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

Issue 2111803002: Revert of 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: 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.
498 virtual bool StartRtcEventLog(rtc::PlatformFile file, 497 virtual bool StartRtcEventLog(rtc::PlatformFile file,
499 int64_t max_size_bytes) { 498 int64_t max_size_bytes) = 0;
500 return false;
501 }
502 499
503 // Stops logging the RtcEventLog. 500 // Stops logging the RtcEventLog.
504 // TODO(ivoc): Make this pure virtual when Chrome is updated. 501 virtual void StopRtcEventLog() = 0;
505 virtual void StopRtcEventLog() {}
506 502
507 // Terminates all media and closes the transport. 503 // Terminates all media and closes the transport.
508 virtual void Close() = 0; 504 virtual void Close() = 0;
509 505
510 protected: 506 protected:
511 // Dtor protected as objects shouldn't be deleted via this interface. 507 // Dtor protected as objects shouldn't be deleted via this interface.
512 ~PeerConnectionInterface() {} 508 ~PeerConnectionInterface() {}
513 }; 509 };
514 510
515 // PeerConnection callback interface. Application should implement these 511 // PeerConnection callback interface. Application should implement these
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730 cricket::WebRtcVideoEncoderFactory* encoder_factory, 726 cricket::WebRtcVideoEncoderFactory* encoder_factory,
731 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 727 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
732 return CreatePeerConnectionFactory( 728 return CreatePeerConnectionFactory(
733 worker_and_network_thread, worker_and_network_thread, signaling_thread, 729 worker_and_network_thread, worker_and_network_thread, signaling_thread,
734 default_adm, encoder_factory, decoder_factory); 730 default_adm, encoder_factory, decoder_factory);
735 } 731 }
736 732
737 } // namespace webrtc 733 } // namespace webrtc
738 734
739 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 735 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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