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TRTCStatistics

Copyright (c) 2021 Tencent. All rights reserved.
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Module: TRTC audio/video metrics (read-only)
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Function: the TRTC SDK reports to you the current real-time audio/video metrics (frame rate, bitrate, lag, etc.) once every two seconds
TRTCStatistics
StructType
FuncList
DESC
TRTCLocalStatistics
Local audio/video metrics.
TRTCRemoteStatistics
Remote audio/video metrics.
TRTCStatistics
Network and performance metrics.
TRTCLocalStatistics
TRTCLocalStatistics
Local audio/video metrics.EnumType
DESC
audioBitrate
Field description: local audio bitrate in Kbps, i.e., how much audio data is generated per second
audioCaptureState
Field description:Audio equipment collection status(
0：Normal；1：Long silence detected；2：Broken sound detected；3：Abnormal intermittent sound detected;)
audioSampleRate
Field description: local audio sample rate (Hz)
frameRate
Field description: local video frame rate in fps, i.e., how many video frames there are per second
height
Field description: local video height in px
streamType
Field description: video stream type (HD big image | smooth small image | substream image)
videoBitrate
Field description: local video bitrate in Kbps, i.e., how much video data is generated per second
width
Field description: local video width in px
TRTCRemoteStatistics
TRTCRemoteStatistics
Remote audio/video metrics.EnumType
DESC
audioBitrate
Field description: local audio bitrate (Kbps)
audioBlockRate
Field description: audio playback lag rate (%)
Audio playback lag rate (audioBlockRate) = cumulative audio playback lag duration (audioTotalBlockTime)/total audio playback duration
audioPacketLoss
Field description: total packet loss rate (%) of the audio stream
 audioPacketLoss  represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".
The smaller the  audioPacketLoss , the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience.
If  downLoss  is  0  but  audioPacketLoss  isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
audioSampleRate
Field description: local audio sample rate (Hz)
audioTotalBlockTime
Field description: cumulative audio playback lag duration (ms)
finalLoss
Field description: total packet loss rate (%) of the audio/video stream
Deprecated, please use audioPacketLoss and videoPacketLoss instead.
frameRate
Field description: remote video frame rate (fps)
height
Field description: remote video height in px
jitterBufferDelay
Field description: playback delay (ms)
In order to avoid audio/video lags caused by network jitters and network packet disorders, TRTC maintains a playback buffer on the playback side to organize the received network data packets.
The size of the buffer is adaptively adjusted according to the current network quality and converted to the length of time in milliseconds, i.e.,  jitterBufferDelay .
point2PointDelay
Field description: end-to-end delay (ms)
 point2PointDelay  represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback".
 point2PointDelay  works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless.
remoteNetworkRTT
Field description: round-trip delay (ms) from the SDK to cloud
This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".
The smaller the value, the better. If  remoteNetworkRTT  is below 50 ms, it means a short audio/video call delay; if  remoteNetworkRTT  is above 200 ms, it means a long audio/video call delay.
It should be explained that  remoteNetworkRTT  represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between  remoteNetworkUpRTT  and  remoteNetworkDownRTT .
remoteNetworkUplinkLoss
Field description: upstream packet loss rate (%) from the SDK to cloud
The smaller the value, the better. If  remoteNetworkUplinkLoss  is  0% , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.
If  remoteNetworkUplinkLoss  is  30% , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.
streamType
Field description: video stream type (HD big image | smooth small image | substream image)
userId
Field description: user ID
videoBitrate
Field description: remote video bitrate (Kbps)
videoBlockRate
Field description: video playback lag rate (%)
Video playback lag rate (videoBlockRate) = cumulative video playback lag duration (videoTotalBlockTime)/total video playback duration
videoPacketLoss
Field description: total packet loss rate (%) of the video stream
 videoPacketLoss  represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".
The smaller the  videoPacketLoss , the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience.
If  downLoss  is  0  but  videoPacketLoss  isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
videoTotalBlockTime
Field description: cumulative video playback lag duration (ms)
width
Field description: remote video width in px
TRTCStatistics
TRTCStatistics
Network and performance metrics.EnumType
DESC
appCpu
Field description: CPU utilization (%) of the current application, Android 8.0 and above systems are not supported
downLoss
Field description: downstream packet loss rate (%) from cloud to the SDK
The smaller the value, the better. If  downLoss  is  0% , the downstream network quality is very good, and the data packets received from the cloud are basically not lost.
If  downLoss  is  30% , 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage.
gatewayRtt
Field description: round-trip delay (ms) from the SDK to gateway
This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> gateway -> SDK".
The smaller the value, the better. If  gatewayRtt  is below 50 ms, it means a short audio/video call delay; if  gatewayRtt  is above 200 ms, it means a long audio/video call delay.
It should be explained that  gatewayRtt  is invalid for cellular network.
localStatistics
Field description: local audio/video statistics
As there may be three local audio/video streams (i.e., HD big image, smooth small image, and substream image), the local audio/video statistics are an array.
receivedBytes
Field description: total number of received bytes (including signaling data and audio/video data)
remoteStatistics
Field description: remote audio/video statistics
As there may be multiple concurrent remote users, and each of them may have multiple concurrent audio/video streams (i.e., HD big image, smooth small image, and substream image), the remote audio/video statistics are an array.
rtt
Field description: round-trip delay (ms) from the SDK to cloud
This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".
The smaller the value, the better. If  rtt  is below 50 ms, it means a short audio/video call delay; if  rtt  is above 200 ms, it means a long audio/video call delay.
It should be explained that  rtt  represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between  upRtt  and  downRtt .
sentBytes
Field description: total number of sent bytes (including signaling data and audio/video data)
systemCpu
Field description: CPU utilization (%) of the current system, Android 8.0 and above systems are not supported
upLoss
Field description: upstream packet loss rate (%) from the SDK to cloud
The smaller the value, the better. If  upLoss  is  0% , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.
If  upLoss  is  30% , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.

FuncList	DESC
TRTCLocalStatistics	Local audio/video metrics.
TRTCRemoteStatistics	Remote audio/video metrics.
TRTCStatistics	Network and performance metrics.

EnumType	DESC
audioBitrate	Field description: local audio bitrate in Kbps, i.e., how much audio data is generated per second
audioCaptureState	Field description:Audio equipment collection status( 0：Normal；1：Long silence detected；2：Broken sound detected；3：Abnormal intermittent sound detected;)
audioSampleRate	Field description: local audio sample rate (Hz)
frameRate	Field description: local video frame rate in fps, i.e., how many video frames there are per second
height	Field description: local video height in px
streamType	Field description: video stream type (HD big image \| smooth small image \| substream image)
videoBitrate	Field description: local video bitrate in Kbps, i.e., how much video data is generated per second
width	Field description: local video width in px

EnumType	DESC
appCpu	Field description: CPU utilization (%) of the current application, Android 8.0 and above systems are not supported
downLoss	Field description: downstream packet loss rate (%) from cloud to the SDK The smaller the value, the better. If `downLoss` is `0%` , the downstream network quality is very good, and the data packets received from the cloud are basically not lost. If `downLoss` is `30%` , 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage.
gatewayRtt	Field description: round-trip delay (ms) from the SDK to gateway This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> gateway -> SDK". The smaller the value, the better. If `gatewayRtt` is below 50 ms, it means a short audio/video call delay; if `gatewayRtt` is above 200 ms, it means a long audio/video call delay. It should be explained that `gatewayRtt` is invalid for cellular network.
localStatistics	Field description: local audio/video statistics As there may be three local audio/video streams (i.e., HD big image, smooth small image, and substream image), the local audio/video statistics are an array.
receivedBytes	Field description: total number of received bytes (including signaling data and audio/video data)
remoteStatistics	Field description: remote audio/video statistics As there may be multiple concurrent remote users, and each of them may have multiple concurrent audio/video streams (i.e., HD big image, smooth small image, and substream image), the remote audio/video statistics are an array.
rtt	Field description: round-trip delay (ms) from the SDK to cloud This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK". The smaller the value, the better. If `rtt` is below 50 ms, it means a short audio/video call delay; if `rtt` is above 200 ms, it means a long audio/video call delay. It should be explained that `rtt` represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between `upRtt` and `downRtt` .
sentBytes	Field description: total number of sent bytes (including signaling data and audio/video data)
systemCpu	Field description: CPU utilization (%) of the current system, Android 8.0 and above systems are not supported
upLoss	Field description: upstream packet loss rate (%) from the SDK to cloud The smaller the value, the better. If `upLoss` is `0%` , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost. If `upLoss` is `30%` , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.