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Commit 7f54ebed authored by TreeHugger Robot's avatar TreeHugger Robot Committed by Automerger Merge Worker
Browse files

Merge "aaudio: free endpoint to prevent crashes" into rvc-dev am: 6e89a070 

Change-Id: I7c0cb0295300e1293caf556273f0cb2655880ab8
parents 620e98a6 6e89a070

media/libaaudio/src/client/AudioEndpoint.cpp

+15 −32
Original line number Diff line number Diff line
@@ -32,19 +32,12 @@ using namespace aaudio; 
#define RIDICULOUSLY_LARGE_FRAME_SIZE        4096



AudioEndpoint::AudioEndpoint()

    : mUpCommandQueue(nullptr)

    , mDataQueue(nullptr)

    , mFreeRunning(false)

    : mFreeRunning(false)

    , mDataReadCounter(0)

    , mDataWriteCounter(0)

{

}



AudioEndpoint::~AudioEndpoint() {

    delete mDataQueue;

    delete mUpCommandQueue;

}



// TODO Consider moving to a method in RingBufferDescriptor

static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,

                                                  const RingBufferDescriptor *descriptor) {
@@ -144,7 +137,7 @@ aaudio_result_t AudioEndpoint::configure(const EndpointDescriptor *pEndpointDesc 
        return AAUDIO_ERROR_INTERNAL;

    }



    mUpCommandQueue = new FifoBuffer(

    mUpCommandQueue = std::make_unique<FifoBuffer>(

            descriptor->bytesPerFrame,

            descriptor->capacityInFrames,

            descriptor->readCounterAddress,
@@ -173,7 +166,7 @@ aaudio_result_t AudioEndpoint::configure(const EndpointDescriptor *pEndpointDesc 
                                  ? &mDataWriteCounter

                                  : descriptor->writeCounterAddress;



    mDataQueue = new FifoBuffer(

    mDataQueue = std::make_unique<FifoBuffer>(

            descriptor->bytesPerFrame,

            descriptor->capacityInFrames,

            readCounterAddress,
@@ -194,18 +187,15 @@ int32_t AudioEndpoint::getEmptyFramesAvailable(WrappingBuffer *wrappingBuffer) { 
    return mDataQueue->getEmptyRoomAvailable(wrappingBuffer);

}



int32_t AudioEndpoint::getEmptyFramesAvailable()

{

int32_t AudioEndpoint::getEmptyFramesAvailable() {

    return mDataQueue->getEmptyFramesAvailable();

}



int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer)

{

int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer) {

    return mDataQueue->getFullDataAvailable(wrappingBuffer);

}



int32_t AudioEndpoint::getFullFramesAvailable()

{

int32_t AudioEndpoint::getFullFramesAvailable() {

    return mDataQueue->getFullFramesAvailable();

}
@@ -217,29 +207,24 @@ void AudioEndpoint::advanceReadIndex(int32_t deltaFrames) { 
    mDataQueue->advanceReadIndex(deltaFrames);

}



void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead)

{

void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead) {

    mDataQueue->setReadCounter(framesRead);

}



fifo_counter_t AudioEndpoint::getDataReadCounter()

{

fifo_counter_t AudioEndpoint::getDataReadCounter() const {

    return mDataQueue->getReadCounter();

}



void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead)

{

void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead) {

    mDataQueue->setWriteCounter(framesRead);

}



fifo_counter_t AudioEndpoint::getDataWriteCounter()

{

fifo_counter_t AudioEndpoint::getDataWriteCounter() const {

    return mDataQueue->getWriteCounter();

}



int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames,

                                            int32_t *actualFrames)

{

                                            int32_t *actualFrames) {

    if (requestedFrames < ENDPOINT_DATA_QUEUE_SIZE_MIN) {

        requestedFrames = ENDPOINT_DATA_QUEUE_SIZE_MIN;

    }
@@ -248,19 +233,17 @@ int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames, 
    return AAUDIO_OK;

}



int32_t AudioEndpoint::getBufferSizeInFrames() const

{

int32_t AudioEndpoint::getBufferSizeInFrames() const {

    return mDataQueue->getThreshold();

}



int32_t AudioEndpoint::getBufferCapacityInFrames() const

{

int32_t AudioEndpoint::getBufferCapacityInFrames() const {

    return (int32_t)mDataQueue->getBufferCapacityInFrames();

}



void AudioEndpoint::dump() const {

    ALOGD("data readCounter  = %lld", (long long) mDataQueue->getReadCounter());

    ALOGD("data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());

    ALOGD("data readCounter  = %lld", (long long) getDataReadCounter());

    ALOGD("data writeCounter = %lld", (long long) getDataWriteCounter());

}



void AudioEndpoint::eraseDataMemory() {

media/libaaudio/src/client/AudioEndpoint.h

+4 −5
Original line number Diff line number Diff line
@@ -35,7 +35,6 @@ class AudioEndpoint { 


public:

    AudioEndpoint();

    virtual ~AudioEndpoint();



    /**

     * Configure based on the EndPointDescriptor_t.
@@ -67,11 +66,11 @@ public: 
     */

    void setDataReadCounter(android::fifo_counter_t framesRead);



    android::fifo_counter_t getDataReadCounter();

    android::fifo_counter_t getDataReadCounter() const;



    void setDataWriteCounter(android::fifo_counter_t framesWritten);



    android::fifo_counter_t getDataWriteCounter();

    android::fifo_counter_t getDataWriteCounter() const;



    /**

     * The result is not valid until after configure() is called.
@@ -94,8 +93,8 @@ public: 
    void dump() const;



private:

    android::FifoBuffer    *mUpCommandQueue;

    android::FifoBuffer    *mDataQueue;

    std::unique_ptr<android::FifoBuffer> mUpCommandQueue;

    std::unique_ptr<android::FifoBuffer> mDataQueue;

    bool                    mFreeRunning;

    android::fifo_counter_t mDataReadCounter; // only used if free-running

    android::fifo_counter_t mDataWriteCounter; // only used if free-running

media/libaaudio/src/client/AudioStreamInternal.cpp

+38 −23
Original line number Diff line number Diff line
@@ -58,7 +58,6 @@ using namespace aaudio; 
AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface  &serviceInterface, bool inService)

        : AudioStream()

        , mClockModel()

        , mAudioEndpoint()

        , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)

        , mInService(inService)

        , mServiceInterface(serviceInterface)
@@ -74,7 +73,6 @@ AudioStreamInternal::~AudioStreamInternal() { 
aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {



    aaudio_result_t result = AAUDIO_OK;

    int32_t capacity;

    int32_t framesPerBurst;

    int32_t framesPerHardwareBurst;

    AAudioStreamRequest request;
@@ -173,7 +171,8 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { 
    }



    // Configure endpoint based on descriptor.

    result = mAudioEndpoint.configure(&mEndpointDescriptor, getDirection());

    mAudioEndpoint = std::make_unique<AudioEndpoint>();

    result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());

    if (result != AAUDIO_OK) {

        goto error;

    }
@@ -201,9 +200,10 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { 
    }

    mFramesPerBurst = framesPerBurst; // only save good value



    capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;

    if (capacity < mFramesPerBurst || capacity > MAX_BUFFER_CAPACITY_IN_FRAMES) {

        ALOGE("%s - bufferCapacity out of range = %d", __func__, capacity);

    mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;

    if (mBufferCapacityInFrames < mFramesPerBurst

            || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {

        ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);

        result = AAUDIO_ERROR_OUT_OF_RANGE;

        goto error;

    }
@@ -239,7 +239,7 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { 
    // You can use this offset to reduce glitching.

    // You can also use this offset to force glitching. By iterating over multiple

    // values you can reveal the distribution of the hardware timing jitter.

    if (mAudioEndpoint.isFreeRunning()) { // MMAP?

    if (mAudioEndpoint->isFreeRunning()) { // MMAP?

        int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)

                ? AAudioProperty_getOutputMMapOffsetMicros()

                : AAudioProperty_getInputMMapOffsetMicros();
@@ -251,7 +251,7 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) { 
        mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;

    }



    setBufferSize(capacity / 2); // Default buffer size to match Q

    setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q



    setState(AAUDIO_STREAM_STATE_OPEN);
@@ -280,6 +280,11 @@ aaudio_result_t AudioStreamInternal::release_l() { 


        mServiceInterface.closeStream(serviceStreamHandle);

        mCallbackBuffer.reset();



        // Update local frame counters so we can query them after releasing the endpoint.

        getFramesRead();

        getFramesWritten();

        mAudioEndpoint.reset();

        result = mEndPointParcelable.close();

        aaudio_result_t result2 = AudioStream::release_l();

        return (result != AAUDIO_OK) ? result : result2;
@@ -538,7 +543,7 @@ aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *mes 
        case AAUDIO_SERVICE_EVENT_DISCONNECTED:

            // Prevent hardware from looping on old data and making buzzing sounds.

            if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {

                mAudioEndpoint.eraseDataMemory();

                mAudioEndpoint->eraseDataMemory();

            }

            result = AAUDIO_ERROR_DISCONNECTED;

            setState(AAUDIO_STREAM_STATE_DISCONNECTED);
@@ -564,7 +569,10 @@ aaudio_result_t AudioStreamInternal::drainTimestampsFromService() { 


    while (result == AAUDIO_OK) {

        AAudioServiceMessage message;

        if (mAudioEndpoint.readUpCommand(&message) != 1) {

        if (!mAudioEndpoint) {

            break;

        }

        if (mAudioEndpoint->readUpCommand(&message) != 1) {

            break; // no command this time, no problem

        }

        switch (message.what) {
@@ -592,7 +600,10 @@ aaudio_result_t AudioStreamInternal::processCommands() { 


    while (result == AAUDIO_OK) {

        AAudioServiceMessage message;

        if (mAudioEndpoint.readUpCommand(&message) != 1) {

        if (!mAudioEndpoint) {

            break;

        }

        if (mAudioEndpoint->readUpCommand(&message) != 1) {

            break; // no command this time, no problem

        }

        switch (message.what) {
@@ -625,7 +636,7 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames 
    const char * fifoName = "aaRdy";

    ATRACE_BEGIN(traceName);

    if (ATRACE_ENABLED()) {

        int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();

        int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();

        ATRACE_INT(fifoName, fullFrames);

    }
@@ -654,7 +665,7 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames 
        if (timeoutNanoseconds == 0) {

            break; // don't block

        } else if (wakeTimeNanos != 0) {

            if (!mAudioEndpoint.isFreeRunning()) {

            if (!mAudioEndpoint->isFreeRunning()) {

                // If there is software on the other end of the FIFO then it may get delayed.

                // So wake up just a little after we expect it to be ready.

                wakeTimeNanos += mWakeupDelayNanos;
@@ -679,12 +690,12 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames 
                ALOGW("processData(): past deadline by %d micros",

                      (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));

                mClockModel.dump();

                mAudioEndpoint.dump();

                mAudioEndpoint->dump();

                break;

            }



            if (ATRACE_ENABLED()) {

                int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();

                int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();

                ATRACE_INT(fifoName, fullFrames);

                int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;

                ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
@@ -696,7 +707,7 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames 
    }



    if (ATRACE_ENABLED()) {

        int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();

        int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();

        ATRACE_INT(fifoName, fullFrames);

    }
@@ -730,11 +741,15 @@ aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) { 
        adjustedFrames = std::min(maximumSize, adjustedFrames);

    }



    if (mAudioEndpoint) {

        // Clip against the actual size from the endpoint.

        int32_t actualFrames = 0;

    mAudioEndpoint.setBufferSizeInFrames(maximumSize, &actualFrames);

        // Set to maximum size so we can write extra data when ready in order to reduce glitches.

        // The amount we keep in the buffer is controlled by mBufferSizeInFrames.

        mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);

        // actualFrames should be <= actual maximum size of endpoint

        adjustedFrames = std::min(actualFrames, adjustedFrames);

    }



    mBufferSizeInFrames = adjustedFrames;

    ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
@@ -746,7 +761,7 @@ int32_t AudioStreamInternal::getBufferSize() const { 
}



int32_t AudioStreamInternal::getBufferCapacity() const {

    return mAudioEndpoint.getBufferCapacityInFrames();

    return mBufferCapacityInFrames;

}



int32_t AudioStreamInternal::getFramesPerBurst() const {
@@ -759,5 +774,5 @@ aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) { 
}



bool AudioStreamInternal::isClockModelInControl() const {

    return isActive() && mAudioEndpoint.isFreeRunning() && mClockModel.isRunning();

    return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();

}

media/libaaudio/src/client/AudioStreamInternal.h

+7 −1
Original line number Diff line number Diff line
@@ -155,7 +155,8 @@ protected: 


    IsochronousClockModel    mClockModel;      // timing model for chasing the HAL



    AudioEndpoint            mAudioEndpoint;   // source for reads or sink for writes

    std::unique_ptr<AudioEndpoint> mAudioEndpoint;   // source for reads or sink for writes



    aaudio_handle_t          mServiceStreamHandle; // opaque handle returned from service



    int32_t                  mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
@@ -178,6 +179,9 @@ protected: 


    float                    mStreamVolume = 1.0f;



    int64_t                  mLastFramesWritten = 0;

    int64_t                  mLastFramesRead = 0;



private:

    /*

     * Asynchronous write with data conversion.
@@ -207,6 +211,8 @@ private: 
    int32_t                  mDeviceChannelCount = 0;



    int32_t                  mBufferSizeInFrames = 0; // local threshold to control latency

    int32_t                  mBufferCapacityInFrames = 0;





};

media/libaaudio/src/client/AudioStreamInternalCapture.cpp

+22 −19
Original line number Diff line number Diff line
@@ -42,8 +42,8 @@ AudioStreamInternalCapture::AudioStreamInternalCapture(AAudioServiceInterface & 
AudioStreamInternalCapture::~AudioStreamInternalCapture() {}



void AudioStreamInternalCapture::advanceClientToMatchServerPosition() {

    int64_t readCounter = mAudioEndpoint.getDataReadCounter();

    int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();

    int64_t readCounter = mAudioEndpoint->getDataReadCounter();

    int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();



    // Bump offset so caller does not see the retrograde motion in getFramesRead().

    int64_t offset = readCounter - writeCounter;
@@ -53,7 +53,7 @@ void AudioStreamInternalCapture::advanceClientToMatchServerPosition() { 


    // Force readCounter to match writeCounter.

    // This is because we cannot change the write counter in the hardware.

    mAudioEndpoint.setDataReadCounter(writeCounter);

    mAudioEndpoint->setDataReadCounter(writeCounter);

}



// Write the data, block if needed and timeoutMillis > 0
@@ -86,7 +86,7 @@ aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t 
    }

    // If we have gotten this far then we have at least one timestamp from server.



    if (mAudioEndpoint.isFreeRunning()) {

    if (mAudioEndpoint->isFreeRunning()) {

        //ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");

        // Update data queue based on the timing model.

        // Jitter in the DSP can cause late writes to the FIFO.
@@ -95,7 +95,7 @@ aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t 
        // that the DSP could have written the data.

        int64_t estimatedRemoteCounter = mClockModel.convertLatestTimeToPosition(currentNanoTime);

        // TODO refactor, maybe use setRemoteCounter()

        mAudioEndpoint.setDataWriteCounter(estimatedRemoteCounter);

        mAudioEndpoint->setDataWriteCounter(estimatedRemoteCounter);

    }



    // This code assumes that we have already received valid timestamps.
@@ -108,8 +108,8 @@ aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t 


    // If the capture buffer is full beyond capacity then consider it an overrun.

    // For shared streams, the xRunCount is passed up from the service.

    if (mAudioEndpoint.isFreeRunning()

        && mAudioEndpoint.getFullFramesAvailable() > mAudioEndpoint.getBufferCapacityInFrames()) {

    if (mAudioEndpoint->isFreeRunning()

        && mAudioEndpoint->getFullFramesAvailable() > mAudioEndpoint->getBufferCapacityInFrames()) {

        mXRunCount++;

        if (ATRACE_ENABLED()) {

            ATRACE_INT("aaOverRuns", mXRunCount);
@@ -143,7 +143,7 @@ aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t 
                // Calculate frame position based off of the readCounter because

                // the writeCounter might have just advanced in the background,

                // causing us to sleep until a later burst.

                int64_t nextPosition = mAudioEndpoint.getDataReadCounter() + mFramesPerBurst;

                int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + mFramesPerBurst;

                wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);

            }

                break;
@@ -166,7 +166,7 @@ aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer, 
    uint8_t *destination = (uint8_t *) buffer;

    int32_t framesLeft = numFrames;



    mAudioEndpoint.getFullFramesAvailable(&wrappingBuffer);

    mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);



    // Read data in one or two parts.

    for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
@@ -208,26 +208,29 @@ aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer, 
    }



    int32_t framesProcessed = numFrames - framesLeft;

    mAudioEndpoint.advanceReadIndex(framesProcessed);

    mAudioEndpoint->advanceReadIndex(framesProcessed);



    //ALOGD("readNowWithConversion() returns %d", framesProcessed);

    return framesProcessed;

}



int64_t AudioStreamInternalCapture::getFramesWritten() {

    if (mAudioEndpoint) {

        const int64_t framesWrittenHardware = isClockModelInControl()

                ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())

            : mAudioEndpoint.getDataWriteCounter();

                : mAudioEndpoint->getDataWriteCounter();

        // Add service offset and prevent retrograde motion.

        mLastFramesWritten = std::max(mLastFramesWritten,

                                      framesWrittenHardware + mFramesOffsetFromService);

    }

    return mLastFramesWritten;

}



int64_t AudioStreamInternalCapture::getFramesRead() {

    int64_t frames = mAudioEndpoint.getDataReadCounter() + mFramesOffsetFromService;

    //ALOGD("getFramesRead() returns %lld", (long long)frames);

    return frames;

    if (mAudioEndpoint) {

        mLastFramesRead = mAudioEndpoint->getDataReadCounter() + mFramesOffsetFromService;

    }

    return mLastFramesRead;

}



// Read data from the stream and pass it to the callback for processing.