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

Issue 1460323002: 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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326 // Note that a SessionDescription negotiation is need before the 326 // Note that a SessionDescription negotiation is need before the
327 // remote peer is notified. 327 // remote peer is notified.
328 virtual void RemoveStream(MediaStreamInterface* stream) = 0; 328 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
329 329
330 // Returns pointer to the created DtmfSender on success. 330 // Returns pointer to the created DtmfSender on success.
331 // Otherwise returns NULL. 331 // Otherwise returns NULL.
332 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( 332 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
333 AudioTrackInterface* track) = 0; 333 AudioTrackInterface* track) = 0;
334 334
335 // TODO(deadbeef): Make these pure virtual once all subclasses implement them. 335 // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
336 // |kind| must be "audio" or "video".
337 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
338 const std::string& kind) {
339 return rtc::scoped_refptr<RtpSenderInterface>();
340 }
341
342 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 336 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
343 const { 337 const {
344 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); 338 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
345 } 339 }
346 340
347 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 341 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
348 const { 342 const {
349 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>(); 343 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
350 } 344 }
351 345
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660 CreatePeerConnectionFactory( 654 CreatePeerConnectionFactory(
661 rtc::Thread* worker_thread, 655 rtc::Thread* worker_thread,
662 rtc::Thread* signaling_thread, 656 rtc::Thread* signaling_thread,
663 AudioDeviceModule* default_adm, 657 AudioDeviceModule* default_adm,
664 cricket::WebRtcVideoEncoderFactory* encoder_factory, 658 cricket::WebRtcVideoEncoderFactory* encoder_factory,
665 cricket::WebRtcVideoDecoderFactory* decoder_factory); 659 cricket::WebRtcVideoDecoderFactory* decoder_factory);
666 660
667 } // namespace webrtc 661 } // namespace webrtc
668 662
669 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 663 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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