Introduce media::AudioPushFifo and a couple of use cases (and clean-ups).

content/renderer/media/webaudio_capturer_source.h

Index: content/renderer/media/webaudio_capturer_source.h
diff --git a/content/renderer/media/webaudio_capturer_source.h b/content/renderer/media/webaudio_capturer_source.h
index b0ee262ccd7c3c06135b9252b3b394db8c0dc7dd..d5b3bb5b554514d8d75ee909bf8ab23f63914279 100644
--- a/content/renderer/media/webaudio_capturer_source.h
+++ b/content/renderer/media/webaudio_capturer_source.h
@@ -11,9 +11,11 @@
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
+#include "base/time/time.h"
 #include "media/audio/audio_parameters.h"
+#include "media/base/audio_bus.h"
 #include "media/base/audio_capturer_source.h"
-#include "media/base/audio_fifo.h"
+#include "media/base/audio_push_fifo.h"
 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
@@ -58,6 +60,12 @@ class WebAudioCapturerSource
   ~WebAudioCapturerSource() override;
 
  private:
+  // Called by AudioPushFifo zero or more times during the call to
+  // consumeAudio().  Delivers audio data with the required buffer size to the
+  // track.
+  void DeliverRebufferedAudio(const media::AudioBus& audio_bus,
+                              int frame_delay);
+
   // Removes this object from a blink::WebMediaStreamSource with which it
   // might be registered. The goal is to avoid dangling pointers.
   void removeFromBlinkSource();
@@ -75,14 +83,18 @@ class WebAudioCapturerSource
   // Flag to help notify the |track_| when the audio format has changed.
   bool audio_format_changed_;
 
-  // Wraps data coming from HandleCapture().
+  // A wrapper used for providing audio to |fifo_|.
   scoped_ptr<media::AudioBus> wrapper_bus_;
 
-  // Bus for reading from FIFO and calling the CaptureCallback.
-  scoped_ptr<media::AudioBus> capture_bus_;
+  // Takes in the audio data passed to consumeAudio() and re-buffers it into 10
+  // ms chunks for the track.  This ensures each chunk of audio delivered to the
+  // track has the required buffer size, regardless of the amount of audio
+  // provided via each consumeAudio() call.
+  media::AudioPushFifo fifo_;
 
-  // Handles mismatch between WebAudio buffer size and WebRTC.
-  scoped_ptr<media::AudioFifo> fifo_;
+  // Used to pass the reference timestamp between DeliverDecodedAudio() and
+  // DeliverRebufferedAudio().
+  base::TimeTicks current_reference_time_;
 
   // Synchronizes HandleCapture() with AudioCapturerSource calls.
   base::Lock lock_;