getAudioStats method

Retrieves real-time audio statistics for the participant.

• This method returns metrics related to the participant’s audio stream, such as jitter, packet loss, bitrate, and round-trip time (RTT). Returns the most recent sample and starts fetching the next one, so call it periodically — from a timer, for example. The first call after the participant joins returns null.

Returns

• A List<dynamic>? containing audio stream statistics objects, or null if stats are unavailable. Each object may include:

  • jitter: Represents distortion in the audio stream.

  • bitrate: The bitrate at which the audio stream is being transmitted.

  • totalPackets: The total number of packets transmitted for the audio stream.

  • packetsLost: The total number of packets lost during transmission.

  • rtt: The round-trip time (RTT) in milliseconds, representing the time taken for the stream to travel from the server to the client.

  • codec: The codec used for encoding the audio stream.

  • network: The network type used to transmit the stream.

Notes

• Each value is the last sample taken, not an average over the room.

• If the reported rtt is consistently greater than 300 ms, it is recommended to use a server region closer to the participant for improved performance.

Example

// Fetch audio and video stats for a participant
final audioStats = widget.participant.getAudioStats();
final videoStats = widget.participant.getVideoStats();

if (audioStats != null && audioStats.isNotEmpty) {
  final stats = audioStats.first;
  print('Audio Bitrate: ${stats['bitrate']}');
  print('Packet Loss: ${stats['packetsLost']}');
  print('RTT: ${stats['rtt']} ms');
}