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client: add jitter to media channel and some "realistic" media channel simulation

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KokaKiwi 2025-08-09 14:37:22 +02:00
parent 0a7291e90f
commit ff738ff045
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GPG key ID: FD333F84686EFE78
1 changed files with 37 additions and 2 deletions
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39
client/lib/webrtc.go
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@ -398,13 +398,48 @@ func (c *WebRTCPeer) openMediaTrack() {
defer ticker.Stop()
for ; true; <-ticker.C {
bufSize := randomInt(1000, 2500)
// Add jitter to simulate "realistic" media patterns
jitterDelay := time.Duration(randomInt(0, 200)) * time.Millisecond
time.Sleep(jitterDelay)
// Vary packet sizes for specific frames types
var bufSize int
frameType := randomInt(1, 100)
switch {
case frameType <= 5: // I-frames: 5% chance, larger
bufSize = randomInt(8000, 15000)
case frameType <= 35: // P-frames: 30% chance, medium
bufSize = randomInt(2000, 5000)
default: // B-frames: 65% chance, smaller
bufSize = randomInt(500, 2000)
}
buf := make([]byte, bufSize)
err := videoTrack.WriteSample(media.Sample{Data: buf, Duration: time.Second})
// Add some timing variation
frameDuration := time.Duration(randomInt(900, 1100)) * time.Millisecond
err = videoTrack.WriteSample(media.Sample{Data: buf, Duration: frameDuration})
if err != nil {
log.Printf("webrtc.WriteSample ERROR: %s", err)
}
// Simulate some burst of smaller packets
if randomInt(1, 10) == 1 { // 10% chance
burstCount := randomInt(2, 5)
for i := 0; i < burstCount; i++ {
smallBuf := make([]byte, randomInt(100, 400))
time.Sleep(time.Duration(randomInt(10, 50)) * time.Millisecond)
frameDuration = time.Duration(randomInt(16, 33)) * time.Millisecond
err = videoTrack.WriteSample(media.Sample{Data: smallBuf, Duration: frameDuration})
if err != nil {
log.Printf("webrtc.WriteSample burst ERROR: %s", err)
break
}
}
}
}
}()