Drop WebRTC audio data if OS has skipped frames.

media/base/audio_pull_fifo.cc

Index: media/base/audio_pull_fifo.cc
diff --git a/media/base/audio_pull_fifo.cc b/media/base/audio_pull_fifo.cc
index cf25142d904f588d11164db363df4d0568ef0757..03aa001a38a98744258ed2a0e48636396f93b120 100644
--- a/media/base/audio_pull_fifo.cc
+++ b/media/base/audio_pull_fifo.cc
@@ -46,6 +46,10 @@ void AudioPullFifo::Consume(AudioBus* destination, int frames_to_consume) {
 
 void AudioPullFifo::Clear() { fifo_index_ = fifo_->frames(); }
 
+int AudioPullFifo::SizeInFrames() const {
+  return fifo_->frames();
+}
+
 int AudioPullFifo::ReadFromFifo(AudioBus* destination,
                                 int frames_to_provide,
                                 int write_pos) {