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

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..d038fe87566e8ce9e7b2a4b71144b07d1dbca07b 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 +13,10 @@ namespace content {
 
 WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
     webrtc::AudioTrackSinkInterface* sink)
-    : sink_(sink) {
+    : sink_(sink),
+      rechunker_(base::TimeDelta::FromMilliseconds(10),
+                 base::Bind(&WebRtcAudioSinkAdapter::DeliverRechunkedAudio,
+                            base::Unretained(this))) {
   DCHECK(sink);
 }
 
@@ -25,26 +30,33 @@ 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
+  // DeliverRechunkedAudio().
+  rechunker_.Push(audio_bus, estimated_capture_time - base::TimeTicks());
+}
+
+void WebRtcAudioSinkAdapter::DeliverRechunkedAudio(
+    const media::AudioBus& audio_bus,
+    base::TimeDelta timestamp) {
   // TODO(henrika): Remove this conversion once the interface in libjingle
   // supports float vectors.
   audio_bus.ToInterleaved(audio_bus.frames(),
                           sizeof(interleaved_data_[0]),
                           interleaved_data_.get());
+  // TODO(miu): Why doesn't a WebRTC sink care about the reference timestamp?
+  // How is AV sync achieved otherwise?
   sink_->OnData(interleaved_data_.get(),
                 16,
-                params_.sample_rate(),
+                rechunker_.sample_rate(),
                 audio_bus.channels(),
                 audio_bus.frames());
 }
 
-void WebRtcAudioSinkAdapter::OnSetFormat(
-    const media::AudioParameters& params) {
+void WebRtcAudioSinkAdapter::OnSetFormat(const media::AudioParameters& params) {
   DCHECK(params.IsValid());
-  params_ = params;
+  rechunker_.SetSampleRate(params.sample_rate());
   const int num_pcm16_data_elements =
-      params_.frames_per_buffer() * params_.channels();
+      rechunker_.output_frames() * params.channels();
   interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);
 }