Drop WebRTC audio data if OS has skipped frames.

media/base/audio_pull_fifo.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 "media/base/audio_pull_fifo.h"
 
 #include <algorithm>
 
 #include "base/logging.h"
 #include "media/base/audio_bus.h"
@@ skipping 28 matching lines @@
     // Try to fulfill the request using what's available in the FIFO.
     frames_read =
         ReadFromFifo(destination, remaining_frames_to_provide, write_pos);
     write_pos += frames_read;
     remaining_frames_to_provide -= frames_read;
   }
 }
 
 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) {
   int frames = std::min(frames_to_provide, fifo_->frames() - fifo_index_);
   if (frames <= 0)
     return 0;
 
   for (int ch = 0; ch < fifo_->channels(); ++ch) {
     const float* src = fifo_->channel(ch) + fifo_index_;
     float* dest = destination->channel(ch) + write_pos;
     memcpy(dest, src, frames * sizeof(*src));
   }
 
   fifo_index_ += frames;
   return frames;
 }
 
 }  // namespace media