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

Issue 1563403002: Adding AddTrack/RemoveTrack to native PeerConnection API. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding default implementation for new methods. Created 4 years, 11 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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90 const MediaConstraintsInterface* constraints, 90 const MediaConstraintsInterface* constraints,
91 rtc::scoped_ptr<cricket::PortAllocator> allocator, 91 rtc::scoped_ptr<cricket::PortAllocator> allocator,
92 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, 92 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
93 PeerConnectionObserver* observer); 93 PeerConnectionObserver* observer);
94 94
95 rtc::scoped_refptr<StreamCollectionInterface> local_streams() override; 95 rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
96 rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override; 96 rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
97 bool AddStream(MediaStreamInterface* local_stream) override; 97 bool AddStream(MediaStreamInterface* local_stream) override;
98 void RemoveStream(MediaStreamInterface* local_stream) override; 98 void RemoveStream(MediaStreamInterface* local_stream) override;
99 99
100 rtc::scoped_refptr<RtpSenderInterface> AddTrack(
101 MediaStreamTrackInterface* track,
102 std::vector<MediaStreamInterface*> streams) override;
103 bool RemoveTrack(RtpSenderInterface* sender) override;
104
100 virtual WebRtcSession* session() { return session_.get(); } 105 virtual WebRtcSession* session() { return session_.get(); }
101 106
102 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( 107 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
103 AudioTrackInterface* track) override; 108 AudioTrackInterface* track) override;
104 109
105 rtc::scoped_refptr<RtpSenderInterface> CreateSender( 110 rtc::scoped_refptr<RtpSenderInterface> CreateSender(
106 const std::string& kind, 111 const std::string& kind,
107 const std::string& stream_id) override; 112 const std::string& stream_id) override;
108 113
109 std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 114 std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
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400 // because its destruction fires signals (such as VoiceChannelDestroyed) 405 // because its destruction fires signals (such as VoiceChannelDestroyed)
401 // which will trigger some final actions in PeerConnection... 406 // which will trigger some final actions in PeerConnection...
402 rtc::scoped_ptr<WebRtcSession> session_; 407 rtc::scoped_ptr<WebRtcSession> session_;
403 // ... But stats_ depends on session_ so it should be destroyed even earlier. 408 // ... But stats_ depends on session_ so it should be destroyed even earlier.
404 rtc::scoped_ptr<StatsCollector> stats_; 409 rtc::scoped_ptr<StatsCollector> stats_;
405 }; 410 };
406 411
407 } // namespace webrtc 412 } // namespace webrtc
408 413
409 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ 414 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_
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