| /* |
| * 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 <memory> |
| |
| #include "webrtc/api/audio/audio_mixer.h" |
| #include "webrtc/api/audio_codecs/audio_encoder.h" |
| #include "webrtc/api/call/audio_sink.h" |
| #include "webrtc/common_audio/resampler/include/push_resampler.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
| #include "webrtc/modules/audio_coding/acm2/rent_a_codec.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_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/rtc_base/event.h" |
| #include "webrtc/rtc_base/optional.h" |
| #include "webrtc/rtc_base/thread_checker.h" |
| #include "webrtc/voice_engine/audio_level.h" |
| #include "webrtc/voice_engine/file_player.h" |
| #include "webrtc/voice_engine/file_recorder.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| #include "webrtc/voice_engine/include/voe_network.h" |
| #include "webrtc/voice_engine/shared_data.h" |
| #include "webrtc/voice_engine/voice_engine_defines.h" |
| |
| namespace rtc { |
| class TimestampWrapAroundHandler; |
| } |
| |
| namespace webrtc { |
| |
| class AudioDeviceModule; |
| class FileWrapper; |
| class PacketRouter; |
| class ProcessThread; |
| class RateLimiter; |
| class ReceiveStatistics; |
| class RemoteNtpTimeEstimator; |
| class RtcEventLog; |
| class RTPPayloadRegistry; |
| class RTPReceiverAudio; |
| class RtpPacketReceived; |
| class RtpRtcp; |
| class RtpTransportControllerSendInterface; |
| class TelephoneEventHandler; |
| class VoERTPObserver; |
| class VoiceEngineObserver; |
| |
| struct CallStatistics; |
| struct ReportBlock; |
| struct SenderInfo; |
| |
| namespace voe { |
| |
| class OutputMixer; |
| class RtcEventLogProxy; |
| class RtcpRttStatsProxy; |
| class RtpPacketSenderProxy; |
| class Statistics; |
| class TransportFeedbackProxy; |
| 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 { |
| bool output_file_playing = false; |
| bool input_file_playing = false; |
| bool playing = false; |
| bool sending = false; |
| }; |
| |
| ChannelState() {} |
| virtual ~ChannelState() {} |
| |
| void Reset() { |
| rtc::CritScope lock(&lock_); |
| state_ = State(); |
| } |
| |
| State Get() const { |
| rtc::CritScope lock(&lock_); |
| return state_; |
| } |
| |
| void SetOutputFilePlaying(bool enable) { |
| rtc::CritScope lock(&lock_); |
| state_.output_file_playing = enable; |
| } |
| |
| void SetInputFilePlaying(bool enable) { |
| rtc::CritScope lock(&lock_); |
| state_.input_file_playing = enable; |
| } |
| |
| void SetPlaying(bool enable) { |
| rtc::CritScope lock(&lock_); |
| state_.playing = enable; |
| } |
| |
| void SetSending(bool enable) { |
| rtc::CritScope lock(&lock_); |
| state_.sending = enable; |
| } |
| |
| private: |
| rtc::CriticalSection lock_; |
| State state_; |
| }; |
| |
| class Channel |
| : public RtpData, |
| public RtpFeedback, |
| public FileCallback, // receiving notification from file player & |
| // recorder |
| public Transport, |
| public AudioPacketizationCallback, // receive encoded packets from the |
| // ACM |
| public MixerParticipant, // supplies output mixer with audio frames |
| public OverheadObserver { |
| public: |
| friend class VoERtcpObserver; |
| |
| enum { KNumSocketThreads = 1 }; |
| enum { KNumberOfSocketBuffers = 8 }; |
| virtual ~Channel(); |
| static int32_t CreateChannel(Channel*& channel, |
| int32_t channelId, |
| uint32_t instanceId, |
| const VoEBase::ChannelConfig& config); |
| Channel(int32_t channelId, |
| uint32_t instanceId, |
| const VoEBase::ChannelConfig& config); |
| int32_t Init(); |
| void RegisterLegacyReceiveCodecs(); |
| void Terminate(); |
| int32_t SetEngineInformation(Statistics& engineStatistics, |
| OutputMixer& outputMixer, |
| ProcessThread& moduleProcessThread, |
| AudioDeviceModule& audioDeviceModule, |
| VoiceEngineObserver* voiceEngineObserver, |
| rtc::CriticalSection* callbackCritSect, |
| rtc::TaskQueue* encoder_queue); |
| |
| void SetSink(std::unique_ptr<AudioSinkInterface> sink); |
| |
| // TODO(ossu): Don't use! It's only here to confirm that the decoder factory |
| // passed into AudioReceiveStream is the same as the one set when creating the |
| // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can |
| // go. |
| const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; |
| |
| void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs); |
| |
| // Send using this encoder, with this payload type. |
| bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder); |
| void ModifyEncoder( |
| rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier); |
| |
| // API methods |
| |
| // VoEBase |
| int32_t StartPlayout(); |
| int32_t StopPlayout(); |
| int32_t StartSend(); |
| void StopSend(); |
| 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, int64_t probing_interval_ms); |
| 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 SetRecPayloadType(int payload_type, const SdpAudioFormat& format); |
| int32_t GetRecPayloadType(CodecInst& codec); |
| int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency); |
| int SetOpusMaxPlaybackRate(int frequency_hz); |
| int SetOpusDtx(bool enable_dtx); |
| int GetOpusDtx(bool* enabled); |
| bool EnableAudioNetworkAdaptor(const std::string& config_string); |
| void DisableAudioNetworkAdaptor(); |
| void SetReceiverFrameLengthRange(int min_frame_length_ms, |
| int max_frame_length_ms); |
| |
| // VoENetwork |
| int32_t RegisterExternalTransport(Transport* transport); |
| int32_t DeRegisterExternalTransport(); |
| int32_t ReceivedRTPPacket(const uint8_t* received_packet, |
| size_t length, |
| const PacketTime& packet_time); |
| // TODO(nisse, solenberg): Delete when VoENetwork is deleted. |
| int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length); |
| void OnRtpPacket(const RtpPacketReceived& packet); |
| |
| // 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); |
| |
| // Muting, Volume and Level. |
| void SetInputMute(bool enable); |
| void SetChannelOutputVolumeScaling(float scaling); |
| int GetSpeechOutputLevel() const; |
| int GetSpeechOutputLevelFullRange() const; |
| // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy |
| double GetTotalOutputEnergy() const; |
| double GetTotalOutputDuration() const; |
| |
| // Stats. |
| int GetNetworkStatistics(NetworkStatistics& stats); |
| void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
| |
| // Audio+Video Sync. |
| uint32_t GetDelayEstimate() const; |
| int SetMinimumPlayoutDelay(int delayMs); |
| int GetPlayoutTimestamp(unsigned int& timestamp); |
| int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
| |
| // DTMF. |
| int SendTelephoneEventOutband(int event, int duration_ms); |
| int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); |
| |
| // 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); |
| void EnableSendTransportSequenceNumber(int id); |
| void EnableReceiveTransportSequenceNumber(int id); |
| |
| void RegisterSenderCongestionControlObjects( |
| RtpTransportControllerSendInterface* transport, |
| RtcpBandwidthObserver* bandwidth_observer); |
| void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router); |
| void ResetSenderCongestionControlObjects(); |
| void ResetReceiverCongestionControlObjects(); |
| void SetRTCPStatus(bool enable); |
| int GetRTCPStatus(bool& enabled); |
| int SetRTCP_CNAME(const char cName[256]); |
| int GetRemoteRTCP_CNAME(char cName[256]); |
| int SendApplicationDefinedRTCPPacket(unsigned char subType, |
| unsigned int name, |
| const char* data, |
| unsigned short dataLengthInBytes); |
| int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks); |
| int GetRTPStatistics(CallStatistics& stats); |
| 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 RtpData in the RTP/RTCP module |
| int32_t OnReceivedPayloadData(const uint8_t* payloadData, |
| size_t payloadSize, |
| const WebRtcRTPHeader* rtpHeader) override; |
| |
| // From RtpFeedback in the RTP/RTCP module |
| int32_t OnInitializeDecoder(int8_t payloadType, |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| int frequency, |
| size_t channels, |
| uint32_t rate) override; |
| void OnIncomingSSRCChanged(uint32_t ssrc) override; |
| void OnIncomingCSRCChanged(uint32_t CSRC, bool added) 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 |
| MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted( |
| int32_t id, |
| AudioFrame* audioFrame) override; |
| int32_t NeededFrequency(int32_t id) const override; |
| |
| // From AudioMixer::Source. |
| AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
| int sample_rate_hz, |
| AudioFrame* audio_frame); |
| |
| // 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 ExternalTransport() const { |
| rtc::CritScope cs(&_callbackCritSect); |
| return _externalTransport; |
| } |
| RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } |
| int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); } |
| |
| // ProcessAndEncodeAudio() creates an audio frame copy and posts a task |
| // on the shared encoder task queue, wich in turn calls (on the queue) |
| // ProcessAndEncodeAudioOnTaskQueue() where the actual processing of the |
| // audio takes place. The processing mainly consists of encoding and preparing |
| // the result for sending by adding it to a send queue. |
| // The main reason for using a task queue here is to release the native, |
| // OS-specific, audio capture thread as soon as possible to ensure that it |
| // can go back to sleep and be prepared to deliver an new captured audio |
| // packet. |
| void ProcessAndEncodeAudio(const AudioFrame& audio_input); |
| |
| // This version of ProcessAndEncodeAudio() is used by PushCaptureData() in |
| // VoEBase and the audio in |audio_data| has not been subject to any APM |
| // processing. Some extra steps are therfore needed when building up the |
| // audio frame copy before using the same task as in the default call to |
| // ProcessAndEncodeAudio(const AudioFrame& audio_input). |
| void ProcessAndEncodeAudio(const int16_t* audio_data, |
| int sample_rate, |
| size_t number_of_frames, |
| size_t number_of_channels); |
| |
| // Associate to a send channel. |
| // Used for obtaining RTT for a receive-only channel. |
| void set_associate_send_channel(const ChannelOwner& channel); |
| // Disassociate a send channel if it was associated. |
| void DisassociateSendChannel(int channel_id); |
| |
| // Set a RtcEventLog logging object. |
| void SetRtcEventLog(RtcEventLog* event_log); |
| |
| void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats); |
| void SetTransportOverhead(size_t transport_overhead_per_packet); |
| |
| // From OverheadObserver in the RTP/RTCP module |
| void OnOverheadChanged(size_t overhead_bytes_per_packet) override; |
| |
| // The existence of this function alongside OnUplinkPacketLossRate is |
| // a compromise. We want the encoder to be agnostic of the PLR source, but |
| // we also don't want it to receive conflicting information from TWCC and |
| // from RTCP-XR. |
| void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate); |
| |
| void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate); |
| |
| std::vector<RtpSource> GetSources() const { |
| return rtp_receiver_->GetSources(); |
| } |
| |
| private: |
| class ProcessAndEncodeAudioTask; |
| |
| void OnUplinkPacketLossRate(float packet_loss_rate); |
| bool InputMute() const; |
| bool OnRtpPacketWithHeader(const uint8_t* received_packet, |
| size_t length, |
| RTPHeader *header); |
| bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length); |
| |
| 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); |
| int32_t MixOrReplaceAudioWithFile(AudioFrame* audio_frame); |
| int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); |
| void UpdatePlayoutTimestamp(bool rtcp); |
| void RegisterReceiveCodecsToRTPModule(); |
| |
| int SetSendRtpHeaderExtension(bool enable, |
| RTPExtensionType type, |
| unsigned char id); |
| |
| void UpdateOverheadForEncoder() |
| EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); |
| |
| int GetRtpTimestampRateHz() const; |
| int64_t GetRTT(bool allow_associate_channel) const; |
| |
| // Called on the encoder task queue when a new input audio frame is ready |
| // for encoding. |
| void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input); |
| |
| uint32_t _instanceId; |
| int32_t _channelId; |
| |
| rtc::CriticalSection _fileCritSect; |
| rtc::CriticalSection _callbackCritSect; |
| rtc::CriticalSection volume_settings_critsect_; |
| |
| ChannelState channel_state_; |
| |
| std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_; |
| std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_; |
| |
| std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
| std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; |
| std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; |
| std::unique_ptr<RtpReceiver> rtp_receiver_; |
| TelephoneEventHandler* telephone_event_handler_; |
| std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
| std::unique_ptr<AudioCodingModule> audio_coding_; |
| acm2::CodecManager codec_manager_; |
| acm2::RentACodec rent_a_codec_; |
| std::unique_ptr<AudioSinkInterface> audio_sink_; |
| AudioLevel _outputAudioLevel; |
| bool _externalTransport; |
| // Downsamples to the codec rate if necessary. |
| PushResampler<int16_t> input_resampler_; |
| std::unique_ptr<FilePlayer> input_file_player_; |
| std::unique_ptr<FilePlayer> output_file_player_; |
| std::unique_ptr<FileRecorder> output_file_recorder_; |
| int _inputFilePlayerId; |
| int _outputFilePlayerId; |
| int _outputFileRecorderId; |
| bool _outputFileRecording; |
| uint32_t _timeStamp ACCESS_ON(encoder_queue_); |
| |
| RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_); |
| |
| // Timestamp of the audio pulled from NetEq. |
| rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_; |
| |
| rtc::CriticalSection video_sync_lock_; |
| uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_); |
| uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_); |
| uint16_t send_sequence_number_; |
| uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes]; |
| |
| rtc::CriticalSection ts_stats_lock_; |
| |
| std::unique_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; |
| ProcessThread* _moduleProcessThreadPtr; |
| AudioDeviceModule* _audioDeviceModulePtr; |
| VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base |
| rtc::CriticalSection* _callbackCritSectPtr; // owned by base |
| Transport* _transportPtr; // WebRtc socket or external transport |
| RmsLevel rms_level_ ACCESS_ON(encoder_queue_); |
| bool input_mute_ GUARDED_BY(volume_settings_critsect_); |
| bool previous_frame_muted_ ACCESS_ON(encoder_queue_); |
| float _outputGain GUARDED_BY(volume_settings_critsect_); |
| // VoEBase |
| bool _mixFileWithMicrophone; |
| // VoeRTP_RTCP |
| // TODO(henrika): can today be accessed on the main thread and on the |
| // task queue; hence potential race. |
| bool _includeAudioLevelIndication; |
| size_t transport_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_); |
| size_t rtp_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_); |
| rtc::CriticalSection overhead_per_packet_lock_; |
| // VoENetwork |
| AudioFrame::SpeechType _outputSpeechType; |
| // DTX. |
| bool restored_packet_in_use_; |
| // RtcpBandwidthObserver |
| std::unique_ptr<VoERtcpObserver> rtcp_observer_; |
| // An associated send channel. |
| rtc::CriticalSection assoc_send_channel_lock_; |
| ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
| |
| bool pacing_enabled_; |
| PacketRouter* packet_router_ = nullptr; |
| std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
| std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
| std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
| std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
| |
| // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
| |
| rtc::Optional<CodecInst> cached_send_codec_; |
| |
| rtc::ThreadChecker construction_thread_; |
| |
| const bool use_twcc_plr_for_ana_; |
| |
| rtc::CriticalSection encoder_queue_lock_; |
| |
| bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; |
| |
| rtc::TaskQueue* encoder_queue_ = nullptr; |
| }; |
| |
| } // namespace voe |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |