Rename mono to rtptime.

group.go

@@ -16,7 +16,7 @@ import (
 	"sync/atomic"
 	"time"
 
-	"sfu/mono"
+	"sfu/rtptime"
 
 	"github.com/pion/webrtc/v2"
 )
@@ -587,12 +587,13 @@ func getClientStats(c *webClient) clientStats {
 	for _, down := range c.down {
 		conns := connStats{id: down.id}
 		for _, t := range down.tracks {
-			loss, jitter := t.stats.Get(mono.Microseconds())
+			us := rtptime.Microseconds()
+			loss, jitter := t.stats.Get(us)
 			j := time.Duration(jitter) * time.Second /
 				time.Duration(t.track.Codec().ClockRate)
 			conns.tracks = append(conns.tracks, trackStats{
 				bitrate:    uint64(t.rate.Estimate()) * 8,
-				maxBitrate: t.GetMaxBitrate(mono.Microseconds()),
+				maxBitrate: t.GetMaxBitrate(us),
 				loss:       uint8(uint32(loss) * 100 / 256),
 				jitter:     j,
 			})

jitter/jitter.go

@@ -3,13 +3,13 @@ package jitter
 import (
 	"sync/atomic"
 
-	"sfu/mono"
+	"sfu/rtptime"
 )
 
 type Estimator struct {
 	hz        uint32
 	timestamp uint32
-	time uint32
+	time      uint32
 
 	jitter uint32 // atomic
 }
@@ -25,11 +25,11 @@ func (e *Estimator) accumulate(timestamp, now uint32) {
 	}
 
 	d := uint32((e.time - now) - (e.timestamp - timestamp))
-	if d & 0x80000000 != 0 {
+	if d&0x80000000 != 0 {
 		d = uint32(-int32(d))
 	}
 	oldjitter := atomic.LoadUint32(&e.jitter)
-	jitter := (oldjitter * 15 + d) / 16
+	jitter := (oldjitter*15 + d) / 16
 	atomic.StoreUint32(&e.jitter, jitter)
 
 	e.timestamp = timestamp
@@ -37,7 +37,7 @@ func (e *Estimator) accumulate(timestamp, now uint32) {
 }
 
 func (e *Estimator) Accumulate(timestamp uint32) {
-	e.accumulate(timestamp, uint32(mono.Now(e.hz)))
+	e.accumulate(timestamp, uint32(rtptime.Now(e.hz)))
 }
 
 func (e *Estimator) Jitter() uint32 {

rtptime/rtptime.go

@@ -1,4 +1,4 @@
-package mono
+package rtptime
 
 import (
 	"time"

rtptime/rtptime_test.go

@@ -1,4 +1,4 @@
-package mono
+package rtptime
 
 import (
 	"testing"

webclient.go

@@ -20,7 +20,7 @@ import (
 
 	"sfu/estimator"
 	"sfu/jitter"
-	"sfu/mono"
+	"sfu/rtptime"
 	"sfu/packetcache"
 
 	"github.com/gorilla/websocket"
@@ -559,7 +559,7 @@ func rtcpUpListener(conn *upConnection, track *upTrack, r *webrtc.RTPReceiver) {
 			switch p := p.(type) {
 			case *rtcp.SenderReport:
 				track.mu.Lock()
-				track.srTime = mono.Now(0x10000)
+				track.srTime = rtptime.Now(0x10000)
 				track.srNTPTime = p.NTPTime
 				track.srRTPTime = p.RTPTime
 				track.mu.Unlock()
@@ -574,7 +574,7 @@ func sendRR(conn *upConnection) error {
 		return nil
 	}
 
-	now := mono.Now(0x10000)
+	now := rtptime.Now(0x10000)
 
 	reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
 	for _, t := range conn.tracks {
@@ -631,7 +631,7 @@ func sendSR(conn *rtpDownConnection) error {
 	packets := make([]rtcp.Packet, 0, len(conn.tracks))
 
 	now := time.Now()
-	nowNTP := mono.TimeToNTP(now)
+	nowNTP := rtptime.TimeToNTP(now)
 
 	for _, t := range conn.tracks {
 		clockrate := t.track.Codec().ClockRate
@@ -643,10 +643,10 @@ func sendSR(conn *rtpDownConnection) error {
 
 		nowRTP := srRTPTime
 		if srNTPTime != 0 {
-			srTime := mono.NTPToTime(srNTPTime)
+			srTime := rtptime.NTPToTime(srNTPTime)
 			delay := now.Sub(srTime)
 			if delay > 0 && delay < time.Hour {
-				d := mono.FromDuration(delay, clockrate)
+				d := rtptime.FromDuration(delay, clockrate)
 				nowRTP = srRTPTime + uint32(d)
 			}
 		}
@@ -918,7 +918,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
 				track.maxREMBBitrate.Set(
-					p.Bitrate, mono.Microseconds(),
+					p.Bitrate, rtptime.Microseconds(),
 				)
 			case *rtcp.ReceiverReport:
 				for _, r := range p.Reports {
@@ -934,7 +934,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 				}
 			case *rtcp.TransportLayerNack:
 				maxBitrate := track.GetMaxBitrate(
-					mono.Microseconds(),
+					rtptime.Microseconds(),
 				)
 				bitrate := track.rate.Estimate()
 				if uint64(bitrate)*7/8 < maxBitrate {
@@ -946,7 +946,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 }
 
 func handleReport(track *rtpDownTrack, report rtcp.ReceptionReport) {
-	now := mono.Microseconds()
+	now := rtptime.Microseconds()
 	track.stats.Set(report.FractionLost, report.Jitter, now)
 	track.updateRate(report.FractionLost, now)
 }
@@ -972,7 +972,7 @@ func trackKinds(down *rtpDownConnection) (audio bool, video bool) {
 }
 
 func updateUpBitrate(up *upConnection, maxVideoRate uint64) {
-	now := mono.Microseconds()
+	now := rtptime.Microseconds()
 
 	for _, track := range up.tracks {
 		isvideo := track.track.Kind() == webrtc.RTPCodecTypeVideo
@@ -1011,7 +1011,7 @@ func (up *upConnection) sendPLI(track *upTrack) error {
 		return ErrUnsupportedFeedback
 	}
 	last := atomic.LoadUint64(&track.lastPLI)
-	now := mono.Microseconds()
+	now := rtptime.Microseconds()
 	if now >= last && now-last < 200000 {
 		return ErrRateLimited
 	}
@@ -1039,7 +1039,7 @@ func (up *upConnection) sendFIR(track *upTrack, increment bool) error {
 		return ErrUnsupportedFeedback
 	}
 	last := atomic.LoadUint64(&track.lastFIR)
-	now := mono.Microseconds()
+	now := rtptime.Microseconds()
 	if now >= last && now-last < 200000 {
 		return ErrRateLimited
 	}