Use audio shifter instead of a fifo for local mediastream playback.

content/renderer/media/webrtc_local_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_LOCAL_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
 
 #include <vector>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace media {
 class AudioBus;
-class AudioFifo;
+class AudioShifter;
 class AudioOutputDevice;
 class AudioParameters;
 }
 
 namespace content {
 
 class WebRtcAudioCapturer;
 
 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
 // local audio media stream tracks,
@@ skipping 78 matching lines @@
   const int source_render_frame_id_;
   const int session_id_;
 
   // MessageLoop associated with the single thread that performs all control
   // tasks.  Set to the MessageLoop that invoked the ctor.
   const scoped_refptr<base::MessageLoopProxy> message_loop_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioOutputDevice> sink_;
 
-  // Contains copies of captured audio frames.
-  scoped_ptr<media::AudioFifo> loopback_fifo_;
+  // This does all the synchronization/resampling/smoothing.
+  scoped_ptr<media::AudioShifter> audio_shifter_;
 
   // Stores last time a render callback was received. The time difference
   // between a new time stamp and this value can be used to derive the
   // total render time.
   base::TimeTicks last_render_time_;
 
   // Keeps track of total time audio has been rendered.
   base::TimeDelta total_render_time_;
 
   // The audio parameters of the capture source.
   // Must only be touched on the main thread.
   media::AudioParameters source_params_;
 
   // The audio parameters used by the sink.
   // Must only be touched on the main thread.
   media::AudioParameters sink_params_;
 
   // Set when playing, cleared when paused.
   bool playing_;
 
-  // Protects |loopback_fifo_|, |playing_| and |sink_|.
+  // Protects |audio_shifter_|, |playing_| and |sink_|.
   mutable base::Lock thread_lock_;
 
   // The preferred buffer size provided via the ctor.
   const int frames_per_buffer_;
 
   // The preferred device id of the output device or empty for the default
   // output device.
   const std::string output_device_id_;
 
   // Cache value for the volume.
   float volume_;
 
   // Flag to indicate whether |sink_| has been started yet.
   bool sink_started_;
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_