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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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436 int ResendPackets(const uint16_t* sequence_numbers, int length); | 436 int ResendPackets(const uint16_t* sequence_numbers, int length); |
437 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); | 437 int32_t MixOrReplaceAudioWithFile(int mixingFrequency); |
438 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); | 438 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); |
439 void UpdatePlayoutTimestamp(bool rtcp); | 439 void UpdatePlayoutTimestamp(bool rtcp); |
440 void RegisterReceiveCodecsToRTPModule(); | 440 void RegisterReceiveCodecsToRTPModule(); |
441 | 441 |
442 int SetSendRtpHeaderExtension(bool enable, | 442 int SetSendRtpHeaderExtension(bool enable, |
443 RTPExtensionType type, | 443 RTPExtensionType type, |
444 unsigned char id); | 444 unsigned char id); |
445 | 445 |
446 void UpdateOverheadForEncoder(); | |
447 | |
446 int GetRtpTimestampRateHz() const; | 448 int GetRtpTimestampRateHz() const; |
447 int64_t GetRTT(bool allow_associate_channel) const; | 449 int64_t GetRTT(bool allow_associate_channel) const; |
448 | 450 |
449 rtc::CriticalSection _fileCritSect; | 451 rtc::CriticalSection _fileCritSect; |
450 rtc::CriticalSection _callbackCritSect; | 452 rtc::CriticalSection _callbackCritSect; |
451 rtc::CriticalSection volume_settings_critsect_; | 453 rtc::CriticalSection volume_settings_critsect_; |
452 uint32_t _instanceId; | 454 uint32_t _instanceId; |
453 int32_t _channelId; | 455 int32_t _channelId; |
454 | 456 |
455 ChannelState channel_state_; | 457 ChannelState channel_state_; |
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522 // VoEVolumeControl | 524 // VoEVolumeControl |
523 bool input_mute_ GUARDED_BY(volume_settings_critsect_); | 525 bool input_mute_ GUARDED_BY(volume_settings_critsect_); |
524 bool previous_frame_muted_; // Only accessed from PrepareEncodeAndSend(). | 526 bool previous_frame_muted_; // Only accessed from PrepareEncodeAndSend(). |
525 float _panLeft GUARDED_BY(volume_settings_critsect_); | 527 float _panLeft GUARDED_BY(volume_settings_critsect_); |
526 float _panRight GUARDED_BY(volume_settings_critsect_); | 528 float _panRight GUARDED_BY(volume_settings_critsect_); |
527 float _outputGain GUARDED_BY(volume_settings_critsect_); | 529 float _outputGain GUARDED_BY(volume_settings_critsect_); |
528 // VoeRTP_RTCP | 530 // VoeRTP_RTCP |
529 uint32_t _lastLocalTimeStamp; | 531 uint32_t _lastLocalTimeStamp; |
530 int8_t _lastPayloadType; | 532 int8_t _lastPayloadType; |
531 bool _includeAudioLevelIndication; | 533 bool _includeAudioLevelIndication; |
534 int _transport_overhead_per_packet; | |
535 int _rtp_overhead_per_packet; | |
michaelt
2016/12/19 14:17:01
it seams that we use in old member-vars "_var" whi
nisse-webrtc
2016/12/19 14:41:04
Ooops, I thought all members here were using the o
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532 // VoENetwork | 536 // VoENetwork |
533 AudioFrame::SpeechType _outputSpeechType; | 537 AudioFrame::SpeechType _outputSpeechType; |
534 // VoEVideoSync | 538 // VoEVideoSync |
535 rtc::CriticalSection video_sync_lock_; | 539 rtc::CriticalSection video_sync_lock_; |
536 // VoEAudioProcessing | 540 // VoEAudioProcessing |
537 bool restored_packet_in_use_; | 541 bool restored_packet_in_use_; |
538 // RtcpBandwidthObserver | 542 // RtcpBandwidthObserver |
539 std::unique_ptr<VoERtcpObserver> rtcp_observer_; | 543 std::unique_ptr<VoERtcpObserver> rtcp_observer_; |
540 // An associated send channel. | 544 // An associated send channel. |
541 rtc::CriticalSection assoc_send_channel_lock_; | 545 rtc::CriticalSection assoc_send_channel_lock_; |
542 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 546 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
543 | 547 |
544 bool pacing_enabled_; | 548 bool pacing_enabled_; |
545 PacketRouter* packet_router_ = nullptr; | 549 PacketRouter* packet_router_ = nullptr; |
546 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 550 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
547 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 551 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
548 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 552 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
549 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 553 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
550 | 554 |
551 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 555 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
552 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 556 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
553 | 557 |
554 SmoothingFilterImpl bitrate_smoother_; | 558 SmoothingFilterImpl bitrate_smoother_; |
555 }; | 559 }; |
556 | 560 |
557 } // namespace voe | 561 } // namespace voe |
558 } // namespace webrtc | 562 } // namespace webrtc |
559 | 563 |
560 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 564 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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