webrtc_connection_to_client.cc | Explore in Territory

// Copyright 2015 The Chromium Authors
// 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/functional/bind.h"
#include "base/location.h"
#include "base/logging.h"
#include "base/task/single_thread_task_runner.h"
#include "components/webrtc/thread_wrapper.h"
#include "net/base/io_buffer.h"
#include "remoting/base/logging.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/desktop_capturer.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_encoder_factory.h"
#include "remoting/protocol/webrtc_video_stream.h"
#include "third_party/webrtc/api/media_stream_interface.h"
#include "third_party/webrtc/api/peer_connection_interface.h"
#include "third_party/webrtc/api/sctp_transport_interface.h"

namespace remoting::protocol {

namespace {

const char kVideoStatsStreamLabel[] = …;

}  // namespace

// Currently the network thread is also used as the 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> audio_task_runner)
    : … { … }

WebrtcConnectionToClient::~WebrtcConnectionToClient() { … }

void WebrtcConnectionToClient::SetEventHandler(
    ConnectionToClient::EventHandler* event_handler) { … }

protocol::Session* WebrtcConnectionToClient::session() { … }

void WebrtcConnectionToClient::Disconnect(ErrorCode error) { … }

std::unique_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
    webrtc::ScreenId screen_id,
    std::unique_ptr<DesktopCapturer> desktop_capturer) { … }

std::unique_ptr<AudioStream> WebrtcConnectionToClient::StartAudioStream(
    std::unique_ptr<AudioSource> audio_source) { … }

// Return pointer to ClientStub.
ClientStub* WebrtcConnectionToClient::client_stub() { … }

void WebrtcConnectionToClient::set_clipboard_stub(
    protocol::ClipboardStub* clipboard_stub) { … }

void WebrtcConnectionToClient::set_host_stub(protocol::HostStub* host_stub) { … }

void WebrtcConnectionToClient::set_input_stub(protocol::InputStub* input_stub) { … }

void WebrtcConnectionToClient::ApplySessionOptions(
    const SessionOptions& options) { … }

void WebrtcConnectionToClient::ApplyNetworkSettings(
    const NetworkSettings& settings) { … }

PeerConnectionControls* WebrtcConnectionToClient::peer_connection_controls() { … }

WebrtcEventLogData* WebrtcConnectionToClient::rtc_event_log() { … }

void WebrtcConnectionToClient::OnSessionStateChange(Session::State state) { … }

void WebrtcConnectionToClient::OnWebrtcTransportConnecting() { … }

void WebrtcConnectionToClient::OnWebrtcTransportConnected() { … }

void WebrtcConnectionToClient::OnWebrtcTransportError(ErrorCode error) { … }

void WebrtcConnectionToClient::OnWebrtcTransportProtocolChanged() { … }

void WebrtcConnectionToClient::OnWebrtcTransportIncomingDataChannel(
    const std::string& name,
    std::unique_ptr<MessagePipe> pipe) { … }

void WebrtcConnectionToClient::OnWebrtcTransportMediaStreamAdded(
    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) { … }

void WebrtcConnectionToClient::OnWebrtcTransportMediaStreamRemoved(
    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) { … }

void WebrtcConnectionToClient::OnWebrtcTransportRouteChanged(
    const TransportRoute& route) { … }

void WebrtcConnectionToClient::OnChannelInitialized(
    ChannelDispatcherBase* channel_dispatcher) { … }

void WebrtcConnectionToClient::OnChannelClosed(
    ChannelDispatcherBase* channel_dispatcher) { … }

bool WebrtcConnectionToClient::allChannelsConnected() { … }

}  // namespace remoting::protocol
chromium/remoting/protocol/webrtc_connection_to_client.cc
