#include <audio_receive_stream.h>

Public Attributes
uint32_t	remote_ssrc = 0

int64_t	bytes_rcvd = 0

uint32_t	packets_rcvd = 0

uint32_t	packets_lost = 0

float	fraction_lost = 0.0f

std::string	codec_name

rtc::Optional< int >	codec_payload_type

uint32_t	ext_seqnum = 0

uint32_t	jitter_ms = 0

uint32_t	jitter_buffer_ms = 0

uint32_t	jitter_buffer_preferred_ms = 0

uint32_t	delay_estimate_ms = 0

int32_t	audio_level = -1

float	expand_rate = 0.0f

float	speech_expand_rate = 0.0f

float	secondary_decoded_rate = 0.0f

float	accelerate_rate = 0.0f

float	preemptive_expand_rate = 0.0f

int32_t	decoding_calls_to_silence_generator = 0

int32_t	decoding_calls_to_neteq = 0

int32_t	decoding_normal = 0

int32_t	decoding_plc = 0

int32_t	decoding_cng = 0

int32_t	decoding_plc_cng = 0

int32_t	decoding_muted_output = 0

int64_t	capture_start_ntp_time_ms = 0

Member Data Documentation

◆ accelerate_rate

float webrtc::AudioReceiveStream::Stats::accelerate_rate = 0.0f

◆ audio_level

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

◆ bytes_rcvd

int64_t webrtc::AudioReceiveStream::Stats::bytes_rcvd = 0

◆ capture_start_ntp_time_ms

int64_t webrtc::AudioReceiveStream::Stats::capture_start_ntp_time_ms = 0

◆ codec_name

std::string webrtc::AudioReceiveStream::Stats::codec_name

◆ codec_payload_type

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

◆ decoding_calls_to_neteq

int32_t webrtc::AudioReceiveStream::Stats::decoding_calls_to_neteq = 0

◆ decoding_calls_to_silence_generator

int32_t webrtc::AudioReceiveStream::Stats::decoding_calls_to_silence_generator = 0

◆ decoding_cng

int32_t webrtc::AudioReceiveStream::Stats::decoding_cng = 0

◆ decoding_muted_output

int32_t webrtc::AudioReceiveStream::Stats::decoding_muted_output = 0

◆ decoding_normal

int32_t webrtc::AudioReceiveStream::Stats::decoding_normal = 0

◆ decoding_plc

int32_t webrtc::AudioReceiveStream::Stats::decoding_plc = 0

◆ decoding_plc_cng

int32_t webrtc::AudioReceiveStream::Stats::decoding_plc_cng = 0

◆ delay_estimate_ms

uint32_t webrtc::AudioReceiveStream::Stats::delay_estimate_ms = 0

◆ expand_rate

float webrtc::AudioReceiveStream::Stats::expand_rate = 0.0f

◆ ext_seqnum

uint32_t webrtc::AudioReceiveStream::Stats::ext_seqnum = 0

◆ fraction_lost

float webrtc::AudioReceiveStream::Stats::fraction_lost = 0.0f

◆ jitter_buffer_ms

uint32_t webrtc::AudioReceiveStream::Stats::jitter_buffer_ms = 0

◆ jitter_buffer_preferred_ms

uint32_t webrtc::AudioReceiveStream::Stats::jitter_buffer_preferred_ms = 0

◆ jitter_ms

uint32_t webrtc::AudioReceiveStream::Stats::jitter_ms = 0

◆ packets_lost

uint32_t webrtc::AudioReceiveStream::Stats::packets_lost = 0

◆ packets_rcvd

uint32_t webrtc::AudioReceiveStream::Stats::packets_rcvd = 0

◆ preemptive_expand_rate

float webrtc::AudioReceiveStream::Stats::preemptive_expand_rate = 0.0f

◆ remote_ssrc

uint32_t webrtc::AudioReceiveStream::Stats::remote_ssrc = 0

◆ secondary_decoded_rate

float webrtc::AudioReceiveStream::Stats::secondary_decoded_rate = 0.0f

◆ speech_expand_rate

float webrtc::AudioReceiveStream::Stats::speech_expand_rate = 0.0f

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

DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/call/audio_receive_stream.h

Public Attributes