media/libaaudio/src/client/AudioStreamInternalPlay.cpp - LeafOS-Project/android_frameworks_av - Gitiles

 /*
  * Copyright (C) 2017 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
  * You may obtain a copy of the License at
  *
  *      http://www.apache.org/licenses/LICENSE-2.0
  *
  * Unless required by applicable law or agreed to in writing, software
  * distributed under the License is distributed on an "AS IS" BASIS,
  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
  * See the License for the specific language governing permissions and
  * limitations under the License.
  */

 //#define LOG_NDEBUG 0
 #include <utils/Log.h>

 #define ATRACE_TAG ATRACE_TAG_AUDIO

 #include <media/MediaMetricsItem.h>
 #include <utils/Trace.h>

 #include "client/AudioStreamInternalPlay.h"
 #include "utility/AudioClock.h"

 // We do this after the #includes because if a header uses ALOG.
 // it would fail on the reference to mInService.
 #undef LOG_TAG
 // This file is used in both client and server processes.
 // This is needed to make sense of the logs more easily.
 #define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
                             : "AudioStreamInternalPlay_Client")

 using android::status_t;
 using android::WrappingBuffer;

 using namespace aaudio;

 AudioStreamInternalPlay::AudioStreamInternalPlay(AAudioServiceInterface  &serviceInterface,
                                                        bool inService)
         : AudioStreamInternal(serviceInterface, inService) {

 }

 constexpr int kRampMSec = 10; // time to apply a change in volume

 aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
     aaudio_result_t result = AudioStreamInternal::open(builder);
     const bool useVolumeRamps = (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE);
     if (result == AAUDIO_OK) {
         result = mFlowGraph.configure(getFormat(),
                              getSamplesPerFrame(),
                              getSampleRate(),
                              getDeviceFormat(),
                              getDeviceSamplesPerFrame(),
                              getDeviceSampleRate(),
                              getRequireMonoBlend(),
                              useVolumeRamps,
                              getAudioBalance(),
                              aaudio::resampler::MultiChannelResampler::Quality::Medium);

         if (result != AAUDIO_OK) {
             safeReleaseClose();
         }
         // Sample rate is constrained to common values by now and should not overflow.
         int32_t numFrames = kRampMSec * getSampleRate() / AAUDIO_MILLIS_PER_SECOND;
         mFlowGraph.setRampLengthInFrames(numFrames);
     }
     return result;
 }

 // This must be called under mStreamLock.
 aaudio_result_t AudioStreamInternalPlay::requestPause_l()
 {
     aaudio_result_t result = stopCallback_l();
     if (result != AAUDIO_OK) {
         return result;
     }
     if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
         ALOGW("%s() mServiceStreamHandle invalid", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }

     mClockModel.stop(AudioClock::getNanoseconds());
     setState(AAUDIO_STREAM_STATE_PAUSING);
     mAtomicInternalTimestamp.clear();
     return mServiceInterface.pauseStream(mServiceStreamHandleInfo);
 }

 aaudio_result_t AudioStreamInternalPlay::requestFlush_l() {
     if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
         ALOGW("%s() mServiceStreamHandle invalid", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }

     setState(AAUDIO_STREAM_STATE_FLUSHING);
     return mServiceInterface.flushStream(mServiceStreamHandleInfo);
 }

 void AudioStreamInternalPlay::prepareBuffersForStart() {
     // Reset volume ramps to avoid a starting noise.
     // This was called here instead of AudioStreamInternal so that
     // it will be easier to backport.
     mFlowGraph.reset();
     // Prevent stale data from being played.
     mAudioEndpoint->eraseDataMemory();
 }

 void AudioStreamInternalPlay::advanceClientToMatchServerPosition(int32_t serverMargin) {
     int64_t readCounter = mAudioEndpoint->getDataReadCounter() + serverMargin;
     int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();

     // Bump offset so caller does not see the retrograde motion in getFramesRead().
     int64_t offset = writeCounter - readCounter;
     mFramesOffsetFromService += offset;
     ALOGV("%s() readN = %lld, writeN = %lld, offset = %lld", __func__,
           (long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);

     // Force writeCounter to match readCounter.
     // This is because we cannot change the read counter in the hardware.
     mAudioEndpoint->setDataWriteCounter(readCounter);
 }

 void AudioStreamInternalPlay::onFlushFromServer() {
     advanceClientToMatchServerPosition(0 /*serverMargin*/);
 }

 // Write the data, block if needed and timeoutMillis > 0
 aaudio_result_t AudioStreamInternalPlay::write(const void *buffer, int32_t numFrames,
                                                int64_t timeoutNanoseconds) {
     return processData((void *)buffer, numFrames, timeoutNanoseconds);
 }

 // Write as much data as we can without blocking.
 aaudio_result_t AudioStreamInternalPlay::processDataNow(void *buffer, int32_t numFrames,
                                               int64_t currentNanoTime, int64_t *wakeTimePtr) {
     aaudio_result_t result = processCommands();
     if (result != AAUDIO_OK) {
         return result;
     }

     const char *traceName = "aaWrNow";
     ATRACE_BEGIN(traceName);

     if (mClockModel.isStarting()) {
         // Still haven't got any timestamps from server.
         // Keep waiting until we get some valid timestamps then start writing to the
         // current buffer position.
         ALOGV("%s() wait for valid timestamps", __func__);
         // Sleep very briefly and hope we get a timestamp soon.
         *wakeTimePtr = currentNanoTime + (2000 * AAUDIO_NANOS_PER_MICROSECOND);
         ATRACE_END();
         return 0;
     }
     // If we have gotten this far then we have at least one timestamp from server.

     // If a DMA channel or DSP is reading the other end then we have to update the readCounter.
     if (mAudioEndpoint->isFreeRunning()) {
         // Update data queue based on the timing model.
         int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
         // ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
         mAudioEndpoint->setDataReadCounter(estimatedReadCounter);
     }

     if (mNeedCatchUp.isRequested()) {
         // Catch an MMAP pointer that is already advancing.
         // This will avoid initial underruns caused by a slow cold start.
         // We add a one burst margin in case the DSP advances before we can write the data.
         // This can help prevent the beginning of the stream from being skipped.
         advanceClientToMatchServerPosition(getFramesPerBurst());
         mNeedCatchUp.acknowledge();
     }

     // If the read index passed the write index then consider it an underrun.
     // For shared streams, the xRunCount is passed up from the service.
     if (mAudioEndpoint->isFreeRunning() && mAudioEndpoint->getFullFramesAvailable() < 0) {
         mXRunCount++;
         if (ATRACE_ENABLED()) {
             ATRACE_INT("aaUnderRuns", mXRunCount);
         }
     }

     // Write some data to the buffer.
     //ALOGD("AudioStreamInternal::processDataNow() - writeNowWithConversion(%d)", numFrames);
     int32_t framesWritten = writeNowWithConversion(buffer, numFrames);
     //ALOGD("AudioStreamInternal::processDataNow() - tried to write %d frames, wrote %d",
     //    numFrames, framesWritten);
     if (ATRACE_ENABLED()) {
         ATRACE_INT("aaWrote", framesWritten);
     }

     // Sleep if there is too much data in the buffer.
     // Calculate an ideal time to wake up.
     if (wakeTimePtr != nullptr
             && (mAudioEndpoint->getFullFramesAvailable() >= getDeviceBufferSize())) {
         // By default wake up a few milliseconds from now.  // TODO review
         int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
         aaudio_stream_state_t state = getState();
         //ALOGD("AudioStreamInternal::processDataNow() - wakeTime based on %s",
         //      AAudio_convertStreamStateToText(state));
         switch (state) {
             case AAUDIO_STREAM_STATE_OPEN:
             case AAUDIO_STREAM_STATE_STARTING:
                 if (framesWritten != 0) {
                     // Don't wait to write more data. Just prime the buffer.
                     wakeTime = currentNanoTime;
                 }
                 break;
             case AAUDIO_STREAM_STATE_STARTED:
             {
                 // Calculate when there will be room available to write to the buffer.
                 // If the appBufferSize is smaller than the endpointBufferSize then
                 // we will have room to write data beyond the appBufferSize.
                 // That is a technique used to reduce glitches without adding latency.
                 const int64_t appBufferSize = getDeviceBufferSize();
                 // The endpoint buffer size is set to the maximum that can be written.
                 // If we use it then we must carve out some room to write data when we wake up.
                 const int64_t endBufferSize = mAudioEndpoint->getBufferSizeInFrames()
                         - getDeviceFramesPerBurst();
                 const int64_t bestBufferSize = std::min(appBufferSize, endBufferSize);
                 int64_t targetReadPosition = mAudioEndpoint->getDataWriteCounter() - bestBufferSize;
                 wakeTime = mClockModel.convertPositionToTime(targetReadPosition);
             }
                 break;
             default:
                 break;
         }
         *wakeTimePtr = wakeTime;

     }

     ATRACE_END();
     return framesWritten;
 }


 aaudio_result_t AudioStreamInternalPlay::writeNowWithConversion(const void *buffer,
                                                             int32_t numFrames) {
     WrappingBuffer wrappingBuffer;
     uint8_t *byteBuffer = (uint8_t *) buffer;
     int32_t framesLeftInByteBuffer = numFrames;

     mAudioEndpoint->getEmptyFramesAvailable(&wrappingBuffer);

     // Write data in one or two parts.
     int partIndex = 0;
     int framesWrittenToAudioEndpoint = 0;
     while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
         int32_t framesAvailableInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
         uint8_t *currentWrappingBuffer = (uint8_t *) wrappingBuffer.data[partIndex];

         if (framesAvailableInWrappingBuffer > 0) {
             // Pull data from the flowgraph in case there is residual data.
             const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.pull(
                 (void*) currentWrappingBuffer,
                 framesAvailableInWrappingBuffer);

             const int32_t numBytesActuallyWrittenToWrappingBuffer =
                 framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
             currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
             framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
             framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;
         } else {
             break;
         }

         // Put data from byteBuffer into the flowgraph one buffer (8 frames) at a time.
         // Continuously pull as much data as possible from the flowgraph into the wrapping buffer.
         // The return value of mFlowGraph.process is the number of frames actually pulled.
         while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
             int32_t framesToWriteFromByteBuffer = std::min(flowgraph::kDefaultBufferSize,
                     framesLeftInByteBuffer);
             // If the wrapping buffer is running low, write one frame at a time.
             if (framesAvailableInWrappingBuffer < flowgraph::kDefaultBufferSize) {
                 framesToWriteFromByteBuffer = 1;
             }

             const int32_t numBytesToWriteFromByteBuffer = getBytesPerFrame() *
                     framesToWriteFromByteBuffer;

             //ALOGD("%s() framesLeftInByteBuffer %d, framesAvailableInWrappingBuffer %d"
             //      "framesToWriteFromByteBuffer %d, numBytesToWriteFromByteBuffer %d"
             //      , __func__, framesLeftInByteBuffer, framesAvailableInWrappingBuffer,
             //      framesToWriteFromByteBuffer, numBytesToWriteFromByteBuffer);

             const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.process(
                     (void *)byteBuffer,
                     framesToWriteFromByteBuffer,
                     (void *)currentWrappingBuffer,
                     framesAvailableInWrappingBuffer);

             byteBuffer += numBytesToWriteFromByteBuffer;
             framesLeftInByteBuffer -= framesToWriteFromByteBuffer;
             const int32_t numBytesActuallyWrittenToWrappingBuffer =
                     framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
             currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
             framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
             framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;

             //ALOGD("%s() numBytesActuallyWrittenToWrappingBuffer %d, framesLeftInByteBuffer %d"
             //      "framesActuallyWrittenToWrappingBuffer %d, numBytesToWriteFromByteBuffer %d"
             //      "framesWrittenToAudioEndpoint %d"
             //      , __func__, numBytesActuallyWrittenToWrappingBuffer, framesLeftInByteBuffer,
             //      framesActuallyWrittenToWrappingBuffer, numBytesToWriteFromByteBuffer,
             //      framesWrittenToAudioEndpoint);
         }
         partIndex++;
     }
     //ALOGD("%s() framesWrittenToAudioEndpoint %d, numFrames %d"
     //              "framesLeftInByteBuffer %d"
     //              , __func__, framesWrittenToAudioEndpoint, numFrames,
     //              framesLeftInByteBuffer);

     // The audio endpoint should reference the number of frames written to the wrapping buffer.
     mAudioEndpoint->advanceWriteIndex(framesWrittenToAudioEndpoint);

     // The internal code should use the number of frames read from the app.
     return numFrames - framesLeftInByteBuffer;
 }

 int64_t AudioStreamInternalPlay::getFramesRead() {
     if (mAudioEndpoint) {
         const int64_t framesReadHardware = isClockModelInControl()
                 ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
                 : mAudioEndpoint->getDataReadCounter();
         // Add service offset and prevent retrograde motion.
         mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
     }
     return mLastFramesRead;
 }

 int64_t AudioStreamInternalPlay::getFramesWritten() {
     if (mAudioEndpoint) {
         mLastFramesWritten = std::max(
                 mLastFramesWritten,
                 mAudioEndpoint->getDataWriteCounter() + mFramesOffsetFromService);
     }
     return mLastFramesWritten;
 }

 // Render audio in the application callback and then write the data to the stream.
 void *AudioStreamInternalPlay::callbackLoop() {
     ALOGD("%s() entering >>>>>>>>>>>>>>>", __func__);
     aaudio_result_t result = AAUDIO_OK;
     aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
     if (!isDataCallbackSet()) return nullptr;
     int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);

     // result might be a frame count
     while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
         // Call application using the AAudio callback interface.
         callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);

         if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
             // Write audio data to stream. This is a BLOCKING WRITE!
             result = write(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
             if ((result != mCallbackFrames)) {
                 if (result >= 0) {
                     // Only wrote some of the frames requested. The stream can be disconnected
                     // or timed out.
                     processCommands();
                     result = isDisconnected() ? AAUDIO_ERROR_DISCONNECTED : AAUDIO_ERROR_TIMEOUT;
                 }
                 maybeCallErrorCallback(result);
                 break;
             }
         } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
             ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
             result = systemStopInternal();
             break;
         }
     }

     ALOGD("%s() exiting, result = %d, isActive() = %d <<<<<<<<<<<<<<",
           __func__, result, (int) isActive());
     return nullptr;
 }

 //------------------------------------------------------------------------------
 // Implementation of PlayerBase
 status_t AudioStreamInternalPlay::doSetVolume() {
     float combinedVolume = mStreamVolume * getDuckAndMuteVolume();
     ALOGD("%s() mStreamVolume * duckAndMuteVolume = %f * %f = %f",
           __func__, mStreamVolume, getDuckAndMuteVolume(), combinedVolume);
     mFlowGraph.setTargetVolume(combinedVolume);
     return android::NO_ERROR;
 }
	/*
	* Copyright (C) 2017 The Android Open Source Project
	*
	* Licensed under the Apache License, Version 2.0 (the "License");
	* you may not use this file except in compliance with the License.
	* You may obtain a copy of the License at
	*
	* http://www.apache.org/licenses/LICENSE-2.0
	*
	* Unless required by applicable law or agreed to in writing, software
	* distributed under the License is distributed on an "AS IS" BASIS,
	* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
	* See the License for the specific language governing permissions and
	* limitations under the License.
	*/

	//#define LOG_NDEBUG 0
	#include <utils/Log.h>

	#define ATRACE_TAG ATRACE_TAG_AUDIO

	#include <media/MediaMetricsItem.h>
	#include <utils/Trace.h>

	#include "client/AudioStreamInternalPlay.h"
	#include "utility/AudioClock.h"

	// We do this after the #includes because if a header uses ALOG.
	// it would fail on the reference to mInService.
	#undef LOG_TAG
	// This file is used in both client and server processes.
	// This is needed to make sense of the logs more easily.
	#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
	: "AudioStreamInternalPlay_Client")

	using android::status_t;
	using android::WrappingBuffer;

	using namespace aaudio;

	AudioStreamInternalPlay::AudioStreamInternalPlay(AAudioServiceInterface &serviceInterface,
	bool inService)
	: AudioStreamInternal(serviceInterface, inService) {

	}

	constexpr int kRampMSec = 10; // time to apply a change in volume

	aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
	aaudio_result_t result = AudioStreamInternal::open(builder);
	const bool useVolumeRamps = (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE);
	if (result == AAUDIO_OK) {
	result = mFlowGraph.configure(getFormat(),
	getSamplesPerFrame(),
	getSampleRate(),
	getDeviceFormat(),
	getDeviceSamplesPerFrame(),
	getDeviceSampleRate(),
	getRequireMonoBlend(),
	useVolumeRamps,
	getAudioBalance(),
	aaudio::resampler::MultiChannelResampler::Quality::Medium);

	if (result != AAUDIO_OK) {
	safeReleaseClose();
	}
	// Sample rate is constrained to common values by now and should not overflow.
	int32_t numFrames = kRampMSec * getSampleRate() / AAUDIO_MILLIS_PER_SECOND;
	mFlowGraph.setRampLengthInFrames(numFrames);
	}
	return result;
	}

	// This must be called under mStreamLock.
	aaudio_result_t AudioStreamInternalPlay::requestPause_l()
	{
	aaudio_result_t result = stopCallback_l();
	if (result != AAUDIO_OK) {
	return result;
	}
	if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
	ALOGW("%s() mServiceStreamHandle invalid", __func__);
	return AAUDIO_ERROR_INVALID_STATE;
	}

	mClockModel.stop(AudioClock::getNanoseconds());
	setState(AAUDIO_STREAM_STATE_PAUSING);
	mAtomicInternalTimestamp.clear();
	return mServiceInterface.pauseStream(mServiceStreamHandleInfo);
	}

	aaudio_result_t AudioStreamInternalPlay::requestFlush_l() {
	if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
	ALOGW("%s() mServiceStreamHandle invalid", __func__);
	return AAUDIO_ERROR_INVALID_STATE;
	}

	setState(AAUDIO_STREAM_STATE_FLUSHING);
	return mServiceInterface.flushStream(mServiceStreamHandleInfo);
	}

	void AudioStreamInternalPlay::prepareBuffersForStart() {
	// Reset volume ramps to avoid a starting noise.
	// This was called here instead of AudioStreamInternal so that
	// it will be easier to backport.
	mFlowGraph.reset();
	// Prevent stale data from being played.
	mAudioEndpoint->eraseDataMemory();
	}

	void AudioStreamInternalPlay::advanceClientToMatchServerPosition(int32_t serverMargin) {
	int64_t readCounter = mAudioEndpoint->getDataReadCounter() + serverMargin;
	int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();

	// Bump offset so caller does not see the retrograde motion in getFramesRead().
	int64_t offset = writeCounter - readCounter;
	mFramesOffsetFromService += offset;
	ALOGV("%s() readN = %lld, writeN = %lld, offset = %lld", __func__,
	(long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);

	// Force writeCounter to match readCounter.
	// This is because we cannot change the read counter in the hardware.
	mAudioEndpoint->setDataWriteCounter(readCounter);
	}

	void AudioStreamInternalPlay::onFlushFromServer() {
	advanceClientToMatchServerPosition(0 /serverMargin/);
	}

	// Write the data, block if needed and timeoutMillis > 0
	aaudio_result_t AudioStreamInternalPlay::write(const void *buffer, int32_t numFrames,
	int64_t timeoutNanoseconds) {
	return processData((void *)buffer, numFrames, timeoutNanoseconds);
	}

	// Write as much data as we can without blocking.
	aaudio_result_t AudioStreamInternalPlay::processDataNow(void *buffer, int32_t numFrames,
	int64_t currentNanoTime, int64_t *wakeTimePtr) {
	aaudio_result_t result = processCommands();
	if (result != AAUDIO_OK) {
	return result;
	}

	const char *traceName = "aaWrNow";
	ATRACE_BEGIN(traceName);

	if (mClockModel.isStarting()) {
	// Still haven't got any timestamps from server.
	// Keep waiting until we get some valid timestamps then start writing to the
	// current buffer position.
	ALOGV("%s() wait for valid timestamps", __func__);
	// Sleep very briefly and hope we get a timestamp soon.
	wakeTimePtr = currentNanoTime + (2000 AAUDIO_NANOS_PER_MICROSECOND);
	ATRACE_END();
	return 0;
	}
	// If we have gotten this far then we have at least one timestamp from server.

	// If a DMA channel or DSP is reading the other end then we have to update the readCounter.
	if (mAudioEndpoint->isFreeRunning()) {
	// Update data queue based on the timing model.
	int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
	// ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
	mAudioEndpoint->setDataReadCounter(estimatedReadCounter);
	}

	if (mNeedCatchUp.isRequested()) {
	// Catch an MMAP pointer that is already advancing.
	// This will avoid initial underruns caused by a slow cold start.
	// We add a one burst margin in case the DSP advances before we can write the data.
	// This can help prevent the beginning of the stream from being skipped.
	advanceClientToMatchServerPosition(getFramesPerBurst());
	mNeedCatchUp.acknowledge();
	}

	// If the read index passed the write index then consider it an underrun.
	// For shared streams, the xRunCount is passed up from the service.
	if (mAudioEndpoint->isFreeRunning() && mAudioEndpoint->getFullFramesAvailable() < 0) {
	mXRunCount++;
	if (ATRACE_ENABLED()) {
	ATRACE_INT("aaUnderRuns", mXRunCount);
	}
	}

	// Write some data to the buffer.
	//ALOGD("AudioStreamInternal::processDataNow() - writeNowWithConversion(%d)", numFrames);
	int32_t framesWritten = writeNowWithConversion(buffer, numFrames);
	//ALOGD("AudioStreamInternal::processDataNow() - tried to write %d frames, wrote %d",
	// numFrames, framesWritten);
	if (ATRACE_ENABLED()) {
	ATRACE_INT("aaWrote", framesWritten);
	}

	// Sleep if there is too much data in the buffer.
	// Calculate an ideal time to wake up.
	if (wakeTimePtr != nullptr
	&& (mAudioEndpoint->getFullFramesAvailable() >= getDeviceBufferSize())) {
	// By default wake up a few milliseconds from now. // TODO review
	int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
	aaudio_stream_state_t state = getState();
	//ALOGD("AudioStreamInternal::processDataNow() - wakeTime based on %s",
	// AAudio_convertStreamStateToText(state));
	switch (state) {
	case AAUDIO_STREAM_STATE_OPEN:
	case AAUDIO_STREAM_STATE_STARTING:
	if (framesWritten != 0) {
	// Don't wait to write more data. Just prime the buffer.
	wakeTime = currentNanoTime;
	}
	break;
	case AAUDIO_STREAM_STATE_STARTED:
	{
	// Calculate when there will be room available to write to the buffer.
	// If the appBufferSize is smaller than the endpointBufferSize then
	// we will have room to write data beyond the appBufferSize.
	// That is a technique used to reduce glitches without adding latency.
	const int64_t appBufferSize = getDeviceBufferSize();
	// The endpoint buffer size is set to the maximum that can be written.
	// If we use it then we must carve out some room to write data when we wake up.
	const int64_t endBufferSize = mAudioEndpoint->getBufferSizeInFrames()
	- getDeviceFramesPerBurst();
	const int64_t bestBufferSize = std::min(appBufferSize, endBufferSize);
	int64_t targetReadPosition = mAudioEndpoint->getDataWriteCounter() - bestBufferSize;
	wakeTime = mClockModel.convertPositionToTime(targetReadPosition);
	}
	break;
	default:
	break;
	}
	*wakeTimePtr = wakeTime;

	}

	ATRACE_END();
	return framesWritten;
	}


	aaudio_result_t AudioStreamInternalPlay::writeNowWithConversion(const void *buffer,
	int32_t numFrames) {
	WrappingBuffer wrappingBuffer;
	uint8_t byteBuffer = (uint8_t ) buffer;
	int32_t framesLeftInByteBuffer = numFrames;

	mAudioEndpoint->getEmptyFramesAvailable(&wrappingBuffer);

	// Write data in one or two parts.
	int partIndex = 0;
	int framesWrittenToAudioEndpoint = 0;
	while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
	int32_t framesAvailableInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
	uint8_t currentWrappingBuffer = (uint8_t ) wrappingBuffer.data[partIndex];

	if (framesAvailableInWrappingBuffer > 0) {
	// Pull data from the flowgraph in case there is residual data.
	const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.pull(
	(void*) currentWrappingBuffer,
	framesAvailableInWrappingBuffer);

	const int32_t numBytesActuallyWrittenToWrappingBuffer =
	framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
	currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
	framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
	framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;
	} else {
	break;
	}

	// Put data from byteBuffer into the flowgraph one buffer (8 frames) at a time.
	// Continuously pull as much data as possible from the flowgraph into the wrapping buffer.
	// The return value of mFlowGraph.process is the number of frames actually pulled.
	while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
	int32_t framesToWriteFromByteBuffer = std::min(flowgraph::kDefaultBufferSize,
	framesLeftInByteBuffer);
	// If the wrapping buffer is running low, write one frame at a time.
	if (framesAvailableInWrappingBuffer < flowgraph::kDefaultBufferSize) {
	framesToWriteFromByteBuffer = 1;
	}

	const int32_t numBytesToWriteFromByteBuffer = getBytesPerFrame() *
	framesToWriteFromByteBuffer;

	//ALOGD("%s() framesLeftInByteBuffer %d, framesAvailableInWrappingBuffer %d"
	// "framesToWriteFromByteBuffer %d, numBytesToWriteFromByteBuffer %d"
	// , __func__, framesLeftInByteBuffer, framesAvailableInWrappingBuffer,
	// framesToWriteFromByteBuffer, numBytesToWriteFromByteBuffer);

	const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.process(
	(void *)byteBuffer,
	framesToWriteFromByteBuffer,
	(void *)currentWrappingBuffer,
	framesAvailableInWrappingBuffer);

	byteBuffer += numBytesToWriteFromByteBuffer;
	framesLeftInByteBuffer -= framesToWriteFromByteBuffer;
	const int32_t numBytesActuallyWrittenToWrappingBuffer =
	framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
	currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
	framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
	framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;

	//ALOGD("%s() numBytesActuallyWrittenToWrappingBuffer %d, framesLeftInByteBuffer %d"
	// "framesActuallyWrittenToWrappingBuffer %d, numBytesToWriteFromByteBuffer %d"
	// "framesWrittenToAudioEndpoint %d"
	// , __func__, numBytesActuallyWrittenToWrappingBuffer, framesLeftInByteBuffer,
	// framesActuallyWrittenToWrappingBuffer, numBytesToWriteFromByteBuffer,
	// framesWrittenToAudioEndpoint);
	}
	partIndex++;
	}
	//ALOGD("%s() framesWrittenToAudioEndpoint %d, numFrames %d"
	// "framesLeftInByteBuffer %d"
	// , __func__, framesWrittenToAudioEndpoint, numFrames,
	// framesLeftInByteBuffer);

	// The audio endpoint should reference the number of frames written to the wrapping buffer.
	mAudioEndpoint->advanceWriteIndex(framesWrittenToAudioEndpoint);

	// The internal code should use the number of frames read from the app.
	return numFrames - framesLeftInByteBuffer;
	}

	int64_t AudioStreamInternalPlay::getFramesRead() {
	if (mAudioEndpoint) {
	const int64_t framesReadHardware = isClockModelInControl()
	? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
	: mAudioEndpoint->getDataReadCounter();
	// Add service offset and prevent retrograde motion.
	mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
	}
	return mLastFramesRead;
	}

	int64_t AudioStreamInternalPlay::getFramesWritten() {
	if (mAudioEndpoint) {
	mLastFramesWritten = std::max(
	mLastFramesWritten,
	mAudioEndpoint->getDataWriteCounter() + mFramesOffsetFromService);
	}
	return mLastFramesWritten;
	}

	// Render audio in the application callback and then write the data to the stream.
	void *AudioStreamInternalPlay::callbackLoop() {
	ALOGD("%s() entering >>>>>>>>>>>>>>>", __func__);
	aaudio_result_t result = AAUDIO_OK;
	aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
	if (!isDataCallbackSet()) return nullptr;
	int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);

	// result might be a frame count
	while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
	// Call application using the AAudio callback interface.
	callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);

	if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
	// Write audio data to stream. This is a BLOCKING WRITE!
	result = write(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
	if ((result != mCallbackFrames)) {
	if (result >= 0) {
	// Only wrote some of the frames requested. The stream can be disconnected
	// or timed out.
	processCommands();
	result = isDisconnected() ? AAUDIO_ERROR_DISCONNECTED : AAUDIO_ERROR_TIMEOUT;
	}
	maybeCallErrorCallback(result);
	break;
	}
	} else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
	ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
	result = systemStopInternal();
	break;
	}
	}

	ALOGD("%s() exiting, result = %d, isActive() = %d <<<<<<<<<<<<<<",
	__func__, result, (int) isActive());
	return nullptr;
	}

	//------------------------------------------------------------------------------
	// Implementation of PlayerBase
	status_t AudioStreamInternalPlay::doSetVolume() {
	float combinedVolume = mStreamVolume * getDuckAndMuteVolume();
	ALOGD("%s() mStreamVolume * duckAndMuteVolume = %f * %f = %f",
	__func__, mStreamVolume, getDuckAndMuteVolume(), combinedVolume);
	mFlowGraph.setTargetVolume(combinedVolume);
	return android::NO_ERROR;
	}