webrtc::internal::AudioReceiveStream Class Reference

#include <audio_receive_stream.h>

Inheritance diagram for webrtc::internal::AudioReceiveStream:

Public Member Functions
	AudioReceiveStream (PacketRouter *packet_router, const webrtc::AudioReceiveStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, webrtc::RtcEventLog *event_log)
	~AudioReceiveStream () override
void	Start () override
void	Stop () override
webrtc::AudioReceiveStream::Stats	GetStats () const override
int	GetOutputLevel () const override
void	SetSink (std::unique_ptr< AudioSinkInterface > sink) override
void	SetGain (float gain) override
void	OnRtpPacket (const RtpPacketReceived &packet)
AudioFrameInfo	GetAudioFrameWithInfo (int sample_rate_hz, AudioFrame *audio_frame) override
int	Ssrc () const override
int	PreferredSampleRate () const override
int	id () const override
rtc::Optional< Syncable::Info >	GetInfo () const override
uint32_t	GetPlayoutTimestamp () const override
void	SetMinimumPlayoutDelay (int delay_ms) override
void	AssociateSendStream (AudioSendStream *send_stream)
void	SignalNetworkState (NetworkState state)
bool	DeliverRtcp (const uint8_t *packet, size_t length)
const webrtc::AudioReceiveStream::Config &	config () const
	AudioReceiveStream (PacketRouter *packet_router, const webrtc::AudioReceiveStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, webrtc::RtcEventLog *event_log)
	~AudioReceiveStream () override
void	Start () override
void	Stop () override
webrtc::AudioReceiveStream::Stats	GetStats () const override
int	GetOutputLevel () const override
void	SetSink (std::unique_ptr< AudioSinkInterface > sink) override
void	SetGain (float gain) override
void	OnRtpPacket (const RtpPacketReceived &packet)
AudioFrameInfo	GetAudioFrameWithInfo (int sample_rate_hz, AudioFrame *audio_frame) override
int	Ssrc () const override
int	PreferredSampleRate () const override
int	id () const override
rtc::Optional< Syncable::Info >	GetInfo () const override
uint32_t	GetPlayoutTimestamp () const override
void	SetMinimumPlayoutDelay (int delay_ms) override
void	AssociateSendStream (AudioSendStream *send_stream)
void	SignalNetworkState (NetworkState state)
bool	DeliverRtcp (const uint8_t *packet, size_t length)
const webrtc::AudioReceiveStream::Config &	config () const
Public Member Functions inherited from webrtc::AudioMixer::Source
virtual	~Source ()
virtual	~Source ()
Public Member Functions inherited from webrtc::Syncable
virtual	~Syncable ()
virtual	~Syncable ()

Additional Inherited Members
Public Types inherited from webrtc::AudioMixer::Source
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 }

Constructor & Destructor Documentation

AudioReceiveStream() [1/2]

webrtc::internal::AudioReceiveStream::AudioReceiveStream	(	PacketRouter *	packet_router,
		const webrtc::AudioReceiveStream::Config &	config,
		const rtc::scoped_refptr< webrtc::AudioState > &	audio_state,
		webrtc::RtcEventLog *	event_log
	)

~AudioReceiveStream() [1/2]

webrtc::internal::AudioReceiveStream::~AudioReceiveStream ( )

overridevirtual

Reimplemented from webrtc::AudioReceiveStream.

AudioReceiveStream() [2/2]

webrtc::internal::AudioReceiveStream::AudioReceiveStream	(	PacketRouter *	packet_router,
		const webrtc::AudioReceiveStream::Config &	config,
		const rtc::scoped_refptr< webrtc::AudioState > &	audio_state,
		webrtc::RtcEventLog *	event_log
	)

~AudioReceiveStream() [2/2]

webrtc::internal::AudioReceiveStream::~AudioReceiveStream ( )

overridevirtual

Reimplemented from webrtc::AudioReceiveStream.

Member Function Documentation

AssociateSendStream() [1/2]

void webrtc::internal::AudioReceiveStream::AssociateSendStream

(

AudioSendStream *

send_stream

)

AssociateSendStream() [2/2]

void webrtc::internal::AudioReceiveStream::AssociateSendStream

(

AudioSendStream *

send_stream

)

config() [1/2]

const webrtc::AudioReceiveStream::Config & webrtc::internal::AudioReceiveStream::config

(

)

const

config() [2/2]

const webrtc::AudioReceiveStream::Config& webrtc::internal::AudioReceiveStream::config

(

)

const

DeliverRtcp() [1/2]

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

DeliverRtcp() [2/2]

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

GetAudioFrameWithInfo() [1/2]

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

overridevirtual

Implements webrtc::AudioMixer::Source.

GetAudioFrameWithInfo() [2/2]

AudioFrameInfo webrtc::internal::AudioReceiveStream::GetAudioFrameWithInfo ( int  sample_rate_hz, AudioFrame *  audio_frame  )

overridevirtual

Implements webrtc::AudioMixer::Source.

GetInfo() [1/2]

rtc::Optional< Syncable::Info > webrtc::internal::AudioReceiveStream::GetInfo ( ) const

overridevirtual

Implements webrtc::Syncable.

GetInfo() [2/2]

rtc::Optional<Syncable::Info> webrtc::internal::AudioReceiveStream::GetInfo ( ) const

overridevirtual

Implements webrtc::Syncable.

GetOutputLevel() [1/2]

int webrtc::internal::AudioReceiveStream::GetOutputLevel ( ) const

overridevirtual

Implements webrtc::AudioReceiveStream.

GetOutputLevel() [2/2]

int webrtc::internal::AudioReceiveStream::GetOutputLevel ( ) const

overridevirtual

Implements webrtc::AudioReceiveStream.

GetPlayoutTimestamp() [1/2]

uint32_t webrtc::internal::AudioReceiveStream::GetPlayoutTimestamp ( ) const

overridevirtual

Implements webrtc::Syncable.

GetPlayoutTimestamp() [2/2]

uint32_t webrtc::internal::AudioReceiveStream::GetPlayoutTimestamp ( ) const

overridevirtual

Implements webrtc::Syncable.

GetStats() [1/2]

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

overridevirtual

Implements webrtc::AudioReceiveStream.

GetStats() [2/2]

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

overridevirtual

Implements webrtc::AudioReceiveStream.

id() [1/2]

int webrtc::internal::AudioReceiveStream::id ( ) const

overridevirtual

Implements webrtc::Syncable.

id() [2/2]

int webrtc::internal::AudioReceiveStream::id ( ) const

overridevirtual

Implements webrtc::Syncable.

OnRtpPacket() [1/2]

void webrtc::internal::AudioReceiveStream::OnRtpPacket

(

const RtpPacketReceived &

packet

)

OnRtpPacket() [2/2]

void webrtc::internal::AudioReceiveStream::OnRtpPacket

(

const RtpPacketReceived &

packet

)

PreferredSampleRate() [1/2]

int webrtc::internal::AudioReceiveStream::PreferredSampleRate ( ) const

overridevirtual

Implements webrtc::AudioMixer::Source.

PreferredSampleRate() [2/2]

int webrtc::internal::AudioReceiveStream::PreferredSampleRate ( ) const

overridevirtual

Implements webrtc::AudioMixer::Source.

SetGain() [1/2]

void webrtc::internal::AudioReceiveStream::SetGain ( float  gain )

overridevirtual

Implements webrtc::AudioReceiveStream.

SetGain() [2/2]

void webrtc::internal::AudioReceiveStream::SetGain ( float  gain )

overridevirtual

Implements webrtc::AudioReceiveStream.

SetMinimumPlayoutDelay() [1/2]

void webrtc::internal::AudioReceiveStream::SetMinimumPlayoutDelay ( int  delay_ms )

overridevirtual

Implements webrtc::Syncable.

SetMinimumPlayoutDelay() [2/2]

void webrtc::internal::AudioReceiveStream::SetMinimumPlayoutDelay ( int  delay_ms )

overridevirtual

Implements webrtc::Syncable.

SetSink() [1/2]

void webrtc::internal::AudioReceiveStream::SetSink ( std::unique_ptr< AudioSinkInterface >  sink )

overridevirtual

Implements webrtc::AudioReceiveStream.

SetSink() [2/2]

void webrtc::internal::AudioReceiveStream::SetSink ( std::unique_ptr< AudioSinkInterface >  sink )

overridevirtual

Implements webrtc::AudioReceiveStream.

SignalNetworkState() [1/2]

void webrtc::internal::AudioReceiveStream::SignalNetworkState

(

NetworkState

state

)

SignalNetworkState() [2/2]

void webrtc::internal::AudioReceiveStream::SignalNetworkState

(

NetworkState

state

)

Ssrc() [1/2]

int webrtc::internal::AudioReceiveStream::Ssrc ( ) const

overridevirtual

Implements webrtc::AudioMixer::Source.

Ssrc() [2/2]

int webrtc::internal::AudioReceiveStream::Ssrc ( ) const

overridevirtual

Implements webrtc::AudioMixer::Source.

Start() [1/2]

void webrtc::internal::AudioReceiveStream::Start ( )

overridevirtual

Implements webrtc::AudioReceiveStream.

Start() [2/2]

void webrtc::internal::AudioReceiveStream::Start ( )

overridevirtual

Implements webrtc::AudioReceiveStream.

Stop() [1/2]

void webrtc::internal::AudioReceiveStream::Stop ( )

overridevirtual

Implements webrtc::AudioReceiveStream.

Stop() [2/2]

void webrtc::internal::AudioReceiveStream::Stop ( )

overridevirtual

Implements webrtc::AudioReceiveStream.

---

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

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