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

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 <utility>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ skipping 384 matching lines @@
     source_->AudioRendererThreadStopped();
   }
   PrepareSink();
   sink_->Start();
   sink_->Play();  // Not all the sinks play on start.
 
   callback.Run(media::OUTPUT_DEVICE_STATUS_OK);
 }
 
 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
-                                uint32_t frames_delayed,
+                                base::TimeDelta delay,
+                                base::TimeTicks delay_timestamp,
                                 uint32_t frames_skipped) {
   DCHECK(sink_->CurrentThreadIsRenderingThread());
   base::AutoLock auto_lock(lock_);
   if (!source_)
     return 0;
 
-  // TODO(grunell): Converting from frames to milliseconds will potentially lose
+  // TODO(grunell): Using of milliseconds will potentially lose

[James West, 2016/10/21 19:40:17]

"Using of milliseconds" => "Using milliseconds"

   // hundreds of microseconds which may cause audio video drift. Update
   // this class and all usage of render delay msec -> frames (possibly even
   // using a double type for frames). See http://crbug.com/586540
-  uint32_t audio_delay_milliseconds = static_cast<double>(frames_delayed) *
-                                      base::Time::kMillisecondsPerSecond /
-                                      sink_params_.sample_rate();
+  int64_t audio_delay_milliseconds = delay.InMilliseconds();

[miu, 2016/10/23 01:19:15]

No need to create a local variable here. Just assign directly to the class
member |audio_delay_milliseconds_| (and delete line 423 below).

Bonus points (because of the TODO comment) if you change:

  int audio_delay_milliseconds_;  -->  base::TimeDelta audio_delay_;

[Mikhail, 2016/10/24 19:50:23]

Done.

[miu, 2016/10/24 22:45:52]

Thanks for the "bonus" change! :)

 
   DVLOG(2) << "WebRtcAudioRenderer::Render()";
   DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
 
   DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
   audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
 
   // If there are skipped frames, pull and throw away the same amount. We always
   // pull 10 ms of data from the source (see PrepareSink()), so the fifo is only
   // required if the number of frames to drop doesn't correspond to 10 ms.
@@ skipping 239 matching lines @@
     sink_params_ = new_sink_params;
   }
 
   // Specify the latency info to be passed to the browser side.
   new_sink_params.set_latency_tag(AudioDeviceFactory::GetSourceLatencyType(
       AudioDeviceFactory::AudioDeviceFactory::kSourceWebRtc));
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content