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

Issue 1537213002: Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years 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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633 VideoSourceInterface* source) = 0; 633 VideoSourceInterface* source) = 0;
634 634
635 // Creates an new AudioTrack. At the moment |source| can be NULL. 635 // Creates an new AudioTrack. At the moment |source| can be NULL.
636 virtual rtc::scoped_refptr<AudioTrackInterface> 636 virtual rtc::scoped_refptr<AudioTrackInterface>
637 CreateAudioTrack(const std::string& label, 637 CreateAudioTrack(const std::string& label,
638 AudioSourceInterface* source) = 0; 638 AudioSourceInterface* source) = 0;
639 639
640 // Starts AEC dump using existing file. Takes ownership of |file| and passes 640 // Starts AEC dump using existing file. Takes ownership of |file| and passes
641 // it on to VoiceEngine (via other objects) immediately, which will take 641 // it on to VoiceEngine (via other objects) immediately, which will take
642 // the ownerhip. If the operation fails, the file will be closed. 642 // the ownerhip. If the operation fails, the file will be closed.
643 // A maximum file size in bytes can be specified. When the file size limit is 643 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
644 // reached, logging is stopped automatically. If max_size_bytes is set to a 644 // http://crbug.com/264611.
645 // value <= 0, no limit will be used, and logging will continue until the 645 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
646 // StopAecDump function is called.
647 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
648 646
649 // Stops logging the AEC dump. 647 // Stops logging the AEC dump.
650 virtual void StopAecDump() = 0; 648 virtual void StopAecDump() = 0;
651 649
652 // Starts RtcEventLog using existing file. Takes ownership of |file| and 650 // Starts RtcEventLog using existing file. Takes ownership of |file| and
653 // passes it on to VoiceEngine, which will take the ownership. If the 651 // passes it on to VoiceEngine, which will take the ownership. If the
654 // operation fails the file will be closed. The logging will stop 652 // operation fails the file will be closed. The logging will stop
655 // automatically after 10 minutes have passed, or when the StopRtcEventLog 653 // automatically after 10 minutes have passed, or when the StopRtcEventLog
656 // function is called. 654 // function is called.
657 // This function as well as the StopRtcEventLog don't really belong on this 655 // This function as well as the StopRtcEventLog don't really belong on this
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683 CreatePeerConnectionFactory( 681 CreatePeerConnectionFactory(
684 rtc::Thread* worker_thread, 682 rtc::Thread* worker_thread,
685 rtc::Thread* signaling_thread, 683 rtc::Thread* signaling_thread,
686 AudioDeviceModule* default_adm, 684 AudioDeviceModule* default_adm,
687 cricket::WebRtcVideoEncoderFactory* encoder_factory, 685 cricket::WebRtcVideoEncoderFactory* encoder_factory,
688 cricket::WebRtcVideoDecoderFactory* decoder_factory); 686 cricket::WebRtcVideoDecoderFactory* decoder_factory);
689 687
690 } // namespace webrtc 688 } // namespace webrtc
691 689
692 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 690 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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