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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 |
11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/audio_sink.h" | 16 #include "webrtc/audio_sink.h" |
17 #include "webrtc/base/criticalsection.h" | 17 #include "webrtc/base/criticalsection.h" |
18 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 18 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
19 #include "webrtc/common_types.h" | 19 #include "webrtc/common_types.h" |
20 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 20 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
21 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" | 21 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" |
22 #include "webrtc/modules/audio_processing/rms_level.h" | 22 #include "webrtc/modules/audio_processing/rms_level.h" |
23 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 23 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
24 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
26 #include "webrtc/modules/utility/include/file_player.h" | 26 #include "webrtc/modules/utility/include/file_player.h" |
27 #include "webrtc/modules/utility/include/file_recorder.h" | 27 #include "webrtc/modules/utility/include/file_recorder.h" |
28 #include "webrtc/voice_engine/dtmf_inband.h" | |
29 #include "webrtc/voice_engine/dtmf_inband_queue.h" | |
30 #include "webrtc/voice_engine/include/voe_audio_processing.h" | 28 #include "webrtc/voice_engine/include/voe_audio_processing.h" |
31 #include "webrtc/voice_engine/include/voe_network.h" | 29 #include "webrtc/voice_engine/include/voe_network.h" |
32 #include "webrtc/voice_engine/level_indicator.h" | 30 #include "webrtc/voice_engine/level_indicator.h" |
33 #include "webrtc/voice_engine/network_predictor.h" | 31 #include "webrtc/voice_engine/network_predictor.h" |
34 #include "webrtc/voice_engine/shared_data.h" | 32 #include "webrtc/voice_engine/shared_data.h" |
35 #include "webrtc/voice_engine/voice_engine_defines.h" | 33 #include "webrtc/voice_engine/voice_engine_defines.h" |
36 | 34 |
37 namespace rtc { | 35 namespace rtc { |
38 | 36 |
39 class TimestampWrapAroundHandler; | 37 class TimestampWrapAroundHandler; |
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291 int SetMinimumPlayoutDelay(int delayMs); | 289 int SetMinimumPlayoutDelay(int delayMs); |
292 int GetPlayoutTimestamp(unsigned int& timestamp); | 290 int GetPlayoutTimestamp(unsigned int& timestamp); |
293 int SetInitTimestamp(unsigned int timestamp); | 291 int SetInitTimestamp(unsigned int timestamp); |
294 int SetInitSequenceNumber(short sequenceNumber); | 292 int SetInitSequenceNumber(short sequenceNumber); |
295 | 293 |
296 // VoEVideoSyncExtended | 294 // VoEVideoSyncExtended |
297 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 295 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
298 | 296 |
299 // VoEDtmf | 297 // VoEDtmf |
300 int SendTelephoneEventOutband(int event, int duration_ms); | 298 int SendTelephoneEventOutband(int event, int duration_ms); |
301 int SendTelephoneEventInband(unsigned char eventCode, | |
302 int lengthMs, | |
303 int attenuationDb, | |
304 bool playDtmfEvent); | |
305 int SetSendTelephoneEventPayloadType(unsigned char type); | 299 int SetSendTelephoneEventPayloadType(unsigned char type); |
306 int GetSendTelephoneEventPayloadType(unsigned char& type); | |
307 | 300 |
308 // VoEAudioProcessingImpl | 301 // VoEAudioProcessingImpl |
309 int UpdateRxVadDetection(AudioFrame& audioFrame); | 302 int UpdateRxVadDetection(AudioFrame& audioFrame); |
310 int RegisterRxVadObserver(VoERxVadCallback& observer); | 303 int RegisterRxVadObserver(VoERxVadCallback& observer); |
311 int DeRegisterRxVadObserver(); | 304 int DeRegisterRxVadObserver(); |
312 int VoiceActivityIndicator(int& activity); | 305 int VoiceActivityIndicator(int& activity); |
313 #ifdef WEBRTC_VOICE_ENGINE_AGC | 306 #ifdef WEBRTC_VOICE_ENGINE_AGC |
314 int SetRxAgcStatus(bool enable, AgcModes mode); | 307 int SetRxAgcStatus(bool enable, AgcModes mode); |
315 int GetRxAgcStatus(bool& enabled, AgcModes& mode); | 308 int GetRxAgcStatus(bool& enabled, AgcModes& mode); |
316 int SetRxAgcConfig(AgcConfig config); | 309 int SetRxAgcConfig(AgcConfig config); |
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454 bool ReceivePacket(const uint8_t* packet, | 447 bool ReceivePacket(const uint8_t* packet, |
455 size_t packet_length, | 448 size_t packet_length, |
456 const RTPHeader& header, | 449 const RTPHeader& header, |
457 bool in_order); | 450 bool in_order); |
458 bool HandleRtxPacket(const uint8_t* packet, | 451 bool HandleRtxPacket(const uint8_t* packet, |
459 size_t packet_length, | 452 size_t packet_length, |
460 const RTPHeader& header); | 453 const RTPHeader& header); |
461 bool IsPacketInOrder(const RTPHeader& header) const; | 454 bool IsPacketInOrder(const RTPHeader& header) const; |
462 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; | 455 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; |
463 int ResendPackets(const uint16_t* sequence_numbers, int length); | 456 int ResendPackets(const uint16_t* sequence_numbers, int length); |
464 int InsertInbandDtmfTone(); | |
465 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); | 457 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); |
466 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); | 458 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); |
467 void UpdatePlayoutTimestamp(bool rtcp); | 459 void UpdatePlayoutTimestamp(bool rtcp); |
468 void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber); | 460 void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber); |
469 void RegisterReceiveCodecsToRTPModule(); | 461 void RegisterReceiveCodecsToRTPModule(); |
470 | 462 |
471 int SetRedPayloadType(int red_payload_type); | 463 int SetRedPayloadType(int red_payload_type); |
472 int SetSendRtpHeaderExtension(bool enable, | 464 int SetSendRtpHeaderExtension(bool enable, |
473 RTPExtensionType type, | 465 RTPExtensionType type, |
474 unsigned char id); | 466 unsigned char id); |
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500 AudioFrame _audioFrame; | 492 AudioFrame _audioFrame; |
501 // Downsamples to the codec rate if necessary. | 493 // Downsamples to the codec rate if necessary. |
502 PushResampler<int16_t> input_resampler_; | 494 PushResampler<int16_t> input_resampler_; |
503 FilePlayer* _inputFilePlayerPtr; | 495 FilePlayer* _inputFilePlayerPtr; |
504 FilePlayer* _outputFilePlayerPtr; | 496 FilePlayer* _outputFilePlayerPtr; |
505 FileRecorder* _outputFileRecorderPtr; | 497 FileRecorder* _outputFileRecorderPtr; |
506 int _inputFilePlayerId; | 498 int _inputFilePlayerId; |
507 int _outputFilePlayerId; | 499 int _outputFilePlayerId; |
508 int _outputFileRecorderId; | 500 int _outputFileRecorderId; |
509 bool _outputFileRecording; | 501 bool _outputFileRecording; |
510 DtmfInbandQueue _inbandDtmfQueue; | |
511 DtmfInband _inbandDtmfGenerator; | |
512 bool _outputExternalMedia; | 502 bool _outputExternalMedia; |
513 VoEMediaProcess* _inputExternalMediaCallbackPtr; | 503 VoEMediaProcess* _inputExternalMediaCallbackPtr; |
514 VoEMediaProcess* _outputExternalMediaCallbackPtr; | 504 VoEMediaProcess* _outputExternalMediaCallbackPtr; |
515 uint32_t _timeStamp; | 505 uint32_t _timeStamp; |
516 uint8_t _sendTelephoneEventPayloadType; | |
tlegrand-webrtc
2016/03/11 13:49:17
I wonder if this should remain? It's used for the
the sun
2016/03/11 13:56:24
Correct, but it is only used here to support retri
tlegrand-webrtc
2016/03/14 08:24:50
Acknowledged.
| |
517 | 506 |
518 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); | 507 RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); |
519 | 508 |
520 // Timestamp of the audio pulled from NetEq. | 509 // Timestamp of the audio pulled from NetEq. |
521 uint32_t jitter_buffer_playout_timestamp_; | 510 uint32_t jitter_buffer_playout_timestamp_; |
522 uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_); | 511 uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_); |
523 uint32_t playout_timestamp_rtcp_; | 512 uint32_t playout_timestamp_rtcp_; |
524 uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_); | 513 uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_); |
525 uint32_t _numberOfDiscardedPackets; | 514 uint32_t _numberOfDiscardedPackets; |
526 uint16_t send_sequence_number_; | 515 uint16_t send_sequence_number_; |
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552 // VoEBase | 541 // VoEBase |
553 bool _externalMixing; | 542 bool _externalMixing; |
554 bool _mixFileWithMicrophone; | 543 bool _mixFileWithMicrophone; |
555 // VoEVolumeControl | 544 // VoEVolumeControl |
556 bool _mute; | 545 bool _mute; |
557 float _panLeft; | 546 float _panLeft; |
558 float _panRight; | 547 float _panRight; |
559 float _outputGain; | 548 float _outputGain; |
560 // VoEDtmf | 549 // VoEDtmf |
561 bool _playOutbandDtmfEvent; | 550 bool _playOutbandDtmfEvent; |
562 bool _playInbandDtmfEvent; | |
563 // VoeRTP_RTCP | 551 // VoeRTP_RTCP |
564 uint32_t _lastLocalTimeStamp; | 552 uint32_t _lastLocalTimeStamp; |
565 int8_t _lastPayloadType; | 553 int8_t _lastPayloadType; |
566 bool _includeAudioLevelIndication; | 554 bool _includeAudioLevelIndication; |
567 // VoENetwork | 555 // VoENetwork |
568 AudioFrame::SpeechType _outputSpeechType; | 556 AudioFrame::SpeechType _outputSpeechType; |
569 // VoEVideoSync | 557 // VoEVideoSync |
570 rtc::CriticalSection video_sync_lock_; | 558 rtc::CriticalSection video_sync_lock_; |
571 uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_); | 559 uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_); |
572 uint32_t _previousTimestamp; | 560 uint32_t _previousTimestamp; |
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587 PacketRouter* packet_router_ = nullptr; | 575 PacketRouter* packet_router_ = nullptr; |
588 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 576 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
589 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 577 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
590 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 578 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
591 }; | 579 }; |
592 | 580 |
593 } // namespace voe | 581 } // namespace voe |
594 } // namespace webrtc | 582 } // namespace webrtc |
595 | 583 |
596 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 584 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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