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

Issue 1974453002: Add a parameter to set a maximum filesize when starting an RTC event log on the PeerConnectionFacto… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed typo in JNI code. Created 4 years, 7 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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645 // StopAecDump function is called. 645 // StopAecDump function is called.
646 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; 646 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
647 647
648 // Stops logging the AEC dump. 648 // Stops logging the AEC dump.
649 virtual void StopAecDump() = 0; 649 virtual void StopAecDump() = 0;
650 650
651 // Starts RtcEventLog using existing file. Takes ownership of |file| and 651 // Starts RtcEventLog using existing file. Takes ownership of |file| and
652 // passes it on to VoiceEngine, which will take the ownership. If the 652 // passes it on to VoiceEngine, which will take the ownership. If the
653 // operation fails the file will be closed. The logging will stop 653 // operation fails the file will be closed. The logging will stop
654 // automatically after 10 minutes have passed, or when the StopRtcEventLog 654 // automatically after 10 minutes have passed, or when the StopRtcEventLog
655 // function is called. 655 // function is called. A maximum filesize in bytes can be set, the logging
656 // will be stopped before exceeding this limit. If max_size_bytes is set to a
657 // value <= 0, no limit will be used.
656 // This function as well as the StopRtcEventLog don't really belong on this 658 // This function as well as the StopRtcEventLog don't really belong on this
657 // interface, this is a temporary solution until we move the logging object 659 // interface, this is a temporary solution until we move the logging object
658 // from inside voice engine to webrtc::Call, which will happen when the VoE 660 // from inside voice engine to webrtc::Call, which will happen when the VoE
659 // restructuring effort is further along. 661 // restructuring effort is further along.
660 // TODO(ivoc): Move this into being: 662 // TODO(ivoc): Move this into being:
661 // PeerConnection => MediaController => webrtc::Call. 663 // PeerConnection => MediaController => webrtc::Call.
664 virtual bool StartRtcEventLog(rtc::PlatformFile file,
665 int64_t max_size_bytes) = 0;
666 // Deprecated, use the version above.
662 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0; 667 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
663 668
664 // Stops logging the RtcEventLog. 669 // Stops logging the RtcEventLog.
665 virtual void StopRtcEventLog() = 0; 670 virtual void StopRtcEventLog() = 0;
666 671
667 protected: 672 protected:
668 // Dtor and ctor protected as objects shouldn't be created or deleted via 673 // Dtor and ctor protected as objects shouldn't be created or deleted via
669 // this interface. 674 // this interface.
670 PeerConnectionFactoryInterface() {} 675 PeerConnectionFactoryInterface() {}
671 ~PeerConnectionFactoryInterface() {} // NOLINT 676 ~PeerConnectionFactoryInterface() {} // NOLINT
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695 CreatePeerConnectionFactory( 700 CreatePeerConnectionFactory(
696 rtc::Thread* worker_thread, 701 rtc::Thread* worker_thread,
697 rtc::Thread* signaling_thread, 702 rtc::Thread* signaling_thread,
698 AudioDeviceModule* default_adm, 703 AudioDeviceModule* default_adm,
699 cricket::WebRtcVideoEncoderFactory* encoder_factory, 704 cricket::WebRtcVideoEncoderFactory* encoder_factory,
700 cricket::WebRtcVideoDecoderFactory* decoder_factory); 705 cricket::WebRtcVideoDecoderFactory* decoder_factory);
701 706
702 } // namespace webrtc 707 } // namespace webrtc
703 708
704 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 709 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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