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Side by Side Diff: webrtc/api/peerconnectioninterface.h

Issue 2020633002: Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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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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