modules/audio_device/android/aaudio_wrapper.cc - src - Git at Google

 /*
  *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "modules/audio_device/android/aaudio_wrapper.h"

 #include "modules/audio_device/android/audio_manager.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/time_utils.h"

 #define LOG_ON_ERROR(op)                                                      \
   do {                                                                        \
     aaudio_result_t result = (op);                                            \
     if (result != AAUDIO_OK) {                                                \
       RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
     }                                                                         \
   } while (0)

 #define RETURN_ON_ERROR(op, ...)                                              \
   do {                                                                        \
     aaudio_result_t result = (op);                                            \
     if (result != AAUDIO_OK) {                                                \
       RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
       return __VA_ARGS__;                                                     \
     }                                                                         \
   } while (0)

 namespace webrtc {

 namespace {

 const char* DirectionToString(aaudio_direction_t direction) {
   switch (direction) {
     case AAUDIO_DIRECTION_OUTPUT:
       return "OUTPUT";
     case AAUDIO_DIRECTION_INPUT:
       return "INPUT";
     default:
       return "UNKNOWN";
   }
 }

 const char* SharingModeToString(aaudio_sharing_mode_t mode) {
   switch (mode) {
     case AAUDIO_SHARING_MODE_EXCLUSIVE:
       return "EXCLUSIVE";
     case AAUDIO_SHARING_MODE_SHARED:
       return "SHARED";
     default:
       return "UNKNOWN";
   }
 }

 const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
   switch (mode) {
     case AAUDIO_PERFORMANCE_MODE_NONE:
       return "NONE";
     case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
       return "POWER_SAVING";
     case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
       return "LOW_LATENCY";
     default:
       return "UNKNOWN";
   }
 }

 const char* FormatToString(int32_t id) {
   switch (id) {
     case AAUDIO_FORMAT_INVALID:
       return "INVALID";
     case AAUDIO_FORMAT_UNSPECIFIED:
       return "UNSPECIFIED";
     case AAUDIO_FORMAT_PCM_I16:
       return "PCM_I16";
     case AAUDIO_FORMAT_PCM_FLOAT:
       return "FLOAT";
     default:
       return "UNKNOWN";
   }
 }

 void ErrorCallback(AAudioStream* stream,
                    void* user_data,
                    aaudio_result_t error) {
   RTC_DCHECK(user_data);
   AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
   RTC_LOG(WARNING) << "ErrorCallback: "
                    << DirectionToString(aaudio_wrapper->direction());
   RTC_DCHECK(aaudio_wrapper->observer());
   aaudio_wrapper->observer()->OnErrorCallback(error);
 }

 aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
                                            void* user_data,
                                            void* audio_data,
                                            int32_t num_frames) {
   RTC_DCHECK(user_data);
   RTC_DCHECK(audio_data);
   AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
   RTC_DCHECK(aaudio_wrapper->observer());
   return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
 }

 // Wraps the stream builder object to ensure that it is released properly when
 // the stream builder goes out of scope.
 class ScopedStreamBuilder {
  public:
   ScopedStreamBuilder() {
     LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
     RTC_DCHECK(builder_);
   }
   ~ScopedStreamBuilder() {
     if (builder_) {
       LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
     }
   }

   AAudioStreamBuilder* get() const { return builder_; }

  private:
   AAudioStreamBuilder* builder_ = nullptr;
 };

 }  // namespace

 AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
                              aaudio_direction_t direction,
                              AAudioObserverInterface* observer)
     : direction_(direction), observer_(observer) {
   RTC_LOG(INFO) << "ctor";
   RTC_DCHECK(observer_);
   direction_ == AAUDIO_DIRECTION_OUTPUT
       ? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
       : audio_parameters_ = audio_manager->GetRecordAudioParameters();
   aaudio_thread_checker_.Detach();
   RTC_LOG(INFO) << audio_parameters_.ToString();
 }

 AAudioWrapper::~AAudioWrapper() {
   RTC_LOG(INFO) << "dtor";
   RTC_DCHECK(thread_checker_.IsCurrent());
   RTC_DCHECK(!stream_);
 }

 bool AAudioWrapper::Init() {
   RTC_LOG(INFO) << "Init";
   RTC_DCHECK(thread_checker_.IsCurrent());
   // Creates a stream builder which can be used to open an audio stream.
   ScopedStreamBuilder builder;
   // Configures the stream builder using audio parameters given at construction.
   SetStreamConfiguration(builder.get());
   // Opens a stream based on options in the stream builder.
   if (!OpenStream(builder.get())) {
     return false;
   }
   // Ensures that the opened stream could activate the requested settings.
   if (!VerifyStreamConfiguration()) {
     return false;
   }
   // Optimizes the buffer scheme for lowest possible latency and creates
   // additional buffer logic to match the 10ms buffer size used in WebRTC.
   if (!OptimizeBuffers()) {
     return false;
   }
   LogStreamState();
   return true;
 }

 bool AAudioWrapper::Start() {
   RTC_LOG(INFO) << "Start";
   RTC_DCHECK(thread_checker_.IsCurrent());
   // TODO(henrika): this state check might not be needed.
   aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
   if (current_state != AAUDIO_STREAM_STATE_OPEN) {
     RTC_LOG(LS_ERROR) << "Invalid state: "
                       << AAudio_convertStreamStateToText(current_state);
     return false;
   }
   // Asynchronous request for the stream to start.
   RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
   LogStreamState();
   return true;
 }

 bool AAudioWrapper::Stop() {
   RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
   RTC_DCHECK(thread_checker_.IsCurrent());
   // Asynchronous request for the stream to stop.
   RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
   CloseStream();
   aaudio_thread_checker_.Detach();
   return true;
 }

 double AAudioWrapper::EstimateLatencyMillis() const {
   RTC_DCHECK(stream_);
   double latency_millis = 0.0;
   if (direction() == AAUDIO_DIRECTION_INPUT) {
     // For input streams. Best guess we can do is to use the current burst size
     // as delay estimate.
     latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
                      rtc::kNumMillisecsPerSec;
   } else {
     int64_t existing_frame_index;
     int64_t existing_frame_presentation_time;
     // Get the time at which a particular frame was presented to audio hardware.
     aaudio_result_t result = AAudioStream_getTimestamp(
         stream_, CLOCK_MONOTONIC, &existing_frame_index,
         &existing_frame_presentation_time);
     // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
     if (result == AAUDIO_OK) {
       // Get write index for next audio frame.
       int64_t next_frame_index = frames_written();
       // Number of frames between next frame and the existing frame.
       int64_t frame_index_delta = next_frame_index - existing_frame_index;
       // Assume the next frame will be written now.
       int64_t next_frame_write_time = rtc::TimeNanos();
       // Calculate time when next frame will be presented to the hardware taking
       // sample rate into account.
       int64_t frame_time_delta =
           (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
       int64_t next_frame_presentation_time =
           existing_frame_presentation_time + frame_time_delta;
       // Derive a latency estimate given results above.
       latency_millis = static_cast<double>(next_frame_presentation_time -
                                            next_frame_write_time) /
                        rtc::kNumNanosecsPerMillisec;
     }
   }
   return latency_millis;
 }

 // Returns new buffer size or a negative error value if buffer size could not
 // be increased.
 bool AAudioWrapper::IncreaseOutputBufferSize() {
   RTC_LOG(INFO) << "IncreaseBufferSize";
   RTC_DCHECK(stream_);
   RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
   RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
   aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
   // Try to increase size of buffer with one burst to reduce risk of underrun.
   buffer_size += frames_per_burst();
   // Verify that the new buffer size is not larger than max capacity.
   // TODO(henrika): keep track of case when we reach the capacity limit.
   const int32_t max_buffer_size = buffer_capacity_in_frames();
   if (buffer_size > max_buffer_size) {
     RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
                       << ") is higher than max: " << max_buffer_size;
     return false;
   }
   RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
                 << " (max=" << max_buffer_size << ")";
   buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
   if (buffer_size < 0) {
     RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
                       << AAudio_convertResultToText(buffer_size);
     return false;
   }
   RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
   return true;
 }

 void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
   RTC_LOG(INFO) << "ClearInputStream";
   RTC_DCHECK(stream_);
   RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
   RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
   aaudio_result_t cleared_frames = 0;
   do {
     cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
   } while (cleared_frames > 0);
 }

 AAudioObserverInterface* AAudioWrapper::observer() const {
   return observer_;
 }

 AudioParameters AAudioWrapper::audio_parameters() const {
   return audio_parameters_;
 }

 int32_t AAudioWrapper::samples_per_frame() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getSamplesPerFrame(stream_);
 }

 int32_t AAudioWrapper::buffer_size_in_frames() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getBufferSizeInFrames(stream_);
 }

 int32_t AAudioWrapper::buffer_capacity_in_frames() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getBufferCapacityInFrames(stream_);
 }

 int32_t AAudioWrapper::device_id() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getDeviceId(stream_);
 }

 int32_t AAudioWrapper::xrun_count() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getXRunCount(stream_);
 }

 int32_t AAudioWrapper::format() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getFormat(stream_);
 }

 int32_t AAudioWrapper::sample_rate() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getSampleRate(stream_);
 }

 int32_t AAudioWrapper::channel_count() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getChannelCount(stream_);
 }

 int32_t AAudioWrapper::frames_per_callback() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getFramesPerDataCallback(stream_);
 }

 aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getSharingMode(stream_);
 }

 aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getPerformanceMode(stream_);
 }

 aaudio_stream_state_t AAudioWrapper::stream_state() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getState(stream_);
 }

 int64_t AAudioWrapper::frames_written() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getFramesWritten(stream_);
 }

 int64_t AAudioWrapper::frames_read() const {
   RTC_DCHECK(stream_);
   return AAudioStream_getFramesRead(stream_);
 }

 void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
   RTC_LOG(INFO) << "SetStreamConfiguration";
   RTC_DCHECK(builder);
   RTC_DCHECK(thread_checker_.IsCurrent());
   // Request usage of default primary output/input device.
   // TODO(henrika): verify that default device follows Java APIs.
   // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
   AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
   // Use preferred sample rate given by the audio parameters.
   AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
   // Use preferred channel configuration given by the audio parameters.
   AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
   // Always use 16-bit PCM audio sample format.
   AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
   // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
   // Ask for exclusive mode since this will give us the lowest possible latency.
   // If exclusive mode isn't available, shared mode will be used instead.
   AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
   // Use the direction that was given at construction.
   AAudioStreamBuilder_setDirection(builder, direction_);
   // TODO(henrika): investigate performance using different performance modes.
   AAudioStreamBuilder_setPerformanceMode(builder,
                                          AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
   // Given that WebRTC applications require low latency, our audio stream uses
   // an asynchronous callback function to transfer data to and from the
   // application. AAudio executes the callback in a higher-priority thread that
   // has better performance.
   AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
   // Request that AAudio calls this functions if any error occurs on a callback
   // thread.
   AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
 }

 bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
   RTC_LOG(INFO) << "OpenStream";
   RTC_DCHECK(builder);
   AAudioStream* stream = nullptr;
   RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
   stream_ = stream;
   LogStreamConfiguration();
   return true;
 }

 void AAudioWrapper::CloseStream() {
   RTC_LOG(INFO) << "CloseStream";
   RTC_DCHECK(stream_);
   LOG_ON_ERROR(AAudioStream_close(stream_));
   stream_ = nullptr;
 }

 void AAudioWrapper::LogStreamConfiguration() {
   RTC_DCHECK(stream_);
   char ss_buf[1024];
   rtc::SimpleStringBuilder ss(ss_buf);
   ss << "Stream Configuration: ";
   ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
   ss << ", samples per frame=" << samples_per_frame();
   ss << ", format=" << FormatToString(format());
   ss << ", sharing mode=" << SharingModeToString(sharing_mode());
   ss << ", performance mode=" << PerformanceModeToString(performance_mode());
   ss << ", direction=" << DirectionToString(direction());
   ss << ", device id=" << AAudioStream_getDeviceId(stream_);
   ss << ", frames per callback=" << frames_per_callback();
   RTC_LOG(INFO) << ss.str();
 }

 void AAudioWrapper::LogStreamState() {
   RTC_LOG(INFO) << "AAudio stream state: "
                 << AAudio_convertStreamStateToText(stream_state());
 }

 bool AAudioWrapper::VerifyStreamConfiguration() {
   RTC_LOG(INFO) << "VerifyStreamConfiguration";
   RTC_DCHECK(stream_);
   // TODO(henrika): should we verify device ID as well?
   if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
     RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
     return false;
   }
   if (AAudioStream_getChannelCount(stream_) !=
       static_cast<int32_t>(audio_parameters().channels())) {
     RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
     return false;
   }
   if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
     RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
     return false;
   }
   if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
     RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
     return false;
   }
   if (AAudioStream_getPerformanceMode(stream_) !=
       AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
     RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
     return false;
   }
   if (AAudioStream_getDirection(stream_) != direction()) {
     RTC_LOG(LS_ERROR) << "Stream direction could not be set";
     return false;
   }
   if (AAudioStream_getSamplesPerFrame(stream_) !=
       static_cast<int32_t>(audio_parameters().channels())) {
     RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
     return false;
   }
   return true;
 }

 bool AAudioWrapper::OptimizeBuffers() {
   RTC_LOG(INFO) << "OptimizeBuffers";
   RTC_DCHECK(stream_);
   // Maximum number of frames that can be filled without blocking.
   RTC_LOG(INFO) << "max buffer capacity in frames: "
                 << buffer_capacity_in_frames();
   // Query the number of frames that the application should read or write at
   // one time for optimal performance.
   int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
   RTC_LOG(INFO) << "frames per burst for optimal performance: "
                 << frames_per_burst;
   frames_per_burst_ = frames_per_burst;
   if (direction() == AAUDIO_DIRECTION_INPUT) {
     // There is no point in calling setBufferSizeInFrames() for input streams
     // since it has no effect on the performance (latency in this case).
     return true;
   }
   // Set buffer size to same as burst size to guarantee lowest possible latency.
   // This size might change for output streams if underruns are detected and
   // automatic buffer adjustment is enabled.
   AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
   int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
   if (buffer_size != frames_per_burst) {
     RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
     return false;
   }
   // Maximum number of frames that can be filled without blocking.
   RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
   return true;
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "modules/audio_device/android/aaudio_wrapper.h"

	#include "modules/audio_device/android/audio_manager.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/strings/string_builder.h"
	#include "rtc_base/time_utils.h"

	#define LOG_ON_ERROR(op) \
	do { \
	aaudio_result_t result = (op); \
	if (result != AAUDIO_OK) { \
	RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
	} \
	} while (0)

	#define RETURN_ON_ERROR(op, ...) \
	do { \
	aaudio_result_t result = (op); \
	if (result != AAUDIO_OK) { \
	RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
	return __VA_ARGS__; \
	} \
	} while (0)

	namespace webrtc {

	namespace {

	const char* DirectionToString(aaudio_direction_t direction) {
	switch (direction) {
	case AAUDIO_DIRECTION_OUTPUT:
	return "OUTPUT";
	case AAUDIO_DIRECTION_INPUT:
	return "INPUT";
	default:
	return "UNKNOWN";
	}
	}

	const char* SharingModeToString(aaudio_sharing_mode_t mode) {
	switch (mode) {
	case AAUDIO_SHARING_MODE_EXCLUSIVE:
	return "EXCLUSIVE";
	case AAUDIO_SHARING_MODE_SHARED:
	return "SHARED";
	default:
	return "UNKNOWN";
	}
	}

	const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
	switch (mode) {
	case AAUDIO_PERFORMANCE_MODE_NONE:
	return "NONE";
	case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
	return "POWER_SAVING";
	case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
	return "LOW_LATENCY";
	default:
	return "UNKNOWN";
	}
	}

	const char* FormatToString(int32_t id) {
	switch (id) {
	case AAUDIO_FORMAT_INVALID:
	return "INVALID";
	case AAUDIO_FORMAT_UNSPECIFIED:
	return "UNSPECIFIED";
	case AAUDIO_FORMAT_PCM_I16:
	return "PCM_I16";
	case AAUDIO_FORMAT_PCM_FLOAT:
	return "FLOAT";
	default:
	return "UNKNOWN";
	}
	}

	void ErrorCallback(AAudioStream* stream,
	void* user_data,
	aaudio_result_t error) {
	RTC_DCHECK(user_data);
	AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
	RTC_LOG(WARNING) << "ErrorCallback: "
	<< DirectionToString(aaudio_wrapper->direction());
	RTC_DCHECK(aaudio_wrapper->observer());
	aaudio_wrapper->observer()->OnErrorCallback(error);
	}

	aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
	void* user_data,
	void* audio_data,
	int32_t num_frames) {
	RTC_DCHECK(user_data);
	RTC_DCHECK(audio_data);
	AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
	RTC_DCHECK(aaudio_wrapper->observer());
	return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
	}

	// Wraps the stream builder object to ensure that it is released properly when
	// the stream builder goes out of scope.
	class ScopedStreamBuilder {
	public:
	ScopedStreamBuilder() {
	LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
	RTC_DCHECK(builder_);
	}
	~ScopedStreamBuilder() {
	if (builder_) {
	LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
	}
	}

	AAudioStreamBuilder* get() const { return builder_; }

	private:
	AAudioStreamBuilder* builder_ = nullptr;
	};

	} // namespace

	AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
	aaudio_direction_t direction,
	AAudioObserverInterface* observer)
	: direction_(direction), observer_(observer) {
	RTC_LOG(INFO) << "ctor";
	RTC_DCHECK(observer_);
	direction_ == AAUDIO_DIRECTION_OUTPUT
	? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
	: audio_parameters_ = audio_manager->GetRecordAudioParameters();
	aaudio_thread_checker_.Detach();
	RTC_LOG(INFO) << audio_parameters_.ToString();
	}

	AAudioWrapper::~AAudioWrapper() {
	RTC_LOG(INFO) << "dtor";
	RTC_DCHECK(thread_checker_.IsCurrent());
	RTC_DCHECK(!stream_);
	}

	bool AAudioWrapper::Init() {
	RTC_LOG(INFO) << "Init";
	RTC_DCHECK(thread_checker_.IsCurrent());
	// Creates a stream builder which can be used to open an audio stream.
	ScopedStreamBuilder builder;
	// Configures the stream builder using audio parameters given at construction.
	SetStreamConfiguration(builder.get());
	// Opens a stream based on options in the stream builder.
	if (!OpenStream(builder.get())) {
	return false;
	}
	// Ensures that the opened stream could activate the requested settings.
	if (!VerifyStreamConfiguration()) {
	return false;
	}
	// Optimizes the buffer scheme for lowest possible latency and creates
	// additional buffer logic to match the 10ms buffer size used in WebRTC.
	if (!OptimizeBuffers()) {
	return false;
	}
	LogStreamState();
	return true;
	}

	bool AAudioWrapper::Start() {
	RTC_LOG(INFO) << "Start";
	RTC_DCHECK(thread_checker_.IsCurrent());
	// TODO(henrika): this state check might not be needed.
	aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
	if (current_state != AAUDIO_STREAM_STATE_OPEN) {
	RTC_LOG(LS_ERROR) << "Invalid state: "
	<< AAudio_convertStreamStateToText(current_state);
	return false;
	}
	// Asynchronous request for the stream to start.
	RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
	LogStreamState();
	return true;
	}

	bool AAudioWrapper::Stop() {
	RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
	RTC_DCHECK(thread_checker_.IsCurrent());
	// Asynchronous request for the stream to stop.
	RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
	CloseStream();
	aaudio_thread_checker_.Detach();
	return true;
	}

	double AAudioWrapper::EstimateLatencyMillis() const {
	RTC_DCHECK(stream_);
	double latency_millis = 0.0;
	if (direction() == AAUDIO_DIRECTION_INPUT) {
	// For input streams. Best guess we can do is to use the current burst size
	// as delay estimate.
	latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
	rtc::kNumMillisecsPerSec;
	} else {
	int64_t existing_frame_index;
	int64_t existing_frame_presentation_time;
	// Get the time at which a particular frame was presented to audio hardware.
	aaudio_result_t result = AAudioStream_getTimestamp(
	stream_, CLOCK_MONOTONIC, &existing_frame_index,
	&existing_frame_presentation_time);
	// Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
	if (result == AAUDIO_OK) {
	// Get write index for next audio frame.
	int64_t next_frame_index = frames_written();
	// Number of frames between next frame and the existing frame.
	int64_t frame_index_delta = next_frame_index - existing_frame_index;
	// Assume the next frame will be written now.
	int64_t next_frame_write_time = rtc::TimeNanos();
	// Calculate time when next frame will be presented to the hardware taking
	// sample rate into account.
	int64_t frame_time_delta =
	(frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
	int64_t next_frame_presentation_time =
	existing_frame_presentation_time + frame_time_delta;
	// Derive a latency estimate given results above.
	latency_millis = static_cast<double>(next_frame_presentation_time -
	next_frame_write_time) /
	rtc::kNumNanosecsPerMillisec;
	}
	}
	return latency_millis;
	}

	// Returns new buffer size or a negative error value if buffer size could not
	// be increased.
	bool AAudioWrapper::IncreaseOutputBufferSize() {
	RTC_LOG(INFO) << "IncreaseBufferSize";
	RTC_DCHECK(stream_);
	RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
	RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
	aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
	// Try to increase size of buffer with one burst to reduce risk of underrun.
	buffer_size += frames_per_burst();
	// Verify that the new buffer size is not larger than max capacity.
	// TODO(henrika): keep track of case when we reach the capacity limit.
	const int32_t max_buffer_size = buffer_capacity_in_frames();
	if (buffer_size > max_buffer_size) {
	RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
	<< ") is higher than max: " << max_buffer_size;
	return false;
	}
	RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
	<< " (max=" << max_buffer_size << ")";
	buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
	if (buffer_size < 0) {
	RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
	<< AAudio_convertResultToText(buffer_size);
	return false;
	}
	RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
	return true;
	}

	void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
	RTC_LOG(INFO) << "ClearInputStream";
	RTC_DCHECK(stream_);
	RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
	RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
	aaudio_result_t cleared_frames = 0;
	do {
	cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
	} while (cleared_frames > 0);
	}

	AAudioObserverInterface* AAudioWrapper::observer() const {
	return observer_;
	}

	AudioParameters AAudioWrapper::audio_parameters() const {
	return audio_parameters_;
	}

	int32_t AAudioWrapper::samples_per_frame() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getSamplesPerFrame(stream_);
	}

	int32_t AAudioWrapper::buffer_size_in_frames() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getBufferSizeInFrames(stream_);
	}

	int32_t AAudioWrapper::buffer_capacity_in_frames() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getBufferCapacityInFrames(stream_);
	}

	int32_t AAudioWrapper::device_id() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getDeviceId(stream_);
	}

	int32_t AAudioWrapper::xrun_count() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getXRunCount(stream_);
	}

	int32_t AAudioWrapper::format() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getFormat(stream_);
	}

	int32_t AAudioWrapper::sample_rate() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getSampleRate(stream_);
	}

	int32_t AAudioWrapper::channel_count() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getChannelCount(stream_);
	}

	int32_t AAudioWrapper::frames_per_callback() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getFramesPerDataCallback(stream_);
	}

	aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getSharingMode(stream_);
	}

	aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getPerformanceMode(stream_);
	}

	aaudio_stream_state_t AAudioWrapper::stream_state() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getState(stream_);
	}

	int64_t AAudioWrapper::frames_written() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getFramesWritten(stream_);
	}

	int64_t AAudioWrapper::frames_read() const {
	RTC_DCHECK(stream_);
	return AAudioStream_getFramesRead(stream_);
	}

	void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
	RTC_LOG(INFO) << "SetStreamConfiguration";
	RTC_DCHECK(builder);
	RTC_DCHECK(thread_checker_.IsCurrent());
	// Request usage of default primary output/input device.
	// TODO(henrika): verify that default device follows Java APIs.
	// https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
	AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
	// Use preferred sample rate given by the audio parameters.
	AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
	// Use preferred channel configuration given by the audio parameters.
	AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
	// Always use 16-bit PCM audio sample format.
	AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
	// TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
	// Ask for exclusive mode since this will give us the lowest possible latency.
	// If exclusive mode isn't available, shared mode will be used instead.
	AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
	// Use the direction that was given at construction.
	AAudioStreamBuilder_setDirection(builder, direction_);
	// TODO(henrika): investigate performance using different performance modes.
	AAudioStreamBuilder_setPerformanceMode(builder,
	AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
	// Given that WebRTC applications require low latency, our audio stream uses
	// an asynchronous callback function to transfer data to and from the
	// application. AAudio executes the callback in a higher-priority thread that
	// has better performance.
	AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
	// Request that AAudio calls this functions if any error occurs on a callback
	// thread.
	AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
	}

	bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
	RTC_LOG(INFO) << "OpenStream";
	RTC_DCHECK(builder);
	AAudioStream* stream = nullptr;
	RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
	stream_ = stream;
	LogStreamConfiguration();
	return true;
	}

	void AAudioWrapper::CloseStream() {
	RTC_LOG(INFO) << "CloseStream";
	RTC_DCHECK(stream_);
	LOG_ON_ERROR(AAudioStream_close(stream_));
	stream_ = nullptr;
	}

	void AAudioWrapper::LogStreamConfiguration() {
	RTC_DCHECK(stream_);
	char ss_buf[1024];
	rtc::SimpleStringBuilder ss(ss_buf);
	ss << "Stream Configuration: ";
	ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
	ss << ", samples per frame=" << samples_per_frame();
	ss << ", format=" << FormatToString(format());
	ss << ", sharing mode=" << SharingModeToString(sharing_mode());
	ss << ", performance mode=" << PerformanceModeToString(performance_mode());
	ss << ", direction=" << DirectionToString(direction());
	ss << ", device id=" << AAudioStream_getDeviceId(stream_);
	ss << ", frames per callback=" << frames_per_callback();
	RTC_LOG(INFO) << ss.str();
	}

	void AAudioWrapper::LogStreamState() {
	RTC_LOG(INFO) << "AAudio stream state: "
	<< AAudio_convertStreamStateToText(stream_state());
	}

	bool AAudioWrapper::VerifyStreamConfiguration() {
	RTC_LOG(INFO) << "VerifyStreamConfiguration";
	RTC_DCHECK(stream_);
	// TODO(henrika): should we verify device ID as well?
	if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
	RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
	return false;
	}
	if (AAudioStream_getChannelCount(stream_) !=
	static_cast<int32_t>(audio_parameters().channels())) {
	RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
	return false;
	}
	if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
	RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
	return false;
	}
	if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
	RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
	return false;
	}
	if (AAudioStream_getPerformanceMode(stream_) !=
	AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
	RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
	return false;
	}
	if (AAudioStream_getDirection(stream_) != direction()) {
	RTC_LOG(LS_ERROR) << "Stream direction could not be set";
	return false;
	}
	if (AAudioStream_getSamplesPerFrame(stream_) !=
	static_cast<int32_t>(audio_parameters().channels())) {
	RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
	return false;
	}
	return true;
	}

	bool AAudioWrapper::OptimizeBuffers() {
	RTC_LOG(INFO) << "OptimizeBuffers";
	RTC_DCHECK(stream_);
	// Maximum number of frames that can be filled without blocking.
	RTC_LOG(INFO) << "max buffer capacity in frames: "
	<< buffer_capacity_in_frames();
	// Query the number of frames that the application should read or write at
	// one time for optimal performance.
	int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
	RTC_LOG(INFO) << "frames per burst for optimal performance: "
	<< frames_per_burst;
	frames_per_burst_ = frames_per_burst;
	if (direction() == AAUDIO_DIRECTION_INPUT) {
	// There is no point in calling setBufferSizeInFrames() for input streams
	// since it has no effect on the performance (latency in this case).
	return true;
	}
	// Set buffer size to same as burst size to guarantee lowest possible latency.
	// This size might change for output streams if underruns are detected and
	// automatic buffer adjustment is enabled.
	AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
	int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
	if (buffer_size != frames_per_burst) {
	RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
	return false;
	}
	// Maximum number of frames that can be filled without blocking.
	RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
	return true;
	}

	} // namespace webrtc