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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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297 int SetLocalSSRC(unsigned int ssrc); | 297 int SetLocalSSRC(unsigned int ssrc); |
298 int GetLocalSSRC(unsigned int& ssrc); | 298 int GetLocalSSRC(unsigned int& ssrc); |
299 int GetRemoteSSRC(unsigned int& ssrc); | 299 int GetRemoteSSRC(unsigned int& ssrc); |
300 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); | 300 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); |
301 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); | 301 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); |
302 int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id); | 302 int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id); |
303 int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id); | 303 int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id); |
304 void EnableSendTransportSequenceNumber(int id); | 304 void EnableSendTransportSequenceNumber(int id); |
305 void EnableReceiveTransportSequenceNumber(int id); | 305 void EnableReceiveTransportSequenceNumber(int id); |
306 | 306 |
307 void RegisterSenderCongestionControlObjects( | 307 void RegisterSenderCongestionControlObjects( |
308 RtpPacketSender* rtp_packet_sender, | 308 RtpPacketSender* rtp_packet_sender, |
309 TransportFeedbackObserver* transport_feedback_observer, | 309 TransportFeedbackObserver* transport_feedback_observer, |
310 PacketRouter* packet_router); | 310 PacketRouter* packet_router, |
311 void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router); | 311 RtcpBandwidthObserver* bandwidth_observer); |
312 void ResetCongestionControlObjects(); | 312 void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router); |
| 313 void ResetCongestionControlObjects(); |
313 | 314 |
314 void SetRTCPStatus(bool enable); | 315 void SetRTCPStatus(bool enable); |
315 int GetRTCPStatus(bool& enabled); | 316 int GetRTCPStatus(bool& enabled); |
316 int SetRTCP_CNAME(const char cName[256]); | 317 int SetRTCP_CNAME(const char cName[256]); |
317 int GetRemoteRTCP_CNAME(char cName[256]); | 318 int GetRemoteRTCP_CNAME(char cName[256]); |
318 int GetRemoteRTCPData(unsigned int& NTPHigh, | 319 int GetRemoteRTCPData(unsigned int& NTPHigh, |
319 unsigned int& NTPLow, | 320 unsigned int& NTPLow, |
320 unsigned int& timestamp, | 321 unsigned int& timestamp, |
321 unsigned int& playoutTimestamp, | 322 unsigned int& playoutTimestamp, |
322 unsigned int* jitter, | 323 unsigned int* jitter, |
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553 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 554 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
554 | 555 |
555 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 556 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
556 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 557 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
557 }; | 558 }; |
558 | 559 |
559 } // namespace voe | 560 } // namespace voe |
560 } // namespace webrtc | 561 } // namespace webrtc |
561 | 562 |
562 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 563 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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