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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,196 @@ | ||
| // Copyright (c) 2022-present Mattermost, Inc. All Rights Reserved. | ||
| // See LICENSE.txt for license information. | ||
|
|
||
| package client | ||
|
|
||
| import ( | ||
| "log/slog" | ||
| "time" | ||
|
|
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| "github.com/pion/interceptor/pkg/stats" | ||
| "github.com/pion/webrtc/v3" | ||
| ) | ||
|
|
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| type rtcMonitor struct { | ||
| log *slog.Logger | ||
| pc *webrtc.PeerConnection | ||
| statsGetter stats.Getter | ||
| interval time.Duration | ||
|
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| lastSndStats map[webrtc.SSRC]*stats.Stats | ||
| lastRcvStats map[webrtc.SSRC]*stats.Stats | ||
|
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| statsCh chan rtcStats | ||
| stopCh chan struct{} | ||
| doneCh chan struct{} | ||
| } | ||
|
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| type rtcStats struct { | ||
| lossRate float64 | ||
| jitter float64 | ||
| } | ||
|
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| func newRTCMonitor(log *slog.Logger, pc *webrtc.PeerConnection, sg stats.Getter, intv time.Duration) *rtcMonitor { | ||
| return &rtcMonitor{ | ||
| log: log, | ||
| pc: pc, | ||
| statsGetter: sg, | ||
| interval: intv, | ||
| statsCh: make(chan rtcStats, 1), | ||
| stopCh: make(chan struct{}), | ||
| doneCh: make(chan struct{}), | ||
| lastSndStats: make(map[webrtc.SSRC]*stats.Stats), | ||
| lastRcvStats: make(map[webrtc.SSRC]*stats.Stats), | ||
| } | ||
| } | ||
|
|
||
| func (m *rtcMonitor) gatherStats() (map[webrtc.SSRC]*stats.Stats, map[webrtc.SSRC]*stats.Stats) { | ||
| sndStats := make(map[webrtc.SSRC]*stats.Stats) | ||
| for _, snd := range m.pc.GetSenders() { | ||
| if snd == nil { | ||
| continue | ||
| } | ||
| for i, enc := range snd.GetParameters().Encodings { | ||
| // For simplicity we only consider audio streams. | ||
| // This lets us more easily make assumptions on the clock rate. | ||
| if snd.GetParameters().Codecs[i].MimeType != webrtc.MimeTypeOpus { | ||
| continue | ||
| } | ||
|
|
||
| stats := m.statsGetter.Get(uint32(enc.SSRC)) | ||
| if stats != nil { | ||
| sndStats[enc.SSRC] = stats | ||
| } | ||
| } | ||
| } | ||
|
|
||
| rcvStats := make(map[webrtc.SSRC]*stats.Stats) | ||
| for _, rcv := range m.pc.GetReceivers() { | ||
| if rcv == nil { | ||
| continue | ||
| } | ||
|
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| track := rcv.Track() | ||
|
|
||
| if track == nil { | ||
| continue | ||
| } | ||
|
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| // For simplicity we only consider audio streams. | ||
| // This lets us more easily make assumptions on the clock rate. | ||
| if track.Codec().MimeType != webrtc.MimeTypeOpus { | ||
| continue | ||
| } | ||
|
|
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| stats := m.statsGetter.Get(uint32(track.SSRC())) | ||
| if stats != nil { | ||
| rcvStats[track.SSRC()] = stats | ||
| } | ||
| } | ||
|
|
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| return sndStats, rcvStats | ||
| } | ||
|
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| func (m *rtcMonitor) getAvgSenderStats(stats map[webrtc.SSRC]*stats.Stats) (avgLossRate, avgJitter, statsCount float64) { | ||
| var totalJitter, totalLossRate float64 | ||
|
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| for ssrc, s := range stats { | ||
| if prevStats := m.lastSndStats[ssrc]; prevStats == nil || s.OutboundRTPStreamStats.PacketsSent == prevStats.OutboundRTPStreamStats.PacketsSent { | ||
| continue | ||
| } | ||
|
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| totalLossRate += s.RemoteInboundRTPStreamStats.FractionLost | ||
| totalJitter += s.RemoteInboundRTPStreamStats.Jitter | ||
| statsCount++ | ||
| } | ||
|
|
||
| if statsCount > 0 { | ||
| avgJitter = (totalJitter / statsCount) * 1000 | ||
| avgLossRate = totalLossRate / statsCount | ||
| } | ||
|
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| return | ||
| } | ||
|
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| func (m *rtcMonitor) getAvgReceiverStats(stats map[webrtc.SSRC]*stats.Stats) (avgLossRate, avgJitter, statsCount float64) { | ||
| var totalJitter, totalLost, totalReceived float64 | ||
|
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| for ssrc, s := range stats { | ||
| prevStats := m.lastRcvStats[ssrc] | ||
| if prevStats == nil || s.InboundRTPStreamStats.PacketsReceived == prevStats.InboundRTPStreamStats.PacketsReceived { | ||
| continue | ||
| } | ||
|
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||
| receivedDiff := s.InboundRTPStreamStats.PacketsReceived - prevStats.InboundRTPStreamStats.PacketsReceived | ||
| potentiallyLost := int64(s.RemoteOutboundRTPStreamStats.PacketsSent) - int64(s.InboundRTPStreamStats.PacketsReceived) | ||
| prevPotentiallyLost := int64(prevStats.RemoteOutboundRTPStreamStats.PacketsSent) - int64(prevStats.InboundRTPStreamStats.PacketsReceived) | ||
| var lostDiff int64 | ||
| if prevPotentiallyLost >= 0 && potentiallyLost > prevPotentiallyLost { | ||
| lostDiff = potentiallyLost - prevPotentiallyLost | ||
| } | ||
| totalLost += float64(lostDiff) | ||
| totalReceived += float64(receivedDiff) | ||
|
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||
| // pion stores inbound jitter in RTP units rather than seconds. | ||
| // 960 is the expected frame size for opus packets. (20ms at 48000Hz) | ||
| totalJitter += s.InboundRTPStreamStats.Jitter / 960 | ||
|
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| statsCount++ | ||
| } | ||
|
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| if statsCount > 0 { | ||
| avgJitter = (totalJitter / statsCount) | ||
| avgLossRate = totalLost / totalReceived | ||
| } | ||
|
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| return | ||
| } | ||
|
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| func (m *rtcMonitor) processStats(sndStats, rcvStats map[webrtc.SSRC]*stats.Stats) { | ||
| defer func() { | ||
| // cache stats for the next iteration | ||
| m.lastSndStats = sndStats | ||
| m.lastRcvStats = rcvStats | ||
| }() | ||
|
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| sndLossRate, sndJitter, sndCnt := m.getAvgSenderStats(sndStats) | ||
| rcvLossRate, rcvJitter, rcvCnt := m.getAvgReceiverStats(rcvStats) | ||
|
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| // nothing to do if we didn't process any stats | ||
| if sndCnt == 0 && rcvCnt == 0 { | ||
| return | ||
| } | ||
|
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| select { | ||
| case m.statsCh <- rtcStats{lossRate: max(sndLossRate, rcvLossRate), jitter: max(sndJitter, rcvJitter)}: | ||
| default: | ||
| m.log.Error("failed to send stats: channel is full") | ||
| } | ||
| } | ||
|
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| func (m *rtcMonitor) Start() { | ||
| go func() { | ||
| defer close(m.doneCh) | ||
| ticker := time.NewTicker(m.interval) | ||
|
|
||
| for { | ||
| select { | ||
| case <-ticker.C: | ||
| sndStats, rcvStats := m.gatherStats() | ||
| m.processStats(sndStats, rcvStats) | ||
| case <-m.stopCh: | ||
| return | ||
| } | ||
| } | ||
| }() | ||
| } | ||
|
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| func (m *rtcMonitor) Stop() { | ||
| close(m.stopCh) | ||
| <-m.doneCh | ||
| close(m.statsCh) | ||
| } | ||
|
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| func (m *rtcMonitor) StatsCh() <-chan rtcStats { | ||
| return m.statsCh | ||
| } |
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