AudioRecord的开始录音 startRecording和 RecordThread

start录音流程走到（Android 应用阶段）
Jni android_media_AudioRecord_start ---> mAudioRecord->start ---> (AudioFlinger阶段开始跨进程)RecordHandle->start ---> RecordTrack->start --->RecordThread->start ---> （录音线程阶段）RecordThread-->threadLoop 走一轮还是来到了RecordThread，直奔解析RecordThread的start。

status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
                                           AudioSystem::sync_event_t event,
                                           int triggerSession)
{
    ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
    sp<ThreadBase> strongMe = this;
    status_t status = NO_ERROR;

  /* 基本默认都用SYNC_EVENT_NONE */
    if (event == AudioSystem::SYNC_EVENT_NONE) {
        recordTrack->clearSyncStartEvent();
    } else if (event != AudioSystem::SYNC_EVENT_SAME) {
        ......
    }

    {
        // This section is a rendezvous between binder thread executing start() and RecordThread
        AutoMutex lock(mLock);
        if (mActiveTracks.indexOf(recordTrack) >= 0) {
  /**这里做判断，如果当前recordTrack已经start过了的，但是在暂停状态的话，
直接调整为ACTIVE状态，并且返回。
*/
           if (recordTrack->mState == TrackBase::PAUSING) {
                ALOGV("active record track PAUSING -> ACTIVE");
                recordTrack->mState = TrackBase::ACTIVE;
            } else {
                ALOGV("active record track state %d", recordTrack->mState);
            }
            return status;
        }

        // TODO consider other ways of handling this, such as changing the state to :STARTING and
        //      adding the track to mActiveTracks after returning from AudioSystem::startInput(),
        //      or using a separate command thread
        recordTrack->mState = TrackBase::STARTING_1;
        mActiveTracks.add(recordTrack); //当前recordTrack存起来
        mActiveTracksGen++;
        status_t status = NO_ERROR;
        if (recordTrack->isExternalTrack()) {
            mLock.unlock();
            status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId());
            mLock.lock();
            // FIXME should verify that recordTrack is still in mActiveTracks
            if (status != NO_ERROR) {
                mActiveTracks.remove(recordTrack);
                mActiveTracksGen++;
                recordTrack->clearSyncStartEvent();
                ALOGV("RecordThread::start error %d", status);
                return status;
            }
        }
        // Catch up with current buffer indices if thread is already running.
        // This is what makes a new client discard all buffered data.  If the track's mRsmpInFront
        // was initialized to some value closer to the thread's mRsmpInFront, then the track could
        // see previously buffered data before it called start(), but with greater risk of overrun.
    /* reset一下mResamplerBufferProvider 和mRecordBufferConverter*/
        recordTrack->mResamplerBufferProvider->reset();
        // clear any converter state as new data will be discontinuous
        recordTrack->mRecordBufferConverter->reset();
        recordTrack->mState = TrackBase::STARTING_2;
        // signal thread to start
        mWaitWorkCV.broadcast();
        if (mActiveTracks.indexOf(recordTrack) < 0) {
            ALOGV("Record failed to start");
            status = BAD_VALUE;
            goto startError;
        }
        return status;
    }

startError:
    if (recordTrack->isExternalTrack()) {
        AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId());
    }
    recordTrack->clearSyncStartEvent();
    // FIXME I wonder why we do not reset the state here?
    return status;
}

AudioSystem::startInput 基本上涉及到AudioSystem都是和AudioPolicy有关的，依然是由AudioPolicyManager::startInput(audio_io_handle_t input,
audio_session_t session)来处理这档事，。现在这里又要绕一圈audioPolicy 到 audioFlinger createAudioPatch创建AuioPatch来连接Source和sink端？不太懂这个AudioPatch概念。AudioPatch连接开始后和状态更新后唤醒录音线程mWaitWorkCV.broadcast(); 到这一步可以看RecordThread->threadLoop了。

bool AudioFlinger::RecordThread::threadLoop()
{
    nsecs_t lastWarning = 0;
  /*  暂停快速截取和让AudioStreamIn 输入流待机状态*/
    inputStandBy();

reacquire_wakelock:
    sp<RecordTrack> activeTrack;
    int activeTracksGen;
    {
        Mutex::Autolock _l(mLock);
        size_t size = mActiveTracks.size();
        activeTracksGen = mActiveTracksGen;
        if (size > 0) {
            // FIXME an arbitrary choice
            activeTrack = mActiveTracks[0];
            acquireWakeLock_l(activeTrack->uid());
            if (size > 1) {
                SortedVector<int> tmp;
                for (size_t i = 0; i < size; i++) {
                    tmp.add(mActiveTracks[i]->uid());
                }
                updateWakeLockUids_l(tmp);
            }
        } else {
            acquireWakeLock_l(-1);
        }
    }

    // used to request a deferred sleep, to be executed later while mutex is unlocked
    uint32_t sleepUs = 0;

    // loop while there is work to do
    for (;;) {
        Vector< sp<EffectChain> > effectChains;

        // sleep with mutex unlocked
        if (sleepUs > 0) {
            ATRACE_BEGIN("sleep");
            usleep(sleepUs);
            ATRACE_END();
            sleepUs = 0;
        }

        // activeTracks accumulates a copy of a subset of mActiveTracks
        Vector< sp<RecordTrack> > activeTracks;

        // reference to the (first and only) active fast track
        sp<RecordTrack> fastTrack;

        // reference to a fast track which is about to be removed
        sp<RecordTrack> fastTrackToRemove;

        { // scope for mLock
            Mutex::Autolock _l(mLock);

            processConfigEvents_l();

            // check exitPending here because checkForNewParameters_l() and
            // checkForNewParameters_l() can temporarily release mLock
            if (exitPending()) {
                break;
            }

            // if no active track(s), then standby and release wakelock
            size_t size = mActiveTracks.size();
            if (size == 0) {
              /*  如果没有正在进行的recordtrack，让input等待状态和释放wakelock
然后睡眠，等待mWaitWorkCV.broadcast() start录音后会唤醒和拥有一个active的track*/
                standbyIfNotAlreadyInStandby();
                // exitPending() can't become true here
                releaseWakeLock_l();
                ALOGV("RecordThread: loop stopping");
                // go to sleep
                mWaitWorkCV.wait(mLock);
                ALOGV("RecordThread: loop starting");
                goto reacquire_wakelock;
            }

            if (mActiveTracksGen != activeTracksGen) {
                activeTracksGen = mActiveTracksGen;
                SortedVector<int> tmp;
                for (size_t i = 0; i < size; i++) {
                    tmp.add(mActiveTracks[i]->uid());
                }
                updateWakeLockUids_l(tmp);
            }

            bool doBroadcast = false;
          /* 这段for循环是为了找出active或准备active状态的recordtrack*/
            for (size_t i = 0; i < size; ) {

                activeTrack = mActiveTracks[i];
              /*  再检测下track是不是已经被关闭了的，清理掉*/
                if (activeTrack->isTerminated()) {
                    if (activeTrack->isFastTrack()) {
                        ALOG_ASSERT(fastTrackToRemove == 0);
                        fastTrackToRemove = activeTrack;
                    }
                    removeTrack_l(activeTrack);
                    mActiveTracks.remove(activeTrack);
                    mActiveTracksGen++;
                    size--;
                    continue;
                }

                TrackBase::track_state activeTrackState = activeTrack->mState;
            /*  状态流程选择开始录音后状态初始是TrackBase::STARTING_2*/
                switch (activeTrackState) {

                case TrackBase::PAUSING:
                    mActiveTracks.remove(activeTrack);
                    mActiveTracksGen++;
                    doBroadcast = true;
                    size--;
                    continue;

                case TrackBase::STARTING_1:
                    sleepUs = 10000;
                    i++;
                    continue;

                case TrackBase::STARTING_2: //here
                    doBroadcast = true;
                    mStandby = false;
                    activeTrack->mState = TrackBase::ACTIVE;
                    break;

                case TrackBase::ACTIVE:
                    break;

                case TrackBase::IDLE:
                    i++;
                    continue;

                default:
                    LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
                }
              /* 涮选出是Active或者是 STARTING_2状态的recordtrack记录*/
                activeTracks.add(activeTrack);
                i++;

                if (activeTrack->isFastTrack()) {
                    ALOG_ASSERT(!mFastTrackAvail);
                    ALOG_ASSERT(fastTrack == 0);
                    fastTrack = activeTrack;
                }
            }
            if (doBroadcast) {
            /* 取消暂停的信号*/
                mStartStopCond.broadcast();
            }

            // sleep if there are no active tracks to process
            if (activeTracks.size() == 0) {
                if (sleepUs == 0) {
                    sleepUs = kRecordThreadSleepUs;
                }
                continue;
            }
            sleepUs = 0;

            lockEffectChains_l(effectChains);
        }

        // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0

        size_t size = effectChains.size();
        for (size_t i = 0; i < size; i++) {
            // thread mutex is not locked, but effect chain is locked
            effectChains[i]->process_l();
        }

        // Push a new fast capture state if fast capture is not already running, or cblk change
 /*  这段还没详细看*/
        if (mFastCapture != 0) {
            FastCaptureStateQueue *sq = mFastCapture->sq();
            FastCaptureState *state = sq->begin();
            bool didModify = false;
            FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
            if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
                    (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
                if (state->mCommand == FastCaptureState::COLD_IDLE) {
                    int32_t old = android_atomic_inc(&mFastCaptureFutex);
                    if (old == -1) {
                        (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
                    }
                }
                state->mCommand = FastCaptureState::READ_WRITE;
                didModify = true;
            }
            audio_track_cblk_t *cblkOld = state->mCblk;
            audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
            if (cblkNew != cblkOld) {
                state->mCblk = cblkNew;
                // block until acked if removing a fast track
                if (cblkOld != NULL) {
                    block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
                }
                didModify = true;
            }
            sq->end(didModify);
            if (didModify) {
                sq->push(block);
            }
        }

        // now run the fast track destructor with thread mutex unlocked
        fastTrackToRemove.clear();

        // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
        // Only the client(s) that are too slow will overrun. But if even the fastest client is too
        // slow, then this RecordThread will overrun by not calling HAL read often enough.
        // If destination is non-contiguous, first read past the nominal end of buffer, then
        // copy to the right place.  Permitted because mRsmpInBuffer was over-allocated.

        int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
        ssize_t framesRead;

        // If an NBAIO source is present, use it to read the normal capture's data
  /* 有俩种方式去读取录音数据管道和HAL 输入流读取 */
        if (mPipeSource != 0) {
            size_t framesToRead = mBufferSize / mFrameSize;
            framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
                    framesToRead, AudioBufferProvider::kInvalidPTS);
            if (framesRead == 0) {
                // since pipe is non-blocking, simulate blocking input
                sleepUs = (framesToRead * 1000000LL) / mSampleRate;
            }
        // otherwise use the HAL / AudioStreamIn directly
        } else {  // 这里mRsmpInBuffer 记住这个采样buffer指针，从流拿出的数据放在这里。
            ssize_t bytesRead = mInput->stream->read(mInput->stream,
                    (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
            if (bytesRead < 0) {
                framesRead = bytesRead;
            } else {
/* 记录有多少帧PCM数据 */
                framesRead = bytesRead / mFrameSize;
            }
        }

        if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
            ALOGE("read failed: framesRead=%d", framesRead);
            // Force input into standby so that it tries to recover at next read attempt
            inputStandBy();
            sleepUs = kRecordThreadSleepUs;
        }
        if (framesRead <= 0) {
            goto unlock;
        }
        ALOG_ASSERT(framesRead > 0);

        if (mTeeSink != 0) { 
      /*官方功能teeSink  会把原始PCM数据写入指定的文件里 */
            (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
        }
        // If destination is non-contiguous, we now correct for reading past end of buffer.
        { /* 这部分比较难看懂，大概是如果mRsmpInFramesP2指向的前面多少帧数据已经被用过了，不需要了，则把后面新加入的尾部数据直接覆盖到前面去？？？循环有效使用内存，点赞 */
            size_t part1 = mRsmpInFramesP2 - rear;
            if ((size_t) framesRead > part1) {
                memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
                        (framesRead - part1) * mFrameSize);
            }
        }
    /* 更新rear尾部帧数量*/
        rear = mRsmpInRear += framesRead;

        size = activeTracks.size();
        // loop over each active track   这里一个for，将当前每个activetrack的处理一下
        for (size_t i = 0; i < size; i++) {
            activeTrack = activeTracks[i];
        
          /*快速模式的track 不走转换和共享客户端数据 */
            // skip fast tracks, as those are handled directly by FastCapture
            if (activeTrack->isFastTrack()) {
                continue;
            }
            // TODO: This code probably should be moved to RecordTrack.
            // TODO: Update the activeTrack buffer converter in case of reconfigure.
      
            // loop over getNextBuffer to handle circular sink
            /* 这段做了转换工作和交付给客户端AudioRecord共享PCM数据的工作*/
            for (;;) {
               
                activeTrack->mSink.frameCount = ~0;
                /* 说明的是这里的getNextBuffer仅仅是通过serverProxy申请了一个循环利用的和共享内存buffer*/
                status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
                size_t framesOut = activeTrack->mSink.frameCount;
                LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));

                // check available frames and handle overrun conditions
                // if the record track isn't draining fast enough.
                bool hasOverrun;
                size_t framesIn;
                activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
                if (hasOverrun) {
                    overrun = OVERRUN_TRUE;
                }
                if (framesOut == 0 || framesIn == 0) {
                    break;
                }

                // Don't allow framesOut to be larger than what is possible with resampling
                // from framesIn.
                // This isn't strictly necessary but helps limit buffer resizing in
                // RecordBufferConverter.  TODO: remove when no longer needed.
                framesOut = min(framesOut,
                        destinationFramesPossible(
                                framesIn, mSampleRate, activeTrack->mSampleRate));
                // process frames from the RecordThread buffer provider to the RecordTrack buffer
              /* 把mResamplerBufferProvider实际是从RecordThread的recordThread->mRsmpInBuffer拿到数据转换到activeTrack->mSink.raw*/
                framesOut = activeTrack->mRecordBufferConverter->convert(
                        activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);

                if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
                    overrun = OVERRUN_FALSE;
                }

                if (activeTrack->mFramesToDrop == 0) {
                    if (framesOut > 0) {
                        activeTrack->mSink.frameCount = framesOut;
                        activeTrack->releaseBuffer(&activeTrack->mSink);
                    }
                } else {
                    // FIXME could do a partial drop of framesOut
                    if (activeTrack->mFramesToDrop > 0) {
                        activeTrack->mFramesToDrop -= framesOut;
                        if (activeTrack->mFramesToDrop <= 0) {
                            activeTrack->clearSyncStartEvent();
                        }
                    } else {
                        activeTrack->mFramesToDrop += framesOut;
                        if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
                                activeTrack->mSyncStartEvent->isCancelled()) {
                            ALOGW("Synced record %s, session %d, trigger session %d",
                                  (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
                                  activeTrack->sessionId(),
                                  (activeTrack->mSyncStartEvent != 0) ?
                                          activeTrack->mSyncStartEvent->triggerSession() : 0);
                            activeTrack->clearSyncStartEvent();
                        }
                    }
                }

                if (framesOut == 0) {
                    break;
                }
            }

            switch (overrun) {
            case OVERRUN_TRUE:
                // client isn't retrieving buffers fast enough
                if (!activeTrack->setOverflow()) {
                    nsecs_t now = systemTime();
                    // FIXME should lastWarning per track?
                    if ((now - lastWarning) > kWarningThrottleNs) {
                        ALOGW("RecordThread: buffer overflow");
                        lastWarning = now;
                    }
                }
                break;
            case OVERRUN_FALSE:
                activeTrack->clearOverflow();
                break;
            case OVERRUN_UNKNOWN:
                break;
            }

        }

unlock:
        // enable changes in effect chain
        unlockEffectChains(effectChains);
        // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
    }

    standbyIfNotAlreadyInStandby();

    {
        Mutex::Autolock _l(mLock);
        for (size_t i = 0; i < mTracks.size(); i++) {
            sp<RecordTrack> track = mTracks[i];
            track->invalidate();
        }
        mActiveTracks.clear();
        mActiveTracksGen++;
        mStartStopCond.broadcast();
    }

    releaseWakeLock();

    ALOGV("RecordThread %p exiting", this);
    return false;
}

代码里有解说，到这里，有疑问应用程序客户端的AudioRecord没看到共享内存的写入是吧，
status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 分析这里即可。

status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
        int64_t pts __unused)
{
    ServerProxy::Buffer buf;
    buf.mFrameCount = buffer->frameCount;
    status_t status = mServerProxy->obtainBuffer(&buf);
    buffer->frameCount = buf.mFrameCount;
    buffer->raw = buf.mRaw;
    if (buf.mFrameCount == 0) {
        // FIXME also wake futex so that overrun is noticed more quickly
        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
    }
    return status;
}

实际走的是mServerProxy->obtainBuffer
在/av/media/libmedia/AudioTrackShared.cpp ServerProxy类实现,
AudioRecordServerProxy初始化传进来的sp<IMemory> cblk的参与分配循环利用共享内存，看了下用到了原子操作实现了数据完整性，这段代码研读起来麻烦，就只看看。

status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
{
    LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0);
    if (mIsShutdown) {
        goto no_init;
    }
    {
    audio_track_cblk_t* cblk = mCblk;
    // compute number of frames available to write (AudioTrack) or read (AudioRecord),
    // or use previous cached value from framesReady(), with added barrier if it omits.
    int32_t front;
    int32_t rear;
    // See notes on barriers at ClientProxy::obtainBuffer()
    if (mIsOut) {
        int32_t flush = cblk->u.mStreaming.mFlush;
        rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
        front = cblk->u.mStreaming.mFront;
        if (flush != mFlush) {
            // effectively obtain then release whatever is in the buffer
            const size_t overflowBit = mFrameCountP2 << 1;
            const size_t mask = overflowBit - 1;
            int32_t newFront = (front & ~mask) | (flush & mask);
            ssize_t filled = rear - newFront;
            if (filled >= (ssize_t)overflowBit) {
                // front and rear offsets span the overflow bit of the p2 mask
                // so rebasing newFront on the front offset is off by the overflow bit.
                // adjust newFront to match rear offset.
                ALOGV("flush wrap: filled %zx >= overflowBit %zx", filled, overflowBit);
                newFront += overflowBit;
                filled -= overflowBit;
            }
            // Rather than shutting down on a corrupt flush, just treat it as a full flush
            if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
                ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
                        "filled %zd=%#x",
                        mFlush, flush, front, rear,
                        (unsigned)mask, newFront, filled, (unsigned)filled);
                newFront = rear;
            }
            mFlush = flush;
            android_atomic_release_store(newFront, &cblk->u.mStreaming.mFront);
            // There is no danger from a false positive, so err on the side of caution
            if (true /*front != newFront*/) {
                int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
                if (!(old & CBLK_FUTEX_WAKE)) {
                    (void) syscall(__NR_futex, &cblk->mFutex,
                            mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
                }
            }
            front = newFront;
        }
    } else {
        front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
        rear = cblk->u.mStreaming.mRear;
    }
    ssize_t filled = rear - front;
    // pipe should not already be overfull
    if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
        ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
        mIsShutdown = true;
    }
    if (mIsShutdown) {
        goto no_init;
    }
    // don't allow filling pipe beyond the nominal size
    size_t availToServer;
    if (mIsOut) {
        availToServer = filled;
        mAvailToClient = mFrameCount - filled;
    } else {
        availToServer = mFrameCount - filled;
        mAvailToClient = filled;
    }
    // 'availToServer' may be non-contiguous, so return only the first contiguous chunk
    size_t part1;
    if (mIsOut) {
        front &= mFrameCountP2 - 1;
        part1 = mFrameCountP2 - front;
    } else {
        rear &= mFrameCountP2 - 1;
        part1 = mFrameCountP2 - rear;
    }
    if (part1 > availToServer) {
        part1 = availToServer;
    }
    size_t ask = buffer->mFrameCount;
    if (part1 > ask) {
        part1 = ask;
    }
    // is assignment redundant in some cases?
    buffer->mFrameCount = part1;
    buffer->mRaw = part1 > 0 ?
            &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL;
    buffer->mNonContig = availToServer - part1;
    // After flush(), allow releaseBuffer() on a previously obtained buffer;
    // see "Acknowledge any pending flush()" in audioflinger/Tracks.cpp.
    if (!ackFlush) {
        mUnreleased = part1;
    }
    return part1 > 0 ? NO_ERROR : WOULD_BLOCK;
    }
no_init:
    buffer->mFrameCount = 0;
    buffer->mRaw = NULL;
    buffer->mNonContig = 0;
    mUnreleased = 0;
    return NO_INIT;
}

AudioRecord 获取read 录音音频数据

前面既然有ServerProxy，那客户端自然有ClientProxy，AudioRecord这边也是直接可以跨进程来读取由IMemory共享出来的内存。
java 代码上面使用方法
int readSize = audioRecord.read(recordData, 0, recordbuffsize);
实测可以直接看下面
status_t ClientProxy::obtainBuffer(Buffer * buffer, const struct timespec *requested,struct timespec *elapsed)
同样是在/av/media/libmedia/AudioTrackShared.cpp ClientProxy类实现。