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

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 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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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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