Interface RemoteAudioTrackStats

Statistics of the remote audio track, such as connection and transmission statistics, which can be retrieved by calling AgoraRTCClient.getRemoteAudioStats.

Hierarchy

  • RemoteAudioTrackStats

Index

Properties

Optional codecType

codecType: "opus" | "aac"

The audio codec.

  • opus: The audio codec is OPUS。
  • aac: The audio codec is AAC。

    Firefox does not support this property.

end2EndDelay

end2EndDelay: number

End-to-end delay (ms).

The delay (ms) between a remote client sampling the audio and the local client playing the audio.

freezeRate

freezeRate: number

The freeze rate of the received audio.

packetLossRate

packetLossRate: number

The packet loss rate of the received audio.

publishDuration

publishDuration: number

receiveBitrate

receiveBitrate: number

The bitrate (bps) of the received audio.

receiveBytes

receiveBytes: number

The total bytes of the received audio.

receiveDelay

receiveDelay: number

The delay (ms) between a remote client sending the audio and the local client playing the audio.

receiveLevel

receiveLevel: number

The energy level of the received audio.

The value range is [0,32767].

This value is retrieved by calling WebRTC-Stats and may not be up-to-date. To get the real-time sound volume, call RemoteAudioTrack.getVolumeLevel.

receivePackets

receivePackets: number

The total packets of the received audio.

receivePacketsLost

receivePacketsLost: number

The total number of lost audio packets that should be received.

totalDuration

totalDuration: number

The total duration of the received audio in seconds.

totalFreezeTime

totalFreezeTime: number

The total freeze time of the received audio in seconds.

transportDelay

transportDelay: number

Transmission delay (ms).

The delay (ms) between a remote client sending the audio and the local client receiving the audio.