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

Issue 1538673002: Adding a MediaStream parameter to createSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years 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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331 // remote peer is notified. 331 // remote peer is notified.
332 virtual void RemoveStream(MediaStreamInterface* stream) = 0; 332 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
333 333
334 // Returns pointer to the created DtmfSender on success. 334 // Returns pointer to the created DtmfSender on success.
335 // Otherwise returns NULL. 335 // Otherwise returns NULL.
336 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( 336 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
337 AudioTrackInterface* track) = 0; 337 AudioTrackInterface* track) = 0;
338 338
339 // TODO(deadbeef): Make these pure virtual once all subclasses implement them. 339 // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
340 // |kind| must be "audio" or "video". 340 // |kind| must be "audio" or "video".
341 // |stream| is optional, and only used to populate the msid.
341 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender( 342 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
342 const std::string& kind) { 343 const std::string& kind,
344 MediaStreamInterface* stream) {
343 return rtc::scoped_refptr<RtpSenderInterface>(); 345 return rtc::scoped_refptr<RtpSenderInterface>();
344 } 346 }
345 347
346 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 348 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
347 const { 349 const {
348 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); 350 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
349 } 351 }
350 352
351 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 353 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
352 const { 354 const {
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678 CreatePeerConnectionFactory( 680 CreatePeerConnectionFactory(
679 rtc::Thread* worker_thread, 681 rtc::Thread* worker_thread,
680 rtc::Thread* signaling_thread, 682 rtc::Thread* signaling_thread,
681 AudioDeviceModule* default_adm, 683 AudioDeviceModule* default_adm,
682 cricket::WebRtcVideoEncoderFactory* encoder_factory, 684 cricket::WebRtcVideoEncoderFactory* encoder_factory,
683 cricket::WebRtcVideoDecoderFactory* decoder_factory); 685 cricket::WebRtcVideoDecoderFactory* decoder_factory);
684 686
685 } // namespace webrtc 687 } // namespace webrtc
686 688
687 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 689 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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