| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
| #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
| |
| #include "webrtc/audio/audio_sink.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/common_audio/resampler/include/push_resampler.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" |
| #include "webrtc/modules/audio_processing/rms_level.h" |
| #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/utility/include/file_player.h" |
| #include "webrtc/modules/utility/include/file_recorder.h" |
| #include "webrtc/voice_engine/dtmf_inband.h" |
| #include "webrtc/voice_engine/dtmf_inband_queue.h" |
| #include "webrtc/voice_engine/include/voe_audio_processing.h" |
| #include "webrtc/voice_engine/include/voe_network.h" |
| #include "webrtc/voice_engine/level_indicator.h" |
| #include "webrtc/voice_engine/network_predictor.h" |
| #include "webrtc/voice_engine/shared_data.h" |
| #include "webrtc/voice_engine/voice_engine_defines.h" |
| |
| #ifdef WEBRTC_DTMF_DETECTION |
| // TelephoneEventDetectionMethods, TelephoneEventObserver |
| #include "webrtc/voice_engine/include/voe_dtmf.h" |
| #endif |
| |
| namespace rtc { |
| |
| class TimestampWrapAroundHandler; |
| } |
| |
| namespace webrtc { |
| |
| class AudioDeviceModule; |
| class Config; |
| class CriticalSectionWrapper; |
| class FileWrapper; |
| class PacketRouter; |
| class ProcessThread; |
| class ReceiveStatistics; |
| class RemoteNtpTimeEstimator; |
| class RtcEventLog; |
| class RTPPayloadRegistry; |
| class RtpReceiver; |
| class RTPReceiverAudio; |
| class RtpRtcp; |
| class TelephoneEventHandler; |
| class VoEMediaProcess; |
| class VoERTPObserver; |
| class VoiceEngineObserver; |
| |
| struct CallStatistics; |
| struct ReportBlock; |
| struct SenderInfo; |
| |
| namespace voe { |
| |
| class OutputMixer; |
| class RtpPacketSenderProxy; |
| class Statistics; |
| class StatisticsProxy; |
| class TransportFeedbackProxy; |
| class TransmitMixer; |
| class TransportSequenceNumberProxy; |
| class VoERtcpObserver; |
| |
| // Helper class to simplify locking scheme for members that are accessed from |
| // multiple threads. |
| // Example: a member can be set on thread T1 and read by an internal audio |
| // thread T2. Accessing the member via this class ensures that we are |
| // safe and also avoid TSan v2 warnings. |
| class ChannelState { |
| public: |
| struct State { |
| State() : rx_apm_is_enabled(false), |
| input_external_media(false), |
| output_file_playing(false), |
| input_file_playing(false), |
| playing(false), |
| sending(false), |
| receiving(false) {} |
| |
| bool rx_apm_is_enabled; |
| bool input_external_media; |
| bool output_file_playing; |
| bool input_file_playing; |
| bool playing; |
| bool sending; |
| bool receiving; |
| }; |
| |
| ChannelState() : lock_(CriticalSectionWrapper::CreateCriticalSection()) { |
| } |
| virtual ~ChannelState() {} |
| |
| void Reset() { |
| CriticalSectionScoped lock(lock_.get()); |
| state_ = State(); |
| } |
| |
| State Get() const { |
| CriticalSectionScoped lock(lock_.get()); |
| return state_; |
| } |
| |
| void SetRxApmIsEnabled(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.rx_apm_is_enabled = enable; |
| } |
| |
| void SetInputExternalMedia(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.input_external_media = enable; |
| } |
| |
| void SetOutputFilePlaying(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.output_file_playing = enable; |
| } |
| |
| void SetInputFilePlaying(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.input_file_playing = enable; |
| } |
| |
| void SetPlaying(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.playing = enable; |
| } |
| |
| void SetSending(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.sending = enable; |
| } |
| |
| void SetReceiving(bool enable) { |
| CriticalSectionScoped lock(lock_.get()); |
| state_.receiving = enable; |
| } |
| |
| private: |
| rtc::scoped_ptr<CriticalSectionWrapper> lock_; |
| State state_; |
| }; |
| |
| class Channel: |
| public RtpData, |
| public RtpFeedback, |
| public FileCallback, // receiving notification from file player & recorder |
| public Transport, |
| public RtpAudioFeedback, |
| public AudioPacketizationCallback, // receive encoded packets from the ACM |
| public ACMVADCallback, // receive voice activity from the ACM |
| public MixerParticipant // supplies output mixer with audio frames |
| { |
| public: |
| friend class VoERtcpObserver; |
| |
| enum {KNumSocketThreads = 1}; |
| enum {KNumberOfSocketBuffers = 8}; |
| virtual ~Channel(); |
| static int32_t CreateChannel(Channel*& channel, |
| int32_t channelId, |
| uint32_t instanceId, |
| RtcEventLog* const event_log, |
| const Config& config); |
| Channel(int32_t channelId, |
| uint32_t instanceId, |
| RtcEventLog* const event_log, |
| const Config& config); |
| int32_t Init(); |
| int32_t SetEngineInformation( |
| Statistics& engineStatistics, |
| OutputMixer& outputMixer, |
| TransmitMixer& transmitMixer, |
| ProcessThread& moduleProcessThread, |
| AudioDeviceModule& audioDeviceModule, |
| VoiceEngineObserver* voiceEngineObserver, |
| CriticalSectionWrapper* callbackCritSect); |
| int32_t UpdateLocalTimeStamp(); |
| |
| void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink); |
| |
| // API methods |
| |
| // VoEBase |
| int32_t StartPlayout(); |
| int32_t StopPlayout(); |
| int32_t StartSend(); |
| int32_t StopSend(); |
| int32_t StartReceiving(); |
| int32_t StopReceiving(); |
| |
| int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); |
| int32_t DeRegisterVoiceEngineObserver(); |
| |
| // VoECodec |
| int32_t GetSendCodec(CodecInst& codec); |
| int32_t GetRecCodec(CodecInst& codec); |
| int32_t SetSendCodec(const CodecInst& codec); |
| void SetBitRate(int bitrate_bps); |
| int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX); |
| int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX); |
| int32_t SetRecPayloadType(const CodecInst& codec); |
| int32_t GetRecPayloadType(CodecInst& codec); |
| int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency); |
| int SetOpusMaxPlaybackRate(int frequency_hz); |
| int SetOpusDtx(bool enable_dtx); |
| |
| // VoENetwork |
| int32_t RegisterExternalTransport(Transport& transport); |
| int32_t DeRegisterExternalTransport(); |
| int32_t ReceivedRTPPacket(const int8_t* data, size_t length, |
| const PacketTime& packet_time); |
| int32_t ReceivedRTCPPacket(const int8_t* data, size_t length); |
| |
| // VoEFile |
| int StartPlayingFileLocally(const char* fileName, bool loop, |
| FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst); |
| int StartPlayingFileLocally(InStream* stream, FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst); |
| int StopPlayingFileLocally(); |
| int IsPlayingFileLocally() const; |
| int RegisterFilePlayingToMixer(); |
| int StartPlayingFileAsMicrophone(const char* fileName, bool loop, |
| FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst); |
| int StartPlayingFileAsMicrophone(InStream* stream, |
| FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst); |
| int StopPlayingFileAsMicrophone(); |
| int IsPlayingFileAsMicrophone() const; |
| int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst); |
| int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst); |
| int StopRecordingPlayout(); |
| |
| void SetMixWithMicStatus(bool mix); |
| |
| // VoEExternalMediaProcessing |
| int RegisterExternalMediaProcessing(ProcessingTypes type, |
| VoEMediaProcess& processObject); |
| int DeRegisterExternalMediaProcessing(ProcessingTypes type); |
| int SetExternalMixing(bool enabled); |
| |
| // VoEVolumeControl |
| int GetSpeechOutputLevel(uint32_t& level) const; |
| int GetSpeechOutputLevelFullRange(uint32_t& level) const; |
| int SetMute(bool enable); |
| bool Mute() const; |
| int SetOutputVolumePan(float left, float right); |
| int GetOutputVolumePan(float& left, float& right) const; |
| int SetChannelOutputVolumeScaling(float scaling); |
| int GetChannelOutputVolumeScaling(float& scaling) const; |
| |
| // VoENetEqStats |
| int GetNetworkStatistics(NetworkStatistics& stats); |
| void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
| |
| // VoEVideoSync |
| bool GetDelayEstimate(int* jitter_buffer_delay_ms, |
| int* playout_buffer_delay_ms) const; |
| uint32_t GetDelayEstimate() const; |
| int LeastRequiredDelayMs() const; |
| int SetMinimumPlayoutDelay(int delayMs); |
| int GetPlayoutTimestamp(unsigned int& timestamp); |
| int SetInitTimestamp(unsigned int timestamp); |
| int SetInitSequenceNumber(short sequenceNumber); |
| |
| // VoEVideoSyncExtended |
| int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
| |
| // VoEDtmf |
| int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs, |
| int attenuationDb, bool playDtmfEvent); |
| int SendTelephoneEventInband(unsigned char eventCode, int lengthMs, |
| int attenuationDb, bool playDtmfEvent); |
| int SetSendTelephoneEventPayloadType(unsigned char type); |
| int GetSendTelephoneEventPayloadType(unsigned char& type); |
| |
| // VoEAudioProcessingImpl |
| int UpdateRxVadDetection(AudioFrame& audioFrame); |
| int RegisterRxVadObserver(VoERxVadCallback &observer); |
| int DeRegisterRxVadObserver(); |
| int VoiceActivityIndicator(int &activity); |
| #ifdef WEBRTC_VOICE_ENGINE_AGC |
| int SetRxAgcStatus(bool enable, AgcModes mode); |
| int GetRxAgcStatus(bool& enabled, AgcModes& mode); |
| int SetRxAgcConfig(AgcConfig config); |
| int GetRxAgcConfig(AgcConfig& config); |
| #endif |
| #ifdef WEBRTC_VOICE_ENGINE_NR |
| int SetRxNsStatus(bool enable, NsModes mode); |
| int GetRxNsStatus(bool& enabled, NsModes& mode); |
| #endif |
| |
| // VoERTP_RTCP |
| int SetLocalSSRC(unsigned int ssrc); |
| int GetLocalSSRC(unsigned int& ssrc); |
| int GetRemoteSSRC(unsigned int& ssrc); |
| int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); |
| int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); |
| int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id); |
| int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id); |
| void EnableSendTransportSequenceNumber(int id); |
| |
| void SetCongestionControlObjects( |
| RtpPacketSender* rtp_packet_sender, |
| TransportFeedbackObserver* transport_feedback_observer, |
| PacketRouter* packet_router); |
| |
| void SetRTCPStatus(bool enable); |
| int GetRTCPStatus(bool& enabled); |
| int SetRTCP_CNAME(const char cName[256]); |
| int GetRemoteRTCP_CNAME(char cName[256]); |
| int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow, |
| unsigned int& timestamp, |
| unsigned int& playoutTimestamp, unsigned int* jitter, |
| unsigned short* fractionLost); |
| int SendApplicationDefinedRTCPPacket(unsigned char subType, |
| unsigned int name, const char* data, |
| unsigned short dataLengthInBytes); |
| int GetRTPStatistics(unsigned int& averageJitterMs, |
| unsigned int& maxJitterMs, |
| unsigned int& discardedPackets); |
| int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks); |
| int GetRTPStatistics(CallStatistics& stats); |
| int SetREDStatus(bool enable, int redPayloadtype); |
| int GetREDStatus(bool& enabled, int& redPayloadtype); |
| int SetCodecFECStatus(bool enable); |
| bool GetCodecFECStatus(); |
| void SetNACKStatus(bool enable, int maxNumberOfPackets); |
| |
| // From AudioPacketizationCallback in the ACM |
| int32_t SendData(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| // From ACMVADCallback in the ACM |
| int32_t InFrameType(FrameType frame_type) override; |
| |
| int32_t OnRxVadDetected(int vadDecision); |
| |
| // From RtpData in the RTP/RTCP module |
| int32_t OnReceivedPayloadData(const uint8_t* payloadData, |
| size_t payloadSize, |
| const WebRtcRTPHeader* rtpHeader) override; |
| bool OnRecoveredPacket(const uint8_t* packet, |
| size_t packet_length) override; |
| |
| // From RtpFeedback in the RTP/RTCP module |
| int32_t OnInitializeDecoder(int8_t payloadType, |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| int frequency, |
| uint8_t channels, |
| uint32_t rate) override; |
| void OnIncomingSSRCChanged(uint32_t ssrc) override; |
| void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; |
| |
| // From RtpAudioFeedback in the RTP/RTCP module |
| void OnPlayTelephoneEvent(uint8_t event, |
| uint16_t lengthMs, |
| uint8_t volume) override; |
| |
| // From Transport (called by the RTP/RTCP module) |
| bool SendRtp(const uint8_t* data, |
| size_t len, |
| const PacketOptions& packet_options) override; |
| bool SendRtcp(const uint8_t* data, size_t len) override; |
| |
| // From MixerParticipant |
| int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; |
| int32_t NeededFrequency(int32_t id) const override; |
| |
| // From FileCallback |
| void PlayNotification(int32_t id, uint32_t durationMs) override; |
| void RecordNotification(int32_t id, uint32_t durationMs) override; |
| void PlayFileEnded(int32_t id) override; |
| void RecordFileEnded(int32_t id) override; |
| |
| uint32_t InstanceId() const |
| { |
| return _instanceId; |
| } |
| int32_t ChannelId() const |
| { |
| return _channelId; |
| } |
| bool Playing() const |
| { |
| return channel_state_.Get().playing; |
| } |
| bool Sending() const |
| { |
| return channel_state_.Get().sending; |
| } |
| bool Receiving() const |
| { |
| return channel_state_.Get().receiving; |
| } |
| bool ExternalTransport() const |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| return _externalTransport; |
| } |
| bool ExternalMixing() const |
| { |
| return _externalMixing; |
| } |
| RtpRtcp* RtpRtcpModulePtr() const |
| { |
| return _rtpRtcpModule.get(); |
| } |
| int8_t OutputEnergyLevel() const |
| { |
| return _outputAudioLevel.Level(); |
| } |
| uint32_t Demultiplex(const AudioFrame& audioFrame); |
| // Demultiplex the data to the channel's |_audioFrame|. The difference |
| // between this method and the overloaded method above is that |audio_data| |
| // does not go through transmit_mixer and APM. |
| void Demultiplex(const int16_t* audio_data, |
| int sample_rate, |
| size_t number_of_frames, |
| int number_of_channels); |
| uint32_t PrepareEncodeAndSend(int mixingFrequency); |
| uint32_t EncodeAndSend(); |
| |
| // Associate to a send channel. |
| // Used for obtaining RTT for a receive-only channel. |
| void set_associate_send_channel(const ChannelOwner& channel) { |
| assert(_channelId != channel.channel()->ChannelId()); |
| CriticalSectionScoped lock(assoc_send_channel_lock_.get()); |
| associate_send_channel_ = channel; |
| } |
| |
| // Disassociate a send channel if it was associated. |
| void DisassociateSendChannel(int channel_id); |
| |
| protected: |
| void OnIncomingFractionLoss(int fraction_lost); |
| |
| private: |
| bool ReceivePacket(const uint8_t* packet, size_t packet_length, |
| const RTPHeader& header, bool in_order); |
| bool HandleRtxPacket(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header); |
| bool IsPacketInOrder(const RTPHeader& header) const; |
| bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; |
| int ResendPackets(const uint16_t* sequence_numbers, int length); |
| int InsertInbandDtmfTone(); |
| int32_t MixOrReplaceAudioWithFile(int mixingFrequency); |
| int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); |
| void UpdatePlayoutTimestamp(bool rtcp); |
| void UpdatePacketDelay(uint32_t timestamp, |
| uint16_t sequenceNumber); |
| void RegisterReceiveCodecsToRTPModule(); |
| |
| int SetRedPayloadType(int red_payload_type); |
| int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, |
| unsigned char id); |
| |
| int32_t GetPlayoutFrequency(); |
| int64_t GetRTT(bool allow_associate_channel) const; |
| |
| CriticalSectionWrapper& _fileCritSect; |
| CriticalSectionWrapper& _callbackCritSect; |
| CriticalSectionWrapper& volume_settings_critsect_; |
| uint32_t _instanceId; |
| int32_t _channelId; |
| |
| ChannelState channel_state_; |
| |
| RtcEventLog* const event_log_; |
| |
| rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
| rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; |
| rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_; |
| rtc::scoped_ptr<StatisticsProxy> statistics_proxy_; |
| rtc::scoped_ptr<RtpReceiver> rtp_receiver_; |
| TelephoneEventHandler* telephone_event_handler_; |
| rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule; |
| rtc::scoped_ptr<AudioCodingModule> audio_coding_; |
| rtc::scoped_ptr<AudioSinkInterface> audio_sink_; |
| AudioLevel _outputAudioLevel; |
| bool _externalTransport; |
| AudioFrame _audioFrame; |
| // Downsamples to the codec rate if necessary. |
| PushResampler<int16_t> input_resampler_; |
| FilePlayer* _inputFilePlayerPtr; |
| FilePlayer* _outputFilePlayerPtr; |
| FileRecorder* _outputFileRecorderPtr; |
| int _inputFilePlayerId; |
| int _outputFilePlayerId; |
| int _outputFileRecorderId; |
| bool _outputFileRecording; |
| DtmfInbandQueue _inbandDtmfQueue; |
| DtmfInband _inbandDtmfGenerator; |
| bool _outputExternalMedia; |
| VoEMediaProcess* _inputExternalMediaCallbackPtr; |
| VoEMediaProcess* _outputExternalMediaCallbackPtr; |
| uint32_t _timeStamp; |
| uint8_t _sendTelephoneEventPayloadType; |
| |
| RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); |
| |
| // Timestamp of the audio pulled from NetEq. |
| uint32_t jitter_buffer_playout_timestamp_; |
| uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_); |
| uint32_t playout_timestamp_rtcp_; |
| uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_); |
| uint32_t _numberOfDiscardedPackets; |
| uint16_t send_sequence_number_; |
| uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes]; |
| |
| rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_; |
| |
| rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; |
| // The rtp timestamp of the first played out audio frame. |
| int64_t capture_start_rtp_time_stamp_; |
| // The capture ntp time (in local timebase) of the first played out audio |
| // frame. |
| int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_); |
| |
| // uses |
| Statistics* _engineStatisticsPtr; |
| OutputMixer* _outputMixerPtr; |
| TransmitMixer* _transmitMixerPtr; |
| ProcessThread* _moduleProcessThreadPtr; |
| AudioDeviceModule* _audioDeviceModulePtr; |
| VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base |
| CriticalSectionWrapper* _callbackCritSectPtr; // owned by base |
| Transport* _transportPtr; // WebRtc socket or external transport |
| RMSLevel rms_level_; |
| rtc::scoped_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing |
| VoERxVadCallback* _rxVadObserverPtr; |
| int32_t _oldVadDecision; |
| int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise |
| // VoEBase |
| bool _externalMixing; |
| bool _mixFileWithMicrophone; |
| // VoEVolumeControl |
| bool _mute; |
| float _panLeft; |
| float _panRight; |
| float _outputGain; |
| // VoEDtmf |
| bool _playOutbandDtmfEvent; |
| bool _playInbandDtmfEvent; |
| // VoeRTP_RTCP |
| uint32_t _lastLocalTimeStamp; |
| int8_t _lastPayloadType; |
| bool _includeAudioLevelIndication; |
| // VoENetwork |
| AudioFrame::SpeechType _outputSpeechType; |
| // VoEVideoSync |
| rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_; |
| uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_); |
| uint32_t _previousTimestamp; |
| uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_); |
| // VoEAudioProcessing |
| bool _RxVadDetection; |
| bool _rxAgcIsEnabled; |
| bool _rxNsIsEnabled; |
| bool restored_packet_in_use_; |
| // RtcpBandwidthObserver |
| rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_; |
| rtc::scoped_ptr<NetworkPredictor> network_predictor_; |
| // An associated send channel. |
| rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; |
| ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
| |
| bool pacing_enabled_; |
| PacketRouter* packet_router_ = nullptr; |
| rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
| rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
| rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
| }; |
| |
| } // namespace voe |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
