services/oboeservice/AAudioServiceStreamShared.cpp - third_party/android/platform/frameworks/av - Git at Google

 /*
  * 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_TAG "AAudioServiceStreamShared"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>

 #include <mutex>

 #include <aaudio/AAudio.h>

 #include "binding/IAAudioService.h"

 #include "binding/AAudioServiceMessage.h"
 #include "AAudioServiceStreamBase.h"
 #include "AAudioServiceStreamShared.h"
 #include "AAudioEndpointManager.h"
 #include "AAudioService.h"
 #include "AAudioServiceEndpoint.h"

 using namespace android;
 using namespace aaudio;

 #define MIN_BURSTS_PER_BUFFER       2
 #define DEFAULT_BURSTS_PER_BUFFER   16
 // This is an arbitrary range. TODO review.
 #define MAX_FRAMES_PER_BUFFER       (32 * 1024)

 AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
     : mAudioService(audioService)
     {
 }

 int32_t AAudioServiceStreamShared::calculateBufferCapacity(int32_t requestedCapacityFrames,
                                                            int32_t framesPerBurst) {

     if (requestedCapacityFrames > MAX_FRAMES_PER_BUFFER) {
         ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() requested capacity %d > max %d",
               requestedCapacityFrames, MAX_FRAMES_PER_BUFFER);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }

     // Determine how many bursts will fit in the buffer.
     int32_t numBursts;
     if (requestedCapacityFrames == AAUDIO_UNSPECIFIED) {
         // Use fewer bursts if default is too many.
         if ((DEFAULT_BURSTS_PER_BUFFER * framesPerBurst) > MAX_FRAMES_PER_BUFFER) {
             numBursts = MAX_FRAMES_PER_BUFFER / framesPerBurst;
         } else {
             numBursts = DEFAULT_BURSTS_PER_BUFFER;
         }
     } else {
         // round up to nearest burst boundary
         numBursts = (requestedCapacityFrames + framesPerBurst - 1) / framesPerBurst;
     }

     // Clip to bare minimum.
     if (numBursts < MIN_BURSTS_PER_BUFFER) {
         numBursts = MIN_BURSTS_PER_BUFFER;
     }
     // Check for numeric overflow.
     if (numBursts > 0x8000 || framesPerBurst > 0x8000) {
         ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() overflow, capacity = %d * %d",
               numBursts, framesPerBurst);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
     int32_t capacityInFrames = numBursts * framesPerBurst;

     // Final sanity check.
     if (capacityInFrames > MAX_FRAMES_PER_BUFFER) {
         ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() calc capacity %d > max %d",
               capacityInFrames, MAX_FRAMES_PER_BUFFER);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
     ALOGD("AAudioServiceStreamShared::calculateBufferCapacity() requested %d frames, actual = %d",
           requestedCapacityFrames, capacityInFrames);
     return capacityInFrames;
 }

 aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request,
                      aaudio::AAudioStreamConfiguration &configurationOutput)  {

     sp<AAudioServiceStreamShared> keep(this);

     aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamBase open() returned %d", result);
         return result;
     }

     const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
     aaudio_direction_t direction = request.getDirection();

     AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
     mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService, configurationOutput, direction);
     if (mServiceEndpoint == nullptr) {
         ALOGE("AAudioServiceStreamShared::open() mServiceEndPoint = %p", mServiceEndpoint);
         return AAUDIO_ERROR_UNAVAILABLE;
     }

     // Is the request compatible with the shared endpoint?
     mAudioFormat = configurationInput.getFormat();
     if (mAudioFormat == AAUDIO_FORMAT_UNSPECIFIED) {
         mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
     } else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
         ALOGE("AAudioServiceStreamShared::open() mAudioFormat = %d, need FLOAT", mAudioFormat);
         result = AAUDIO_ERROR_INVALID_FORMAT;
         goto error;
     }

     mSampleRate = configurationInput.getSampleRate();
     if (mSampleRate == AAUDIO_UNSPECIFIED) {
         mSampleRate = mServiceEndpoint->getSampleRate();
     } else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
         ALOGE("AAudioServiceStreamShared::open() mSampleRate = %d, need %d",
               mSampleRate, mServiceEndpoint->getSampleRate());
         result = AAUDIO_ERROR_INVALID_RATE;
         goto error;
     }

     mSamplesPerFrame = configurationInput.getSamplesPerFrame();
     if (mSamplesPerFrame == AAUDIO_UNSPECIFIED) {
         mSamplesPerFrame = mServiceEndpoint->getSamplesPerFrame();
     } else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
         ALOGE("AAudioServiceStreamShared::open() mSamplesPerFrame = %d, need %d",
               mSamplesPerFrame, mServiceEndpoint->getSamplesPerFrame());
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }

     mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
     ALOGD("AAudioServiceStreamShared::open() mSampleRate = %d, mFramesPerBurst = %d",
           mSampleRate, mFramesPerBurst);

     mCapacityInFrames = calculateBufferCapacity(configurationInput.getBufferCapacity(),
                                      mFramesPerBurst);
     if (mCapacityInFrames < 0) {
         result = mCapacityInFrames; // negative error code
         mCapacityInFrames = 0;
         goto error;
     }

     // Create audio data shared memory buffer for client.
     mAudioDataQueue = new SharedRingBuffer();
     result = mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::open() could not allocate FIFO with %d frames",
               mCapacityInFrames);
         result = AAUDIO_ERROR_NO_MEMORY;
         goto error;
     }

     ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
           mSampleRate, mSamplesPerFrame, mServiceEndpoint->getDeviceId());

     // Fill in configuration for client.
     configurationOutput.setSampleRate(mSampleRate);
     configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
     configurationOutput.setFormat(mAudioFormat);
     configurationOutput.setDeviceId(mServiceEndpoint->getDeviceId());

     result = mServiceEndpoint->registerStream(keep);
     if (result != AAUDIO_OK) {
         goto error;
     }

     setState(AAUDIO_STREAM_STATE_OPEN);
     return AAUDIO_OK;

 error:
     close();
     return result;
 }

 /**
  * Start the flow of audio data.
  *
  * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
  */
 aaudio_result_t AAudioServiceStreamShared::start()  {
     if (isRunning()) {
         return AAUDIO_OK;
     }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
     // For output streams, this will add the stream to the mixer.
     aaudio_result_t result = endpoint->startStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
         disconnect();
     } else {
         result = endpoint->getStreamInternal()->startClient(mMmapClient, &mClientHandle);
         if (result == AAUDIO_OK) {
             result = AAudioServiceStreamBase::start();
         }
     }
     return result;
 }

 /**
  * Stop the flow of data so that start() can resume without loss of data.
  *
  * An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
 */
 aaudio_result_t AAudioServiceStreamShared::pause()  {
     if (!isRunning()) {
         return AAUDIO_OK;
     }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
     endpoint->getStreamInternal()->stopClient(mClientHandle);
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
         disconnect(); // TODO should we return or pause Base first?
     }
     return AAudioServiceStreamBase::pause();
 }

 aaudio_result_t AAudioServiceStreamShared::stop()  {
     if (!isRunning()) {
         return AAUDIO_OK;
     }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
     endpoint->getStreamInternal()->stopClient(mClientHandle);
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
         disconnect();
     }
     return AAudioServiceStreamBase::stop();
 }

 /**
  *  Discard any data held by the underlying HAL or Service.
  *
  * An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
  */
 aaudio_result_t AAudioServiceStreamShared::flush()  {
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
     if (mState != AAUDIO_STREAM_STATE_PAUSED) {
          ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
             AAudio_convertStreamStateToText(mState));
         return AAUDIO_ERROR_INVALID_STATE;
     }
     // Data will get flushed when the client receives the FLUSHED event.
     return AAudioServiceStreamBase::flush();
 }

 aaudio_result_t AAudioServiceStreamShared::close()  {
     if (mState == AAUDIO_STREAM_STATE_CLOSED) {
         return AAUDIO_OK;
     }

     stop();

     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }

     endpoint->unregisterStream(this);

     AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
     mEndpointManager.closeEndpoint(endpoint);
     mServiceEndpoint = nullptr;

     if (mAudioDataQueue != nullptr) {
         delete mAudioDataQueue;
         mAudioDataQueue = nullptr;
     }
     return AAudioServiceStreamBase::close();
 }

 /**
  * Get an immutable description of the data queue created by this service.
  */
 aaudio_result_t AAudioServiceStreamShared::getDownDataDescription(AudioEndpointParcelable &parcelable)
 {
     // Gather information on the data queue.
     mAudioDataQueue->fillParcelable(parcelable,
                                     parcelable.mDownDataQueueParcelable);
     parcelable.mDownDataQueueParcelable.setFramesPerBurst(getFramesPerBurst());
     return AAUDIO_OK;
 }

 void AAudioServiceStreamShared::markTransferTime(int64_t nanoseconds) {
     mMarkedPosition = mAudioDataQueue->getFifoBuffer()->getReadCounter();
     mMarkedTime = nanoseconds;
 }

 aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
                                                                 int64_t *timeNanos) {
     // TODO get these two numbers as an atomic pair
     *positionFrames = mMarkedPosition;
     *timeNanos = mMarkedTime;
     return AAUDIO_OK;
 }
	/*
	* 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_TAG "AAudioServiceStreamShared"
	//#define LOG_NDEBUG 0
	#include <utils/Log.h>

	#include <mutex>

	#include <aaudio/AAudio.h>

	#include "binding/IAAudioService.h"

	#include "binding/AAudioServiceMessage.h"
	#include "AAudioServiceStreamBase.h"
	#include "AAudioServiceStreamShared.h"
	#include "AAudioEndpointManager.h"
	#include "AAudioService.h"
	#include "AAudioServiceEndpoint.h"

	using namespace android;
	using namespace aaudio;

	#define MIN_BURSTS_PER_BUFFER 2
	#define DEFAULT_BURSTS_PER_BUFFER 16
	// This is an arbitrary range. TODO review.
	#define MAX_FRAMES_PER_BUFFER (32 * 1024)

	AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
	: mAudioService(audioService)
	{
	}

	int32_t AAudioServiceStreamShared::calculateBufferCapacity(int32_t requestedCapacityFrames,
	int32_t framesPerBurst) {

	if (requestedCapacityFrames > MAX_FRAMES_PER_BUFFER) {
	ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() requested capacity %d > max %d",
	requestedCapacityFrames, MAX_FRAMES_PER_BUFFER);
	return AAUDIO_ERROR_OUT_OF_RANGE;
	}

	// Determine how many bursts will fit in the buffer.
	int32_t numBursts;
	if (requestedCapacityFrames == AAUDIO_UNSPECIFIED) {
	// Use fewer bursts if default is too many.
	if ((DEFAULT_BURSTS_PER_BUFFER * framesPerBurst) > MAX_FRAMES_PER_BUFFER) {
	numBursts = MAX_FRAMES_PER_BUFFER / framesPerBurst;
	} else {
	numBursts = DEFAULT_BURSTS_PER_BUFFER;
	}
	} else {
	// round up to nearest burst boundary
	numBursts = (requestedCapacityFrames + framesPerBurst - 1) / framesPerBurst;
	}

	// Clip to bare minimum.
	if (numBursts < MIN_BURSTS_PER_BUFFER) {
	numBursts = MIN_BURSTS_PER_BUFFER;
	}
	// Check for numeric overflow.
	if (numBursts > 0x8000 \|\| framesPerBurst > 0x8000) {
	ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() overflow, capacity = %d * %d",
	numBursts, framesPerBurst);
	return AAUDIO_ERROR_OUT_OF_RANGE;
	}
	int32_t capacityInFrames = numBursts * framesPerBurst;

	// Final sanity check.
	if (capacityInFrames > MAX_FRAMES_PER_BUFFER) {
	ALOGE("AAudioServiceStreamShared::calculateBufferCapacity() calc capacity %d > max %d",
	capacityInFrames, MAX_FRAMES_PER_BUFFER);
	return AAUDIO_ERROR_OUT_OF_RANGE;
	}
	ALOGD("AAudioServiceStreamShared::calculateBufferCapacity() requested %d frames, actual = %d",
	requestedCapacityFrames, capacityInFrames);
	return capacityInFrames;
	}

	aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request,
	aaudio::AAudioStreamConfiguration &configurationOutput) {

	sp<AAudioServiceStreamShared> keep(this);

	aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
	if (result != AAUDIO_OK) {
	ALOGE("AAudioServiceStreamBase open() returned %d", result);
	return result;
	}

	const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
	aaudio_direction_t direction = request.getDirection();

	AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
	mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService, configurationOutput, direction);
	if (mServiceEndpoint == nullptr) {
	ALOGE("AAudioServiceStreamShared::open() mServiceEndPoint = %p", mServiceEndpoint);
	return AAUDIO_ERROR_UNAVAILABLE;
	}

	// Is the request compatible with the shared endpoint?
	mAudioFormat = configurationInput.getFormat();
	if (mAudioFormat == AAUDIO_FORMAT_UNSPECIFIED) {
	mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
	} else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
	ALOGE("AAudioServiceStreamShared::open() mAudioFormat = %d, need FLOAT", mAudioFormat);
	result = AAUDIO_ERROR_INVALID_FORMAT;
	goto error;
	}

	mSampleRate = configurationInput.getSampleRate();
	if (mSampleRate == AAUDIO_UNSPECIFIED) {
	mSampleRate = mServiceEndpoint->getSampleRate();
	} else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
	ALOGE("AAudioServiceStreamShared::open() mSampleRate = %d, need %d",
	mSampleRate, mServiceEndpoint->getSampleRate());
	result = AAUDIO_ERROR_INVALID_RATE;
	goto error;
	}

	mSamplesPerFrame = configurationInput.getSamplesPerFrame();
	if (mSamplesPerFrame == AAUDIO_UNSPECIFIED) {
	mSamplesPerFrame = mServiceEndpoint->getSamplesPerFrame();
	} else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
	ALOGE("AAudioServiceStreamShared::open() mSamplesPerFrame = %d, need %d",
	mSamplesPerFrame, mServiceEndpoint->getSamplesPerFrame());
	result = AAUDIO_ERROR_OUT_OF_RANGE;
	goto error;
	}

	mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
	ALOGD("AAudioServiceStreamShared::open() mSampleRate = %d, mFramesPerBurst = %d",
	mSampleRate, mFramesPerBurst);

	mCapacityInFrames = calculateBufferCapacity(configurationInput.getBufferCapacity(),
	mFramesPerBurst);
	if (mCapacityInFrames < 0) {
	result = mCapacityInFrames; // negative error code
	mCapacityInFrames = 0;
	goto error;
	}

	// Create audio data shared memory buffer for client.
	mAudioDataQueue = new SharedRingBuffer();
	result = mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
	if (result != AAUDIO_OK) {
	ALOGE("AAudioServiceStreamShared::open() could not allocate FIFO with %d frames",
	mCapacityInFrames);
	result = AAUDIO_ERROR_NO_MEMORY;
	goto error;
	}

	ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
	mSampleRate, mSamplesPerFrame, mServiceEndpoint->getDeviceId());

	// Fill in configuration for client.
	configurationOutput.setSampleRate(mSampleRate);
	configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
	configurationOutput.setFormat(mAudioFormat);
	configurationOutput.setDeviceId(mServiceEndpoint->getDeviceId());

	result = mServiceEndpoint->registerStream(keep);
	if (result != AAUDIO_OK) {
	goto error;
	}

	setState(AAUDIO_STREAM_STATE_OPEN);
	return AAUDIO_OK;

	error:
	close();
	return result;
	}

	/**
	* Start the flow of audio data.
	*
	* An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
	*/
	aaudio_result_t AAudioServiceStreamShared::start() {
	if (isRunning()) {
	return AAUDIO_OK;
	}
	AAudioServiceEndpoint *endpoint = mServiceEndpoint;
	if (endpoint == nullptr) {
	return AAUDIO_ERROR_INVALID_STATE;
	}
	// For output streams, this will add the stream to the mixer.
	aaudio_result_t result = endpoint->startStream(this);
	if (result != AAUDIO_OK) {
	ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
	disconnect();
	} else {
	result = endpoint->getStreamInternal()->startClient(mMmapClient, &mClientHandle);
	if (result == AAUDIO_OK) {
	result = AAudioServiceStreamBase::start();
	}
	}
	return result;
	}

	/**
	* Stop the flow of data so that start() can resume without loss of data.
	*
	* An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
	*/
	aaudio_result_t AAudioServiceStreamShared::pause() {
	if (!isRunning()) {
	return AAUDIO_OK;
	}
	AAudioServiceEndpoint *endpoint = mServiceEndpoint;
	if (endpoint == nullptr) {
	return AAUDIO_ERROR_INVALID_STATE;
	}
	endpoint->getStreamInternal()->stopClient(mClientHandle);
	aaudio_result_t result = endpoint->stopStream(this);
	if (result != AAUDIO_OK) {
	ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
	disconnect(); // TODO should we return or pause Base first?
	}
	return AAudioServiceStreamBase::pause();
	}

	aaudio_result_t AAudioServiceStreamShared::stop() {
	if (!isRunning()) {
	return AAUDIO_OK;
	}
	AAudioServiceEndpoint *endpoint = mServiceEndpoint;
	if (endpoint == nullptr) {
	return AAUDIO_ERROR_INVALID_STATE;
	}
	endpoint->getStreamInternal()->stopClient(mClientHandle);
	aaudio_result_t result = endpoint->stopStream(this);
	if (result != AAUDIO_OK) {
	ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
	disconnect();
	}
	return AAudioServiceStreamBase::stop();
	}

	/**
	* Discard any data held by the underlying HAL or Service.
	*
	* An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
	*/
	aaudio_result_t AAudioServiceStreamShared::flush() {
	AAudioServiceEndpoint *endpoint = mServiceEndpoint;
	if (endpoint == nullptr) {
	return AAUDIO_ERROR_INVALID_STATE;
	}
	if (mState != AAUDIO_STREAM_STATE_PAUSED) {
	ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
	AAudio_convertStreamStateToText(mState));
	return AAUDIO_ERROR_INVALID_STATE;
	}
	// Data will get flushed when the client receives the FLUSHED event.
	return AAudioServiceStreamBase::flush();
	}

	aaudio_result_t AAudioServiceStreamShared::close() {
	if (mState == AAUDIO_STREAM_STATE_CLOSED) {
	return AAUDIO_OK;
	}

	stop();

	AAudioServiceEndpoint *endpoint = mServiceEndpoint;
	if (endpoint == nullptr) {
	return AAUDIO_ERROR_INVALID_STATE;
	}

	endpoint->unregisterStream(this);

	AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
	mEndpointManager.closeEndpoint(endpoint);
	mServiceEndpoint = nullptr;

	if (mAudioDataQueue != nullptr) {
	delete mAudioDataQueue;
	mAudioDataQueue = nullptr;
	}
	return AAudioServiceStreamBase::close();
	}

	/**
	* Get an immutable description of the data queue created by this service.
	*/
	aaudio_result_t AAudioServiceStreamShared::getDownDataDescription(AudioEndpointParcelable &parcelable)
	{
	// Gather information on the data queue.
	mAudioDataQueue->fillParcelable(parcelable,
	parcelable.mDownDataQueueParcelable);
	parcelable.mDownDataQueueParcelable.setFramesPerBurst(getFramesPerBurst());
	return AAUDIO_OK;
	}

	void AAudioServiceStreamShared::markTransferTime(int64_t nanoseconds) {
	mMarkedPosition = mAudioDataQueue->getFifoBuffer()->getReadCounter();
	mMarkedTime = nanoseconds;
	}

	aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
	int64_t *timeNanos) {
	// TODO get these two numbers as an atomic pair
	*positionFrames = mMarkedPosition;
	*timeNanos = mMarkedTime;
	return AAUDIO_OK;
	}