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

Issue 1374253002: Added functions on libjingle API to start and stop the recording of an RtcEventLog. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added functions to start and stop logging. Created 5 years, 2 months 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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573 CreateAudioTrack(const std::string& label, 573 CreateAudioTrack(const std::string& label,
574 AudioSourceInterface* source) = 0; 574 AudioSourceInterface* source) = 0;
575 575
576 // Starts AEC dump using existing file. Takes ownership of |file| and passes 576 // Starts AEC dump using existing file. Takes ownership of |file| and passes
577 // it on to VoiceEngine (via other objects) immediately, which will take 577 // it on to VoiceEngine (via other objects) immediately, which will take
578 // the ownerhip. If the operation fails, the file will be closed. 578 // the ownerhip. If the operation fails, the file will be closed.
579 // TODO(grunell): Remove when Chromium has started to use AEC in each source. 579 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
580 // http://crbug.com/264611. 580 // http://crbug.com/264611.
581 virtual bool StartAecDump(rtc::PlatformFile file) = 0; 581 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
582 582
583 // Starts RtcEventLog using existing file. Takes ownership of |file| and
584 // passes it on to VoiceEngine, which will take the ownership. If the
585 // operation fails the file will be closed. The logging will stop
586 // automatically after 10 minutes have passed, or when the StopRtcEventLog
587 // function is called.
588 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
589
590 // Stops logging the RtcEventLog.
591 virtual void StopRtcEventLog() = 0;
592
583 protected: 593 protected:
584 // Dtor and ctor protected as objects shouldn't be created or deleted via 594 // Dtor and ctor protected as objects shouldn't be created or deleted via
585 // this interface. 595 // this interface.
586 PeerConnectionFactoryInterface() {} 596 PeerConnectionFactoryInterface() {}
587 ~PeerConnectionFactoryInterface() {} // NOLINT 597 ~PeerConnectionFactoryInterface() {} // NOLINT
588 }; 598 };
589 599
590 // Create a new instance of PeerConnectionFactoryInterface. 600 // Create a new instance of PeerConnectionFactoryInterface.
591 rtc::scoped_refptr<PeerConnectionFactoryInterface> 601 rtc::scoped_refptr<PeerConnectionFactoryInterface>
592 CreatePeerConnectionFactory(); 602 CreatePeerConnectionFactory();
593 603
594 // Create a new instance of PeerConnectionFactoryInterface. 604 // Create a new instance of PeerConnectionFactoryInterface.
595 // Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and 605 // Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
596 // |decoder_factory| transferred to the returned factory. 606 // |decoder_factory| transferred to the returned factory.
597 rtc::scoped_refptr<PeerConnectionFactoryInterface> 607 rtc::scoped_refptr<PeerConnectionFactoryInterface>
598 CreatePeerConnectionFactory( 608 CreatePeerConnectionFactory(
599 rtc::Thread* worker_thread, 609 rtc::Thread* worker_thread,
600 rtc::Thread* signaling_thread, 610 rtc::Thread* signaling_thread,
601 AudioDeviceModule* default_adm, 611 AudioDeviceModule* default_adm,
602 cricket::WebRtcVideoEncoderFactory* encoder_factory, 612 cricket::WebRtcVideoEncoderFactory* encoder_factory,
603 cricket::WebRtcVideoDecoderFactory* decoder_factory); 613 cricket::WebRtcVideoDecoderFactory* decoder_factory);
604 614
605 } // namespace webrtc 615 } // namespace webrtc
606 616
607 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 617 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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