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

Issue 1426443007: Revert of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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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330 // Note that a SessionDescription negotiation is need before the 330 // Note that a SessionDescription negotiation is need before the
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".
341 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
342 const std::string& kind) {
343 return rtc::scoped_refptr<RtpSenderInterface>();
344 }
345
346 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 340 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
347 const { 341 const {
348 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); 342 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
349 } 343 }
350 344
351 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 345 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
352 const { 346 const {
353 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>(); 347 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
354 } 348 }
355 349
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664 CreatePeerConnectionFactory( 658 CreatePeerConnectionFactory(
665 rtc::Thread* worker_thread, 659 rtc::Thread* worker_thread,
666 rtc::Thread* signaling_thread, 660 rtc::Thread* signaling_thread,
667 AudioDeviceModule* default_adm, 661 AudioDeviceModule* default_adm,
668 cricket::WebRtcVideoEncoderFactory* encoder_factory, 662 cricket::WebRtcVideoEncoderFactory* encoder_factory,
669 cricket::WebRtcVideoDecoderFactory* decoder_factory); 663 cricket::WebRtcVideoDecoderFactory* decoder_factory);
670 664
671 } // namespace webrtc 665 } // namespace webrtc
672 666
673 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 667 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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