Add audio support in WebrtcConnectionToHost, audio unittest

remoting/protocol/webrtc_video_renderer_adapter.cc

 // Copyright 2015 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 "remoting/protocol/webrtc_video_renderer_adapter.h"
 
 #include <memory>
 #include <string>
 #include <utility>
 
@@ skipping 60 matching lines @@
   video_tracks[0]->RemoveSink(this);
 }
 
 void WebrtcVideoRendererAdapter::SetMediaStream(
     scoped_refptr<webrtc::MediaStreamInterface> media_stream) {
   DCHECK_EQ(media_stream->label(), label());
 
   media_stream_ = std::move(media_stream);
 
   webrtc::VideoTrackVector video_tracks = media_stream_->GetVideoTracks();
-  if (video_tracks.empty()) {
-    LOG(ERROR) << "Received media stream with no video tracks.";
-    return;
-  }
+
+  // Caller must verify that the media stream contains video tracks.
+  DCHECK(!video_tracks.empty());
 
   if (video_tracks.size() > 1U) {
     LOG(WARNING) << "Received media stream with multiple video tracks.";
   }
 
   video_tracks[0]->AddOrUpdateSink(this, rtc::VideoSinkWants());
 }
 
 void WebrtcVideoRendererAdapter::SetVideoStatsChannel(
     std::unique_ptr<MessagePipe> message_pipe) {
@@ skipping 146 matching lines @@
   frame_stats.client_stats = *client_stats;
   host_stats_queue_.pop_front();
   FrameStatsConsumer* frame_stats_consumer =
       video_renderer_->GetFrameStatsConsumer();
   if (frame_stats_consumer)
     frame_stats_consumer->OnVideoFrameStats(frame_stats);
 }
 
 }  // namespace protocol
 }  // namespace remoting