Reland: 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 142 matching lines @@
       SourcePlayingStates;
 
   // 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,
-             uint32_t frames_skipped) override;
+  int Render(base::TimeDelta delay,
+             base::TimeTicks delay_timestamp,
+             int prior_frames_skipped,
+             media::AudioBus* audio_bus) 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 45 matching lines @@
 
   // Ref count for the MediaPlayers which have called Start() but not Stop().
   int start_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.
   std::unique_ptr<media::AudioPullFifo> audio_fifo_;
 
   // Contains the accumulated delay estimate which is provided to the WebRTC
   // AEC.
-  int audio_delay_milliseconds_;
+  base::TimeDelta audio_delay_;
 
   base::TimeDelta current_time_;
 
   // Saved volume and playing state of the root renderer.
   PlayingState playing_state_;
 
   // Audio params used by the sink of the renderer.
   media::AudioParameters sink_params_;
 
   // The preferred device id of the output device or empty for the default
@@ skipping 11 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_