
}
//system/libhidl/transport/include/hidl/HidlTransportSupport.h
template <typename BpType, typename IType = typename BpType::Pure,
typename = std::enable_if_t<std::is_same<i_tag, typename IType::_hidl_tag>::value>,
typename = std::enable_if_t<std::is_same<bphw_tag, typename BpType::_hidl_tag>::value>>
sp<IType> getServiceInternal(const std::string& instance, bool retry, bool getStub) {
using ::android::hidl::base::V1_0::IBase;
//IBase与IBinder类似
sp<IBase> base = getRawServiceInternal(IType::descriptor, instance, retry, getStub);
if (base == nullptr) {
return nullptr;
}
//假如为远程服务，返回代理端
if (base->isRemote()) {
//从这里看，返回的是BpType泛型类别，即BpHwDevicesFactory。
//传入参数是IBinder类型。
return sp<IType>(new BpType(toBinder<IBase>(base)));
}
//与Binder的interfacecast类似
return IType::castFrom(base);
}
//system/libhidl/transport/ServiceManagement.cpp
sp<::android::hidl::base::V1_0::IBase> getRawServiceInternal(const std::string& descriptor,
const std::string& instance,
bool retry, bool getStub) {
...
sp<Waiter> waiter;
//获取hidl的ServiceManager,打开/dev/hwbinder节点
const sp<IServiceManager1_1> sm = defaultServiceManager1_1();
...
Return<Transport> transportRet = sm->getTransport(descriptor, instance);
...
Transport transport = transportRet;
//使用的传输模式是HWBINDER还是PASSTHROUGH?
const bool vintfHwbinder = (transport == Transport::HWBINDER);
const bool vintfPassthru = (transport == Transport::PASSTHROUGH);
...
//Legacy模式一般为false
const bool vintfLegacy = false;
...
//尝试一定次数去获得IBase对象，仅限于HwBinder以及Legacy模式
for (int tries = 0; !getStub && (vintfHwbinder || vintfLegacy); tries++) {
if (waiter == nullptr && tries > 0) {
//创建了Waiter对象，用于等待
waiter = new Waiter(descriptor, instance, sm);
}
if (waiter != nullptr) {
waiter->reset();
}
//根据descriptor获取IBase对象。IBase通过封装能转化成Bp类。
Return<sp<IBase>> ret = sm->get(descriptor, instance);
//假如返回结果不为ok，则跳出循环，走PASSTHROUGH逻辑
if (!ret.isOk()) {
break;
}
sp<IBase> base = ret;
if (base != nullptr) {
//检验是否可以进行转换
Return<bool> canCastRet =
details::canCastInterface(base.get(), descriptor.c_str(), true /* emitError */);
//可以转换，则waiter调用done完成工作，返回base对象
if (canCastRet.isOk() && canCastRet) {
if (waiter != nullptr) {
waiter->done();
}
return base; // still needs to be wrapped by Bp class.
}
if (!handleCastError(canCastRet, descriptor, instance)) break;
}
if (vintfLegacy || !retry) break;
//base为空，waiter不为空时，等待
if (waiter != nullptr) {
waiter->wait();
}
}
if (waiter != nullptr) {
waiter->done();
}
if (getStub || vintfPassthru || vintfLegacy) {
//假如是直通模式，获取直通模式的ServiceManager
const sp<IServiceManager> pm = getPassthroughServiceManager();
if (pm != nullptr) {
sp<IBase> base = pm->get(descriptor, instance).withDefault(nullptr);
if (!getStub || trebleTestingOverride) {
base = wrapPassthrough(base);
}
return base;
}
}
return nullptr;
}
当获取到了serviceManager后,即可通过binder方式去调用服务,最后通过binderized调用openDevice:
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status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
if (mDevicesFactory == 0) return NO_INIT;
IDevicesFactory::Device hidlDevice;
//通过name，如primary，a2dp调用相应的静态hidlDevice对象。
status_t status = nameFromHal(name, &hidlDevice);
if (status != OK) return status;
Result retval = Result::NOT_INITIALIZED;
//此处使用了c++ lambda，mDevicesFactory为BpHwDevicesFactory。
//参数2为回调方法,当openDevice动作成功时，会新建一个DeviceHalHidl的对象，由device指向。
Return<void> ret = mDevicesFactory->openDevice(
hidlDevice,
[&](Result r, const sp<IDevice>& result) {
retval = r;
if (retval == Result::OK) {
*device = new DeviceHalHidl(result);
}
});
//错误处理
....
}
//out/soong/.intermediates/hardware/interfaces/audio/2.0/android.hardware.audio@2.0_genc++/gen/android/hardware/audio/2.0/DevicesFactoryAll.cpp
::android::hardware::Return<void> BpHwDevicesFactory::openDevice(::android::hardware::audio::V2_0::IDevicesFactory::Device device, openDevice_cb _hidl_cb){
//调用代理端的BpHwDevicesFactory::_hidl_openDevice方法，经过Binder进行通信。
::android::hardware::Return<void> _hidl_out = ::android::hardware::audio::V2_0::BpHwDevicesFactory::_hidl_openDevice(this, this, device, _hidl_cb);
return _hidl_out;
}
Binderized运行openDevice会创建DeviceHalHidl，Local时则会创建DeviceHalLocal。接下来看_hidl_openDevice的实现:
BpHwDevicesFactory::_hidl_openDevice为代理端实现:
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//out/soong/.intermediates/hardware/interfaces/audio/2.0/android.hardware.audio@2.0_genc++/gen/android/hardware/audio/2.0/DevicesFactoryAll.cpp
::android::hardware::Return<void> BpHwDevicesFactory::_hidl_openDevice(::android::hardware::IInterface *_hidl_this, ::android::hardware::details::HidlInstrumentor *_hidl_this_instrumentor, ::android::hardware::audio::V2_0::IDevicesFactory::Device device, openDevice_cb _hidl_cb) {
...
(void) _hidl_this_instrumentor;
if (_hidl_cb == nullptr) {
return ::android::hardware::Status::fromExceptionCode(
::android::hardware::Status::EX_ILLEGAL_ARGUMENT,
"Null synchronous callback passed.");
}
::android::hardware::Parcel _hidl_data;
::android::hardware::Parcel _hidl_reply;
::android::status_t _hidl_err;
::android::hardware::Status _hidl_status;
::android::hardware::audio::V2_0::Result _hidl_out_retval;
::android::sp<::android::hardware::audio::V2_0::IDevice> _hidl_out_result;
//writeInterface，descriptor为"android.hardware.audio@2.0::IDevicesFactory"
_hidl_err = _hidl_data.writeInterfaceToken(BpHwDevicesFactory::descriptor);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
_hidl_err = _hidl_data.writeInt32((int32_t)device);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
//通过_hidl_this发起transact通信，1为openDevice协议，_hidl_reply为返回的信息
_hidl_err = ::android::hardware::IInterface::asBinder(_hidl_this)->transact(1 /* openDevice */, _hidl_data, &_hidl_reply);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
//读取_hidl_reply的信息包括_hidl_status,_hidl_out_retrval
_hidl_err = ::android::hardware::readFromParcel(&_hidl_status, _hidl_reply);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
if (!_hidl_status.isOk()) { return _hidl_status; }
_hidl_err = _hidl_reply.readInt32((int32_t *)&_hidl_out_retval);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
{
::android::sp<::android::hardware::IBinder> _hidl_binder;
_hidl_err = _hidl_reply.readNullableStrongBinder(&_hidl_binder);
if (_hidl_err != ::android::OK) { goto _hidl_error; }
_hidl_out_result = ::android::hardware::fromBinder<::android::hardware::audio::V2_0::IDevice,::android::hardware::audio::V2_0::BpHwDevice,::android::hardware::audio::V2_0::BnHwDevice>(_hidl_binder);
}
//调用回调函数，假如值_hidl_out_result为Result::OK，则新建DeviceHalHidl
_hidl_cb(_hidl_out_retval, _hidl_out_result);
_hidl_status.setFromStatusT(_hidl_err);
return ::android::hardware::Return<void>();
_hidl_error:
_hidl_status.setFromStatusT(_hidl_err);
return ::android::hardware::Return<void>(_hidl_status);
}
BnHwDevicesFactory::_hidl_openDevice为对应的服务端实现:
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//out/soong/.intermediates/hardware/interfaces/audio/2.0/android.hardware.audio@2.0_genc++/gen/android/hardware/audio/2.0/DevicesFactoryAll.cpp
::android::status_t BnHwDevicesFactory::_hidl_openDevice(
::android::hidl::base::V1_0::BnHwBase* _hidl_this,
const ::android::hardware::Parcel &_hidl_data,
::android::hardware::Parcel *_hidl_reply,
TransactCallback _hidl_cb) {
::android::status_t _hidl_err = ::android::OK;
//校验descriptor
if (!_hidl_data.enforceInterface(BnHwDevicesFactory::Pure::descriptor)) {
_hidl_err = ::android::BAD_TYPE;
return _hidl_err;
}
::android::hardware::audio::V2_0::IDevicesFactory::Device device;
//从请求服务的进程的_hidl_data Parcel包中读取device
_hidl_err = _hidl_data.readInt32((int32_t *)&device);
if (_hidl_err != ::android::OK) { return _hidl_err; }
bool _hidl_callbackCalled = false;
//调用HAL层openDevice
static_cast<IDevicesFactory*>(_hidl_this->getImpl().get())->openDevice(device, [&](const auto &_hidl_out_retval, const auto &_hidl_out_result) {
if (_hidl_callbackCalled) {
LOG_ALWAYS_FATAL("openDevice: _hidl_cb called a second time, but must be called once.");
}
_hidl_callbackCalled = true;
//将状态返回到_hidl_reply,使得上层得到回复
::android::hardware::writeToParcel(::android::hardware::Status::ok(), _hidl_reply);
_hidl_err = _hidl_reply->writeInt32((int32_t)_hidl_out_retval);
/* _hidl_err ignored! */
if (_hidl_out_result == nullptr) {
_hidl_err = _hidl_reply->writeStrongBinder(nullptr);
} else {
::android::sp<::android::hardware::IBinder> _hidl_binder = ::android::hardware::toBinder<
::android::hardware::audio::V2_0::IDevice>(_hidl_out_result);
if (_hidl_binder.get() != nullptr) {
//返回了_hidl_binder
_hidl_err = _hidl_reply->writeStrongBinder(_hidl_binder);
} else {
_hidl_err = ::android::UNKNOWN_ERROR;
}
}
//回调方法
_hidl_cb(*_hidl_reply);
});
if (!_hidl_callbackCalled) {
LOG_ALWAYS_FATAL("openDevice: _hidl_cb not called, but must be called once.");
}
return _hidl_err;
}
至此,研究了上层是如何调用到HAL层的adev_open接口.
4.3.3 SwAudioOutputDescriptor && open
从initialize的流程可得,遍历AudioProfile并对每一个新建一个相应的SwAudioOutputDescriptor(继承于AudioOutputDescriptor).当调用open时:
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status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
audio_devices_t device,
const String8& address,
audio_stream_type_t stream,
audio_output_flags_t flags,
audio_io_handle_t *output)
{
//本例中config为空,会新建一个lConfig,同时为采样率.通道数,采样精度赋值
audio_config_t lConfig;
if (config == nullptr) {
lConfig = AUDIO_CONFIG_INITIALIZER;
lConfig.sample_rate = mSamplingRate;
lConfig.channel_mask = mChannelMask;
lConfig.format = mFormat;
} else {
lConfig = *config;
}
mDevice = device;
//假如选定的AudioProfile是offload模式,且没有指定offload info,则会创建一个默认的offload_info
if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
lConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
lConfig.offload_info = AUDIO_INFO_INITIALIZER;
lConfig.offload_info.sample_rate = lConfig.sample_rate;
lConfig.offload_info.channel_mask = lConfig.channel_mask;
lConfig.offload_info.format = lConfig.format;
lConfig.offload_info.stream_type = stream;
lConfig.offload_info.duration_us = -1;
lConfig.offload_info.has_video = true; // conservative
lConfig.offload_info.is_streaming = true; // likely
}
//mFlags置位
mFlags = (audio_output_flags_t)(mFlags | flags);
//调用AudioFlinger的openOutput
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
&lConfig,
&mDevice,
address,
&mLatency,
mFlags);
if (status == NO_ERROR) {
mSamplingRate = lConfig.sample_rate;
mChannelMask = lConfig.channel_mask;
mFormat = lConfig.format;
mId = AudioPort::getNextUniqueId();
mIoHandle = *output;
mProfile->curOpenCount++;
}
return status;
}
mClientInterface的类型为AudioPolicyClient,其实现如下:
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status_t AudioPolicyService::AudioPolicyClient::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)
{
//获取audioFlinger的proxy端
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
return PERMISSION_DENIED;
}
return af->openOutput(module, output, config, devices, address, latencyMs, flags);
}
暂且忽略AudioFlinger的Binder通信过程,直接看AudioFlinger的实现:
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//frameworks/av/services/audioflinger/AudioFlinger.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)
{
if (devices == NULL || *devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
//openOutput_l
sp<ThreadBase> thread = openOutput_l(module, output, config, *devices, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
*latencyMs = playbackThread->latency();
//通知clients进程output已经创建完毕了
playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
return NO_ERROR;
}
return NO_INIT;
}
sp<AudioFlinger::ThreadBase> 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)
{
//Step 1.查找对应的audio interface
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
if (outHwDev == NULL) {
return 0;
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
} else {
ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
return 0;
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
//.....
AudioStreamOut *outputStream = NULL;
//Step 2.为设备打开一个输出流,
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
devices,
flags,
config,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
//Step 3.创建播放线程
if (status == NO_ERROR) {
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
sp<MmapPlaybackThread> thread =
new MmapPlaybackThread(this, *output, outHwDev, outputStream,
devices, AUDIO_DEVICE_NONE, mSystemReady);
mMmapThreads.add(*output, thread);
return thread;
} else {
sp<PlaybackThread> thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
//直接输出,不需要混音
thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
} else {
//需要混音线程
thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
}
//加入到mPlaybackThreads播放线程中
mPlaybackThreads.add(*output, thread);
return thread;
}
}
return 0;
}
4.3.3.1 findSuitableHwDev_l
findSuitableHwDev_l是为了找到适合的audio interface:
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AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t devices)
{
if (module == 0) {
//假如module为0,会对所有audio interfaces调用loadHwModule_l
for (size_t i = 0; i < arraysize(audio_interfaces); i++) {
loadHwModule_l(audio_interfaces[i]);
}
//遍历所有设备,并尝试找可支持的设备
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
sp<DeviceHalInterface> dev = audioHwDevice->hwDevice();
uint32_t supportedDevices;
//找到了可支持的设备
if (dev->getSupportedDevices(&supportedDevices) == OK &&
(supportedDevices & devices) == devices) {
return audioHwDevice;
}
}
} else {
//一般会提供不为0的module,直接通过键值对获取
//在之前AudioPolicyMananger中的initialize时,就会调用loadHwModule并对mAudioHwDevs赋值
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
4.3.3.2 openOutputStream
openOutputStream是为了给设备打开输出流:
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status_t AudioHwDevice::openOutputStream(
AudioStreamOut **ppStreamOut,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
const char *address)
{
struct audio_config originalConfig = *config;
AudioStreamOut *outputStream = new AudioStreamOut(this, flags);
status_t status = outputStream->open(handle, devices, config, address);
//....
*ppStreamOut = outputStream;
return status;
}
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//frameworks/av/services/audioflinger/AudioStreamOut.cpp
status_t AudioStreamOut::open(
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
const char *address)
{
sp<StreamOutHalInterface> outStream;
audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
? (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
: flags;
int status = hwDev()->openOutputStream(
handle,
devices,
customFlags,
config,
address,
&outStream);
if (status != NO_ERROR && config->format == AUDIO_FORMAT_IEC61937) {
struct audio_config customConfig = *config;
customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
status = hwDev()->openOutputStream(
handle,
devices,
customFlags,
&customConfig,
address,
&outStream);
}
if (status == NO_ERROR) {
stream = outStream;
mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
status = stream->getFrameSize(&mHalFrameSize);
}
return status;
}
至于openOutputStream的实现,会根据设备类型,选择调用的是hidl还是LegacyHal,最终会到厂商HAL层的adev_open_output_stream方法.主要目的即创建了音频输出流,并返回了outputStream
4.3.3.3 PlayBackThread
PlayBackThread是基类,这里暂且只分析DirectOutputThread以及MixerThread,给出线程的类图:
A. PlayBackThread
为了更好的了解DirectOutputThread以及MixerThread,先要了解PlayBackThread的实现,首先自然好奇其命名,可以查看AudioFlinger的线程,可输入如下命令:
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ps -T | grep -i audioserver
此时展示出所有与audioserver相关的进程与线程:
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#由于audioserver关键字的进程有两个,首先剔除第一行,只显示进程名的进程,再获取进程号,并赋值到ps -T显示其下所有线程
ps -T `ps -A |awk -F " " '{for(i=2;i<=NF;i++)printf("%s ",$i);printf "\n"}'| grep -i audioserver| awk '{print $1}'`
显示的结果如下:
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USER PID TID PPID VSZ RSS WCHAN ADDR S CMD
audioserver 1783 1783 1 54100 16856 binder_thread_read 0 S audioserver
audioserver 1783 1895 1 54100 16856 binder_thread_read 0 S HwBinder:1783_1
audioserver 1783 1899 1 54100 16856 futex_wait_queue_me 0 S ApmTone
audioserver 1783 1900 1 54100 16856 futex_wait_queue_me 0 S ApmAudio
audioserver 1783 1901 1 54100 16856 futex_wait_queue_me 0 S ApmOutput
audioserver 1783 1903 1 54100 16856 binder_thread_read 0 S HwBinder:1783_2
audioserver 1783 1906 1 54100 16856 binder_thread_read 0 S Binder:1783_1
audioserver 1783 1907 1 54100 16856 binder_thread_read 0 S Binder:1783_2
audioserver 1783 1947 1 54100 16856 futex_wait_queue_me 0 S AudioOut_D
audioserver 1783 2062 1 54100 16856 futex_wait_queue_me 0 S soundTrigger cb
audioserver 1783 2092 1 54100 16856 futex_wait_queue_me 0 S TimeCheckThread
audioserver 1783 14919 1 54100 16856 binder_thread_read 0 S Binder:1783_3
audioserver 1783 17914 1 54100 16856 binder_thread_read 0 S Binder:1783_4
可以快速剔除三个Apm相关线程,audioserver主线程,Binder线程,以及HW相关的线程.十分怀疑AudioOut_D就是指PlayBackThread线程了.结合到代码看PlayBackThread的构造方法,有如下逻辑,就更证实了之前的猜想.
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snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
其threadLoop的内容十分冗长,为了方便分析,可以简化其流程:
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//frameworks/av/services/audioflinger/Threads.cpp
bool AudioFlinger::PlaybackThread::threadLoop()
{
....
acquireWakeLock();
//判断线程循环条件
while (!exitPending())
{
{
Mutex::Autolock _l(mLock);
//Step 1. 循环处理config
processConfigEvents_l();
saveOutputTracks();
...
if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
isSuspended()) {
if (shouldStandby_l()) {
//step 2. threadLoop_standby
threadLoop_standby();
mStandby = true;
}
}
//step 3.准备音频数据
mMixerStatus = prepareTracks_l(&tracksToRemove);
...
} // mLock scope ends
if (mBytesRemaining == 0) {
...
if (mMixerStatus == MIXER_TRACKS_READY) {
//step 4. 开始混音
threadLoop_mix();
} else if ((mMixerStatus != MIXER_DRAIN_TRACK)
&& (mMixerStatus != MIXER_DRAIN_ALL)) {
//休眠
threadLoop_sleepTime();
...
}
}
...
if (!waitingAsyncCallback()) {
if (mSleepTimeUs == 0) {
ssize_t ret = 0;
nsecs_t previousLastWriteFinished = lastWriteFinished;
nsecs_t delta = 0;
if (mBytesRemaining) {
...
//step 5. 将数据写入音频硬件设备
ret = threadLoop_write();
...
} else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
(mMixerStatus == MIXER_DRAIN_ALL)) {
threadLoop_drain();
}
...
} else {
...
}
}
//step 6. 移除Tracks
threadLoop_removeTracks(tracksToRemove);
tracksToRemove.clear();
clearOutputTracks();
}
//step 7.退出线程
threadLoop_exit();
if (!mStandby) {
threadLoop_standby();
mStandby = true;
}
releaseWakeLock();
return false;
}
至此,可以总结PlaybackThread的循环流程为:
processConfigEvents_l循环处理configthreadLoop_standby进入休眠?prepareTracks_l准备音频数据threadLoop_mix开始混音threadLoop_write写入音频硬件threadLoop_removeTracks移除相关TracksthreadLoop_exit跳出循环后,结束线程
A.1 processConfigEvents_l
ThreadBase(PlayBackThread的父类)中实现了几个方法将events加入到mConfigEvents中,如下方法均可以变更配置:
- sendPrioConfigEvent 设置优先级
- sendIoConfigEvent 发送IO配置
- setParameters 设置参数
- sendCreateAudioPatchConfigEvent 发送创建AudioPatch配置
- sendReleaseAudioPatchConfigEvent 发送释放AudioPatch配置
关于AudioPatch的资料可以参考链接: 在Android5.0上Audio Patch和Patch Panel的一些分析
现在可以分析processConfigEvents_l了:
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//frameworks/av/services/audioflinger/Threads.cpp
void AudioFlinger::ThreadBase::processConfigEvents_l()
{
bool configChanged = false;
//当mConfigEvents不为空时
while (!mConfigEvents.isEmpty()) {
//每次从头部处理event
sp<ConfigEvent> event = mConfigEvents[0];
mConfigEvents.removeAt(0);
switch (event->mType) {
case CFG_EVENT_PRIO: {
PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
//调用libmediautils库的方法修改优先级
int err = requestPriority(data->mPid, data->mTid, data->mPrio, data->mForApp,
true /*asynchronous*/);
...
} break;
case CFG_EVENT_IO: {
IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
//最终调用到AudioSystem的ioConfigChanged,ioConfig类型包括:
//enum audio_io_config_event {
// AUDIO_OUTPUT_REGISTERED, //Output注册
// AUDIO_OUTPUT_OPENED, //Output打开
// AUDIO_OUTPUT_CLOSED, //Output关闭
// AUDIO_OUTPUT_CONFIG_CHANGED,//Output配置改变
// AUDIO_INPUT_REGISTERED, //INPUT注册
// AUDIO_INPUT_OPENED, //INPUT打开
// AUDIO_INPUT_CLOSED, //INPUT关闭
// AUDIO_INPUT_CONFIG_CHANGED, //INPUT配置改变
//};
ioConfigChanged(data->mEvent, data->mPid);
} break;
case CFG_EVENT_SET_PARAMETER: {
SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
//检查并设置属性,具体线程有自己的实现方式
if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
configChanged = true;
}
} break;
case CFG_EVENT_CREATE_AUDIO_PATCH: {
const audio_devices_t oldDevice = getDevice();
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)event->mData.get();
//创建AudioPatch,具体线程需要实现createAudioPatch_l
event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
const audio_devices_t newDevice = getDevice();
...
} break;
case CFG_EVENT_RELEASE_AUDIO_PATCH: {
const audio_devices_t oldDevice = getDevice();
ReleaseAudioPatchConfigEventData *data =
(ReleaseAudioPatchConfigEventData *)event->mData.get();
//释放AudioPatch,具体线程需要实现releaseAudioPatch_l
event->mStatus = releaseAudioPatch_l(data->mHandle);
const audio_devices_t newDevice = getDevice();
} break;
default:
break;
}
{
Mutex::Autolock _l(event->mLock);
if (event->mWaitStatus) {
event->mWaitStatus = false;
event->mCond.signal();
}
}
}
if (configChanged) {
cacheParameters_l();
}
}
A.2 threadLoop_standby
threadLoop_standby最终调用到HAL层的output_standby方法.
A.3 prepareTracks_l && threadLoop_mix
prepareTracks_l和threadLoop_mix会在具体线程实现,将放在后续进行分析.
A.4 threadLoop_write
threadLoop_write这里展示的是PlaybackThread的实现,一般DirectOutputThread以及OffloadThread会调用到这里.
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ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
mInWrite = true;
ssize_t bytesWritten;
const size_t offset = mCurrentWriteLength - mBytesRemaining;
//NBAIO模式
if (mNormalSink != 0) {
const size_t count = mBytesRemaining / mFrameSize;
uint32_t screenState = AudioFlinger::mScreenState;
if (screenState != mScreenState) {
mScreenState = screenState;
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
if (pipe != NULL) {
pipe->setAvgFrames((mScreenState & 1) ?
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
//也是将SinkBuffer写入
ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
if (framesWritten > 0) {
bytesWritten = framesWritten * mFrameSize;
} else {
bytesWritten = framesWritten;
}
// otherwise use the HAL / AudioStreamOut directly
} else {
//Direct模式以及Offload模式
if (mUseAsyncWrite) {
mWriteAckSequence += 2;
mWriteAckSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
//在这里将数据写入到音频硬件设备中,具体需要看HAL层接口,写的数据保存在了mSinkBuffer中
bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
mWriteAckSequence &= ~1;
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
}
mNumWrites++;
mInWrite = false;
mStandby = false;
return bytesWritten;
}
6. threadLoop_removeTracks
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void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
const Vector< sp<Track> >& tracksToRemove)
{
size_t count = tracksToRemove.size();
if (count > 0) {
//遍历tracksToRemove中的track进行移除
for (size_t i = 0 ; i < count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
if (track->isExternalTrack()) {
//通过AudioSystem调用StopOutput停用
AudioSystem::stopOutput(mId, track->streamType(),
track->sessionId());
if (track->isTerminated()) {
AudioSystem::releaseOutput(mId, track->streamType(),
track->sessionId());
}
}
}
}
}
B. DirectOutputThread
至此具体到DirectOutputThread,即不需要进行混音即可输出.当满足flags置位了AUDIO_OUTPUT_FLAG_DIRECT,或采样精度,通道数不满足混音时,此时会新建DirectOutputThread.
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else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
}
首先来看其构造函数,type类型定义为DIRECT.
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AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady)
: PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady)
{
}
DirectOutThread并没有实现自己的ThreadLoop,而是使用了父类PlaybackThread的,但对关键的步骤进行了覆盖,先再展示ThreadLoop的关键步骤:
processConfigEvents_l循环处理configthreadLoop_standby进入休眠?prepareTracks_l准备音频数据threadLoop_mix开始混音threadLoop_write写入音频硬件threadLoop_removeTracks移除相关TracksthreadLoop_exit跳出循环后,结束线程
DirectOutThread在上述方法进行重写的方法包括prepareTracks_l以及threadLoop_mix,其余步骤都是使用了父类PlaybackThread或者ThreadBase的方法.
prepareTracks_l进行准备性的工作,并处理了许多underrun的情景.
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AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove
)
{
size_t count = mActiveTracks.size();
mixer_state mixerStatus = MIXER_IDLE;
bool doHwPause = false;
bool doHwResume = false;
//当AudioTrack自上层创建后,会通过PlaybackThread的createTrack_l创建Track.
//假如将Track加入到了mActiveTracks中,表明这个Track是Active(活跃的).这里就是希望从ActiveTracks中找到需要处理的Track
for (const sp<Track> &t : mActiveTracks) {
if (t->isInvalid()) {
tracksToRemove->add(t);
continue;
}
Track* const track = t.get();
#ifdef VERY_VERY_VERBOSE_LOGGING
//准备音频数据块
audio_track_cblk_t* cblk = track->cblk();
#endif
//获取mLatestActiveTrack,即最近一次调用add时,会将add的Track赋值为mLatestActiveTrack.
sp<Track> l = mActiveTracks.getLatest();
//假如上次处理的Track与当前Track相同
bool last = l.get() == track;
if (track->isPausing()) {
track->setPaused();
//HAL层支持暂停并且mLatestActiveTrack不为空,会设置doHwPause,mHwPaused为true,后续将会调用到HAL层的暂停
if (mHwSupportsPause && last && !mHwPaused) {
doHwPause = true;
mHwPaused = true;
}
tracksToRemove->add(track);
} else if (track->isFlushPending()) {
track->flushAck();
if (last) {
mFlushPending = true;
}
} else if (track->isResumePending()) {
track->resumeAck();
if (last) {
mLeftVolFloat = mRightVolFloat = -1.0;
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
}
//计算最小帧数,在createTrack_l时,mNormalFrameCount在readOutputParameters_l时被更新
//sharedBuffer为0表明当前为stream模式.
uint32_t minFrames;
if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
&& (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) {
minFrames = mNormalFrameCount;
} else {
minFrames = 1;
}
//当前准备帧数大于最小帧数,track已经准备就绪,且不是暂停或停止状态.
if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
!track->isStopping_2() && !track->isStopped())
{
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
if (last) {
mLeftVolFloat = mRightVolFloat = -1.0;
}
if (!mHwSupportsPause) {
track->resumeAck();
}
}
//处理音量,该方法DirectOutThread也进行了重写
processVolume_l(track, last);
if (last) {//假如上次处理的Track为当前循环遍历的Track
sp<Track> previousTrack = mPreviousTrack.promote();
if (previousTrack != 0) {
if (track != previousTrack.get()) {
mBytesRemaining = 0;
previousTrack->invalidate();
}
}
//将当前track赋值到mPreviousTrack,表明这是最近处理的Track
mPreviousTrack = track;
//最大重试次数为2
track->mRetryCount = kMaxTrackRetriesDirect;
mActiveTrack = t;
mixerStatus = MIXER_TRACKS_READY;
//假如mHwPause当前为True,设置doHwResume为True,mHwPause改为false
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
} else {//数据未准备好
if (!mEffectChains.isEmpty() && last) {
mEffectChains[0]->clearInputBuffer();
}
//STOPPING_1即出现了第一次underrun,生产者速度不及消费
if (track->isStopping_1()) {
//STOPPING_2等待当前完成
track->mState = TrackBase::STOPPING_2;
//上次处理的Track跟当前处理的相同,且mHwPaused为true
if (last && mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
//当前为stream模式或者track处于停止或者暂停状态
if ((track->sharedBuffer() != 0) || track->isStopped() ||
track->isStopping_2() || track->isPaused()) {
//此时表明已经消费完所有Track中的buffer了,需要将Track从activeTracks中移除
size_t audioHALFrames;
if (audio_has_proportional_frames(mFormat)) {
audioHALFrames = (latency_l() * mSampleRate) / 1000;
} else {
audioHALFrames = 0;
}
int64_t framesWritten = mBytesWritten / mFrameSize;
if (mStandby || !last ||
track->presentationComplete(framesWritten, audioHALFrames)) {
if (track->isStopping_2()) {
track->mState = TrackBase::STOPPED;
}
if (track->isStopped()) {
track->reset();
}
tracksToRemove->add(track);
}
} else {
//该Track没有buffer,给它机会去填充buffer,然后从ActiveTracks中移除
if (--(track->mRetryCount) <= 0) {
tracksToRemove->add(track);
//disable该track,当数据有效时,会自动重启
track->disable();
} else if (last) {
//因为underrun而暂停,将doHwPause,mHwPaused设置为true
mixerStatus = MIXER_TRACKS_ENABLED;
if (mHwSupportsPause && !mHwPaused && !mStandby) {
doHwPause = true;
mHwPaused = true;
}
}
}
}
}
// if an active track did not command a flush, check for pending flush on stopped tracks
if (!mFlushPending) {
for (size_t i = 0; i < mTracks.size(); i++) {
if (mTracks[i]->isFlushPending()) {
mTracks[i]->flushAck();
mFlushPending = true;
}
}
}
//保证pause/flush/resume的序列以正确的顺序执行.
//假如flush是待执行而且track是active的,但硬件并没有暂停,会强制使硬件先于flush执行暂停,再继续运行.
if (mHwSupportsPause && !mStandby &&
(doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
//调用到HAL层的暂停.
status_t result = mOutput->stream->pause();
}
//同步数据
if (mFlushPending) {
flushHw_l();
}
if (mHwSupportsPause && !mStandby && doHwResume) {
//调用到HAL层的resume
status_t result = mOutput->stream->resume();
}
//移除所有需要移除的Track
removeTracks_l(*tracksToRemove);
return mixerStatus;
}
再来看一下当数据准备完成后,DirectOutputThread调用processVolume_l的流程:
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void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
{
float left, right;
//当mMasterMute为true或者流音量中的mute设置为true,左右声道音量都设置为0
if (mMasterMute || mStreamTypes[track->streamType()].mute) {
left = right = 0;
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
//mMasterVolume恒为1.0,上层AudioManager已经废弃相关接口不使用MasterVolume了
float v = mMasterVolume * typeVolume;
sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
// Get volumeshaper scaling
std::pair<float /* volume */, bool /* active */>
vh = track->getVolumeHandler()->getVolume(
track->mAudioTrackServerProxy->framesReleased());
v *= vh.first;
mVolumeShaperActive = vh.second;
//获取左右音量,并进行转换赋值到left,right中,最终值还需要与v进行相乘.
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
left = float_from_gain(gain_minifloat_unpack_left(vlr));
if (left > GAIN_FLOAT_UNITY) {
left = GAIN_FLOAT_UNITY;
}
left *= v;
right = float_from_gain(gain_minifloat_unpack_right(vlr));
if (right > GAIN_FLOAT_UNITY) {
right = GAIN_FLOAT_UNITY;
}
right *= v;
}
if (lastTrack) {
track->setFinalVolume((left + right) / 2.f);
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
if (!mEffectChains.isEmpty()) {
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
(void)mEffectChains[0]->setVolume_l(&vl, &vr);
} else {
setVolumeForOutput_l(left, right);
}
}
}
}
DirectOutputThread中的threadLoop_mix将数据从Track中搬运到了SinkBuffer中.
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void AudioFlinger::DirectOutputThread::threadLoop_mix()
{
size_t frameCount = mFrameCount;
//mSinkBuffer是在readOutputParameters_l时申请的内存,其大小是帧数乘以帧大小
//const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
//(void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
//在后续的threadLoop_write,会将mSinkBuffer的数据写入到硬件中
int8_t *curBuf = (int8_t *)mSinkBuffer;
//输出Audio到HAL
while (frameCount) {
AudioBufferProvider::Buffer buffer;
buffer.frameCount = frameCount;
//获取mActiveTrack的buffer
status_t status = mActiveTrack->getNextBuffer(&buffer);
if (status != NO_ERROR || buffer.raw == NULL) {
// no need to pad with 0 for compressed audio
if (audio_has_proportional_frames(mFormat)) {
memset(curBuf, 0, frameCount * mFrameSize);
}
break;
}
//拷贝数据到mSinkBuffer
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
frameCount -= buffer.frameCount;
curBuf += buffer.frameCount * mFrameSize;
mActiveTrack->releaseBuffer(&buffer);
}
mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
mSleepTimeUs = 0;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mActiveTrack.clear();
}
C. MixerThread
MixerThread一般用的场景更多,先来看下其构造方法:
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AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
: PlaybackThread(audioFlinger, output, id, device, type, systemReady),
mFastMixerFutex(0),
mMasterMono(false)
{
//MixerThread中都会有唯一的AudioMixer用于混音
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
//DuplicatingThread也是MixerThread,但不需要执行后续的初始化工作
if (type == DUPLICATING) {
return;
}
//为HAL层的output stream创建NBAIO sink端
mOutputSink = new AudioStreamOutSink(output->stream);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
#if !LOG_NDEBUG
ssize_t index =
#else
(void)
#endif
mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
//初始化FastMixer,暂不分析
....
}
}
关于FastMixer的内容,可以查看:
再继续看MixerThread::prepareTracks_l
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AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove)
{
//在分配新Track前清理AudioMixer中的tracks
(void)mTracks.processDeletedTrackNames([this](int name) {
// for each name, destroy it in the AudioMixer
if (mAudioMixer->exists(name)) {
mAudioMixer->destroy(name);
}
});
mTracks.clearDeletedTrackNames();
mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = mActiveTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
// counts only _active_ fast tracks
size_t fastTracks = 0;
uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
}
// Delegate master volume control to effect in output mix effect chain if needed
sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume_l(&v, &v);
masterVolume = (float)((v + (1 << 23)) >> 24);
chain.clear();
}
// prepare a new state to push
FastMixerStateQueue *sq = NULL;
FastMixerState *state = NULL;
bool didModify = false;
FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
bool coldIdle = false;
if (mFastMixer != 0) {
sq = mFastMixer->sq();
state = sq->begin();
coldIdle = state->mCommand == FastMixerState::COLD_IDLE;
}
mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
//遍历mActiveTracks
for (size_t i=0 ; i<count ; i++) {
const sp<Track> t = mActiveTracks[i];
//获取当前track
Track* const track = t.get();
//处理FastTrack
if (track->isFastTrack()) {
int j = track->mFastIndex;
FastTrack *fastTrack = &state->mFastTracks[j];
FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
FastTrackUnderruns underruns = ftDump->mUnderruns;
uint32_t recentFull = (underruns.mBitFields.mFull -
track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
uint32_t recentPartial = (underruns.mBitFields.mPartial -
track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
uint32_t recentUnderruns = recentPartial + recentEmpty;
track->mObservedUnderruns = underruns;
if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
recentUnderruns > 0) {
track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
} else {
track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
// This is similar to the state machine for normal tracks,
// with a few modifications for fast tracks.
bool isActive = true;
switch (track->mState) {
case TrackBase::STOPPING_1:
if (recentUnderruns > 0 || track->isTerminated()) {
track->mState = TrackBase::STOPPING_2;
}
break;
case TrackBase::PAUSING:
track->setPaused();
break;
case TrackBase::RESUMING:
track->mState = TrackBase::ACTIVE;
break;
case TrackBase::ACTIVE:
if (recentFull > 0 || recentPartial > 0) {
track->mRetryCount = kMaxTrackRetries;
}
if (recentUnderruns == 0) {
// no recent underruns: stay active
break;
}
// there has recently been an underrun of some kind
if (track->sharedBuffer() == 0) {
// were any of the recent underruns "empty" (no frames available)?
if (recentEmpty == 0) {
// no, then ignore the partial underruns as they are allowed indefinitely
break;
}
// there has recently been an "empty" underrun: decrement the retry counter
if (--(track->mRetryCount) > 0) {
break;
}
// indicate to client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
track->disable();
// remove from active list, but state remains ACTIVE [confusing but true]
isActive = false;
break;
}
FALLTHROUGH_INTENDED;
case TrackBase::STOPPING_2:
case TrackBase::PAUSED:
case TrackBase::STOPPED:
case TrackBase::FLUSHED: // flush() while active
// Check for presentation complete if track is inactive
// We have consumed all the buffers of this track.
// This would be incomplete if we auto-paused on underrun
{
uint32_t latency = 0;
status_t result = mOutput->stream->getLatency(&latency);
ALOGE_IF(result != OK,
"Error when retrieving output stream latency: %d", result);
size_t audioHALFrames = (latency * mSampleRate) / 1000;
int64_t framesWritten = mBytesWritten / mFrameSize;
if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
// track stays in active list until presentation is complete
break;
}
}
if (track->isStopping_2()) {
track->mState = TrackBase::STOPPED;
}
if (track->isStopped()) {
// Can't reset directly, as fast mixer is still polling this track
// track->reset();
// So instead mark this track as needing to be reset after push with ack
resetMask |= 1 << i;
}
isActive = false;
break;
case TrackBase::IDLE:
default:
LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
}
if (isActive) {
// was it previously inactive?
if (!(state->mTrackMask & (1 << j))) {
ExtendedAudioBufferProvider *eabp = track;
VolumeProvider *vp = track;
fastTrack->mBufferProvider = eabp;
fastTrack->mVolumeProvider = vp;
fastTrack->mChannelMask = track->mChannelMask;
fastTrack->mFormat = track->mFormat;
fastTrack->mGeneration++;
state->mTrackMask |= 1 << j;
didModify = true;
// no acknowledgement required for newly active tracks
}
sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
// cache the combined master volume and stream type volume for fast mixer; this
// lacks any synchronization or barrier so VolumeProvider may read a stale value
const float vh = track->getVolumeHandler()->getVolume(
proxy->framesReleased()).first;
float volume = masterVolume
* mStreamTypes[track->streamType()].volume
* vh;
track->mCachedVolume = volume;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
float vlf = volume * float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = volume * float_from_gain(gain_minifloat_unpack_right(vlr));
track->setFinalVolume((vlf + vrf) / 2.f);
++fastTracks;
} else {
// was it previously active?
if (state->mTrackMask & (1 << j)) {
fastTrack->mBufferProvider = NULL;
fastTrack->mGeneration++;
state->mTrackMask &= ~(1 << j);
didModify = true;
// If any fast tracks were removed, we must wait for acknowledgement
// because we're about to decrement the last sp<> on those tracks.
block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
} else {
// ALOGW rather than LOG_ALWAYS_FATAL because it seems there are cases where an
// AudioTrack may start (which may not be with a start() but with a write()
// after underrun) and immediately paused or released. In that case the
// FastTrack state hasn't had time to update.
// TODO Remove the ALOGW when this theory is confirmed.
ALOGW("fast track %d should have been active; "
"mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
j, track->mState, state->mTrackMask, recentUnderruns,
track->sharedBuffer() != 0);
// Since the FastMixer state already has the track inactive, do nothing here.
}
tracksToRemove->add(track);
// Avoids a misleading display in dumpsys
track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
}
continue;
}
{ // local variable scope to avoid goto warning
//准备数据块
audio_track_cblk_t* cblk = track->cblk();
int name = track->name();
//AudioMixer中不存在该Track,则创建一个
if (!mAudioMixer->exists(name)) {
status_t status = mAudioMixer->create(
name,
track->mChannelMask,
track->mFormat,
track->mSessionId);
if (status != OK) {
tracksToRemove->add(track);
track->invalidate(); // consider it dead.
continue;
}
}
size_t desiredFrames;
const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
desiredFrames = sourceFramesNeededWithTimestretch(
sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
minFrames = desiredFrames;
}
size_t framesReady = track->framesReady();
if (ATRACE_ENABLED()) {
std::string traceName("nRdy");
traceName += std::to_string(track->name());
}
//数据已准备好
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
mixedTracks++;
chain.clear();
if (track->mainBuffer() != mSinkBuffer &&
track->mainBuffer() != mMixerBuffer) {
if (mEffectBufferEnabled) {
mEffectBufferValid = true; // Later can set directly.
}
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
tracksWithEffect++;
} else {
ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
"session %d",
name, track->sessionId());
}
}
int param = AudioMixer::VOLUME;
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
//AudioMixer设置参数
mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
mLeftVolFloat = -1.0;
} else if (cblk->mServer != 0) {
param = AudioMixer::RAMP_VOLUME;
}
//计算Track的音量
uint32_t vl, vr; // in U8.24 integer format
float vlf, vrf, vaf; // in [0.0, 1.0] float format
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
vl = vr = 0;
vlf = vrf = vaf = 0.;
if (track->isPausing()) {
track->setPaused();
}
} else {
sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
if (vlf > GAIN_FLOAT_UNITY) {
vlf = GAIN_FLOAT_UNITY;
}
if (vrf > GAIN_FLOAT_UNITY) {
vrf = GAIN_FLOAT_UNITY;
}
const float vh = track->getVolumeHandler()->getVolume(
track->mAudioTrackServerProxy->framesReleased()).first;
vlf *= v * vh;
vrf *= v * vh;
const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
vl = (uint32_t) (scaleto8_24 * vlf);
vr = (uint32_t) (scaleto8_24 * vrf);
uint16_t sendLevel = proxy->getSendLevel_U4_12();
if (sendLevel > MAX_GAIN_INT) {
sendLevel = MAX_GAIN_INT;
}
vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
//设置音量
track->setFinalVolume((vrf + vlf) / 2.f);
if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
param = AudioMixer::VOLUME;
vlf = (float)vl / (1 << 24);
vrf = (float)vr / (1 << 24);
track->mHasVolumeController = true;
} else {
if (track->mHasVolumeController) {
param = AudioMixer::VOLUME;
}
track->mHasVolumeController = false;
}
if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
v = mStreamTypes[track->streamType()].mute ? 0.0f : v;
if (v != mLeftVolFloat) {
status_t result = mOutput->stream->setVolume(v, v);
if (result == OK) {
mLeftVolFloat = v;
}
}
// if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we
// remove stream volume contribution from software volume.
if (v != 0.0f && mLeftVolFloat == v) {
vlf = min(1.0f, vlf / v);
vrf = min(1.0f, vrf / v);
vaf = min(1.0f, vaf / v);
}
}
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(name, track);
mAudioMixer->enable(name);
//通过AudioMixer设置音量
mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
if (reqSampleRate == 0) {
reqSampleRate = mSampleRate;
} else if (reqSampleRate > maxSampleRate) {
reqSampleRate = maxSampleRate;
}
mAudioMixer->setParameter(
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(uintptr_t)reqSampleRate);
AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
mAudioMixer->setParameter(
name,
AudioMixer::TIMESTRETCH,
AudioMixer::PLAYBACK_RATE,
&playbackRate);
if (mMixerBufferEnabled
&& (track->mainBuffer() == mSinkBuffer
|| track->mainBuffer() == mMixerBuffer)) {
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
mMixerBufferValid = true;
} else {
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)EFFECT_BUFFER_FORMAT);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
}
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
track->mRetryCount = kMaxTrackRetries;
if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_ENABLED) {
mixerStatus = MIXER_TRACKS_READY;
}
} else {
if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
} else {
track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->clearInputBuffer();
}
if ((track->sharedBuffer() != 0) || track->isTerminated() ||
track->isStopped() || track->isPaused()) {
size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
int64_t framesWritten = mBytesWritten / mFrameSize;
if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
if (track->isStopped()) {
track->reset();
}
tracksToRemove->add(track);
}
} else {
if (--(track->mRetryCount) <= 0) {
tracksToRemove->add(track);
track->disable();
} else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(name);
}
} // local variable scope to avoid goto warning
}
bool pauseAudioWatchdog = false;
if (didModify) {
state->mFastTracksGen++;
if (kUseFastMixer == FastMixer_Dynamic &&
state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
state->mCommand = FastMixerState::COLD_IDLE;
state->mColdFutexAddr = &mFastMixerFutex;
state->mColdGen++;
mFastMixerFutex = 0;
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mOutputSink;
}
block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
pauseAudioWatchdog = true;
}
}
if (sq != NULL) {
sq->end(didModify);
sq->push(coldIdle ? FastMixerStateQueue::BLOCK_NEVER : block);
}
while (resetMask != 0) {
size_t i = __builtin_ctz(resetMask);
ALOG_ASSERT(i < count);
resetMask &= ~(1 << i);
sp<Track> track = mActiveTracks[i];
ALOG_ASSERT(track->isFastTrack() && track->isStopped());
track->reset();
}
for (const auto &track : *tracksToRemove) {
const int name = track->name();
if (mAudioMixer->exists(name)) { // Normal tracks here, fast tracks in FastMixer.
mAudioMixer->setBufferProvider(name, nullptr /* bufferProvider */);
}
}
removeTracks_l(*tracksToRemove);
if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
mEffectBufferValid = true;
}
if (mEffectBufferValid) {
memset(mEffectBuffer, 0, mEffectBufferSize);
}
if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
(mixedTracks == 0 && fastTracks > 0))) {
// FIXME as a performance optimization, should remember previous zero status
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
}
memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
}
mMixerStatusIgnoringFastTracks = mixerStatus;
if (fastTracks > 0) {
mixerStatus = MIXER_TRACKS_READY;
}
return mixerStatus;
}
prepareTracks_l的流程很长,但目前只需要把握几点,即创建AudioMixer,并设置参数.但还没有传到HAL层,还需要调用thread_mix:
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void AudioFlinger::MixerThread::threadLoop_mix()
{
//处理AudioMixer
mAudioMixer->process();
...
}
AudioMixer的process类似与钩子函数,在不同的场景将会调用不同的接口.
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//frameworks/av/include/media/AudioMixer.h
void process() {
(this->*mHook)();
}
对应的场景接口包括:
| 接口 | 描述 |
|---|---|
process__nop() |
初始值 |
process__genericNoResampling |
不重采样 |
process__genericResampling |
需要重采样 |
process__noResampleOneTrack |
单路Track,不需要重采样 |
process__oneTrack16BitsStereoNoResampling |
单路16bit双通道不需要重采样 |
实际AudioMixer的操作在这里暂不讨论,后续深入学习后再进行总结.
4.3.4 addOutput
再回到AudioPolicyManager的流程:
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void AudioPolicyManager::addOutput(audio_io_handle_t output,
const sp<SwAudioOutputDescriptor>& outputDesc)
{
mOutputs.add(output, outputDesc);
applyStreamVolumes(outputDesc, AUDIO_DEVICE_NONE, 0 /* delayMs */, true /* force */);
updateMono(output); // update mono status when adding to output list
selectOutputForMusicEffects();
nextAudioPortGeneration();
}
4.3.5 setOutputDevice
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uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
bool force,
int delayMs,
audio_patch_handle_t *patchHandle,
const char *address,
bool requiresMuteCheck)
{
AudioParameter param;
uint32_t muteWaitMs;
//假如outputDesc重复了,即mOutput1,mOuptut2都不为空,则分别进行设置,这里用了递归
if (outputDesc->isDuplicated()) {
muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs,
nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs,
nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
return muteWaitMs;
}
if ((device != AUDIO_DEVICE_NONE) &&
((device & outputDesc->supportedDevices()) == AUDIO_DEVICE_NONE)) {
return 0;
}
//outputDesc获取支持的设备
device = (audio_devices_t)(device & outputDesc->supportedDevices());
audio_devices_t prevDevice = outputDesc->mDevice;
if (device != AUDIO_DEVICE_NONE) {
outputDesc->mDevice = device;
}
if (requiresMuteCheck) {
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
} else {
muteWaitMs = 0;
}
if ((device == AUDIO_DEVICE_NONE || device == prevDevice) &&
!force &&
outputDesc->getPatchHandle() != 0) {
return muteWaitMs;
}
//假如device为AUDIO_DEVICE_NONE,需要进行routing,重新设置device
if (device == AUDIO_DEVICE_NONE) {
resetOutputDevice(outputDesc, delayMs, NULL);
} else {
DeviceVector deviceList;
//获取deviceList
if ((address == NULL) || (strlen(address) == 0)) {
deviceList = mAvailableOutputDevices.getDevicesFromType(device);
} else {
deviceList = mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
}
if (!deviceList.isEmpty()) {
struct audio_patch patch;
outputDesc->toAudioPortConfig(&patch.sources[0]);
patch.num_sources = 1;
patch.num_sinks = 0;
for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
patch.num_sinks++;
}
ssize_t index;
if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
}
sp< AudioPatch> patchDesc;
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
afPatchHandle = patchDesc->mAfPatchHandle;
}
//创建AudioPatch
status_t status = mpClientInterface->createAudioPatch(&patch,
&afPatchHandle,
delayMs);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
//设置Patch的Handle值
outputDesc->setPatchHandle(patchDesc->mHandle);
nextAudioPortGeneration();
//调用回调通知Patch已经更新了
mpClientInterface->onAudioPatchListUpdate();
}
}
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (!is_virtual_input_device(inputDescriptor->mDevice)) {
AudioParameter inputCmd = AudioParameter();
inputCmd.addInt(String8(AudioParameter::keyRouting),device);
//通知inptut
mpClientInterface->setParameters(inputDescriptor->mIoHandle,
inputCmd.toString(),
delayMs);
}
}
}
applyStreamVolumes(outputDesc, device, delayMs);
return muteWaitMs;
}
5. 小结
分析至此,仅仅是大致了解了音频系统的初始化流程,其中许多细节仍然是还不了解,需要再实际调试或者解问题中加深了解.本博客也会适时查漏补缺,及时更新.
