Android Audio代码分析7 - stream type

在看AudioTrack代码的时候，我们看到，要创建一个AudioTrack对象，需要指定一个StreamType。
今天我们只把stream type相关的代码抽取出来，详细看看stream type相关的东东。

java层的代码就不看了。
从函数android_media_AudioTrack_native_setup开始说起。

static int
android_media_AudioTrack_native_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jint streamType, jint sampleRateInHertz, jint channels,
jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession)
{
...

// 获取Frame Count和Sampling Rate的依据都是stream type。
// 其实现方法是通过stream type取得output，然后取得output的描述
// 如果取得成功，则取描述中的frame count，否则通过audio flinger取得output对应的frame count。
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
LOGE("Error creating AudioTrack: Could not get AudioSystem frame count.");
return AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
}
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
LOGE("Error creating AudioTrack: Could not get AudioSystem sampling rate.");
return AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
}

...

// 下面的工作是将java 侧的stream type转换为native 侧的stream type。
// 以后使用的都是转换后的strea type。
// check the stream type
AudioSystem::stream_type atStreamType;
if (streamType == javaAudioTrackFields.STREAM_VOICE_CALL) {
atStreamType = AudioSystem::VOICE_CALL;
} else if (streamType == javaAudioTrackFields.STREAM_SYSTEM) {
atStreamType = AudioSystem::SYSTEM;
} else if (streamType == javaAudioTrackFields.STREAM_RING) {
atStreamType = AudioSystem::RING;
} else if (streamType == javaAudioTrackFields.STREAM_MUSIC) {
atStreamType = AudioSystem::MUSIC;
} else if (streamType == javaAudioTrackFields.STREAM_ALARM) {
atStreamType = AudioSystem::ALARM;
} else if (streamType == javaAudioTrackFields.STREAM_NOTIFICATION) {
atStreamType = AudioSystem::NOTIFICATION;
} else if (streamType == javaAudioTrackFields.STREAM_BLUETOOTH_SCO) {
atStreamType = AudioSystem::BLUETOOTH_SCO;
} else if (streamType == javaAudioTrackFields.STREAM_DTMF) {
atStreamType = AudioSystem::DTMF;
} else {
LOGE("Error creating AudioTrack: unknown stream type.");
return AUDIOTRACK_ERROR_SETUP_INVALIDSTREAMTYPE;
}

...

// 将stream type保存在AudioTrackJniStorage对象中
lpJniStorage->mStreamType = atStreamType;

...

// 调用AudioTrack对象的set函数
// initialize the native AudioTrack object
if (memoryMode == javaAudioTrackFields.MODE_STREAM) {

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioTrack::set(
int streamType,
uint32_t sampleRate,
int format,
int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId)
{

...

// 前面已经说过
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}

// stream type如果是DEFAULT，将其设置为MUSIC
// handle default values first.
if (streamType == AudioSystem::DEFAULT) {
streamType = AudioSystem::MUSIC;
}

...

// 获取output
audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
sampleRate, format, channels, (AudioSystem::output_flags)flags);

if (output == 0) {
LOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}

mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;

// 创建IAudioTrack对象
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioTrack::createTrack(
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount)
{
...

// 这几个又出现了
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}

...

// 调用audio flinger中的函数创建IAudioTrack对象
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
int lSessionId;

// 参数检查
if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}

...

// 并没有对stream type作其他处理，调用playback thread的函数
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
status_t *status)
{
...

// 创建Track对象
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
...

// 没有再往下传，赋值给了成员变量mStreamType

// 看看哪些地方使用到了该成员变量mStreamType
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// desctructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
// TrackHandle destructor, the TrackHandle still holds a strong ref on
// this Track with its member mTrack.
sp<Track> keep(this);
{ // scope for mLock
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioSystem::stopOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
if (mpPolicyManager == NULL) {
return NO_INIT;
}
LOGV("stopOutput() tid %d", gettid());
Mutex::Autolock _l(mLock);
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
LOGW("stopOutput() unknow output %d", output);
return BAD_VALUE;
}

AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
// 根据stream type获取strategy
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
AudioSystem::stream_type stream) {
// stream to strategy mapping
switch (stream) {
case AudioSystem::VOICE_CALL:
case AudioSystem::BLUETOOTH_SCO:
return STRATEGY_PHONE;
case AudioSystem::RING:
case AudioSystem::NOTIFICATION:
case AudioSystem::ALARM:
case AudioSystem::ENFORCED_AUDIBLE:
return STRATEGY_SONIFICATION;
case AudioSystem::DTMF:
return STRATEGY_DTMF;
default:
LOGE("unknown stream type");
case AudioSystem::SYSTEM:
// NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
// while key clicks are played produces a poor result
case AudioSystem::TTS:
case AudioSystem::MUSIC:
return STRATEGY_MEDIA;
}
}
----------------------------------------------------------------
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);

// handle special case for sonification while in call
if (isInCall()) {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange)
{
// if the stream pertains to sonification strategy and we are in call we must
// mute the stream if it is low visibility. If it is high visibility, we must play a tone
// in the device used for phone strategy and play the tone if the selected device does not
// interfere with the device used for phone strategy
// if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
// many times as there are active tracks on the output

if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
LOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
stream, starting, outputDesc->mDevice, stateChange);
if (outputDesc->mRefCount[stream]) {
int muteCount = 1;
if (stateChange) {
muteCount = outputDesc->mRefCount[stream];
}
if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
{
StreamDescriptor &streamDesc = mStreams[stream];
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);

LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);

if (on) {
if (outputDesc->mMuteCount[stream] == 0) {
if (streamDesc.mCanBeMuted) {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
{

// do not change actual stream volume if the stream is muted
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
LOGV("checkAndSetVolume() stream %d muted count %d", stream, mOutputs.valueFor(output)->mMuteCount[stream]);
return NO_ERROR;
}

// do not change in call volume if bluetooth is connected and vice versa
if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
(stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
return INVALID_OPERATION;
}

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
{
float volume = 1.0;
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
StreamDescriptor &streamDesc = mStreams[stream];

if (device == 0) {
device = outputDesc->device();
}

int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
volume = AudioSystem::linearToLog(volInt);

// if a headset is connected, apply the following rules to ring tones and notifications
// to avoid sound level bursts in user's ears:
// - always attenuate ring tones and notifications volume by 6dB
// - if music is playing, always limit the volume to current music volume,
// with a minimum threshold at -36dB so that notification is always perceived.
if ((device &
(AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
AudioSystem::DEVICE_OUT_WIRED_HEADSET |
AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
((getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) ||
(stream == AudioSystem::SYSTEM)) &&
streamDesc.mCanBeMuted) {
volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
// when the phone is ringing we must consider that music could have been paused just before
// by the music application and behave as if music was active if the last music track was
// just stopped
if (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
if (volume > minVol) {
volume = minVol;
LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
}
}
}

return volume;
}
----------------------------------------------------------------
float volume = computeVolume(stream, index, output, device);
// We actually change the volume if:
// - the float value returned by computeVolume() changed
// - the force flag is set
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
force) {
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::DTMF ||
stream == AudioSystem::BLUETOOTH_SCO) {
// offset value to reflect actual hardware volume that never reaches 0
// 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
volume = 0.01 + 0.99 * volume;
}

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyService::setStreamVolume(AudioSystem::stream_type stream,
float volume,
audio_io_handle_t output,
int delayMs)
{
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream,
float volume,
int output,
int delayMs)
{
status_t status = NO_ERROR;

AudioCommand *command = new AudioCommand();
command->mCommand = SET_VOLUME;
VolumeData *data = new VolumeData();
data->mStream = stream;
data->mVolume = volume;
data->mIO = output;
command->mParam = data;
if (delayMs == 0) {
command->mWaitStatus = true;
} else {
command->mWaitStatus = false;
}
Mutex::Autolock _l(mLock);
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
// insertCommand_l() must be called with mLock held
void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *command, int delayMs)
{
ssize_t i;
Vector <AudioCommand *> removedCommands;

command->mTime = systemTime() + milliseconds(delayMs);

// acquire wake lock to make sure delayed commands are processed
if (mName != "" && mAudioCommands.isEmpty()) {
acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string());
}

// check same pending commands with later time stamps and eliminate them
for (i = mAudioCommands.size()-1; i >= 0; i--) {
AudioCommand *command2 = mAudioCommands[i];
// commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands
if (command2->mTime <= command->mTime) break;
if (command2->mCommand != command->mCommand) continue;

switch (command->mCommand) {
case SET_PARAMETERS: {
ParametersData *data = (ParametersData *)command->mParam;
ParametersData *data2 = (ParametersData *)command2->mParam;
if (data->mIO != data2->mIO) break;
LOGV("Comparing parameter command %s to new command %s",
data2->mKeyValuePairs.string(), data->mKeyValuePairs.string());
AudioParameter param = AudioParameter(data->mKeyValuePairs);
AudioParameter param2 = AudioParameter(data2->mKeyValuePairs);
for (size_t j = 0; j < param.size(); j++) {
String8 key;
String8 value;
param.getAt(j, key, value);
for (size_t k = 0; k < param2.size(); k++) {
String8 key2;
String8 value2;
param2.getAt(k, key2, value2);
if (key2 == key) {
param2.remove(key2);
LOGV("Filtering out parameter %s", key2.string());
break;
}
}
}
// if all keys have been filtered out, remove the command.
// otherwise, update the key value pairs
if (param2.size() == 0) {
removedCommands.add(command2);
} else {
data2->mKeyValuePairs = param2.toString();
}
} break;

case SET_VOLUME: {
VolumeData *data = (VolumeData *)command->mParam;
VolumeData *data2 = (VolumeData *)command2->mParam;
if (data->mIO != data2->mIO) break;
if (data->mStream != data2->mStream) break;
LOGV("Filtering out volume command on output %d for stream %d",
data->mIO, data->mStream);
removedCommands.add(command2);
} break;
case START_TONE:
case STOP_TONE:
default:
break;
}
}

// remove filtered commands
for (size_t j = 0; j < removedCommands.size(); j++) {
// removed commands always have time stamps greater than current command
for (size_t k = i + 1; k < mAudioCommands.size(); k++) {
if (mAudioCommands[k] == removedCommands[j]) {
LOGV("suppressing command: %d", mAudioCommands[k]->mCommand);
mAudioCommands.removeAt(k);
break;
}
}
}
removedCommands.clear();

// insert command at the right place according to its time stamp
LOGV("inserting command: %d at index %d, num commands %d",
command->mCommand, (int)i+1, mAudioCommands.size());
// 将command保存到了成员变量mAudioCommands中。
// 函数AudioPolicyService::AudioCommandThread::threadLoop中，处理mAudioCommands中的command。
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
bool AudioPolicyService::AudioCommandThread::threadLoop()
{
nsecs_t waitTime = INT64_MAX;

mLock.lock();
while (!exitPending())
{
while(!mAudioCommands.isEmpty()) {
nsecs_t curTime = systemTime();
// commands are sorted by increasing time stamp: execute them from index 0 and up
if (mAudioCommands[0]->mTime <= curTime) {
AudioCommand *command = mAudioCommands[0];
mAudioCommands.removeAt(0);
mLastCommand = *command;

switch (command->mCommand) {
case START_TONE: {
mLock.unlock();
ToneData *data = (ToneData *)command->mParam;
LOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
if (mpToneGenerator != NULL)
delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
mLock.lock();
}break;
case STOP_TONE: {
mLock.unlock();
LOGV("AudioCommandThread() processing stop tone");
if (mpToneGenerator != NULL) {
mpToneGenerator->stopTone();
delete mpToneGenerator;
mpToneGenerator = NULL;
}
mLock.lock();
}break;
case SET_VOLUME: {
VolumeData *data = (VolumeData *)command->mParam;
LOGV("AudioCommandThread() processing set volume stream %d, \
volume %f, output %d", data->mStream, data->mVolume, data->mIO);
// 调用到AudioSystem的函数中
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioSystem::setStreamVolume(int stream, float value, int output)
{
if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
// 又调到了audio flinger中
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}

if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return BAD_VALUE;
}

AutoMutex lock(mLock);
PlaybackThread *thread = NULL;
if (output) {
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
}

// audio flinger中保存的各stream的volume
mStreamTypes[stream].volume = value;

if (thread == NULL) {
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setStreamVolume(audioOutputType, stream, value);
}
#endif
// playback thread中保存的各stream的volume
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
----------------------------------------------------------------
thread->setStreamVolume(stream, value);
}

return NO_ERROR;
}
----------------------------------------------------------------
af->setStreamVolume(stream, value, output);
return NO_ERROR;
}
----------------------------------------------------------------
command->mStatus = AudioSystem::setStreamVolume(data->mStream,
data->mVolume,
data->mIO);
if (command->mWaitStatus) {
command->mCond.signal();
mWaitWorkCV.wait(mLock);
}
delete data;
}break;
case SET_PARAMETERS: {
ParametersData *data = (ParametersData *)command->mParam;
LOGV("AudioCommandThread() processing set parameters string %s, io %d",
data->mKeyValuePairs.string(), data->mIO);
command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
if (command->mWaitStatus) {
command->mCond.signal();
mWaitWorkCV.wait(mLock);
}
delete data;
}break;
case SET_VOICE_VOLUME: {
VoiceVolumeData *data = (VoiceVolumeData *)command->mParam;
LOGV("AudioCommandThread() processing set voice volume volume %f",
data->mVolume);
command->mStatus = AudioSystem::setVoiceVolume(data->mVolume);
if (command->mWaitStatus) {
command->mCond.signal();
mWaitWorkCV.wait(mLock);
}
delete data;
}break;
default:
LOGW("AudioCommandThread() unknown command %d", command->mCommand);
}
delete command;
waitTime = INT64_MAX;
} else {
waitTime = mAudioCommands[0]->mTime - curTime;
break;
}
}
// release delayed commands wake lock
if (mName != "" && mAudioCommands.isEmpty()) {
release_wake_lock(mName.string());
}
LOGV("AudioCommandThread() going to sleep");
mWaitWorkCV.waitRelative(mLock, waitTime);
LOGV("AudioCommandThread() waking up");
}
mLock.unlock();
return false;
}
----------------------------------------------------------------
mAudioCommands.insertAt(command, i + 1);
}
----------------------------------------------------------------
insertCommand_l(command, delayMs);
LOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %d",
stream, volume, output);
mWaitWorkCV.signal();
if (command->mWaitStatus) {
command->mCond.wait(mLock);
status = command->mStatus;
mWaitWorkCV.signal();
}
return status;
}
----------------------------------------------------------------
return mAudioCommandThread->volumeCommand((int)stream, volume, (int)output, delayMs);
}
----------------------------------------------------------------
mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
}

if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
float voiceVolume;
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
if (stream == AudioSystem::VOICE_CALL) {
voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
} else {
voiceVolume = 1.0;
}
if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) {
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
mLastVoiceVolume = voiceVolume;
}
}

return NO_ERROR;
}
----------------------------------------------------------------
checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
}
}
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
outputDesc->mMuteCount[stream]++;
} else {
if (outputDesc->mMuteCount[stream] == 0) {
LOGW("setStreamMute() unmuting non muted stream!");
return;
}
if (--outputDesc->mMuteCount[stream] == 0) {
checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs);
}
}
}
----------------------------------------------------------------
setStreamMute(stream, starting, mHardwareOutput);
}
} else {
LOGV("handleIncallSonification() high visibility");
if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, mHardwareOutput);
}
}
if (starting) {
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyService::startTone(ToneGenerator::tone_type tone,
AudioSystem::stream_type stream)
{
// 应该与刚看过的set volume command类似
// command的处理中threadloop函数中，处理代码如下：
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
case START_TONE: {
mLock.unlock();
ToneData *data = (ToneData *)command->mParam;
LOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
if (mpToneGenerator != NULL)
delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
mLock.lock();
}break;
----------------------------------------------------------------
mTonePlaybackThread->startToneCommand(tone, stream);
return NO_ERROR;
}
----------------------------------------------------------------
mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
} else {
mpClientInterface->stopTone();
}
}
}
}
}
----------------------------------------------------------------
handleIncallSonification(stream, false, false);
}

if (outputDesc->mRefCount[stream] > 0) {
// decrement usage count of this stream on the output
outputDesc->changeRefCount(stream, -1);
// store time at which the last music track was stopped - see computeVolume()
if (stream == AudioSystem::MUSIC) {
mMusicStopTime = systemTime();
}

setOutputDevice(output, getNewDevice(output));

#ifdef WITH_A2DP
if (mA2dpOutput != 0 && !a2dpUsedForSonification() &&
strategy == STRATEGY_SONIFICATION) {
setStrategyMute(STRATEGY_MEDIA,
false,
mA2dpOutput,
mOutputs.valueFor(mHardwareOutput)->mLatency*2);
}
#endif
if (output != mHardwareOutput) {
setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
}
return NO_ERROR;
} else {
LOGW("stopOutput() refcount is already 0 for output %d", output);
return INVALID_OPERATION;
}
}
----------------------------------------------------------------
return mpPolicyManager->stopOutput(output, stream, session);
}
----------------------------------------------------------------
return aps->stopOutput(output, stream, session);
}
----------------------------------------------------------------
AudioSystem::stopOutput(thread->id(),
(AudioSystem::stream_type)mStreamType,
mSessionId);
}
AudioSystem::releaseOutput(thread->id());
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->destroyTrack_l(this);
}
}
}
----------------------------------------------------------------

// 看看下一个使用成员变量 mStreamType 的地方是哪儿
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
LOGV("start(%d), calling thread %d session %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (mState == PAUSED) {
mState = TrackBase::RESUMING;
LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
} else {
mState = TrackBase::ACTIVE;
LOGV("? => ACTIVE (%d) on thread %p", mName, this);
}

if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioSystem::startOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
status_t AudioPolicyService::startOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
if (mpPolicyManager == NULL) {
return NO_INIT;
}
LOGV("startOutput() tid %d", gettid());
Mutex::Autolock _l(mLock);
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
// 与刚看过的函数AudioPolicyManagerBase::stopOutput中调用的函数类似
status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
LOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
LOGW("startOutput() unknow output %d", output);
return BAD_VALUE;
}

AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);

#ifdef WITH_A2DP
if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
}
#endif

// incremenent usage count for this stream on the requested output:
// NOTE that the usage count is the same for duplicated output and hardware output which is
// necassary for a correct control of hardware output routing by startOutput() and stopOutput()
outputDesc->changeRefCount(stream, 1);

setOutputDevice(output, getNewDevice(output));

// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, true, false);
}

// apply volume rules for current stream and device if necessary
checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());

return NO_ERROR;
}
----------------------------------------------------------------
return mpPolicyManager->startOutput(output, stream, session);
}
----------------------------------------------------------------
return aps->startOutput(output, stream, session);
}
----------------------------------------------------------------
status = AudioSystem::startOutput(thread->id(),
(AudioSystem::stream_type)mStreamType,
mSessionId);
thread->mLock.lock();
}
if (status == NO_ERROR) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->addTrack_l(this);
} else {
mState = state;
}
} else {
status = BAD_VALUE;
}
return status;
}
----------------------------------------------------------------

// 看看下一个使用成员变量 mStreamType 的地方是哪儿
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
void AudioFlinger::PlaybackThread::Track::stop()
{
LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
// 调用的函数也是AudioSystem::stopOutput，刚才已看过
AudioSystem::stopOutput(thread->id(),
(AudioSystem::stream_type)mStreamType,
mSessionId);
thread->mLock.lock();
}
}
}
----------------------------------------------------------------

// 看看下一个使用成员变量 mStreamType 的地方是哪儿
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
void AudioFlinger::PlaybackThread::Track::pause()
{
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState == ACTIVE || mState == RESUMING) {
mState = PAUSING;
LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
// 调用的函数也是AudioSystem::stopOutput，刚才已看过
AudioSystem::stopOutput(thread->id(),
(AudioSystem::stream_type)mStreamType,
mSessionId);
thread->mLock.lock();
}
}
}
}
----------------------------------------------------------------

mStreamType = streamType;

...

}
----------------------------------------------------------------
track = new Track(this, client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, sessionId);

...

return track;
}
----------------------------------------------------------------
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);

...

return trackHandle;
}
----------------------------------------------------------------
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
format,
channelCount,
frameCount,
((uint16_t)flags) << 16,
sharedBuffer,
output,
&mSessionId,
&status);

if (track == 0) {
LOGE("AudioFlinger could not create track, status: %d", status);
return status;
}

...

return NO_ERROR;
}
----------------------------------------------------------------
// create the IAudioTrack
status_t status = createTrack(streamType, sampleRate, format, channelCount,
frameCount, flags, sharedBuffer, output, true);

if (status != NO_ERROR) {
return status;
}

if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
LOGE("Could not create callback thread");
return NO_INIT;
}
}

mStatus = NO_ERROR;

// 将stream type保存到成员变量中
// 以下函数中会使用到该成员变量
// AudioTrack::streamType - 获取stream type
// AudioTrack::start - 如果status为DEAD_OBJECT，将调用函数createTrack创建IAudioTrack对象
// AudioTrack::setSampleRate - 会根据stream type获取sampling rate。
// AudioTrack::getOutput - 根据stream type获取output
// AudioTrack::obtainBuffer - 如果status为DEAD_OBJECT，将调用函数createTrack创建IAudioTrack对象
mStreamType = streamType;
mFormat = format;
mChannels = channels;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = 0;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlags = flags;

return NO_ERROR;
}
----------------------------------------------------------------
lpTrack->set(
atStreamType,// stream type
sampleRateInHertz,
format,// word length, PCM
channels,
frameCount,
0,// flags
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)
0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
0,// shared mem
true,// thread can call Java
sessionId);// audio session ID

} else if (memoryMode == javaAudioTrackFields.MODE_STATIC) {
// AudioTrack is using shared memory

if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
LOGE("Error creating AudioTrack in static mode: error creating mem heap base");
goto native_init_failure;
}

lpTrack->set(
atStreamType,// stream type
sampleRateInHertz,
format,// word length, PCM
channels,
frameCount,
0,// flags
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));
0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
lpJniStorage->mMemBase,// shared mem
true,// thread can call Java
sessionId);// audio session ID
}

...

}

&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&总结&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
stream type是audio native层管理stream用的。
再往下，HAL层中，并没有stream type的概念，
如函数AudioHardwareALSA::openOutputStream的参数中并不包含stream type的信息：
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
// set/get global audio parameters
//virtual status_t setParameters(const String8& keyValuePairs);
//virtual String8 getParameters(const String8& keys);

// Returns audio input buffer size according to parameters passed or 0 if one of the
// parameters is not supported
//virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channels);

/** This method creates and opens the audio hardware output stream */
virtual AudioStreamOut* openOutputStream(
uint32_t devices,
int *format=0,
uint32_t *channels=0,
uint32_t *sampleRate=0,
status_t *status=0);
----------------------------------------------------------------

audio flinger和play back thread分别用数组保存了各stream type的相关信息。
audio policy中保存了各stream type的strategy。
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
audio flinger中定义的stream type的信息的数组：
PlaybackThread::stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
struct stream_type_t {
stream_type_t()
: volume(1.0f),
mute(false)
{
}
float volume;
bool mute;
};
----------------------------------------------------------------
----------------------------------------------------------------
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&