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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: Adding some stubs so that Chromium build won't break. Created 5 years, 2 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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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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