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

content/renderer/media/webrtc_audio_device_impl.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 108 matching lines @@
   return new_volume;
 }
 
 void WebRtcAudioDeviceImpl::OnSetFormat(
     const media::AudioParameters& params) {
   DVLOG(1) << "WebRtcAudioDeviceImpl::OnSetFormat()";
 }
 
 void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                        int sample_rate,
-                                       int audio_delay_milliseconds) {
+                                       int audio_delay_milliseconds,
+                                       base::TimeDelta* current_time) {
   render_buffer_.resize(audio_bus->frames() * audio_bus->channels());
 
   {
     base::AutoLock auto_lock(lock_);
     DCHECK(audio_transport_callback_);
     // Store the reported audio delay locally.
     output_delay_ms_ = audio_delay_milliseconds;
   }
 
   int frames_per_10_ms = (sample_rate / 100);
@@ skipping 32 matching lines @@
       audio_transport_callback_->NeedMorePlayData(frames_per_10_ms,
                                                   bytes_per_sample,
                                                   audio_bus->channels(),
                                                   sample_rate,
                                                   audio_data,
                                                   num_audio_frames,
                                                   &elapsed_time_ms,
                                                   &ntp_time_ms);
       accumulated_audio_frames += num_audio_frames;
     }
-
+    if (elapsed_time_ms >= 0) {
+      *current_time = base::TimeDelta::FromMilliseconds(elapsed_time_ms);
+    }
     audio_data += bytes_per_10_ms;
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(&render_buffer_[0],
                              audio_bus->frames(),
                              bytes_per_sample);
 
   // Pass the render data to the playout sinks.
@@ skipping 366 matching lines @@
 
   // Start the Aec dump on the current default capturer.
   scoped_refptr<WebRtcAudioCapturer> default_capturer(GetDefaultCapturer());
   if (!default_capturer)
     return;
 
   default_capturer->StartAecDump(aec_dump_file_.Pass());
 }
 
 }  // namespace content