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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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61 #include "webrtc/api/dtmfsenderinterface.h" | 61 #include "webrtc/api/dtmfsenderinterface.h" |
62 #include "webrtc/api/jsep.h" | 62 #include "webrtc/api/jsep.h" |
63 #include "webrtc/api/mediastreaminterface.h" | 63 #include "webrtc/api/mediastreaminterface.h" |
64 #include "webrtc/api/rtpreceiverinterface.h" | 64 #include "webrtc/api/rtpreceiverinterface.h" |
65 #include "webrtc/api/rtpsenderinterface.h" | 65 #include "webrtc/api/rtpsenderinterface.h" |
66 #include "webrtc/api/statstypes.h" | 66 #include "webrtc/api/statstypes.h" |
67 #include "webrtc/api/umametrics.h" | 67 #include "webrtc/api/umametrics.h" |
68 #include "webrtc/base/fileutils.h" | 68 #include "webrtc/base/fileutils.h" |
69 #include "webrtc/base/network.h" | 69 #include "webrtc/base/network.h" |
70 #include "webrtc/base/rtccertificate.h" | 70 #include "webrtc/base/rtccertificate.h" |
71 #include "webrtc/base/rtccertificategenerator.h" | |
72 #include "webrtc/base/socketaddress.h" | 71 #include "webrtc/base/socketaddress.h" |
73 #include "webrtc/base/sslstreamadapter.h" | 72 #include "webrtc/base/sslstreamadapter.h" |
74 #include "webrtc/media/base/mediachannel.h" | 73 #include "webrtc/media/base/mediachannel.h" |
75 #include "webrtc/p2p/base/portallocator.h" | 74 #include "webrtc/p2p/base/portallocator.h" |
76 | 75 |
77 namespace rtc { | 76 namespace rtc { |
78 class SSLIdentity; | 77 class SSLIdentity; |
79 class Thread; | 78 class Thread; |
80 } | 79 } |
81 | 80 |
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574 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. | 573 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. |
575 rtc::SSLProtocolVersion ssl_max_version; | 574 rtc::SSLProtocolVersion ssl_max_version; |
576 }; | 575 }; |
577 | 576 |
578 virtual void SetOptions(const Options& options) = 0; | 577 virtual void SetOptions(const Options& options) = 0; |
579 | 578 |
580 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | 579 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( |
581 const PeerConnectionInterface::RTCConfiguration& configuration, | 580 const PeerConnectionInterface::RTCConfiguration& configuration, |
582 const MediaConstraintsInterface* constraints, | 581 const MediaConstraintsInterface* constraints, |
583 std::unique_ptr<cricket::PortAllocator> allocator, | 582 std::unique_ptr<cricket::PortAllocator> allocator, |
584 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | |
585 PeerConnectionObserver* observer) = 0; | |
586 // TODO(hbos): To be removed in favor of the |cert_generator| version as soon | |
587 // as Chromium stops using this version. See bugs.webrtc.org/5707, | |
588 // bugs.webrtc.org/5708. | |
589 rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | |
590 const PeerConnectionInterface::RTCConfiguration& configuration, | |
591 const MediaConstraintsInterface* constraints, | |
592 std::unique_ptr<cricket::PortAllocator> allocator, | |
593 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 583 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
594 PeerConnectionObserver* observer) { | 584 PeerConnectionObserver* observer) = 0; |
595 return CreatePeerConnection( | |
596 configuration, | |
597 constraints, | |
598 std::move(allocator), | |
599 std::unique_ptr<rtc::RTCCertificateGeneratorInterface>( | |
600 dtls_identity_store ? new RTCCertificateGeneratorStoreWrapper( | |
601 std::move(dtls_identity_store)) : nullptr), | |
602 observer); | |
603 } | |
604 | 585 |
605 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | 586 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( |
606 const PeerConnectionInterface::RTCConfiguration& configuration, | 587 const PeerConnectionInterface::RTCConfiguration& configuration, |
607 std::unique_ptr<cricket::PortAllocator> allocator, | 588 std::unique_ptr<cricket::PortAllocator> allocator, |
608 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | 589 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
609 PeerConnectionObserver* observer) = 0; | 590 PeerConnectionObserver* observer) = 0; |
610 // TODO(hbos): To be removed in favor of the |cert_generator| version as soon | |
611 // as Chromium stops using this version. See bugs.webrtc.org/5707, | |
612 // bugs.webrtc.org/5708. | |
613 rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | |
614 const PeerConnectionInterface::RTCConfiguration& configuration, | |
615 std::unique_ptr<cricket::PortAllocator> allocator, | |
616 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | |
617 PeerConnectionObserver* observer) { | |
618 return CreatePeerConnection( | |
619 configuration, | |
620 std::move(allocator), | |
621 std::unique_ptr<rtc::RTCCertificateGeneratorInterface>( | |
622 dtls_identity_store ? new RTCCertificateGeneratorStoreWrapper( | |
623 std::move(dtls_identity_store)) : nullptr), | |
624 observer); | |
625 } | |
626 | 591 |
627 virtual rtc::scoped_refptr<MediaStreamInterface> | 592 virtual rtc::scoped_refptr<MediaStreamInterface> |
628 CreateLocalMediaStream(const std::string& label) = 0; | 593 CreateLocalMediaStream(const std::string& label) = 0; |
629 | 594 |
630 // Creates a AudioSourceInterface. | 595 // Creates a AudioSourceInterface. |
631 // |constraints| decides audio processing settings but can be NULL. | 596 // |constraints| decides audio processing settings but can be NULL. |
632 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 597 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
633 const cricket::AudioOptions& options) = 0; | 598 const cricket::AudioOptions& options) = 0; |
634 // Deprecated - use version above. | 599 // Deprecated - use version above. |
635 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 600 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
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736 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 701 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
737 cricket::WebRtcVideoDecoderFactory* decoder_factory) { | 702 cricket::WebRtcVideoDecoderFactory* decoder_factory) { |
738 return CreatePeerConnectionFactory( | 703 return CreatePeerConnectionFactory( |
739 worker_and_network_thread, worker_and_network_thread, signaling_thread, | 704 worker_and_network_thread, worker_and_network_thread, signaling_thread, |
740 default_adm, encoder_factory, decoder_factory); | 705 default_adm, encoder_factory, decoder_factory); |
741 } | 706 } |
742 | 707 |
743 } // namespace webrtc | 708 } // namespace webrtc |
744 | 709 |
745 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 710 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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