Drop WebRTC audio data if OS has skipped frames.

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index 640d8dc4ad689ba6db96781017231662b532d0be..e33b92039c78fd1e5e9a8643bee227ee8c732112 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -187,7 +187,6 @@ WebRtcAudioRenderer::WebRtcAudioRenderer(
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
-      fifo_delay_milliseconds_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO,
                    0,
@@ -434,6 +433,26 @@ int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
   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.
+  if (frames_skipped > 0) {
+    const uint32_t frames_per_10ms =
+        static_cast<uint32_t>(sink_params_.sample_rate() / 100);
+    if (!audio_fifo_ && frames_skipped != frames_per_10ms) {
+      audio_fifo_.reset(new media::AudioPullFifo(
+          sink_params_.channels(), frames_per_10ms,
+          base::Bind(&WebRtcAudioRenderer::SourceCallback,
+                     base::Unretained(this))));
+    }
+
+    scoped_ptr<media::AudioBus> drop_bus =
+        media::AudioBus::Create(audio_bus->channels(), frames_skipped);
+    if (audio_fifo_)
+      audio_fifo_->Consume(drop_bus.get(), drop_bus->frames());
+    else
+      SourceCallback(0, drop_bus.get());
+  }
+
+  // Pull the data we will deliver.
   if (audio_fifo_)
     audio_fifo_->Consume(audio_bus, audio_bus->frames());
   else
@@ -457,7 +476,9 @@ void WebRtcAudioRenderer::SourceCallback(
            << audio_bus->frames() << ")";
 
   int output_delay_milliseconds = audio_delay_milliseconds_;
-  output_delay_milliseconds += fifo_delay_milliseconds_;
+  output_delay_milliseconds += fifo_frame_delay *
+                               base::Time::kMillisecondsPerSecond /
+                               sink_params_.sample_rate();
   DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;
 
   // We need to keep render data for the |source_| regardless of |state_|,
@@ -606,8 +627,7 @@ void WebRtcAudioRenderer::PrepareSink() {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected", sample_rate);
   }
 
-  // Set up audio parameters for the source, i.e., the WebRTC client.
-
+  // Set up audio parameters for the source, i.e. the WebRTC client.
   // The WebRTC client only supports multiples of 10ms as buffer size where
   // 10ms is preferred for lowest possible delay.
   const int frames_per_10ms = (sample_rate / 100);
@@ -627,34 +647,24 @@ void WebRtcAudioRenderer::PrepareSink() {
   // Create a FIFO if re-buffering is required to match the source input with
   // the sink request. The source acts as provider here and the sink as
   // consumer.
-  int new_fifo_delay_milliseconds = 0;
-  scoped_ptr<media::AudioPullFifo> new_audio_fifo;
-  if (source_params.frames_per_buffer() !=
-      new_sink_params.frames_per_buffer()) {
+  const bool different_source_sink_frames =
+      source_params.frames_per_buffer() != new_sink_params.frames_per_buffer();
+  if (different_source_sink_frames) {
     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
              << " to " << new_sink_params.frames_per_buffer();
-    new_audio_fifo.reset(new media::AudioPullFifo(
-        source_params.channels(), source_params.frames_per_buffer(),
-        base::Bind(&WebRtcAudioRenderer::SourceCallback,
-                   base::Unretained(this))));
-
-    if (new_sink_params.frames_per_buffer() >
-        source_params.frames_per_buffer()) {
-      int frame_duration_milliseconds =
-          base::Time::kMillisecondsPerSecond /
-          static_cast<double>(source_params.sample_rate());
-      new_fifo_delay_milliseconds = (new_sink_params.frames_per_buffer() -
-                                     source_params.frames_per_buffer()) *
-                                    frame_duration_milliseconds;
-    }
   }
 
   {
     base::AutoLock lock(lock_);
+    if ((!audio_fifo_ && different_source_sink_frames) ||
+        (audio_fifo_ &&
+         audio_fifo_->SizeInFrames() != source_params.frames_per_buffer())) {
+      audio_fifo_.reset(new media::AudioPullFifo(
+          source_params.channels(), source_params.frames_per_buffer(),
+          base::Bind(&WebRtcAudioRenderer::SourceCallback,
+                     base::Unretained(this))));
+    }
     sink_params_ = new_sink_params;
-    fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
-    if (new_audio_fifo.get())
-      audio_fifo_ = std::move(new_audio_fifo);
   }
 
   sink_->Initialize(new_sink_params, this);