初始化AudioTrace学习笔记------深入理解Android

AudioTrack类管理并播放Java应用的音频资源。它将PCM音频缓冲流传输到音频硬件播放。使用write(byte[], int, int)和write(short[], int, int)方法推送音频流到AudioTrace对象。

AudioTrace实例有两种工作方式：静态，流态

流态模式，应用将连续的数据流使用write()函数写入AudioTrack。数据流将阻塞，当数据从Java层传递到Native层后返回，并放入Playback队列。流态模式适用于播放音频数据块：

• 因声音播放时间过长而不适宜存储
• 因音频数据特性(高采样录、bits/s)而不适宜存储
• 之前的音频数据正在播放，同时接收或产生新的数据

静态模式适用于可存储的很短的，播放延迟最小的声音。因此该模式适用于播放UI或者游戏中的声音。

根据创造方式，AudioTrack对象可决定是否初始化它的音频缓冲。缓冲的大小在构造时指定，并决定一个AudioTrack播放时间。

静态模式的AudioTrack，缓冲的大小是他可播放声音的最大长度。

流态模式的AudioTrack，数据将以长度小于总缓冲大小的小块写到硬件。

AuidoTrack->set()@AudioTrack.cpp

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)
{
    ......
    uint32_t channelCount = AudioSystem::popCount(channels);

    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
            sampleRate, format, channels, (AudioSystem::output_flags)flags);
    ......
    mVolume[LEFT] = 1.0f;
    mVolume[RIGHT] = 1.0f;
    mSendLevel = 0;
    mFrameCount = frameCount;
    mNotificationFramesReq = notificationFrames;
    mSessionId = sessionId;
    mAuxEffectId = 0;

    // create the IAudioTrack
    status_t status = createTrack(streamType, sampleRate, format, channelCount,-------------------------------步骤1
                                  frameCount, flags, sharedBuffer, output, true);

    ......
    if (cbf != 0) {
        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);----------------------------------步骤2
        if (mAudioTrackThread == 0) {
          LOGE("Could not create callback thread");
          return NO_INIT;
        }
    }
    ......
    mSharedBuffer = sharedBuffer;--------------------------------------------------------------------------步骤3
    ......
    mCbf = cbf;
    mUserData = user;
    ......
    mAudioSession = -1;
    return NO_ERROR;
}

AuidoTrack->createTrack()@AudioTrack.cpp

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)
{
    status_t status;
    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();//获取AudioFlinger服务
    ......
    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
                                                      streamType,
                                                      sampleRate,
                                                      format,
                                                      channelCount,
                                                      frameCount,
                                                      ((uint16_t)flags) << 16,
                                                      sharedBuffer,
                                                      output,
                                                      &mSessionId,
                                                      &status);

    ......
    sp<IMemory> cblk = track->getCblk();
    ......
    mAudioTrack.clear();
    mAudioTrack = track;
    mCblkMemory.clear();
    mCblkMemory = cblk;
    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());//获取AudioTrack共享buffer首地址
    mCblk->flags |= CBLK_DIRECTION_OUT;
    if (sharedBuffer == 0) {
        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
    } else {
        mCblk->buffers = sharedBuffer->pointer();
         // Force buffer full condition as data is already present in shared memory
        mCblk->stepUser(mCblk->frameCount);
    }
    ......
    mAudioTrack->attachAuxEffect(mAuxEffectId);
    ......
    return NO_ERROR;
}

AudioFlinger->createTrack()@AudioFlinger.cpp

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;

    {
        Mutex::Autolock _l(mLock);
        PlaybackThread *thread = checkPlaybackThread_l(output);-----------------------获取output对应的playbackthread
        PlaybackThread *effectThread = NULL;
        if (thread == NULL) {
            LOGE("unknown output thread");
            lStatus = BAD_VALUE;
            goto Exit;
        }

        wclient = mClients.valueFor(pid);

        if (wclient != NULL) {
            client = wclient.promote();
        } else {
            client = new Client(this, pid);----------------------------------------------初始化1024*1024大小的memoryheap
            mClients.add(pid, client);----------------------------------------------------pid，client对
        }

        LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);-------sessionId不为空
        if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
                if (mPlaybackThreads.keyAt(i) != output) {
                    // prevent same audio session on different output threads
                    uint32_t sessions = t->hasAudioSession(*sessionId);
                    if (sessions & PlaybackThread::TRACK_SESSION) {
                        lStatus = BAD_VALUE;
                        goto Exit;
                    }
                    // check if an effect with same session ID is waiting for a track to be created
                    if (sessions & PlaybackThread::EFFECT_SESSION) {
                        effectThread = t.get();
`                    }
                }
            }
            lSessionId = *sessionId;
        } else {
            // if no audio session id is provided, create one here
            lSessionId = nextUniqueId();------------------------------------------------*sessionId赋值
            if (sessionId != NULL) {
                *sessionId = lSessionId;
            }
        }

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

        // move effect chain to this output thread if an effect on same session was waiting
        // for a track to be created
        if (lStatus == NO_ERROR && effectThread != NULL) {
            Mutex::Autolock _dl(thread->mLock);
            Mutex::Autolock _sl(effectThread->mLock);
            moveEffectChain_l(lSessionId, effectThread, thread, true);
        }
    }
    if (lStatus == NO_ERROR) {
        trackHandle = new TrackHandle(track);--------------------------------------------TrackHandle初始化
    } else {
        // remove local strong reference to Client before deleting the Track so that the Client
        // destructor is called by the TrackBase destructor with mLock held
        client.clear();
        track.clear();
    }

Exit:
    if(status) {
        *status = lStatus;
    }
    return trackHandle;
}

AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l()

AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
{
    PlaybackThread *thread = NULL;
    if (mPlaybackThreads.indexOfKey(output) >= 0) {
        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();-----------------mPlaybackThreads.add(id, thread)@openOutput;id为output号，thread为mixerthread
    }
    return thread;
}

class Client

AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
    :   RefBase(),
        mAudioFlinger(audioFlinger),
        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
        mPid(pid)
{
    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}

PlaybackThread::createTrack_l()

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)
{
    sp<Track> track;
    status_t lStatus;
    ......
    { // scope for mLock
        Mutex::Autolock _l(mLock);

        // all tracks in same audio session must share the same routing strategy otherwise
        // conflicts will happen when tracks are moved from one output to another by audio policy
        // manager
        uint32_t strategy =
                AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType);
        for (size_t i = 0; i < mTracks.size(); ++i) {
            sp<Track> t = mTracks[i];
            if (t != 0) {
                if (sessionId == t->sessionId() &&
                        strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) {
                    lStatus = BAD_VALUE;
                    goto Exit;
                }
            }
        }

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

Exit:
    if(status) {
        *status = lStatus;
    }
    return track;
}

class Track

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)
{
    if (mCblk != NULL) {
        sp<ThreadBase> baseThread = thread.promote();
        if (baseThread != 0) {
            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
            mName = playbackThread->getTrackName_l();
            mMainBuffer = playbackThread->mixBuffer();
        }
        ......
        LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
        if (mName < 0) {
            LOGE("no more track names available");
        }
        ......
        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
        mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
    }
    ......
}

AudioFlinger::ThreadBase::TrackBase::TrackBase

AudioFlinger::ThreadBase::TrackBase::TrackBase(
            const wp<ThreadBase>& thread,
            const sp<Client>& client,
            uint32_t sampleRate,
            int format,
            int channelCount,
            int frameCount,
            uint32_t flags,
            const sp<IMemory>& sharedBuffer,
            int sessionId)
    :   RefBase(),
        mThread(thread),
        mClient(client),
        mCblk(0),
        mFrameCount(0),
        mState(IDLE),
        mClientTid(-1),
        mFormat(format),
        mFlags(flags & ~SYSTEM_FLAGS_MASK),
        mSessionId(sessionId)
{
    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());

    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
   size_t size = sizeof(audio_track_cblk_t);
   size_t bufferSize = 0;
   if ( (format == AudioSystem::PCM_16_BIT) ||
        (format == AudioSystem::PCM_8_BIT))
   {
       bufferSize = frameCount*channelCount*sizeof(int16_t);
   }
   else if (format == AudioSystem::AMR_NB)
   {
       bufferSize = frameCount*channelCount*32; // full rate frame size
   }
   ......
   if (sharedBuffer == 0) {
       size += bufferSize;
   }

   if (client != NULL) {
        mCblkMemory = client->heap()->allocate(size);------------------------------------------在已有的heap上分配空间
        if (mCblkMemory != 0) {
            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
            if (mCblk) { // construct the shared structure in-place.
                new(mCblk) audio_track_cblk_t();-----------------------------------------------placement new，音频信息头
                // clear all buffers
                mCblk->frameCount = frameCount;
                mCblk->sampleRate = sampleRate;
                mCblk->channelCount = (uint8_t)channelCount;
                if (sharedBuffer == 0) {
                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);------------------------mBuffer音频数据的存储起始位置
                    // Change for Codec type
                    if ( (format == AudioSystem::PCM_16_BIT) ||
                    (format == AudioSystem::PCM_8_BIT))
                    {
                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
                    }
                    else if (format == AudioSystem::AMR_NB)
                    {
                      memset(mBuffer, 0, frameCount*channelCount*32); // full rate frame size
                    }
                    ......
                    // Force underrun condition to avoid false underrun callback until first data is
                    // written to buffer (other flags are cleared)
                    mCblk->flags = CBLK_UNDERRUN_ON;
                } else {
                    mBuffer = sharedBuffer->pointer();
                }
                mBufferEnd = (uint8_t *)mBuffer + bufferSize;-----------------------------------音频数据的结束为止
            }
            ......
        } 
        ......
   } 
   ......
}

audio_track_cblk_t

struct audio_track_cblk_t
{

    // The data members are grouped so that members accessed frequently and in the same context
    // are in the same line of data cache.
                Mutex       lock;
                Condition   cv;
    volatile    uint32_t    user;
    volatile    uint32_t    server;
                uint32_t    userBase;
                uint32_t    serverBase;
                void*       buffers;
                uint32_t    frameCount;
                // Cache line boundary
                uint32_t    loopStart;
                uint32_t    loopEnd;
                int         loopCount;
    volatile    union {
                    uint16_t    volume[2];
                    uint32_t    volumeLR;
                };
                uint32_t    sampleRate;
                // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
                // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
                // 16 bit because data is converted to 16 bit before being stored in buffer

                uint8_t     frameSize;
                uint8_t     channelCount;
                uint16_t    flags;

                uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
                uint16_t    waitTimeMs;      // Cumulated wait time

                uint16_t    sendLevel;
                uint16_t    reserved;
                // Cache line boundary (32 bytes)
                            audio_track_cblk_t();
                uint32_t    stepUser(uint32_t frameCount);
                bool        stepServer(uint32_t frameCount);
                void*       buffer(uint32_t offset) const;
                uint32_t    framesAvailable();
                uint32_t    framesAvailable_l();
                uint32_t    framesReady();
};

int AudioFlinger::MixerThread::getTrackName_l()

int AudioFlinger::MixerThread::getTrackName_l()
{
    return mAudioMixer->getTrackName();
}

int AudioMixer::getTrackName()

 int AudioMixer::getTrackName()
 {
    uint32_t names = mTrackNames;
    uint32_t mask = 1;
    int n = 0;
    while (names & mask) {
        mask <<= 1;
        n++;
    }
    if (mask) {
        LOGV("add track (%d)", n);
        mTrackNames |= mask;
        return TRACK0 + n;
    }
    return -1;
 }

int16_t     *mixBuffer()

 int16_t     *mixBuffer() { return mMixBuffer; };

mMixBuffer是在PlayBackThread的构造函数中调用readOutputParameters()中初始化的

void AudioFlinger::PlaybackThread::readOutputParameters()

void AudioFlinger::PlaybackThread::readOutputParameters()
{
    ......
    // FIXME - Current mixer implementation only supports stereo output: Always
    // Allocate a stereo buffer even if HW output is mono.
    if (mMixBuffer != NULL) delete[] mMixBuffer;
    mMixBuffer = new int16_t[mFrameCount * 2];
    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
    ......
}

class TrackHandle

AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
    : BnAudioTrack(),
      mTrack(track)
{
}

TrackHandle使用Proxy模式，真正干活的AudioFlinger::PlaybackThread::Track，最终发给底层音频驱动设备。

AuidoTrack->createTrack()函数最终经过AudioFlinger->createTrack()函数获得了AudioFlinger::TrackHandle。mAudioTrack记录下了TrackHandle，将来AudioTrack就靠它与下层AudioFlinger协作，调用start，stop等函数。

mCblkMemory记录下共享buffer首地址。

track->getCblk()

sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
    return mTrack->getCblk();
}

sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
    return mCblkMemory;
}

mCblkMemory@TrackBase是在TrackBase构造函数中init的，mCblkMemory = client->heap()->allocate(size);。client也就是前面Client类中预先分配的1M的heap空间。