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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.
#ifndef REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
#define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
#include "base/macros.h"
#include "base/memory/scoped_ptr.h"
#include "base/memory/scoped_vector.h"
#include "base/memory/weak_ptr.h"
#include "base/threading/thread.h"
#include "base/threading/thread_checker.h"
#include "base/timer/timer.h"
#include "remoting/protocol/transport.h"
#include "remoting/protocol/webrtc_data_stream_adapter.h"
#include "remoting/signaling/signal_strategy.h"
#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
namespace webrtc {
class FakeAudioDeviceModule;
} // namespace webrtc
namespace remoting {
namespace protocol {
class WebrtcTransport : public Transport,
public webrtc::PeerConnectionObserver {
public:
WebrtcTransport(
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
port_allocator_factory,
TransportRole role,
scoped_refptr<base::SingleThreadTaskRunner> worker_task_runner);
~WebrtcTransport() override;
// Transport interface.
void Start(EventHandler* event_handler,
Authenticator* authenticator) override;
bool ProcessTransportInfo(buzz::XmlElement* transport_info) override;
DatagramChannelFactory* GetDatagramChannelFactory() override;
StreamChannelFactory* GetStreamChannelFactory() override;
StreamChannelFactory* GetMultiplexedChannelFactory() override;
private:
void DoStart(rtc::Thread* worker_thread);
void OnLocalSessionDescriptionCreated(
scoped_ptr<webrtc::SessionDescriptionInterface> description,
const std::string& error);
void OnLocalDescriptionSet(bool success, const std::string& error);
void OnRemoteDescriptionSet(bool success, const std::string& error);
// webrtc::PeerConnectionObserver interface.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(webrtc::MediaStreamInterface* stream) override;
void OnRemoveStream(webrtc::MediaStreamInterface* stream) override;
void OnDataChannel(webrtc::DataChannelInterface* data_channel) override;
void OnRenegotiationNeeded() override;
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
void EnsurePendingTransportInfoMessage();
void SendTransportInfo();
void AddPendingCandidatesIfPossible();
void Close(ErrorCode error);
base::ThreadChecker thread_checker_;
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
port_allocator_factory_;
TransportRole role_;
EventHandler* event_handler_ = nullptr;
scoped_refptr<base::SingleThreadTaskRunner> worker_task_runner_;
scoped_ptr<webrtc::FakeAudioDeviceModule> fake_audio_device_module_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
scoped_ptr<buzz::XmlElement> pending_transport_info_message_;
base::OneShotTimer transport_info_timer_;
ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_;
std::list<rtc::scoped_refptr<webrtc::MediaStreamInterface>>
unclaimed_streams_;
WebrtcDataStreamAdapter data_stream_adapter_;
base::WeakPtrFactory<WebrtcTransport> weak_factory_;
DISALLOW_COPY_AND_ASSIGN(WebrtcTransport);
};
class WebrtcTransportFactory : public TransportFactory {
public:
WebrtcTransportFactory(
SignalStrategy* signal_strategy,
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
port_allocator_factory,
TransportRole role);
~WebrtcTransportFactory() override;
// TransportFactory interface.
scoped_ptr<Transport> CreateTransport() override;
private:
SignalStrategy* signal_strategy_;
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
port_allocator_factory_;
TransportRole role_;
base::Thread worker_thread_;
DISALLOW_COPY_AND_ASSIGN(WebrtcTransportFactory);
};
} // namespace protocol
} // namespace remoting
#endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
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