webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Classes | Public Member Functions | Public Attributes | Static Public Attributes | List of all members
webrtc::VideoSendStream::Config Struct Reference

#include <video_send_stream.h>

Classes

struct  EncoderSettings
 
struct  Rtp
 

Public Member Functions

 Config ()=delete
 
 Config (Config &&)=default
 
 Config (Transport *send_transport)
 
Configoperator= (Config &&)=default
 
Configoperator= (const Config &)=delete
 
Config Copy () const
 
std::string ToString () const
 
 Config ()=delete
 
 Config (Config &&)=default
 
 Config (Transport *send_transport)
 
Configoperator= (Config &&)=default
 
Configoperator= (const Config &)=delete
 
Config Copy () const
 
std::string ToString () const
 

Public Attributes

struct webrtc::VideoSendStream::Config::EncoderSettings encoder_settings
 
struct webrtc::VideoSendStream::Config::Rtp rtp
 
Transportsend_transport = nullptr
 
rtc::VideoSinkInterface< VideoFrame > * pre_encode_callback = nullptr
 
EncodedFrameObserverpost_encode_callback = nullptr
 
int render_delay_ms = 0
 
int target_delay_ms = 0
 
bool suspend_below_min_bitrate = false
 
bool periodic_alr_bandwidth_probing = false
 

Static Public Attributes

static const size_t kDefaultMaxPacketSize = 1500 - 40
 

Constructor & Destructor Documentation

◆ Config() [1/6]

webrtc::VideoSendStream::Config::Config ( )
delete

◆ Config() [2/6]

webrtc::VideoSendStream::Config::Config ( Config &&  )
default

◆ Config() [3/6]

webrtc::VideoSendStream::Config::Config ( Transport send_transport)
inlineexplicit

◆ Config() [4/6]

webrtc::VideoSendStream::Config::Config ( )
delete

◆ Config() [5/6]

webrtc::VideoSendStream::Config::Config ( Config &&  )
default

◆ Config() [6/6]

webrtc::VideoSendStream::Config::Config ( Transport send_transport)
inlineexplicit

Member Function Documentation

◆ Copy() [1/2]

Config webrtc::VideoSendStream::Config::Copy ( ) const
inline

◆ Copy() [2/2]

Config webrtc::VideoSendStream::Config::Copy ( ) const
inline

◆ operator=() [1/4]

Config& webrtc::VideoSendStream::Config::operator= ( Config &&  )
default

◆ operator=() [2/4]

Config& webrtc::VideoSendStream::Config::operator= ( Config &&  )
default

◆ operator=() [3/4]

Config& webrtc::VideoSendStream::Config::operator= ( const Config )
delete

◆ operator=() [4/4]

Config& webrtc::VideoSendStream::Config::operator= ( const Config )
delete

◆ ToString() [1/2]

std::string webrtc::VideoSendStream::Config::ToString ( ) const

◆ ToString() [2/2]

std::string webrtc::VideoSendStream::Config::ToString ( ) const

Member Data Documentation

◆ encoder_settings

struct webrtc::VideoSendStream::Config::EncoderSettings webrtc::VideoSendStream::Config::encoder_settings

◆ kDefaultMaxPacketSize

static const size_t webrtc::VideoSendStream::Config::kDefaultMaxPacketSize = 1500 - 40
static

◆ periodic_alr_bandwidth_probing

bool webrtc::VideoSendStream::Config::periodic_alr_bandwidth_probing = false

◆ post_encode_callback

EncodedFrameObserver * webrtc::VideoSendStream::Config::post_encode_callback = nullptr

◆ pre_encode_callback

rtc::VideoSinkInterface< VideoFrame > * webrtc::VideoSendStream::Config::pre_encode_callback = nullptr

◆ render_delay_ms

int webrtc::VideoSendStream::Config::render_delay_ms = 0

◆ rtp

struct webrtc::VideoSendStream::Config::Rtp webrtc::VideoSendStream::Config::rtp

◆ send_transport

Transport * webrtc::VideoSendStream::Config::send_transport = nullptr

◆ suspend_below_min_bitrate

bool webrtc::VideoSendStream::Config::suspend_below_min_bitrate = false

◆ target_delay_ms

int webrtc::VideoSendStream::Config::target_delay_ms = 0

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