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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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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 TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
69 #define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 69 #define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
70 | 70 |
71 #include <string> | 71 #include <string> |
| 72 #include <utility> |
72 #include <vector> | 73 #include <vector> |
73 | 74 |
74 #include "talk/app/webrtc/datachannelinterface.h" | 75 #include "talk/app/webrtc/datachannelinterface.h" |
75 #include "talk/app/webrtc/dtlsidentitystore.h" | 76 #include "talk/app/webrtc/dtlsidentitystore.h" |
76 #include "talk/app/webrtc/dtmfsenderinterface.h" | 77 #include "talk/app/webrtc/dtmfsenderinterface.h" |
77 #include "talk/app/webrtc/dtlsidentitystore.h" | 78 #include "talk/app/webrtc/dtlsidentitystore.h" |
78 #include "talk/app/webrtc/jsep.h" | 79 #include "talk/app/webrtc/jsep.h" |
79 #include "talk/app/webrtc/mediastreaminterface.h" | 80 #include "talk/app/webrtc/mediastreaminterface.h" |
80 #include "talk/app/webrtc/rtpreceiverinterface.h" | 81 #include "talk/app/webrtc/rtpreceiverinterface.h" |
81 #include "talk/app/webrtc/rtpsenderinterface.h" | 82 #include "talk/app/webrtc/rtpsenderinterface.h" |
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597 inline rtc::scoped_refptr<PeerConnectionInterface> | 598 inline rtc::scoped_refptr<PeerConnectionInterface> |
598 CreatePeerConnection( | 599 CreatePeerConnection( |
599 const PeerConnectionInterface::IceServers& servers, | 600 const PeerConnectionInterface::IceServers& servers, |
600 const MediaConstraintsInterface* constraints, | 601 const MediaConstraintsInterface* constraints, |
601 PortAllocatorFactoryInterface* allocator_factory, | 602 PortAllocatorFactoryInterface* allocator_factory, |
602 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 603 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
603 PeerConnectionObserver* observer) { | 604 PeerConnectionObserver* observer) { |
604 PeerConnectionInterface::RTCConfiguration rtc_config; | 605 PeerConnectionInterface::RTCConfiguration rtc_config; |
605 rtc_config.servers = servers; | 606 rtc_config.servers = servers; |
606 return CreatePeerConnection(rtc_config, constraints, allocator_factory, | 607 return CreatePeerConnection(rtc_config, constraints, allocator_factory, |
607 dtls_identity_store.Pass(), observer); | 608 std::move(dtls_identity_store), observer); |
608 } | 609 } |
609 | 610 |
610 virtual rtc::scoped_refptr<MediaStreamInterface> | 611 virtual rtc::scoped_refptr<MediaStreamInterface> |
611 CreateLocalMediaStream(const std::string& label) = 0; | 612 CreateLocalMediaStream(const std::string& label) = 0; |
612 | 613 |
613 // Creates a AudioSourceInterface. | 614 // Creates a AudioSourceInterface. |
614 // |constraints| decides audio processing settings but can be NULL. | 615 // |constraints| decides audio processing settings but can be NULL. |
615 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 616 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
616 const MediaConstraintsInterface* constraints) = 0; | 617 const MediaConstraintsInterface* constraints) = 0; |
617 | 618 |
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677 CreatePeerConnectionFactory( | 678 CreatePeerConnectionFactory( |
678 rtc::Thread* worker_thread, | 679 rtc::Thread* worker_thread, |
679 rtc::Thread* signaling_thread, | 680 rtc::Thread* signaling_thread, |
680 AudioDeviceModule* default_adm, | 681 AudioDeviceModule* default_adm, |
681 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 682 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
682 cricket::WebRtcVideoDecoderFactory* decoder_factory); | 683 cricket::WebRtcVideoDecoderFactory* decoder_factory); |
683 | 684 |
684 } // namespace webrtc | 685 } // namespace webrtc |
685 | 686 |
686 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 687 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
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