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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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322 // Remove a MediaStream from this PeerConnection. | 324 // Remove a MediaStream from this PeerConnection. |
323 // Note that a SessionDescription negotiation is need before the | 325 // Note that a SessionDescription negotiation is need before the |
324 // remote peer is notified. | 326 // remote peer is notified. |
325 virtual void RemoveStream(MediaStreamInterface* stream) = 0; | 327 virtual void RemoveStream(MediaStreamInterface* stream) = 0; |
326 | 328 |
327 // Returns pointer to the created DtmfSender on success. | 329 // Returns pointer to the created DtmfSender on success. |
328 // Otherwise returns NULL. | 330 // Otherwise returns NULL. |
329 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( | 331 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( |
330 AudioTrackInterface* track) = 0; | 332 AudioTrackInterface* track) = 0; |
331 | 333 |
| 334 // TODO(deadbeef): Make these pure virtual once all subclasses implement them. |
| 335 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() |
| 336 const { |
| 337 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); |
| 338 } |
| 339 |
| 340 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() |
| 341 const { |
| 342 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>(); |
| 343 } |
| 344 |
332 virtual bool GetStats(StatsObserver* observer, | 345 virtual bool GetStats(StatsObserver* observer, |
333 MediaStreamTrackInterface* track, | 346 MediaStreamTrackInterface* track, |
334 StatsOutputLevel level) = 0; | 347 StatsOutputLevel level) = 0; |
335 | 348 |
336 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( | 349 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( |
337 const std::string& label, | 350 const std::string& label, |
338 const DataChannelInit* config) = 0; | 351 const DataChannelInit* config) = 0; |
339 | 352 |
340 virtual const SessionDescriptionInterface* local_description() const = 0; | 353 virtual const SessionDescriptionInterface* local_description() const = 0; |
341 virtual const SessionDescriptionInterface* remote_description() const = 0; | 354 virtual const SessionDescriptionInterface* remote_description() const = 0; |
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605 CreatePeerConnectionFactory( | 618 CreatePeerConnectionFactory( |
606 rtc::Thread* worker_thread, | 619 rtc::Thread* worker_thread, |
607 rtc::Thread* signaling_thread, | 620 rtc::Thread* signaling_thread, |
608 AudioDeviceModule* default_adm, | 621 AudioDeviceModule* default_adm, |
609 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 622 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
610 cricket::WebRtcVideoDecoderFactory* decoder_factory); | 623 cricket::WebRtcVideoDecoderFactory* decoder_factory); |
611 | 624 |
612 } // namespace webrtc | 625 } // namespace webrtc |
613 | 626 |
614 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 627 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
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