| _mixHistory | webrtc::MixerParticipant | |
| AudioFrameInfo enum name | webrtc::MixerParticipant | |
| AudioFrameInfo enum name | webrtc::MixerParticipant | |
| Channel(int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config) | webrtc::voe::Channel | |
| Channel(int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config) | webrtc::voe::Channel | |
| ChannelId() const | webrtc::voe::Channel | inline |
| ChannelId() const | webrtc::voe::Channel | inline |
| CreateChannel(Channel *&channel, int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config) | webrtc::voe::Channel | static |
| CreateChannel(Channel *&channel, int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config) | webrtc::voe::Channel | static |
| Demultiplex(const AudioFrame &audioFrame) | webrtc::voe::Channel | |
| Demultiplex(const int16_t *audio_data, int sample_rate, size_t number_of_frames, size_t number_of_channels) | webrtc::voe::Channel | |
| Demultiplex(const AudioFrame &audioFrame) | webrtc::voe::Channel | |
| Demultiplex(const int16_t *audio_data, int sample_rate, size_t number_of_frames, size_t number_of_channels) | webrtc::voe::Channel | |
| DeRegisterExternalTransport() | webrtc::voe::Channel | |
| DeRegisterExternalTransport() | webrtc::voe::Channel | |
| DeRegisterVoiceEngineObserver() | webrtc::voe::Channel | |
| DeRegisterVoiceEngineObserver() | webrtc::voe::Channel | |
| DisableAudioNetworkAdaptor() | webrtc::voe::Channel | |
| DisableAudioNetworkAdaptor() | webrtc::voe::Channel | |
| DisassociateSendChannel(int channel_id) | webrtc::voe::Channel | |
| DisassociateSendChannel(int channel_id) | webrtc::voe::Channel | |
| EnableAudioNetworkAdaptor(const std::string &config_string) | webrtc::voe::Channel | |
| EnableAudioNetworkAdaptor(const std::string &config_string) | webrtc::voe::Channel | |
| EnableReceiveTransportSequenceNumber(int id) | webrtc::voe::Channel | |
| EnableReceiveTransportSequenceNumber(int id) | webrtc::voe::Channel | |
| EnableSendTransportSequenceNumber(int id) | webrtc::voe::Channel | |
| EnableSendTransportSequenceNumber(int id) | webrtc::voe::Channel | |
| EncodeAndSend() | webrtc::voe::Channel | |
| EncodeAndSend() | webrtc::voe::Channel | |
| ExternalTransport() const | webrtc::voe::Channel | inline |
| ExternalTransport() const | webrtc::voe::Channel | inline |
| FileCallback() | webrtc::FileCallback | inlineprotected |
| FileCallback() | webrtc::FileCallback | inlineprotected |
| GetAudioDecoderFactory() const | webrtc::voe::Channel | |
| GetAudioDecoderFactory() const | webrtc::voe::Channel | |
| GetAudioFrame(int32_t id, AudioFrame *audioFrame) | webrtc::MixerParticipant | inlinevirtual |
| GetAudioFrame(int32_t id, AudioFrame *audioFrame) | webrtc::MixerParticipant | inlinevirtual |
| GetAudioFrameWithInfo(int sample_rate_hz, AudioFrame *audio_frame) | webrtc::voe::Channel | |
| GetAudioFrameWithInfo(int sample_rate_hz, AudioFrame *audio_frame) | webrtc::voe::Channel | |
| GetAudioFrameWithMuted(int32_t id, AudioFrame *audioFrame) override | webrtc::voe::Channel | virtual |
| GetAudioFrameWithMuted(int32_t id, AudioFrame *audioFrame) override | webrtc::voe::Channel | virtual |
| GetChannelOutputVolumeScaling(float &scaling) const | webrtc::voe::Channel | |
| GetChannelOutputVolumeScaling(float &scaling) const | webrtc::voe::Channel | |
| GetCodecFECStatus() | webrtc::voe::Channel | |
| GetCodecFECStatus() | webrtc::voe::Channel | |
| GetDecodingCallStatistics(AudioDecodingCallStats *stats) const | webrtc::voe::Channel | |
| GetDecodingCallStatistics(AudioDecodingCallStats *stats) const | webrtc::voe::Channel | |
| GetDelayEstimate() const | webrtc::voe::Channel | |
| GetDelayEstimate() const | webrtc::voe::Channel | |
| GetLocalSSRC(unsigned int &ssrc) | webrtc::voe::Channel | |
| GetLocalSSRC(unsigned int &ssrc) | webrtc::voe::Channel | |
| GetNetworkStatistics(NetworkStatistics &stats) | webrtc::voe::Channel | |
| GetNetworkStatistics(NetworkStatistics &stats) | webrtc::voe::Channel | |
| GetOpusDtx(bool *enabled) | webrtc::voe::Channel | |
| GetOpusDtx(bool *enabled) | webrtc::voe::Channel | |
| GetOutputVolumePan(float &left, float &right) const | webrtc::voe::Channel | |
| GetOutputVolumePan(float &left, float &right) const | webrtc::voe::Channel | |
| GetPlayoutTimestamp(unsigned int ×tamp) | webrtc::voe::Channel | |
| GetPlayoutTimestamp(unsigned int ×tamp) | webrtc::voe::Channel | |
| GetRecCodec(CodecInst &codec) | webrtc::voe::Channel | |
| GetRecCodec(CodecInst &codec) | webrtc::voe::Channel | |
| GetRecPayloadType(CodecInst &codec) | webrtc::voe::Channel | |
| GetRecPayloadType(CodecInst &codec) | webrtc::voe::Channel | |
| GetRemoteRTCP_CNAME(char cName[256]) | webrtc::voe::Channel | |
| GetRemoteRTCP_CNAME(char cName[256]) | webrtc::voe::Channel | |
| GetRemoteRTCPReportBlocks(std::vector< ReportBlock > *report_blocks) | webrtc::voe::Channel | |
| GetRemoteRTCPReportBlocks(std::vector< ReportBlock > *report_blocks) | webrtc::voe::Channel | |
| GetRemoteSSRC(unsigned int &ssrc) | webrtc::voe::Channel | |
| GetRemoteSSRC(unsigned int &ssrc) | webrtc::voe::Channel | |
| GetRTCPStatus(bool &enabled) | webrtc::voe::Channel | |
| GetRTCPStatus(bool &enabled) | webrtc::voe::Channel | |
| GetRtpRtcp(RtpRtcp **rtpRtcpModule, RtpReceiver **rtp_receiver) const | webrtc::voe::Channel | |
| GetRtpRtcp(RtpRtcp **rtpRtcpModule, RtpReceiver **rtp_receiver) const | webrtc::voe::Channel | |
| GetRTPStatistics(CallStatistics &stats) | webrtc::voe::Channel | |
| GetRTPStatistics(CallStatistics &stats) | webrtc::voe::Channel | |
| GetSendCodec(CodecInst &codec) | webrtc::voe::Channel | |
| GetSendCodec(CodecInst &codec) | webrtc::voe::Channel | |
| GetSpeechOutputLevel(uint32_t &level) const | webrtc::voe::Channel | |
| GetSpeechOutputLevel(uint32_t &level) const | webrtc::voe::Channel | |
| GetSpeechOutputLevelFullRange(uint32_t &level) const | webrtc::voe::Channel | |
| GetSpeechOutputLevelFullRange(uint32_t &level) const | webrtc::voe::Channel | |
| GetVADStatus(bool &enabledVAD, ACMVADMode &mode, bool &disabledDTX) | webrtc::voe::Channel | |
| GetVADStatus(bool &enabledVAD, ACMVADMode &mode, bool &disabledDTX) | webrtc::voe::Channel | |
| InFrameType(FrameType frame_type) override | webrtc::voe::Channel | virtual |
| InFrameType(FrameType frame_type) override | webrtc::voe::Channel | virtual |
| Init() | webrtc::voe::Channel | |
| Init() | webrtc::voe::Channel | |
| InputMute() const | webrtc::voe::Channel | |
| InputMute() const | webrtc::voe::Channel | |
| InstanceId() const | webrtc::voe::Channel | inline |
| InstanceId() const | webrtc::voe::Channel | inline |
| IsMixed() const | webrtc::MixerParticipant | |
| IsMixed() const | webrtc::MixerParticipant | |
| IsPlayingFileAsMicrophone() const | webrtc::voe::Channel | |
| IsPlayingFileAsMicrophone() const | webrtc::voe::Channel | |
| IsPlayingFileLocally() const | webrtc::voe::Channel | |
| IsPlayingFileLocally() const | webrtc::voe::Channel | |
| KNumberOfSocketBuffers enum value | webrtc::voe::Channel | |
| KNumSocketThreads enum value | webrtc::voe::Channel | |
| MixerParticipant() | webrtc::MixerParticipant | protected |
| MixerParticipant() | webrtc::MixerParticipant | protected |
| NeededFrequency(int32_t id) const override | webrtc::voe::Channel | virtual |
| NeededFrequency(int32_t id) const override | webrtc::voe::Channel | virtual |
| OnIncomingCSRCChanged(uint32_t CSRC, bool added) override | webrtc::voe::Channel | virtual |
| OnIncomingCSRCChanged(uint32_t CSRC, bool added) override | webrtc::voe::Channel | virtual |
| OnIncomingFractionLoss(int fraction_lost) | webrtc::voe::Channel | protected |
| OnIncomingFractionLoss(int fraction_lost) | webrtc::voe::Channel | protected |
| OnIncomingSSRCChanged(uint32_t ssrc) override | webrtc::voe::Channel | virtual |
| OnIncomingSSRCChanged(uint32_t ssrc) override | webrtc::voe::Channel | virtual |
| OnInitializeDecoder(int8_t payloadType, const char payloadName[RTP_PAYLOAD_NAME_SIZE], int frequency, size_t channels, uint32_t rate) override | webrtc::voe::Channel | virtual |
| OnInitializeDecoder(int8_t payloadType, const char payloadName[RTP_PAYLOAD_NAME_SIZE], int frequency, size_t channels, uint32_t rate) override | webrtc::voe::Channel | virtual |
| OnOverheadChanged(size_t overhead_bytes_per_packet) override | webrtc::voe::Channel | virtual |
| OnOverheadChanged(size_t overhead_bytes_per_packet) override | webrtc::voe::Channel | virtual |
| OnReceivedPayloadData(const uint8_t *payloadData, size_t payloadSize, const WebRtcRTPHeader *rtpHeader) override | webrtc::voe::Channel | virtual |
| OnReceivedPayloadData(const uint8_t *payloadData, size_t payloadSize, const WebRtcRTPHeader *rtpHeader) override | webrtc::voe::Channel | virtual |
| OnRecoveredPacket(const uint8_t *packet, size_t packet_length) override | webrtc::voe::Channel | virtual |
| OnRecoveredPacket(const uint8_t *packet, size_t packet_length) override | webrtc::voe::Channel | virtual |
| OnRtpPacket(const RtpPacketReceived &packet) | webrtc::voe::Channel | |
| OnRtpPacket(const RtpPacketReceived &packet) | webrtc::voe::Channel | |
| OutputEnergyLevel() const | webrtc::voe::Channel | inline |
| OutputEnergyLevel() const | webrtc::voe::Channel | inline |
| PlayFileEnded(int32_t id) override | webrtc::voe::Channel | virtual |
| PlayFileEnded(int32_t id) override | webrtc::voe::Channel | virtual |
| Playing() const | webrtc::voe::Channel | inline |
| Playing() const | webrtc::voe::Channel | inline |
| PlayNotification(int32_t id, uint32_t durationMs) override | webrtc::voe::Channel | virtual |
| PlayNotification(int32_t id, uint32_t durationMs) override | webrtc::voe::Channel | virtual |
| PrepareEncodeAndSend(int mixingFrequency) | webrtc::voe::Channel | |
| PrepareEncodeAndSend(int mixingFrequency) | webrtc::voe::Channel | |
| ReceivedRTCPPacket(const uint8_t *data, size_t length) | webrtc::voe::Channel | |
| ReceivedRTCPPacket(const uint8_t *data, size_t length) | webrtc::voe::Channel | |
| ReceivedRTPPacket(const uint8_t *received_packet, size_t length, const PacketTime &packet_time) | webrtc::voe::Channel | |
| ReceivedRTPPacket(const uint8_t *received_packet, size_t length, const PacketTime &packet_time) | webrtc::voe::Channel | |
| RecordFileEnded(int32_t id) override | webrtc::voe::Channel | virtual |
| RecordFileEnded(int32_t id) override | webrtc::voe::Channel | virtual |
| RecordNotification(int32_t id, uint32_t durationMs) override | webrtc::voe::Channel | virtual |
| RecordNotification(int32_t id, uint32_t durationMs) override | webrtc::voe::Channel | virtual |
| RegisterExternalTransport(Transport *transport) | webrtc::voe::Channel | |
| RegisterExternalTransport(Transport *transport) | webrtc::voe::Channel | |
| RegisterFilePlayingToMixer() | webrtc::voe::Channel | |
| RegisterFilePlayingToMixer() | webrtc::voe::Channel | |
| RegisterReceiverCongestionControlObjects(PacketRouter *packet_router) | webrtc::voe::Channel | |
| RegisterReceiverCongestionControlObjects(PacketRouter *packet_router) | webrtc::voe::Channel | |
| RegisterSenderCongestionControlObjects(RtpPacketSender *rtp_packet_sender, TransportFeedbackObserver *transport_feedback_observer, PacketRouter *packet_router, RtcpBandwidthObserver *bandwidth_observer) | webrtc::voe::Channel | |
| RegisterSenderCongestionControlObjects(RtpPacketSender *rtp_packet_sender, TransportFeedbackObserver *transport_feedback_observer, PacketRouter *packet_router, RtcpBandwidthObserver *bandwidth_observer) | webrtc::voe::Channel | |
| RegisterVoiceEngineObserver(VoiceEngineObserver &observer) | webrtc::voe::Channel | |
| RegisterVoiceEngineObserver(VoiceEngineObserver &observer) | webrtc::voe::Channel | |
| ResetCongestionControlObjects() | webrtc::voe::Channel | |
| ResetCongestionControlObjects() | webrtc::voe::Channel | |
| RtpRtcpModulePtr() const | webrtc::voe::Channel | inline |
| RtpRtcpModulePtr() const | webrtc::voe::Channel | inline |
| SendApplicationDefinedRTCPPacket(unsigned char subType, unsigned int name, const char *data, unsigned short dataLengthInBytes) | webrtc::voe::Channel | |
| SendApplicationDefinedRTCPPacket(unsigned char subType, unsigned int name, const char *data, unsigned short dataLengthInBytes) | webrtc::voe::Channel | |
| SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override | webrtc::voe::Channel | virtual |
| SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override | webrtc::voe::Channel | virtual |
| Sending() const | webrtc::voe::Channel | inline |
| Sending() const | webrtc::voe::Channel | inline |
| SendRtcp(const uint8_t *data, size_t len) override | webrtc::voe::Channel | virtual |
| SendRtcp(const uint8_t *data, size_t len) override | webrtc::voe::Channel | virtual |
| SendRtp(const uint8_t *data, size_t len, const PacketOptions &packet_options) override | webrtc::voe::Channel | virtual |
| SendRtp(const uint8_t *data, size_t len, const PacketOptions &packet_options) override | webrtc::voe::Channel | virtual |
| SendTelephoneEventOutband(int event, int duration_ms) | webrtc::voe::Channel | |
| SendTelephoneEventOutband(int event, int duration_ms) | webrtc::voe::Channel | |
| set_associate_send_channel(const ChannelOwner &channel) | webrtc::voe::Channel | |
| set_associate_send_channel(const ChannelOwner &channel) | webrtc::voe::Channel | |
| SetBitRate(int bitrate_bps, int64_t probing_interval_ms) | webrtc::voe::Channel | |
| SetBitRate(int bitrate_bps, int64_t probing_interval_ms) | webrtc::voe::Channel | |
| SetChannelOutputVolumeScaling(float scaling) | webrtc::voe::Channel | |
| SetChannelOutputVolumeScaling(float scaling) | webrtc::voe::Channel | |
| SetCodecFECStatus(bool enable) | webrtc::voe::Channel | |
| SetCodecFECStatus(bool enable) | webrtc::voe::Channel | |
| SetEngineInformation(Statistics &engineStatistics, OutputMixer &outputMixer, ProcessThread &moduleProcessThread, AudioDeviceModule &audioDeviceModule, VoiceEngineObserver *voiceEngineObserver, rtc::CriticalSection *callbackCritSect) | webrtc::voe::Channel | |
| SetEngineInformation(Statistics &engineStatistics, OutputMixer &outputMixer, ProcessThread &moduleProcessThread, AudioDeviceModule &audioDeviceModule, VoiceEngineObserver *voiceEngineObserver, rtc::CriticalSection *callbackCritSect) | webrtc::voe::Channel | |
| SetInputMute(bool enable) | webrtc::voe::Channel | |
| SetInputMute(bool enable) | webrtc::voe::Channel | |
| SetLocalSSRC(unsigned int ssrc) | webrtc::voe::Channel | |
| SetLocalSSRC(unsigned int ssrc) | webrtc::voe::Channel | |
| SetMinimumPlayoutDelay(int delayMs) | webrtc::voe::Channel | |
| SetMinimumPlayoutDelay(int delayMs) | webrtc::voe::Channel | |
| SetMixWithMicStatus(bool mix) | webrtc::voe::Channel | |
| SetMixWithMicStatus(bool mix) | webrtc::voe::Channel | |
| SetNACKStatus(bool enable, int maxNumberOfPackets) | webrtc::voe::Channel | |
| SetNACKStatus(bool enable, int maxNumberOfPackets) | webrtc::voe::Channel | |
| SetOpusDtx(bool enable_dtx) | webrtc::voe::Channel | |
| SetOpusDtx(bool enable_dtx) | webrtc::voe::Channel | |
| SetOpusMaxPlaybackRate(int frequency_hz) | webrtc::voe::Channel | |
| SetOpusMaxPlaybackRate(int frequency_hz) | webrtc::voe::Channel | |
| SetOutputVolumePan(float left, float right) | webrtc::voe::Channel | |
| SetOutputVolumePan(float left, float right) | webrtc::voe::Channel | |
| SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id) | webrtc::voe::Channel | |
| SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id) | webrtc::voe::Channel | |
| SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) | webrtc::voe::Channel | |
| SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) | webrtc::voe::Channel | |
| SetRecPayloadType(const CodecInst &codec) | webrtc::voe::Channel | |
| SetRecPayloadType(int payload_type, const SdpAudioFormat &format) | webrtc::voe::Channel | |
| SetRecPayloadType(const CodecInst &codec) | webrtc::voe::Channel | |
| SetRecPayloadType(int payload_type, const SdpAudioFormat &format) | webrtc::voe::Channel | |
| SetRtcEventLog(RtcEventLog *event_log) | webrtc::voe::Channel | |
| SetRtcEventLog(RtcEventLog *event_log) | webrtc::voe::Channel | |
| SetRTCP_CNAME(const char cName[256]) | webrtc::voe::Channel | |
| SetRTCP_CNAME(const char cName[256]) | webrtc::voe::Channel | |
| SetRtcpRttStats(RtcpRttStats *rtcp_rtt_stats) | webrtc::voe::Channel | |
| SetRtcpRttStats(RtcpRttStats *rtcp_rtt_stats) | webrtc::voe::Channel | |
| SetRTCPStatus(bool enable) | webrtc::voe::Channel | |
| SetRTCPStatus(bool enable) | webrtc::voe::Channel | |
| SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) | webrtc::voe::Channel | |
| SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) | webrtc::voe::Channel | |
| SetSendCNPayloadType(int type, PayloadFrequencies frequency) | webrtc::voe::Channel | |
| SetSendCNPayloadType(int type, PayloadFrequencies frequency) | webrtc::voe::Channel | |
| SetSendCodec(const CodecInst &codec) | webrtc::voe::Channel | |
| SetSendCodec(const CodecInst &codec) | webrtc::voe::Channel | |
| SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency) | webrtc::voe::Channel | |
| SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency) | webrtc::voe::Channel | |
| SetSink(std::unique_ptr< AudioSinkInterface > sink) | webrtc::voe::Channel | |
| SetSink(std::unique_ptr< AudioSinkInterface > sink) | webrtc::voe::Channel | |
| SetTransportOverhead(size_t transport_overhead_per_packet) | webrtc::voe::Channel | |
| SetTransportOverhead(size_t transport_overhead_per_packet) | webrtc::voe::Channel | |
| SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX) | webrtc::voe::Channel | |
| SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX) | webrtc::voe::Channel | |
| StartPlayingFileAsMicrophone(const char *fileName, bool loop, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileAsMicrophone(InStream *stream, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileAsMicrophone(const char *fileName, bool loop, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileAsMicrophone(InStream *stream, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileLocally(const char *fileName, bool loop, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileLocally(InStream *stream, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileLocally(const char *fileName, bool loop, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayingFileLocally(InStream *stream, FileFormats format, int startPosition, float volumeScaling, int stopPosition, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartPlayout() | webrtc::voe::Channel | |
| StartPlayout() | webrtc::voe::Channel | |
| StartRecordingPlayout(const char *fileName, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartRecordingPlayout(OutStream *stream, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartRecordingPlayout(const char *fileName, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartRecordingPlayout(OutStream *stream, const CodecInst *codecInst) | webrtc::voe::Channel | |
| StartSend() | webrtc::voe::Channel | |
| StartSend() | webrtc::voe::Channel | |
| StopPlayingFileAsMicrophone() | webrtc::voe::Channel | |
| StopPlayingFileAsMicrophone() | webrtc::voe::Channel | |
| StopPlayingFileLocally() | webrtc::voe::Channel | |
| StopPlayingFileLocally() | webrtc::voe::Channel | |
| StopPlayout() | webrtc::voe::Channel | |
| StopPlayout() | webrtc::voe::Channel | |
| StopRecordingPlayout() | webrtc::voe::Channel | |
| StopRecordingPlayout() | webrtc::voe::Channel | |
| StopSend() | webrtc::voe::Channel | |
| StopSend() | webrtc::voe::Channel | |
| UpdateLocalTimeStamp() | webrtc::voe::Channel | |
| UpdateLocalTimeStamp() | webrtc::voe::Channel | |
| VoERtcpObserver class | webrtc::voe::Channel | friend |
| VoiceActivityIndicator(int &activity) | webrtc::voe::Channel | |
| VoiceActivityIndicator(int &activity) | webrtc::voe::Channel | |
| ~ACMVADCallback() | webrtc::ACMVADCallback | inlinevirtual |
| ~ACMVADCallback() | webrtc::ACMVADCallback | inlinevirtual |
| ~AudioPacketizationCallback() | webrtc::AudioPacketizationCallback | inlinevirtual |
| ~AudioPacketizationCallback() | webrtc::AudioPacketizationCallback | inlinevirtual |
| ~Channel() | webrtc::voe::Channel | virtual |
| ~Channel() | webrtc::voe::Channel | virtual |
| ~FileCallback() | webrtc::FileCallback | inlinevirtual |
| ~FileCallback() | webrtc::FileCallback | inlinevirtual |
| ~MixerParticipant() | webrtc::MixerParticipant | protectedvirtual |
| ~MixerParticipant() | webrtc::MixerParticipant | protectedvirtual |
| ~OverheadObserver()=default | webrtc::OverheadObserver | virtual |
| ~OverheadObserver()=default | webrtc::OverheadObserver | virtual |
| ~RtpData() | webrtc::RtpData | inlinevirtual |
| ~RtpData() | webrtc::RtpData | inlinevirtual |
| ~RtpFeedback() | webrtc::RtpFeedback | inlinevirtual |
| ~RtpFeedback() | webrtc::RtpFeedback | inlinevirtual |
| ~Transport() | webrtc::Transport | inlineprotectedvirtual |
| ~Transport() | webrtc::Transport | inlineprotectedvirtual |
