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Side by Side Diff: talk/session/media/channel.h

Issue 1588693002: Revert of Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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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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