| /* |
| * libjingle |
| * Copyright 2012 Google Inc. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #ifndef TALK_APP_WEBRTC_PEERCONNECTION_H_ |
| #define TALK_APP_WEBRTC_PEERCONNECTION_H_ |
| |
| #include <string> |
| |
| #include "talk/app/webrtc/dtlsidentitystore.h" |
| #include "talk/app/webrtc/mediastreamsignaling.h" |
| #include "talk/app/webrtc/peerconnectionfactory.h" |
| #include "talk/app/webrtc/peerconnectioninterface.h" |
| #include "talk/app/webrtc/statscollector.h" |
| #include "talk/app/webrtc/streamcollection.h" |
| #include "talk/app/webrtc/webrtcsession.h" |
| #include "webrtc/base/scoped_ptr.h" |
| |
| namespace webrtc { |
| class MediaStreamHandlerContainer; |
| |
| typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration> |
| StunConfigurations; |
| typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration> |
| TurnConfigurations; |
| |
| // PeerConnectionImpl implements the PeerConnection interface. |
| // It uses MediaStreamSignaling and WebRtcSession to implement |
| // the PeerConnection functionality. |
| class PeerConnection : public PeerConnectionInterface, |
| public MediaStreamSignalingObserver, |
| public IceObserver, |
| public rtc::MessageHandler, |
| public sigslot::has_slots<> { |
| public: |
| explicit PeerConnection(PeerConnectionFactory* factory); |
| |
| bool Initialize( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| const MediaConstraintsInterface* constraints, |
| PortAllocatorFactoryInterface* allocator_factory, |
| rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
| PeerConnectionObserver* observer); |
| virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams(); |
| virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams(); |
| virtual bool AddStream(MediaStreamInterface* local_stream); |
| virtual void RemoveStream(MediaStreamInterface* local_stream); |
| |
| virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( |
| AudioTrackInterface* track); |
| |
| virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( |
| const std::string& label, |
| const DataChannelInit* config); |
| virtual bool GetStats(StatsObserver* observer, |
| webrtc::MediaStreamTrackInterface* track, |
| StatsOutputLevel level); |
| |
| virtual SignalingState signaling_state(); |
| |
| // TODO(bemasc): Remove ice_state() when callers are removed. |
| virtual IceState ice_state(); |
| virtual IceConnectionState ice_connection_state(); |
| virtual IceGatheringState ice_gathering_state(); |
| |
| virtual const SessionDescriptionInterface* local_description() const; |
| virtual const SessionDescriptionInterface* remote_description() const; |
| |
| // JSEP01 |
| virtual void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints); |
| virtual void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options); |
| virtual void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints); |
| virtual void SetLocalDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc); |
| virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc); |
| virtual void SetIceConnectionReceivingTimeout(int timeout_ms); |
| // TODO(mallinath) : Deprecated version, remove after all clients are updated. |
| virtual bool UpdateIce(const IceServers& configuration, |
| const MediaConstraintsInterface* constraints); |
| virtual bool UpdateIce( |
| const PeerConnectionInterface::RTCConfiguration& config); |
| virtual bool AddIceCandidate(const IceCandidateInterface* candidate); |
| |
| virtual void RegisterUMAObserver(UMAObserver* observer); |
| |
| virtual void Close(); |
| |
| protected: |
| virtual ~PeerConnection(); |
| |
| private: |
| // Implements MessageHandler. |
| virtual void OnMessage(rtc::Message* msg); |
| |
| // Implements MediaStreamSignalingObserver. |
| void OnAddRemoteStream(MediaStreamInterface* stream) override; |
| void OnRemoveRemoteStream(MediaStreamInterface* stream) override; |
| void OnAddDataChannel(DataChannelInterface* data_channel) override; |
| void OnAddRemoteAudioTrack(MediaStreamInterface* stream, |
| AudioTrackInterface* audio_track, |
| uint32 ssrc) override; |
| void OnAddRemoteVideoTrack(MediaStreamInterface* stream, |
| VideoTrackInterface* video_track, |
| uint32 ssrc) override; |
| void OnRemoveRemoteAudioTrack(MediaStreamInterface* stream, |
| AudioTrackInterface* audio_track) override; |
| void OnRemoveRemoteVideoTrack(MediaStreamInterface* stream, |
| VideoTrackInterface* video_track) override; |
| void OnAddLocalAudioTrack(MediaStreamInterface* stream, |
| AudioTrackInterface* audio_track, |
| uint32 ssrc) override; |
| void OnAddLocalVideoTrack(MediaStreamInterface* stream, |
| VideoTrackInterface* video_track, |
| uint32 ssrc) override; |
| void OnRemoveLocalAudioTrack(MediaStreamInterface* stream, |
| AudioTrackInterface* audio_track, |
| uint32 ssrc) override; |
| void OnRemoveLocalVideoTrack(MediaStreamInterface* stream, |
| VideoTrackInterface* video_track) override; |
| void OnRemoveLocalStream(MediaStreamInterface* stream) override; |
| |
| // Implements IceObserver |
| void OnIceConnectionChange(IceConnectionState new_state) override; |
| void OnIceGatheringChange(IceGatheringState new_state) override; |
| void OnIceCandidate(const IceCandidateInterface* candidate) override; |
| void OnIceComplete() override; |
| void OnIceConnectionReceivingChange(bool receiving) override; |
| |
| // Signals from WebRtcSession. |
| void OnSessionStateChange(cricket::BaseSession* session, |
| cricket::BaseSession::State state); |
| void ChangeSignalingState(SignalingState signaling_state); |
| |
| rtc::Thread* signaling_thread() const { |
| return factory_->signaling_thread(); |
| } |
| |
| void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer, |
| const std::string& error); |
| |
| bool IsClosed() const { |
| return signaling_state_ == PeerConnectionInterface::kClosed; |
| } |
| |
| // Storing the factory as a scoped reference pointer ensures that the memory |
| // in the PeerConnectionFactoryImpl remains available as long as the |
| // PeerConnection is running. It is passed to PeerConnection as a raw pointer. |
| // However, since the reference counting is done in the |
| // PeerConnectionFactoryInteface all instances created using the raw pointer |
| // will refer to the same reference count. |
| rtc::scoped_refptr<PeerConnectionFactory> factory_; |
| PeerConnectionObserver* observer_; |
| UMAObserver* uma_observer_; |
| SignalingState signaling_state_; |
| // TODO(bemasc): Remove ice_state_. |
| IceState ice_state_; |
| IceConnectionState ice_connection_state_; |
| IceGatheringState ice_gathering_state_; |
| |
| rtc::scoped_ptr<cricket::PortAllocator> port_allocator_; |
| rtc::scoped_ptr<WebRtcSession> session_; |
| rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_; |
| rtc::scoped_ptr<MediaStreamHandlerContainer> stream_handler_container_; |
| rtc::scoped_ptr<StatsCollector> stats_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ |
