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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 |
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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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