Add WebrtcConnectionToClient.

remoting/protocol/webrtc_connection_to_client.cc

Index: remoting/protocol/webrtc_connection_to_client.cc
diff --git a/remoting/protocol/webrtc_connection_to_client.cc b/remoting/protocol/webrtc_connection_to_client.cc
new file mode 100644
index 0000000000000000000000000000000000000000..d36ac2485639269203bb903fd94ac08dc3df9ece
--- /dev/null
+++ b/remoting/protocol/webrtc_connection_to_client.cc
@@ -0,0 +1,203 @@
+// 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 "base/bind.h"
+#include "base/location.h"
+#include "net/base/io_buffer.h"
+#include "remoting/codec/video_encoder.h"
+#include "remoting/codec/video_encoder_verbatim.h"
+#include "remoting/codec/video_encoder_vpx.h"
+#include "remoting/protocol/audio_writer.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/webrtc_transport.h"
+#include "remoting/protocol/webrtc_video_capturer_adapter.h"
+#include "remoting/protocol/webrtc_video_stream.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
+#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h"
+#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h"
+
+namespace remoting {
+namespace protocol {
+
+const char kStreamLabel[] = "screen_stream";
+const char kVideoLabel[] = "screen_video";
+
+WebrtcConnectionToClient::WebrtcConnectionToClient(
+    scoped_ptr<protocol::Session> session)
+    : session_(session.Pass()) {
+  session_->SetEventHandler(this);
+}
+
+WebrtcConnectionToClient::~WebrtcConnectionToClient() {}
+
+void WebrtcConnectionToClient::SetEventHandler(
+    ConnectionToClient::EventHandler* event_handler) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  event_handler_ = event_handler;
+}
+
+protocol::Session* WebrtcConnectionToClient::session() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return session_.get();
+}
+
+void WebrtcConnectionToClient::Disconnect(ErrorCode error) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  control_dispatcher_.reset();
+  event_dispatcher_.reset();
+
+  // This should trigger OnConnectionClosed() event and this object
+  // may be destroyed as the result.
+  session_->Close(error);
+}
+
+void WebrtcConnectionToClient::OnInputEventReceived(int64_t timestamp) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  event_handler_->OnInputEventReceived(this, timestamp);
+}
+
+scoped_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
+    scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
+  // TODO(sergeyu): Reconsider Transport interface and how it's used here.
+  WebrtcTransport* transport = session_->GetTransport()->AsWebrtcTransport();
+  CHECK(transport);
+
+  scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter(
+      new WebrtcVideoCapturerAdapter(desktop_capturer.Pass()));
+
+  // Set video stream constraints.
+  webrtc::FakeConstraints video_constraints;
+  video_constraints.AddMandatory(
+      webrtc::MediaConstraintsInterface::kMinFrameRate, 5);
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
+      transport->peer_connection_factory()->CreateVideoTrack(
+          kVideoLabel,
+          transport->peer_connection_factory()->CreateVideoSource(
+              video_capturer_adapter.release(), &video_constraints));
+
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> video_stream =
+      transport->peer_connection_factory()->CreateLocalMediaStream(
+          kStreamLabel);
+
+  if (!video_stream->AddTrack(video_track) ||
+      !transport->peer_connection()->AddStream(video_stream)) {
+    return nullptr;
+  }
+
+  scoped_ptr<VideoStream> result(
+      new WebrtcVideoStream(transport->peer_connection(), video_stream));
+  return result.Pass();
+}
+
+AudioStub* WebrtcConnectionToClient::audio_stub() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return nullptr;
+}
+
+// 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) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  control_dispatcher_->set_clipboard_stub(clipboard_stub);
+}
+
+void WebrtcConnectionToClient::set_host_stub(protocol::HostStub* host_stub) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  control_dispatcher_->set_host_stub(host_stub);
+}
+
+void WebrtcConnectionToClient::set_input_stub(protocol::InputStub* input_stub) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  event_dispatcher_->set_input_stub(input_stub);
+}
+
+void WebrtcConnectionToClient::OnSessionStateChange(Session::State state) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  DCHECK(event_handler_);
+  switch(state) {
+    case Session::INITIALIZING:
+    case Session::CONNECTING:
+    case Session::ACCEPTING:
+    case Session::ACCEPTED:
+      // Don't care about these events.
+      break;
+    case Session::AUTHENTICATING:
+      event_handler_->OnConnectionAuthenticating(this);
+      break;
+    case Session::AUTHENTICATED: {
+      // Initialize channels.
+      control_dispatcher_.reset(new HostControlDispatcher());
+      control_dispatcher_->Init(
+          session_.get(),
+          ChannelConfig(ChannelConfig::TRANSPORT_STREAM, kDefaultStreamVersion,
+                        ChannelConfig::CODEC_UNDEFINED),
+          this);
+
+      event_dispatcher_.reset(new HostEventDispatcher());
+      event_dispatcher_->Init(
+          session_.get(),
+          ChannelConfig(ChannelConfig::TRANSPORT_STREAM, kDefaultStreamVersion,
+                        ChannelConfig::CODEC_UNDEFINED),
+          this);
+      event_dispatcher_->set_on_input_event_callback(base::Bind(
+          &ConnectionToClient::OnInputEventReceived, base::Unretained(this)));
+
+      // Notify the handler after initializing the channels, so that
+      // ClientSession can get a client clipboard stub.
+      event_handler_->OnConnectionAuthenticated(this);
+      break;
+    }
+
+    case Session::CONNECTED:
+      event_handler_->OnConnectionChannelsConnected(this);
+      break;
+
+    case Session::CLOSED:
+    case Session::FAILED:
+      control_dispatcher_.reset();
+      event_dispatcher_.reset();
+      event_handler_->OnConnectionClosed(
+          this, state == Session::CLOSED ? OK : session_->error());
+      break;
+  }
+}
+
+void WebrtcConnectionToClient::OnSessionRouteChange(
+    const std::string& channel_name,
+    const TransportRoute& route) {
+  event_handler_->OnRouteChange(this, channel_name, route);
+}
+
+void WebrtcConnectionToClient::OnChannelInitialized(
+    ChannelDispatcherBase* channel_dispatcher) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void WebrtcConnectionToClient::OnChannelError(
+    ChannelDispatcherBase* channel_dispatcher,
+    ErrorCode error) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  LOG(ERROR) << "Failed to connect channel "
+             << channel_dispatcher->channel_name();
+  session_->Close(CHANNEL_CONNECTION_ERROR);
+}
+
+}  // namespace protocol
+}  // namespace remoting