Feed the WebRTC APM with empty far-end frames for the number of frames skipped by OS.

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 167 matching lines @@
     : state_(UNINITIALIZED),
       source_render_frame_id_(source_render_frame_id),
       session_id_(session_id),
       signaling_thread_(signaling_thread),
       media_stream_(media_stream),
       source_(NULL),
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
       fifo_delay_milliseconds_(0),
+      skipped_frames_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO,
                    0,
                    16,
                    0),
       output_device_id_(device_id),
       security_origin_(security_origin),
       render_callback_count_(0) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_frame_id=%d, session_id=%d, effects=%i",
@@ skipping 225 matching lines @@
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   if (!source_)
     return 0;
 
   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);
+  skipped_frames_ += frames_skipped;
 
   if (audio_fifo_)
     audio_fifo_->Consume(audio_bus, audio_bus->frames());
   else
     SourceCallback(0, audio_bus);
 
   return (state_ == PLAYING) ? audio_bus->frames() : 0;
 }
 
 void WebRtcAudioRenderer::OnRenderError() {
@@ skipping 11 matching lines @@
            << 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,
+                      skipped_frames_,
                       &current_time_);
+  skipped_frames_ = 0;
 
   // 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();
 
   if (++render_callback_count_ == kNumCallbacksBetweenRenderTimeHistograms) {
     base::TimeDelta elapsed = base::TimeTicks::Now() - start_time;
     render_callback_count_ = 0;
     UMA_HISTOGRAM_TIMES("WebRTC.AudioRenderTimes", elapsed);
@@ skipping 176 matching lines @@
     sink_params_ = new_sink_params;
     fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
     if (new_audio_fifo.get())
       audio_fifo_ = new_audio_fifo.Pass();
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content