Decouple 'give me more data' and 'rendered end of stream' audio callbacks.

content/renderer/media/audio_renderer_impl.h

 // Copyright (c) 2012 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.
 //
 // Audio rendering unit utilizing AudioDevice.
 //
 // This class lives inside three threads during it's lifetime, namely:
 // 1. Render thread.
 //    This object is created on the render thread.
 // 2. Pipeline thread
 //    OnInitialize() is called here with the audio format.
 //    Play/Pause/Seek also happens here.
 // 3. Audio thread created by the AudioDevice.
 //    Render() is called here where audio data is decoded into raw PCM data.
 
 #ifndef CONTENT_RENDERER_MEDIA_AUDIO_RENDERER_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_AUDIO_RENDERER_IMPL_H_
 #pragma once
 
 #include <vector>
 
+#include "base/atomicops.h"
 #include "base/gtest_prod_util.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "content/renderer/media/audio_device.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_renderer_sink.h"
 #include "media/filters/audio_renderer_base.h"
 
 class AudioMessageFilter;
@@ skipping 38 matching lines @@
   // Helper methods.
   // Convert number of bytes to duration of time using information about the
   // number of channels, sample rate and sample bits.
   base::TimeDelta ConvertToDuration(int bytes);
 
   // Methods called on pipeline thread ----------------------------------------
   void DoPlay();
   void DoPause();
   void DoSeek();
 
+  // Methods called on IO thread ----------------------------------------------
+  void DoSignalEndOfStream(base::subtle::AtomicWord stream_id);
+
   // media::AudioRendererSink::RenderCallback implementation.
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
   virtual void OnError() OVERRIDE;
 
+  // Returns delay in ms before call to OnRenderEndOfStream().
+  // Made virtual so test can override.
+  virtual int64 OnRenderEndOfStreamDelay();
+
   // Accessors used by tests.
   base::Time earliest_end_time() const {
     return earliest_end_time_;
   }
 
   void set_earliest_end_time(const base::Time& earliest_end_time) {
     earliest_end_time_ = earliest_end_time;
   }
 
   uint32 bytes_per_second() const {
@@ skipping 10 matching lines @@
 
   // A flag that indicates this filter is called to stop.
   bool stopped_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioRendererSink> sink_;
 
   // Set to true when OnInitialize() is called.
   bool is_initialized_;
 
+  // Set to true when we post delayed task to signal 'ended' event.
+  bool ended_event_scheduled_;
+
   // We're supposed to know amount of audio data OS or hardware buffered, but
   // that is not always so -- on my Linux box
   // AudioBuffersState::hardware_delay_bytes never reaches 0.
   //
   // As a result we cannot use it to find when stream ends. If we just ignore
   // buffered data we will notify host that stream ended before it is actually
   // did so, I've seen it done ~140ms too early when playing ~150ms file.
   //
   // Instead of trying to invent OS-specific solution for each and every OS we
   // are supporting, use simple workaround: every time we fill the buffer we
   // remember when it should stop playing, and do not assume that buffer is
   // empty till that time. Workaround is not bulletproof, as we don't exactly
   // know when that particular data would start playing, but it is much better
   // than nothing.
   base::Time earliest_end_time_;
 
   AudioParameters audio_parameters_;
 
+  // Use message loop proxy, not message loop itself, to avoid crash
+  // because of message loop that ended while we are still playing.
+  // We don't need complex shutdown operations, just not posting tasks
+  // is enough, so we can use message loop proxy.
+  scoped_refptr<base::MessageLoopProxy> io_message_loop_proxy_;
+
+  // Stream id -- necessary because there is no way to cancel scheduled task.
+  // We want be sure we are still playing the same stream when delayed task is
+  // called. If seek or pause happened after scheduling but before task was
+  // called, there would be mismatch between expected and actual ids and delayed
+  // task would not do anything.
+  base::subtle::AtomicWord stream_id_;
+
   DISALLOW_COPY_AND_ASSIGN(AudioRendererImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_AUDIO_RENDERER_IMPL_H_