抖動的概念：在接收端計算。拿音頻來說，接收連續兩包音頻的時間差(R2-R1)，減去這個2包rtp包的時間差--轉換成ms時間。對視頻來說，是一幀數據，而不是一包。?

WebRtcVoiceMediaChannel::GetStats --| AudioReceiveStream::GetStats--除以采樣頻率--| ChannelReceive::GetRTCPStatistics()--| StreamStatisticianImpl::GetStats()stats.jitter = jitter_q4_ >> 4;StreamStatisticianImpl::UpdateJitter{int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_;//R2-R1的時間uint32_t receive_diff_rtp = static_cast<uint32_t>((receive_diff_ms * packet.payload_type_frequency()) / 1000); //換算成rtp時間int32_t time_diff_samples =receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);time_diff_samples = std::abs(time_diff_samples); //rtp的時間戳 if (time_diff_samples < 450000) {// Note we calculate in Q4 to avoid using float.int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);}}

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