#include <channel.h>

Inheritance diagram for webrtc::voe::Channel:

Public Types
enum	{ KNumSocketThreads = 1 }

enum	{ KNumberOfSocketBuffers = 8 }

enum	{ KNumSocketThreads = 1 }

enum	{ KNumberOfSocketBuffers = 8 }

Public Types inherited from webrtc::MixerParticipant
enum	AudioFrameInfo { AudioFrameInfo::kNormal, AudioFrameInfo::kMuted, AudioFrameInfo::kError, AudioFrameInfo::kNormal, AudioFrameInfo::kMuted, AudioFrameInfo::kError }

enum	AudioFrameInfo { AudioFrameInfo::kNormal, AudioFrameInfo::kMuted, AudioFrameInfo::kError, AudioFrameInfo::kNormal, AudioFrameInfo::kMuted, AudioFrameInfo::kError }

Public Member Functions
virtual	~Channel ()

	Channel (int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config)

int32_t	Init ()

int32_t	SetEngineInformation (Statistics &engineStatistics, OutputMixer &outputMixer, ProcessThread &moduleProcessThread, AudioDeviceModule &audioDeviceModule, VoiceEngineObserver voiceEngineObserver, rtc::CriticalSection callbackCritSect)

int32_t	UpdateLocalTimeStamp ()

void	SetSink (std::unique_ptr< AudioSinkInterface > sink)

const rtc::scoped_refptr< AudioDecoderFactory > &	GetAudioDecoderFactory () const

int32_t	StartPlayout ()

int32_t	StopPlayout ()

int32_t	StartSend ()

int32_t	StopSend ()

int32_t	RegisterVoiceEngineObserver (VoiceEngineObserver &observer)

int32_t	DeRegisterVoiceEngineObserver ()

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)

int32_t	RegisterExternalTransport (Transport *transport)

int32_t	DeRegisterExternalTransport ()

int32_t	ReceivedRTPPacket (const uint8_t *received_packet, size_t length, const PacketTime &packet_time)

int32_t	ReceivedRTCPPacket (const uint8_t *data, size_t length)

void	OnRtpPacket (const RtpPacketReceived &packet)

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)

int	GetSpeechOutputLevel (uint32_t &level) const

int	GetSpeechOutputLevelFullRange (uint32_t &level) const

int	SetInputMute (bool enable)

bool	InputMute () const

int	SetOutputVolumePan (float left, float right)

int	GetOutputVolumePan (float &left, float &right) const

int	SetChannelOutputVolumeScaling (float scaling)

int	GetChannelOutputVolumeScaling (float &scaling) const

int	GetNetworkStatistics (NetworkStatistics &stats)

void	GetDecodingCallStatistics (AudioDecodingCallStats *stats) const

uint32_t	GetDelayEstimate () const

int	SetMinimumPlayoutDelay (int delayMs)

int	GetPlayoutTimestamp (unsigned int &timestamp)

int	GetRtpRtcp (RtpRtcp rtpRtcpModule, RtpReceiver rtp_receiver) const

int	SendTelephoneEventOutband (int event, int duration_ms)

int	SetSendTelephoneEventPayloadType (int payload_type, int payload_frequency)

int	VoiceActivityIndicator (int &activity)

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 (RtpPacketSender rtp_packet_sender, TransportFeedbackObserver transport_feedback_observer, PacketRouter packet_router, RtcpBandwidthObserver bandwidth_observer)

void	RegisterReceiverCongestionControlObjects (PacketRouter *packet_router)

void	ResetCongestionControlObjects ()

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)

int32_t	SendData (FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t payloadData, size_t payloadSize, const RTPFragmentationHeader fragmentation) override

int32_t	InFrameType (FrameType frame_type) override

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

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

bool	SendRtp (const uint8_t *data, size_t len, const PacketOptions &packet_options) override

bool	SendRtcp (const uint8_t *data, size_t len) override

MixerParticipant::AudioFrameInfo	GetAudioFrameWithMuted (int32_t id, AudioFrame *audioFrame) override

int32_t	NeededFrequency (int32_t id) const override

AudioMixer::Source::AudioFrameInfo	GetAudioFrameWithInfo (int sample_rate_hz, AudioFrame *audio_frame)

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

int32_t	ChannelId () const

bool	Playing () const

bool	Sending () const

bool	ExternalTransport () const

RtpRtcp *	RtpRtcpModulePtr () const

int8_t	OutputEnergyLevel () const

uint32_t	Demultiplex (const AudioFrame &audioFrame)

void	Demultiplex (const int16_t *audio_data, int sample_rate, size_t number_of_frames, size_t number_of_channels)

uint32_t	PrepareEncodeAndSend (int mixingFrequency)

uint32_t	EncodeAndSend ()

void	set_associate_send_channel (const ChannelOwner &channel)

void	DisassociateSendChannel (int channel_id)

void	SetRtcEventLog (RtcEventLog *event_log)

void	SetRtcpRttStats (RtcpRttStats *rtcp_rtt_stats)

void	SetTransportOverhead (size_t transport_overhead_per_packet)

void	OnOverheadChanged (size_t overhead_bytes_per_packet) override

virtual	~Channel ()

	Channel (int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config)

int32_t	Init ()

int32_t	SetEngineInformation (Statistics &engineStatistics, OutputMixer &outputMixer, ProcessThread &moduleProcessThread, AudioDeviceModule &audioDeviceModule, VoiceEngineObserver voiceEngineObserver, rtc::CriticalSection callbackCritSect)

int32_t	UpdateLocalTimeStamp ()

void	SetSink (std::unique_ptr< AudioSinkInterface > sink)

const rtc::scoped_refptr< AudioDecoderFactory > &	GetAudioDecoderFactory () const

int32_t	StartPlayout ()

int32_t	StopPlayout ()

int32_t	StartSend ()

int32_t	StopSend ()

int32_t	RegisterVoiceEngineObserver (VoiceEngineObserver &observer)

int32_t	DeRegisterVoiceEngineObserver ()

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)

int32_t	RegisterExternalTransport (Transport *transport)

int32_t	DeRegisterExternalTransport ()

int32_t	ReceivedRTPPacket (const uint8_t *received_packet, size_t length, const PacketTime &packet_time)

int32_t	ReceivedRTCPPacket (const uint8_t *data, size_t length)

void	OnRtpPacket (const RtpPacketReceived &packet)

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)

int	GetSpeechOutputLevel (uint32_t &level) const

int	GetSpeechOutputLevelFullRange (uint32_t &level) const

int	SetInputMute (bool enable)

bool	InputMute () const

int	SetOutputVolumePan (float left, float right)

int	GetOutputVolumePan (float &left, float &right) const

int	SetChannelOutputVolumeScaling (float scaling)

int	GetChannelOutputVolumeScaling (float &scaling) const

int	GetNetworkStatistics (NetworkStatistics &stats)

void	GetDecodingCallStatistics (AudioDecodingCallStats *stats) const

uint32_t	GetDelayEstimate () const

int	SetMinimumPlayoutDelay (int delayMs)

int	GetPlayoutTimestamp (unsigned int &timestamp)

int	GetRtpRtcp (RtpRtcp rtpRtcpModule, RtpReceiver rtp_receiver) const

int	SendTelephoneEventOutband (int event, int duration_ms)

int	SetSendTelephoneEventPayloadType (int payload_type, int payload_frequency)

int	VoiceActivityIndicator (int &activity)

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 (RtpPacketSender rtp_packet_sender, TransportFeedbackObserver transport_feedback_observer, PacketRouter packet_router, RtcpBandwidthObserver bandwidth_observer)

void	RegisterReceiverCongestionControlObjects (PacketRouter *packet_router)

void	ResetCongestionControlObjects ()

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)

int32_t	SendData (FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t payloadData, size_t payloadSize, const RTPFragmentationHeader fragmentation) override

int32_t	InFrameType (FrameType frame_type) override

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

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

bool	SendRtp (const uint8_t *data, size_t len, const PacketOptions &packet_options) override

bool	SendRtcp (const uint8_t *data, size_t len) override

MixerParticipant::AudioFrameInfo	GetAudioFrameWithMuted (int32_t id, AudioFrame *audioFrame) override

int32_t	NeededFrequency (int32_t id) const override

AudioMixer::Source::AudioFrameInfo	GetAudioFrameWithInfo (int sample_rate_hz, AudioFrame *audio_frame)

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

int32_t	ChannelId () const

bool	Playing () const

bool	Sending () const

bool	ExternalTransport () const

RtpRtcp *	RtpRtcpModulePtr () const

int8_t	OutputEnergyLevel () const

uint32_t	Demultiplex (const AudioFrame &audioFrame)

void	Demultiplex (const int16_t *audio_data, int sample_rate, size_t number_of_frames, size_t number_of_channels)

uint32_t	PrepareEncodeAndSend (int mixingFrequency)

uint32_t	EncodeAndSend ()

void	set_associate_send_channel (const ChannelOwner &channel)

void	DisassociateSendChannel (int channel_id)

void	SetRtcEventLog (RtcEventLog *event_log)

void	SetRtcpRttStats (RtcpRttStats *rtcp_rtt_stats)

void	SetTransportOverhead (size_t transport_overhead_per_packet)

void	OnOverheadChanged (size_t overhead_bytes_per_packet) override

Public Member Functions inherited from webrtc::RtpData
virtual	~RtpData ()

virtual	~RtpData ()

Public Member Functions inherited from webrtc::RtpFeedback
virtual	~RtpFeedback ()

virtual	~RtpFeedback ()

Public Member Functions inherited from webrtc::FileCallback
virtual	~FileCallback ()

virtual	~FileCallback ()

Public Member Functions inherited from webrtc::AudioPacketizationCallback
virtual	~AudioPacketizationCallback ()

virtual	~AudioPacketizationCallback ()

Public Member Functions inherited from webrtc::ACMVADCallback
virtual	~ACMVADCallback ()

virtual	~ACMVADCallback ()

Public Member Functions inherited from webrtc::MixerParticipant
virtual int32_t	GetAudioFrame (int32_t id, AudioFrame *audioFrame)

bool	IsMixed () const

virtual int32_t	GetAudioFrame (int32_t id, AudioFrame *audioFrame)

bool	IsMixed () const

Public Member Functions inherited from webrtc::OverheadObserver
virtual	~OverheadObserver ()=default

virtual	~OverheadObserver ()=default

Static Public Member Functions
static int32_t	CreateChannel (Channel *&channel, int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config)

static int32_t	CreateChannel (Channel *&channel, int32_t channelId, uint32_t instanceId, const VoEBase::ChannelConfig &config)

Protected Member Functions
void	OnIncomingFractionLoss (int fraction_lost)

void	OnIncomingFractionLoss (int fraction_lost)

Protected Member Functions inherited from webrtc::FileCallback
	FileCallback ()

	FileCallback ()

Protected Member Functions inherited from webrtc::Transport
virtual	~Transport ()

virtual	~Transport ()

Protected Member Functions inherited from webrtc::MixerParticipant
	MixerParticipant ()

virtual	~MixerParticipant ()

	MixerParticipant ()

virtual	~MixerParticipant ()

Friends
class	VoERtcpObserver

Additional Inherited Members
Public Attributes inherited from webrtc::MixerParticipant
MixHistory *	_mixHistory

Member Enumeration Documentation

◆ anonymous enum

anonymous enum

Enumerator
KNumSocketThreads

◆ anonymous enum

anonymous enum

Enumerator
KNumberOfSocketBuffers

◆ anonymous enum

anonymous enum

Enumerator
KNumSocketThreads

◆ anonymous enum

anonymous enum

Enumerator
KNumberOfSocketBuffers

Constructor & Destructor Documentation

◆ ~Channel() [1/2]

webrtc::voe::Channel::~Channel ( )

virtual

◆ Channel() [1/2]

webrtc::voe::Channel::Channel	(	int32_t	channelId,
		uint32_t	instanceId,
		const VoEBase::ChannelConfig &	config
	)

◆ ~Channel() [2/2]

virtual webrtc::voe::Channel::~Channel ( )

virtual

◆ Channel() [2/2]

webrtc::voe::Channel::Channel	(	int32_t	channelId,
		uint32_t	instanceId,
		const VoEBase::ChannelConfig &	config
	)

Member Function Documentation

◆ ChannelId() [1/2]

int32_t webrtc::voe::Channel::ChannelId ( ) const

inline

◆ ChannelId() [2/2]

int32_t webrtc::voe::Channel::ChannelId ( ) const

inline

◆ CreateChannel() [1/2]

int32_t webrtc::voe::Channel::CreateChannel	(	Channel *&	channel,
		int32_t	channelId,
		uint32_t	instanceId,
		const VoEBase::ChannelConfig &	config
	)

static

◆ CreateChannel() [2/2]

static int32_t webrtc::voe::Channel::CreateChannel	(	Channel *&	channel,
		int32_t	channelId,
		uint32_t	instanceId,
		const VoEBase::ChannelConfig &	config
	)

static

◆ Demultiplex() [1/4]

uint32_t webrtc::voe::Channel::Demultiplex ( const AudioFrame & audioFrame )

◆ Demultiplex() [2/4]

uint32_t webrtc::voe::Channel::Demultiplex ( const AudioFrame & audioFrame )

◆ Demultiplex() [3/4]

void webrtc::voe::Channel::Demultiplex	(	const int16_t *	audio_data,
		int	sample_rate,
		size_t	number_of_frames,
		size_t	number_of_channels
	)

◆ Demultiplex() [4/4]

void webrtc::voe::Channel::Demultiplex	(	const int16_t *	audio_data,
		int	sample_rate,
		size_t	number_of_frames,
		size_t	number_of_channels
	)

◆ DeRegisterExternalTransport() [1/2]

int32_t webrtc::voe::Channel::DeRegisterExternalTransport ( )

◆ DeRegisterExternalTransport() [2/2]

int32_t webrtc::voe::Channel::DeRegisterExternalTransport ( )

◆ DeRegisterVoiceEngineObserver() [1/2]

int32_t webrtc::voe::Channel::DeRegisterVoiceEngineObserver ( )

◆ DeRegisterVoiceEngineObserver() [2/2]

int32_t webrtc::voe::Channel::DeRegisterVoiceEngineObserver ( )

◆ DisableAudioNetworkAdaptor() [1/2]

void webrtc::voe::Channel::DisableAudioNetworkAdaptor ( )

◆ DisableAudioNetworkAdaptor() [2/2]

void webrtc::voe::Channel::DisableAudioNetworkAdaptor ( )

◆ DisassociateSendChannel() [1/2]

void webrtc::voe::Channel::DisassociateSendChannel ( int channel_id )

◆ DisassociateSendChannel() [2/2]

void webrtc::voe::Channel::DisassociateSendChannel ( int channel_id )

◆ EnableAudioNetworkAdaptor() [1/2]

bool webrtc::voe::Channel::EnableAudioNetworkAdaptor ( const std::string & config_string )

◆ EnableAudioNetworkAdaptor() [2/2]

bool webrtc::voe::Channel::EnableAudioNetworkAdaptor ( const std::string & config_string )

◆ EnableReceiveTransportSequenceNumber() [1/2]

void webrtc::voe::Channel::EnableReceiveTransportSequenceNumber ( int id )

◆ EnableReceiveTransportSequenceNumber() [2/2]

void webrtc::voe::Channel::EnableReceiveTransportSequenceNumber ( int id )

◆ EnableSendTransportSequenceNumber() [1/2]

void webrtc::voe::Channel::EnableSendTransportSequenceNumber ( int id )

◆ EnableSendTransportSequenceNumber() [2/2]

void webrtc::voe::Channel::EnableSendTransportSequenceNumber ( int id )

◆ EncodeAndSend() [1/2]

uint32_t webrtc::voe::Channel::EncodeAndSend ( )

◆ EncodeAndSend() [2/2]

uint32_t webrtc::voe::Channel::EncodeAndSend ( )

◆ ExternalTransport() [1/2]

bool webrtc::voe::Channel::ExternalTransport ( ) const

inline

◆ ExternalTransport() [2/2]

bool webrtc::voe::Channel::ExternalTransport ( ) const

inline

◆ GetAudioDecoderFactory() [1/2]

const rtc::scoped_refptr< AudioDecoderFactory > & webrtc::voe::Channel::GetAudioDecoderFactory ( ) const

◆ GetAudioDecoderFactory() [2/2]

const rtc::scoped_refptr<AudioDecoderFactory>& webrtc::voe::Channel::GetAudioDecoderFactory ( ) const

◆ GetAudioFrameWithInfo() [1/2]

AudioMixer::Source::AudioFrameInfo webrtc::voe::Channel::GetAudioFrameWithInfo	(	int	sample_rate_hz,
		AudioFrame *	audio_frame
	)

◆ GetAudioFrameWithInfo() [2/2]

AudioMixer::Source::AudioFrameInfo webrtc::voe::Channel::GetAudioFrameWithInfo	(	int	sample_rate_hz,
		AudioFrame *	audio_frame
	)

◆ GetAudioFrameWithMuted() [1/2]

MixerParticipant::AudioFrameInfo webrtc::voe::Channel::GetAudioFrameWithMuted	(	int32_t	id,
		AudioFrame *	audioFrame
	)

overridevirtual

Reimplemented from webrtc::MixerParticipant.

◆ GetAudioFrameWithMuted() [2/2]

MixerParticipant::AudioFrameInfo webrtc::voe::Channel::GetAudioFrameWithMuted	(	int32_t	id,
		AudioFrame *	audioFrame
	)

overridevirtual

Reimplemented from webrtc::MixerParticipant.

◆ GetChannelOutputVolumeScaling() [1/2]

int webrtc::voe::Channel::GetChannelOutputVolumeScaling ( float & scaling ) const

◆ GetChannelOutputVolumeScaling() [2/2]

int webrtc::voe::Channel::GetChannelOutputVolumeScaling ( float & scaling ) const

◆ GetCodecFECStatus() [1/2]

bool webrtc::voe::Channel::GetCodecFECStatus ( )

◆ GetCodecFECStatus() [2/2]

bool webrtc::voe::Channel::GetCodecFECStatus ( )

◆ GetDecodingCallStatistics() [1/2]

void webrtc::voe::Channel::GetDecodingCallStatistics ( AudioDecodingCallStats * stats ) const

◆ GetDecodingCallStatistics() [2/2]

void webrtc::voe::Channel::GetDecodingCallStatistics ( AudioDecodingCallStats * stats ) const

◆ GetDelayEstimate() [1/2]

uint32_t webrtc::voe::Channel::GetDelayEstimate ( ) const

◆ GetDelayEstimate() [2/2]

uint32_t webrtc::voe::Channel::GetDelayEstimate ( ) const

◆ GetLocalSSRC() [1/2]

int webrtc::voe::Channel::GetLocalSSRC ( unsigned int & ssrc )

◆ GetLocalSSRC() [2/2]

int webrtc::voe::Channel::GetLocalSSRC ( unsigned int & ssrc )

◆ GetNetworkStatistics() [1/2]

int webrtc::voe::Channel::GetNetworkStatistics ( NetworkStatistics & stats )

◆ GetNetworkStatistics() [2/2]

int webrtc::voe::Channel::GetNetworkStatistics ( NetworkStatistics & stats )

◆ GetOpusDtx() [1/2]

int webrtc::voe::Channel::GetOpusDtx ( bool * enabled )

◆ GetOpusDtx() [2/2]

int webrtc::voe::Channel::GetOpusDtx ( bool * enabled )

◆ GetOutputVolumePan() [1/2]

int webrtc::voe::Channel::GetOutputVolumePan	(	float &	left,
		float &	right
	)		const

◆ GetOutputVolumePan() [2/2]

int webrtc::voe::Channel::GetOutputVolumePan	(	float &	left,
		float &	right
	)		const

◆ GetPlayoutTimestamp() [1/2]

int webrtc::voe::Channel::GetPlayoutTimestamp ( unsigned int & timestamp )

◆ GetPlayoutTimestamp() [2/2]

int webrtc::voe::Channel::GetPlayoutTimestamp ( unsigned int & timestamp )

◆ GetRecCodec() [1/2]

int32_t webrtc::voe::Channel::GetRecCodec ( CodecInst & codec )

◆ GetRecCodec() [2/2]

int32_t webrtc::voe::Channel::GetRecCodec ( CodecInst & codec )

◆ GetRecPayloadType() [1/2]

int32_t webrtc::voe::Channel::GetRecPayloadType ( CodecInst & codec )

◆ GetRecPayloadType() [2/2]

int32_t webrtc::voe::Channel::GetRecPayloadType ( CodecInst & codec )

◆ GetRemoteRTCP_CNAME() [1/2]

int webrtc::voe::Channel::GetRemoteRTCP_CNAME ( char cName[256] )

◆ GetRemoteRTCP_CNAME() [2/2]

int webrtc::voe::Channel::GetRemoteRTCP_CNAME ( char cName[256] )

◆ GetRemoteRTCPReportBlocks() [1/2]

int webrtc::voe::Channel::GetRemoteRTCPReportBlocks ( std::vector< ReportBlock > * report_blocks )

◆ GetRemoteRTCPReportBlocks() [2/2]

int webrtc::voe::Channel::GetRemoteRTCPReportBlocks ( std::vector< ReportBlock > * report_blocks )

◆ GetRemoteSSRC() [1/2]

int webrtc::voe::Channel::GetRemoteSSRC ( unsigned int & ssrc )

◆ GetRemoteSSRC() [2/2]

int webrtc::voe::Channel::GetRemoteSSRC ( unsigned int & ssrc )

◆ GetRTCPStatus() [1/2]

int webrtc::voe::Channel::GetRTCPStatus ( bool & enabled )

◆ GetRTCPStatus() [2/2]

int webrtc::voe::Channel::GetRTCPStatus ( bool & enabled )

◆ GetRtpRtcp() [1/2]

int webrtc::voe::Channel::GetRtpRtcp	(	RtpRtcp **	rtpRtcpModule,
		RtpReceiver **	rtp_receiver
	)		const

◆ GetRtpRtcp() [2/2]

int webrtc::voe::Channel::GetRtpRtcp	(	RtpRtcp **	rtpRtcpModule,
		RtpReceiver **	rtp_receiver
	)		const

◆ GetRTPStatistics() [1/2]

int webrtc::voe::Channel::GetRTPStatistics ( CallStatistics & stats )

◆ GetRTPStatistics() [2/2]

int webrtc::voe::Channel::GetRTPStatistics ( CallStatistics & stats )

◆ GetSendCodec() [1/2]

int32_t webrtc::voe::Channel::GetSendCodec ( CodecInst & codec )

◆ GetSendCodec() [2/2]

int32_t webrtc::voe::Channel::GetSendCodec ( CodecInst & codec )

◆ GetSpeechOutputLevel() [1/2]

int webrtc::voe::Channel::GetSpeechOutputLevel ( uint32_t & level ) const

◆ GetSpeechOutputLevel() [2/2]

int webrtc::voe::Channel::GetSpeechOutputLevel ( uint32_t & level ) const

◆ GetSpeechOutputLevelFullRange() [1/2]

int webrtc::voe::Channel::GetSpeechOutputLevelFullRange ( uint32_t & level ) const

◆ GetSpeechOutputLevelFullRange() [2/2]

int webrtc::voe::Channel::GetSpeechOutputLevelFullRange ( uint32_t & level ) const

◆ GetVADStatus() [1/2]

int32_t webrtc::voe::Channel::GetVADStatus	(	bool &	enabledVAD,
		ACMVADMode &	mode,
		bool &	disabledDTX
	)

◆ GetVADStatus() [2/2]

int32_t webrtc::voe::Channel::GetVADStatus	(	bool &	enabledVAD,
		ACMVADMode &	mode,
		bool &	disabledDTX
	)

◆ InFrameType() [1/2]

int32_t webrtc::voe::Channel::InFrameType ( FrameType frame_type )

overridevirtual

Implements webrtc::ACMVADCallback.

◆ InFrameType() [2/2]

int32_t webrtc::voe::Channel::InFrameType ( FrameType frame_type )

overridevirtual

Implements webrtc::ACMVADCallback.

◆ Init() [1/2]

int32_t webrtc::voe::Channel::Init ( )

◆ Init() [2/2]

int32_t webrtc::voe::Channel::Init ( )

◆ InputMute() [1/2]

bool webrtc::voe::Channel::InputMute ( ) const

◆ InputMute() [2/2]

bool webrtc::voe::Channel::InputMute ( ) const

◆ InstanceId() [1/2]

uint32_t webrtc::voe::Channel::InstanceId ( ) const

inline

◆ InstanceId() [2/2]

uint32_t webrtc::voe::Channel::InstanceId ( ) const

inline

◆ IsPlayingFileAsMicrophone() [1/2]

int webrtc::voe::Channel::IsPlayingFileAsMicrophone ( ) const

◆ IsPlayingFileAsMicrophone() [2/2]

int webrtc::voe::Channel::IsPlayingFileAsMicrophone ( ) const

◆ IsPlayingFileLocally() [1/2]

int webrtc::voe::Channel::IsPlayingFileLocally ( ) const

◆ IsPlayingFileLocally() [2/2]

int webrtc::voe::Channel::IsPlayingFileLocally ( ) const

◆ NeededFrequency() [1/2]

int32_t webrtc::voe::Channel::NeededFrequency ( int32_t id ) const

overridevirtual

Implements webrtc::MixerParticipant.

◆ NeededFrequency() [2/2]

int32_t webrtc::voe::Channel::NeededFrequency ( int32_t id ) const

overridevirtual

Implements webrtc::MixerParticipant.

◆ OnIncomingCSRCChanged() [1/2]

void webrtc::voe::Channel::OnIncomingCSRCChanged	(	uint32_t	CSRC,
		bool	added
	)

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnIncomingCSRCChanged() [2/2]

void webrtc::voe::Channel::OnIncomingCSRCChanged	(	uint32_t	CSRC,
		bool	added
	)

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnIncomingFractionLoss() [1/2]

void webrtc::voe::Channel::OnIncomingFractionLoss ( int fraction_lost )

protected

◆ OnIncomingFractionLoss() [2/2]

void webrtc::voe::Channel::OnIncomingFractionLoss ( int fraction_lost )

protected

◆ OnIncomingSSRCChanged() [1/2]

void webrtc::voe::Channel::OnIncomingSSRCChanged ( uint32_t ssrc )

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnIncomingSSRCChanged() [2/2]

void webrtc::voe::Channel::OnIncomingSSRCChanged ( uint32_t ssrc )

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnInitializeDecoder() [1/2]

int32_t webrtc::voe::Channel::OnInitializeDecoder	(	int8_t	payloadType,
		const char	payloadName[RTP_PAYLOAD_NAME_SIZE],
		int	frequency,
		size_t	channels,
		uint32_t	rate
	)

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnInitializeDecoder() [2/2]

int32_t webrtc::voe::Channel::OnInitializeDecoder	(	int8_t	payloadType,
		const char	payloadName[RTP_PAYLOAD_NAME_SIZE],
		int	frequency,
		size_t	channels,
		uint32_t	rate
	)

overridevirtual

Implements webrtc::RtpFeedback.

◆ OnOverheadChanged() [1/2]

void webrtc::voe::Channel::OnOverheadChanged ( size_t overhead_bytes_per_packet )

overridevirtual

Implements webrtc::OverheadObserver.

◆ OnOverheadChanged() [2/2]

void webrtc::voe::Channel::OnOverheadChanged ( size_t overhead_bytes_per_packet )

overridevirtual

Implements webrtc::OverheadObserver.

◆ OnReceivedPayloadData() [1/2]

int32_t webrtc::voe::Channel::OnReceivedPayloadData	(	const uint8_t *	payloadData,
		size_t	payloadSize,
		const WebRtcRTPHeader *	rtpHeader
	)

overridevirtual

Implements webrtc::RtpData.

◆ OnReceivedPayloadData() [2/2]

int32_t webrtc::voe::Channel::OnReceivedPayloadData	(	const uint8_t *	payloadData,
		size_t	payloadSize,
		const WebRtcRTPHeader *	rtpHeader
	)

overridevirtual

Implements webrtc::RtpData.

◆ OnRecoveredPacket() [1/2]

bool webrtc::voe::Channel::OnRecoveredPacket	(	const uint8_t *	packet,
		size_t	packet_length
	)

overridevirtual

Implements webrtc::RtpData.

◆ OnRecoveredPacket() [2/2]

bool webrtc::voe::Channel::OnRecoveredPacket	(	const uint8_t *	packet,
		size_t	packet_length
	)

overridevirtual

Implements webrtc::RtpData.

◆ OnRtpPacket() [1/2]

void webrtc::voe::Channel::OnRtpPacket ( const RtpPacketReceived & packet )

◆ OnRtpPacket() [2/2]

void webrtc::voe::Channel::OnRtpPacket ( const RtpPacketReceived & packet )

◆ OutputEnergyLevel() [1/2]

int8_t webrtc::voe::Channel::OutputEnergyLevel ( ) const

inline

◆ OutputEnergyLevel() [2/2]

int8_t webrtc::voe::Channel::OutputEnergyLevel ( ) const

inline

◆ PlayFileEnded() [1/2]

void webrtc::voe::Channel::PlayFileEnded ( int32_t id )

overridevirtual

Implements webrtc::FileCallback.

◆ PlayFileEnded() [2/2]

void webrtc::voe::Channel::PlayFileEnded ( int32_t id )

overridevirtual

Implements webrtc::FileCallback.

◆ Playing() [1/2]

bool webrtc::voe::Channel::Playing ( ) const

inline

◆ Playing() [2/2]

bool webrtc::voe::Channel::Playing ( ) const

inline

◆ PlayNotification() [1/2]

void webrtc::voe::Channel::PlayNotification	(	int32_t	id,
		uint32_t	durationMs
	)

overridevirtual

Implements webrtc::FileCallback.

◆ PlayNotification() [2/2]

void webrtc::voe::Channel::PlayNotification	(	int32_t	id,
		uint32_t	durationMs
	)

overridevirtual

Implements webrtc::FileCallback.

◆ PrepareEncodeAndSend() [1/2]

uint32_t webrtc::voe::Channel::PrepareEncodeAndSend ( int mixingFrequency )

◆ PrepareEncodeAndSend() [2/2]

uint32_t webrtc::voe::Channel::PrepareEncodeAndSend ( int mixingFrequency )

◆ ReceivedRTCPPacket() [1/2]

int32_t webrtc::voe::Channel::ReceivedRTCPPacket	(	const uint8_t *	data,
		size_t	length
	)

◆ ReceivedRTCPPacket() [2/2]

int32_t webrtc::voe::Channel::ReceivedRTCPPacket	(	const uint8_t *	data,
		size_t	length
	)

◆ ReceivedRTPPacket() [1/2]

int32_t webrtc::voe::Channel::ReceivedRTPPacket	(	const uint8_t *	received_packet,
		size_t	length,
		const PacketTime &	packet_time
	)

◆ ReceivedRTPPacket() [2/2]

int32_t webrtc::voe::Channel::ReceivedRTPPacket	(	const uint8_t *	received_packet,
		size_t	length,
		const PacketTime &	packet_time
	)

◆ RecordFileEnded() [1/2]

void webrtc::voe::Channel::RecordFileEnded ( int32_t id )

overridevirtual

Implements webrtc::FileCallback.

◆ RecordFileEnded() [2/2]

void webrtc::voe::Channel::RecordFileEnded ( int32_t id )

overridevirtual

Implements webrtc::FileCallback.

◆ RecordNotification() [1/2]

void webrtc::voe::Channel::RecordNotification	(	int32_t	id,
		uint32_t	durationMs
	)

overridevirtual

Implements webrtc::FileCallback.

◆ RecordNotification() [2/2]

void webrtc::voe::Channel::RecordNotification	(	int32_t	id,
		uint32_t	durationMs
	)

overridevirtual

Implements webrtc::FileCallback.

◆ RegisterExternalTransport() [1/2]

int32_t webrtc::voe::Channel::RegisterExternalTransport ( Transport * transport )

◆ RegisterExternalTransport() [2/2]

int32_t webrtc::voe::Channel::RegisterExternalTransport ( Transport * transport )

◆ RegisterFilePlayingToMixer() [1/2]

int webrtc::voe::Channel::RegisterFilePlayingToMixer ( )

◆ RegisterFilePlayingToMixer() [2/2]

int webrtc::voe::Channel::RegisterFilePlayingToMixer ( )

◆ RegisterReceiverCongestionControlObjects() [1/2]

void webrtc::voe::Channel::RegisterReceiverCongestionControlObjects ( PacketRouter * packet_router )

◆ RegisterReceiverCongestionControlObjects() [2/2]

void webrtc::voe::Channel::RegisterReceiverCongestionControlObjects ( PacketRouter * packet_router )

◆ RegisterSenderCongestionControlObjects() [1/2]

void webrtc::voe::Channel::RegisterSenderCongestionControlObjects	(	RtpPacketSender *	rtp_packet_sender,
		TransportFeedbackObserver *	transport_feedback_observer,
		PacketRouter *	packet_router,
		RtcpBandwidthObserver *	bandwidth_observer
	)

◆ RegisterSenderCongestionControlObjects() [2/2]

void webrtc::voe::Channel::RegisterSenderCongestionControlObjects	(	RtpPacketSender *	rtp_packet_sender,
		TransportFeedbackObserver *	transport_feedback_observer,
		PacketRouter *	packet_router,
		RtcpBandwidthObserver *	bandwidth_observer
	)

◆ RegisterVoiceEngineObserver() [1/2]

int32_t webrtc::voe::Channel::RegisterVoiceEngineObserver ( VoiceEngineObserver & observer )

◆ RegisterVoiceEngineObserver() [2/2]

int32_t webrtc::voe::Channel::RegisterVoiceEngineObserver ( VoiceEngineObserver & observer )

◆ ResetCongestionControlObjects() [1/2]

void webrtc::voe::Channel::ResetCongestionControlObjects ( )

◆ ResetCongestionControlObjects() [2/2]

void webrtc::voe::Channel::ResetCongestionControlObjects ( )

◆ RtpRtcpModulePtr() [1/2]

RtpRtcp* webrtc::voe::Channel::RtpRtcpModulePtr ( ) const

inline

◆ RtpRtcpModulePtr() [2/2]

RtpRtcp* webrtc::voe::Channel::RtpRtcpModulePtr ( ) const

inline

◆ SendApplicationDefinedRTCPPacket() [1/2]

int webrtc::voe::Channel::SendApplicationDefinedRTCPPacket	(	unsigned char	subType,
		unsigned int	name,
		const char *	data,
		unsigned short	dataLengthInBytes
	)

◆ SendApplicationDefinedRTCPPacket() [2/2]

int webrtc::voe::Channel::SendApplicationDefinedRTCPPacket	(	unsigned char	subType,
		unsigned int	name,
		const char *	data,
		unsigned short	dataLengthInBytes
	)

◆ SendData() [1/2]

int32_t webrtc::voe::Channel::SendData	(	FrameType	frameType,
		uint8_t	payloadType,
		uint32_t	timeStamp,
		const uint8_t *	payloadData,
		size_t	payloadSize,
		const RTPFragmentationHeader *	fragmentation
	)

overridevirtual

Implements webrtc::AudioPacketizationCallback.

◆ SendData() [2/2]

int32_t webrtc::voe::Channel::SendData	(	FrameType	frameType,
		uint8_t	payloadType,
		uint32_t	timeStamp,
		const uint8_t *	payloadData,
		size_t	payloadSize,
		const RTPFragmentationHeader *	fragmentation
	)

overridevirtual

Implements webrtc::AudioPacketizationCallback.

◆ Sending() [1/2]

bool webrtc::voe::Channel::Sending ( ) const

inline

◆ Sending() [2/2]

bool webrtc::voe::Channel::Sending ( ) const

inline

◆ SendRtcp() [1/2]

bool webrtc::voe::Channel::SendRtcp	(	const uint8_t *	data,
		size_t	len
	)

overridevirtual

Implements webrtc::Transport.

◆ SendRtcp() [2/2]

bool webrtc::voe::Channel::SendRtcp	(	const uint8_t *	data,
		size_t	len
	)

overridevirtual

Implements webrtc::Transport.

◆ SendRtp() [1/2]

bool webrtc::voe::Channel::SendRtp	(	const uint8_t *	data,
		size_t	len,
		const PacketOptions &	packet_options
	)

overridevirtual

Implements webrtc::Transport.

◆ SendRtp() [2/2]

bool webrtc::voe::Channel::SendRtp	(	const uint8_t *	data,
		size_t	len,
		const PacketOptions &	packet_options
	)

overridevirtual

Implements webrtc::Transport.

◆ SendTelephoneEventOutband() [1/2]

int webrtc::voe::Channel::SendTelephoneEventOutband	(	int	event,
		int	duration_ms
	)

◆ SendTelephoneEventOutband() [2/2]

int webrtc::voe::Channel::SendTelephoneEventOutband	(	int	event,
		int	duration_ms
	)

◆ set_associate_send_channel() [1/2]

void webrtc::voe::Channel::set_associate_send_channel ( const ChannelOwner & channel )

◆ set_associate_send_channel() [2/2]

void webrtc::voe::Channel::set_associate_send_channel ( const ChannelOwner & channel )

◆ SetBitRate() [1/2]

void webrtc::voe::Channel::SetBitRate	(	int	bitrate_bps,
		int64_t	probing_interval_ms
	)

◆ SetBitRate() [2/2]

void webrtc::voe::Channel::SetBitRate	(	int	bitrate_bps,
		int64_t	probing_interval_ms
	)

◆ SetChannelOutputVolumeScaling() [1/2]

int webrtc::voe::Channel::SetChannelOutputVolumeScaling ( float scaling )

◆ SetChannelOutputVolumeScaling() [2/2]

int webrtc::voe::Channel::SetChannelOutputVolumeScaling ( float scaling )

◆ SetCodecFECStatus() [1/2]

int webrtc::voe::Channel::SetCodecFECStatus ( bool enable )

◆ SetCodecFECStatus() [2/2]

int webrtc::voe::Channel::SetCodecFECStatus ( bool enable )

◆ SetEngineInformation() [1/2]

int32_t webrtc::voe::Channel::SetEngineInformation	(	Statistics &	engineStatistics,
		OutputMixer &	outputMixer,
		ProcessThread &	moduleProcessThread,
		AudioDeviceModule &	audioDeviceModule,
		VoiceEngineObserver *	voiceEngineObserver,
		rtc::CriticalSection *	callbackCritSect
	)

◆ SetEngineInformation() [2/2]

int32_t webrtc::voe::Channel::SetEngineInformation	(	Statistics &	engineStatistics,
		OutputMixer &	outputMixer,
		ProcessThread &	moduleProcessThread,
		AudioDeviceModule &	audioDeviceModule,
		VoiceEngineObserver *	voiceEngineObserver,
		rtc::CriticalSection *	callbackCritSect
	)

◆ SetInputMute() [1/2]

int webrtc::voe::Channel::SetInputMute ( bool enable )

◆ SetInputMute() [2/2]

int webrtc::voe::Channel::SetInputMute ( bool enable )

◆ SetLocalSSRC() [1/2]

int webrtc::voe::Channel::SetLocalSSRC ( unsigned int ssrc )

◆ SetLocalSSRC() [2/2]

int webrtc::voe::Channel::SetLocalSSRC ( unsigned int ssrc )

◆ SetMinimumPlayoutDelay() [1/2]

int webrtc::voe::Channel::SetMinimumPlayoutDelay ( int delayMs )

◆ SetMinimumPlayoutDelay() [2/2]

int webrtc::voe::Channel::SetMinimumPlayoutDelay ( int delayMs )

◆ SetMixWithMicStatus() [1/2]

void webrtc::voe::Channel::SetMixWithMicStatus ( bool mix )

◆ SetMixWithMicStatus() [2/2]

void webrtc::voe::Channel::SetMixWithMicStatus ( bool mix )

◆ SetNACKStatus() [1/2]

void webrtc::voe::Channel::SetNACKStatus	(	bool	enable,
		int	maxNumberOfPackets
	)

◆ SetNACKStatus() [2/2]

void webrtc::voe::Channel::SetNACKStatus	(	bool	enable,
		int	maxNumberOfPackets
	)

◆ SetOpusDtx() [1/2]

int webrtc::voe::Channel::SetOpusDtx ( bool enable_dtx )

◆ SetOpusDtx() [2/2]

int webrtc::voe::Channel::SetOpusDtx ( bool enable_dtx )

◆ SetOpusMaxPlaybackRate() [1/2]

int webrtc::voe::Channel::SetOpusMaxPlaybackRate ( int frequency_hz )

◆ SetOpusMaxPlaybackRate() [2/2]

int webrtc::voe::Channel::SetOpusMaxPlaybackRate ( int frequency_hz )

◆ SetOutputVolumePan() [1/2]

int webrtc::voe::Channel::SetOutputVolumePan	(	float	left,
		float	right
	)

◆ SetOutputVolumePan() [2/2]

int webrtc::voe::Channel::SetOutputVolumePan	(	float	left,
		float	right
	)

◆ SetReceiveAudioLevelIndicationStatus() [1/2]

int webrtc::voe::Channel::SetReceiveAudioLevelIndicationStatus	(	bool	enable,
		unsigned char	id
	)

◆ SetReceiveAudioLevelIndicationStatus() [2/2]

int webrtc::voe::Channel::SetReceiveAudioLevelIndicationStatus	(	bool	enable,
		unsigned char	id
	)

◆ SetReceiverFrameLengthRange() [1/2]

void webrtc::voe::Channel::SetReceiverFrameLengthRange	(	int	min_frame_length_ms,
		int	max_frame_length_ms
	)

◆ SetReceiverFrameLengthRange() [2/2]

void webrtc::voe::Channel::SetReceiverFrameLengthRange	(	int	min_frame_length_ms,
		int	max_frame_length_ms
	)

◆ SetRecPayloadType() [1/4]

int32_t webrtc::voe::Channel::SetRecPayloadType ( const CodecInst & codec )

◆ SetRecPayloadType() [2/4]

int32_t webrtc::voe::Channel::SetRecPayloadType ( const CodecInst & codec )

◆ SetRecPayloadType() [3/4]

int32_t webrtc::voe::Channel::SetRecPayloadType	(	int	payload_type,
		const SdpAudioFormat &	format
	)

◆ SetRecPayloadType() [4/4]

int32_t webrtc::voe::Channel::SetRecPayloadType	(	int	payload_type,
		const SdpAudioFormat &	format
	)

◆ SetRtcEventLog() [1/2]

void webrtc::voe::Channel::SetRtcEventLog ( RtcEventLog * event_log )

◆ SetRtcEventLog() [2/2]

void webrtc::voe::Channel::SetRtcEventLog ( RtcEventLog * event_log )

◆ SetRTCP_CNAME() [1/2]

int webrtc::voe::Channel::SetRTCP_CNAME ( const char cName[256] )

◆ SetRTCP_CNAME() [2/2]

int webrtc::voe::Channel::SetRTCP_CNAME ( const char cName[256] )

◆ SetRtcpRttStats() [1/2]

void webrtc::voe::Channel::SetRtcpRttStats ( RtcpRttStats * rtcp_rtt_stats )

◆ SetRtcpRttStats() [2/2]

void webrtc::voe::Channel::SetRtcpRttStats ( RtcpRttStats * rtcp_rtt_stats )

◆ SetRTCPStatus() [1/2]

void webrtc::voe::Channel::SetRTCPStatus ( bool enable )

◆ SetRTCPStatus() [2/2]

void webrtc::voe::Channel::SetRTCPStatus ( bool enable )

◆ SetSendAudioLevelIndicationStatus() [1/2]

int webrtc::voe::Channel::SetSendAudioLevelIndicationStatus	(	bool	enable,
		unsigned char	id
	)

◆ SetSendAudioLevelIndicationStatus() [2/2]

int webrtc::voe::Channel::SetSendAudioLevelIndicationStatus	(	bool	enable,
		unsigned char	id
	)

◆ SetSendCNPayloadType() [1/2]

int32_t webrtc::voe::Channel::SetSendCNPayloadType	(	int	type,
		PayloadFrequencies	frequency
	)

◆ SetSendCNPayloadType() [2/2]

int32_t webrtc::voe::Channel::SetSendCNPayloadType	(	int	type,
		PayloadFrequencies	frequency
	)

◆ SetSendCodec() [1/2]

int32_t webrtc::voe::Channel::SetSendCodec ( const CodecInst & codec )

◆ SetSendCodec() [2/2]

int32_t webrtc::voe::Channel::SetSendCodec ( const CodecInst & codec )

◆ SetSendTelephoneEventPayloadType() [1/2]

int webrtc::voe::Channel::SetSendTelephoneEventPayloadType	(	int	payload_type,
		int	payload_frequency
	)

◆ SetSendTelephoneEventPayloadType() [2/2]

int webrtc::voe::Channel::SetSendTelephoneEventPayloadType	(	int	payload_type,
		int	payload_frequency
	)

◆ SetSink() [1/2]

void webrtc::voe::Channel::SetSink ( std::unique_ptr< AudioSinkInterface > sink )

◆ SetSink() [2/2]

void webrtc::voe::Channel::SetSink ( std::unique_ptr< AudioSinkInterface > sink )

◆ SetTransportOverhead() [1/2]

void webrtc::voe::Channel::SetTransportOverhead ( size_t transport_overhead_per_packet )

◆ SetTransportOverhead() [2/2]

void webrtc::voe::Channel::SetTransportOverhead ( size_t transport_overhead_per_packet )

◆ SetVADStatus() [1/2]

int32_t webrtc::voe::Channel::SetVADStatus	(	bool	enableVAD,
		ACMVADMode	mode,
		bool	disableDTX
	)

◆ SetVADStatus() [2/2]

int32_t webrtc::voe::Channel::SetVADStatus	(	bool	enableVAD,
		ACMVADMode	mode,
		bool	disableDTX
	)

◆ StartPlayingFileAsMicrophone() [1/4]

int webrtc::voe::Channel::StartPlayingFileAsMicrophone	(	const char *	fileName,
		bool	loop,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileAsMicrophone() [2/4]

int webrtc::voe::Channel::StartPlayingFileAsMicrophone	(	const char *	fileName,
		bool	loop,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileAsMicrophone() [3/4]

int webrtc::voe::Channel::StartPlayingFileAsMicrophone	(	InStream *	stream,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileAsMicrophone() [4/4]

int webrtc::voe::Channel::StartPlayingFileAsMicrophone	(	InStream *	stream,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileLocally() [1/4]

int webrtc::voe::Channel::StartPlayingFileLocally	(	const char *	fileName,
		bool	loop,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileLocally() [2/4]

int webrtc::voe::Channel::StartPlayingFileLocally	(	const char *	fileName,
		bool	loop,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileLocally() [3/4]

int webrtc::voe::Channel::StartPlayingFileLocally	(	InStream *	stream,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayingFileLocally() [4/4]

int webrtc::voe::Channel::StartPlayingFileLocally	(	InStream *	stream,
		FileFormats	format,
		int	startPosition,
		float	volumeScaling,
		int	stopPosition,
		const CodecInst *	codecInst
	)

◆ StartPlayout() [1/2]

int32_t webrtc::voe::Channel::StartPlayout ( )

◆ StartPlayout() [2/2]

int32_t webrtc::voe::Channel::StartPlayout ( )

◆ StartRecordingPlayout() [1/4]

int webrtc::voe::Channel::StartRecordingPlayout	(	const char *	fileName,
		const CodecInst *	codecInst
	)

◆ StartRecordingPlayout() [2/4]

int webrtc::voe::Channel::StartRecordingPlayout	(	const char *	fileName,
		const CodecInst *	codecInst
	)

◆ StartRecordingPlayout() [3/4]

int webrtc::voe::Channel::StartRecordingPlayout	(	OutStream *	stream,
		const CodecInst *	codecInst
	)

◆ StartRecordingPlayout() [4/4]

int webrtc::voe::Channel::StartRecordingPlayout	(	OutStream *	stream,
		const CodecInst *	codecInst
	)

◆ StartSend() [1/2]

int32_t webrtc::voe::Channel::StartSend ( )

◆ StartSend() [2/2]

int32_t webrtc::voe::Channel::StartSend ( )

◆ StopPlayingFileAsMicrophone() [1/2]

int webrtc::voe::Channel::StopPlayingFileAsMicrophone ( )

◆ StopPlayingFileAsMicrophone() [2/2]

int webrtc::voe::Channel::StopPlayingFileAsMicrophone ( )

◆ StopPlayingFileLocally() [1/2]

int webrtc::voe::Channel::StopPlayingFileLocally ( )

◆ StopPlayingFileLocally() [2/2]

int webrtc::voe::Channel::StopPlayingFileLocally ( )

◆ StopPlayout() [1/2]

int32_t webrtc::voe::Channel::StopPlayout ( )

◆ StopPlayout() [2/2]

int32_t webrtc::voe::Channel::StopPlayout ( )

◆ StopRecordingPlayout() [1/2]

int webrtc::voe::Channel::StopRecordingPlayout ( )

◆ StopRecordingPlayout() [2/2]

int webrtc::voe::Channel::StopRecordingPlayout ( )

◆ StopSend() [1/2]

int32_t webrtc::voe::Channel::StopSend ( )

◆ StopSend() [2/2]

int32_t webrtc::voe::Channel::StopSend ( )

◆ UpdateLocalTimeStamp() [1/2]

int32_t webrtc::voe::Channel::UpdateLocalTimeStamp ( )

◆ UpdateLocalTimeStamp() [2/2]

int32_t webrtc::voe::Channel::UpdateLocalTimeStamp ( )

◆ VoiceActivityIndicator() [1/2]

int webrtc::voe::Channel::VoiceActivityIndicator ( int & activity )

◆ VoiceActivityIndicator() [2/2]

int webrtc::voe::Channel::VoiceActivityIndicator ( int & activity )

Friends And Related Function Documentation

◆ VoERtcpObserver

VoERtcpObserver

friend

The documentation for this class was generated from the following files:

DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/voice_engine/channel.h
Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.cc

Public Types

Public Member Functions

Static Public Member Functions

Protected Member Functions

Friends

Additional Inherited Members

Member Enumeration Documentation

◆ anonymous enum

◆ anonymous enum

◆ anonymous enum

◆ anonymous enum

Constructor & Destructor Documentation

◆ ~Channel() [1/2]

◆ Channel() [1/2]

◆ ~Channel() [2/2]

◆ Channel() [2/2]

Member Function Documentation

◆ ChannelId() [1/2]

◆ ChannelId() [2/2]

◆ CreateChannel() [1/2]

◆ CreateChannel() [2/2]

◆ Demultiplex() [1/4]

◆ Demultiplex() [2/4]

◆ Demultiplex() [3/4]

◆ Demultiplex() [4/4]

◆ DeRegisterExternalTransport() [1/2]

◆ DeRegisterExternalTransport() [2/2]

◆ DeRegisterVoiceEngineObserver() [1/2]

◆ DeRegisterVoiceEngineObserver() [2/2]

◆ DisableAudioNetworkAdaptor() [1/2]

◆ DisableAudioNetworkAdaptor() [2/2]

◆ DisassociateSendChannel() [1/2]

◆ DisassociateSendChannel() [2/2]

◆ EnableAudioNetworkAdaptor() [1/2]

◆ EnableAudioNetworkAdaptor() [2/2]

◆ EnableReceiveTransportSequenceNumber() [1/2]

◆ EnableReceiveTransportSequenceNumber() [2/2]

◆ EnableSendTransportSequenceNumber() [1/2]

◆ EnableSendTransportSequenceNumber() [2/2]

◆ EncodeAndSend() [1/2]

◆ EncodeAndSend() [2/2]

◆ ExternalTransport() [1/2]

◆ ExternalTransport() [2/2]

◆ GetAudioDecoderFactory() [1/2]

◆ GetAudioDecoderFactory() [2/2]

◆ GetAudioFrameWithInfo() [1/2]

◆ GetAudioFrameWithInfo() [2/2]

◆ GetAudioFrameWithMuted() [1/2]

◆ GetAudioFrameWithMuted() [2/2]

◆ GetChannelOutputVolumeScaling() [1/2]

◆ GetChannelOutputVolumeScaling() [2/2]

◆ GetCodecFECStatus() [1/2]

◆ GetCodecFECStatus() [2/2]

◆ GetDecodingCallStatistics() [1/2]

◆ GetDecodingCallStatistics() [2/2]

◆ GetDelayEstimate() [1/2]

◆ GetDelayEstimate() [2/2]

◆ GetLocalSSRC() [1/2]

◆ GetLocalSSRC() [2/2]

◆ GetNetworkStatistics() [1/2]

◆ GetNetworkStatistics() [2/2]

◆ GetOpusDtx() [1/2]

◆ GetOpusDtx() [2/2]

◆ GetOutputVolumePan() [1/2]

◆ GetOutputVolumePan() [2/2]

◆ GetPlayoutTimestamp() [1/2]

◆ GetPlayoutTimestamp() [2/2]

◆ GetRecCodec() [1/2]

◆ GetRecCodec() [2/2]

◆ GetRecPayloadType() [1/2]

◆ GetRecPayloadType() [2/2]

◆ GetRemoteRTCP_CNAME() [1/2]

◆ GetRemoteRTCP_CNAME() [2/2]

◆ GetRemoteRTCPReportBlocks() [1/2]

◆ GetRemoteRTCPReportBlocks() [2/2]

◆ GetRemoteSSRC() [1/2]

◆ GetRemoteSSRC() [2/2]

◆ GetRTCPStatus() [1/2]

◆ GetRTCPStatus() [2/2]

◆ GetRtpRtcp() [1/2]