webrtc::internal::AudioSendStream Class Reference

#include <audio_send_stream.h>

Inheritance diagram for webrtc::internal::AudioSendStream:

Public Member Functions
	AudioSendStream (const webrtc::AudioSendStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, rtc::TaskQueue *worker_queue, PacketRouter *packet_router, CongestionController *congestion_controller, BitrateAllocator *bitrate_allocator, RtcEventLog *event_log, RtcpRttStats *rtcp_rtt_stats)
	~AudioSendStream () override
void	Start () override
void	Stop () override
bool	SendTelephoneEvent (int payload_type, int payload_frequency, int event, int duration_ms) override
void	SetMuted (bool muted) override
webrtc::AudioSendStream::Stats	GetStats () const override
void	SignalNetworkState (NetworkState state)
bool	DeliverRtcp (const uint8_t *packet, size_t length)
uint32_t	OnBitrateUpdated (uint32_t bitrate_bps, uint8_t fraction_loss, int64_t rtt, int64_t probing_interval_ms) override
const webrtc::AudioSendStream::Config &	config () const
void	SetTransportOverhead (int transport_overhead_per_packet)
	AudioSendStream (const webrtc::AudioSendStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, rtc::TaskQueue *worker_queue, PacketRouter *packet_router, CongestionController *congestion_controller, BitrateAllocator *bitrate_allocator, RtcEventLog *event_log, RtcpRttStats *rtcp_rtt_stats)
	~AudioSendStream () override
void	Start () override
void	Stop () override
bool	SendTelephoneEvent (int payload_type, int payload_frequency, int event, int duration_ms) override
void	SetMuted (bool muted) override
webrtc::AudioSendStream::Stats	GetStats () const override
void	SignalNetworkState (NetworkState state)
bool	DeliverRtcp (const uint8_t *packet, size_t length)
uint32_t	OnBitrateUpdated (uint32_t bitrate_bps, uint8_t fraction_loss, int64_t rtt, int64_t probing_interval_ms) override
const webrtc::AudioSendStream::Config &	config () const
void	SetTransportOverhead (int transport_overhead_per_packet)

Additional Inherited Members
Protected Member Functions inherited from webrtc::BitrateAllocatorObserver
virtual	~BitrateAllocatorObserver ()
virtual	~BitrateAllocatorObserver ()

Constructor & Destructor Documentation

AudioSendStream() [1/2]

webrtc::internal::AudioSendStream::AudioSendStream	(	const webrtc::AudioSendStream::Config &	config,
		const rtc::scoped_refptr< webrtc::AudioState > &	audio_state,
		rtc::TaskQueue *	worker_queue,
		PacketRouter *	packet_router,
		CongestionController *	congestion_controller,
		BitrateAllocator *	bitrate_allocator,
		RtcEventLog *	event_log,
		RtcpRttStats *	rtcp_rtt_stats
	)

~AudioSendStream() [1/2]

webrtc::internal::AudioSendStream::~AudioSendStream ( )

overridevirtual

Reimplemented from webrtc::AudioSendStream.

AudioSendStream() [2/2]

webrtc::internal::AudioSendStream::AudioSendStream	(	const webrtc::AudioSendStream::Config &	config,
		const rtc::scoped_refptr< webrtc::AudioState > &	audio_state,
		rtc::TaskQueue *	worker_queue,
		PacketRouter *	packet_router,
		CongestionController *	congestion_controller,
		BitrateAllocator *	bitrate_allocator,
		RtcEventLog *	event_log,
		RtcpRttStats *	rtcp_rtt_stats
	)

~AudioSendStream() [2/2]

webrtc::internal::AudioSendStream::~AudioSendStream ( )

overridevirtual

Reimplemented from webrtc::AudioSendStream.

Member Function Documentation

config() [1/2]

const webrtc::AudioSendStream::Config & webrtc::internal::AudioSendStream::config

(

)

const

config() [2/2]

const webrtc::AudioSendStream::Config& webrtc::internal::AudioSendStream::config

(

)

const

DeliverRtcp() [1/2]

bool webrtc::internal::AudioSendStream::DeliverRtcp	(	const uint8_t *	packet,
		size_t	length
	)

DeliverRtcp() [2/2]

bool webrtc::internal::AudioSendStream::DeliverRtcp	(	const uint8_t *	packet,
		size_t	length
	)

GetStats() [1/2]

webrtc::AudioSendStream::Stats webrtc::internal::AudioSendStream::GetStats ( ) const

overridevirtual

Implements webrtc::AudioSendStream.

GetStats() [2/2]

webrtc::AudioSendStream::Stats webrtc::internal::AudioSendStream::GetStats ( ) const

overridevirtual

Implements webrtc::AudioSendStream.

OnBitrateUpdated() [1/2]

uint32_t webrtc::internal::AudioSendStream::OnBitrateUpdated ( uint32_t  bitrate_bps, uint8_t  fraction_loss, int64_t  rtt, int64_t  probing_interval_ms  )

overridevirtual

Implements webrtc::BitrateAllocatorObserver.

OnBitrateUpdated() [2/2]

uint32_t webrtc::internal::AudioSendStream::OnBitrateUpdated ( uint32_t  bitrate_bps, uint8_t  fraction_loss, int64_t  rtt, int64_t  probing_interval_ms  )

overridevirtual

Implements webrtc::BitrateAllocatorObserver.

SendTelephoneEvent() [1/2]

bool webrtc::internal::AudioSendStream::SendTelephoneEvent ( int  payload_type, int  payload_frequency, int  event, int  duration_ms  )

overridevirtual

Implements webrtc::AudioSendStream.

SendTelephoneEvent() [2/2]

bool webrtc::internal::AudioSendStream::SendTelephoneEvent ( int  payload_type, int  payload_frequency, int  event, int  duration_ms  )

overridevirtual

Implements webrtc::AudioSendStream.

SetMuted() [1/2]

void webrtc::internal::AudioSendStream::SetMuted ( bool  muted )

overridevirtual

Implements webrtc::AudioSendStream.

SetMuted() [2/2]

void webrtc::internal::AudioSendStream::SetMuted ( bool  muted )

overridevirtual

Implements webrtc::AudioSendStream.

SetTransportOverhead() [1/2]

void webrtc::internal::AudioSendStream::SetTransportOverhead

(

int

transport_overhead_per_packet

)

SetTransportOverhead() [2/2]

void webrtc::internal::AudioSendStream::SetTransportOverhead

(

int

transport_overhead_per_packet

)

SignalNetworkState() [1/2]

void webrtc::internal::AudioSendStream::SignalNetworkState

(

NetworkState

state

)

SignalNetworkState() [2/2]

void webrtc::internal::AudioSendStream::SignalNetworkState

(

NetworkState

state

)

Start() [1/2]

void webrtc::internal::AudioSendStream::Start ( )

overridevirtual

Implements webrtc::AudioSendStream.

Start() [2/2]

void webrtc::internal::AudioSendStream::Start ( )

overridevirtual

Implements webrtc::AudioSendStream.

Stop() [1/2]

void webrtc::internal::AudioSendStream::Stop ( )

overridevirtual

Implements webrtc::AudioSendStream.

Stop() [2/2]

void webrtc::internal::AudioSendStream::Stop ( )

overridevirtual

Implements webrtc::AudioSendStream.

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/audio/audio_send_stream.h
• Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc