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Side by Side Diff: talk/app/webrtc/peerconnectioninterface.h

Issue 1460043002: Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Restore the Pass methods Created 5 years ago
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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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