WebRTC的Jitter计算
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WebRTC的Jitter计算
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抖動(dòng)的概念:在接收端計(jì)算。拿音頻來(lái)說(shuō),接收連續(xù)兩包音頻的時(shí)間差(R2-R1),減去這個(gè)2包rtp包的時(shí)間差--轉(zhuǎn)換成ms時(shí)間。對(duì)視頻來(lái)說(shuō),是一幀數(shù)據(jù),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(shí)間uint32_t receive_diff_rtp = static_cast<uint32_t>((receive_diff_ms * packet.payload_type_frequency()) / 1000); //換算成rtp時(shí)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(shí)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);}}總結(jié)
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