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

#include <peerconnectioninterface.h>

Public Member Functions

 RTCConfiguration ()=default
 
 RTCConfiguration (RTCConfigurationType type)
 
bool operator== (const RTCConfiguration &o) const
 
bool operator!= (const RTCConfiguration &o) const
 
bool dscp ()
 
void set_dscp (bool enable)
 
bool cpu_adaptation ()
 
void set_cpu_adaptation (bool enable)
 
bool suspend_below_min_bitrate ()
 
void set_suspend_below_min_bitrate (bool enable)
 
bool prerenderer_smoothing ()
 
void set_prerenderer_smoothing (bool enable)
 
 RTCConfiguration ()=default
 
 RTCConfiguration (RTCConfigurationType type)
 
bool operator== (const RTCConfiguration &o) const
 
bool operator!= (const RTCConfiguration &o) const
 
bool dscp ()
 
void set_dscp (bool enable)
 
bool cpu_adaptation ()
 
void set_cpu_adaptation (bool enable)
 
bool suspend_below_min_bitrate ()
 
void set_suspend_below_min_bitrate (bool enable)
 
bool prerenderer_smoothing ()
 
void set_prerenderer_smoothing (bool enable)
 

Public Attributes

IceServers servers
 
IceTransportsType type = kAll
 
BundlePolicy bundle_policy = kBundlePolicyBalanced
 
RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire
 
std::vector< rtc::scoped_refptr< rtc::RTCCertificate > > certificates
 
int ice_candidate_pool_size = 0
 
bool disable_ipv6 = false
 
bool enable_rtp_data_channel = false
 
rtc::Optional< int > screencast_min_bitrate
 
rtc::Optional< bool > combined_audio_video_bwe
 
rtc::Optional< bool > enable_dtls_srtp
 
TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled
 
CandidateNetworkPolicy candidate_network_policy
 
int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets
 
bool audio_jitter_buffer_fast_accelerate = false
 
int ice_connection_receiving_timeout = kUndefined
 
int ice_backup_candidate_pair_ping_interval = kUndefined
 
ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE
 
bool prioritize_most_likely_ice_candidate_pairs = false
 
struct cricket::MediaConfig media_config
 
bool enable_quic = false
 
bool prune_turn_ports = false
 
bool presume_writable_when_fully_relayed = false
 
bool enable_ice_renomination = false
 
bool redetermine_role_on_ice_restart = true
 
rtc::Optional< int > ice_check_min_interval
 

Static Public Attributes

static const int kUndefined = -1
 
static const int kAudioJitterBufferMaxPackets = 50
 
static const int kAggressiveIceConnectionReceivingTimeout = 1000
 

Constructor & Destructor Documentation

◆ RTCConfiguration() [1/4]

webrtc::PeerConnectionInterface::RTCConfiguration::RTCConfiguration ( )
default

◆ RTCConfiguration() [2/4]

webrtc::PeerConnectionInterface::RTCConfiguration::RTCConfiguration ( RTCConfigurationType  type)
inlineexplicit

◆ RTCConfiguration() [3/4]

webrtc::PeerConnectionInterface::RTCConfiguration::RTCConfiguration ( )
default

◆ RTCConfiguration() [4/4]

webrtc::PeerConnectionInterface::RTCConfiguration::RTCConfiguration ( RTCConfigurationType  type)
inlineexplicit

Member Function Documentation

◆ cpu_adaptation() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::cpu_adaptation ( )
inline

◆ cpu_adaptation() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::cpu_adaptation ( )
inline

◆ dscp() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::dscp ( )
inline

◆ dscp() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::dscp ( )
inline

◆ operator!=() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::operator!= ( const RTCConfiguration &  o) const

◆ operator!=() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::operator!= ( const RTCConfiguration &  o) const

◆ operator==() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::operator== ( const RTCConfiguration &  o) const

◆ operator==() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::operator== ( const RTCConfiguration &  o) const

◆ prerenderer_smoothing() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::prerenderer_smoothing ( )
inline

◆ prerenderer_smoothing() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::prerenderer_smoothing ( )
inline

◆ set_cpu_adaptation() [1/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_cpu_adaptation ( bool  enable)
inline

◆ set_cpu_adaptation() [2/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_cpu_adaptation ( bool  enable)
inline

◆ set_dscp() [1/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_dscp ( bool  enable)
inline

◆ set_dscp() [2/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_dscp ( bool  enable)
inline

◆ set_prerenderer_smoothing() [1/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_prerenderer_smoothing ( bool  enable)
inline

◆ set_prerenderer_smoothing() [2/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_prerenderer_smoothing ( bool  enable)
inline

◆ set_suspend_below_min_bitrate() [1/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_suspend_below_min_bitrate ( bool  enable)
inline

◆ set_suspend_below_min_bitrate() [2/2]

void webrtc::PeerConnectionInterface::RTCConfiguration::set_suspend_below_min_bitrate ( bool  enable)
inline

◆ suspend_below_min_bitrate() [1/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::suspend_below_min_bitrate ( )
inline

◆ suspend_below_min_bitrate() [2/2]

bool webrtc::PeerConnectionInterface::RTCConfiguration::suspend_below_min_bitrate ( )
inline

Member Data Documentation

◆ audio_jitter_buffer_fast_accelerate

bool webrtc::PeerConnectionInterface::RTCConfiguration::audio_jitter_buffer_fast_accelerate = false

◆ audio_jitter_buffer_max_packets

int webrtc::PeerConnectionInterface::RTCConfiguration::audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets

◆ bundle_policy

BundlePolicy webrtc::PeerConnectionInterface::RTCConfiguration::bundle_policy = kBundlePolicyBalanced

◆ candidate_network_policy

CandidateNetworkPolicy webrtc::PeerConnectionInterface::RTCConfiguration::candidate_network_policy
Initial value:
=
kCandidateNetworkPolicyAll

◆ certificates

std::vector< rtc::scoped_refptr< rtc::RTCCertificate > > webrtc::PeerConnectionInterface::RTCConfiguration::certificates

◆ combined_audio_video_bwe

rtc::Optional< bool > webrtc::PeerConnectionInterface::RTCConfiguration::combined_audio_video_bwe

◆ continual_gathering_policy

ContinualGatheringPolicy webrtc::PeerConnectionInterface::RTCConfiguration::continual_gathering_policy = GATHER_ONCE

◆ disable_ipv6

bool webrtc::PeerConnectionInterface::RTCConfiguration::disable_ipv6 = false

◆ enable_dtls_srtp

rtc::Optional< bool > webrtc::PeerConnectionInterface::RTCConfiguration::enable_dtls_srtp

◆ enable_ice_renomination

bool webrtc::PeerConnectionInterface::RTCConfiguration::enable_ice_renomination = false

◆ enable_quic

bool webrtc::PeerConnectionInterface::RTCConfiguration::enable_quic = false

◆ enable_rtp_data_channel

bool webrtc::PeerConnectionInterface::RTCConfiguration::enable_rtp_data_channel = false

◆ ice_backup_candidate_pair_ping_interval

int webrtc::PeerConnectionInterface::RTCConfiguration::ice_backup_candidate_pair_ping_interval = kUndefined

◆ ice_candidate_pool_size

int webrtc::PeerConnectionInterface::RTCConfiguration::ice_candidate_pool_size = 0

◆ ice_check_min_interval

rtc::Optional< int > webrtc::PeerConnectionInterface::RTCConfiguration::ice_check_min_interval

◆ ice_connection_receiving_timeout

int webrtc::PeerConnectionInterface::RTCConfiguration::ice_connection_receiving_timeout = kUndefined

◆ kAggressiveIceConnectionReceivingTimeout

static const int webrtc::PeerConnectionInterface::RTCConfiguration::kAggressiveIceConnectionReceivingTimeout = 1000
static

◆ kAudioJitterBufferMaxPackets

static const int webrtc::PeerConnectionInterface::RTCConfiguration::kAudioJitterBufferMaxPackets = 50
static

◆ kUndefined

static const int webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined = -1
static

◆ media_config

struct cricket::MediaConfig webrtc::PeerConnectionInterface::RTCConfiguration::media_config

◆ presume_writable_when_fully_relayed

bool webrtc::PeerConnectionInterface::RTCConfiguration::presume_writable_when_fully_relayed = false

◆ prioritize_most_likely_ice_candidate_pairs

bool webrtc::PeerConnectionInterface::RTCConfiguration::prioritize_most_likely_ice_candidate_pairs = false

◆ prune_turn_ports

bool webrtc::PeerConnectionInterface::RTCConfiguration::prune_turn_ports = false

◆ redetermine_role_on_ice_restart

bool webrtc::PeerConnectionInterface::RTCConfiguration::redetermine_role_on_ice_restart = true

◆ rtcp_mux_policy

RtcpMuxPolicy webrtc::PeerConnectionInterface::RTCConfiguration::rtcp_mux_policy = kRtcpMuxPolicyRequire

◆ screencast_min_bitrate

rtc::Optional< int > webrtc::PeerConnectionInterface::RTCConfiguration::screencast_min_bitrate

◆ servers

IceServers webrtc::PeerConnectionInterface::RTCConfiguration::servers

◆ tcp_candidate_policy

TcpCandidatePolicy webrtc::PeerConnectionInterface::RTCConfiguration::tcp_candidate_policy = kTcpCandidatePolicyEnabled

◆ type

IceTransportsType webrtc::PeerConnectionInterface::RTCConfiguration::type = kAll
The documentation for this struct was generated from the following files:
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