/**
 * Sample RTC statistics. See [[Call.sampleEvent]]
 */
export default interface RTCSample {
  [key: string]: any;

  /**
   * Audio input level in last second. Between 0 and 32767, representing -100 to -30 dB.
   */
  audioInputLevel: number;

  /**
   * Audio output level in last second. Between 0 and 32767, representing -100 to -30 dB.
   */
  audioOutputLevel: number;

  /**
   * Bytes received in last second.
   */
  bytesReceived: number;

  /**
   * Bytes sent in last second.
   */
  bytesSent: number;

  /**
   * Audio codec used, either pcmu or opus
   */
  codecName: string;

  /**
   * Packets delay variation
   */
  jitter: number;

  /**
   * Mean opinion score, 1.0 through roughly 4.5
   */
  mos: number | null;

  /**
   * Number of packets lost in last second.
   */
  packetsLost: number;

  /**
   * Packets lost to inbound packets ratio in last second.
   */
  packetsLostFraction: number;

  /**
   * Number of packets received in last second.
   */
  packetsReceived: number;

  /**
   * Number of packets sent in last second.
   */
  packetsSent: number;

  /**
   * Round trip time, to the server back to the client.
   */
  rtt: number;

  /**
   * Timestamp
   */
  timestamp: number;

  /**
   * Totals for packets and bytes related information
   */
  totals: RTCSampleTotals;
}

/**
 * Totals included in RTC statistics samples
 */
export interface RTCSampleTotals {
  /**
   * Total bytes received.
   */
  bytesReceived: number;

  /**
   * Total bytes sent.
   */
  bytesSent: number;

  /**
   * Total packets lost.
   */
  packetsLost: number;

  /**
   * Total packets lost to total inbound packets ratio
   */
  packetsLostFraction: number;

  /**
   * Total packets received.
   */
  packetsReceived: number;

  /**
   * Total packets sent.
   */
  packetsSent: number;
}