Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_connection_to_client.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_connection_to_client.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/location.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "net/base/io_buffer.h"
 #include "remoting/codec/video_encoder.h"
 #include "remoting/codec/webrtc_video_encoder_vpx.h"
 #include "remoting/protocol/audio_source.h"
 #include "remoting/protocol/audio_stream.h"
 #include "remoting/protocol/clipboard_stub.h"
 #include "remoting/protocol/host_control_dispatcher.h"
 #include "remoting/protocol/host_event_dispatcher.h"
 #include "remoting/protocol/host_stub.h"
 #include "remoting/protocol/input_stub.h"
 #include "remoting/protocol/message_pipe.h"
 #include "remoting/protocol/transport_context.h"
+#include "remoting/protocol/webrtc_audio_stream.h"
 #include "remoting/protocol/webrtc_transport.h"
 #include "remoting/protocol/webrtc_video_stream.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 #include "third_party/webrtc/api/test/fakeconstraints.h"
 
 namespace remoting {
 namespace protocol {
 
 // Currently the network thread is also used as worker thread for webrtc.
 //
 // TODO(sergeyu): Figure out if we would benefit from using a separate
 // thread as a worker thread.
 WebrtcConnectionToClient::WebrtcConnectionToClient(
     std::unique_ptr<protocol::Session> session,
     scoped_refptr<protocol::TransportContext> transport_context,
-    scoped_refptr<base::SingleThreadTaskRunner> video_encode_task_runner)
+    scoped_refptr<base::SingleThreadTaskRunner> video_encode_task_runner,
+    scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
     : transport_(
           new WebrtcTransport(jingle_glue::JingleThreadWrapper::current(),
                               transport_context,
                               this)),
       session_(std::move(session)),
       video_encode_task_runner_(video_encode_task_runner),
+      audio_task_runner_(audio_task_runner),
       control_dispatcher_(new HostControlDispatcher()),
       event_dispatcher_(new HostEventDispatcher()),
       weak_factory_(this) {
   session_->SetEventHandler(this);
   session_->SetTransport(transport_.get());
 }
 
 WebrtcConnectionToClient::~WebrtcConnectionToClient() {}
 
 void WebrtcConnectionToClient::SetEventHandler(
@@ skipping 10 matching lines @@
 void WebrtcConnectionToClient::Disconnect(ErrorCode error) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   // This should trigger OnConnectionClosed() event and this object
   // may be destroyed as the result.
   session_->Close(error);
 }
 
 std::unique_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
     std::unique_ptr<webrtc::DesktopCapturer> desktop_capturer) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(transport_);
+
   std::unique_ptr<WebrtcVideoStream> stream(new WebrtcVideoStream());
-  if (!stream->Start(std::move(desktop_capturer), transport_.get(),
-                     video_encode_task_runner_)) {
-    return nullptr;
-  }
+  stream->Start(std::move(desktop_capturer), transport_.get(),
+                video_encode_task_runner_);
   return std::move(stream);
 }
 
 std::unique_ptr<AudioStream> WebrtcConnectionToClient::StartAudioStream(
     std::unique_ptr<AudioSource> audio_source) {
-  NOTIMPLEMENTED();
-  return nullptr;
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(transport_);
+
+  std::unique_ptr<WebrtcAudioStream> stream(new WebrtcAudioStream());
+  stream->Start(audio_task_runner_, std::move(audio_source), transport_.get());
+  return std::move(stream);
 }
 
 // Return pointer to ClientStub.
 ClientStub* WebrtcConnectionToClient::client_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return control_dispatcher_.get();
 }
 
 void WebrtcConnectionToClient::set_clipboard_stub(
     protocol::ClipboardStub* clipboard_stub) {
@@ skipping 36 matching lines @@
       if (self)
         event_handler_->CreateMediaStreams();
       break;
     }
 
     case Session::CLOSED:
     case Session::FAILED:
       control_dispatcher_.reset();
       event_dispatcher_.reset();
       transport_->Close(state == Session::CLOSED ? OK : session_->error());
+      transport_.reset();
       event_handler_->OnConnectionClosed(
           state == Session::CLOSED ? OK : session_->error());
       break;
   }
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportConnecting() {
+  DCHECK(thread_checker_.CalledOnValidThread());
   // Create outgoing control channel. |event_dispatcher_| is initialized later
   // because event channel is expected to be created by the client.
   control_dispatcher_->Init(
       transport_->CreateOutgoingChannel(control_dispatcher_->channel_name()),
       this);
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportConnected() {
   DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportError(ErrorCode error) {
   DCHECK(thread_checker_.CalledOnValidThread());
   Disconnect(error);
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportIncomingDataChannel(
     const std::string& name,
     std::unique_ptr<MessagePipe> pipe) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   if (name == event_dispatcher_->channel_name() &&
       !event_dispatcher_->is_connected()) {
     event_dispatcher_->set_on_input_event_callback(
         base::Bind(&WebrtcConnectionToClient::OnInputEventReceived,
                    base::Unretained(this)));
     event_dispatcher_->Init(std::move(pipe), this);
   }
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportMediaStreamAdded(
     scoped_refptr<webrtc::MediaStreamInterface> stream) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   LOG(WARNING) << "The client created an unexpected media stream.";
 }
 
 void WebrtcConnectionToClient::OnWebrtcTransportMediaStreamRemoved(
-    scoped_refptr<webrtc::MediaStreamInterface> stream) {}
+    scoped_refptr<webrtc::MediaStreamInterface> stream) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+}
 
 void WebrtcConnectionToClient::OnChannelInitialized(
     ChannelDispatcherBase* channel_dispatcher) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (control_dispatcher_ && control_dispatcher_->is_connected() &&
       event_dispatcher_ && event_dispatcher_->is_connected()) {
     event_handler_->OnConnectionChannelsConnected();
   }
 }
 
 void WebrtcConnectionToClient::OnChannelClosed(
     ChannelDispatcherBase* channel_dispatcher) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   LOG(ERROR) << "Channel " << channel_dispatcher->channel_name()
              << " was closed unexpectedly.";
   Disconnect(INCOMPATIBLE_PROTOCOL);
 }
 
 void WebrtcConnectionToClient::OnInputEventReceived(int64_t timestamp) {
   DCHECK(thread_checker_.CalledOnValidThread());
   event_handler_->OnInputEventReceived(timestamp);
 }
 
 }  // namespace protocol
 }  // namespace remoting