Optional codecThe audio codec.
"opus": The audio codec is OPUS。"aac": The audio codec is AAC。"pcmu": Reserved for future use."pcma": Reserved for future use."g722": Reserved for future use.Firefox does not support this property.
The packet loss rate of the received audio.
End-to-end delay (ms).
The delay (ms) between a remote client sampling the audio and the local client playing the audio. This delay does not include the time spent in encoding at the remote client and the time spent in decoding at the local client.
The freeze rate of the received audio.
The packet loss rate of the received audio.
The bitrate (bps) of the received audio.
The total bytes of the received audio.
The delay (ms) between a remote client sending the audio and the local client playing the audio.
This property is inaccurate on Safari and Firefox.
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]IRemoteAudioTrack.getVolumeLevel.
The total packets of the received audio.
The number of packets discarded by the jitter buffer due to early or late arrival.
The total number of lost audio packets that should be received.
The total duration of the received audio in seconds.
The total freeze time of the received audio in seconds.
Transmission delay (ms).
The delay (ms) between a remote client sending the audio and the local client receiving the audio.
Statistics of the remote audio track, such as connection and transmission statistics, which can be retrieved by calling [AgoraRTCClient.getRemoteAudioStats]IAgoraRTCClient.getRemoteAudioStats.