Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_video_stream.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_stream.h"
 
 #include "base/logging.h"
 #include "base/single_thread_task_runner.h"
 #include "base/task_runner_util.h"
 #include "base/threading/thread_task_runner_handle.h"
@@ skipping 59 matching lines @@
 WebrtcVideoStream::~WebrtcVideoStream() {
   if (stream_) {
     for (const auto& track : stream_->GetVideoTracks()) {
       stream_->RemoveTrack(track.get());
     }
     peer_connection_->RemoveStream(stream_.get());
   }
   encode_task_runner_->DeleteSoon(FROM_HERE, encoder_.release());
 }
 
-bool WebrtcVideoStream::Start(
+void WebrtcVideoStream::Start(
     std::unique_ptr<webrtc::DesktopCapturer> desktop_capturer,
     WebrtcTransport* webrtc_transport,
     scoped_refptr<base::SingleThreadTaskRunner> encode_task_runner) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(webrtc_transport);
   DCHECK(desktop_capturer);
   DCHECK(encode_task_runner);
 
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_connection_factory(
       webrtc_transport->peer_connection_factory());
@@ skipping 14 matching lines @@
       webrtc::MediaConstraintsInterface::kMinFrameRate, 5);
 
   rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> src =
       peer_connection_factory->CreateVideoSource(new WebrtcDummyVideoCapturer(),
                                                  &video_constraints);
   rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
       peer_connection_factory->CreateVideoTrack(kVideoLabel, src);
 
   stream_ = peer_connection_factory->CreateLocalMediaStream(kStreamLabel);
 
-  if (!stream_->AddTrack(video_track.get()) ||
-      !peer_connection_->AddStream(stream_.get())) {
-    stream_ = nullptr;
-    peer_connection_ = nullptr;
-    return false;
-  }
+  // AddTrack() may fail only if there is another track with the same name,
+  // which is impossible because it's a brand new stream.
+  bool result = stream_->AddTrack(video_track.get());
+  DCHECK(result);
+
+  // AddStream() may fail if there is another stream with the same name or when
+  // the PeerConnection is closed, neither is expected.
+  result = peer_connection_->AddStream(stream_.get());
+  DCHECK(result);
 
   // Register for PLI requests.
   webrtc_transport_->video_encoder_factory()->SetKeyFrameRequestCallback(
       base::Bind(&PostTaskOnTaskRunner, base::ThreadTaskRunnerHandle::Get(),
                  base::Bind(&WebrtcVideoStream::SetKeyFrameRequest,
                             weak_factory_.GetWeakPtr())));
 
   // Register for target bitrate notifications.
   webrtc_transport_->video_encoder_factory()->SetTargetBitrateCallback(
       base::Bind(&PostTaskOnTaskRunnerWithParam<int>,
                  base::ThreadTaskRunnerHandle::Get(),
                  base::Bind(&WebrtcVideoStream::SetTargetBitrate,
                             weak_factory_.GetWeakPtr())));
 
   video_stats_dispatcher_.Init(webrtc_transport_->CreateOutgoingChannel(
                                    video_stats_dispatcher_.channel_name()),
                                this);
 
   scheduler_.reset(new WebrtcFrameSchedulerSimple());
-
-  return true;
 }
 
 void WebrtcVideoStream::Pause(bool pause) {
   DCHECK(thread_checker_.CalledOnValidThread());
   scheduler_->Pause(pause);
 }
 
 void WebrtcVideoStream::OnInputEventReceived(int64_t event_timestamp) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
@@ skipping 153 matching lines @@
     // TODO(sergeyu): Figure out how to measure send_pending time with WebRTC
     // and set it here.
     stats.send_pending_delay = base::TimeDelta();
 
     video_stats_dispatcher_.OnVideoFrameStats(result.frame_id, stats);
   }
 }
 
 }  // namespace protocol
 }  // namespace remoting