Revert of 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 7708b50ccff296332a857b6cf8729cdb11cc9d13..3c34d794dded57cd712436cbad564dca909e531b 100644
--- a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
@@ -5,12 +5,11 @@
 #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"
@@ -28,12 +27,12 @@
   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
   web_stream_.audioTracks(audio_tracks);
   for (blink::WebMediaStreamTrack& audio_track : audio_tracks)
-    AddAudioSinkToTrack(audio_track);
+    CreateAudioTrack(audio_track);
 
   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
   web_stream_.videoTracks(video_tracks);
   for (blink::WebMediaStreamTrack& video_track : video_tracks)
-    AddVideoSinkToTrack(video_track);
+    CreateVideoTrack(video_track);
 
   MediaStream* const native_stream = MediaStream::GetMediaStream(web_stream_);
   native_stream->AddObserver(this);
@@ -42,110 +41,72 @@
 WebRtcMediaStreamAdapter::~WebRtcMediaStreamAdapter() {
   MediaStream* const native_stream = MediaStream::GetMediaStream(web_stream_);
   native_stream->RemoveObserver(this);
-
-  blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
-  web_stream_.audioTracks(audio_tracks);
-  for (blink::WebMediaStreamTrack& audio_track : audio_tracks)
-    TrackRemoved(audio_track);
-  DCHECK(audio_sinks_.empty());
-  blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
-  web_stream_.videoTracks(video_tracks);
-  for (blink::WebMediaStreamTrack& video_track : video_tracks)
-    TrackRemoved(video_track);
-  DCHECK(video_sinks_.empty());
 }
 
 void WebRtcMediaStreamAdapter::TrackAdded(
     const blink::WebMediaStreamTrack& track) {
   if (track.source().getType() == blink::WebMediaStreamSource::TypeAudio)
-    AddAudioSinkToTrack(track);
+    CreateAudioTrack(track);
   else
-    AddVideoSinkToTrack(track);
+    CreateVideoTrack(track);
 }
 
 void WebRtcMediaStreamAdapter::TrackRemoved(
     const blink::WebMediaStreamTrack& track) {
   const std::string track_id = track.id().utf8();
   if (track.source().getType() == blink::WebMediaStreamSource::TypeAudio) {
-    scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
-        make_scoped_refptr(
-            webrtc_media_stream_->FindAudioTrack(track_id).get());
-    if (!webrtc_track)
-      return;
-    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;
-      }
-    }
+    webrtc_media_stream_->RemoveTrack(
+        webrtc_media_stream_->FindAudioTrack(track_id));
   } else {
     DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo);
     scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
-        make_scoped_refptr(
-            webrtc_media_stream_->FindVideoTrack(track_id).get());
-    if (!webrtc_track)
-      return;
+        webrtc_media_stream_->FindVideoTrack(track_id).get();
     webrtc_media_stream_->RemoveTrack(webrtc_track.get());
 
-    for (auto it = video_sinks_.begin(); it != video_sinks_.end(); ++it) {
+    for (ScopedVector<MediaStreamVideoWebRtcSink>::iterator it =
+             video_adapters_.begin(); it != video_adapters_.end(); ++it) {
       if ((*it)->webrtc_video_track() == webrtc_track.get()) {
-        video_sinks_.erase(it);
+        video_adapters_.erase(it);
         break;
       }
     }
   }
 }
 
-void WebRtcMediaStreamAdapter::AddAudioSinkToTrack(
+void WebRtcMediaStreamAdapter::CreateAudioTrack(
     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;
   }
 
-  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::AudioTrackInterface* audio_track = native_track->GetAudioAdapter();
+  if (!audio_track) {
+    DLOG(ERROR) << "Audio track doesn't support webrtc.";
+    return;
   }
 
-  audio_sinks_.push_back(std::unique_ptr<WebRtcAudioSink>(audio_sink));
-  native_track->AddSink(audio_sink);
-  webrtc_media_stream_->AddTrack(audio_sink->webrtc_audio_track());
+  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();
+  }
+
+  webrtc_media_stream_->AddTrack(audio_track);
 }
 
-void WebRtcMediaStreamAdapter::AddVideoSinkToTrack(
+void WebRtcMediaStreamAdapter::CreateVideoTrack(
     const blink::WebMediaStreamTrack& track) {
   DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo);
-  MediaStreamVideoWebRtcSink* video_sink =
+  MediaStreamVideoWebRtcSink* adapter =
       new MediaStreamVideoWebRtcSink(track, factory_);
-  video_sinks_.push_back(
-      std::unique_ptr<MediaStreamVideoWebRtcSink>(video_sink));
-  webrtc_media_stream_->AddTrack(video_sink->webrtc_video_track());
+  video_adapters_.push_back(adapter);
+  webrtc_media_stream_->AddTrack(adapter->webrtc_video_track());
 }
 
 }  // namespace content