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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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290 int LeastRequiredDelayMs() const; | 290 int LeastRequiredDelayMs() const; |
291 int SetMinimumPlayoutDelay(int delayMs); | 291 int SetMinimumPlayoutDelay(int delayMs); |
292 int GetPlayoutTimestamp(unsigned int& timestamp); | 292 int GetPlayoutTimestamp(unsigned int& timestamp); |
293 int SetInitTimestamp(unsigned int timestamp); | 293 int SetInitTimestamp(unsigned int timestamp); |
294 int SetInitSequenceNumber(short sequenceNumber); | 294 int SetInitSequenceNumber(short sequenceNumber); |
295 | 295 |
296 // VoEVideoSyncExtended | 296 // VoEVideoSyncExtended |
297 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 297 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
298 | 298 |
299 // VoEDtmf | 299 // VoEDtmf |
300 int SendTelephoneEventOutband(unsigned char eventCode, | 300 int SendTelephoneEventOutband(int event, int duration_ms); |
301 int lengthMs, | |
302 int attenuationDb, | |
303 bool playDtmfEvent); | |
304 int SendTelephoneEventInband(unsigned char eventCode, | 301 int SendTelephoneEventInband(unsigned char eventCode, |
305 int lengthMs, | 302 int lengthMs, |
306 int attenuationDb, | 303 int attenuationDb, |
307 bool playDtmfEvent); | 304 bool playDtmfEvent); |
308 int SetSendTelephoneEventPayloadType(unsigned char type); | 305 int SetSendTelephoneEventPayloadType(unsigned char type); |
309 int GetSendTelephoneEventPayloadType(unsigned char& type); | 306 int GetSendTelephoneEventPayloadType(unsigned char& type); |
310 | 307 |
311 // VoEAudioProcessingImpl | 308 // VoEAudioProcessingImpl |
312 int UpdateRxVadDetection(AudioFrame& audioFrame); | 309 int UpdateRxVadDetection(AudioFrame& audioFrame); |
313 int RegisterRxVadObserver(VoERxVadCallback& observer); | 310 int RegisterRxVadObserver(VoERxVadCallback& observer); |
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590 PacketRouter* packet_router_ = nullptr; | 587 PacketRouter* packet_router_ = nullptr; |
591 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 588 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
592 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 589 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
593 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 590 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
594 }; | 591 }; |
595 | 592 |
596 } // namespace voe | 593 } // namespace voe |
597 } // namespace webrtc | 594 } // namespace webrtc |
598 | 595 |
599 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 596 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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