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Tencent Real-Time Communication
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    TRTCStatistics

    Last updated: 2024-10-25 17:04:23
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      Copyright (c) 2021 Tencent. All rights reserved.
      
      Module: TRTC audio/video metrics (read-only)
      
      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
      Local audio/video metrics
      Remote audio/video metrics
      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.
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