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// 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> 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(
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());
// 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());
stream->Start(std::move(desktop_capturer), transport_.get(),
video_encode_task_runner_);
stream->SetEventTimestampsSource(
event_dispatcher_->event_timestamps_source());
return std::move(stream);
}
std::unique_ptr<AudioStream> WebrtcConnectionToClient::StartAudioStream(
std::unique_ptr<AudioSource> audio_source) {
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) {
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();
break;
case Session::AUTHENTICATED: {
base::WeakPtr<WebrtcConnectionToClient> self = weak_factory_.GetWeakPtr();
event_handler_->OnConnectionAuthenticated();
// OnConnectionAuthenticated() call above may result in the connection
// being torn down.
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_->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) {
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);
}
} // namespace protocol
} // namespace remoting
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