Pass the elapsed time from VoE to WebRtcAudioRenderer as the current time for the audio/video eleme…

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 171 matching lines @@
 class WebRtcAudioCapturer;
 class WebRtcAudioRenderer;
 
 // TODO(xians): Move the following two interfaces to webrtc so that
 // libjingle can own references to the renderer and capturer.
 class WebRtcAudioRendererSource {
  public:
   // Callback to get the rendered data.
   virtual void RenderData(media::AudioBus* audio_bus,
                           int sample_rate,
-                          int audio_delay_milliseconds) = 0;
+                          int audio_delay_milliseconds,
+                          base::TimeDelta* current_time) = 0;
 
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
  protected:
   virtual ~WebRtcAudioRendererSource() {}
 };
 
 class PeerConnectionAudioSink {
  public:
@@ skipping 180 matching lines @@
                      bool key_pressed) OVERRIDE;
 
   // Called on the AudioInputDevice worker thread.
   virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioOutputDevice worker thread.
   virtual void RenderData(media::AudioBus* audio_bus,
                           int sample_rate,
-                          int audio_delay_milliseconds) OVERRIDE;
+                          int audio_delay_milliseconds,
+                          base::TimeDelta* current_time) OVERRIDE;
 
   // Called on the main render thread.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
 
   // WebRtcPlayoutDataSource implementation.
   virtual void AddPlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
   virtual void RemovePlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
 
   // Helper to start the Aec dump if the default capturer exists.
   void MaybeStartAecDump();
@@ skipping 51 matching lines @@
 
   // Flag to tell if audio processing is enabled in MediaStreamAudioProcessor.
   const bool is_audio_track_processing_enabled_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_