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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /talk/app/webrtc/peerconnectionproxy.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'talk/app/webrtc/peerconnectionproxy.h')
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diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h
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+/*
+ * 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_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_