webrtc::AudioEncoderOpus::Config Struct Reference

#include <audio_encoder_opus.h>

Public Member Functions
	Config ()
	Config (const Config &)
	~Config ()
Config &	operator= (const Config &)
bool	IsOk () const
int	GetBitrateBps () const
rtc::Optional< int >	GetNewComplexity () const
	Config ()
	Config (const Config &)
	~Config ()
Config &	operator= (const Config &)
bool	IsOk () const
int	GetBitrateBps () const
rtc::Optional< int >	GetNewComplexity () const

Public Attributes
int	frame_size_ms = 20
size_t	num_channels = 1
int	payload_type = 120
ApplicationMode	application = kVoip
rtc::Optional< int >	bitrate_bps
bool	fec_enabled = false
int	max_playback_rate_hz = 48000
int	complexity = kDefaultComplexity
int	low_rate_complexity = kDefaultComplexity
int	complexity_threshold_bps = 12500
int	complexity_threshold_window_bps = 1500
bool	dtx_enabled = false
std::vector< int >	supported_frame_lengths_ms
const Clock *	clock = Clock::GetRealTimeClock()
int	uplink_bandwidth_update_interval_ms = 200

Constructor & Destructor Documentation

Config() [1/4]

webrtc::AudioEncoderOpus::Config::Config

(

)

Config() [2/4]

webrtc::AudioEncoderOpus::Config::Config ( const Config &  )

default

~Config() [1/2]

webrtc::AudioEncoderOpus::Config::~Config ( )

default

Config() [3/4]

webrtc::AudioEncoderOpus::Config::Config

(

)

Config() [4/4]

webrtc::AudioEncoderOpus::Config::Config

(

const Config &

)

~Config() [2/2]

webrtc::AudioEncoderOpus::Config::~Config

(

)

Member Function Documentation

GetBitrateBps() [1/2]

int webrtc::AudioEncoderOpus::Config::GetBitrateBps

(

)

const

GetBitrateBps() [2/2]

int webrtc::AudioEncoderOpus::Config::GetBitrateBps

(

)

const

GetNewComplexity() [1/2]

rtc::Optional< int > webrtc::AudioEncoderOpus::Config::GetNewComplexity

(

)

const

GetNewComplexity() [2/2]

rtc::Optional<int> webrtc::AudioEncoderOpus::Config::GetNewComplexity

(

)

const

IsOk() [1/2]

bool webrtc::AudioEncoderOpus::Config::IsOk

(

)

const

IsOk() [2/2]

bool webrtc::AudioEncoderOpus::Config::IsOk

(

)

const

operator=() [1/2]

auto webrtc::AudioEncoderOpus::Config::operator= ( const Config &  )

default

operator=() [2/2]

Config& webrtc::AudioEncoderOpus::Config::operator=

(

const Config &

)

Member Data Documentation

application

ApplicationMode webrtc::AudioEncoderOpus::Config::application = kVoip

bitrate_bps

rtc::Optional< int > webrtc::AudioEncoderOpus::Config::bitrate_bps

clock

const Clock * webrtc::AudioEncoderOpus::Config::clock = Clock::GetRealTimeClock()

complexity

int webrtc::AudioEncoderOpus::Config::complexity = kDefaultComplexity

complexity_threshold_bps

int webrtc::AudioEncoderOpus::Config::complexity_threshold_bps = 12500

complexity_threshold_window_bps

int webrtc::AudioEncoderOpus::Config::complexity_threshold_window_bps = 1500

dtx_enabled

bool webrtc::AudioEncoderOpus::Config::dtx_enabled = false

fec_enabled

bool webrtc::AudioEncoderOpus::Config::fec_enabled = false

frame_size_ms

int webrtc::AudioEncoderOpus::Config::frame_size_ms = 20

low_rate_complexity

int webrtc::AudioEncoderOpus::Config::low_rate_complexity = kDefaultComplexity

max_playback_rate_hz

int webrtc::AudioEncoderOpus::Config::max_playback_rate_hz = 48000

num_channels

size_t webrtc::AudioEncoderOpus::Config::num_channels = 1

payload_type

int webrtc::AudioEncoderOpus::Config::payload_type = 120

supported_frame_lengths_ms

std::vector< int > webrtc::AudioEncoderOpus::Config::supported_frame_lengths_ms

uplink_bandwidth_update_interval_ms

int webrtc::AudioEncoderOpus::Config::uplink_bandwidth_update_interval_ms = 200

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc