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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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629 VideoSourceInterface* source) = 0; | 629 VideoSourceInterface* source) = 0; |
630 | 630 |
631 // Creates an new AudioTrack. At the moment |source| can be NULL. | 631 // Creates an new AudioTrack. At the moment |source| can be NULL. |
632 virtual rtc::scoped_refptr<AudioTrackInterface> | 632 virtual rtc::scoped_refptr<AudioTrackInterface> |
633 CreateAudioTrack(const std::string& label, | 633 CreateAudioTrack(const std::string& label, |
634 AudioSourceInterface* source) = 0; | 634 AudioSourceInterface* source) = 0; |
635 | 635 |
636 // Starts AEC dump using existing file. Takes ownership of |file| and passes | 636 // Starts AEC dump using existing file. Takes ownership of |file| and passes |
637 // it on to VoiceEngine (via other objects) immediately, which will take | 637 // it on to VoiceEngine (via other objects) immediately, which will take |
638 // the ownerhip. If the operation fails, the file will be closed. | 638 // the ownerhip. If the operation fails, the file will be closed. |
639 // TODO(grunell): Remove when Chromium has started to use AEC in each source. | 639 // A maximum file size in bytes can be specified. When the file size limit is |
640 // http://crbug.com/264611. | 640 // reached, logging is stopped automatically. If max_size_bytes is set to a |
641 virtual bool StartAecDump(rtc::PlatformFile file) = 0; | 641 // value <= 0, no limit will be used, and logging will continue until the |
| 642 // StopAecDump function is called. |
| 643 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; |
642 | 644 |
643 // Stops logging the AEC dump. | 645 // Stops logging the AEC dump. |
644 virtual void StopAecDump() = 0; | 646 virtual void StopAecDump() = 0; |
645 | 647 |
646 // Starts RtcEventLog using existing file. Takes ownership of |file| and | 648 // Starts RtcEventLog using existing file. Takes ownership of |file| and |
647 // passes it on to VoiceEngine, which will take the ownership. If the | 649 // passes it on to VoiceEngine, which will take the ownership. If the |
648 // operation fails the file will be closed. The logging will stop | 650 // operation fails the file will be closed. The logging will stop |
649 // automatically after 10 minutes have passed, or when the StopRtcEventLog | 651 // automatically after 10 minutes have passed, or when the StopRtcEventLog |
650 // function is called. | 652 // function is called. |
651 // This function as well as the StopRtcEventLog don't really belong on this | 653 // This function as well as the StopRtcEventLog don't really belong on this |
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677 CreatePeerConnectionFactory( | 679 CreatePeerConnectionFactory( |
678 rtc::Thread* worker_thread, | 680 rtc::Thread* worker_thread, |
679 rtc::Thread* signaling_thread, | 681 rtc::Thread* signaling_thread, |
680 AudioDeviceModule* default_adm, | 682 AudioDeviceModule* default_adm, |
681 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 683 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
682 cricket::WebRtcVideoDecoderFactory* decoder_factory); | 684 cricket::WebRtcVideoDecoderFactory* decoder_factory); |
683 | 685 |
684 } // namespace webrtc | 686 } // namespace webrtc |
685 | 687 |
686 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 688 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ |
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