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

content/renderer/media/webrtc_audio_renderer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 432 matching lines @@
 
 void WebRtcAudioRenderer::SetVolume(float volume) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(volume >= 0.0f && volume <= 1.0f);
 
   playing_state_.set_volume(volume);
   OnPlayStateChanged(media_stream_, &playing_state_);
 }
 
 base::TimeDelta WebRtcAudioRenderer::GetCurrentRenderTime() const {
-  return base::TimeDelta();
+  DCHECK(thread_checker_.CalledOnValidThread());
+  base::AutoLock auto_lock(lock_);
+  return current_time_;
 }
 
 bool WebRtcAudioRenderer::IsLocalRenderer() const {
   return false;
 }
 
 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
                                 int audio_delay_milliseconds) {
   base::AutoLock auto_lock(lock_);
   if (!source_)
@@ skipping 24 matching lines @@
            << fifo_frame_delay << ", "
            << audio_bus->frames() << ")";
 
   int output_delay_milliseconds = audio_delay_milliseconds_;
   output_delay_milliseconds += fifo_delay_milliseconds_;
   DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;
 
   // We need to keep render data for the |source_| regardless of |state_|,
   // otherwise the data will be buffered up inside |source_|.
   source_->RenderData(audio_bus, sink_params_.sample_rate(),
-                      output_delay_milliseconds);
+                      output_delay_milliseconds,
+                      &current_time_);
 
   // Avoid filling up the audio bus if we are not playing; instead
   // return here and ensure that the returned value in Render() is 0.
   if (state_ != PLAYING)
     audio_bus->Zero();
 }
 
 void WebRtcAudioRenderer::UpdateSourceVolume(
     webrtc::AudioSourceInterface* source) {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 72 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content