| /* |
| * 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_PEERCONNECTIONPROXY_H_ |
| #define TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_ |
| |
| #include "talk/app/webrtc/peerconnectioninterface.h" |
| #include "talk/app/webrtc/proxy.h" |
| |
| namespace webrtc { |
| |
| // Define proxy for PeerConnectionInterface. |
| BEGIN_PROXY_MAP(PeerConnection) |
| PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, |
| local_streams) |
| PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, |
| remote_streams) |
| PROXY_METHOD1(bool, AddStream, MediaStreamInterface*) |
| PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*) |
| PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>, |
| CreateDtmfSender, AudioTrackInterface*) |
| PROXY_METHOD1(rtc::scoped_refptr<RtpSenderInterface>, |
| CreateSender, |
| const std::string&) |
| PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>, |
| GetSenders) |
| PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>, |
| GetReceivers) |
| PROXY_METHOD3(bool, GetStats, StatsObserver*, |
| MediaStreamTrackInterface*, |
| StatsOutputLevel) |
| PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>, |
| CreateDataChannel, const std::string&, const DataChannelInit*) |
| PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description) |
| PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description) |
| PROXY_METHOD2(void, CreateOffer, CreateSessionDescriptionObserver*, |
| const MediaConstraintsInterface*) |
| PROXY_METHOD2(void, CreateAnswer, CreateSessionDescriptionObserver*, |
| const MediaConstraintsInterface*) |
| PROXY_METHOD2(void, SetLocalDescription, SetSessionDescriptionObserver*, |
| SessionDescriptionInterface*) |
| PROXY_METHOD2(void, SetRemoteDescription, SetSessionDescriptionObserver*, |
| SessionDescriptionInterface*) |
| PROXY_METHOD1(bool, |
| SetConfiguration, |
| const PeerConnectionInterface::RTCConfiguration&); |
| PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*) |
| PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*) |
| PROXY_METHOD0(SignalingState, signaling_state) |
| PROXY_METHOD0(IceState, ice_state) |
| PROXY_METHOD0(IceConnectionState, ice_connection_state) |
| PROXY_METHOD0(IceGatheringState, ice_gathering_state) |
| PROXY_METHOD0(void, Close) |
| END_PROXY() |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_ |