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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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