webrtc/talk/app/webrtc/peerconnection.h
henrike@webrtc.org 28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00

193 lines
8.6 KiB
C++

/*
* 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/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/peerconnectionfactory.h"
#include "talk/app/webrtc/statscollector.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/webrtcsession.h"
#include "talk/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 talk_base::MessageHandler,
public sigslot::has_slots<> {
public:
explicit PeerConnection(PeerConnectionFactory* factory);
bool Initialize(const PeerConnectionInterface::IceServers& configuration,
const MediaConstraintsInterface* constraints,
webrtc::PortAllocatorFactoryInterface* allocator_factory,
PeerConnectionObserver* observer);
virtual talk_base::scoped_refptr<StreamCollectionInterface> local_streams();
virtual talk_base::scoped_refptr<StreamCollectionInterface> remote_streams();
virtual bool AddStream(MediaStreamInterface* local_stream,
const MediaConstraintsInterface* constraints);
virtual void RemoveStream(MediaStreamInterface* local_stream);
virtual talk_base::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track);
virtual talk_base::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const DataChannelInit* config);
virtual bool GetStats(StatsObserver* observer,
webrtc::MediaStreamTrackInterface* track);
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 CreateAnswer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc);
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc);
virtual bool UpdateIce(const IceServers& configuration,
const MediaConstraintsInterface* constraints);
virtual bool AddIceCandidate(const IceCandidateInterface* candidate);
virtual void Close();
protected:
virtual ~PeerConnection();
private:
// Implements MessageHandler.
virtual void OnMessage(talk_base::Message* msg);
// Implements MediaStreamSignalingObserver.
virtual void OnAddRemoteStream(MediaStreamInterface* stream) OVERRIDE;
virtual void OnRemoveRemoteStream(MediaStreamInterface* stream) OVERRIDE;
virtual void OnAddDataChannel(DataChannelInterface* data_channel) OVERRIDE;
virtual void OnAddRemoteAudioTrack(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32 ssrc) OVERRIDE;
virtual void OnAddRemoteVideoTrack(MediaStreamInterface* stream,
VideoTrackInterface* video_track,
uint32 ssrc) OVERRIDE;
virtual void OnRemoveRemoteAudioTrack(
MediaStreamInterface* stream,
AudioTrackInterface* audio_track) OVERRIDE;
virtual void OnRemoveRemoteVideoTrack(
MediaStreamInterface* stream,
VideoTrackInterface* video_track) OVERRIDE;
virtual void OnAddLocalAudioTrack(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32 ssrc) OVERRIDE;
virtual void OnAddLocalVideoTrack(MediaStreamInterface* stream,
VideoTrackInterface* video_track,
uint32 ssrc) OVERRIDE;
virtual void OnRemoveLocalAudioTrack(
MediaStreamInterface* stream,
AudioTrackInterface* audio_track) OVERRIDE;
virtual void OnRemoveLocalVideoTrack(
MediaStreamInterface* stream,
VideoTrackInterface* video_track) OVERRIDE;
virtual void OnRemoveLocalStream(MediaStreamInterface* stream);
// Implements IceObserver
virtual void OnIceConnectionChange(IceConnectionState new_state);
virtual void OnIceGatheringChange(IceGatheringState new_state);
virtual void OnIceCandidate(const IceCandidateInterface* candidate);
virtual void OnIceComplete();
// Signals from WebRtcSession.
void OnSessionStateChange(cricket::BaseSession* session,
cricket::BaseSession::State state);
void ChangeSignalingState(SignalingState signaling_state);
bool DoInitialize(const StunConfigurations& stun_config,
const TurnConfigurations& turn_config,
const MediaConstraintsInterface* constraints,
webrtc::PortAllocatorFactoryInterface* allocator_factory,
PeerConnectionObserver* observer);
talk_base::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.
talk_base::scoped_refptr<PeerConnectionFactory> factory_;
PeerConnectionObserver* observer_;
SignalingState signaling_state_;
// TODO(bemasc): Remove ice_state_.
IceState ice_state_;
IceConnectionState ice_connection_state_;
IceGatheringState ice_gathering_state_;
talk_base::scoped_ptr<cricket::PortAllocator> port_allocator_;
talk_base::scoped_ptr<WebRtcSession> session_;
talk_base::scoped_ptr<MediaStreamSignaling> mediastream_signaling_;
talk_base::scoped_ptr<MediaStreamHandlerContainer> stream_handler_container_;
StatsCollector stats_;
};
} // namespace webrtc
#endif // TALK_APP_WEBRTC_PEERCONNECTION_H_