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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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