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 } ... }

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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);

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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; };

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