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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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