Land Henrik's patch: http://codereview.chromium.org/6721027/

content/renderer/audio_device.cc

Index: content/renderer/audio_device.cc
===================================================================
--- content/renderer/audio_device.cc	(revision 79311)
+++ content/renderer/audio_device.cc	(working copy)
@@ -1,12 +1,14 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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/audio_device.h"
 
+#include "base/message_loop.h"
 #include "base/singleton.h"
 #include "chrome/renderer/render_thread.h"
 #include "content/common/audio_messages.h"
+#include "content/common/child_process.h"
 #include "content/common/view_messages.h"
 #include "media/audio/audio_util.h"
 
@@ -41,7 +43,7 @@
   scoped_refptr<AudioMessageFilter> filter_;
 };
 
-}
+}  // namespace
 
 AudioDevice::AudioDevice(size_t buffer_size,
                          int channels,
@@ -49,17 +51,24 @@
                          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) {
   audio_data_.reserve(channels);
   for (int i = 0; i < channels; ++i) {
     float* channel_data = new float[buffer_size];
     audio_data_.push_back(channel_data);
   }
+  // Lazily create the message filter and share across AudioDevice instances.
+  filter_ = AudioMessageFilterCreator::SharedFilter();
 }
 
 AudioDevice::~AudioDevice() {
+  // Make sure we have been shut down.
+  DCHECK_EQ(0, stream_id_);
   Stop();
   for (int i = 0; i < channels_; ++i)
     delete [] audio_data_[i];
@@ -71,53 +80,102 @@
   if (stream_id_)
     return false;
 
-  // Lazily create the message filter and share across AudioDevice instances.
-  filter_ = AudioMessageFilterCreator::SharedFilter();
-
-  stream_id_ = filter_->AddDelegate(this);
-
   AudioParameters params;
   params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
   params.channels = channels_;
   params.sample_rate = static_cast<int>(sample_rate_);
-  params.bits_per_sample = 16;
+  params.bits_per_sample = bits_per_sample_;
   params.samples_per_packet = buffer_size_;
 
-  filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true));
+  // Ensure that the initialization task is posted on the I/O thread by
+  // accessing the I/O message loop directly. This approach avoids a race
+  // condition which could exist if the message loop of the filter was
+  // used instead.
+  DCHECK(ChildProcess::current()) << "Must be in the renderer";
+  MessageLoop* message_loop = ChildProcess::current()->io_message_loop();
+  if (!message_loop)
+    return false;
 
+  message_loop->PostTask(FROM_HERE,
+      NewRunnableMethod(this, &AudioDevice::InitializeOnIOThread, params));
+
   return true;
 }
 
 bool AudioDevice::Stop() {
-  if (stream_id_) {
-    OnDestroy();
-    return true;
+  if (!stream_id_)
+    return false;
+
+  filter_->message_loop()->PostTask(FROM_HERE,
+      NewRunnableMethod(this, &AudioDevice::ShutDownOnIOThread));
+
+  if (audio_thread_.get()) {
+    socket_->Close();
+    audio_thread_->Join();
   }
-  return false;
+
+  return true;
 }
 
-void AudioDevice::OnDestroy() {
-  // Make sure we don't call destroy more than once.
-  DCHECK_NE(0, stream_id_);
+bool AudioDevice::SetVolume(double volume) {
   if (!stream_id_)
+    return false;
+
+  if (volume < 0 || volume > 1.0)
+    return false;
+
+  filter_->message_loop()->PostTask(FROM_HERE,
+      NewRunnableMethod(this, &AudioDevice::SetVolumeOnIOThread, volume));
+
+  volume_ = volume;
+
+  return true;
+}
+
+bool AudioDevice::GetVolume(double* volume) {
+  if (!stream_id_)
+    return false;
+
+  // Return a locally cached version of the current scaling factor.
+  *volume = volume_;
+
+  return true;
+}
+
+void AudioDevice::InitializeOnIOThread(const AudioParameters& params) {
+  stream_id_ = filter_->AddDelegate(this);
+  filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true));
+}
+
+void AudioDevice::StartOnIOThread() {
+  if (stream_id_)
+    filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_));
+}
+
+void AudioDevice::ShutDownOnIOThread() {
+  // Make sure we don't call shutdown more than once.
+  if (!stream_id_)
     return;
 
+  filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_));
   filter_->RemoveDelegate(stream_id_);
-  filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_));
   stream_id_ = 0;
-  if (audio_thread_.get()) {
-    socket_->Close();
-    audio_thread_->Join();
-  }
 }
 
+void AudioDevice::SetVolumeOnIOThread(double volume) {
+  if (stream_id_)
+    filter_->Send(new AudioHostMsg_SetVolume(0, 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) {
-  // Not needed in this simple implementation.
+  if (state == kAudioStreamError) {
+    DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
+  }
   NOTIMPLEMENTED();
 }
 
@@ -131,7 +189,6 @@
     base::SharedMemoryHandle handle,
     base::SyncSocket::Handle socket_handle,
     uint32 length) {
-
 #if defined(OS_WIN)
   DCHECK(handle);
   DCHECK(socket_handle);
@@ -140,7 +197,6 @@
   DCHECK_GE(socket_handle, 0);
 #endif
   DCHECK(length);
-  DCHECK(!audio_thread_.get());
 
   // TODO(crogers) : check that length is big enough for buffer_size_
 
@@ -156,28 +212,39 @@
       new base::DelegateSimpleThread(this, "renderer_audio_thread"));
   audio_thread_->Start();
 
-  filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_));
+  if (filter_) {
+    filter_->message_loop()->PostTask(FROM_HERE,
+        NewRunnableMethod(this, &AudioDevice::StartOnIOThread));
+  }
 }
 
 void AudioDevice::OnVolume(double volume) {
-  // Not needed in this simple implementation.
   NOTIMPLEMENTED();
 }
 
 // Our audio thread runs here.
 void AudioDevice::Run() {
   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) {
+    {
+      // Convert the number of pending bytes in the render buffer
+      // into milliseconds.
+      audio_delay_milliseconds_ = pending_data / bytes_per_ms;
+    }
+
     FireRenderCallback();
   }
 }
 
 void AudioDevice::FireRenderCallback() {
   if (callback_) {
-    // Ask client to render audio.
-    callback_->Render(audio_data_, buffer_size_);
+    // Update the audio-delay measurement then ask client to render audio.
+    callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_);
 
     // Interleave, scale, and clip to int16.
     int16* output_buffer16 = static_cast<int16*>(shared_memory_data());