Android音频数据抓取(杂音pop音定位必备)

framework层音频数据抓取：

1.抓取解码出来的原始PCM数据(自定义“config.audioData.afterDecoder”属性开关控制处)

frameworks/av/media/libmedia/AudioTrack.cpp

ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
    if (mTransfer != TRANSFER_SYNC || mIsTimed) {
        return INVALID_OPERATION;
    }

    if (isDirect()) {
        AutoMutex lock(mLock);
        int32_t flags = android_atomic_and(
                            ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
                            &mCblk->mFlags);
        if (flags & CBLK_INVALID) {
            return DEAD_OBJECT;
        }
    }

    if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
        // Sanity-check: user is most-likely passing an error code, and it would
        // make the return value ambiguous (actualSize vs error).
        ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);
        return BAD_VALUE;
    }

    size_t written = 0;
    Buffer audioBuffer;

    while (userSize >= mFrameSize) {
        audioBuffer.frameCount = userSize / mFrameSize;

        status_t err = obtainBuffer(&audioBuffer,
                blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
        if (err < 0) {
            if (written > 0) {
                break;
            }
            return ssize_t(err);
        }

        size_t toWrite = audioBuffer.size;
        memcpy(audioBuffer.i8, buffer, toWrite);
        buffer = ((const char *) buffer) + toWrite;
        userSize -= toWrite;
        written += toWrite;	
		
<span style="color:#ff0000;"><span style="white-space:pre">	</span>char buf[PROPERTY_VALUE_MAX];
        property_get("config.audioData.afterDecoder", buf, "");
	if( (0 == strcmp("buf","on")) && (toWrite !=0) ){
		FILE* fp = fopen("/data/audio/afterDecoder.pcm", "w+");
		fwrite(audioBuffer.i8,toWrite,1,fp);
                ... ...
	}</span>

      releaseBuffer(&audioBuffer);
    }
    return written;
}

或者

nsecs_t AudioTrack::processAudioBuffer()
{
    // Currently the AudioTrack thread is not created if there are no callbacks.
    // Would it ever make sense to run the thread, even without callbacks?
    // If so, then replace this by checks at each use for mCbf != NULL.
    LOG_ALWAYS_FATAL_IF(mCblk == NULL);

    mLock.lock();
    if (mAwaitBoost) {
        mAwaitBoost = false;
        mLock.unlock();
        static const int32_t kMaxTries = 5;
        int32_t tryCounter = kMaxTries;
        uint32_t pollUs = 10000;
        do {
            int policy = sched_getscheduler(0);
            if (policy == SCHED_FIFO || policy == SCHED_RR) {
                break;
            }
            usleep(pollUs);
            pollUs <<= 1;
        } while (tryCounter-- > 0);
        if (tryCounter < 0) {
            ALOGE("did not receive expected priority boost on time");
        }
        // Run again immediately
        return 0;
    }

    // Can only reference mCblk while locked
    int32_t flags = android_atomic_and(
        ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);

    // Check for track invalidation
    if (flags & CBLK_INVALID) {
        // for offloaded tracks restoreTrack_l() will just update the sequence and clear
        // AudioSystem cache. We should not exit here but after calling the callback so
        // that the upper layers can recreate the track
        if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
            status_t status __unused = restoreTrack_l("processAudioBuffer");
            // FIXME unused status
            // after restoration, continue below to make sure that the loop and buffer events
            // are notified because they have been cleared from mCblk->mFlags above.
        }
    }

    bool waitStreamEnd = mState == STATE_STOPPING;
    bool active = mState == STATE_ACTIVE;

    // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
    bool newUnderrun = false;
    if (flags & CBLK_UNDERRUN) {
#if 0
        // Currently in shared buffer mode, when the server reaches the end of buffer,
        // the track stays active in continuous underrun state.  It's up to the application
        // to pause or stop the track, or set the position to a new offset within buffer.
        // This was some experimental code to auto-pause on underrun.   Keeping it here
        // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
        if (mTransfer == TRANSFER_SHARED) {
            mState = STATE_PAUSED;
            active = false;
        }
#endif
        if (!mInUnderrun) {
            mInUnderrun = true;
            newUnderrun = true;
        }
    }

    // Get current position of server
    size_t position = updateAndGetPosition_l();

    // Manage marker callback
    bool markerReached = false;
    size_t markerPosition = mMarkerPosition;
    // FIXME fails for wraparound, need 64 bits
    if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
        mMarkerReached = markerReached = true;
    }

    // Determine number of new position callback(s) that will be needed, while locked
    size_t newPosCount = 0;
    size_t newPosition = mNewPosition;
    size_t updatePeriod = mUpdatePeriod;
    // FIXME fails for wraparound, need 64 bits
    if (updatePeriod > 0 && position >= newPosition) {
        newPosCount = ((position - newPosition) / updatePeriod) + 1;
        mNewPosition += updatePeriod * newPosCount;
    }

    // Cache other fields that will be needed soon
    uint32_t sampleRate = mSampleRate;
    float speed = mPlaybackRate.mSpeed;
    const uint32_t notificationFrames = mNotificationFramesAct;
    if (mRefreshRemaining) {
        mRefreshRemaining = false;
        mRemainingFrames = notificationFrames;
        mRetryOnPartialBuffer = false;
    }
    size_t misalignment = mProxy->getMisalignment();
    uint32_t sequence = mSequence;
    sp<AudioTrackClientProxy> proxy = mProxy;

    // Determine the number of new loop callback(s) that will be needed, while locked.
    int loopCountNotifications = 0;
    uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END

    if (mLoopCount > 0) {
        int loopCount;
        size_t bufferPosition;
        mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
        loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
        loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
        mLoopCountNotified = loopCount; // discard any excess notifications
    } else if (mLoopCount < 0) {
        // FIXME: We're not accurate with notification count and position with infinite looping
        // since loopCount from server side will always return -1 (we could decrement it).
        size_t bufferPosition = mStaticProxy->getBufferPosition();
        loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
        loopPeriod = mLoopEnd - bufferPosition;
    } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
        size_t bufferPosition = mStaticProxy->getBufferPosition();
        loopPeriod = mFrameCount - bufferPosition;
    }

    // These fields don't need to be cached, because they are assigned only by set():
    //     mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
    // mFlags is also assigned by createTrack_l(), but not the bit we care about.

    mLock.unlock();

    // get anchor time to account for callbacks.
    const nsecs_t timeBeforeCallbacks = systemTime();

    if (waitStreamEnd) {
        // FIXME:  Instead of blocking in proxy->waitStreamEndDone(), Callback thread
        // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
        // (and make sure we don't callback for more data while we're stopping).
        // This helps with position, marker notifications, and track invalidation.
        struct timespec timeout;
        timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
        timeout.tv_nsec = 0;

        status_t status = proxy->waitStreamEndDone(&timeout);
        switch (status) {
        case NO_ERROR:
        case DEAD_OBJECT:
        case TIMED_OUT:
            if (status != DEAD_OBJECT) {
                // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
                // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
                mCbf(EVENT_STREAM_END, mUserData, NULL);
            }
            {
                AutoMutex lock(mLock);
                // The previously assigned value of waitStreamEnd is no longer valid,
                // since the mutex has been unlocked and either the callback handler
                // or another thread could have re-started the AudioTrack during that time.
                waitStreamEnd = mState == STATE_STOPPING;
                if (waitStreamEnd) {
                    mState = STATE_STOPPED;
                    mReleased = 0;
                }
            }
            if (waitStreamEnd && status != DEAD_OBJECT) {
               return NS_INACTIVE;
            }
            break;
        }
        return 0;
    }

    // perform callbacks while unlocked
    if (newUnderrun) {
        mCbf(EVENT_UNDERRUN, mUserData, NULL);
    }
    while (loopCountNotifications > 0) {
        mCbf(EVENT_LOOP_END, mUserData, NULL);
        --loopCountNotifications;
    }
    if (flags & CBLK_BUFFER_END) {
        mCbf(EVENT_BUFFER_END, mUserData, NULL);
    }
    if (markerReached) {
        mCbf(EVENT_MARKER, mUserData, &markerPosition);
    }
    while (newPosCount > 0) {
        size_t temp = newPosition;
        mCbf(EVENT_NEW_POS, mUserData, &temp);
        newPosition += updatePeriod;
        newPosCount--;
    }

    if (mObservedSequence != sequence) {
        mObservedSequence = sequence;
        mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
        // for offloaded tracks, just wait for the upper layers to recreate the track
        if (isOffloadedOrDirect()) {
            return NS_INACTIVE;
        }
    }

    // if inactive, then don't run me again until re-started
    if (!active) {
        return NS_INACTIVE;
    }

    // Compute the estimated time until the next timed event (position, markers, loops)
    // FIXME only for non-compressed audio
    uint32_t minFrames = ~0;
    if (!markerReached && position < markerPosition) {
        minFrames = markerPosition - position;
    }
    if (loopPeriod > 0 && loopPeriod < minFrames) {
        // loopPeriod is already adjusted for actual position.
        minFrames = loopPeriod;
    }
    if (updatePeriod > 0) {
        minFrames = min(minFrames, uint32_t(newPosition - position));
    }

    // If > 0, poll periodically to recover from a stuck server.  A good value is 2.
    static const uint32_t kPoll = 0;
    if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
        minFrames = kPoll * notificationFrames;
    }

    // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
    static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
    const nsecs_t timeAfterCallbacks = systemTime();

    // Convert frame units to time units
    nsecs_t ns = NS_WHENEVER;
    if (minFrames != (uint32_t) ~0) {
        ns = framesToNanoseconds(minFrames, sampleRate, speed) + kWaitPeriodNs;
        ns -= (timeAfterCallbacks - timeBeforeCallbacks);  // account for callback time
        // TODO: Should we warn if the callback time is too long?
        if (ns < 0) ns = 0;
    }

    // If not supplying data by EVENT_MORE_DATA, then we're done
    if (mTransfer != TRANSFER_CALLBACK) {
        return ns;
    }

    // EVENT_MORE_DATA callback handling.
    // Timing for linear pcm audio data formats can be derived directly from the
    // buffer fill level.
    // Timing for compressed data is not directly available from the buffer fill level,
    // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
    // to return a certain fill level.

    struct timespec timeout;
    const struct timespec *requested = &ClientProxy::kForever;
    if (ns != NS_WHENEVER) {
        timeout.tv_sec = ns / 1000000000LL;
        timeout.tv_nsec = ns % 1000000000LL;
        ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
        requested = &timeout;
    }

    while (mRemainingFrames > 0) {

        Buffer audioBuffer;
        audioBuffer.frameCount = mRemainingFrames;
        size_t nonContig;
        status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
        LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
                "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
        requested = &ClientProxy::kNonBlocking;
        size_t avail = audioBuffer.frameCount + nonContig;
        ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
                mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
        if (err != NO_ERROR) {
            if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
                    (isOffloaded() && (err == DEAD_OBJECT))) {
                // FIXME bug 25195759
                return 1000000;
            }
            ALOGE("Error %d obtaining an audio buffer, giving up.", err);
            return NS_NEVER;
        }

        if (mRetryOnPartialBuffer && audio_is_linear_pcm(mFormat)) {
            mRetryOnPartialBuffer = false;
            if (avail < mRemainingFrames) {
                if (ns > 0) { // account for obtain time
                    const nsecs_t timeNow = systemTime();
                    ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
                }
                nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
                if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
                    ns = myns;
                }
                return ns;
            }
        }

        size_t reqSize = audioBuffer.size;
        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
        size_t writtenSize = audioBuffer.size;

<span style="color:#ff0000;">	char buf[PROPERTY_VALUE_MAX];
        property_get("config.audioData.afterDecoder", buf, "");
	if( (0 == strcmp("buf","on")) && (writtenSize !=0)){
		FILE* fp = fopen("/data/audio/afterDecoder.pcm", "w+");
		fwrite(mUserData,writtenSize,1,fp);</span>
<span style="white-space:pre">		</span>... ...
	}</span>

        // Sanity check on returned size
        if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
            ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
                    reqSize, ssize_t(writtenSize));
            return NS_NEVER;
        }

        if (writtenSize == 0) {
            // The callback is done filling buffers
            // Keep this thread going to handle timed events and
            // still try to get more data in intervals of WAIT_PERIOD_MS
            // but don't just loop and block the CPU, so wait

            // mCbf(EVENT_MORE_DATA, ...) might either
            // (1) Block until it can fill the buffer, returning 0 size on EOS.
            // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
            // (3) Return 0 size when no data is available, does not wait for more data.
            //
            // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
            // We try to compute the wait time to avoid a tight sleep-wait cycle,
            // especially for case (3).
            //
            // The decision to support (1) and (2) affect the sizing of mRemainingFrames
            // and this loop; whereas for case (3) we could simply check once with the full
            // buffer size and skip the loop entirely.

            nsecs_t myns;
            if (audio_is_linear_pcm(mFormat)) {
                // time to wait based on buffer occupancy
                const nsecs_t datans = mRemainingFrames <= avail ? 0 :
                        framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
                // audio flinger thread buffer size (TODO: adjust for fast tracks)
                const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
                // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
                myns = datans + (afns / 2);
            } else {
                // FIXME: This could ping quite a bit if the buffer isn't full.
                // Note that when mState is stopping we waitStreamEnd, so it never gets here.
                myns = kWaitPeriodNs;
            }
            if (ns > 0) { // account for obtain and callback time
                const nsecs_t timeNow = systemTime();
                ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
            }
            if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
                ns = myns;
            }
            return ns;
        }

        size_t releasedFrames = writtenSize / mFrameSize;
        audioBuffer.frameCount = releasedFrames;
        mRemainingFrames -= releasedFrames;
        if (misalignment >= releasedFrames) {
            misalignment -= releasedFrames;
        } else {
            misalignment = 0;
        }

        releaseBuffer(&audioBuffer);

        // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
        // if callback doesn't like to accept the full chunk
        if (writtenSize < reqSize) {
            continue;
        }

        // There could be enough non-contiguous frames available to satisfy the remaining request
        if (mRemainingFrames <= nonContig) {
            continue;
        }

#if 0
        // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
        // sum <= notificationFrames.  It replaces that series by at most two EVENT_MORE_DATA
        // that total to a sum == notificationFrames.
        if (0 < misalignment && misalignment <= mRemainingFrames) {
            mRemainingFrames = misalignment;
            return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
        }
#endif

    }
    mRemainingFrames = notificationFrames;
    mRetryOnPartialBuffer = true;

    // A lot has transpired since ns was calculated, so run again immediately and re-calculate
    return 0;
}

2.重采样之前的数据（"config.audioData.beforeResample"属性开关控制处）
frameworks/av/services/audioflinger/AudioResamplerSinc.cpp

size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
        AudioBufferProvider* provider)
{
    const Constants& c(*mConstants);
    const size_t headOffset = c.halfNumCoefs*CHANNELS;
    int16_t* impulse = mImpulse;
    uint32_t vRL = mVolumeRL;
    size_t inputIndex = mInputIndex;
    uint32_t phaseFraction = mPhaseFraction;
    uint32_t phaseIncrement = mPhaseIncrement;
    size_t outputIndex = 0;
    size_t outputSampleCount = outFrameCount * 2;
    size_t inFrameCount = getInFrameCountRequired(outFrameCount);

    while (outputIndex < outputSampleCount) {
        // buffer is empty, fetch a new one
        while (mBuffer.frameCount == 0) {
            mBuffer.frameCount = inFrameCount;
            provider->getNextBuffer(&mBuffer,calculateOutputPTS(outputIndex / 2));
            if (mBuffer.raw == NULL) {
                goto resample_exit;
            }
			
<span style="color:#ff0000;"><span style="white-space:pre">	</span>char buf[PROPERTY_VALUE_MAX];
	property_get("config.audioData.beforeResample", buf, "");
	if( 0 == strcmp("buf","on") ){
		FILE* fp = fopen("/data/audio/beforeResample.pcm", "w+");
		fwrite(mBuffer.i16,toWrite,1,fp);
		... ...
	}</span>
			
            const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
            if (phaseIndex == 1) {
                // read one frame
                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
            } else if (phaseIndex == 2) {
                // read 2 frames
                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
                inputIndex++;
                if (inputIndex >= mBuffer.frameCount) {
                    inputIndex -= mBuffer.frameCount;
                    provider->releaseBuffer(&mBuffer);
                } else {
                    read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
                }
            }
        }
        int16_t const * const in = mBuffer.i16;
        const size_t frameCount = mBuffer.frameCount;

        // Always read-in the first samples from the input buffer
        int16_t* head = impulse + headOffset;
        for (size_t i=0 ; i<CHANNELS ; i++) {
            head[i] = in[inputIndex*CHANNELS + i];
        }

        // handle boundary case
        while (CC_LIKELY(outputIndex < outputSampleCount)) {
            filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL);
            outputIndex += 2;

            phaseFraction += phaseIncrement;
            const size_t phaseIndex = phaseFraction >> kNumPhaseBits;
            for (size_t i=0 ; i<phaseIndex ; i++) {
                inputIndex++;
                if (inputIndex >= frameCount) {
                    goto done;  // need a new buffer
                }
                read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
            }
        }
done:
        // if done with buffer, save samples
        if (inputIndex >= frameCount) {
            inputIndex -= frameCount;
            provider->releaseBuffer(&mBuffer);
        }
    }

resample_exit:
    mImpulse = impulse;
    mInputIndex = inputIndex;
    mPhaseFraction = phaseFraction;
    return outputIndex / CHANNELS;
}

3.重采样之后的数据(“config.audioData.afterResample”属性开关控制处)
frameworks/av/services/threads.cpp

3.重采样之后的数据(“config.audioData.afterResample”属性开关控制处)
frameworks/av/services/threads.cpp

// shared by MIXER and DIRECT, overridden by DUPLICATING
ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
    // FIXME rewrite to reduce number of system calls
    mLastWriteTime = systemTime();
    mInWrite = true;
    ssize_t bytesWritten;
    const size_t offset = mCurrentWriteLength - mBytesRemaining;

    // If an NBAIO sink is present, use it to write the normal mixer's submix
    if (mNormalSink != 0) {

        const size_t count = mBytesRemaining / mFrameSize;

        ATRACE_BEGIN("write");
        // update the setpoint when AudioFlinger::mScreenState changes
        uint32_t screenState = AudioFlinger::mScreenState;
        if (screenState != mScreenState) {
            mScreenState = screenState;
            MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
            if (pipe != NULL) {
                pipe->setAvgFrames((mScreenState & 1) ?
                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
            }
        }
        ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
<span style="color:#ff0000;">
<span style="white-space:pre">	</span>char buf[PROPERTY_VALUE_MAX];
	property_get("config.audioData.afterResample", buf, "");
	if( 0 == strcmp("buf","on") ){
		FILE* fp = fopen("/data/audio/afterResample.pcm", "w+");
		fwrite((char *)mSinkBuffer + offset,framesWritten,1,fp);
		... ...
	}</span>
		
        ATRACE_END();
        if (framesWritten > 0) {
            bytesWritten = framesWritten * mFrameSize;
        } else {
            bytesWritten = framesWritten;
        }
        mLatchDValid = false;
        status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
        if (status == NO_ERROR) {
            size_t totalFramesWritten = mNormalSink->framesWritten();
            if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) {
                mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition;
                // mLatchD.mFramesReleased is set immediately before D is clocked into Q
                mLatchDValid = true;
            }
        }
    // otherwise use the HAL / AudioStreamOut directly
    } else {
        // Direct output and offload threads

        if (mUseAsyncWrite) {
            ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
            mWriteAckSequence += 2;
            mWriteAckSequence |= 1;
            ALOG_ASSERT(mCallbackThread != 0);
            mCallbackThread->setWriteBlocked(mWriteAckSequence);
        }
        // FIXME We should have an implementation of timestamps for direct output threads.
        // They are used e.g for multichannel PCM playback over HDMI.
        bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
        if (mUseAsyncWrite &&
                ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
            // do not wait for async callback in case of error of full write
            mWriteAckSequence &= ~1;
            ALOG_ASSERT(mCallbackThread != 0);
            mCallbackThread->setWriteBlocked(mWriteAckSequence);
        }
    }

    mNumWrites++;
    mInWrite = false;
    mStandby = false;
    return bytesWritten;
}

4.soundEffects音效处理前后音频数据抓取（未完待续）

HAL层音频数据抓取：

1.Ap侧写入pcm设备之前HAL层音频数据("config.audioData.HAL")
hardware/libhardware/modules/audio/audio_hw.c

static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
                         size_t bytes)
{
    ALOGV("out_write: bytes: %d", bytes);
	
<span style="white-space:pre">	</span>char buf[PROPERTY_VALUE_MAX];
	property_get("config.audioData.HAL", buf, "");
	if( 0 == strcmp("buf","on") ){
		FILE* fp = fopen("/data/audio/HAL.pcm", "w+");
		fwrite((buffer,bytes,1,fp);
		... ...
	}
	
    /* XXX: fake timing for audio output */
    usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
           out_get_sample_rate(&stream->common));
    return bytes;
}

DSP侧音频数据抓取：(未完待续)