| Index: remoting/protocol/webrtc_transport.h
|
| diff --git a/remoting/protocol/webrtc_transport.h b/remoting/protocol/webrtc_transport.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..cbc6458d8bc550c2aa9ebf0a118ecb19a0b4a004
|
| --- /dev/null
|
| +++ b/remoting/protocol/webrtc_transport.h
|
| @@ -0,0 +1,124 @@
|
| +// 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/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_;
|
| +
|
| + 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_
|
|
|