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

content/renderer/media/webrtc/webrtc_media_stream_adapter.cc

Index: content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
diff --git a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
index 3c34d794dded57cd712436cbad564dca909e531b..577bb0742281dc596dd2a0d45bc7aad7709d6766 100644
--- a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
@@ -5,11 +5,12 @@
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 
 #include "base/logging.h"
-#include "content/renderer/media/media_stream_audio_source.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"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
+#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebString.h"
@@ -27,12 +28,12 @@ WebRtcMediaStreamAdapter::WebRtcMediaStreamAdapter(
   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
   web_stream_.audioTracks(audio_tracks);
   for (blink::WebMediaStreamTrack& audio_track : audio_tracks)
-    CreateAudioTrack(audio_track);
+    AddAudioSinkToTrack(audio_track);
 
   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
   web_stream_.videoTracks(video_tracks);
   for (blink::WebMediaStreamTrack& video_track : video_tracks)
-    CreateVideoTrack(video_track);
+    AddVideoSinkToTrack(video_track);
 
   MediaStream* const native_stream = MediaStream::GetMediaStream(web_stream_);
   native_stream->AddObserver(this);
@@ -46,67 +47,90 @@ WebRtcMediaStreamAdapter::~WebRtcMediaStreamAdapter() {
 void WebRtcMediaStreamAdapter::TrackAdded(
     const blink::WebMediaStreamTrack& track) {
   if (track.source().getType() == blink::WebMediaStreamSource::TypeAudio)
-    CreateAudioTrack(track);
+    AddAudioSinkToTrack(track);
   else
-    CreateVideoTrack(track);
+    AddVideoSinkToTrack(track);
 }
 
 void WebRtcMediaStreamAdapter::TrackRemoved(
     const blink::WebMediaStreamTrack& track) {
   const std::string track_id = track.id().utf8();
   if (track.source().getType() == blink::WebMediaStreamSource::TypeAudio) {
-    webrtc_media_stream_->RemoveTrack(
-        webrtc_media_stream_->FindAudioTrack(track_id));
+    scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
+        make_scoped_refptr(
+            webrtc_media_stream_->FindAudioTrack(track_id).get());
+    webrtc_media_stream_->RemoveTrack(webrtc_track.get());
+
+    for (auto it = audio_sinks_.begin(); it != audio_sinks_.end(); ++it) {
+      if ((*it)->webrtc_audio_track() == webrtc_track.get()) {
+        if (auto* media_stream_track = MediaStreamAudioTrack::From(track))
+          media_stream_track->RemoveSink(it->get());
+        audio_sinks_.erase(it);
+        break;
+      }
+    }
   } else {
     DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo);
     scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
-        webrtc_media_stream_->FindVideoTrack(track_id).get();
+        make_scoped_refptr(
+            webrtc_media_stream_->FindVideoTrack(track_id).get());
     webrtc_media_stream_->RemoveTrack(webrtc_track.get());
 
-    for (ScopedVector<MediaStreamVideoWebRtcSink>::iterator it =
-             video_adapters_.begin(); it != video_adapters_.end(); ++it) {
+    for (auto it = video_sinks_.begin(); it != video_sinks_.end(); ++it) {
       if ((*it)->webrtc_video_track() == webrtc_track.get()) {
-        video_adapters_.erase(it);
+        video_sinks_.erase(it);
         break;
       }
     }
   }
 }
 
-void WebRtcMediaStreamAdapter::CreateAudioTrack(
+void WebRtcMediaStreamAdapter::AddAudioSinkToTrack(
     const blink::WebMediaStreamTrack& track) {
-  DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeAudio);
-  // A media stream is connected to a peer connection, enable the
-  // peer connection mode for the sources.
   MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track);
   if (!native_track) {
     DLOG(ERROR) << "No native track for blink audio track.";
     return;
   }
 
-  webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter();
-  if (!audio_track) {
-    DLOG(ERROR) << "Audio track doesn't support webrtc.";
-    return;
-  }
-
-  if (native_track->is_local_track()) {
-    const blink::WebMediaStreamSource& source = track.source();
-    MediaStreamAudioSource* audio_source = MediaStreamAudioSource::From(source);
-    if (audio_source && audio_source->audio_capturer())
-      audio_source->audio_capturer()->EnablePeerConnectionMode();
+  WebRtcAudioSink* audio_sink;
+  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());
   }
 
-  webrtc_media_stream_->AddTrack(audio_track);
+  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::CreateVideoTrack(
+void WebRtcMediaStreamAdapter::AddVideoSinkToTrack(
     const blink::WebMediaStreamTrack& track) {
   DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo);
-  MediaStreamVideoWebRtcSink* adapter =
+  MediaStreamVideoWebRtcSink* video_sink =
       new MediaStreamVideoWebRtcSink(track, factory_);
-  video_adapters_.push_back(adapter);
-  webrtc_media_stream_->AddTrack(adapter->webrtc_video_track());
+  video_sinks_.push_back(
+      std::unique_ptr<MediaStreamVideoWebRtcSink>(video_sink));
+  webrtc_media_stream_->AddTrack(video_sink->webrtc_video_track());
 }
 
 }  // namespace content