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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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267 int SetChannelOutputVolumeScaling(float scaling); | 267 int SetChannelOutputVolumeScaling(float scaling); |
268 int GetChannelOutputVolumeScaling(float& scaling) const; | 268 int GetChannelOutputVolumeScaling(float& scaling) const; |
269 | 269 |
270 // VoENetEqStats | 270 // VoENetEqStats |
271 int GetNetworkStatistics(NetworkStatistics& stats); | 271 int GetNetworkStatistics(NetworkStatistics& stats); |
272 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; | 272 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
273 | 273 |
274 // VoEVideoSync | 274 // VoEVideoSync |
275 bool GetDelayEstimate(int* jitter_buffer_delay_ms, | 275 bool GetDelayEstimate(int* jitter_buffer_delay_ms, |
276 int* playout_buffer_delay_ms) const; | 276 int* playout_buffer_delay_ms) const; |
277 int least_required_delay_ms() const { return least_required_delay_ms_; } | 277 int LeastRequiredDelayMs() const; |
278 int SetInitialPlayoutDelay(int delay_ms); | 278 int SetInitialPlayoutDelay(int delay_ms); |
279 int SetMinimumPlayoutDelay(int delayMs); | 279 int SetMinimumPlayoutDelay(int delayMs); |
280 int GetPlayoutTimestamp(unsigned int& timestamp); | 280 int GetPlayoutTimestamp(unsigned int& timestamp); |
281 void UpdatePlayoutTimestamp(bool rtcp); | |
282 int SetInitTimestamp(unsigned int timestamp); | 281 int SetInitTimestamp(unsigned int timestamp); |
283 int SetInitSequenceNumber(short sequenceNumber); | 282 int SetInitSequenceNumber(short sequenceNumber); |
284 | 283 |
285 // VoEVideoSyncExtended | 284 // VoEVideoSyncExtended |
286 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 285 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
287 | 286 |
288 // VoEDtmf | 287 // VoEDtmf |
289 int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs, | 288 int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs, |
290 int attenuationDb, bool playDtmfEvent); | 289 int attenuationDb, bool playDtmfEvent); |
291 int SendTelephoneEventInband(unsigned char eventCode, int lengthMs, | 290 int SendTelephoneEventInband(unsigned char eventCode, int lengthMs, |
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457 bool HandleRtxPacket(const uint8_t* packet, | 456 bool HandleRtxPacket(const uint8_t* packet, |
458 size_t packet_length, | 457 size_t packet_length, |
459 const RTPHeader& header); | 458 const RTPHeader& header); |
460 bool IsPacketInOrder(const RTPHeader& header) const; | 459 bool IsPacketInOrder(const RTPHeader& header) const; |
461 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; | 460 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; |
462 int ResendPackets(const uint16_t* sequence_numbers, int length); | 461 int ResendPackets(const uint16_t* sequence_numbers, int length); |
463 int InsertInbandDtmfTone(); | 462 int InsertInbandDtmfTone(); |
464 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); | 463 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); |
465 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); | 464 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); |
466 int32_t SendPacketRaw(const void *data, size_t len, bool RTCP); | 465 int32_t SendPacketRaw(const void *data, size_t len, bool RTCP); |
| 466 void UpdatePlayoutTimestamp(bool rtcp); |
467 void UpdatePacketDelay(uint32_t timestamp, | 467 void UpdatePacketDelay(uint32_t timestamp, |
468 uint16_t sequenceNumber); | 468 uint16_t sequenceNumber); |
469 void RegisterReceiveCodecsToRTPModule(); | 469 void RegisterReceiveCodecsToRTPModule(); |
470 | 470 |
471 int SetRedPayloadType(int red_payload_type); | 471 int SetRedPayloadType(int red_payload_type); |
472 int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, | 472 int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, |
473 unsigned char id); | 473 unsigned char id); |
474 | 474 |
475 int32_t GetPlayoutFrequency(); | 475 int32_t GetPlayoutFrequency(); |
476 int64_t GetRTT(bool allow_associate_channel) const; | 476 int64_t GetRTT(bool allow_associate_channel) const; |
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509 bool _outputExternalMedia; | 509 bool _outputExternalMedia; |
510 VoEMediaProcess* _inputExternalMediaCallbackPtr; | 510 VoEMediaProcess* _inputExternalMediaCallbackPtr; |
511 VoEMediaProcess* _outputExternalMediaCallbackPtr; | 511 VoEMediaProcess* _outputExternalMediaCallbackPtr; |
512 uint32_t _timeStamp; | 512 uint32_t _timeStamp; |
513 uint8_t _sendTelephoneEventPayloadType; | 513 uint8_t _sendTelephoneEventPayloadType; |
514 | 514 |
515 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); | 515 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); |
516 | 516 |
517 // Timestamp of the audio pulled from NetEq. | 517 // Timestamp of the audio pulled from NetEq. |
518 uint32_t jitter_buffer_playout_timestamp_; | 518 uint32_t jitter_buffer_playout_timestamp_; |
519 uint32_t playout_timestamp_rtp_; | 519 uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_); |
520 uint32_t playout_timestamp_rtcp_; | 520 uint32_t playout_timestamp_rtcp_; |
521 uint32_t playout_delay_ms_; | 521 uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_); |
522 uint32_t _numberOfDiscardedPackets; | 522 uint32_t _numberOfDiscardedPackets; |
523 uint16_t send_sequence_number_; | 523 uint16_t send_sequence_number_; |
524 uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes]; | 524 uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes]; |
525 | 525 |
526 rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_; | 526 rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_; |
527 | 527 |
528 rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; | 528 rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; |
529 // The rtp timestamp of the first played out audio frame. | 529 // The rtp timestamp of the first played out audio frame. |
530 int64_t capture_start_rtp_time_stamp_; | 530 int64_t capture_start_rtp_time_stamp_; |
531 // The capture ntp time (in local timebase) of the first played out audio | 531 // The capture ntp time (in local timebase) of the first played out audio |
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557 // VoEDtmf | 557 // VoEDtmf |
558 bool _playOutbandDtmfEvent; | 558 bool _playOutbandDtmfEvent; |
559 bool _playInbandDtmfEvent; | 559 bool _playInbandDtmfEvent; |
560 // VoeRTP_RTCP | 560 // VoeRTP_RTCP |
561 uint32_t _lastLocalTimeStamp; | 561 uint32_t _lastLocalTimeStamp; |
562 int8_t _lastPayloadType; | 562 int8_t _lastPayloadType; |
563 bool _includeAudioLevelIndication; | 563 bool _includeAudioLevelIndication; |
564 // VoENetwork | 564 // VoENetwork |
565 AudioFrame::SpeechType _outputSpeechType; | 565 AudioFrame::SpeechType _outputSpeechType; |
566 // VoEVideoSync | 566 // VoEVideoSync |
567 uint32_t _average_jitter_buffer_delay_us; | 567 rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_; |
568 int least_required_delay_ms_; | 568 uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_); |
569 uint32_t _previousTimestamp; | 569 uint32_t _previousTimestamp; |
570 uint16_t _recPacketDelayMs; | 570 uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_); |
571 // VoEAudioProcessing | 571 // VoEAudioProcessing |
572 bool _RxVadDetection; | 572 bool _RxVadDetection; |
573 bool _rxAgcIsEnabled; | 573 bool _rxAgcIsEnabled; |
574 bool _rxNsIsEnabled; | 574 bool _rxNsIsEnabled; |
575 bool restored_packet_in_use_; | 575 bool restored_packet_in_use_; |
576 // RtcpBandwidthObserver | 576 // RtcpBandwidthObserver |
577 rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_; | 577 rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_; |
578 rtc::scoped_ptr<NetworkPredictor> network_predictor_; | 578 rtc::scoped_ptr<NetworkPredictor> network_predictor_; |
579 // An associated send channel. | 579 // An associated send channel. |
580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; | 580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; |
581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
582 }; | 582 }; |
583 | 583 |
584 } // namespace voe | 584 } // namespace voe |
585 } // namespace webrtc | 585 } // namespace webrtc |
586 | 586 |
587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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