15583c19d7
The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545}
89 lines
4.1 KiB
C++
89 lines
4.1 KiB
C++
/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef WEBRTC_API_PEERCONNECTIONPROXY_H_
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#define WEBRTC_API_PEERCONNECTIONPROXY_H_
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#include "webrtc/api/peerconnectioninterface.h"
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#include "webrtc/api/proxy.h"
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namespace webrtc {
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// Define proxy for PeerConnectionInterface.
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BEGIN_PROXY_MAP(PeerConnection)
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PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
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local_streams)
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PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
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remote_streams)
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PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
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PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
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PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
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AddTrack,
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MediaStreamTrackInterface*,
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std::vector<MediaStreamInterface*>)
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PROXY_METHOD1(bool, RemoveTrack, RtpSenderInterface*)
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PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
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CreateDtmfSender, AudioTrackInterface*)
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PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
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CreateSender,
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const std::string&,
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const std::string&)
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PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
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GetSenders)
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PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
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GetReceivers)
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PROXY_METHOD3(bool, GetStats, StatsObserver*,
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MediaStreamTrackInterface*,
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StatsOutputLevel)
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PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
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CreateDataChannel, const std::string&, const DataChannelInit*)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description)
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PROXY_METHOD2(void, CreateOffer, CreateSessionDescriptionObserver*,
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const MediaConstraintsInterface*)
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PROXY_METHOD2(void, CreateAnswer, CreateSessionDescriptionObserver*,
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const MediaConstraintsInterface*)
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PROXY_METHOD2(void, SetLocalDescription, SetSessionDescriptionObserver*,
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SessionDescriptionInterface*)
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PROXY_METHOD2(void, SetRemoteDescription, SetSessionDescriptionObserver*,
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SessionDescriptionInterface*)
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PROXY_METHOD1(bool,
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SetConfiguration,
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const PeerConnectionInterface::RTCConfiguration&);
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PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*)
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PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
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PROXY_METHOD0(SignalingState, signaling_state)
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PROXY_METHOD0(IceState, ice_state)
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PROXY_METHOD0(IceConnectionState, ice_connection_state)
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PROXY_METHOD0(IceGatheringState, ice_gathering_state)
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PROXY_METHOD0(void, Close)
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END_PROXY()
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} // namespace webrtc
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#endif // WEBRTC_API_PEERCONNECTIONPROXY_H_
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