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Side by Side Diff: webrtc/api/peerconnectioninterface.h

Issue 2107253002: Fix to make the start/stop functions for the Rtc Eventlog non-virtual. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added empty/default implementation. Created 4 years, 5 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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487 // Returns the current IceState. 487 // Returns the current IceState.
488 virtual IceState ice_state() = 0; 488 virtual IceState ice_state() = 0;
489 virtual IceConnectionState ice_connection_state() = 0; 489 virtual IceConnectionState ice_connection_state() = 0;
490 virtual IceGatheringState ice_gathering_state() = 0; 490 virtual IceGatheringState ice_gathering_state() = 0;
491 491
492 // Starts RtcEventLog using existing file. Takes ownership of |file| and 492 // Starts RtcEventLog using existing file. Takes ownership of |file| and
493 // passes it on to Call, which will take the ownership. If the 493 // passes it on to Call, which will take the ownership. If the
494 // operation fails the file will be closed. The logging will stop 494 // operation fails the file will be closed. The logging will stop
495 // automatically after 10 minutes have passed, or when the StopRtcEventLog 495 // automatically after 10 minutes have passed, or when the StopRtcEventLog
496 // function is called. 496 // function is called.
497 // TODO(ivoc): Make this pure virtual when Chrome is updated.
497 virtual bool StartRtcEventLog(rtc::PlatformFile file, 498 virtual bool StartRtcEventLog(rtc::PlatformFile file,
498 int64_t max_size_bytes) = 0; 499 int64_t max_size_bytes) {
500 return false;
501 }
tommi 2016/06/29 14:55:13 ah :)
499 502
500 // Stops logging the RtcEventLog. 503 // Stops logging the RtcEventLog.
501 virtual void StopRtcEventLog() = 0; 504 // TODO(ivoc): Make this pure virtual when Chrome is updated.
505 virtual void StopRtcEventLog() {}
502 506
503 // Terminates all media and closes the transport. 507 // Terminates all media and closes the transport.
504 virtual void Close() = 0; 508 virtual void Close() = 0;
505 509
506 protected: 510 protected:
507 // Dtor protected as objects shouldn't be deleted via this interface. 511 // Dtor protected as objects shouldn't be deleted via this interface.
508 ~PeerConnectionInterface() {} 512 ~PeerConnectionInterface() {}
509 }; 513 };
510 514
511 // PeerConnection callback interface. Application should implement these 515 // PeerConnection callback interface. Application should implement these
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726 cricket::WebRtcVideoEncoderFactory* encoder_factory, 730 cricket::WebRtcVideoEncoderFactory* encoder_factory,
727 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 731 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
728 return CreatePeerConnectionFactory( 732 return CreatePeerConnectionFactory(
729 worker_and_network_thread, worker_and_network_thread, signaling_thread, 733 worker_and_network_thread, worker_and_network_thread, signaling_thread,
730 default_adm, encoder_factory, decoder_factory); 734 default_adm, encoder_factory, decoder_factory);
731 } 735 }
732 736
733 } // namespace webrtc 737 } // namespace webrtc
734 738
735 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 739 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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