Make more media APIs aware of |delay| and |delay_timestamp|

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 <stdint.h>
 
 #include <map>
@@ skipping 143 matching lines @@
 
   // 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.
   int Render(media::AudioBus* audio_bus,
-             uint32_t frames_delayed,
+             base::TimeDelta delay,
+             base::TimeTicks delay_timestamp,
              uint32_t frames_skipped) override;
   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);
 
   // Goes through all renderers for the |source| and applies the proper
   // volume scaling for the source based on the volume(s) of the renderer(s).
   void UpdateSourceVolume(webrtc::AudioSourceInterface* source);
@@ skipping 78 matching lines @@
   // Stores the maximum time spent waiting for render data from the source. Used
   // for logging UMA data. Logged and reset when Stop() is called.
   base::TimeDelta max_render_time_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_