MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc

Index: content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
diff --git a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
index 2679aff9648a438574fc06749be4af57b5e724fb..e2cfd01ae958ab0f298153695e88314d1c5c6d93 100644
--- a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
@@ -2,6 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include "base/bind.h"
+#include "base/bind_helpers.h"
 #include "base/logging.h"
 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
 #include "media/base/audio_bus.h"
@@ -11,7 +13,9 @@ namespace content {
 
 WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
     webrtc::AudioTrackSinkInterface* sink)
-    : sink_(sink) {
+    : sink_(sink),
+      fifo_(base::Bind(&WebRtcAudioSinkAdapter::DeliverRebufferedAudio,
+                       base::Unretained(this))) {
   DCHECK(sink);
 }
 
@@ -25,8 +29,17 @@ bool WebRtcAudioSinkAdapter::IsEqual(
 
 void WebRtcAudioSinkAdapter::OnData(const media::AudioBus& audio_bus,
                                     base::TimeTicks estimated_capture_time) {
-  DCHECK_EQ(audio_bus.frames(), params_.frames_per_buffer());
-  DCHECK_EQ(audio_bus.channels(), params_.channels());
+  // The following will result in zero, one, or multiple synchronous calls to
+  // DeliverRebufferedAudio().
+  fifo_.Push(audio_bus);
+}
+
+void WebRtcAudioSinkAdapter::DeliverRebufferedAudio(
+    const media::AudioBus& audio_bus,
+    int frame_delay) {
+  // TODO(miu): Why doesn't a WebRTC sink care about reference time passed to
+  // OnData(), and the |frame_delay| here?  How is AV sync achieved otherwise?
+
   // TODO(henrika): Remove this conversion once the interface in libjingle
   // supports float vectors.
   audio_bus.ToInterleaved(audio_bus.frames(),
@@ -43,6 +56,7 @@ void WebRtcAudioSinkAdapter::OnSetFormat(
     const media::AudioParameters& params) {
   DCHECK(params.IsValid());
   params_ = params;
+  fifo_.Reset(params_.frames_per_buffer());
   const int num_pcm16_data_elements =
       params_.frames_per_buffer() * params_.channels();
   interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);