Webrtc音频发送延迟

一、计算公式

单个数据包发送延迟 = 音频数据包发送时间 - 采集时间

(1)、通过统计自己计算

总累计发送延时 = sum(音频数据包发送时间 - 采集时间)

发送延迟 = (两次统计间隔 总累计延迟差) / （两次统计间隔 发送数据包数量）

(2)、求得一定时间的内的平均发送延迟

// 会计算10s内的平均延迟、和最大延迟
DEPRECATED_RtpSenderEgress::UpdateDelayStatistics
{
 // 音频的send_side_delay_observer_默认空
 if (!send_side_delay_observer_ || capture_time_ms <= 0)
    return;

  int avg_delay_ms = 0;
  int max_delay_ms = 0;
  uint64_t total_packet_send_delay_ms = 0;
  {
    MutexLock lock(&lock_);
    // Compute the max and average of the recent capture-to-send delays.
    // The time complexity of the current approach depends on the distribution
    // of the delay values. This could be done more efficiently.

    // Remove elements older than kSendSideDelayWindowMs.
    auto lower_bound =
        send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
    for (auto it = send_dela