Simplify AudioRendererImpl by using AudioDevice.

content/renderer/media/audio_device.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/audio_device.h"
 
 #include "base/bind.h"
 #include "base/debug/trace_event.h"
 #include "base/message_loop.h"
 #include "base/time.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
 #include "content/common/view_messages.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 
+AudioDevice::AudioDevice()
+    : buffer_size_(0),
+      channels_(0),
+      bits_per_sample_(16),
+      sample_rate_(0),
+      latency_format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
+      callback_(0),
+      audio_delay_milliseconds_(0),
+      volume_(1.0),
+      stream_id_(0) {
+  filter_ = RenderThreadImpl::current()->audio_message_filter();
+}
+
 AudioDevice::AudioDevice(size_t buffer_size,
                          int channels,
                          double sample_rate,
                          RenderCallback* callback)
     : buffer_size_(buffer_size),
       channels_(channels),
       bits_per_sample_(16),
       sample_rate_(sample_rate),
       callback_(callback),
       audio_delay_milliseconds_(0),
       volume_(1.0),
       stream_id_(0) {
   filter_ = RenderThreadImpl::current()->audio_message_filter();
+  Initialize(buffer_size,
+             channels,
+             sample_rate,
+             AudioParameters::AUDIO_PCM_LOW_LATENCY,
+             callback);
+}
+
+void AudioDevice::Initialize(size_t buffer_size,
+                             int channels,
+                             double sample_rate,
+                             AudioParameters::Format latency_format,
+                             RenderCallback* callback) {
+  CHECK_EQ(0, stream_id_) <<
+      "AudioDevice::Initialize() must be called before Start()";
+
+  buffer_size_ = buffer_size;
+  channels_ = channels;
+  sample_rate_ = sample_rate;
+  latency_format_ = latency_format;
+  callback_ = callback;
+
+  // Cleanup from any previous initialization.
+  for (size_t i = 0; i < audio_data_.size(); ++i)
+    delete [] audio_data_[i];
+
   audio_data_.reserve(channels);
   for (int i = 0; i < channels; ++i) {
     float* channel_data = new float[buffer_size];
     audio_data_.push_back(channel_data);
   }
 }
 
+bool AudioDevice::IsInitialized() {
+  return audio_data_.size() > 0;
+}
+
 AudioDevice::~AudioDevice() {
   // The current design requires that the user calls Stop() before deleting
   // this class.
   CHECK_EQ(0, stream_id_);
   for (int i = 0; i < channels_; ++i)
     delete [] audio_data_[i];
 }
 
 void AudioDevice::Start() {
   AudioParameters params;
-  params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
+  params.format = latency_format_;
   params.channels = channels_;
   params.sample_rate = static_cast<int>(sample_rate_);
   params.bits_per_sample = bits_per_sample_;
   params.samples_per_packet = buffer_size_;
 
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&AudioDevice::InitializeOnIOThread, this, params));
 }
 
@@ skipping 19 matching lines @@
       audio_thread_.reset(NULL);
     }
   } else {
     LOG(ERROR) << "Failed to shut down audio output on IO thread";
     return false;
   }
 
   return true;
 }
 
+void AudioDevice::Play() {
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioDevice::PlayOnIOThread, this));
+}
+
+void AudioDevice::Pause(bool flush) {
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioDevice::PauseOnIOThread, this, flush));
+}
+
 bool AudioDevice::SetVolume(double volume) {
   if (volume < 0 || volume > 1.0)
     return false;
 
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&AudioDevice::SetVolumeOnIOThread, this, volume));
 
   volume_ = volume;
 
   return true;
 }
 
 void AudioDevice::GetVolume(double* volume) {
   // Return a locally cached version of the current scaling factor.
   *volume = volume_;
 }
 
 void AudioDevice::InitializeOnIOThread(const AudioParameters& params) {
-  // Make sure we don't call Start() more than once.
+  // Make sure we don't create the stream more than once.
   DCHECK_EQ(0, stream_id_);
   if (stream_id_)
     return;
 
   stream_id_ = filter_->AddDelegate(this);
   Send(new AudioHostMsg_CreateStream(stream_id_, params, true));
 }
 
-void AudioDevice::StartOnIOThread() {
+void AudioDevice::PlayOnIOThread() {
   if (stream_id_)
     Send(new AudioHostMsg_PlayStream(stream_id_));
 }
 
+void AudioDevice::PauseOnIOThread(bool flush) {
+  if (stream_id_) {
+    Send(new AudioHostMsg_PauseStream(stream_id_));
+    if (flush)
+      Send(new AudioHostMsg_FlushStream(stream_id_));
+  }
+}
+
 void AudioDevice::ShutDownOnIOThread(base::WaitableEvent* completion) {
   // Make sure we don't call shutdown more than once.
   if (!stream_id_) {
     completion->Signal();
     return;
   }
 
   filter_->RemoveDelegate(stream_id_);
   Send(new AudioHostMsg_CloseStream(stream_id_));
   stream_id_ = 0;
 
   completion->Signal();
 }
 
 void AudioDevice::SetVolumeOnIOThread(double volume) {
   if (stream_id_)
     Send(new AudioHostMsg_SetVolume(stream_id_, volume));
 }
 
 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
   // This method does not apply to the low-latency system.
-  NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnStateChanged(AudioStreamState state) {
   if (state == kAudioStreamError) {
     DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
   }
-  NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnCreated(
     base::SharedMemoryHandle handle, uint32 length) {
   // Not needed in this simple implementation.
-  NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnLowLatencyCreated(
     base::SharedMemoryHandle handle,
     base::SyncSocket::Handle socket_handle,
     uint32 length) {
   DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
 #if defined(OS_WIN)
   DCHECK(handle);
   DCHECK(socket_handle);
@@ skipping 19 matching lines @@
   socket_.reset(new base::SyncSocket(socket_handle));
   // Allow the client to pre-populate the buffer.
   FireRenderCallback();
 
   audio_thread_.reset(
       new base::DelegateSimpleThread(this, "renderer_audio_thread"));
   audio_thread_->Start();
 
   MessageLoop::current()->PostTask(
       FROM_HERE,
-      base::Bind(&AudioDevice::StartOnIOThread, this));
+      base::Bind(&AudioDevice::PlayOnIOThread, this));
 }
 
 void AudioDevice::OnVolume(double volume) {
   NOTIMPLEMENTED();
 }
 
 void AudioDevice::Send(IPC::Message* message) {
   filter_->Send(message);
 }
 
 // Our audio thread runs here.
 void AudioDevice::Run() {
   audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
 
   int pending_data;
   const int samples_per_ms = static_cast<int>(sample_rate_) / 1000;
   const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms;
 
-  while ((sizeof(pending_data) == socket_->Receive(&pending_data,
-                                                   sizeof(pending_data))) &&
-         (pending_data >= 0)) {
-
+  while (sizeof(pending_data) ==
+      socket_->Receive(&pending_data, sizeof(pending_data))) {
     // Convert the number of pending bytes in the render buffer
     // into milliseconds.
     audio_delay_milliseconds_ = pending_data / bytes_per_ms;
     FireRenderCallback();
   }
 }
 
 void AudioDevice::FireRenderCallback() {
   TRACE_EVENT0("audio", "AudioDevice::FireRenderCallback");
 
@@ skipping 20 matching lines @@
 
 size_t AudioDevice::GetAudioHardwareBufferSize() {
   // Uses cached value if possible.
   static size_t buffer_size = 0;
 
   if (!buffer_size)
     buffer_size = media::GetAudioHardwareBufferSize();
 
   return buffer_size;
 }