MediaStream audio rendering: Bypass audio processing for non-WebRTC cases.

media/base/audio_shifter.h

Index: media/base/audio_shifter.h
diff --git a/media/base/audio_shifter.h b/media/base/audio_shifter.h
index e7bc502af3ea9abd3a1fbb4dada563f15f0d21fd..015626922cfe59ca05e31498e42e0a24a92401c0 100644
--- a/media/base/audio_shifter.h
+++ b/media/base/audio_shifter.h
@@ -56,10 +56,13 @@ class MEDIA_EXPORT AudioShifter {
   AudioShifter(base::TimeDelta max_buffer_size,
                base::TimeDelta clock_accuracy,
                base::TimeDelta adjustment_time,
-               size_t rate,
+               int rate,
                int channels);
   ~AudioShifter();
 
+  int sample_rate() const { return rate_; }
+  int channels() const { return channels_; }
+
   // Push Audio into the shifter. All inputs must have the same number of
   // channels, but bus size can vary. The playout time can be noisy and
   // does not have to line up perfectly with the number of samples pushed
@@ -79,9 +82,6 @@ class MEDIA_EXPORT AudioShifter {
   // calculate playout_time would be now + audio pipeline delay.
   void Pull(AudioBus* output, base::TimeTicks playout_time);
 
-  // Flush audio (but leave timing info)
-  void Flush();
-
 private:
   void Zero(AudioBus* output);
   void ResamplerCallback(int frame_delay, AudioBus* destination);
@@ -100,7 +100,8 @@ private:
   const base::TimeDelta max_buffer_size_;
   const base::TimeDelta clock_accuracy_;
   const base::TimeDelta adjustment_time_;
-  const size_t rate_;
+  const int rate_;
+  const int channels_;
 
   // The clock smoothers are used to smooth out timestamps
   // and adjust for drift and inaccurate clocks.