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

Issue 1413983004: Reland of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing merge issue. 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
336 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 342 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
337 const { 343 const {
338 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>(); 344 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
339 } 345 }
340 346
341 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 347 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
342 const { 348 const {
343 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>(); 349 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
344 } 350 }
345 351
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654 CreatePeerConnectionFactory( 660 CreatePeerConnectionFactory(
655 rtc::Thread* worker_thread, 661 rtc::Thread* worker_thread,
656 rtc::Thread* signaling_thread, 662 rtc::Thread* signaling_thread,
657 AudioDeviceModule* default_adm, 663 AudioDeviceModule* default_adm,
658 cricket::WebRtcVideoEncoderFactory* encoder_factory, 664 cricket::WebRtcVideoEncoderFactory* encoder_factory,
659 cricket::WebRtcVideoDecoderFactory* decoder_factory); 665 cricket::WebRtcVideoDecoderFactory* decoder_factory);
660 666
661 } // namespace webrtc 667 } // namespace webrtc
662 668
663 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 669 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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