webrtc::AudioSendStream::Config::Rtp Struct Reference

#include <audio_send_stream.h>

Public Member Functions
	Rtp ()
	~Rtp ()
std::string	ToString () const
	Rtp ()
	~Rtp ()
std::string	ToString () const

Public Attributes
uint32_t	ssrc = 0
std::vector< RtpExtension >	extensions
NackConfig	nack
std::string	c_name

Constructor & Destructor Documentation

Rtp() [1/2]

webrtc::AudioSendStream::Config::Rtp::Rtp ( )

default

~Rtp() [1/2]

webrtc::AudioSendStream::Config::Rtp::~Rtp ( )

default

Rtp() [2/2]

webrtc::AudioSendStream::Config::Rtp::Rtp

(

)

~Rtp() [2/2]

webrtc::AudioSendStream::Config::Rtp::~Rtp

(

)

Member Function Documentation

ToString() [1/2]

std::string webrtc::AudioSendStream::Config::Rtp::ToString

(

)

const

ToString() [2/2]

std::string webrtc::AudioSendStream::Config::Rtp::ToString

(

)

const

Member Data Documentation

c_name

std::string webrtc::AudioSendStream::Config::Rtp::c_name

extensions

std::vector< RtpExtension > webrtc::AudioSendStream::Config::Rtp::extensions

nack

NackConfig webrtc::AudioSendStream::Config::Rtp::nack

ssrc

uint32_t webrtc::AudioSendStream::Config::Rtp::ssrc = 0

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

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