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

Issue 1930463002: Replace scoped_ptr with unique_ptr in webrtc/api/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 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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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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