#include <audio_send_stream.h>

Public Member Functions
	Stats ()

	~Stats ()

	Stats ()

	~Stats ()

Public Attributes
uint32_t	local_ssrc = 0

int64_t	bytes_sent = 0

int32_t	packets_sent = 0

int32_t	packets_lost = -1

float	fraction_lost = -1.0f

std::string	codec_name

rtc::Optional< int >	codec_payload_type

int32_t	ext_seqnum = -1

int32_t	jitter_ms = -1

int64_t	rtt_ms = -1

int32_t	audio_level = -1

float	aec_quality_min = -1.0f

int32_t	echo_delay_median_ms = -1

int32_t	echo_delay_std_ms = -1

int32_t	echo_return_loss = -100

int32_t	echo_return_loss_enhancement = -100

float	residual_echo_likelihood = -1.0f

float	residual_echo_likelihood_recent_max = -1.0f

bool	typing_noise_detected = false

Constructor & Destructor Documentation

◆ Stats() [1/2]

webrtc::AudioSendStream::Stats::Stats ( )

default

◆ ~Stats() [1/2]

webrtc::AudioSendStream::Stats::~Stats ( )

default

◆ Stats() [2/2]

webrtc::AudioSendStream::Stats::Stats ( )

◆ ~Stats() [2/2]

webrtc::AudioSendStream::Stats::~Stats ( )

Member Data Documentation

◆ aec_quality_min

float webrtc::AudioSendStream::Stats::aec_quality_min = -1.0f

◆ audio_level

int32_t webrtc::AudioSendStream::Stats::audio_level = -1

◆ bytes_sent

int64_t webrtc::AudioSendStream::Stats::bytes_sent = 0

◆ codec_name

std::string webrtc::AudioSendStream::Stats::codec_name

◆ codec_payload_type

rtc::Optional< int > webrtc::AudioSendStream::Stats::codec_payload_type

◆ echo_delay_median_ms

int32_t webrtc::AudioSendStream::Stats::echo_delay_median_ms = -1

◆ echo_delay_std_ms

int32_t webrtc::AudioSendStream::Stats::echo_delay_std_ms = -1

◆ echo_return_loss

int32_t webrtc::AudioSendStream::Stats::echo_return_loss = -100

◆ echo_return_loss_enhancement

int32_t webrtc::AudioSendStream::Stats::echo_return_loss_enhancement = -100

◆ ext_seqnum

int32_t webrtc::AudioSendStream::Stats::ext_seqnum = -1

◆ fraction_lost

float webrtc::AudioSendStream::Stats::fraction_lost = -1.0f

◆ jitter_ms

int32_t webrtc::AudioSendStream::Stats::jitter_ms = -1

◆ local_ssrc

uint32_t webrtc::AudioSendStream::Stats::local_ssrc = 0

◆ packets_lost

int32_t webrtc::AudioSendStream::Stats::packets_lost = -1

◆ packets_sent

int32_t webrtc::AudioSendStream::Stats::packets_sent = 0

◆ residual_echo_likelihood

float webrtc::AudioSendStream::Stats::residual_echo_likelihood = -1.0f

◆ residual_echo_likelihood_recent_max

float webrtc::AudioSendStream::Stats::residual_echo_likelihood_recent_max = -1.0f

◆ rtt_ms

int64_t webrtc::AudioSendStream::Stats::rtt_ms = -1

◆ typing_noise_detected

bool webrtc::AudioSendStream::Stats::typing_noise_detected = false

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

Public Member Functions

Public Attributes