Remove WebRtc audio source references for local audio tracks.

content/renderer/media/webrtc/webrtc_media_stream_adapter.cc

 // Copyright 2014 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 "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 
 #include "base/logging.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h"
@@ skipping 90 matching lines @@
 }
 
 void WebRtcMediaStreamAdapter::AddAudioSinkToTrack(
     const blink::WebMediaStreamTrack& track) {
   MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track);
   if (!native_track) {
     DLOG(ERROR) << "No native track for blink audio track.";
     return;
   }
 
-  WebRtcAudioSink* audio_sink;
+  // Non-WebRtc remote sources and local sources do not provide an instance of
+  // the webrtc::AudioSourceInterface, and also do not need references to the
+  // audio level calculator or audio processor passed to the sink.
+  webrtc::AudioSourceInterface* source_interface = nullptr;
+  WebRtcAudioSink* audio_sink = new WebRtcAudioSink(
+      track.id().utf8(), source_interface,
+      factory_->GetWebRtcSignalingThread());
+
   if (auto* media_stream_source = ProcessedLocalAudioSource::From(
           MediaStreamAudioSource::From(track.source()))) {
-    audio_sink = new WebRtcAudioSink(
-        track.id().utf8(), media_stream_source->rtc_source(),
-        factory_->GetWebRtcSignalingThread());
     audio_sink->SetLevel(media_stream_source->audio_level());
     // The sink only grabs stats from the audio processor. Stats are only
     // available if audio processing is turned on. Therefore, only provide the
     // sink a reference to the processor if audio processing is turned on.
     if (auto processor = media_stream_source->audio_processor()) {
       if (processor && processor->has_audio_processing())
         audio_sink->SetAudioProcessor(processor);
     }
-  } else {
-    // Remote sources and other non-WebRtc local sources do not provide an
-    // instance of the webrtc::AudioSourceInterface, and also do not need
-    // references to the audio level calculator or audio processor passed to the
-    // sink.
-    webrtc::AudioSourceInterface* source_interface = nullptr;
-    audio_sink = new WebRtcAudioSink(
-        track.id().utf8(), source_interface,
-        factory_->GetWebRtcSignalingThread());
   }
 
   audio_sinks_.push_back(std::unique_ptr<WebRtcAudioSink>(audio_sink));
   native_track->AddSink(audio_sink);
   webrtc_media_stream_->AddTrack(audio_sink->webrtc_audio_track());
 }
 
 void WebRtcMediaStreamAdapter::AddVideoSinkToTrack(
     const blink::WebMediaStreamTrack& track) {
   DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo);
   MediaStreamVideoWebRtcSink* video_sink =
       new MediaStreamVideoWebRtcSink(track, factory_);
   video_sinks_.push_back(
       std::unique_ptr<MediaStreamVideoWebRtcSink>(video_sink));
   webrtc_media_stream_->AddTrack(video_sink->webrtc_video_track());
 }
 
 }  // namespace content