Fix start/stop of html5 audio stream and race condition when pausing.

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"
@@ skipping 269 matching lines @@
 
 void AudioDevice::Send(IPC::Message* message) {
   filter_->Send(message);
 }
 
 // Our audio thread runs here.
 void AudioDevice::Run() {
   audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
 
   base::SharedMemory shared_memory(shared_memory_handle_, false);
-  shared_memory.Map(memory_length_);
+  shared_memory.Map(media::TotalSharedMemorySizeInBytes(memory_length_));
   // Allow the client to pre-populate the buffer.
-  FireRenderCallback(reinterpret_cast<int16*>(shared_memory.memory()));
+  // If it cannot it just returns zero.
+  size_t num_frames = FireRenderCallback(
+      reinterpret_cast<int16*>(shared_memory.memory()));
+  if (num_frames == 0)
+    media::SetUnknownDataSize(&shared_memory, memory_length_);
+  else
+    media::SetActualDataSizeInBytes(&shared_memory,
+                                    memory_length_,
+                                    num_frames * channels_ * sizeof(int16));
 
   base::SyncSocket socket(socket_handle_);
 
   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))) {
     if (pending_data == media::AudioOutputController::kPauseMark) {
       memset(shared_memory.memory(), 0, memory_length_);
+      media::SetActualDataSizeInBytes(&shared_memory, memory_length_, 0);
       continue;
     } else if (pending_data < 0) {
       break;
     }
     // Convert the number of pending bytes in the render buffer
     // into milliseconds.
     audio_delay_milliseconds_ = pending_data / bytes_per_ms;
-    FireRenderCallback(reinterpret_cast<int16*>(shared_memory.memory()));
+    num_frames = FireRenderCallback(
+        reinterpret_cast<int16*>(shared_memory.memory()));
+    // Let the host know we are done.
+    media::SetActualDataSizeInBytes(&shared_memory,
+                                    memory_length_,
+                                    num_frames * channels_ * sizeof(int16));
   }
 }
 
-void AudioDevice::FireRenderCallback(int16* data) {
+size_t AudioDevice::FireRenderCallback(int16* data) {
   TRACE_EVENT0("audio", "AudioDevice::FireRenderCallback");
 
+  size_t num_frames = 0;
   if (callback_) {
     // Update the audio-delay measurement then ask client to render audio.
-    callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_);
+    num_frames = callback_->Render(audio_data_,
+                                   buffer_size_,
+                                   audio_delay_milliseconds_);
 
     // Interleave, scale, and clip to int16.
     // TODO(crogers): avoid converting to integer here, and pass the data
     // to the browser process as float, so we don't lose precision for
     // audio hardware which has better than 16bit precision.
     media::InterleaveFloatToInt16(audio_data_,
                                   data,
                                   buffer_size_);
   }
+  return num_frames;
 }
 
 void AudioDevice::ShutDownAudioThread() {
   // Synchronize with OnLowLatencyCreated().
   base::AutoLock auto_lock(lock_);
   if (audio_thread_.get()) {
     // Close the socket handler to terminate the main thread function in the
     // audio thread.
     {
       base::SyncSocket socket(socket_handle_);
     }
     audio_thread_->Join();
     audio_thread_.reset(NULL);
   }
 }