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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 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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368 | 368 |
369 // Returns if the telephone-event has been negotiated. | 369 // Returns if the telephone-event has been negotiated. |
370 bool CanInsertDtmf(); | 370 bool CanInsertDtmf(); |
371 // Send and/or play a DTMF |event| according to the |flags|. | 371 // Send and/or play a DTMF |event| according to the |flags|. |
372 // The DTMF out-of-band signal will be used on sending. | 372 // The DTMF out-of-band signal will be used on sending. |
373 // The |ssrc| should be either 0 or a valid send stream ssrc. | 373 // The |ssrc| should be either 0 or a valid send stream ssrc. |
374 // The valid value for the |event| are 0 which corresponding to DTMF | 374 // The valid value for the |event| are 0 which corresponding to DTMF |
375 // event 0-9, *, #, A-D. | 375 // event 0-9, *, #, A-D. |
376 bool InsertDtmf(uint32_t ssrc, int event_code, int duration); | 376 bool InsertDtmf(uint32_t ssrc, int event_code, int duration); |
377 bool SetOutputVolume(uint32_t ssrc, double volume); | 377 bool SetOutputVolume(uint32_t ssrc, double volume); |
378 void SetRawAudioSink( | 378 void SetRawAudioSink(uint32_t ssrc, |
379 uint32_t ssrc, | 379 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink); |
380 const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink); | |
381 | 380 |
382 // Get statistics about the current media session. | 381 // Get statistics about the current media session. |
383 bool GetStats(VoiceMediaInfo* stats); | 382 bool GetStats(VoiceMediaInfo* stats); |
384 | 383 |
385 // Monitoring functions | 384 // Monitoring functions |
386 sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&> | 385 sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&> |
387 SignalConnectionMonitor; | 386 SignalConnectionMonitor; |
388 | 387 |
389 void StartMediaMonitor(int cms); | 388 void StartMediaMonitor(int cms); |
390 void StopMediaMonitor(); | 389 void StopMediaMonitor(); |
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644 // SetSendParameters. | 643 // SetSendParameters. |
645 DataSendParameters last_send_params_; | 644 DataSendParameters last_send_params_; |
646 // Last DataRecvParameters sent down to the media_channel() via | 645 // Last DataRecvParameters sent down to the media_channel() via |
647 // SetRecvParameters. | 646 // SetRecvParameters. |
648 DataRecvParameters last_recv_params_; | 647 DataRecvParameters last_recv_params_; |
649 }; | 648 }; |
650 | 649 |
651 } // namespace cricket | 650 } // namespace cricket |
652 | 651 |
653 #endif // TALK_SESSION_MEDIA_CHANNEL_H_ | 652 #endif // TALK_SESSION_MEDIA_CHANNEL_H_ |
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