There is a racing between SyncSocket::Receive in audio_thread_ and SyncSocket::Close in renderer ...

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(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) {
+      stream_id_(0),
+      audio_event_(true, false) {

[enal1, 2011/11/29 16:14:48]

Can you please explain what exactly arguments to event constructor mean...

[tommi (sloooow) - chröme, 2011/11/29 16:41:53]

I think it's ok to set |manual_reset| to false in this case (first argument).

   filter_ = RenderThreadImpl::current()->audio_message_filter();
   audio_data_.reserve(channels);
   for (int i = 0; i < channels; ++i) {
     float* channel_data = new float[buffer_size];
     audio_data_.push_back(channel_data);
   }
 }
 
 AudioDevice::~AudioDevice() {
   // The current design requires that the user calls Stop() before deleting
@@ skipping 26 matching lines @@
 
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&AudioDevice::ShutDownOnIOThread, this, &completion));
 
   // We wait here for the IO task to be completed to remove race conflicts
   // with OnLowLatencyCreated() and to ensure that Stop() acts as a synchronous
   // function call.
   if (completion.TimedWait(kMaxTimeOut)) {
     if (audio_thread_.get()) {
-      socket_->Close();
+      audio_event_.Signal();
       audio_thread_->Join();
       audio_thread_.reset(NULL);
     }
   } else {
     LOG(ERROR) << "Failed to shut down audio output on IO thread";
     return false;
   }
 
   return true;
 }
@@ skipping 92 matching lines @@
 
   shared_memory_.reset(new base::SharedMemory(handle, false));
   shared_memory_->Map(length);
 
   DCHECK_GE(length, buffer_size_ * sizeof(int16) * channels_);
 
   socket_.reset(new base::SyncSocket(socket_handle));
   // Allow the client to pre-populate the buffer.
   FireRenderCallback();
 
+  audio_event_.Reset();

[tommi (sloooow) - chröme, 2011/11/29 16:41:53]

if you set manual_reset to false, you don't have to do this step.

   audio_thread_.reset(
       new base::DelegateSimpleThread(this, "renderer_audio_thread"));
   audio_thread_->Start();
 
   MessageLoop::current()->PostTask(
       FROM_HERE,
       base::Bind(&AudioDevice::StartOnIOThread, 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,
+  while (!audio_event_.IsSignaled() &&

[enal1, 2011/11/29 16:14:48]

I don't fully understand how event helps in a case when thread is blocked on
socket wait. I.e. suggested solution works if there is consistent stream of data
coming in, but not for case when there is no data...

[tommi (sloooow) - chröme, 2011/11/29 16:41:53]

good point!  Is there a way to read from the socket asynchronously?

[no longer working on chromium, 2011/11/30 17:02:55]

Really good question. No, event does not help at all when the socket is waiting
for receiving, but the current solution, which calls socket_.Close() does not
fix the problem either, the Close() can only jumps in after socket finishes
receiving.

+         (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();
   }
+
+  socket_->Close();
 }
 
 void AudioDevice::FireRenderCallback() {
   TRACE_EVENT0("audio", "AudioDevice::FireRenderCallback");
 
   if (callback_) {
     // 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.
     media::InterleaveFloatToInt16(audio_data_,
                                   static_cast<int16*>(shared_memory_data()),
                                   buffer_size_);
   }
 }