src/starboard/shared/alsa/alsa_audio_sink_type.cc - cobalt - Git at Google

 // Copyright 2016 The Cobalt Authors. All Rights Reserved.
 //
 // 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.

 #include "starboard/shared/alsa/alsa_audio_sink_type.h"

 #include <alsa/asoundlib.h>

 #include <algorithm>
 #include <vector>

 #include "starboard/audio_sink.h"
 #include "starboard/common/condition_variable.h"
 #include "starboard/common/log.h"
 #include "starboard/common/mutex.h"
 #include "starboard/configuration.h"
 #include "starboard/memory.h"
 #include "starboard/shared/alsa/alsa_util.h"
 #include "starboard/thread.h"
 #include "starboard/time.h"

 namespace starboard {
 namespace shared {
 namespace alsa {
 namespace {

 using starboard::ScopedLock;
 using starboard::ScopedTryLock;
 using starboard::shared::alsa::AlsaGetBufferedFrames;
 using starboard::shared::alsa::AlsaWriteFrames;

 // The maximum number of frames that can be written to ALSA once.  It must be a
 // power of 2.  It is also used as the ALSA polling size.  A small number will
 // lead to more CPU being used as the callbacks will be called more
 // frequently and also it will be more likely to cause underflow but can make
 // the audio clock more accurate.
 const int kFramesPerRequest = 512;
 // When the frames inside ALSA buffer is less than |kMinimumFramesInALSA|, the
 // class will try to write more frames.  The larger the number is, the less
 // likely an underflow will happen but to use a larger number will cause longer
 // delays after pause and stop.
 const int kMinimumFramesInALSA = 2048;
 // The size of the audio buffer ALSA allocates internally.  Ideally this value
 // should be greater than the sum of the above two constants.  Choose a value
 // that is too large can waste some memory as the extra buffer is never used.
 const int kALSABufferSizeInFrames = 8192;

 // Helper function to compute the size of the two valid starboard audio sample
 // types.
 size_t GetSampleSize(SbMediaAudioSampleType sample_type) {
   switch (sample_type) {
     case kSbMediaAudioSampleTypeFloat32:
       return sizeof(float);
     case kSbMediaAudioSampleTypeInt16Deprecated:
       return sizeof(int16_t);
   }
   SB_NOTREACHED();
   return 0u;
 }

 void* IncrementPointerByBytes(void* pointer, size_t offset) {
   return static_cast<void*>(static_cast<uint8_t*>(pointer) + offset);
 }

 // This class is an ALSA based audio sink with the following features:
 // 1. It doesn't cache any data internally and maintains minimum data inside
 //    the ALSA buffer.  It relies on pulling data from its source in high
 //    frequency to playback audio.
 // 2. It never stops the underlying ALSA audio sink once created.  When its
 //    source cannot provide enough data to continue playback, it simply writes
 //    silence to ALSA.
 class AlsaAudioSink : public SbAudioSinkPrivate {
  public:
   AlsaAudioSink(Type* type,
                 int channels,
                 int sampling_frequency_hz,
                 SbMediaAudioSampleType sample_type,
                 SbAudioSinkFrameBuffers frame_buffers,
                 int frames_per_channel,
                 SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
                 ConsumeFramesFunc consume_frames_func,
                 void* context);
   ~AlsaAudioSink() override;

   bool IsType(Type* type) override { return type_ == type; }

   void SetPlaybackRate(double playback_rate) override {
     ScopedLock lock(mutex_);
     playback_rate_ = playback_rate;
   }

   void SetVolume(double volume) override {
     ScopedLock lock(mutex_);
     volume_ = volume;
   }

   bool is_valid() { return playback_handle_ != NULL; }

  private:
   AlsaAudioSink(const AlsaAudioSink&) = delete;
   AlsaAudioSink& operator=(const AlsaAudioSink&) = delete;

   static void* ThreadEntryPoint(void* context);
   void AudioThreadFunc();
   // Write silence to ALSA when there is not enough data in source or when the
   // sink is paused.
   // Return true to continue to play.  Return false when destroying.
   bool IdleLoop();
   // Keep pulling frames from source until there is no frames to keep playback.
   // When the sink is paused or there is no frames in source, it returns true
   // so we can continue into the IdleLoop().  It returns false when destroying.
   bool PlaybackLoop();
   // Helper function to write frames contained in a ring buffer to ALSA.
   void WriteFrames(double playback_rate,
                    int frames_to_write,
                    int frames_in_buffer,
                    int offset_in_frames);

   Type* type_;
   SbAudioSinkUpdateSourceStatusFunc update_source_status_func_;
   ConsumeFramesFunc consume_frames_func_;
   void* context_;

   double playback_rate_;
   double volume_;
   std::vector<uint8_t> resample_buffer_;

   int channels_;
   int sampling_frequency_hz_;
   SbMediaAudioSampleType sample_type_;

   SbThread audio_out_thread_;
   starboard::Mutex mutex_;
   starboard::ConditionVariable creation_signal_;

   SbTime time_to_wait_;

   bool destroying_;

   void* frame_buffer_;
   int frames_per_channel_;
   void* silence_frames_;

   void* playback_handle_;
 };

 AlsaAudioSink::AlsaAudioSink(
     Type* type,
     int channels,
     int sampling_frequency_hz,
     SbMediaAudioSampleType sample_type,
     SbAudioSinkFrameBuffers frame_buffers,
     int frames_per_channel,
     SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
     ConsumeFramesFunc consume_frames_func,
     void* context)
     : type_(type),
       playback_rate_(1.0),
       volume_(1.0),
       resample_buffer_(channels * kFramesPerRequest *
                        GetSampleSize(sample_type)),
       channels_(channels),
       sampling_frequency_hz_(sampling_frequency_hz),
       sample_type_(sample_type),
       update_source_status_func_(update_source_status_func),
       consume_frames_func_(consume_frames_func),
       context_(context),
       audio_out_thread_(kSbThreadInvalid),
       creation_signal_(mutex_),
       time_to_wait_(kFramesPerRequest * kSbTimeSecond / sampling_frequency_hz /
                     2),
       destroying_(false),
       frame_buffer_(frame_buffers[0]),
       frames_per_channel_(frames_per_channel),
       silence_frames_(new uint8_t[channels * kFramesPerRequest *
                                   GetSampleSize(sample_type)]),
       playback_handle_(NULL) {
   SB_DCHECK(update_source_status_func_);
   SB_DCHECK(consume_frames_func_);
   SB_DCHECK(frame_buffer_);
   SB_DCHECK(SbAudioSinkIsAudioSampleTypeSupported(sample_type_));

   memset(silence_frames_, 0,
               channels * kFramesPerRequest * GetSampleSize(sample_type));

   ScopedLock lock(mutex_);
   audio_out_thread_ =
       SbThreadCreate(0, kSbThreadPriorityRealTime, kSbThreadNoAffinity, true,
                      "alsa_audio_out", &AlsaAudioSink::ThreadEntryPoint, this);
   SB_DCHECK(SbThreadIsValid(audio_out_thread_));
   creation_signal_.Wait();
 }

 AlsaAudioSink::~AlsaAudioSink() {
   {
     ScopedLock lock(mutex_);
     destroying_ = true;
   }
   SbThreadJoin(audio_out_thread_, NULL);

   delete[] static_cast<uint8_t*>(silence_frames_);
 }

 // static
 void* AlsaAudioSink::ThreadEntryPoint(void* context) {
   SB_DCHECK(context);
   AlsaAudioSink* sink = reinterpret_cast<AlsaAudioSink*>(context);
   sink->AudioThreadFunc();

   return NULL;
 }

 void AlsaAudioSink::AudioThreadFunc() {
   snd_pcm_format_t alsa_sample_type =
       sample_type_ == kSbMediaAudioSampleTypeFloat32 ? SND_PCM_FORMAT_FLOAT_LE
                                                      : SND_PCM_FORMAT_S16;

   playback_handle_ = starboard::shared::alsa::AlsaOpenPlaybackDevice(
       channels_, sampling_frequency_hz_, kFramesPerRequest,
       kALSABufferSizeInFrames, alsa_sample_type);
   {
     ScopedLock lock(mutex_);
     creation_signal_.Signal();
   }

   if (!playback_handle_) {
     return;
   }

   for (;;) {
     if (!IdleLoop()) {
       break;
     }
     if (!PlaybackLoop()) {
       break;
     }
   }

   starboard::shared::alsa::AlsaCloseDevice(playback_handle_);
   ScopedLock lock(mutex_);
   playback_handle_ = NULL;
 }

 bool AlsaAudioSink::IdleLoop() {
   SB_DLOG(INFO) << "alsa::AlsaAudioSink enters idle loop";

   bool drain = true;

   for (;;) {
     double playback_rate;
     {
       ScopedLock lock(mutex_);
       if (destroying_) {
         SB_DLOG(INFO) << "alsa::AlsaAudioSink exits idle loop : destroying";
         break;
       }
       playback_rate = playback_rate_;
     }
     int frames_in_buffer, offset_in_frames;
     bool is_playing, is_eos_reached;
     update_source_status_func_(&frames_in_buffer, &offset_in_frames,
                                &is_playing, &is_eos_reached, context_);
     if (is_playing && frames_in_buffer > 0 && playback_rate > 0.0) {
       SB_DLOG(INFO) << "alsa::AlsaAudioSink exits idle loop : is playing "
                     << is_playing << " frames in buffer " << frames_in_buffer
                     << " playback_rate " << playback_rate;
       return true;
     }
     if (drain) {
       drain = false;
       AlsaWriteFrames(playback_handle_, silence_frames_, kFramesPerRequest);
       AlsaDrain(playback_handle_);
     }
     SbThreadSleep(time_to_wait_);
   }

   return false;
 }

 bool AlsaAudioSink::PlaybackLoop() {
   SB_DLOG(INFO) << "alsa::AlsaAudioSink enters playback loop";

   // TODO: Also handle |volume_| here.
   double playback_rate = 1.0;
   for (;;) {
     int delayed_frame = AlsaGetBufferedFrames(playback_handle_);
     {
       ScopedTryLock lock(mutex_);
       if (lock.is_locked()) {
         if (destroying_) {
           SB_DLOG(INFO)
               << "alsa::AlsaAudioSink exits playback loop : destroying";
           break;
         }
         playback_rate = playback_rate_;
       }
     }

     if (delayed_frame < kMinimumFramesInALSA) {
       if (playback_rate == 0.0) {
         SB_DLOG(INFO)
             << "alsa::AlsaAudioSink exits playback loop: playback rate 0";
         return true;
       }
       int frames_in_buffer, offset_in_frames;
       bool is_playing, is_eos_reached;
       update_source_status_func_(&frames_in_buffer, &offset_in_frames,
                                  &is_playing, &is_eos_reached, context_);
       if (!is_playing || frames_in_buffer == 0) {
         SB_DLOG(INFO) << "alsa::AlsaAudioSink exits playback loop: is playing "
                       << is_playing << " frames in buffer " << frames_in_buffer;
         return true;
       }
       WriteFrames(playback_rate, std::min(kFramesPerRequest, frames_in_buffer),
                   frames_in_buffer, offset_in_frames);
     } else {
       SbThreadSleep(time_to_wait_);
     }
   }

   return false;
 }

 void AlsaAudioSink::WriteFrames(double playback_rate,
                                 int frames_to_write,
                                 int frames_in_buffer,
                                 int offset_in_frames) {
   const int bytes_per_frame = channels_ * GetSampleSize(sample_type_);
   if (playback_rate == 1.0) {
     SB_DCHECK(frames_to_write <= frames_in_buffer);

     int frames_to_buffer_end = frames_per_channel_ - offset_in_frames;
     if (frames_to_write > frames_to_buffer_end) {
       int consumed = AlsaWriteFrames(
           playback_handle_,
           IncrementPointerByBytes(frame_buffer_,
                                   offset_in_frames * bytes_per_frame),
           frames_to_buffer_end);
       consume_frames_func_(consumed, SbTimeGetMonotonicNow(), context_);
       if (consumed != frames_to_buffer_end) {
         SB_DLOG(INFO) << "alsa::AlsaAudioSink exits write frames : consumed "
                       << consumed << " frames, with " << frames_to_buffer_end
                       << " frames to buffer end";
         return;
       }

       frames_to_write -= frames_to_buffer_end;
       offset_in_frames = 0;
     }

     int consumed =
         AlsaWriteFrames(playback_handle_,
                         IncrementPointerByBytes(
                             frame_buffer_, offset_in_frames * bytes_per_frame),
                         frames_to_write);
     consume_frames_func_(consumed, SbTimeGetMonotonicNow(), context_);
   } else {
     // A very low quality resampler that simply shift the audio frames to play
     // at the right time.
     // TODO: The playback rate adjustment should be done in AudioRenderer.  We
     // should provide a default sinc resampler.
     double source_frames = 0.0;
     int buffer_size_in_frames = resample_buffer_.size() / bytes_per_frame;
     int target_frames = 0;
     SB_DCHECK(buffer_size_in_frames <= frames_to_write);

     // Use |playback_rate| as the granularity of increment for source buffer.
     // For example, when |playback_rate| is 0.25, every time a frame is copied
     // to the target buffer, the offset of source buffer will be increased by
     // 0.25, this effectively repeat the same frame four times into the target
     // buffer and it takes 4 times longer to finish playing the frames.
     while (static_cast<int>(source_frames) < frames_in_buffer &&
            target_frames < buffer_size_in_frames) {
       const uint8_t* source_addr = static_cast<uint8_t*>(frame_buffer_);
       source_addr += static_cast<int>(offset_in_frames + source_frames) %
                      frames_per_channel_ * bytes_per_frame;
       memcpy(&resample_buffer_[0] + bytes_per_frame * target_frames,
                    source_addr, bytes_per_frame);
       ++target_frames;
       source_frames += playback_rate;
     }

     int consumed =
         AlsaWriteFrames(playback_handle_, &resample_buffer_[0], target_frames);
     consume_frames_func_(consumed * playback_rate_, SbTimeGetMonotonicNow(),
                          context_);
   }
 }

 class AlsaAudioSinkType : public SbAudioSinkPrivate::Type {
  public:
   SbAudioSink Create(
       int channels,
       int sampling_frequency_hz,
       SbMediaAudioSampleType audio_sample_type,
       SbMediaAudioFrameStorageType audio_frame_storage_type,
       SbAudioSinkFrameBuffers frame_buffers,
       int frames_per_channel,
       SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
       SbAudioSinkPrivate::ConsumeFramesFunc consume_frames_func,
 #if SB_API_VERSION >= 12
       SbAudioSinkPrivate::ErrorFunc error_func,
 #endif  // SB_API_VERSION >= 12
       void* context) override;

   bool IsValid(SbAudioSink audio_sink) override {
     return audio_sink != kSbAudioSinkInvalid && audio_sink->IsType(this);
   }

   void Destroy(SbAudioSink audio_sink) override {
     if (audio_sink != kSbAudioSinkInvalid && !IsValid(audio_sink)) {
       SB_LOG(WARNING) << "audio_sink is invalid.";
       return;
     }
     delete audio_sink;
   }
 };

 SbAudioSink AlsaAudioSinkType::Create(
     int channels,
     int sampling_frequency_hz,
     SbMediaAudioSampleType audio_sample_type,
     SbMediaAudioFrameStorageType audio_frame_storage_type,
     SbAudioSinkFrameBuffers frame_buffers,
     int frames_per_channel,
     SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
     SbAudioSinkPrivate::ConsumeFramesFunc consume_frames_func,
 #if SB_API_VERSION >= 12
     SbAudioSinkPrivate::ErrorFunc error_func,
 #endif  // SB_API_VERSION >= 12
     void* context) {
   AlsaAudioSink* audio_sink = new AlsaAudioSink(
       this, channels, sampling_frequency_hz, audio_sample_type, frame_buffers,
       frames_per_channel, update_source_status_func, consume_frames_func,
       context);
   if (!audio_sink->is_valid()) {
     delete audio_sink;
     return kSbAudioSinkInvalid;
   }

   return audio_sink;
 }

 }  // namespace

 namespace {
 AlsaAudioSinkType* alsa_audio_sink_type_;
 }  // namespace

 // static
 void PlatformInitialize() {
   SB_DCHECK(!alsa_audio_sink_type_);
   alsa_audio_sink_type_ = new AlsaAudioSinkType();
   SbAudioSinkPrivate::SetPrimaryType(alsa_audio_sink_type_);
 }

 // static
 void PlatformTearDown() {
   SB_DCHECK(alsa_audio_sink_type_);
   SB_DCHECK(alsa_audio_sink_type_ == SbAudioSinkPrivate::GetPrimaryType());

   SbAudioSinkPrivate::SetPrimaryType(NULL);
   delete alsa_audio_sink_type_;
   alsa_audio_sink_type_ = NULL;
 }

 }  // namespace alsa
 }  // namespace shared
 }  // namespace starboard
	// Copyright 2016 The Cobalt Authors. All Rights Reserved.
	//
	// 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.

	#include "starboard/shared/alsa/alsa_audio_sink_type.h"

	#include <alsa/asoundlib.h>

	#include <algorithm>
	#include <vector>

	#include "starboard/audio_sink.h"
	#include "starboard/common/condition_variable.h"
	#include "starboard/common/log.h"
	#include "starboard/common/mutex.h"
	#include "starboard/configuration.h"
	#include "starboard/memory.h"
	#include "starboard/shared/alsa/alsa_util.h"
	#include "starboard/thread.h"
	#include "starboard/time.h"

	namespace starboard {
	namespace shared {
	namespace alsa {
	namespace {

	using starboard::ScopedLock;
	using starboard::ScopedTryLock;
	using starboard::shared::alsa::AlsaGetBufferedFrames;
	using starboard::shared::alsa::AlsaWriteFrames;

	// The maximum number of frames that can be written to ALSA once. It must be a
	// power of 2. It is also used as the ALSA polling size. A small number will
	// lead to more CPU being used as the callbacks will be called more
	// frequently and also it will be more likely to cause underflow but can make
	// the audio clock more accurate.
	const int kFramesPerRequest = 512;
	// When the frames inside ALSA buffer is less than \|kMinimumFramesInALSA\|, the
	// class will try to write more frames. The larger the number is, the less
	// likely an underflow will happen but to use a larger number will cause longer
	// delays after pause and stop.
	const int kMinimumFramesInALSA = 2048;
	// The size of the audio buffer ALSA allocates internally. Ideally this value
	// should be greater than the sum of the above two constants. Choose a value
	// that is too large can waste some memory as the extra buffer is never used.
	const int kALSABufferSizeInFrames = 8192;

	// Helper function to compute the size of the two valid starboard audio sample
	// types.
	size_t GetSampleSize(SbMediaAudioSampleType sample_type) {
	switch (sample_type) {
	case kSbMediaAudioSampleTypeFloat32:
	return sizeof(float);
	case kSbMediaAudioSampleTypeInt16Deprecated:
	return sizeof(int16_t);
	}
	SB_NOTREACHED();
	return 0u;
	}

	void* IncrementPointerByBytes(void* pointer, size_t offset) {
	return static_cast<void>(static_cast<uint8_t>(pointer) + offset);
	}

	// This class is an ALSA based audio sink with the following features:
	// 1. It doesn't cache any data internally and maintains minimum data inside
	// the ALSA buffer. It relies on pulling data from its source in high
	// frequency to playback audio.
	// 2. It never stops the underlying ALSA audio sink once created. When its
	// source cannot provide enough data to continue playback, it simply writes
	// silence to ALSA.
	class AlsaAudioSink : public SbAudioSinkPrivate {
	public:
	AlsaAudioSink(Type* type,
	int channels,
	int sampling_frequency_hz,
	SbMediaAudioSampleType sample_type,
	SbAudioSinkFrameBuffers frame_buffers,
	int frames_per_channel,
	SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
	ConsumeFramesFunc consume_frames_func,
	void* context);
	~AlsaAudioSink() override;

	bool IsType(Type* type) override { return type_ == type; }

	void SetPlaybackRate(double playback_rate) override {
	ScopedLock lock(mutex_);
	playback_rate_ = playback_rate;
	}

	void SetVolume(double volume) override {
	ScopedLock lock(mutex_);
	volume_ = volume;
	}

	bool is_valid() { return playback_handle_ != NULL; }

	private:
	AlsaAudioSink(const AlsaAudioSink&) = delete;
	AlsaAudioSink& operator=(const AlsaAudioSink&) = delete;

	static void* ThreadEntryPoint(void* context);
	void AudioThreadFunc();
	// Write silence to ALSA when there is not enough data in source or when the
	// sink is paused.
	// Return true to continue to play. Return false when destroying.
	bool IdleLoop();
	// Keep pulling frames from source until there is no frames to keep playback.
	// When the sink is paused or there is no frames in source, it returns true
	// so we can continue into the IdleLoop(). It returns false when destroying.
	bool PlaybackLoop();
	// Helper function to write frames contained in a ring buffer to ALSA.
	void WriteFrames(double playback_rate,
	int frames_to_write,
	int frames_in_buffer,
	int offset_in_frames);

	Type* type_;
	SbAudioSinkUpdateSourceStatusFunc update_source_status_func_;
	ConsumeFramesFunc consume_frames_func_;
	void* context_;

	double playback_rate_;
	double volume_;
	std::vector<uint8_t> resample_buffer_;

	int channels_;
	int sampling_frequency_hz_;
	SbMediaAudioSampleType sample_type_;

	SbThread audio_out_thread_;
	starboard::Mutex mutex_;
	starboard::ConditionVariable creation_signal_;

	SbTime time_to_wait_;

	bool destroying_;

	void* frame_buffer_;
	int frames_per_channel_;
	void* silence_frames_;

	void* playback_handle_;
	};

	AlsaAudioSink::AlsaAudioSink(
	Type* type,
	int channels,
	int sampling_frequency_hz,
	SbMediaAudioSampleType sample_type,
	SbAudioSinkFrameBuffers frame_buffers,
	int frames_per_channel,
	SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
	ConsumeFramesFunc consume_frames_func,
	void* context)
	: type_(type),
	playback_rate_(1.0),
	volume_(1.0),
	resample_buffer_(channels * kFramesPerRequest *
	GetSampleSize(sample_type)),
	channels_(channels),
	sampling_frequency_hz_(sampling_frequency_hz),
	sample_type_(sample_type),
	update_source_status_func_(update_source_status_func),
	consume_frames_func_(consume_frames_func),
	context_(context),
	audio_out_thread_(kSbThreadInvalid),
	creation_signal_(mutex_),
	time_to_wait_(kFramesPerRequest * kSbTimeSecond / sampling_frequency_hz /
	2),
	destroying_(false),
	frame_buffer_(frame_buffers[0]),
	frames_per_channel_(frames_per_channel),
	silence_frames_(new uint8_t[channels * kFramesPerRequest *
	GetSampleSize(sample_type)]),
	playback_handle_(NULL) {
	SB_DCHECK(update_source_status_func_);
	SB_DCHECK(consume_frames_func_);
	SB_DCHECK(frame_buffer_);
	SB_DCHECK(SbAudioSinkIsAudioSampleTypeSupported(sample_type_));

	memset(silence_frames_, 0,
	channels * kFramesPerRequest * GetSampleSize(sample_type));

	ScopedLock lock(mutex_);
	audio_out_thread_ =
	SbThreadCreate(0, kSbThreadPriorityRealTime, kSbThreadNoAffinity, true,
	"alsa_audio_out", &AlsaAudioSink::ThreadEntryPoint, this);
	SB_DCHECK(SbThreadIsValid(audio_out_thread_));
	creation_signal_.Wait();
	}

	AlsaAudioSink::~AlsaAudioSink() {
	{
	ScopedLock lock(mutex_);
	destroying_ = true;
	}
	SbThreadJoin(audio_out_thread_, NULL);

	delete[] static_cast<uint8_t*>(silence_frames_);
	}

	// static
	void* AlsaAudioSink::ThreadEntryPoint(void* context) {
	SB_DCHECK(context);
	AlsaAudioSink* sink = reinterpret_cast<AlsaAudioSink*>(context);
	sink->AudioThreadFunc();

	return NULL;
	}

	void AlsaAudioSink::AudioThreadFunc() {
	snd_pcm_format_t alsa_sample_type =
	sample_type_ == kSbMediaAudioSampleTypeFloat32 ? SND_PCM_FORMAT_FLOAT_LE
	: SND_PCM_FORMAT_S16;

	playback_handle_ = starboard::shared::alsa::AlsaOpenPlaybackDevice(
	channels_, sampling_frequency_hz_, kFramesPerRequest,
	kALSABufferSizeInFrames, alsa_sample_type);
	{
	ScopedLock lock(mutex_);
	creation_signal_.Signal();
	}

	if (!playback_handle_) {
	return;
	}

	for (;;) {
	if (!IdleLoop()) {
	break;
	}
	if (!PlaybackLoop()) {
	break;
	}
	}

	starboard::shared::alsa::AlsaCloseDevice(playback_handle_);
	ScopedLock lock(mutex_);
	playback_handle_ = NULL;
	}

	bool AlsaAudioSink::IdleLoop() {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink enters idle loop";

	bool drain = true;

	for (;;) {
	double playback_rate;
	{
	ScopedLock lock(mutex_);
	if (destroying_) {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink exits idle loop : destroying";
	break;
	}
	playback_rate = playback_rate_;
	}
	int frames_in_buffer, offset_in_frames;
	bool is_playing, is_eos_reached;
	update_source_status_func_(&frames_in_buffer, &offset_in_frames,
	&is_playing, &is_eos_reached, context_);
	if (is_playing && frames_in_buffer > 0 && playback_rate > 0.0) {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink exits idle loop : is playing "
	<< is_playing << " frames in buffer " << frames_in_buffer
	<< " playback_rate " << playback_rate;
	return true;
	}
	if (drain) {
	drain = false;
	AlsaWriteFrames(playback_handle_, silence_frames_, kFramesPerRequest);
	AlsaDrain(playback_handle_);
	}
	SbThreadSleep(time_to_wait_);
	}

	return false;
	}

	bool AlsaAudioSink::PlaybackLoop() {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink enters playback loop";

	// TODO: Also handle \|volume_\| here.
	double playback_rate = 1.0;
	for (;;) {
	int delayed_frame = AlsaGetBufferedFrames(playback_handle_);
	{
	ScopedTryLock lock(mutex_);
	if (lock.is_locked()) {
	if (destroying_) {
	SB_DLOG(INFO)
	<< "alsa::AlsaAudioSink exits playback loop : destroying";
	break;
	}
	playback_rate = playback_rate_;
	}
	}

	if (delayed_frame < kMinimumFramesInALSA) {
	if (playback_rate == 0.0) {
	SB_DLOG(INFO)
	<< "alsa::AlsaAudioSink exits playback loop: playback rate 0";
	return true;
	}
	int frames_in_buffer, offset_in_frames;
	bool is_playing, is_eos_reached;
	update_source_status_func_(&frames_in_buffer, &offset_in_frames,
	&is_playing, &is_eos_reached, context_);
	if (!is_playing \|\| frames_in_buffer == 0) {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink exits playback loop: is playing "
	<< is_playing << " frames in buffer " << frames_in_buffer;
	return true;
	}
	WriteFrames(playback_rate, std::min(kFramesPerRequest, frames_in_buffer),
	frames_in_buffer, offset_in_frames);
	} else {
	SbThreadSleep(time_to_wait_);
	}
	}

	return false;
	}

	void AlsaAudioSink::WriteFrames(double playback_rate,
	int frames_to_write,
	int frames_in_buffer,
	int offset_in_frames) {
	const int bytes_per_frame = channels_ * GetSampleSize(sample_type_);
	if (playback_rate == 1.0) {
	SB_DCHECK(frames_to_write <= frames_in_buffer);

	int frames_to_buffer_end = frames_per_channel_ - offset_in_frames;
	if (frames_to_write > frames_to_buffer_end) {
	int consumed = AlsaWriteFrames(
	playback_handle_,
	IncrementPointerByBytes(frame_buffer_,
	offset_in_frames * bytes_per_frame),
	frames_to_buffer_end);
	consume_frames_func_(consumed, SbTimeGetMonotonicNow(), context_);
	if (consumed != frames_to_buffer_end) {
	SB_DLOG(INFO) << "alsa::AlsaAudioSink exits write frames : consumed "
	<< consumed << " frames, with " << frames_to_buffer_end
	<< " frames to buffer end";
	return;
	}

	frames_to_write -= frames_to_buffer_end;
	offset_in_frames = 0;
	}

	int consumed =
	AlsaWriteFrames(playback_handle_,
	IncrementPointerByBytes(
	frame_buffer_, offset_in_frames * bytes_per_frame),
	frames_to_write);
	consume_frames_func_(consumed, SbTimeGetMonotonicNow(), context_);
	} else {
	// A very low quality resampler that simply shift the audio frames to play
	// at the right time.
	// TODO: The playback rate adjustment should be done in AudioRenderer. We
	// should provide a default sinc resampler.
	double source_frames = 0.0;
	int buffer_size_in_frames = resample_buffer_.size() / bytes_per_frame;
	int target_frames = 0;
	SB_DCHECK(buffer_size_in_frames <= frames_to_write);

	// Use \|playback_rate\| as the granularity of increment for source buffer.
	// For example, when \|playback_rate\| is 0.25, every time a frame is copied
	// to the target buffer, the offset of source buffer will be increased by
	// 0.25, this effectively repeat the same frame four times into the target
	// buffer and it takes 4 times longer to finish playing the frames.
	while (static_cast<int>(source_frames) < frames_in_buffer &&
	target_frames < buffer_size_in_frames) {
	const uint8_t* source_addr = static_cast<uint8_t*>(frame_buffer_);
	source_addr += static_cast<int>(offset_in_frames + source_frames) %
	frames_per_channel_ * bytes_per_frame;
	memcpy(&resample_buffer_[0] + bytes_per_frame * target_frames,
	source_addr, bytes_per_frame);
	++target_frames;
	source_frames += playback_rate;
	}

	int consumed =
	AlsaWriteFrames(playback_handle_, &resample_buffer_[0], target_frames);
	consume_frames_func_(consumed * playback_rate_, SbTimeGetMonotonicNow(),
	context_);
	}
	}

	class AlsaAudioSinkType : public SbAudioSinkPrivate::Type {
	public:
	SbAudioSink Create(
	int channels,
	int sampling_frequency_hz,
	SbMediaAudioSampleType audio_sample_type,
	SbMediaAudioFrameStorageType audio_frame_storage_type,
	SbAudioSinkFrameBuffers frame_buffers,
	int frames_per_channel,
	SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
	SbAudioSinkPrivate::ConsumeFramesFunc consume_frames_func,
	#if SB_API_VERSION >= 12
	SbAudioSinkPrivate::ErrorFunc error_func,
	#endif // SB_API_VERSION >= 12
	void* context) override;

	bool IsValid(SbAudioSink audio_sink) override {
	return audio_sink != kSbAudioSinkInvalid && audio_sink->IsType(this);
	}

	void Destroy(SbAudioSink audio_sink) override {
	if (audio_sink != kSbAudioSinkInvalid && !IsValid(audio_sink)) {
	SB_LOG(WARNING) << "audio_sink is invalid.";
	return;
	}
	delete audio_sink;
	}
	};

	SbAudioSink AlsaAudioSinkType::Create(
	int channels,
	int sampling_frequency_hz,
	SbMediaAudioSampleType audio_sample_type,
	SbMediaAudioFrameStorageType audio_frame_storage_type,
	SbAudioSinkFrameBuffers frame_buffers,
	int frames_per_channel,
	SbAudioSinkUpdateSourceStatusFunc update_source_status_func,
	SbAudioSinkPrivate::ConsumeFramesFunc consume_frames_func,
	#if SB_API_VERSION >= 12
	SbAudioSinkPrivate::ErrorFunc error_func,
	#endif // SB_API_VERSION >= 12
	void* context) {
	AlsaAudioSink* audio_sink = new AlsaAudioSink(
	this, channels, sampling_frequency_hz, audio_sample_type, frame_buffers,
	frames_per_channel, update_source_status_func, consume_frames_func,
	context);
	if (!audio_sink->is_valid()) {
	delete audio_sink;
	return kSbAudioSinkInvalid;
	}

	return audio_sink;
	}

	} // namespace

	namespace {
	AlsaAudioSinkType* alsa_audio_sink_type_;
	} // namespace

	// static
	void PlatformInitialize() {
	SB_DCHECK(!alsa_audio_sink_type_);
	alsa_audio_sink_type_ = new AlsaAudioSinkType();
	SbAudioSinkPrivate::SetPrimaryType(alsa_audio_sink_type_);
	}

	// static
	void PlatformTearDown() {
	SB_DCHECK(alsa_audio_sink_type_);
	SB_DCHECK(alsa_audio_sink_type_ == SbAudioSinkPrivate::GetPrimaryType());

	SbAudioSinkPrivate::SetPrimaryType(NULL);
	delete alsa_audio_sink_type_;
	alsa_audio_sink_type_ = NULL;
	}

	} // namespace alsa
	} // namespace shared
	} // namespace starboard