Avoids irregular OnMoreData callbacks on Windows using Core Audio

content/renderer/media/webrtc_audio_renderer.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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
+#include "base/threading/thread_checker.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_decoder.h"
+#include "media/base/audio_pull_fifo.h"
 #include "media/base/audio_renderer_sink.h"
 #include "webkit/media/media_stream_audio_renderer.h"
 
 namespace content {
 
 class RendererAudioOutputDevice;
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
-// for connecting WebRtc MediaStream with pipeline.
+// for connecting WebRtc MediaStream with the audio pipeline.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public webkit_media::MediaStreamAudioRenderer) {
  public:
   explicit WebRtcAudioRenderer(int source_render_view_id);
 
-  // Initialize function called by clients like WebRtcAudioDeviceImpl. Note,
+  // Initialize function called by clients like WebRtcAudioDeviceImpl.
   // Stop() has to be called before |source| is deleted.
-  // Returns false if Initialize() fails.
   bool Initialize(WebRtcAudioRendererSource* source);
 
   // Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
   // MediaStreamAudioRenderer implementation.
   virtual void Start() OVERRIDE;
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void SetVolume(float volume) OVERRIDE;
   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
   virtual bool IsLocalRenderer() const OVERRIDE;
 
  protected:
   virtual ~WebRtcAudioRenderer();
 
  private:
   enum State {
     UNINITIALIZED,
     PLAYING,
     PAUSED,
   };
+
+  // Used to DCHECK that we are called on the correct thread.
+  base::ThreadChecker thread_checker_;
+
   // Flag to keep track the state of the renderer.
   State state_;
 
   // media::AudioRendererSink::RenderCallback implementation.
+  // These two methods are called on the AudioOutputDevice worker thread.
   virtual int Render(media::AudioBus* audio_bus,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
+  // Called by AudioPullFifo when more data is necessary.
+  // This method is called on the AudioOutputDevice worker thread.
+  void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
+
   // The render view in which the audio is rendered into |sink_|.
   const int source_render_view_id_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<RendererAudioOutputDevice> sink_;
 
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
-  // Cached values of utilized audio parameters. Platform dependent.
-  media::AudioParameters params_;
-
   // Buffers used for temporary storage during render callbacks.
   // Allocated during initialization.
   scoped_array<int16> buffer_;
 
-  // Protect access to |state_|.
+  // Protects access to |state_|, |source_| and |sink_|.
   base::Lock lock_;
 
   // Ref count for the MediaPlayers which are playing audio.
   int play_ref_count_;
 
+  // Used to buffer data between the client and the output device in cases where
+  // the client buffer size is not the same as the output device buffer size.
+  scoped_ptr<media::AudioPullFifo> audio_fifo_;
+
+  double frame_duration_milliseconds_;

[tommi (sloooow) - chröme, 2013/01/31 13:42:08]

why is this a double and audio_delay_milliseconds_ an int?

[henrika (OOO until Aug 14), 2013/01/31 14:29:38]

I want to maintain precision. For fs=48kHz, we have frame_duration_milliseconds_
= 0.020833333333333 and it is used to derive a new (compensated) delay value.

+
+  int audio_delay_milliseconds_;
+
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_