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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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| 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/dtlsidentitystore.h" |
| 76 #include "talk/app/webrtc/dtmfsenderinterface.h" | 76 #include "talk/app/webrtc/dtmfsenderinterface.h" |
| 77 #include "talk/app/webrtc/dtlsidentitystore.h" | 77 #include "talk/app/webrtc/dtlsidentitystore.h" |
| 78 #include "talk/app/webrtc/jsep.h" | 78 #include "talk/app/webrtc/jsep.h" |
| 79 #include "talk/app/webrtc/mediastreaminterface.h" | 79 #include "talk/app/webrtc/mediastreaminterface.h" |
| 80 #include "talk/app/webrtc/rtpreceiverinterface.h" | |
| 81 #include "talk/app/webrtc/rtpsenderinterface.h" | |
| 80 #include "talk/app/webrtc/statstypes.h" | 82 #include "talk/app/webrtc/statstypes.h" |
| 81 #include "talk/app/webrtc/umametrics.h" | 83 #include "talk/app/webrtc/umametrics.h" |
| 82 #include "webrtc/base/fileutils.h" | 84 #include "webrtc/base/fileutils.h" |
| 83 #include "webrtc/base/network.h" | 85 #include "webrtc/base/network.h" |
| 84 #include "webrtc/base/rtccertificate.h" | 86 #include "webrtc/base/rtccertificate.h" |
| 85 #include "webrtc/base/sslstreamadapter.h" | 87 #include "webrtc/base/sslstreamadapter.h" |
| 86 #include "webrtc/base/socketaddress.h" | 88 #include "webrtc/base/socketaddress.h" |
| 87 | 89 |
| 88 namespace rtc { | 90 namespace rtc { |
| 89 class SSLIdentity; | 91 class SSLIdentity; |
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| 315 // Remove a MediaStream from this PeerConnection. | 317 // Remove a MediaStream from this PeerConnection. |
| 316 // Note that a SessionDescription negotiation is need before the | 318 // Note that a SessionDescription negotiation is need before the |
| 317 // remote peer is notified. | 319 // remote peer is notified. |
| 318 virtual void RemoveStream(MediaStreamInterface* stream) = 0; | 320 virtual void RemoveStream(MediaStreamInterface* stream) = 0; |
| 319 | 321 |
| 320 // Returns pointer to the created DtmfSender on success. | 322 // Returns pointer to the created DtmfSender on success. |
| 321 // Otherwise returns NULL. | 323 // Otherwise returns NULL. |
| 322 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( | 324 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( |
| 323 AudioTrackInterface* track) = 0; | 325 AudioTrackInterface* track) = 0; |
| 324 | 326 |
| 327 typedef std::vector<rtc::scoped_refptr<RtpSenderInterface>> RtpSenders; | |
|
pthatcher1
2015/09/17 04:25:46
I think we should call it something like Refcounte
Taylor Brandstetter
2015/09/23 00:10:45
I assume you mean "out of PeerConnectionInterface"
pthatcher1
2015/09/23 14:47:39
Yes, out of PeerConnectionInterface.
Actually, th
| |
| 328 virtual RtpSenders GetSenders() = 0; | |
| 329 | |
| 330 typedef std::vector<rtc::scoped_refptr<RtpReceiverInterface>> RtpReceivers; | |
| 331 virtual RtpReceivers GetReceivers() = 0; | |
| 332 | |
| 325 virtual bool GetStats(StatsObserver* observer, | 333 virtual bool GetStats(StatsObserver* observer, |
| 326 MediaStreamTrackInterface* track, | 334 MediaStreamTrackInterface* track, |
| 327 StatsOutputLevel level) = 0; | 335 StatsOutputLevel level) = 0; |
| 328 | 336 |
| 329 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( | 337 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( |
| 330 const std::string& label, | 338 const std::string& label, |
| 331 const DataChannelInit* config) = 0; | 339 const DataChannelInit* config) = 0; |
| 332 | 340 |
| 333 virtual const SessionDescriptionInterface* local_description() const = 0; | 341 virtual const SessionDescriptionInterface* local_description() const = 0; |
| 334 virtual const SessionDescriptionInterface* remote_description() const = 0; | 342 virtual const SessionDescriptionInterface* remote_description() const = 0; |
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| 598 CreatePeerConnectionFactory( | 606 CreatePeerConnectionFactory( |
| 599 rtc::Thread* worker_thread, | 607 rtc::Thread* worker_thread, |
| 600 rtc::Thread* signaling_thread, | 608 rtc::Thread* signaling_thread, |
| 601 AudioDeviceModule* default_adm, | 609 AudioDeviceModule* default_adm, |
| 602 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 610 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
| 603 cricket::WebRtcVideoDecoderFactory* decoder_factory); | 611 cricket::WebRtcVideoDecoderFactory* decoder_factory); |
| 604 | 612 |
| 605 } // namespace webrtc | 613 } // namespace webrtc |
| 606 | 614 |
| 607 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 615 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
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