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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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44 // back to the remote peer. | 44 // back to the remote peer. |
45 // 5. Provide the local answer to the new PeerConnection by calling | 45 // 5. Provide the local answer to the new PeerConnection by calling |
46 // SetLocalSessionDescription with the answer. | 46 // SetLocalSessionDescription with the answer. |
47 // 6. Provide the remote ice candidates by calling AddIceCandidate. | 47 // 6. Provide the remote ice candidates by calling AddIceCandidate. |
48 // 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 |
49 // function OnIceCandidate. Send these candidates to the remote peer. | 49 // function OnIceCandidate. Send these candidates to the remote peer. |
50 | 50 |
51 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 51 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
52 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 52 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
53 | 53 |
| 54 #include <memory> |
54 #include <string> | 55 #include <string> |
55 #include <utility> | 56 #include <utility> |
56 #include <vector> | 57 #include <vector> |
57 | 58 |
58 #include "webrtc/api/datachannelinterface.h" | 59 #include "webrtc/api/datachannelinterface.h" |
59 #include "webrtc/api/dtlsidentitystore.h" | 60 #include "webrtc/api/dtlsidentitystore.h" |
60 #include "webrtc/api/dtlsidentitystore.h" | 61 #include "webrtc/api/dtlsidentitystore.h" |
61 #include "webrtc/api/dtmfsenderinterface.h" | 62 #include "webrtc/api/dtmfsenderinterface.h" |
62 #include "webrtc/api/jsep.h" | 63 #include "webrtc/api/jsep.h" |
63 #include "webrtc/api/mediastreaminterface.h" | 64 #include "webrtc/api/mediastreaminterface.h" |
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584 // supported by both ends will be used for the connection, i.e. if one | 585 // supported by both ends will be used for the connection, i.e. if one |
585 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. | 586 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. |
586 rtc::SSLProtocolVersion ssl_max_version; | 587 rtc::SSLProtocolVersion ssl_max_version; |
587 }; | 588 }; |
588 | 589 |
589 virtual void SetOptions(const Options& options) = 0; | 590 virtual void SetOptions(const Options& options) = 0; |
590 | 591 |
591 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | 592 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( |
592 const PeerConnectionInterface::RTCConfiguration& configuration, | 593 const PeerConnectionInterface::RTCConfiguration& configuration, |
593 const MediaConstraintsInterface* constraints, | 594 const MediaConstraintsInterface* constraints, |
594 rtc::scoped_ptr<cricket::PortAllocator> allocator, | 595 std::unique_ptr<cricket::PortAllocator> allocator, |
595 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 596 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
596 PeerConnectionObserver* observer) = 0; | 597 PeerConnectionObserver* observer) = 0; |
597 | 598 |
598 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | 599 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( |
599 const PeerConnectionInterface::RTCConfiguration& configuration, | 600 const PeerConnectionInterface::RTCConfiguration& configuration, |
600 rtc::scoped_ptr<cricket::PortAllocator> allocator, | 601 std::unique_ptr<cricket::PortAllocator> allocator, |
601 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 602 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
602 PeerConnectionObserver* observer) = 0; | 603 PeerConnectionObserver* observer) = 0; |
603 | 604 |
604 virtual rtc::scoped_refptr<MediaStreamInterface> | 605 virtual rtc::scoped_refptr<MediaStreamInterface> |
605 CreateLocalMediaStream(const std::string& label) = 0; | 606 CreateLocalMediaStream(const std::string& label) = 0; |
606 | 607 |
607 // Creates a AudioSourceInterface. | 608 // Creates a AudioSourceInterface. |
608 // |constraints| decides audio processing settings but can be NULL. | 609 // |constraints| decides audio processing settings but can be NULL. |
609 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 610 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
610 const cricket::AudioOptions& options) = 0; | 611 const cricket::AudioOptions& options) = 0; |
611 // Deprecated - use version above. | 612 // Deprecated - use version above. |
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694 CreatePeerConnectionFactory( | 695 CreatePeerConnectionFactory( |
695 rtc::Thread* worker_thread, | 696 rtc::Thread* worker_thread, |
696 rtc::Thread* signaling_thread, | 697 rtc::Thread* signaling_thread, |
697 AudioDeviceModule* default_adm, | 698 AudioDeviceModule* default_adm, |
698 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 699 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
699 cricket::WebRtcVideoDecoderFactory* decoder_factory); | 700 cricket::WebRtcVideoDecoderFactory* decoder_factory); |
700 | 701 |
701 } // namespace webrtc | 702 } // namespace webrtc |
702 | 703 |
703 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 704 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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