webrtc::AudioRtpSender Member List

This is the complete list of members for webrtc::AudioRtpSender, including all inherited members.

$AddTrackToStats()	webrtc::AudioRtpSender
$AttachmentId() const	webrtc::RtpSenderBase
$AttachTrack()	webrtc::AudioRtpSender
$CanInsertDtmf()	webrtc::AudioRtpSender
$CheckCodecParameters(::webrtc::RtpParameters const &parameters)	webrtc::RtpSenderBase
$ClearSend()	webrtc::AudioRtpSender
$ctor(::rtc::Thread *worker_thread, ::std::string const &id, ::webrtc::LegacyStatsCollectorInterface *legacy_stats, ::webrtc::RtpSenderBase::SetStreamsObserver *set_streams_observer)	webrtc::AudioRtpSender
webrtc::RtpSenderBase::$ctor(::rtc::Thread *worker_thread, ::std::string const &id, ::webrtc::RtpSenderBase::SetStreamsObserver *set_streams_observer)	webrtc::RtpSenderBase
$DetachTrack()	webrtc::AudioRtpSender
$DisableEncodingLayers(::std::vector<::std::string > const &rids)	webrtc::RtpSenderBase
$dtls_transport() const	webrtc::RtpSenderBase
$dtor()	webrtc::AudioRtpSender
$GenerateKeyFrame(::std::vector<::std::string > const &rids)	webrtc::AudioRtpSender
$GetDtmfSender() const	webrtc::AudioRtpSender
$GetFrameEncryptor() const	webrtc::RtpSenderBase
$GetParameters() const	webrtc::RtpSenderBase
$GetParametersInternal() const	webrtc::RtpSenderBase
$GetParametersInternalWithAllLayers() const	webrtc::RtpSenderBase
$id() const	webrtc::RtpSenderBase
$init_send_encodings() const	webrtc::RtpSenderBase
$InsertDtmf(int code, int duration)	webrtc::AudioRtpSender
$media_type() const	webrtc::AudioRtpSender
$OnChanged()	webrtc::AudioRtpSender
$RemoveTrackFromStats()	webrtc::AudioRtpSender
$set_init_send_encodings(::std::vector<::webrtc::RtpEncodingParameters > const &init_send_encodings)	webrtc::RtpSenderBase
$set_stream_ids(::std::vector<::std::string > const &stream_ids)	webrtc::RtpSenderBase
$set_transport(::webrtc::scoped_refptr<::webrtc::DtlsTransportInterface > dtls_transport)	webrtc::RtpSenderBase
$SetEncoderSelector(::std::unique_ptr<::webrtc::VideoEncoderFactory::EncoderSelectorInterface > encoder_selector)	webrtc::RtpSenderBase
$SetEncoderToPacketizerFrameTransformer(::webrtc::scoped_refptr<::webrtc::FrameTransformerInterface > frame_transformer)	webrtc::RtpSenderBase
$SetFrameEncryptor(::webrtc::scoped_refptr<::webrtc::FrameEncryptorInterface > frame_encryptor)	webrtc::RtpSenderBase
$SetMediaChannel(::cricket::MediaSendChannelInterface *media_channel)	webrtc::RtpSenderBase
$SetParameters(::webrtc::RtpParameters const &parameters)	webrtc::RtpSenderBase
$SetParametersAsync(::webrtc::RtpParameters const &parameters, ::absl::AnyInvocable< void(::webrtc::RTCError) && > callback)	webrtc::RtpSenderBase
$SetParametersInternal(::webrtc::RtpParameters const &parameters, ::absl::AnyInvocable< void(::webrtc::RTCError) && > callback, bool blocking)	webrtc::RtpSenderBase
$SetParametersInternalWithAllLayers(::webrtc::RtpParameters const &parameters)	webrtc::RtpSenderBase
$SetSend()	webrtc::AudioRtpSender
$SetSendCodecs(::std::vector<::cricket::Codec > send_codecs)	webrtc::RtpSenderBase
$SetSsrc(uint ssrc)	webrtc::RtpSenderBase
$SetStreams(::std::vector<::std::string > const &stream_ids)	webrtc::RtpSenderBase
$SetTrack(::webrtc::MediaStreamTrackInterface *track)	webrtc::RtpSenderBase
$SetTransceiverAsStopped()	webrtc::RtpSenderBase
$ssrc() const	webrtc::RtpSenderBase
$Stop()	webrtc::RtpSenderBase
$stream_ids() const	webrtc::RtpSenderBase
$track() const	webrtc::RtpSenderBase
$track_kind() const	webrtc::AudioRtpSender
$vftable()	webrtc::AudioRtpSender	static
$vftableForObserverInterface()	webrtc::AudioRtpSender	static
$vftableForRtpSenderInternal()	webrtc::AudioRtpSender	static
AddRef() const =0 (defined in webrtc::RefCountInterface)	webrtc::RefCountInterface	pure virtual
AddTrackToStats() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
AttachmentId() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
AttachTrack() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
AudioRtpSender(AudioRtpSender const &) (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
AudioRtpSender() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
AudioRtpSender(::rtc::Thread *worker_thread, ::std::string const &id, ::webrtc::LegacyStatsCollectorInterface *legacy_stats, ::webrtc::RtpSenderBase::SetStreamsObserver *set_streams_observer)	webrtc::AudioRtpSender
CanInsertDtmf() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
CheckCodecParameters(::webrtc::RtpParameters const &parameters) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
CheckSetParameters(::webrtc::RtpParameters const &parameters)	webrtc::RtpSenderBase
ClearSend() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
Create(::rtc::Thread *worker_thread, ::std::string const &id, ::webrtc::LegacyStatsCollectorInterface *stats, ::webrtc::RtpSenderBase::SetStreamsObserver *set_streams_observer)	webrtc::AudioRtpSender	static
DetachTrack() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
DisableEncodingLayers(::std::vector<::std::string > const &rids) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
dtls_transport() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
GenerateKeyFrame(::std::vector<::std::string > const &rids) (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
GetDtmfSender() const (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
GetFrameEncryptor() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
GetParameters() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
GetParametersInternal() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
GetParametersInternalWithAllLayers() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
id() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
init_send_encodings() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
InsertDtmf(int code, int duration) (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
media_type() const (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
mUnk1a7b11 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk22488a (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk2e12f7 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk393418 (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
mUnk3a48da (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk3e58b8 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk403096 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk4f7245 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk6b9467 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk74471a (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
mUnk7d6a99 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk871216 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnk8d3d1e (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
mUnk92bc60 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnka71bc1 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkb1c1d7 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkb28e63 (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
mUnkcfb04f (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkd68b10 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnke1b367 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkec7d87 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkedb02c (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
mUnkeeca2f (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
mUnkffabe7 (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
OnChanged() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
operator=(AudioRtpSender const &) (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender
operator=(RtpSenderBase const &) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
Release() const =0 (defined in webrtc::RefCountInterface)	webrtc::RefCountInterface	pure virtual
RemoveTrackFromStats() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
RtpSenderBase(RtpSenderBase const &) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
RtpSenderBase() (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase
RtpSenderBase(::rtc::Thread *worker_thread, ::std::string const &id, ::webrtc::RtpSenderBase::SetStreamsObserver *set_streams_observer)	webrtc::RtpSenderBase
set_init_send_encodings(::std::vector<::webrtc::RtpEncodingParameters > const &init_send_encodings) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
set_stream_ids(::std::vector<::std::string > const &stream_ids) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
set_transport(::webrtc::scoped_refptr<::webrtc::DtlsTransportInterface > dtls_transport) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetEncoderSelector(::std::unique_ptr<::webrtc::VideoEncoderFactory::EncoderSelectorInterface > encoder_selector) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetEncoderSelectorOnChannel()	webrtc::RtpSenderBase
SetEncoderToPacketizerFrameTransformer(::webrtc::scoped_refptr<::webrtc::FrameTransformerInterface > frame_transformer) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetFrameEncryptor(::webrtc::scoped_refptr<::webrtc::FrameEncryptorInterface > frame_encryptor) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetMediaChannel(::cricket::MediaSendChannelInterface *media_channel) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetParameters(::webrtc::RtpParameters const &parameters) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetParametersAsync(::webrtc::RtpParameters const &parameters, ::absl::AnyInvocable< void(::webrtc::RTCError) && > callback) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetParametersInternal(::webrtc::RtpParameters const &parameters, ::absl::AnyInvocable< void(::webrtc::RTCError) && > callback, bool blocking) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetParametersInternalWithAllLayers(::webrtc::RtpParameters const &parameters) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetSend() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
SetSendCodecs(::std::vector<::cricket::Codec > send_codecs) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetSsrc(uint ssrc) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetStreams(::std::vector<::std::string > const &stream_ids) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetTrack(::webrtc::MediaStreamTrackInterface *track) (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
SetTransceiverAsStopped() (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
ssrc() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
Stop() (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
stream_ids() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
track() const (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
track_kind() const (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
~AudioRtpSender() (defined in webrtc::AudioRtpSender)	webrtc::AudioRtpSender	virtual
~DtmfProviderInterface() (defined in webrtc::DtmfProviderInterface)	webrtc::DtmfProviderInterface	virtual
~ObserverInterface() (defined in webrtc::ObserverInterface)	webrtc::ObserverInterface	virtual
~RefCountInterface()=default (defined in webrtc::RefCountInterface)	webrtc::RefCountInterface	virtual
~RtpSenderBase() (defined in webrtc::RtpSenderBase)	webrtc::RtpSenderBase	virtual
~RtpSenderInterface()=default (defined in webrtc::RtpSenderInterface)	webrtc::RtpSenderInterface	virtual
~RtpSenderInternal() (defined in webrtc::RtpSenderInternal)	webrtc::RtpSenderInternal	virtual