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

Issue 1268363002: DtlsIdentityStoreInterface added. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Making trybots compile (hopefully) Created 5 years, 4 months 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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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 <vector> 72 #include <vector>
73 73
74 #include "talk/app/webrtc/datachannelinterface.h" 74 #include "talk/app/webrtc/datachannelinterface.h"
75 #include "talk/app/webrtc/dtlsidentitystore.h"
75 #include "talk/app/webrtc/dtmfsenderinterface.h" 76 #include "talk/app/webrtc/dtmfsenderinterface.h"
76 #include "talk/app/webrtc/jsep.h" 77 #include "talk/app/webrtc/jsep.h"
77 #include "talk/app/webrtc/mediastreaminterface.h" 78 #include "talk/app/webrtc/mediastreaminterface.h"
78 #include "talk/app/webrtc/statstypes.h" 79 #include "talk/app/webrtc/statstypes.h"
79 #include "talk/app/webrtc/umametrics.h" 80 #include "talk/app/webrtc/umametrics.h"
80 #include "webrtc/base/fileutils.h" 81 #include "webrtc/base/fileutils.h"
81 #include "webrtc/base/network.h" 82 #include "webrtc/base/network.h"
82 #include "webrtc/base/sslstreamadapter.h" 83 #include "webrtc/base/sslstreamadapter.h"
83 #include "webrtc/base/socketaddress.h" 84 #include "webrtc/base/socketaddress.h"
84 85
85 namespace rtc { 86 namespace rtc {
86 class SSLIdentity; 87 class SSLIdentity;
87 class Thread; 88 class Thread;
88 } 89 }
89 90
90 namespace cricket { 91 namespace cricket {
91 class PortAllocator; 92 class PortAllocator;
92 class WebRtcVideoDecoderFactory; 93 class WebRtcVideoDecoderFactory;
93 class WebRtcVideoEncoderFactory; 94 class WebRtcVideoEncoderFactory;
94 } 95 }
95 96
96 namespace webrtc { 97 namespace webrtc {
97 class AudioDeviceModule; 98 class AudioDeviceModule;
99 class DTLSIdentityRequestObserver;
hbos 2015/08/05 11:41:15 Oops, unnecessary, will remove.
98 class MediaConstraintsInterface; 100 class MediaConstraintsInterface;
99 101
100 // MediaStream container interface. 102 // MediaStream container interface.
101 class StreamCollectionInterface : public rtc::RefCountInterface { 103 class StreamCollectionInterface : public rtc::RefCountInterface {
102 public: 104 public:
103 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find. 105 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
104 virtual size_t count() = 0; 106 virtual size_t count() = 0;
105 virtual MediaStreamInterface* at(size_t index) = 0; 107 virtual MediaStreamInterface* at(size_t index) = 0;
106 virtual MediaStreamInterface* find(const std::string& label) = 0; 108 virtual MediaStreamInterface* find(const std::string& label) = 0;
107 virtual MediaStreamTrackInterface* FindAudioTrack( 109 virtual MediaStreamTrackInterface* FindAudioTrack(
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459 // After this method is called, the port allocator should consider loopback 461 // After this method is called, the port allocator should consider loopback
460 // network interfaces as well. 462 // network interfaces as well.
461 virtual void SetNetworkIgnoreMask(int network_ignore_mask) { 463 virtual void SetNetworkIgnoreMask(int network_ignore_mask) {
462 } 464 }
463 465
464 protected: 466 protected:
465 PortAllocatorFactoryInterface() {} 467 PortAllocatorFactoryInterface() {}
466 ~PortAllocatorFactoryInterface() {} 468 ~PortAllocatorFactoryInterface() {}
467 }; 469 };
468 470
469 // Used to receive callbacks of DTLS identity requests.
470 class DTLSIdentityRequestObserver : public rtc::RefCountInterface {
471 public:
472 virtual void OnFailure(int error) = 0;
473 // TODO(jiayl): Unify the OnSuccess method once Chrome code is updated.
474 virtual void OnSuccess(const std::string& der_cert,
475 const std::string& der_private_key) = 0;
476 // |identity| is a scoped_ptr because rtc::SSLIdentity is not copyable and the
477 // client has to get the ownership of the object to make use of it.
478 virtual void OnSuccessWithIdentityObj(
479 rtc::scoped_ptr<rtc::SSLIdentity> identity) = 0;
480
481 protected:
482 virtual ~DTLSIdentityRequestObserver() {}
483 };
484
485 class DTLSIdentityServiceInterface { 471 class DTLSIdentityServiceInterface {
486 public: 472 public:
487 // Asynchronously request a DTLS identity, including a self-signed certificate 473 // Asynchronously request a DTLS identity, including a self-signed certificate
488 // and the private key used to sign the certificate, from the identity store 474 // and the private key used to sign the certificate, from the identity store
489 // for the given identity name. 475 // for the given identity name.
490 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if 476 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
491 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be 477 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
492 // called with an error code if the request failed. 478 // called with an error code if the request failed.
493 // 479 //
494 // Only one request can be made at a time. If a second request is called 480 // Only one request can be made at a time. If a second request is called
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541 int network_ignore_mask; 527 int network_ignore_mask;
542 528
543 // Sets the maximum supported protocol version. The highest version 529 // Sets the maximum supported protocol version. The highest version
544 // supported by both ends will be used for the connection, i.e. if one 530 // supported by both ends will be used for the connection, i.e. if one
545 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. 531 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
546 rtc::SSLProtocolVersion ssl_max_version; 532 rtc::SSLProtocolVersion ssl_max_version;
547 }; 533 };
548 534
549 virtual void SetOptions(const Options& options) = 0; 535 virtual void SetOptions(const Options& options) = 0;
550 536
537 // TODO(hbos): Temporary CreatePeerConnection function while we transition
538 // from DTLSIdentityServiceInterface to DtlsIdentityStoreInterface.
539 rtc::scoped_refptr<PeerConnectionInterface>
540 CreatePeerConnection(
541 const PeerConnectionInterface::RTCConfiguration& configuration,
542 const MediaConstraintsInterface* constraints,
543 PortAllocatorFactoryInterface* allocator_factory,
544 DTLSIdentityServiceInterface* dtls_identity_service,
545 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
546 PeerConnectionObserver* observer) {
547 return CreatePeerConnection(configuration, constraints, allocator_factory,
548 dtls_identity_service, observer);
549 }
550
551 // This method takes the ownership of |dtls_identity_service|. 551 // This method takes the ownership of |dtls_identity_service|.
552 virtual rtc::scoped_refptr<PeerConnectionInterface> 552 virtual rtc::scoped_refptr<PeerConnectionInterface>
553 CreatePeerConnection( 553 CreatePeerConnection(
554 const PeerConnectionInterface::RTCConfiguration& configuration, 554 const PeerConnectionInterface::RTCConfiguration& configuration,
555 const MediaConstraintsInterface* constraints, 555 const MediaConstraintsInterface* constraints,
556 PortAllocatorFactoryInterface* allocator_factory, 556 PortAllocatorFactoryInterface* allocator_factory,
557 DTLSIdentityServiceInterface* dtls_identity_service, 557 DTLSIdentityServiceInterface* dtls_identity_service,
558 PeerConnectionObserver* observer) = 0; 558 PeerConnectionObserver* observer) = 0;
559 559
560 // TODO(mallinath) : Remove below versions after clients are updated 560 // TODO(mallinath) : Remove below versions after clients are updated
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627 CreatePeerConnectionFactory( 627 CreatePeerConnectionFactory(
628 rtc::Thread* worker_thread, 628 rtc::Thread* worker_thread,
629 rtc::Thread* signaling_thread, 629 rtc::Thread* signaling_thread,
630 AudioDeviceModule* default_adm, 630 AudioDeviceModule* default_adm,
631 cricket::WebRtcVideoEncoderFactory* encoder_factory, 631 cricket::WebRtcVideoEncoderFactory* encoder_factory,
632 cricket::WebRtcVideoDecoderFactory* decoder_factory); 632 cricket::WebRtcVideoDecoderFactory* decoder_factory);
633 633
634 } // namespace webrtc 634 } // namespace webrtc
635 635
636 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 636 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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