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

Issue 1413713003: Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some unit tests for new methods on the sender. 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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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".
pthatcher1 2015/10/20 17:42:49 I think we could use cricket::MediaType here and s
Taylor Brandstetter 2015/10/21 00:22:08 Done.
341 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
342 const std::string& kind) {
343 return rtc::scoped_refptr<RtpSenderInterface>();
344 }
345
340 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 346 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
341 const { 347 const {
342 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); 348 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
343 } 349 }
344 350
345 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 351 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
346 const { 352 const {
347 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>(); 353 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
348 } 354 }
349 355
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637 CreatePeerConnectionFactory( 643 CreatePeerConnectionFactory(
638 rtc::Thread* worker_thread, 644 rtc::Thread* worker_thread,
639 rtc::Thread* signaling_thread, 645 rtc::Thread* signaling_thread,
640 AudioDeviceModule* default_adm, 646 AudioDeviceModule* default_adm,
641 cricket::WebRtcVideoEncoderFactory* encoder_factory, 647 cricket::WebRtcVideoEncoderFactory* encoder_factory,
642 cricket::WebRtcVideoDecoderFactory* decoder_factory); 648 cricket::WebRtcVideoDecoderFactory* decoder_factory);
643 649
644 } // namespace webrtc 650 } // namespace webrtc
645 651
646 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 652 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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