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Side by Side Diff: webrtc/api/peerconnectioninterface.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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58 // 3. Provide the remote offer to the new PeerConnection object by calling 58 // 3. Provide the remote offer to the new PeerConnection object by calling
59 // SetRemoteSessionDescription. 59 // SetRemoteSessionDescription.
60 // 4. Generate an answer to the remote offer by calling CreateAnswer and send it 60 // 4. Generate an answer to the remote offer by calling CreateAnswer and send it
61 // back to the remote peer. 61 // back to the remote peer.
62 // 5. Provide the local answer to the new PeerConnection by calling 62 // 5. Provide the local answer to the new PeerConnection by calling
63 // SetLocalSessionDescription with the answer. 63 // SetLocalSessionDescription with the answer.
64 // 6. Provide the remote ice candidates by calling AddIceCandidate. 64 // 6. Provide the remote ice candidates by calling AddIceCandidate.
65 // 7. Once a candidate have been found PeerConnection will call the observer 65 // 7. Once a candidate have been found PeerConnection will call the observer
66 // function OnIceCandidate. Send these candidates to the remote peer. 66 // function OnIceCandidate. Send these candidates to the remote peer.
67 67
68 #ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 68 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
69 #define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 69 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
70 70
71 #include <string> 71 #include <string>
72 #include <utility> 72 #include <utility>
73 #include <vector> 73 #include <vector>
74 74
75 #include "talk/app/webrtc/datachannelinterface.h" 75 #include "webrtc/api/datachannelinterface.h"
76 #include "talk/app/webrtc/dtlsidentitystore.h" 76 #include "webrtc/api/dtlsidentitystore.h"
77 #include "talk/app/webrtc/dtlsidentitystore.h" 77 #include "webrtc/api/dtlsidentitystore.h"
78 #include "talk/app/webrtc/dtmfsenderinterface.h" 78 #include "webrtc/api/dtmfsenderinterface.h"
79 #include "talk/app/webrtc/jsep.h" 79 #include "webrtc/api/jsep.h"
80 #include "talk/app/webrtc/mediastreaminterface.h" 80 #include "webrtc/api/mediastreaminterface.h"
81 #include "talk/app/webrtc/rtpreceiverinterface.h" 81 #include "webrtc/api/rtpreceiverinterface.h"
82 #include "talk/app/webrtc/rtpsenderinterface.h" 82 #include "webrtc/api/rtpsenderinterface.h"
83 #include "talk/app/webrtc/statstypes.h" 83 #include "webrtc/api/statstypes.h"
84 #include "talk/app/webrtc/umametrics.h" 84 #include "webrtc/api/umametrics.h"
85 #include "webrtc/base/fileutils.h" 85 #include "webrtc/base/fileutils.h"
86 #include "webrtc/base/network.h" 86 #include "webrtc/base/network.h"
87 #include "webrtc/base/rtccertificate.h" 87 #include "webrtc/base/rtccertificate.h"
88 #include "webrtc/base/socketaddress.h" 88 #include "webrtc/base/socketaddress.h"
89 #include "webrtc/base/sslstreamadapter.h" 89 #include "webrtc/base/sslstreamadapter.h"
90 #include "webrtc/p2p/base/portallocator.h" 90 #include "webrtc/p2p/base/portallocator.h"
91 91
92 namespace rtc { 92 namespace rtc {
93 class SSLIdentity; 93 class SSLIdentity;
94 class Thread; 94 class Thread;
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612 rtc::scoped_refptr<PeerConnectionFactoryInterface> 612 rtc::scoped_refptr<PeerConnectionFactoryInterface>
613 CreatePeerConnectionFactory( 613 CreatePeerConnectionFactory(
614 rtc::Thread* worker_thread, 614 rtc::Thread* worker_thread,
615 rtc::Thread* signaling_thread, 615 rtc::Thread* signaling_thread,
616 AudioDeviceModule* default_adm, 616 AudioDeviceModule* default_adm,
617 cricket::WebRtcVideoEncoderFactory* encoder_factory, 617 cricket::WebRtcVideoEncoderFactory* encoder_factory,
618 cricket::WebRtcVideoDecoderFactory* decoder_factory); 618 cricket::WebRtcVideoDecoderFactory* decoder_factory);
619 619
620 } // namespace webrtc 620 } // namespace webrtc
621 621
622 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 622 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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