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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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