oboe 从使用到源码详解

#include <oboe/Oboe.h>
#include <math.h>
using namespace oboe;

class OboeSinePlayer: public oboe::AudioStreamDataCallback {
public:

    virtual ~OboeSinePlayer() = default;

    // Call this from Activity onResume()
    int32_t startAudio() {
        std::lock_guard<std::mutex> lock(mLock);
        oboe::AudioStreamBuilder builder;
        // The builder set methods can be chained for convenience.
        Result result = builder.setSharingMode(oboe::SharingMode::Exclusive)
                ->setPerformanceMode(oboe::PerformanceMode::LowLatency)
                ->setChannelCount(kChannelCount)
                ->setSampleRate(kSampleRate)
        ->setSampleRateConversionQuality(oboe::SampleRateConversionQuality::Medium)
                ->setFormat(oboe::AudioFormat::Float)
                ->setDataCallback(this)
                ->openStream(mStream);
    if (result != Result::OK) return (int32_t) result;
    
        // Typically, start the stream after querying some stream information, as well as some input from the user
        result = outStream->requestStart();
    return (int32_t) result;
    }
   
    // Call this from Activity onPause()
    void stopAudio() {
        // Stop, close and delete in case not already closed.
        std::lock_guard<std::mutex> lock(mLock);
        if (mStream) {
            mStream->stop();
            mStream->close();
            mStream.reset();
        }
    }

    oboe::DataCallbackResult onAudioReady(oboe::AudioStream *oboeStream, void *audioData, int32_t numFrames) override {
        float *floatData = (float *) audioData;
        for (int i = 0; i < numFrames; ++i) {
            float sampleValue = kAmplitude * sinf(mPhase);
            for (int j = 0; j < kChannelCount; j++) {
                floatData[i * kChannelCount + j] = sampleValue;
            }
            mPhase += mPhaseIncrement;
            if (mPhase >= kTwoPi) mPhase -= kTwoPi;
        }
        return oboe::DataCallbackResult::Continue;
    }

private:
    std::mutex         mLock;
    std::shared_ptr<oboe::AudioStream> mStream;

    // Stream params
    static int constexpr kChannelCount = 2;
    static int constexpr kSampleRate = 48000;
    // Wave params, these could be instance variables in order to modify at runtime
    static float constexpr kAmplitude = 0.5f;
    static float constexpr kFrequency = 440;
    static float constexpr kPI = M_PI;
    static float constexpr kTwoPi = kPI * 2;
    static double constexpr mPhaseIncrement = kFrequency * kTwoPi / (double) kSampleRate;
    // Keeps track of where the wave is
    float mPhase = 0.0;
};

Result AudioStreamBuilder::openStream(AudioStream **streamPP) {
    auto result = isValidConfig();
    if (result != Result::OK) {
        LOGW("%s() invalid config %d", __func__, result);
        return result;
    }

    LOGI("%s() %s -------- %s --------",
         __func__, getDirection() == Direction::Input ? "INPUT" : "OUTPUT", getVersionText());

    if (streamPP == nullptr) {
        return Result::ErrorNull;
    }
    *streamPP = nullptr;

    AudioStream *streamP = nullptr;

    // Maybe make a FilterInputStream.
    AudioStreamBuilder childBuilder(*this);
    // Check need for conversion and modify childBuilder for optimal stream.
    bool conversionNeeded = QuirksManager::getInstance().isConversionNeeded(*this, childBuilder); // 如果允许调整的话，内部会针对通道数，采样率等进行调整
    // Do we need to make a child stream and convert.
   // 这儿是一个递归，针对需要调整的场景，可以再创建一个自己去构造，因为childBuilder已经调整了，因此不会陷入无限递归
    if (conversionNeeded) {
        AudioStream *tempStream;
        result = childBuilder.openStream(&tempStream);
        if (result != Result::OK) {
            return result;
        }

        if (isCompatible(*tempStream)) {
            // The child stream would work as the requested stream so we can just use it directly.
            *streamPP = tempStream;
            return result;
        } else {
            AudioStreamBuilder parentBuilder = *this;
            // Build a stream that is as close as possible to the childStream.
            if (getFormat() == oboe::AudioFormat::Unspecified) {
                parentBuilder.setFormat(tempStream->getFormat());
            }
            if (getChannelCount() == oboe::Unspecified) {
                parentBuilder.setChannelCount(tempStream->getChannelCount());
            }
            if (getSampleRate() == oboe::Unspecified) {
                parentBuilder.setSampleRate(tempStream->getSampleRate());
            }
            if (getFramesPerDataCallback() == oboe::Unspecified) {
                parentBuilder.setFramesPerCallback(tempStream->getFramesPerDataCallback());
            }

            // Use childStream in a FilterAudioStream.
            LOGI("%s() create a FilterAudioStream for data conversion.", __func__);
            FilterAudioStream *filterStream = new FilterAudioStream(parentBuilder, tempStream);
            result = filterStream->configureFlowGraph();
            if (result !=  Result::OK) {
                filterStream->close();
                delete filterStream;
                // Just open streamP the old way.
            } else {
                streamP = static_cast<AudioStream *>(filterStream);
            }
        }
    }

    if (streamP == nullptr) {
        streamP = build();
        if (streamP == nullptr) {
            return Result::ErrorNull;
        }
    }
 // 这个逻辑是oboe的设备兼容性方案
    // If MMAP has a problem in this case then disable it temporarily.
    bool wasMMapOriginallyEnabled = AAudioExtensions::getInstance().isMMapEnabled();
    bool wasMMapTemporarilyDisabled = false;
    if (wasMMapOriginallyEnabled) {
        bool isMMapSafe = QuirksManager::getInstance().isMMapSafe(childBuilder);
        if (!isMMapSafe) {
            AAudioExtensions::getInstance().setMMapEnabled(false);
            wasMMapTemporarilyDisabled = true;
        }
    }
    result = streamP->open();
    if (wasMMapTemporarilyDisabled) {
        AAudioExtensions::getInstance().setMMapEnabled(wasMMapOriginallyEnabled); // restore original
    }
    if (result == Result::OK) {

        int32_t  optimalBufferSize = -1;
        // Use a reasonable default buffer size.
        if (streamP->getDirection() == Direction::Input) {
            // For input, small size does not improve latency because the stream is usually
            // run close to empty. And a low size can result in XRuns so always use the maximum.
            optimalBufferSize = streamP->getBufferCapacityInFrames();
        } else if (streamP->getPerformanceMode() == PerformanceMode::LowLatency
                && streamP->getDirection() == Direction::Output)  { // Output check is redundant.
            optimalBufferSize = streamP->getFramesPerBurst() *
                                    kBufferSizeInBurstsForLowLatencyStreams;
        }
        if (optimalBufferSize >= 0) {
            auto setBufferResult = streamP->setBufferSizeInFrames(optimalBufferSize);
            if (!setBufferResult) {
                LOGW("Failed to setBufferSizeInFrames(%d). Error was %s",
                     optimalBufferSize,
                     convertToText(setBufferResult.error()));
            }
        }

        *streamPP = streamP;
    } else {
        delete streamP;
    }
    return result;
}

Result AudioInputStreamOpenSLES::open() {
    ...
    // 初始化 opensl engine
    Result oboeResult = AudioStreamOpenSLES::open();
    if (Result::OK != oboeResult) return oboeResult;

    SLuint32 bitsPerSample = static_cast<SLuint32>(getBytesPerSample() * kBitsPerByte);

    // configure audio sink
    mBufferQueueLength = calculateOptimalBufferQueueLength(); // 计算最佳buffer，需要至少是2倍的回调数据
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
            SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,    // locatorType
            static_cast<SLuint32>(mBufferQueueLength)};   // numBuffers

    // Define the audio data format.
    SLDataFormat_PCM format_pcm = {
            SL_DATAFORMAT_PCM,       // formatType
            static_cast<SLuint32>(mChannelCount),           // numChannels
            static_cast<SLuint32>(mSampleRate * kMillisPerSecond), // milliSamplesPerSec
            bitsPerSample,                      // bitsPerSample
            bitsPerSample,                      // containerSize;
            channelCountToChannelMask(mChannelCount), // channelMask
            getDefaultByteOrder(),
    };

    SLDataSink audioSink = {&amp;loc_bufq, &amp;format_pcm};

    /**
     * API 23 (Marshmallow) introduced support for floating-point data representation and an
     * extended data format type: SLAndroidDataFormat_PCM_EX for recording streams (playback streams
     * got this in API 21). If running on API 23+ use this newer format type, creating it from our
     * original format.
     */
    SLAndroidDataFormat_PCM_EX format_pcm_ex;
    if (getSdkVersion() >= __ANDROID_API_M__) {
        SLuint32 representation = OpenSLES_ConvertFormatToRepresentation(getFormat());
        // Fill in the format structure.
        format_pcm_ex = OpenSLES_createExtendedFormat(format_pcm, representation);
        // Use in place of the previous format.
        audioSink.pFormat = &amp;format_pcm_ex;
    }


    // configure audio source
    SLDataLocator_IODevice loc_dev = {SL_DATALOCATOR_IODEVICE,
                                      SL_IODEVICE_AUDIOINPUT,
                                      SL_DEFAULTDEVICEID_AUDIOINPUT,
                                      NULL};
    SLDataSource audioSrc = {&amp;loc_dev, NULL};

    SLresult result = EngineOpenSLES::getInstance().createAudioRecorder(&amp;mObjectInterface,
                                                                        &amp;audioSrc,
                                                                        &amp;audioSink);

    if (SL_RESULT_SUCCESS != result) {
        LOGE("createAudioRecorder() result:%s", getSLErrStr(result));
        goto error;
    }

    // Configure the stream.
    result = (*mObjectInterface)->GetInterface(mObjectInterface,
                                            SL_IID_ANDROIDCONFIGURATION,
                                            &amp;configItf);

    if (SL_RESULT_SUCCESS != result) {
        LOGW("%s() GetInterface(SL_IID_ANDROIDCONFIGURATION) failed with %s",
             __func__, getSLErrStr(result));
    } else {
        if (getInputPreset() == InputPreset::VoicePerformance) {
            LOGD("OpenSL ES does not support InputPreset::VoicePerformance. Use VoiceRecognition.");
            mInputPreset = InputPreset::VoiceRecognition;
        }
        SLuint32 presetValue = OpenSLES_convertInputPreset(getInputPreset());
        result = (*configItf)->SetConfiguration(configItf,
                                         SL_ANDROID_KEY_RECORDING_PRESET,
                                         &amp;presetValue,
                                         sizeof(SLuint32));
        if (SL_RESULT_SUCCESS != result
                &amp;&amp; presetValue != SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION) {
            presetValue = SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION;
            LOGD("Setting InputPreset %d failed. Using VoiceRecognition instead.", getInputPreset());
            mInputPreset = InputPreset::VoiceRecognition;
            (*configItf)->SetConfiguration(configItf,
                                             SL_ANDROID_KEY_RECORDING_PRESET,
                                             &amp;presetValue,
                                             sizeof(SLuint32));
        }

        result = configurePerformanceMode(configItf);
        if (SL_RESULT_SUCCESS != result) {
            goto error;
        }
    }

    result = (*mObjectInterface)->Realize(mObjectInterface, SL_BOOLEAN_FALSE);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("Realize recorder object result:%s", getSLErrStr(result));
        goto error;
    }

    result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_RECORD, &amp;mRecordInterface);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("GetInterface RECORD result:%s", getSLErrStr(result));
        goto error;
    }

    result = finishCommonOpen(configItf);
    if (SL_RESULT_SUCCESS != result) {
        goto error;
    }

    setState(StreamState::Open);
    return Result::OK;

error:
    close(); // Clean up various OpenSL objects and prevent resource leaks.
    return Result::ErrorInternal; // TODO convert error from SLES to OBOE
}

static SLresult IEngine_CreateAudioRecorder(SLEngineItf self, SLObjectItf *pRecorder,
    SLDataSource *pAudioSrc, SLDataSink *pAudioSnk, SLuint32 numInterfaces,
    const SLInterfaceID *pInterfaceIds, const SLboolean *pInterfaceRequired)
{
    SL_ENTER_INTERFACE

#if (USE_PROFILES & USE_PROFILES_OPTIONAL) || defined(ANDROID)
    if (NULL == pRecorder) {
        result = SL_RESULT_PARAMETER_INVALID;
    } else {
        *pRecorder = NULL;
        unsigned exposedMask;
        const ClassTable *pCAudioRecorder_class = objectIDtoClass(SL_OBJECTID_AUDIORECORDER);
        if (NULL == pCAudioRecorder_class) {
            result = SL_RESULT_FEATURE_UNSUPPORTED;
        } else {
            result = checkInterfaces(pCAudioRecorder_class, numInterfaces,
                    pInterfaceIds, pInterfaceRequired, &exposedMask, NULL);
        }

        if (SL_RESULT_SUCCESS == result) {

            // Construct our new AudioRecorder instance
            CAudioRecorder *thiz = (CAudioRecorder *) construct(pCAudioRecorder_class, exposedMask,
                    self);
            if (NULL == thiz) {
                result = SL_RESULT_MEMORY_FAILURE;
            } else {

                do {

                    // Initialize fields not associated with any interface

                    // Default data source in case of failure in checkDataSource
                    thiz->mDataSource.mLocator.mLocatorType = SL_DATALOCATOR_NULL;
                    thiz->mDataSource.mFormat.mFormatType = SL_DATAFORMAT_NULL;

                    // Default data sink in case of failure in checkDataSink
                    thiz->mDataSink.mLocator.mLocatorType = SL_DATALOCATOR_NULL;
                    thiz->mDataSink.mFormat.mFormatType = SL_DATAFORMAT_NULL;

                    // These fields are set to real values by
                    // android_audioRecorder_checkSourceSink.  Note that the data sink is
                    // always PCM buffer queue, so we know the channel count and sample rate early.
                    thiz->mNumChannels = UNKNOWN_NUMCHANNELS;
                    thiz->mSampleRateMilliHz = UNKNOWN_SAMPLERATE;
#ifdef ANDROID
                    // placement new (explicit constructor)
                    // FIXME unnecessary once those fields are encapsulated in one class, rather
                    //   than a structure
                    // 创建采集
                    (void) new (&amp;thiz->mAudioRecord) android::sp<android::AudioRecord>();
                    (void) new (&amp;thiz->mCallbackProtector)
                            android::sp<android::CallbackProtector>();
                    thiz->mRecordSource = AUDIO_SOURCE_DEFAULT;
#endif

                    // Check the source and sink parameters, and make a local copy of all parameters
                    result = checkDataSource("pAudioSrc", pAudioSrc, &amp;thiz->mDataSource,
                            DATALOCATOR_MASK_IODEVICE, DATAFORMAT_MASK_NULL);
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }
                    result = checkDataSink("pAudioSnk", pAudioSnk, &amp;thiz->mDataSink,
                            DATALOCATOR_MASK_URI
#ifdef ANDROID
                            | DATALOCATOR_MASK_ANDROIDSIMPLEBUFFERQUEUE
#endif
                            , DATAFORMAT_MASK_MIME | DATAFORMAT_MASK_PCM | DATAFORMAT_MASK_PCM_EX
                    );
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }

                    // It would be unsafe to ever refer to the application pointers again
                    pAudioSrc = NULL;
                    pAudioSnk = NULL;

                    // check the audio source and sink parameters against platform support
#ifdef ANDROID
                    result = android_audioRecorder_checkSourceSink(thiz);
                    if (SL_RESULT_SUCCESS != result) {
                        SL_LOGE("Cannot create AudioRecorder: invalid source or sink");
                        break;
                    }
#endif

#ifdef ANDROID
                    // Allocate memory for buffer queue
                    SLuint32 locatorType = thiz->mDataSink.mLocator.mLocatorType;
                    if (locatorType == SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE) {
                        thiz->mBufferQueue.mNumBuffers =
                            thiz->mDataSink.mLocator.mBufferQueue.numBuffers;
                        // inline allocation of circular Buffer Queue mArray, up to a typical max
                        if (BUFFER_HEADER_TYPICAL >= thiz->mBufferQueue.mNumBuffers) {
                            thiz->mBufferQueue.mArray = thiz->mBufferQueue.mTypical;
                        } else {
                            // Avoid possible integer overflow during multiplication; this arbitrary
                            // maximum is big enough to not interfere with real applications, but
                            // small enough to not overflow.
                            if (thiz->mBufferQueue.mNumBuffers >= 256) {
                                result = SL_RESULT_MEMORY_FAILURE;
                                break;
                            }
                            thiz->mBufferQueue.mArray = (BufferHeader *) malloc((thiz->mBufferQueue.
                                    mNumBuffers + 1) * sizeof(BufferHeader));
                            if (NULL == thiz->mBufferQueue.mArray) {
                                result = SL_RESULT_MEMORY_FAILURE;
                                break;
                            }
                        }
                        thiz->mBufferQueue.mFront = thiz->mBufferQueue.mArray;
                        thiz->mBufferQueue.mRear = thiz->mBufferQueue.mArray;
                    }
#endif

                    // platform-specific initialization
#ifdef ANDROID
                    android_audioRecorder_create(thiz);
#endif

                } while (0);

                if (SL_RESULT_SUCCESS != result) {
                    IObject_Destroy(&amp;thiz->mObject.mItf);
                } else {
                    IObject_Publish(&amp;thiz->mObject);
                    // return the new audio recorder object
                    *pRecorder = &amp;thiz->mObject.mItf;
                }
            }

        }

    }
#else
    result = SL_RESULT_FEATURE_UNSUPPORTED;
#endif

    SL_LEAVE_INTERFACE
}

/*typedef*/ struct CAudioRecorder_struct {
    // mandated interfaces
    IObject mObject;
#ifdef ANDROID
#define INTERFACES_AudioRecorder 14 // see MPH_to_AudioRecorder in MPH_to.c for list of interfaces
#else
#define INTERFACES_AudioRecorder 9  // see MPH_to_AudioRecorder in MPH_to.c for list of interfaces
#endif
    SLuint8 mInterfaceStates2[INTERFACES_AudioRecorder - INTERFACES_Default];
    IDynamicInterfaceManagement mDynamicInterfaceManagement;
    IRecord mRecord;
    IAudioEncoder mAudioEncoder;
    // optional interfaces
    IBassBoost mBassBoost;
    IDynamicSource mDynamicSource;
    IEqualizer mEqualizer;
    IVisualization mVisualization;
    IVolume mVolume;
#ifdef ANDROID
    IBufferQueue mBufferQueue;
    IAndroidConfiguration mAndroidConfiguration;
    IAndroidAcousticEchoCancellation  mAcousticEchoCancellation;
    IAndroidAutomaticGainControl mAutomaticGainControl;
    IAndroidNoiseSuppression mNoiseSuppression;
#endif
    // remaining are per-instance private fields not associated with an interface
    DataLocatorFormat mDataSource;
    DataLocatorFormat mDataSink;
    // cached data for this instance
    SLuint8 mNumChannels;   // initially UNKNOWN_NUMCHANNELS, then const once it is known,
                            // range 1 <= x <= FCC_8
    SLuint32 mSampleRateMilliHz;// initially UNKNOWN_SAMPLERATE, then const once it is known
    // implementation-specific data for this instance
#ifdef ANDROID
    // FIXME consolidate the next several variables into ARecorder class to avoid placement new
    enum AndroidObjectType mAndroidObjType;
    android::sp<android::AudioRecord> mAudioRecord;
    android::sp<android::CallbackProtector> mCallbackProtector;
    audio_source_t mRecordSource;
    SLuint32 mPerformanceMode;
#endif
} /*CAudioRecorder*/;

Result AudioOutputStreamOpenSLES::open() {
...
  // 初始化 opensl
    Result oboeResult = AudioStreamOpenSLES::open();
    if (Result::OK != oboeResult)  return oboeResult;
    // 创建 output mixer
    SLresult result = OutputMixerOpenSL::getInstance().open();
    if (SL_RESULT_SUCCESS != result) {
        AudioStreamOpenSLES::close();
        return Result::ErrorInternal;
    }

    SLuint32 bitsPerSample = static_cast<SLuint32>(getBytesPerSample() * kBitsPerByte);

    // configure audio source
    mBufferQueueLength = calculateOptimalBufferQueueLength();
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
            SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,    // locatorType
            static_cast<SLuint32>(mBufferQueueLength)};   // numBuffers

    // Define the audio data format.
    SLDataFormat_PCM format_pcm = {
            SL_DATAFORMAT_PCM,       // formatType
            static_cast<SLuint32>(mChannelCount),           // numChannels
            static_cast<SLuint32>(mSampleRate * kMillisPerSecond),    // milliSamplesPerSec
            bitsPerSample,                      // bitsPerSample
            bitsPerSample,                      // containerSize;
            channelCountToChannelMask(mChannelCount), // channelMask
            getDefaultByteOrder(),
    };

    SLDataSource audioSrc = {&loc_bufq, &format_pcm};

    /**
     * API 21 (Lollipop) introduced support for floating-point data representation and an extended
     * data format type: SLAndroidDataFormat_PCM_EX. If running on API 21+ use this newer format
     * type, creating it from our original format.
     */
    SLAndroidDataFormat_PCM_EX format_pcm_ex;
    if (getSdkVersion() >= __ANDROID_API_L__) {
        SLuint32 representation = OpenSLES_ConvertFormatToRepresentation(getFormat());
        // Fill in the format structure.
        format_pcm_ex = OpenSLES_createExtendedFormat(format_pcm, representation);
        // Use in place of the previous format.
        audioSrc.pFormat = &format_pcm_ex;
    }
   // 创建player
    result = OutputMixerOpenSL::getInstance().createAudioPlayer(&mObjectInterface,
                                                                          &audioSrc);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("createAudioPlayer() result:%s", getSLErrStr(result));
        goto error;
    }

    // Configure the stream.
    result = (*mObjectInterface)->GetInterface(mObjectInterface,
                                               SL_IID_ANDROIDCONFIGURATION,
                                               (void *)&configItf);
    if (SL_RESULT_SUCCESS != result) {
        LOGW("%s() GetInterface(SL_IID_ANDROIDCONFIGURATION) failed with %s",
             __func__, getSLErrStr(result));
    } else {
        result = configurePerformanceMode(configItf);
        if (SL_RESULT_SUCCESS != result) {
            goto error;
        }

        SLuint32 presetValue = OpenSLES_convertOutputUsage(getUsage());
        result = (*configItf)->SetConfiguration(configItf,
                                                SL_ANDROID_KEY_STREAM_TYPE,
                                                &presetValue,
                                                sizeof(presetValue));
        if (SL_RESULT_SUCCESS != result) {
            goto error;
        }
    }

    result = (*mObjectInterface)->Realize(mObjectInterface, SL_BOOLEAN_FALSE);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("Realize player object result:%s", getSLErrStr(result));
        goto error;
    }

    result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_PLAY, &mPlayInterface);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("GetInterface PLAY result:%s", getSLErrStr(result));
        goto error;
    }

    result = finishCommonOpen(configItf);
    if (SL_RESULT_SUCCESS != result) {
        goto error;
    }

    setState(StreamState::Open);
    return Result::OK;

error:
    close();  // Clean up various OpenSL objects and prevent resource leaks.
    return Result::ErrorInternal; // TODO convert error from SLES to OBOE
}

static SLresult IEngine_CreateAudioPlayer(SLEngineItf self, SLObjectItf *pPlayer,
    SLDataSource *pAudioSrc, SLDataSink *pAudioSnk, SLuint32 numInterfaces,
    const SLInterfaceID *pInterfaceIds, const SLboolean *pInterfaceRequired)
{
    SL_ENTER_INTERFACE

    if (NULL == pPlayer) {
       result = SL_RESULT_PARAMETER_INVALID;
    } else {
        *pPlayer = NULL;
        unsigned exposedMask, requiredMask;
        const ClassTable *pCAudioPlayer_class = objectIDtoClass(SL_OBJECTID_AUDIOPLAYER);
        assert(NULL != pCAudioPlayer_class);
        result = checkInterfaces(pCAudioPlayer_class, numInterfaces,
            pInterfaceIds, pInterfaceRequired, &exposedMask, &requiredMask);
        if (SL_RESULT_SUCCESS == result) {

            // Construct our new AudioPlayer instance
            CAudioPlayer *thiz = (CAudioPlayer *) construct(pCAudioPlayer_class, exposedMask, self);
            if (NULL == thiz) {
                result = SL_RESULT_MEMORY_FAILURE;
            } else {

                do {

                    // Initialize private fields not associated with an interface

                    // Default data source in case of failure in checkDataSource
                    thiz->mDataSource.mLocator.mLocatorType = SL_DATALOCATOR_NULL;
                    thiz->mDataSource.mFormat.mFormatType = SL_DATAFORMAT_NULL;

                    // Default data sink in case of failure in checkDataSink
                    thiz->mDataSink.mLocator.mLocatorType = SL_DATALOCATOR_NULL;
                    thiz->mDataSink.mFormat.mFormatType = SL_DATAFORMAT_NULL;

                    // Default is no per-channel mute or solo
                    thiz->mMuteMask = 0;
                    thiz->mSoloMask = 0;

                    // Will be set soon for PCM buffer queues, or later by platform-specific code
                    // during Realize or Prefetch
                    thiz->mNumChannels = UNKNOWN_NUMCHANNELS;
                    thiz->mSampleRateMilliHz = UNKNOWN_SAMPLERATE;

                    // More default values, in case destructor needs to be called early
                    thiz->mDirectLevel = 0; // no attenuation
#ifdef USE_OUTPUTMIXEXT
                    thiz->mTrack = NULL;
                    thiz->mGains[0] = 1.0f;
                    thiz->mGains[1] = 1.0f;
                    thiz->mDestroyRequested = SL_BOOLEAN_FALSE;
#endif
#ifdef USE_SNDFILE
                    thiz->mSndFile.mPathname = NULL;
                    thiz->mSndFile.mSNDFILE = NULL;
                    memset(&thiz->mSndFile.mSfInfo, 0, sizeof(SF_INFO));
                    memset(&thiz->mSndFile.mMutex, 0, sizeof(pthread_mutex_t));
                    thiz->mSndFile.mEOF = SL_BOOLEAN_FALSE;
                    thiz->mSndFile.mWhich = 0;
                    memset(thiz->mSndFile.mBuffer, 0, sizeof(thiz->mSndFile.mBuffer));
#endif
#ifdef ANDROID
                    // placement new (explicit constructor)
                    // FIXME unnecessary once those fields are encapsulated in one class, rather
                    //   than a structure
                    //###(void) new (&thiz->mAudioTrack) android::sp<android::AudioTrack>();
                    (void) new (&thiz->mTrackPlayer) android::sp<android::TrackPlayerBase>();
                    (void) new (&thiz->mCallbackProtector)
                            android::sp<android::CallbackProtector>();
                    (void) new (&thiz->mAuxEffect) android::sp<android::AudioEffect>();
                    (void) new (&thiz->mAPlayer) android::sp<android::GenericPlayer>();
                    // Android-specific POD fields are initialized in android_audioPlayer_create,
                    // and assume calloc or memset 0 during allocation
#endif

                    // Check the source and sink parameters against generic constraints,
                    // and make a local copy of all parameters in case other application threads
                    // change memory concurrently.

                    result = checkDataSource("pAudioSrc", pAudioSrc, &thiz->mDataSource,
                            DATALOCATOR_MASK_URI | DATALOCATOR_MASK_ADDRESS |
                            DATALOCATOR_MASK_BUFFERQUEUE
#ifdef ANDROID
                            | DATALOCATOR_MASK_ANDROIDFD | DATALOCATOR_MASK_ANDROIDSIMPLEBUFFERQUEUE
                            | DATALOCATOR_MASK_ANDROIDBUFFERQUEUE
#endif
                            , DATAFORMAT_MASK_MIME | DATAFORMAT_MASK_PCM | DATAFORMAT_MASK_PCM_EX);

                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }

                    result = checkDataSink("pAudioSnk", pAudioSnk, &thiz->mDataSink,
                            DATALOCATOR_MASK_OUTPUTMIX                  // for playback
#ifdef ANDROID
                            | DATALOCATOR_MASK_ANDROIDSIMPLEBUFFERQUEUE // for decode to a BQ
                            | DATALOCATOR_MASK_BUFFERQUEUE              // for decode to a BQ
#endif
                            , DATAFORMAT_MASK_NULL
#ifdef ANDROID
                            | DATAFORMAT_MASK_PCM | DATAFORMAT_MASK_PCM_EX  // for decode to PCM
#endif
                            );
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }

                    // It would be unsafe to ever refer to the application pointers again
                    pAudioSrc = NULL;
                    pAudioSnk = NULL;

                    // Check that the requested interfaces are compatible with data source and sink
                    result = checkSourceSinkVsInterfacesCompatibility(&thiz->mDataSource,
                            &thiz->mDataSink, pCAudioPlayer_class, requiredMask);
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }

                    // copy the buffer queue count from source locator (for playback) / from the
                    // sink locator (for decode on ANDROID build) to the buffer queue interface
                    // we have already range-checked the value down to a smaller width
                    SLuint16 nbBuffers = 0;
                    bool usesAdvancedBufferHeaders = false;
                    bool usesSimpleBufferQueue = false;
                    // creating an AudioPlayer which decodes AAC ADTS buffers to a PCM buffer queue
                    //  will cause usesAdvancedBufferHeaders and usesSimpleBufferQueue to be true
                    switch (thiz->mDataSource.mLocator.mLocatorType) {
                    case SL_DATALOCATOR_BUFFERQUEUE:
#ifdef ANDROID
                    case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE:
#endif
                        usesSimpleBufferQueue = true;
                        nbBuffers = (SLuint16) thiz->mDataSource.mLocator.mBufferQueue.numBuffers;
                        assert(SL_DATAFORMAT_PCM == thiz->mDataSource.mFormat.mFormatType
                                || SL_ANDROID_DATAFORMAT_PCM_EX
                                    == thiz->mDataSource.mFormat.mFormatType);
                        thiz->mNumChannels = thiz->mDataSource.mFormat.mPCM.numChannels;
                        thiz->mSampleRateMilliHz = thiz->mDataSource.mFormat.mPCM.samplesPerSec;
                        break;
#ifdef ANDROID
                    case SL_DATALOCATOR_ANDROIDBUFFERQUEUE:
                        usesAdvancedBufferHeaders = true;
                        nbBuffers = (SLuint16) thiz->mDataSource.mLocator.mABQ.numBuffers;
                        thiz->mAndroidBufferQueue.mNumBuffers = nbBuffers;
                        break;
#endif
                    default:
                        nbBuffers = 0;
                        break;
                    }
#ifdef ANDROID
                    switch (thiz->mDataSink.mLocator.mLocatorType) {
                    case SL_DATALOCATOR_BUFFERQUEUE:
                    case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE:
                        usesSimpleBufferQueue = true;
                        nbBuffers = thiz->mDataSink.mLocator.mBufferQueue.numBuffers;
                        assert(SL_DATAFORMAT_PCM == thiz->mDataSink.mFormat.mFormatType
                                || SL_ANDROID_DATAFORMAT_PCM_EX
                                    == thiz->mDataSink.mFormat.mFormatType);
                        // FIXME The values specified by the app are meaningless. We get the
                        // real values from the decoder.  But the data sink checks currently require
                        // that the app specify these useless values.  Needs doc/fix.
                        // Instead use the "unknown" values, as needed by prepare completion.
                        // thiz->mNumChannels = thiz->mDataSink.mFormat.mPCM.numChannels;
                        // thiz->mSampleRateMilliHz = thiz->mDataSink.mFormat.mPCM.samplesPerSec;
                        thiz->mNumChannels = UNKNOWN_NUMCHANNELS;
                        thiz->mSampleRateMilliHz = UNKNOWN_SAMPLERATE;
                        break;
                    default:
                        // leave nbBuffers unchanged
                        break;
                    }
#endif
                    thiz->mBufferQueue.mNumBuffers = nbBuffers;

                    // check the audio source and sink parameters against platform support
#ifdef ANDROID
                    result = android_audioPlayer_checkSourceSink(thiz);
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }
#endif

#ifdef USE_SNDFILE
                    result = SndFile_checkAudioPlayerSourceSink(thiz);
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }
#endif

#ifdef USE_OUTPUTMIXEXT
                    result = IOutputMixExt_checkAudioPlayerSourceSink(thiz);
                    if (SL_RESULT_SUCCESS != result) {
                        break;
                    }
#endif

                    // Allocate memory for buffer queue
                    if (usesAdvancedBufferHeaders) {
#ifdef ANDROID
                        // locator is SL_DATALOCATOR_ANDROIDBUFFERQUEUE
                        result = initializeAndroidBufferQueueMembers(thiz);
#else
                        assert(false);
#endif
                    }

                    if (usesSimpleBufferQueue) {
                        // locator is SL_DATALOCATOR_BUFFERQUEUE
                        //         or SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE
                        result = initializeBufferQueueMembers(thiz);
                    }

                    // used to store the data source of our audio player
                    thiz->mDynamicSource.mDataSource = &thiz->mDataSource.u.mSource;

                    // platform-specific initialization
#ifdef ANDROID
                    android_audioPlayer_create(thiz);
#endif

                } while (0);

                if (SL_RESULT_SUCCESS != result) {
                    IObject_Destroy(&thiz->mObject.mItf);
                } else {
                    IObject_Publish(&thiz->mObject);
                    // return the new audio player object
                    *pPlayer = &thiz->mObject.mItf;
                }

            }
        }

    }

    SL_LEAVE_INTERFACE
}

void android_audioPlayer_create(CAudioPlayer *pAudioPlayer) {

    // pAudioPlayer->mAndroidObjType has been set in android_audioPlayer_checkSourceSink()
    // and if it was == INVALID_TYPE, then IEngine_CreateAudioPlayer would never call us
    assert(INVALID_TYPE != pAudioPlayer->mAndroidObjType);

    // These initializations are in the same order as the field declarations in classes.h

    // FIXME Consolidate initializations (many of these already in IEngine_CreateAudioPlayer)
    // mAndroidObjType: see above comment
    pAudioPlayer->mAndroidObjState = ANDROID_UNINITIALIZED;
    pAudioPlayer->mSessionId = (audio_session_t) android::AudioSystem::newAudioUniqueId(
            AUDIO_UNIQUE_ID_USE_SESSION);
    pAudioPlayer->mPIId = PLAYER_PIID_INVALID;

    // placeholder: not necessary yet as session ID lifetime doesn't extend beyond player
    // android::AudioSystem::acquireAudioSessionId(pAudioPlayer->mSessionId);

    pAudioPlayer->mStreamType = ANDROID_DEFAULT_OUTPUT_STREAM_TYPE;
    pAudioPlayer->mPerformanceMode = ANDROID_PERFORMANCE_MODE_DEFAULT;

    // mAudioTrack lifecycle is handled through mTrackPlayer
    pAudioPlayer->mTrackPlayer = new android::TrackPlayerBase();
    assert(pAudioPlayer->mTrackPlayer != 0);
    pAudioPlayer->mCallbackProtector = new android::CallbackProtector();
    // mAPLayer
    // mAuxEffect

    pAudioPlayer->mAuxSendLevel = 0;
    pAudioPlayer->mAmplFromDirectLevel = 1.0f; // matches initial mDirectLevel value
    pAudioPlayer->mDeferredStart = false;

    // This section re-initializes interface-specific fields that
    // can be set or used regardless of whether the interface is
    // exposed on the AudioPlayer or not

    switch (pAudioPlayer->mAndroidObjType) {
    case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE:
        pAudioPlayer->mPlaybackRate.mMinRate = AUDIOTRACK_MIN_PLAYBACKRATE_PERMILLE;
        pAudioPlayer->mPlaybackRate.mMaxRate = AUDIOTRACK_MAX_PLAYBACKRATE_PERMILLE;
        break;
    case AUDIOPLAYER_FROM_URIFD:
        pAudioPlayer->mPlaybackRate.mMinRate = MEDIAPLAYER_MIN_PLAYBACKRATE_PERMILLE;
        pAudioPlayer->mPlaybackRate.mMaxRate = MEDIAPLAYER_MAX_PLAYBACKRATE_PERMILLE;
        break;
    default:
        // use the default range
        break;
    }

}

SLresult CAudioPlayer_Realize(void *self, SLboolean async)
{
    CAudioPlayer *thiz = (CAudioPlayer *) self;
    SLresult result = SL_RESULT_SUCCESS;

#ifdef ANDROID
    result = android_audioPlayer_realize(thiz, async);
#endif

#ifdef USE_SNDFILE
    result = SndFile_Realize(thiz);
#endif

    // At this point the channel count and sample rate might still be unknown,
    // depending on the data source and the platform implementation.
    // If they are unknown here, then they will be determined during prefetch.

    return result;
}

SLresult android_audioPlayer_realize(CAudioPlayer *pAudioPlayer, SLboolean async) {

    SLresult result = SL_RESULT_SUCCESS;
    SL_LOGV("Realize pAudioPlayer=%p", pAudioPlayer);
    AudioPlayback_Parameters app;
    app.sessionId = pAudioPlayer->mSessionId;
    app.streamType = pAudioPlayer->mStreamType;

    switch (pAudioPlayer->mAndroidObjType) {

    //-----------------------------------
    // AudioTrack
    case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: {
        // initialize platform-specific CAudioPlayer fields

        SLDataFormat_PCM *df_pcm = (SLDataFormat_PCM *)
                pAudioPlayer->mDynamicSource.mDataSource->pFormat;

        uint32_t sampleRate = sles_to_android_sampleRate(df_pcm->samplesPerSec);

        audio_channel_mask_t channelMask;
        channelMask = sles_to_audio_output_channel_mask(df_pcm->channelMask);

        // To maintain backward compatibility with previous releases, ignore
        // channel masks that are not indexed.
        if (channelMask == AUDIO_CHANNEL_INVALID
                || audio_channel_mask_get_representation(channelMask)
                        == AUDIO_CHANNEL_REPRESENTATION_POSITION) {
            channelMask = audio_channel_out_mask_from_count(df_pcm->numChannels);
            SL_LOGI("Emulating old channel mask behavior "
                    "(ignoring positional mask %#x, using default mask %#x based on "
                    "channel count of %d)", df_pcm->channelMask, channelMask,
                    df_pcm->numChannels);
        }
        SL_LOGV("AudioPlayer: mapped SLES channel mask %#x to android channel mask %#x",
            df_pcm->channelMask,
            channelMask);

        checkAndSetPerformanceModePre(pAudioPlayer);

        audio_output_flags_t policy;
        switch (pAudioPlayer->mPerformanceMode) {
        case ANDROID_PERFORMANCE_MODE_POWER_SAVING:
            policy = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
            break;
        case ANDROID_PERFORMANCE_MODE_NONE:
            policy = AUDIO_OUTPUT_FLAG_NONE;
            break;
        case ANDROID_PERFORMANCE_MODE_LATENCY_EFFECTS:
            policy = AUDIO_OUTPUT_FLAG_FAST;
            break;
        case ANDROID_PERFORMANCE_MODE_LATENCY:
        default:
            policy = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_RAW);
            break;
        }

        int32_t notificationFrames;
        if ((policy & AUDIO_OUTPUT_FLAG_FAST) != 0) {
            // negative notificationFrames is the number of notifications (sub-buffers) per track
            // buffer for details see the explanation at frameworks/av/include/media/AudioTrack.h
            notificationFrames = -pAudioPlayer->mBufferQueue.mNumBuffers;
        } else {
            notificationFrames = 0;
        }

        android::AudioTrack* pat = new android::AudioTrack(
                pAudioPlayer->mStreamType,                           // streamType
                sampleRate,                                          // sampleRate
                sles_to_android_sampleFormat(df_pcm),                // format
                channelMask,                                         // channel mask
                0,                                                   // frameCount
                policy,                                              // flags
                audioTrack_callBack_pullFromBuffQueue,               // callback
                (void *) pAudioPlayer,                               // user
                notificationFrames,                                  // see comment above
                pAudioPlayer->mSessionId);

        // Set it here so it can be logged by the destructor if the open failed.
        pat->setCallerName(ANDROID_OPENSLES_CALLER_NAME);

        android::status_t status = pat->initCheck();
        if (status != android::NO_ERROR) {
            // AudioTracks are meant to be refcounted, so their dtor is protected.
            static_cast<void>(android::sp<android::AudioTrack>(pat));

            SL_LOGE("AudioTrack::initCheck status %u", status);
            // FIXME should return a more specific result depending on status
            result = SL_RESULT_CONTENT_UNSUPPORTED;
            return result;
        }

        pAudioPlayer->mTrackPlayer->init(pat, android::PLAYER_TYPE_SLES_AUDIOPLAYER_BUFFERQUEUE,
                usageForStreamType(pAudioPlayer->mStreamType));

        // update performance mode according to actual flags granted to AudioTrack
        checkAndSetPerformanceModePost(pAudioPlayer);

        // initialize platform-independent CAudioPlayer fields

        pAudioPlayer->mNumChannels = df_pcm->numChannels;
        pAudioPlayer->mSampleRateMilliHz = df_pcm->samplesPerSec; // Note: bad field name in SL ES

        // This use case does not have a separate "prepare" step
        pAudioPlayer->mAndroidObjState = ANDROID_READY;

        // If there is a JavaAudioRoutingProxy associated with this player, hook it up...
        JNIEnv* j_env = NULL;
        jclass clsAudioTrack = NULL;
        jmethodID midRoutingProxy_connect = NULL;
        if (pAudioPlayer->mAndroidConfiguration.mRoutingProxy != NULL &&
                (j_env = android::AndroidRuntime::getJNIEnv()) != NULL &&
                (clsAudioTrack = j_env->FindClass("android/media/AudioTrack")) != NULL &&
                (midRoutingProxy_connect =
                    j_env->GetMethodID(clsAudioTrack, "deferred_connect", "(J)V")) != NULL) {
            j_env->ExceptionClear();
            j_env->CallVoidMethod(pAudioPlayer->mAndroidConfiguration.mRoutingProxy,
                                  midRoutingProxy_connect,
                                  (jlong)pAudioPlayer->mTrackPlayer->mAudioTrack.get());
            if (j_env->ExceptionCheck()) {
                SL_LOGE("Java exception releasing player routing object.");
                result = SL_RESULT_INTERNAL_ERROR;
                pAudioPlayer->mTrackPlayer->mAudioTrack.clear();
                return result;
            }
        }
    }
        break;

    //-----------------------------------
    // MediaPlayer
    case AUDIOPLAYER_FROM_URIFD: {
        pAudioPlayer->mAPlayer = new android::LocAVPlayer(&app, false /*hasVideo*/);
        pAudioPlayer->mAPlayer->init(sfplayer_handlePrefetchEvent,
                        (void*)pAudioPlayer /*notifUSer*/);

        switch (pAudioPlayer->mDataSource.mLocator.mLocatorType) {
            case SL_DATALOCATOR_URI: {
                // The legacy implementation ran Stagefright within the application process, and
                // so allowed local pathnames specified by URI that were openable by
                // the application but were not openable by mediaserver.
                // The current implementation runs Stagefright (mostly) within mediaserver,
                // which runs as a different UID and likely a different current working directory.
                // For backwards compatibility with any applications which may have relied on the
                // previous behavior, we convert an openable file URI into an FD.
                // Note that unlike SL_DATALOCATOR_ANDROIDFD, this FD is owned by us
                // and so we close it as soon as we've passed it (via Binder dup) to mediaserver.
                const char *uri = (const char *)pAudioPlayer->mDataSource.mLocator.mURI.URI;
                if (!isDistantProtocol(uri)) {
                    // don't touch the original uri, we may need it later
                    const char *pathname = uri;
                    // skip over an optional leading file:// prefix
                    if (!strncasecmp(pathname, "file://", 7)) {
                        pathname += 7;
                    }
                    // attempt to open it as a file using the application's credentials
                    int fd = ::open(pathname, O_RDONLY);
                    if (fd >= 0) {
                        // if open is successful, then check to see if it's a regular file
                        struct stat statbuf;
                        if (!::fstat(fd, &statbuf) && S_ISREG(statbuf.st_mode)) {
                            // treat similarly to an FD data locator, but
                            // let setDataSource take responsibility for closing fd
                            pAudioPlayer->mAPlayer->setDataSource(fd, 0, statbuf.st_size, true);
                            break;
                        }
                        // we were able to open it, but it's not a file, so let mediaserver try
                        (void) ::close(fd);
                    }
                }
                // if either the URI didn't look like a file, or open failed, or not a file
                pAudioPlayer->mAPlayer->setDataSource(uri);
                } break;
            case SL_DATALOCATOR_ANDROIDFD: {
                int64_t offset = (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.offset;
                pAudioPlayer->mAPlayer->setDataSource(
                        (int)pAudioPlayer->mDataSource.mLocator.mFD.fd,
                        offset == SL_DATALOCATOR_ANDROIDFD_USE_FILE_SIZE ?
                                (int64_t)PLAYER_FD_FIND_FILE_SIZE : offset,
                        (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.length);
                }
                break;
            default:
                SL_LOGE(ERROR_PLAYERREALIZE_UNKNOWN_DATASOURCE_LOCATOR);
                break;
        }

        if (pAudioPlayer->mObject.mEngine->mAudioManager == 0) {
            SL_LOGE("AudioPlayer realize: no audio service, player will not be registered");
            pAudioPlayer->mPIId = 0;
        } else {
            pAudioPlayer->mPIId = pAudioPlayer->mObject.mEngine->mAudioManager->trackPlayer(
                    android::PLAYER_TYPE_SLES_AUDIOPLAYER_URI_FD,
                    usageForStreamType(pAudioPlayer->mStreamType), AUDIO_CONTENT_TYPE_UNKNOWN,
                    pAudioPlayer->mTrackPlayer);
        }
        }
        break;

    //-----------------------------------
    // StreamPlayer
    case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: {
        android::StreamPlayer* splr = new android::StreamPlayer(&app, false /*hasVideo*/,
                &pAudioPlayer->mAndroidBufferQueue, pAudioPlayer->mCallbackProtector);
        pAudioPlayer->mAPlayer = splr;
        splr->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer);
        }
        break;

    //-----------------------------------
    // AudioToCbRenderer
    case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: {
        android::AudioToCbRenderer* decoder = new android::AudioToCbRenderer(&app);
        pAudioPlayer->mAPlayer = decoder;
        // configures the callback for the sink buffer queue
        decoder->setDataPushListener(adecoder_writeToBufferQueue, pAudioPlayer);
        // configures the callback for the notifications coming from the SF code
        decoder->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer);

        switch (pAudioPlayer->mDataSource.mLocator.mLocatorType) {
        case SL_DATALOCATOR_URI:
            decoder->setDataSource(
                    (const char*)pAudioPlayer->mDataSource.mLocator.mURI.URI);
            break;
        case SL_DATALOCATOR_ANDROIDFD: {
            int64_t offset = (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.offset;
            decoder->setDataSource(
                    (int)pAudioPlayer->mDataSource.mLocator.mFD.fd,
                    offset == SL_DATALOCATOR_ANDROIDFD_USE_FILE_SIZE ?
                            (int64_t)PLAYER_FD_FIND_FILE_SIZE : offset,
                            (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.length);
            }
            break;
        default:
            SL_LOGE(ERROR_PLAYERREALIZE_UNKNOWN_DATASOURCE_LOCATOR);
            break;
        }

        }
        break;

    //-----------------------------------
    // AacBqToPcmCbRenderer
    case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: {
        android::AacBqToPcmCbRenderer* bqtobq = new android::AacBqToPcmCbRenderer(&app,
                &pAudioPlayer->mAndroidBufferQueue);
        // configures the callback for the sink buffer queue
        bqtobq->setDataPushListener(adecoder_writeToBufferQueue, pAudioPlayer);
        pAudioPlayer->mAPlayer = bqtobq;
        // configures the callback for the notifications coming from the SF code,
        // but also implicitly configures the AndroidBufferQueue from which ADTS data is read
        pAudioPlayer->mAPlayer->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer);
        }
        break;

    //-----------------------------------
    default:
        SL_LOGE(ERROR_PLAYERREALIZE_UNEXPECTED_OBJECT_TYPE_D, pAudioPlayer->mAndroidObjType);
        result = SL_RESULT_INTERNAL_ERROR;
        break;
    }

    if (result == SL_RESULT_SUCCESS) {
        // proceed with effect initialization
        // initialize EQ
        // FIXME use a table of effect descriptors when adding support for more effects

        // No session effects allowed even in latency with effects performance mode because HW
        // accelerated effects are only tolerated as post processing in this mode
        if ((pAudioPlayer->mAndroidObjType != AUDIOPLAYER_FROM_PCM_BUFFERQUEUE) ||
                ((pAudioPlayer->mPerformanceMode != ANDROID_PERFORMANCE_MODE_LATENCY) &&
                 (pAudioPlayer->mPerformanceMode != ANDROID_PERFORMANCE_MODE_LATENCY_EFFECTS))) {
            if (memcmp(SL_IID_EQUALIZER, &pAudioPlayer->mEqualizer.mEqDescriptor.type,
                    sizeof(effect_uuid_t)) == 0) {
                SL_LOGV("Need to initialize EQ for AudioPlayer=%p", pAudioPlayer);
                android_eq_init(pAudioPlayer->mSessionId, &pAudioPlayer->mEqualizer);
            }
            // initialize BassBoost
            if (memcmp(SL_IID_BASSBOOST, &pAudioPlayer->mBassBoost.mBassBoostDescriptor.type,
                    sizeof(effect_uuid_t)) == 0) {
                SL_LOGV("Need to initialize BassBoost for AudioPlayer=%p", pAudioPlayer);
                android_bb_init(pAudioPlayer->mSessionId, &pAudioPlayer->mBassBoost);
            }
            // initialize Virtualizer
            if (memcmp(SL_IID_VIRTUALIZER, &pAudioPlayer->mVirtualizer.mVirtualizerDescriptor.type,
                       sizeof(effect_uuid_t)) == 0) {
                SL_LOGV("Need to initialize Virtualizer for AudioPlayer=%p", pAudioPlayer);
                android_virt_init(pAudioPlayer->mSessionId, &pAudioPlayer->mVirtualizer);
            }
        }
    }

    // initialize EffectSend
    // FIXME initialize EffectSend

    return result;
}

Result AudioStreamAAudio::open() {
    Result result = Result::OK;

    if (mAAudioStream != nullptr) {
        return Result::ErrorInvalidState;
    }

    result = AudioStream::open();
    if (result != Result::OK) {
        return result;
    }

    AAudioStreamBuilder *aaudioBuilder;
    result = static_cast<Result>(mLibLoader->createStreamBuilder(&aaudioBuilder));
    if (result != Result::OK) {
        return result;
    }

    // Do not set INPUT capacity below 4096 because that prevents us from getting a FAST track
    // when using the Legacy data path.
    // If the app requests > 4096 then we allow it but we are less likely to get LowLatency.
    // See internal bug b/80308183 for more details.
    // Fixed in Q but let's still clip the capacity because high input capacity
    // does not increase latency.
    int32_t capacity = mBufferCapacityInFrames;
    constexpr int kCapacityRequiredForFastLegacyTrack = 4096; // matches value in AudioFinger
    if (OboeGlobals::areWorkaroundsEnabled()
            && mDirection == oboe::Direction::Input
            && capacity != oboe::Unspecified
            && capacity < kCapacityRequiredForFastLegacyTrack
            && mPerformanceMode == oboe::PerformanceMode::LowLatency) {
        capacity = kCapacityRequiredForFastLegacyTrack;
        LOGD("AudioStreamAAudio.open() capacity changed from %d to %d for lower latency",
             static_cast<int>(mBufferCapacityInFrames), capacity);
    }
    mLibLoader->builder_setBufferCapacityInFrames(aaudioBuilder, capacity);

    // Channel mask was added in SC_V2. Given the corresponding channel count of selected channel
    // mask may be different from selected channel count, the last set value will be respected.
    // If channel count is set after channel mask, the previously set channel mask will be cleared.
    // If channel mask is set after channel count, the channel count will be automatically
    // calculated from selected channel mask. In that case, only set channel mask when the API
    // is available and the channel mask is specified.
    if (mLibLoader->builder_setChannelMask != nullptr && mChannelMask != ChannelMask::Unspecified) {
        mLibLoader->builder_setChannelMask(aaudioBuilder,
                                           static_cast<aaudio_channel_mask_t>(mChannelMask));
    } else {
        mLibLoader->builder_setChannelCount(aaudioBuilder, mChannelCount);
    }
    mLibLoader->builder_setDeviceId(aaudioBuilder, mDeviceId);
    mLibLoader->builder_setDirection(aaudioBuilder, static_cast<aaudio_direction_t>(mDirection));
    mLibLoader->builder_setFormat(aaudioBuilder, static_cast<aaudio_format_t>(mFormat));
    mLibLoader->builder_setSampleRate(aaudioBuilder, mSampleRate);
    mLibLoader->builder_setSharingMode(aaudioBuilder,
                                       static_cast<aaudio_sharing_mode_t>(mSharingMode));
    mLibLoader->builder_setPerformanceMode(aaudioBuilder,
                                           static_cast<aaudio_performance_mode_t>(mPerformanceMode));

    // These were added in P so we have to check for the function pointer.
    if (mLibLoader->builder_setUsage != nullptr) {
        mLibLoader->builder_setUsage(aaudioBuilder,
                                     static_cast<aaudio_usage_t>(mUsage));
    }

    if (mLibLoader->builder_setContentType != nullptr) {
        mLibLoader->builder_setContentType(aaudioBuilder,
                                           static_cast<aaudio_content_type_t>(mContentType));
    }

    if (mLibLoader->builder_setInputPreset != nullptr) {
        aaudio_input_preset_t inputPreset = mInputPreset;
        if (getSdkVersion() <= __ANDROID_API_P__ && inputPreset == InputPreset::VoicePerformance) {
            LOGD("InputPreset::VoicePerformance not supported before Q. Using VoiceRecognition.");
            inputPreset = InputPreset::VoiceRecognition; // most similar preset
        }
        mLibLoader->builder_setInputPreset(aaudioBuilder,
                                           static_cast<aaudio_input_preset_t>(inputPreset));
    }

    if (mLibLoader->builder_setSessionId != nullptr) {
        mLibLoader->builder_setSessionId(aaudioBuilder,
                                         static_cast<aaudio_session_id_t>(mSessionId));
    }

    // These were added in S so we have to check for the function pointer.
    if (mLibLoader->builder_setPackageName != nullptr && !mPackageName.empty()) {
        mLibLoader->builder_setPackageName(aaudioBuilder,
                                           mPackageName.c_str());
    }

    if (mLibLoader->builder_setAttributionTag != nullptr && !mAttributionTag.empty()) {
        mLibLoader->builder_setAttributionTag(aaudioBuilder,
                                           mAttributionTag.c_str());
    }

    if (isDataCallbackSpecified()) {
        mLibLoader->builder_setDataCallback(aaudioBuilder, oboe_aaudio_data_callback_proc, this);
        mLibLoader->builder_setFramesPerDataCallback(aaudioBuilder, getFramesPerDataCallback());

        if (!isErrorCallbackSpecified()) {
            // The app did not specify a callback so we should specify
            // our own so the stream gets closed and stopped.
            mErrorCallback = &mDefaultErrorCallback;
        }
        mLibLoader->builder_setErrorCallback(aaudioBuilder, internalErrorCallback, this);
    }
    // Else if the data callback is not being used then the write method will return an error
    // and the app can stop and close the stream.

    // ============= OPEN THE STREAM ================
    {
        AAudioStream *stream = nullptr;
        result = static_cast<Result>(mLibLoader->builder_openStream(aaudioBuilder, &stream));
        mAAudioStream.store(stream);
    }
    if (result != Result::OK) {
        // Warn developer because ErrorInternal is not very informative.
        if (result == Result::ErrorInternal && mDirection == Direction::Input) {
            LOGW("AudioStreamAAudio.open() may have failed due to lack of "
                 "audio recording permission.");
        }
        goto error2;
    }

    // Query and cache the stream properties
    mDeviceId = mLibLoader->stream_getDeviceId(mAAudioStream);
    mChannelCount = mLibLoader->stream_getChannelCount(mAAudioStream);
    mSampleRate = mLibLoader->stream_getSampleRate(mAAudioStream);
    mFormat = static_cast<AudioFormat>(mLibLoader->stream_getFormat(mAAudioStream));
    mSharingMode = static_cast<SharingMode>(mLibLoader->stream_getSharingMode(mAAudioStream));
    mPerformanceMode = static_cast<PerformanceMode>(
            mLibLoader->stream_getPerformanceMode(mAAudioStream));
    mBufferCapacityInFrames = mLibLoader->stream_getBufferCapacity(mAAudioStream);
    mBufferSizeInFrames = mLibLoader->stream_getBufferSize(mAAudioStream);
    mFramesPerBurst = mLibLoader->stream_getFramesPerBurst(mAAudioStream);

    // These were added in P so we have to check for the function pointer.
    if (mLibLoader->stream_getUsage != nullptr) {
        mUsage = static_cast<Usage>(mLibLoader->stream_getUsage(mAAudioStream));
    }
    if (mLibLoader->stream_getContentType != nullptr) {
        mContentType = static_cast<ContentType>(mLibLoader->stream_getContentType(mAAudioStream));
    }
    if (mLibLoader->stream_getInputPreset != nullptr) {
        mInputPreset = static_cast<InputPreset>(mLibLoader->stream_getInputPreset(mAAudioStream));
    }
    if (mLibLoader->stream_getSessionId != nullptr) {
        mSessionId = static_cast<SessionId>(mLibLoader->stream_getSessionId(mAAudioStream));
    } else {
        mSessionId = SessionId::None;
    }

    if (mLibLoader->stream_getChannelMask != nullptr) {
        mChannelMask = static_cast<ChannelMask>(mLibLoader->stream_getChannelMask(mAAudioStream));
    }

    LOGD("AudioStreamAAudio.open() format=%d, sampleRate=%d, capacity = %d",
            static_cast<int>(mFormat), static_cast<int>(mSampleRate),
            static_cast<int>(mBufferCapacityInFrames));

    calculateDefaultDelayBeforeCloseMillis();

error2:
    mLibLoader->builder_delete(aaudioBuilder);
    LOGD("AudioStreamAAudio.open: AAudioStream_Open() returned %s",
         mLibLoader->convertResultToText(static_cast<aaudio_result_t>(result)));
    return result;
}

Result AudioInputStreamOpenSLES::requestStart() {
    LOGD("AudioInputStreamOpenSLES(): %s() called", __func__);
    std::lock_guard<std::mutex> lock(mLock);
    StreamState initialState = getState();
    switch (initialState) {
        case StreamState::Starting:
        case StreamState::Started:
            return Result::OK;
        case StreamState::Closed:
            return Result::ErrorClosed;
        default:
            break;
    }

    // We use a callback if the user requests one
    // OR if we have an internal callback to fill the blocking IO buffer.
    setDataCallbackEnabled(true);

    setState(StreamState::Starting);
    Result result = setRecordState_l(SL_RECORDSTATE_RECORDING);
    if (result == Result::OK) {
        setState(StreamState::Started);
        // Enqueue the first buffer to start the streaming.
        // This does not call the callback function.
        enqueueCallbackBuffer(mSimpleBufferQueueInterface);
    } else {
        setState(initialState);
    }
    return result;
}

Result AudioInputStreamOpenSLES::setRecordState_l(SLuint32 newState) {
    LOGD("AudioInputStreamOpenSLES::%s(%u)", __func__, newState);
    Result result = Result::OK;

    if (mRecordInterface == nullptr) {
        LOGW("AudioInputStreamOpenSLES::%s() mRecordInterface is null", __func__);
        return Result::ErrorInvalidState;
    }
    SLresult slResult = (*mRecordInterface)->SetRecordState(mRecordInterface, newState);
    //LOGD("AudioInputStreamOpenSLES::%s(%u) returned %u", __func__, newState, slResult);
    if (SL_RESULT_SUCCESS != slResult) {
        LOGE("AudioInputStreamOpenSLES::%s(%u) returned error %s",
                __func__, newState, getSLErrStr(slResult));
        result = Result::ErrorInternal; // TODO review
    }
    return result;
}

static SLresult IRecord_SetRecordState(SLRecordItf self, SLuint32 state)
{
    SL_ENTER_INTERFACE

    switch (state) {
    case SL_RECORDSTATE_STOPPED:
    case SL_RECORDSTATE_PAUSED:
    case SL_RECORDSTATE_RECORDING:
        {
        IRecord *thiz = (IRecord *) self;
        interface_lock_exclusive(thiz);
        thiz->mState = state;
#ifdef ANDROID
        android_audioRecorder_setRecordState(InterfaceToCAudioRecorder(thiz), state);
#endif
        interface_unlock_exclusive(thiz);
        result = SL_RESULT_SUCCESS;
        }
        break;
    default:
        result = SL_RESULT_PARAMETER_INVALID;
        break;
    }

    SL_LEAVE_INTERFACE
}

void android_audioRecorder_setRecordState(CAudioRecorder* ar, SLuint32 state) {
    SL_LOGV("android_audioRecorder_setRecordState(%p, %u) entering", ar, state);

    if (ar->mAudioRecord == 0) {
        return;
    }

    switch (state) {
     case SL_RECORDSTATE_STOPPED:
         ar->mAudioRecord->stop();
         break;
     case SL_RECORDSTATE_PAUSED:
         // Note that pausing is treated like stop as this implementation only records to a buffer
         //  queue, so there is no notion of destination being "opened" or "closed" (See description
         //  of SL_RECORDSTATE in specification)
         ar->mAudioRecord->stop();
         break;
     case SL_RECORDSTATE_RECORDING:
         ar->mAudioRecord->start();
         break;
     default:
         break;
     }

}

SLresult AudioStreamOpenSLES::finishCommonOpen(SLAndroidConfigurationItf configItf) {
    SLresult result = registerBufferQueueCallback();
    if (SL_RESULT_SUCCESS != result) {
        return result;
    }

    result = updateStreamParameters(configItf);
    if (SL_RESULT_SUCCESS != result) {
        return result;
    }

    Result oboeResult = configureBufferSizes(mSampleRate);
    if (Result::OK != oboeResult) {
        return (SLresult) oboeResult;
    }

    allocateFifo();

    calculateDefaultDelayBeforeCloseMillis();

    return SL_RESULT_SUCCESS;
}

SLresult AudioStreamOpenSLES::registerBufferQueueCallback() {
    // The BufferQueue
    SLresult result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
                                                &mSimpleBufferQueueInterface);
    if (SL_RESULT_SUCCESS != result) {
        LOGE("get buffer queue interface:%p result:%s",
             mSimpleBufferQueueInterface,
             getSLErrStr(result));
    } else {
        // Register the BufferQueue callback
        result = (*mSimpleBufferQueueInterface)->RegisterCallback(mSimpleBufferQueueInterface,
                                                                  bqCallbackGlue, this);
        if (SL_RESULT_SUCCESS != result) {
            LOGE("RegisterCallback result:%s", getSLErrStr(result));
        }
    }
    return result;
}

SLresult IBufferQueue_RegisterCallback(SLBufferQueueItf self,
    slBufferQueueCallback callback, void *pContext)
{
    SL_ENTER_INTERFACE

    IBufferQueue *thiz = (IBufferQueue *) self;
    interface_lock_exclusive(thiz);
    // verify pre-condition that media object is in the SL_PLAYSTATE_STOPPED state
    if (SL_PLAYSTATE_STOPPED == getAssociatedState(thiz)) {
        thiz->mCallback = callback;
        thiz->mContext = pContext;
        result = SL_RESULT_SUCCESS;
    } else {
        result = SL_RESULT_PRECONDITIONS_VIOLATED;
    }
    interface_unlock_exclusive(thiz);

    SL_LEAVE_INTERFACE
}

    ar->mAudioRecord = new android::AudioRecord(
            ar->mRecordSource,     // source
            sampleRate,            // sample rate in Hertz
            sles_to_android_sampleFormat(df_pcm),               // format
            channelMask,           // channel mask
            android::String16(),   // app ops
            0,                     // frameCount
            audioRecorder_callback,// callback_t  // 采集回调
            (void*)ar,             // user, callback data, here the AudioRecorder
            0,                     // notificationFrames
            AUDIO_SESSION_ALLOCATE,
            android::AudioRecord::TRANSFER_CALLBACK,
                                   // transfer type
            policy);   

static void audioRecorder_callback(int event, void* user, void *info) {
    //SL_LOGV("audioRecorder_callback(%d, %p, %p) entering", event, user, info);

    CAudioRecorder *ar = (CAudioRecorder *)user;

    if (!android::CallbackProtector::enterCbIfOk(ar->mCallbackProtector)) {
        // it is not safe to enter the callback (the track is about to go away)
        return;
    }

    void * callbackPContext = NULL;

    switch (event) {
    case android::AudioRecord::EVENT_MORE_DATA: {
        slBufferQueueCallback callback = NULL;
        android::AudioRecord::Buffer* pBuff = (android::AudioRecord::Buffer*)info;

        // push data to the buffer queue
        interface_lock_exclusive(&ar->mBufferQueue);

        if (ar->mBufferQueue.mState.count != 0) {
            assert(ar->mBufferQueue.mFront != ar->mBufferQueue.mRear);

            BufferHeader *oldFront = ar->mBufferQueue.mFront;
            BufferHeader *newFront = &oldFront[1];

            size_t availSink = oldFront->mSize - ar->mBufferQueue.mSizeConsumed;
            size_t availSource = pBuff->size;
            size_t bytesToCopy = availSink < availSource ? availSink : availSource;
            void *pDest = (char *)oldFront->mBuffer + ar->mBufferQueue.mSizeConsumed;
            memcpy(pDest, pBuff->raw, bytesToCopy);

            if (bytesToCopy < availSink) {
                // can't consume the whole or rest of the buffer in one shot
                ar->mBufferQueue.mSizeConsumed += availSource;
                // pBuff->size is already equal to bytesToCopy in this case
            } else {
                // finish pushing the buffer or push the buffer in one shot
                pBuff->size = bytesToCopy;
                ar->mBufferQueue.mSizeConsumed = 0;
                if (newFront == &ar->mBufferQueue.mArray[ar->mBufferQueue.mNumBuffers + 1]) {
                    newFront = ar->mBufferQueue.mArray;
                }
                ar->mBufferQueue.mFront = newFront;

                ar->mBufferQueue.mState.count--;
                ar->mBufferQueue.mState.playIndex++;

                // data has been copied to the buffer, and the buffer queue state has been updated
                // we will notify the client if applicable
                callback = ar->mBufferQueue.mCallback;
                // save callback data
                callbackPContext = ar->mBufferQueue.mContext;
            }
        } else { // empty queue
            // no destination to push the data
            pBuff->size = 0;
        }

        interface_unlock_exclusive(&ar->mBufferQueue);

        // notify client
        if (NULL != callback) {
            (*callback)(&ar->mBufferQueue.mItf, callbackPContext);
        }
        }
        break;

    case android::AudioRecord::EVENT_OVERRUN:
        audioRecorder_handleOverrun_lockRecord(ar);
        break;

    case android::AudioRecord::EVENT_MARKER:
        audioRecorder_handleMarker_lockRecord(ar);
        break;

    case android::AudioRecord::EVENT_NEW_POS:
        audioRecorder_handleNewPos_lockRecord(ar);
        break;

    case android::AudioRecord::EVENT_NEW_IAUDIORECORD:
        // ignore for now
        break;

    default:
        SL_LOGE("Encountered unknown AudioRecord event %d for CAudioRecord %p", event, ar);
        break;
    }

    ar->mCallbackProtector->exitCb();
}

// This callback handler is called every time a buffer has been processed by OpenSL ES.
static void bqCallbackGlue(SLAndroidSimpleBufferQueueItf bq, void *context) {
    bool shouldStopStream = (reinterpret_cast<AudioStreamOpenSLES *>(context))
            ->processBufferCallback(bq);
    if (shouldStopStream) {
        (reinterpret_cast<AudioStreamOpenSLES *>(context))->requestStop();
    }
}

bool AudioStreamOpenSLES::processBufferCallback(SLAndroidSimpleBufferQueueItf bq) {
    bool shouldStopStream = false;
    // Ask the app callback to process the buffer.
    DataCallbackResult result =
            fireDataCallback(mCallbackBuffer[mCallbackBufferIndex].get(), mFramesPerCallback);
    if (result == DataCallbackResult::Continue) {
        // Pass the buffer to OpenSLES.
        SLresult enqueueResult = enqueueCallbackBuffer(bq);
        if (enqueueResult != SL_RESULT_SUCCESS) {
            LOGE("%s() returned %d", __func__, enqueueResult);
            shouldStopStream = true;
        }
        // Update Oboe client position with frames handled by the callback.
        if (getDirection() == Direction::Input) {
            mFramesRead += mFramesPerCallback;
        } else {
            mFramesWritten += mFramesPerCallback;
        }
    } else if (result == DataCallbackResult::Stop) {
        LOGD("Oboe callback returned Stop");
        shouldStopStream = true;
    } else {
        LOGW("Oboe callback returned unexpected value = %d", result);
        shouldStopStream = true;
    }
    if (shouldStopStream) {
        mCallbackBufferIndex = 0;
    }
    return shouldStopStream;
}

DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) {
    if (!isDataCallbackEnabled()) {
        LOGW("AudioStream::%s() called with data callback disabled!", __func__);
        return DataCallbackResult::Stop; // Should not be getting called
    }

    DataCallbackResult result;
    if (mDataCallback) {
        result = mDataCallback->onAudioReady(this, audioData, numFrames);
    } else {
        result = onDefaultCallback(audioData, numFrames);
    }
    // On Oreo, we might get called after returning stop.
    // So block that here.
    setDataCallbackEnabled(result == DataCallbackResult::Continue);

    return result;
}

        android::AudioTrack* pat = new android::AudioTrack(
                pAudioPlayer->mStreamType,                           // streamType
                sampleRate,                                          // sampleRate
                sles_to_android_sampleFormat(df_pcm),                // format
                channelMask,                                         // channel mask
                0,                                                   // frameCount
                policy,                                              // flags
                audioTrack_callBack_pullFromBuffQueue,               // callback
                (void *) pAudioPlayer,                               // user
                notificationFrames,                                  // see comment above
                pAudioPlayer->mSessionId);

//-----------------------------------------------------------------------------
// Callback associated with an AudioTrack of an SL ES AudioPlayer that gets its data
// from a buffer queue. This will not be called once the AudioTrack has been destroyed.
static void audioTrack_callBack_pullFromBuffQueue(int event, void* user, void *info) {
    CAudioPlayer *ap = (CAudioPlayer *)user;

    if (!android::CallbackProtector::enterCbIfOk(ap->mCallbackProtector)) {
        // it is not safe to enter the callback (the track is about to go away)
        return;
    }

    void * callbackPContext = NULL;
    switch (event) {

    case android::AudioTrack::EVENT_MORE_DATA: {
        //SL_LOGV("received event EVENT_MORE_DATA from AudioTrack TID=%d", gettid());
        slPrefetchCallback prefetchCallback = NULL;
        void *prefetchContext = NULL;
        SLuint32 prefetchEvents = SL_PREFETCHEVENT_NONE;
        android::AudioTrack::Buffer* pBuff = (android::AudioTrack::Buffer*)info;

        // retrieve data from the buffer queue
        interface_lock_exclusive(&ap->mBufferQueue);

        if (ap->mBufferQueue.mCallbackPending) {
            // call callback with lock not held
            slBufferQueueCallback callback = ap->mBufferQueue.mCallback;
            if (NULL != callback) {
                callbackPContext = ap->mBufferQueue.mContext;
                interface_unlock_exclusive(&ap->mBufferQueue);
                (*callback)(&ap->mBufferQueue.mItf, callbackPContext);
                interface_lock_exclusive(&ap->mBufferQueue);
                ap->mBufferQueue.mCallbackPending = false;
            }
        }

        if (ap->mBufferQueue.mState.count != 0) {
            //SL_LOGV("nbBuffers in queue = %u",ap->mBufferQueue.mState.count);
            assert(ap->mBufferQueue.mFront != ap->mBufferQueue.mRear);

            BufferHeader *oldFront = ap->mBufferQueue.mFront;
            BufferHeader *newFront = &oldFront[1];

            size_t availSource = oldFront->mSize - ap->mBufferQueue.mSizeConsumed;
            size_t availSink = pBuff->size;
            size_t bytesToCopy = availSource < availSink ? availSource : availSink;
            void *pSrc = (char *)oldFront->mBuffer + ap->mBufferQueue.mSizeConsumed;
            memcpy(pBuff->raw, pSrc, bytesToCopy);

            if (bytesToCopy < availSource) {
                ap->mBufferQueue.mSizeConsumed += bytesToCopy;
                // pBuff->size is already equal to bytesToCopy in this case
            } else {
                // consumed an entire buffer, dequeue
                pBuff->size = bytesToCopy;
                ap->mBufferQueue.mSizeConsumed = 0;
                if (newFront ==
                        &ap->mBufferQueue.mArray
                            [ap->mBufferQueue.mNumBuffers + 1])
                {
                    newFront = ap->mBufferQueue.mArray;
                }
                ap->mBufferQueue.mFront = newFront;

                ap->mBufferQueue.mState.count--;
                ap->mBufferQueue.mState.playIndex++;
                ap->mBufferQueue.mCallbackPending = true;
            }
        } else { // empty queue
            // signal no data available
            pBuff->size = 0;

            // signal we're at the end of the content, but don't pause (see note in function)
            audioPlayer_dispatch_headAtEnd_lockPlay(ap, false /*set state to paused?*/, false);

            // signal underflow to prefetch status itf
            if (IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) {
                ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_UNDERFLOW;
                ap->mPrefetchStatus.mLevel = 0;
                // callback or no callback?
                prefetchEvents = ap->mPrefetchStatus.mCallbackEventsMask &
                        (SL_PREFETCHEVENT_STATUSCHANGE | SL_PREFETCHEVENT_FILLLEVELCHANGE);
                if (SL_PREFETCHEVENT_NONE != prefetchEvents) {
                    prefetchCallback = ap->mPrefetchStatus.mCallback;
                    prefetchContext  = ap->mPrefetchStatus.mContext;
                }
            }

            // stop the track so it restarts playing faster when new data is enqueued
            ap->mTrackPlayer->stop();
        }
        interface_unlock_exclusive(&ap->mBufferQueue);

        // notify client
        if (NULL != prefetchCallback) {
            assert(SL_PREFETCHEVENT_NONE != prefetchEvents);
            // spec requires separate callbacks for each event
            if (prefetchEvents & SL_PREFETCHEVENT_STATUSCHANGE) {
                (*prefetchCallback)(&ap->mPrefetchStatus.mItf, prefetchContext,
                        SL_PREFETCHEVENT_STATUSCHANGE);
            }
            if (prefetchEvents & SL_PREFETCHEVENT_FILLLEVELCHANGE) {
                (*prefetchCallback)(&ap->mPrefetchStatus.mItf, prefetchContext,
                        SL_PREFETCHEVENT_FILLLEVELCHANGE);
            }
        }
    }
    break;

    case android::AudioTrack::EVENT_MARKER:
        //SL_LOGI("received event EVENT_MARKER from AudioTrack");
        audioTrack_handleMarker_lockPlay(ap);
        break;

    case android::AudioTrack::EVENT_NEW_POS:
        //SL_LOGI("received event EVENT_NEW_POS from AudioTrack");
        audioTrack_handleNewPos_lockPlay(ap);
        break;

    case android::AudioTrack::EVENT_UNDERRUN:
        //SL_LOGI("received event EVENT_UNDERRUN from AudioTrack");
        audioTrack_handleUnderrun_lockPlay(ap);
        break;

    case android::AudioTrack::EVENT_NEW_IAUDIOTRACK:
        // ignore for now
        break;

    case android::AudioTrack::EVENT_BUFFER_END:
    case android::AudioTrack::EVENT_LOOP_END:
    case android::AudioTrack::EVENT_STREAM_END:
        // These are unexpected so fall through
        FALLTHROUGH_INTENDED;
    default:
        // FIXME where does the notification of SL_PLAYEVENT_HEADMOVING fit?
        SL_LOGE("Encountered unknown AudioTrack event %d for CAudioPlayer %p", event,
                (CAudioPlayer *)user);
        break;
    }

    ap->mCallbackProtector->exitCb();
}

Result AudioStreamAAudio::requestStart() {
    std::lock_guard<std::mutex> lock(mLock);
    AAudioStream *stream = mAAudioStream.load();
    if (stream != nullptr) {
        // Avoid state machine errors in O_MR1.
        if (getSdkVersion() <= __ANDROID_API_O_MR1__) {
            StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream));
            if (state == StreamState::Starting || state == StreamState::Started) {
                // WARNING: On P, AAudio is returning ErrorInvalidState for Output and OK for Input.
                return Result::OK;
            }
        }
        if (isDataCallbackSpecified()) {
            setDataCallbackEnabled(true);
        }
        mStopThreadAllowed = true;
        return static_cast<Result>(mLibLoader->stream_requestStart(stream));
    } else {
        return Result::ErrorClosed;
    }
}

if (isDataCallbackSpecified()) {
        mLibLoader->builder_setDataCallback(aaudioBuilder, oboe_aaudio_data_callback_proc, this);
        mLibLoader->builder_setFramesPerDataCallback(aaudioBuilder, getFramesPerDataCallback());

        if (!isErrorCallbackSpecified()) {
            // The app did not specify a callback so we should specify
            // our own so the stream gets closed and stopped.
            mErrorCallback = &mDefaultErrorCallback;
        }
        mLibLoader->builder_setErrorCallback(aaudioBuilder, internalErrorCallback, this);
    }

static aaudio_data_callback_result_t oboe_aaudio_data_callback_proc(
        AAudioStream *stream,
        void *userData,
        void *audioData,
        int32_t numFrames) {

    AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(userData);
    if (oboeStream != nullptr) {
        return static_cast<aaudio_data_callback_result_t>(
                oboeStream->callOnAudioReady(stream, audioData, numFrames));

    } else {
        return static_cast<aaudio_data_callback_result_t>(DataCallbackResult::Stop);
    }
}

DataCallbackResult AudioStreamAAudio::callOnAudioReady(AAudioStream * /*stream*/,
                                                                 void *audioData,
                                                                 int32_t numFrames) {
    DataCallbackResult result = fireDataCallback(audioData, numFrames);
    if (result == DataCallbackResult::Continue) {
        return result;
    } else {
        if (result == DataCallbackResult::Stop) {
            LOGD("Oboe callback returned DataCallbackResult::Stop");
        } else {
            LOGE("Oboe callback returned unexpected value = %d", result);
        }

        // Returning Stop caused various problems before S. See #1230
        if (OboeGlobals::areWorkaroundsEnabled() &amp;&amp; getSdkVersion() <= __ANDROID_API_R__) {
            launchStopThread();
            return DataCallbackResult::Continue;
        } else {
            return DataCallbackResult::Stop; // OK >= API_S
        }
    }
}

DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) {
    if (!isDataCallbackEnabled()) {
        LOGW("AudioStream::%s() called with data callback disabled!", __func__);
        return DataCallbackResult::Stop; // Should not be getting called
    }

    DataCallbackResult result;
    if (mDataCallback) {
        result = mDataCallback->onAudioReady(this, audioData, numFrames);
    } else {
        result = onDefaultCallback(audioData, numFrames);
    }
    // On Oreo, we might get called after returning stop.
    // So block that here.
    setDataCallbackEnabled(result == DataCallbackResult::Continue);

    return result;
}