Simplify AudioRendererImpl by using AudioDevice.

content/renderer/media/audio_renderer_impl.cc

Index: content/renderer/media/audio_renderer_impl.cc
diff --git a/content/renderer/media/audio_renderer_impl.cc b/content/renderer/media/audio_renderer_impl.cc
index 3e617f549dcd2f12f8e3e465ad0a8e1ef5d2941b..768142ad5ee726b8b777cb577252724eebc588ea 100644
--- a/content/renderer/media/audio_renderer_impl.cc
+++ b/content/renderer/media/audio_renderer_impl.cc
@@ -9,61 +9,47 @@
 #include <algorithm>
 
 #include "base/bind.h"
-#include "base/command_line.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
-#include "content/public/common/content_switches.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_buffers_state.h"
-#include "media/audio/audio_output_controller.h"
 #include "media/audio/audio_util.h"
-#include "media/base/filter_host.h"
 
-// Static variable that says what code path we are using -- low or high
-// latency. Made separate variable so we don't have to go to command line
-// for every DCHECK().
-AudioRendererImpl::LatencyType AudioRendererImpl::latency_type_ =
-    AudioRendererImpl::kUninitializedLatency;
+// We define GetBufferSizeForSampleRate() instead of using
+// GetAudioHardwareBufferSize() in audio_util because we're using
+// the AUDIO_PCM_LINEAR flag, instead of AUDIO_PCM_LOW_LATENCY,
+// which the audio_util functions assume.
+//
+// See: http://code.google.com/p/chromium/issues/detail?id=103627
+// for a more detailed description of the subtleties.
+static size_t GetBufferSizeForSampleRate(int sample_rate) {
+  // kNominalBufferSize has been tested on Windows, Mac OS X, and Linux
+  // using the low-latency audio codepath (SyncSocket implementation)
+  // with the AUDIO_PCM_LINEAR flag.
+  const size_t kNominalBufferSize = 2048;
+
+  if (sample_rate <= 48000)
+    return kNominalBufferSize;
+  else if (sample_rate <= 96000)
+    return kNominalBufferSize * 2;
+  return kNominalBufferSize * 4;
+}
 
 AudioRendererImpl::AudioRendererImpl()
     : AudioRendererBase(),
       bytes_per_second_(0),
-      stream_created_(false),
-      stream_id_(0),
-      shared_memory_(NULL),
-      shared_memory_size_(0),
-      stopped_(false),
-      pending_request_(false) {
-  filter_ = RenderThreadImpl::current()->audio_message_filter();
-  // Figure out if we are planning to use high or low latency code path.
-  // We are initializing only one variable and double initialization is Ok,
-  // so there would not be any issues caused by CPU memory model.
-  if (latency_type_ == kUninitializedLatency) {
-    // Urgent workaround for
-    // http://code.google.com/p/chromium-os/issues/detail?id=21491
-    // TODO(enal): Fix it properly.
-#if defined(OS_CHROMEOS)
-    latency_type_ = kHighLatency;
-#else
-    if (!CommandLine::ForCurrentProcess()->HasSwitch(
-        switches::kHighLatencyAudio)) {
-      latency_type_ = kLowLatency;
-    } else {
-      latency_type_ = kHighLatency;
-    }
-#endif
-  }
+      stopped_(false) {
+  // We create the AudioDevice here because it must be created in the
+  // main thread.  But we don't yet know the audio format (sample-rate, etc.)
+  // at this point.  Later, when OnInitialize() is called, we have
+  // the audio format information and call the AudioDevice::Initialize()
+  // method to fully initialize it.
+  audio_device_ = new AudioDevice();
 }
 
 AudioRendererImpl::~AudioRendererImpl() {
 }
 
-// static
-void AudioRendererImpl::set_latency_type(LatencyType latency_type) {
-  DCHECK_EQ(kUninitializedLatency, latency_type_);
-  latency_type_ = latency_type;
-}
-
 base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
   if (bytes_per_second_) {
     return base::TimeDelta::FromMicroseconds(
@@ -92,69 +78,47 @@ void AudioRendererImpl::UpdateEarliestEndTime(int bytes_filled,
 bool AudioRendererImpl::OnInitialize(int bits_per_channel,
                                      ChannelLayout channel_layout,
                                      int sample_rate) {
-  AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR, channel_layout,
-                         sample_rate, bits_per_channel, 0);
-
-  bytes_per_second_ = params.GetBytesPerSecond();
+  // We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
+  // does not currently support all the sample-rates that we require.
+  // Please see: http://code.google.com/p/chromium/issues/detail?id=103627
+  // for more details.
+  audio_parameters_ = AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                                      channel_layout,
+                                      sample_rate,
+                                      bits_per_channel,
+                                      0);
 
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      base::Bind(&AudioRendererImpl::CreateStreamTask, this, params));
-  return true;
-}
+  bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
 
-void AudioRendererImpl::OnStop() {
-  // Since joining with the audio thread can acquire lock_, we make sure to
-  // Join() with it not under lock.
-  base::DelegateSimpleThread* audio_thread = NULL;
-  {
-    base::AutoLock auto_lock(lock_);
-    if (stopped_)
-      return;
-    stopped_ = true;
-
-    DCHECK_EQ(!audio_thread_.get(), !socket_.get());
-    if (socket_.get())
-      socket_->Close();
-    if (audio_thread_.get())
-      audio_thread = audio_thread_.get();
-
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::DestroyTask, this));
-  }
+  DCHECK(audio_device_.get());
 
-  if (audio_thread)
-    audio_thread->Join();
-}
+  if (!audio_device_->IsInitialized()) {
+    audio_device_->Initialize(
+        GetBufferSizeForSampleRate(sample_rate),
+        audio_parameters_.channels,
+        audio_parameters_.sample_rate,
+        audio_parameters_.format,
+        this);
 
-void AudioRendererImpl::NotifyDataAvailableIfNecessary() {
-  if (latency_type_ == kHighLatency) {
-    // Post a task to render thread to notify a packet reception.
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::NotifyPacketReadyTask, this));
+    audio_device_->Start();
   }
+
+  return true;
 }
 
-void AudioRendererImpl::ConsumeAudioSamples(
-    scoped_refptr<media::Buffer> buffer_in) {
-  base::AutoLock auto_lock(lock_);
+void AudioRendererImpl::OnStop() {
   if (stopped_)
     return;
 
-  // TODO(hclam): handle end of stream here.
+  DCHECK(audio_device_.get());
+  audio_device_->Stop();
 
-  // Use the base class to queue the buffer.
-  AudioRendererBase::ConsumeAudioSamples(buffer_in);
-
-  NotifyDataAvailableIfNecessary();
+  stopped_ = true;
 }
 
 void AudioRendererImpl::SetPlaybackRate(float rate) {
   DCHECK_LE(0.0f, rate);
 
-  base::AutoLock auto_lock(lock_);
   // Handle the case where we stopped due to IO message loop dying.
   if (stopped_) {
     AudioRendererBase::SetPlaybackRate(rate);
@@ -165,363 +129,121 @@ void AudioRendererImpl::SetPlaybackRate(float rate) {
   // Play: GetPlaybackRate() == 0.0 && rate != 0.0
   // Pause: GetPlaybackRate() != 0.0 && rate == 0.0
   if (GetPlaybackRate() == 0.0f && rate != 0.0f) {
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::PlayTask, this));
+    DoPlay();
   } else if (GetPlaybackRate() != 0.0f && rate == 0.0f) {
     // Pause is easy, we can always pause.
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::PauseTask, this));
+    DoPause();
   }
   AudioRendererBase::SetPlaybackRate(rate);
-
-  // If we are playing, give a kick to try fulfilling the packet request as
-  // the previous packet request may be stalled by a pause.
-  if (rate > 0.0f) {
-    NotifyDataAvailableIfNecessary();
-  }
 }
 
 void AudioRendererImpl::Pause(const base::Closure& callback) {
   AudioRendererBase::Pause(callback);
-  base::AutoLock auto_lock(lock_);
   if (stopped_)
     return;
 
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      base::Bind(&AudioRendererImpl::PauseTask, this));
+  DoPause();
 }
 
 void AudioRendererImpl::Seek(base::TimeDelta time,
                              const media::FilterStatusCB& cb) {
   AudioRendererBase::Seek(time, cb);
-  base::AutoLock auto_lock(lock_);
   if (stopped_)
     return;
 
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      base::Bind(&AudioRendererImpl::SeekTask, this));
+  DoSeek();
 }
 
-
 void AudioRendererImpl::Play(const base::Closure& callback) {
   AudioRendererBase::Play(callback);
-  base::AutoLock auto_lock(lock_);
   if (stopped_)
     return;
 
   if (GetPlaybackRate() != 0.0f) {
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::PlayTask, this));
+    DoPlay();
   } else {
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioRendererImpl::PauseTask, this));
+    DoPause();
   }
 }
 
 void AudioRendererImpl::SetVolume(float volume) {
-  base::AutoLock auto_lock(lock_);
   if (stopped_)
     return;
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      base::Bind(&AudioRendererImpl::SetVolumeTask, this, volume));
+  DCHECK(audio_device_.get());
+  audio_device_->SetVolume(volume);
 }
 
-void AudioRendererImpl::OnCreated(base::SharedMemoryHandle handle,
-                                  uint32 length) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  shared_memory_.reset(new base::SharedMemory(handle, false));
-  shared_memory_->Map(length);
-  shared_memory_size_ = length;
-}
-
-void AudioRendererImpl::CreateSocket(base::SyncSocket::Handle socket_handle) {
-  DCHECK_EQ(kLowLatency, latency_type_);
-#if defined(OS_WIN)
-  DCHECK(socket_handle);
-#else
-  DCHECK_GE(socket_handle, 0);
-#endif
-  socket_.reset(new base::SyncSocket(socket_handle));
-}
-
-void AudioRendererImpl::CreateAudioThread() {
-  DCHECK_EQ(kLowLatency, latency_type_);
-  audio_thread_.reset(
-      new base::DelegateSimpleThread(this, "renderer_audio_thread"));
-  audio_thread_->Start();
-}
-
-void AudioRendererImpl::OnLowLatencyCreated(
-    base::SharedMemoryHandle handle,
-    base::SyncSocket::Handle socket_handle,
-    uint32 length) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kLowLatency, latency_type_);
-#if defined(OS_WIN)
-  DCHECK(handle);
-#else
-  DCHECK_GE(handle.fd, 0);
-#endif
-  DCHECK_NE(0u, length);
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  shared_memory_.reset(new base::SharedMemory(handle, false));
-  shared_memory_->Map(media::TotalSharedMemorySizeInBytes(length));
-  shared_memory_size_ = length;
-
-  CreateSocket(socket_handle);
-  CreateAudioThread();
-}
-
-void AudioRendererImpl::OnRequestPacket(AudioBuffersState buffers_state) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-  {
-    base::AutoLock auto_lock(lock_);
-    DCHECK(!pending_request_);
-    pending_request_ = true;
-    request_buffers_state_ = buffers_state;
-  }
-
-  // Try to fill in the fulfill the packet request.
-  NotifyPacketReadyTask();
-}
-
-void AudioRendererImpl::OnStateChanged(AudioStreamState state) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  switch (state) {
-    case kAudioStreamError:
-      // We receive this error if we counter an hardware error on the browser
-      // side. We can proceed with ignoring the audio stream.
-      // TODO(hclam): We need more handling of these kind of error. For example
-      // re-try creating the audio output stream on the browser side or fail
-      // nicely and report to demuxer that the whole audio stream is discarded.
-      host()->DisableAudioRenderer();
-      break;
-    // TODO(hclam): handle these events.
-    case kAudioStreamPlaying:
-    case kAudioStreamPaused:
-      break;
-    default:
-      NOTREACHED();
-      break;
-  }
-}
-
-void AudioRendererImpl::OnVolume(double volume) {
-  // TODO(hclam): decide whether we need to report the current volume to
-  // pipeline.
-}
-
-void AudioRendererImpl::CreateStreamTask(const AudioParameters& audio_params) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  stream_created_ = true;
-
-  // Make sure we don't call create more than once.
-  DCHECK_EQ(0, stream_id_);
-  stream_id_ = filter_->AddDelegate(this);
-  ChildProcess::current()->io_message_loop()->AddDestructionObserver(this);
-
-  AudioParameters params_to_send(audio_params);
-  // Let the browser choose packet size.
-  params_to_send.samples_per_packet = 0;
-
-  Send(new AudioHostMsg_CreateStream(stream_id_,
-                                     params_to_send,
-                                     latency_type_ == kLowLatency));
-}
-
-void AudioRendererImpl::PlayTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
+void AudioRendererImpl::DoPlay() {
   earliest_end_time_ = base::Time::Now();
-  Send(new AudioHostMsg_PlayStream(stream_id_));
+  DCHECK(audio_device_.get());
+  audio_device_->Play();
 }
 
-void AudioRendererImpl::PauseTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  Send(new AudioHostMsg_PauseStream(stream_id_));
+void AudioRendererImpl::DoPause() {
+  DCHECK(audio_device_.get());
+  audio_device_->Pause(false);
 }
 
-void AudioRendererImpl::SeekTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
+void AudioRendererImpl::DoSeek() {
   earliest_end_time_ = base::Time::Now();
-  // We have to pause the audio stream before we can flush.
-  Send(new AudioHostMsg_PauseStream(stream_id_));
-  Send(new AudioHostMsg_FlushStream(stream_id_));
-}
-
-void AudioRendererImpl::DestroyTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  base::AutoLock auto_lock(lock_);
-  // Errors can cause us to get here before CreateStreamTask ever ran, in which
-  // case there's nothing to do.
-  if (!stream_created_)
-    return;
-
-  // Make sure we don't call destroy more than once.
-  DCHECK_NE(0, stream_id_);
-  filter_->RemoveDelegate(stream_id_);
-  Send(new AudioHostMsg_CloseStream(stream_id_));
-  // During shutdown this may be NULL; don't worry about deregistering in that
-  // case.
-  if (ChildProcess::current())
-    ChildProcess::current()->io_message_loop()->RemoveDestructionObserver(this);
-  stream_id_ = 0;
-}
-
-void AudioRendererImpl::SetVolumeTask(double volume) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
 
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-  Send(new AudioHostMsg_SetVolume(stream_id_, volume));
+  // Pause and flush the stream when we seek to a new location.
+  DCHECK(audio_device_.get());
+  audio_device_->Pause(true);
 }
 
-void AudioRendererImpl::NotifyPacketReadyTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
+void AudioRendererImpl::Render(const std::vector<float*>& audio_data,
+                               size_t number_of_frames,
+                               size_t audio_delay_milliseconds) {
+  if (stopped_ || GetPlaybackRate() == 0.0f) {
+    // Output silence if stopped.
+    for (size_t i = 0; i < audio_data.size(); ++i)
+      memset(audio_data[i], 0, sizeof(float) * number_of_frames);
     return;
-  if (pending_request_ && GetPlaybackRate() > 0.0f) {
-    DCHECK(shared_memory_.get());
-
-    // Adjust the playback delay.
-    base::Time current_time = base::Time::Now();
-
-    base::TimeDelta request_delay =
-        ConvertToDuration(request_buffers_state_.total_bytes());
-
-    // Add message delivery delay.
-    if (current_time > request_buffers_state_.timestamp) {
-      base::TimeDelta receive_latency =
-          current_time - request_buffers_state_.timestamp;
-
-      // If the receive latency is too much it may offset all the delay.
-      if (receive_latency >= request_delay) {
-        request_delay = base::TimeDelta();
-      } else {
-        request_delay -= receive_latency;
-      }
-    }
-
-    // Finally we need to adjust the delay according to playback rate.
-    if (GetPlaybackRate() != 1.0f) {
-      request_delay = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                  GetPlaybackRate())));
-    }
-
-    bool buffer_empty = (request_buffers_state_.pending_bytes == 0) &&
-                        (current_time >= earliest_end_time_);
-
-    // For high latency mode we don't write length into shared memory,
-    // it is explicit part of AudioHostMsg_NotifyPacketReady() message,
-    // so no need to reserve first word of buffer for length.
-    uint32 filled = FillBuffer(static_cast<uint8*>(shared_memory_->memory()),
-                               shared_memory_size_, request_delay,
-                               buffer_empty);
-    UpdateEarliestEndTime(filled, request_delay, current_time);
-    pending_request_ = false;
-
-    // Then tell browser process we are done filling into the buffer.
-    Send(new AudioHostMsg_NotifyPacketReady(stream_id_, filled));
   }
-}
 
-void AudioRendererImpl::WillDestroyCurrentMessageLoop() {
-  DCHECK(!ChildProcess::current() ||  // During shutdown.
-         (MessageLoop::current() ==
-          ChildProcess::current()->io_message_loop()));
+  // Adjust the playback delay.
+  base::Time current_time = base::Time::Now();
 
-  // We treat the IO loop going away the same as stopping.
-  {
-    base::AutoLock auto_lock(lock_);
-    if (stopped_)
-      return;
+  base::TimeDelta request_delay =
+      base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
 
-    stopped_ = true;
+  // Finally we need to adjust the delay according to playback rate.
+  if (GetPlaybackRate() != 1.0f) {
+    request_delay = base::TimeDelta::FromMicroseconds(
+        static_cast<int64>(ceil(request_delay.InMicroseconds() *
+                                GetPlaybackRate())));
   }
-  DestroyTask();
-}
 
-// Our audio thread runs here. We receive requests for more data and send it
-// on this thread.
-void AudioRendererImpl::Run() {
-  DCHECK_EQ(kLowLatency, latency_type_);
-  audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
-
-  int bytes;
-  while (sizeof(bytes) == socket_->Receive(&bytes, sizeof(bytes))) {
-    if (bytes == media::AudioOutputController::kPauseMark) {
-      // When restarting playback, host should get new data,
-      // not what is currently in the buffer.
-      media::SetActualDataSizeInBytes(shared_memory_.get(),
-                                      shared_memory_size_,
-                                      0);
-      continue;
-    }
-    else if (bytes < 0)
-      break;
-    base::AutoLock auto_lock(lock_);
-    if (stopped_)
-      break;
-    float playback_rate = GetPlaybackRate();
-    if (playback_rate <= 0.0f)
-      continue;
-    DCHECK(shared_memory_.get());
-    base::TimeDelta request_delay = ConvertToDuration(bytes);
+  uint32 bytes_per_frame =
+      audio_parameters_.bits_per_sample * audio_parameters_.channels / 8;
 
-    // We need to adjust the delay according to playback rate.
-    if (playback_rate != 1.0f) {
-      request_delay = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                  playback_rate)));
-    }
-    base::Time time_now = base::Time::Now();
-    uint32 size = FillBuffer(static_cast<uint8*>(shared_memory_->memory()),
-                             shared_memory_size_,
+  const size_t buf_size = number_of_frames * bytes_per_frame;
+  scoped_array<uint8> buf(new uint8[buf_size]);
+
+  base::Time time_now = base::Time::Now();
+  uint32 filled = FillBuffer(buf.get(),
+                             buf_size,
                              request_delay,
                              time_now >= earliest_end_time_);
-    media::SetActualDataSizeInBytes(shared_memory_.get(),
-                                    shared_memory_size_,
-                                    size);
-    UpdateEarliestEndTime(size, request_delay, time_now);
+  DCHECK_LE(filled, buf_size);
+
+  uint32 filled_frames = filled / bytes_per_frame;
+
+  // Deinterleave each audio channel.
+  int channels = audio_data.size();
+  for (int channel_index = 0; channel_index < channels; ++channel_index) {
+    media::DeinterleaveAudioChannel(buf.get(),
+                                    audio_data[channel_index],
+                                    channels,
+                                    channel_index,
+                                    bytes_per_frame / channels,
+                                    filled_frames);
+
+    // If FillBuffer() didn't give us enough data then zero out the remainder.
+    if (filled_frames < number_of_frames) {
+      int frames_to_zero = number_of_frames - filled_frames;
+      memset(audio_data[channel_index], 0, sizeof(float) * frames_to_zero);
+    }
   }
 }
-
-void AudioRendererImpl::Send(IPC::Message* message) {
-  filter_->Send(message);
-}