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Commit c473ac09 authored by Atneya Nair's avatar Atneya Nair Committed by Android (Google) Code Review
Browse files

Merge "Consolidate libaudioclient buffers p1"

parents 95dd9990 03079276

cmds/stagefright/AudioPlayer.cpp

+1 −1
Original line number Diff line number Diff line
@@ -453,7 +453,7 @@ size_t AudioPlayer::AudioSinkCallback( 
}



size_t AudioPlayer::onMoreData(const AudioTrack::Buffer& buffer) {

    return fillBuffer(buffer.raw, buffer.size);

    return fillBuffer(buffer.data(), buffer.size());

}



void AudioPlayer::onStreamEnd() {

media/libaaudio/src/legacy/AudioStreamLegacy.cpp

+72 −7
Original line number Diff line number Diff line
@@ -76,6 +76,7 @@ size_t AudioStreamLegacy::onMoreData(const android::AudioTrack::Buffer& buffer) 
    // This takes advantage of them killing the stream when they see a size out of range.

    // That is an undocumented behavior.

    // TODO add to API in AudioRecord and AudioTrack

    // TODO(b/216175830) cleanup size re-computation

    const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;

    aaudio_data_callback_result_t callbackResult;

    (void) checkForDisconnectRequest(true);
@@ -83,7 +84,7 @@ size_t AudioStreamLegacy::onMoreData(const android::AudioTrack::Buffer& buffer) 
    // Note that this code assumes an AudioTrack::Buffer is the same as

    // AudioRecord::Buffer

    // TODO define our own AudioBuffer and pass it from the subclasses.

    size_t written = buffer.size;

    size_t written = buffer.size();

    if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) {

        ALOGW("%s() data, stream disconnected", __func__);

        // This will kill the stream and prevent it from being restarted.
@@ -96,23 +97,23 @@ size_t AudioStreamLegacy::onMoreData(const android::AudioTrack::Buffer& buffer) 
        // caused by Legacy callbacks running after the track is "stopped".

        written = 0;

    } else {

        if (buffer.frameCount == 0) {

        if (buffer.getFrameCount() == 0) {

            ALOGW("%s() data, frameCount is zero", __func__);

            return written;

        }



        // If the caller specified an exact size then use a block size adapter.

        if (mBlockAdapter != nullptr) {

            int32_t byteCount = buffer.frameCount * getBytesPerDeviceFrame();

            int32_t byteCount = buffer.getFrameCount() * getBytesPerDeviceFrame();

            callbackResult = mBlockAdapter->processVariableBlock(

                    static_cast<uint8_t*>(buffer.raw), byteCount);

                    buffer.data(), byteCount);

        } else {

            // Call using the AAudio callback interface.

            callbackResult = callDataCallbackFrames(static_cast<uint8_t *>(buffer.raw),

                                                    buffer.frameCount);

            callbackResult = callDataCallbackFrames(buffer.data(),

                                                    buffer.getFrameCount());

        }

        if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {

            written = buffer.frameCount * getBytesPerDeviceFrame();

            written = buffer.getFrameCount() * getBytesPerDeviceFrame();

        } else {

            if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {

                ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
@@ -134,6 +135,70 @@ size_t AudioStreamLegacy::onMoreData(const android::AudioTrack::Buffer& buffer) 
    return written;

}



// TODO (b/216175830) this method is duplicated in order to ease refactoring which will

// reconsolidate.

size_t AudioStreamLegacy::onMoreData(const android::AudioRecord::Buffer& buffer) {

    // This illegal size can be used to tell AudioRecord or AudioTrack to stop calling us.

    // This takes advantage of them killing the stream when they see a size out of range.

    // That is an undocumented behavior.

    // TODO add to API in AudioRecord and AudioTrack

    const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;

    aaudio_data_callback_result_t callbackResult;

    (void) checkForDisconnectRequest(true);



    // Note that this code assumes an AudioTrack::Buffer is the same as

    // AudioRecord::Buffer

    // TODO define our own AudioBuffer and pass it from the subclasses.

    size_t written = buffer.size();

    if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) {

        ALOGW("%s() data, stream disconnected", __func__);

        // This will kill the stream and prevent it from being restarted.

        // That is OK because the stream is disconnected.

        written = SIZE_STOP_CALLBACKS;

    } else if (!mCallbackEnabled.load()) {

        ALOGW("%s() no data because callback disabled, set size=0", __func__);

        // Do NOT use SIZE_STOP_CALLBACKS here because that will kill the stream and

        // prevent it from being restarted. This can occur because of a race condition

        // caused by Legacy callbacks running after the track is "stopped".

        written = 0;

    } else {

        if (buffer.getFrameCount() == 0) {

            ALOGW("%s() data, frameCount is zero", __func__);

            return written;

        }



        // If the caller specified an exact size then use a block size adapter.

        if (mBlockAdapter != nullptr) {

            int32_t byteCount = buffer.getFrameCount() * getBytesPerDeviceFrame();

            callbackResult = mBlockAdapter->processVariableBlock(

                    buffer.data(), byteCount);

        } else {

            // Call using the AAudio callback interface.

            callbackResult = callDataCallbackFrames(buffer.data(),

                                                    buffer.getFrameCount());

        }

        if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {

            written = buffer.getFrameCount() * getBytesPerDeviceFrame();

        } else {

            if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {

                ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);

            } else {

                ALOGW("%s() callback returned invalid result = %d",

                      __func__, callbackResult);

            }

            written = 0;

            systemStopInternal();

            // Disable the callback just in case the system keeps trying to call us.

            mCallbackEnabled.store(false);

        }



        if (updateStateMachine() != AAUDIO_OK) {

            forceDisconnect();

            mCallbackEnabled.store(false);

        }

    }

    return written;

}



aaudio_result_t AudioStreamLegacy::checkForDisconnectRequest(bool errorCallbackEnabled) {

    if (mRequestDisconnect.isRequested()) {

media/libaaudio/src/legacy/AudioStreamLegacy.h

+8 −2
Original line number Diff line number Diff line
@@ -17,9 +17,10 @@ 
#ifndef LEGACY_AUDIO_STREAM_LEGACY_H

#define LEGACY_AUDIO_STREAM_LEGACY_H



#include <media/AudioRecord.h>

#include <media/AudioSystem.h>

#include <media/AudioTimestamp.h>

#include <media/AudioTrack.h>

#include <media/AudioSystem.h>



#include <aaudio/AAudio.h>
@@ -57,7 +58,8 @@ typedef int32_t aaudio_callback_operation_t; 


class AudioStreamLegacy : public AudioStream,

                          public FixedBlockProcessor,

                          protected android::AudioTrack::IAudioTrackCallback {

                          protected android::AudioTrack::IAudioTrackCallback,

                          protected android::AudioRecord::IAudioRecordCallback {

public:

    AudioStreamLegacy();
@@ -82,7 +84,11 @@ public: 


protected:

    size_t onMoreData(const android::AudioTrack::Buffer& buffer) override;

    // TODO (b/216175830) this method is duplicated in order to ease refactoring which will

    // reconsolidate.

    size_t onMoreData(const android::AudioRecord::Buffer& buffer) override;

    void onNewIAudioTrack() override;

    void onNewIAudioRecord() override { onNewIAudioTrack(); }

    aaudio_result_t getBestTimestamp(clockid_t clockId,

                                     int64_t *framePosition,

                                     int64_t *timeNanoseconds,

media/libaaudio/src/legacy/AudioStreamRecord.cpp

+2 −27
Original line number Diff line number Diff line
@@ -37,10 +37,6 @@ using android::content::AttributionSourceState; 
using namespace android;

using namespace aaudio;



static void sCallbackWrapper(int event, void* userData, void* info) {

    static_cast<AudioStreamRecord*>(userData)->processCallback(event, info);

}



AudioStreamRecord::AudioStreamRecord()

    : AudioStreamLegacy()

    , mFixedBlockWriter(*this)
@@ -128,13 +124,11 @@ aaudio_result_t AudioStreamRecord::open(const AudioStreamBuilder& builder) 
    uint32_t notificationFrames = 0;



    // Setup the callback if there is one.

    AudioRecord::legacy_callback_t callback = nullptr;

    void *callbackData = nullptr;

    sp<AudioRecord::IAudioRecordCallback> callback;

    AudioRecord::transfer_type streamTransferType = AudioRecord::transfer_type::TRANSFER_SYNC;

    if (builder.getDataCallbackProc() != nullptr) {

        streamTransferType = AudioRecord::transfer_type::TRANSFER_CALLBACK;

        callback = sCallbackWrapper;

        callbackData = this;

        callback = sp<AudioRecord::IAudioRecordCallback>::fromExisting(this);

    }

    mCallbackBufferSize = builder.getFramesPerDataCallback();
@@ -181,7 +175,6 @@ aaudio_result_t AudioStreamRecord::open(const AudioStreamBuilder& builder) 
                channelMask,

                frameCount,

                callback,

                callbackData,

                notificationFrames,

                false /*threadCanCallJava*/,

                sessionId,
@@ -354,24 +347,6 @@ const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, in 
    }

}



void AudioStreamRecord::processCallback(int event, void *info) {

    switch (event) {

        case AudioRecord::EVENT_MORE_DATA:

        {

            AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(info);

            audioBuffer->size = onMoreData(*audioBuffer);

            break;

        }

            // Stream got rerouted so we disconnect.

        case AudioRecord::EVENT_NEW_IAUDIORECORD:

            onNewIAudioTrack();

            break;

        default:

            break;

    }

    return;

}



aaudio_result_t AudioStreamRecord::requestStart_l()

{

    if (mAudioRecord.get() == nullptr) {

media/libaudioclient/AudioRecord.cpp

+8 −8
Original line number Diff line number Diff line
@@ -267,7 +267,7 @@ class LegacyCallbackWrapper : public AudioRecord::IAudioRecordCallback { 
    size_t onMoreData(const AudioRecord::Buffer& buffer) override {

        AudioRecord::Buffer copy = buffer;

        mCallback(AudioRecord::EVENT_MORE_DATA, mData, &copy);

        return copy.size;

        return copy.size();

    }



    void onOverrun() override { mCallback(AudioRecord::EVENT_OVERRUN, mData, nullptr); }
@@ -1142,7 +1142,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_ 
    }

    if (mTransfer != TRANSFER_OBTAIN) {

        audioBuffer->frameCount = 0;

        audioBuffer->size = 0;

        audioBuffer->mSize = 0;

        audioBuffer->raw = NULL;

        if (nonContig != NULL) {

            *nonContig = 0;
@@ -1231,7 +1231,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r 
    } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));



    audioBuffer->frameCount = buffer.mFrameCount;

    audioBuffer->size = buffer.mFrameCount * mServerFrameSize;

    audioBuffer->mSize = buffer.mFrameCount * mServerFrameSize;

    audioBuffer->raw = buffer.mRaw;

    audioBuffer->sequence = oldSequence;

    if (nonContig != NULL) {
@@ -1308,7 +1308,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking) 


        size_t bytesRead = audioBuffer.frameCount * mFrameSize;

        memcpy_by_audio_format(buffer, mFormat, audioBuffer.raw, mServerConfig.format,

                               audioBuffer.size / mServerSampleSize);

                               audioBuffer.mSize / mServerSampleSize);

        buffer = ((char *) buffer) + bytesRead;

        userSize -= bytesRead;

        read += bytesRead;
@@ -1514,15 +1514,15 @@ nsecs_t AudioRecord::processAudioBuffer() 
        if (mServerConfig.format != mFormat) {

            buffer = &mFormatConversionBuffer;

            buffer->frameCount = audioBuffer.frameCount;

            buffer->size = buffer->frameCount * mFrameSize;

            buffer->mSize = buffer->frameCount * mFrameSize;

            buffer->sequence = audioBuffer.sequence;

            memcpy_by_audio_format(buffer->raw, mFormat, audioBuffer.raw,

                                   mServerConfig.format, audioBuffer.size / mServerSampleSize);

                                   mServerConfig.format, audioBuffer.size() / mServerSampleSize);

        }



        const size_t reqSize = buffer->size;

        const size_t reqSize = buffer->size();

        const size_t readSize = callback->onMoreData(*buffer);

        buffer->size = readSize;

        buffer->mSize = readSize;



        // Validate on returned size

        if (ssize_t(readSize) < 0 || readSize > reqSize) {