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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: Updated location for peerconnection_unittests.isolate Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * Copyright 2012 Google Inc.
4 * 3 *
5 * Redistribution and use in source and binary forms, with or without 4 * Use of this source code is governed by a BSD-style license
6 * modification, are permitted provided that the following conditions are met: 5 * that can be found in the LICENSE file in the root of the source
7 * 6 * tree. An additional intellectual property rights grant can be found
8 * 1. Redistributions of source code must retain the above copyright notice, 7 * in the file PATENTS. All contributing project authors may
9 * this list of conditions and the following disclaimer. 8 * be found in the AUTHORS file in the root of the source tree.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 9 */
27 10
28 // This file contains the PeerConnection interface as defined in 11 // This file contains the PeerConnection interface as defined in
29 // http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections. 12 // http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
30 // Applications must use this interface to implement peerconnection. 13 // Applications must use this interface to implement peerconnection.
31 // PeerConnectionFactory class provides factory methods to create 14 // PeerConnectionFactory class provides factory methods to create
32 // peerconnection, mediastream and media tracks objects. 15 // peerconnection, mediastream and media tracks objects.
33 // 16 //
34 // The Following steps are needed to setup a typical call using Jsep. 17 // The Following steps are needed to setup a typical call using Jsep.
35 // 1. Create a PeerConnectionFactoryInterface. Check constructors for more 18 // 1. Create a PeerConnectionFactoryInterface. Check constructors for more
(...skipping 22 matching lines...) Expand all
58 // 3. Provide the remote offer to the new PeerConnection object by calling 41 // 3. Provide the remote offer to the new PeerConnection object by calling
59 // SetRemoteSessionDescription. 42 // SetRemoteSessionDescription.
60 // 4. Generate an answer to the remote offer by calling CreateAnswer and send it 43 // 4. Generate an answer to the remote offer by calling CreateAnswer and send it
61 // back to the remote peer. 44 // back to the remote peer.
62 // 5. Provide the local answer to the new PeerConnection by calling 45 // 5. Provide the local answer to the new PeerConnection by calling
63 // SetLocalSessionDescription with the answer. 46 // SetLocalSessionDescription with the answer.
64 // 6. Provide the remote ice candidates by calling AddIceCandidate. 47 // 6. Provide the remote ice candidates by calling AddIceCandidate.
65 // 7. Once a candidate have been found PeerConnection will call the observer 48 // 7. Once a candidate have been found PeerConnection will call the observer
66 // function OnIceCandidate. Send these candidates to the remote peer. 49 // function OnIceCandidate. Send these candidates to the remote peer.
67 50
68 #ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 51 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
69 #define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 52 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
70 53
71 #include <string> 54 #include <string>
72 #include <utility> 55 #include <utility>
73 #include <vector> 56 #include <vector>
74 57
75 #include "talk/app/webrtc/datachannelinterface.h" 58 #include "webrtc/api/datachannelinterface.h"
76 #include "talk/app/webrtc/dtlsidentitystore.h" 59 #include "webrtc/api/dtlsidentitystore.h"
77 #include "talk/app/webrtc/dtmfsenderinterface.h" 60 #include "webrtc/api/dtlsidentitystore.h"
78 #include "talk/app/webrtc/dtlsidentitystore.h" 61 #include "webrtc/api/dtmfsenderinterface.h"
79 #include "talk/app/webrtc/jsep.h" 62 #include "webrtc/api/jsep.h"
80 #include "talk/app/webrtc/mediastreaminterface.h" 63 #include "webrtc/api/mediastreaminterface.h"
81 #include "talk/app/webrtc/rtpreceiverinterface.h" 64 #include "webrtc/api/rtpreceiverinterface.h"
82 #include "talk/app/webrtc/rtpsenderinterface.h" 65 #include "webrtc/api/rtpsenderinterface.h"
83 #include "talk/app/webrtc/statstypes.h" 66 #include "webrtc/api/statstypes.h"
84 #include "talk/app/webrtc/umametrics.h" 67 #include "webrtc/api/umametrics.h"
85 #include "webrtc/base/fileutils.h" 68 #include "webrtc/base/fileutils.h"
86 #include "webrtc/base/network.h" 69 #include "webrtc/base/network.h"
87 #include "webrtc/base/rtccertificate.h" 70 #include "webrtc/base/rtccertificate.h"
71 #include "webrtc/base/socketaddress.h"
88 #include "webrtc/base/sslstreamadapter.h" 72 #include "webrtc/base/sslstreamadapter.h"
89 #include "webrtc/base/socketaddress.h"
90 #include "webrtc/p2p/base/portallocator.h" 73 #include "webrtc/p2p/base/portallocator.h"
91 74
92 namespace rtc { 75 namespace rtc {
93 class SSLIdentity; 76 class SSLIdentity;
94 class Thread; 77 class Thread;
95 } 78 }
96 79
97 namespace cricket { 80 namespace cricket {
98 class WebRtcVideoDecoderFactory; 81 class WebRtcVideoDecoderFactory;
99 class WebRtcVideoEncoderFactory; 82 class WebRtcVideoEncoderFactory;
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620 rtc::scoped_refptr<PeerConnectionFactoryInterface> 603 rtc::scoped_refptr<PeerConnectionFactoryInterface>
621 CreatePeerConnectionFactory( 604 CreatePeerConnectionFactory(
622 rtc::Thread* worker_thread, 605 rtc::Thread* worker_thread,
623 rtc::Thread* signaling_thread, 606 rtc::Thread* signaling_thread,
624 AudioDeviceModule* default_adm, 607 AudioDeviceModule* default_adm,
625 cricket::WebRtcVideoEncoderFactory* encoder_factory, 608 cricket::WebRtcVideoEncoderFactory* encoder_factory,
626 cricket::WebRtcVideoDecoderFactory* decoder_factory); 609 cricket::WebRtcVideoDecoderFactory* decoder_factory);
627 610
628 } // namespace webrtc 611 } // namespace webrtc
629 612
630 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 613 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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