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Author SHA1 Message Date
Cédric Verstraeten
8c7a46e3ae Merge pull request #234 from kerberos-io/fix/fps-gop-size
fix/fps-gop-size
2026-02-11 15:05:31 +01:00
Cédric Verstraeten
57ccfaabf5 Merge branch 'fix/fps-gop-size' of github.com:kerberos-io/agent into fix/fps-gop-size 2026-02-11 14:59:34 +01:00
Cédric Verstraeten
4a9cb51e95 Update machinery/src/capture/gortsplib.go
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
2026-02-11 14:59:15 +01:00
Cédric Verstraeten
ab6f621e76 Merge branch 'fix/fps-gop-size' of github.com:kerberos-io/agent into fix/fps-gop-size 2026-02-11 14:58:44 +01:00
Cédric Verstraeten
c365ae5af2 Ensure thread-safe closure of peer connections in InitializeWebRTCConnection 2026-02-11 13:58:29 +00:00
Cédric Verstraeten
b05c3d1baa Update machinery/src/capture/gortsplib.go
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
2026-02-11 14:52:40 +01:00
Cédric Verstraeten
c7c7203fad Merge branch 'master' into fix/fps-gop-size 2026-02-11 14:48:05 +01:00
Cédric Verstraeten
d93f85b4f3 Refactor FPS calculation to use per-stream trackers for improved accuracy 2026-02-11 13:45:07 +00:00
Cédric Verstraeten
031212b98c Merge pull request #232 from kerberos-io/fix/fps-gop-size
fix/fps-gop-size
2026-02-11 14:27:18 +01:00
Cédric Verstraeten
a4837b3cb3 Implement PTS-based FPS calculation and GOP size adjustments 2026-02-11 13:14:29 +00:00
2 changed files with 146 additions and 87 deletions

View File

@@ -85,12 +85,8 @@ type Golibrtsp struct {
Streams []packets.Stream
// FPS calculation fields
lastFrameTime time.Time
frameTimeBuffer []time.Duration
frameBufferSize int
frameBufferIndex int
fpsMutex sync.Mutex
// Per-stream FPS calculation (keyed by stream index)
fpsTrackers map[int8]*fpsTracker
// I-frame interval tracking fields
packetsSinceLastKeyframe int
@@ -101,6 +97,78 @@ type Golibrtsp struct {
keyframeMutex sync.Mutex
}
// fpsTracker holds per-stream state for PTS-based FPS calculation.
// Each video stream (H264 / H265) gets its own tracker so PTS
// samples from different codecs never interleave.
type fpsTracker struct {
mu sync.Mutex
lastPTS time.Duration
hasPTS bool
frameTimeBuffer []time.Duration
bufferSize int
bufferIndex int
cachedFPS float64 // latest computed FPS
}
func newFPSTracker(bufferSize int) *fpsTracker {
return &fpsTracker{
frameTimeBuffer: make([]time.Duration, bufferSize),
bufferSize: bufferSize,
}
}
// update records a new PTS sample and returns the latest FPS estimate.
// It must be called once per complete decoded frame (after Decode()
// succeeds), not on every RTP packet fragment.
func (ft *fpsTracker) update(pts time.Duration) float64 {
ft.mu.Lock()
defer ft.mu.Unlock()
if !ft.hasPTS {
ft.lastPTS = pts
ft.hasPTS = true
return 0
}
interval := pts - ft.lastPTS
ft.lastPTS = pts
// Skip invalid intervals (zero, negative, or very large which
// indicate a PTS discontinuity or wrap).
if interval <= 0 || interval > 5*time.Second {
return ft.cachedFPS
}
ft.frameTimeBuffer[ft.bufferIndex] = interval
ft.bufferIndex = (ft.bufferIndex + 1) % ft.bufferSize
var totalInterval time.Duration
validSamples := 0
for _, iv := range ft.frameTimeBuffer {
if iv > 0 {
totalInterval += iv
validSamples++
}
}
if validSamples == 0 {
return ft.cachedFPS
}
avgInterval := totalInterval / time.Duration(validSamples)
if avgInterval == 0 {
return ft.cachedFPS
}
ft.cachedFPS = float64(time.Second) / float64(avgInterval)
return ft.cachedFPS
}
// fps returns the most recent FPS estimate without recording a new sample.
func (ft *fpsTracker) fps() float64 {
ft.mu.Lock()
defer ft.mu.Unlock()
return ft.cachedFPS
}
// Init function
var H264FrameDecoder *Decoder
var H265FrameDecoder *Decoder
@@ -548,18 +616,17 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
if len(rtppkt.Payload) > 0 {
// decode timestamp
pts, ok := g.Client.PacketPTS(g.VideoH264Media, rtppkt)
pts2, ok := g.Client.PacketPTS2(g.VideoH264Media, rtppkt)
if !ok {
log.Log.Debug("capture.golibrtsp.Start(): " + "unable to get PTS")
// decode timestamps — validate each call separately
pts, okPTS := g.Client.PacketPTS(g.VideoH264Media, rtppkt)
pts2, okPTS2 := g.Client.PacketPTS2(g.VideoH264Media, rtppkt)
if !okPTS2 {
log.Log.Debug("capture.golibrtsp.Start(): unable to get PTS2 from PacketPTS2")
return
}
// Extract access units from RTP packets
// We need to do this, because the decoder expects a full
// access unit. Once we have a full access unit, we can
// decode it, and know if it's a keyframe or not.
// Extract access units from RTP packets.
// We need a complete access unit to determine whether
// this is a keyframe.
au, errDecode := g.VideoH264Decoder.Decode(rtppkt)
if errDecode != nil {
if errDecode != rtph264.ErrNonStartingPacketAndNoPrevious && errDecode != rtph264.ErrMorePacketsNeeded {
@@ -568,6 +635,15 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
return
}
// Frame is complete — update per-stream FPS from PTS.
if okPTS {
if ft := g.fpsTrackers[g.VideoH264Index]; ft != nil {
if ptsFPS := ft.update(pts); ptsFPS > 0 && ptsFPS <= 120 {
g.Streams[g.VideoH264Index].FPS = ptsFPS
}
}
}
// We'll need to read out a few things.
// prepend an AUD. This is required by some players
filteredAU = [][]byte{
@@ -651,7 +727,11 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
keyframeInterval := g.trackKeyframeInterval(idrPresent)
if idrPresent && keyframeInterval > 0 {
avgInterval := g.getAverageKeyframeInterval()
gopDuration := float64(keyframeInterval) / g.Streams[g.VideoH265Index].FPS
fps := g.Streams[g.VideoH264Index].FPS
if fps <= 0 {
fps = 25.0 // Default fallback FPS
}
gopDuration := float64(keyframeInterval) / fps
gopSize := int(avgInterval) // Store GOP size in a separate variable
g.Streams[g.VideoH264Index].GopSize = gopSize
log.Log.Debug(fmt.Sprintf("capture.golibrtsp.Start(%s): Keyframe interval=%d packets, Avg=%.1f, GOP=%.1fs, GOPSize=%d",
@@ -716,18 +796,17 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
if len(rtppkt.Payload) > 0 {
// decode timestamp
pts, ok := g.Client.PacketPTS(g.VideoH265Media, rtppkt)
pts2, ok := g.Client.PacketPTS2(g.VideoH265Media, rtppkt)
if !ok {
log.Log.Debug("capture.golibrtsp.Start(): " + "unable to get PTS")
// decode timestamps — validate each call separately
pts, okPTS := g.Client.PacketPTS(g.VideoH265Media, rtppkt)
pts2, okPTS2 := g.Client.PacketPTS2(g.VideoH265Media, rtppkt)
if !okPTS2 {
log.Log.Debug("capture.golibrtsp.Start(): unable to get PTS")
return
}
// Extract access units from RTP packets
// We need to do this, because the decoder expects a full
// access unit. Once we have a full access unit, we can
// decode it, and know if it's a keyframe or not.
// Extract access units from RTP packets.
// We need a complete access unit to determine whether
// this is a keyframe.
au, errDecode := g.VideoH265Decoder.Decode(rtppkt)
if errDecode != nil {
if errDecode != rtph265.ErrNonStartingPacketAndNoPrevious && errDecode != rtph265.ErrMorePacketsNeeded {
@@ -736,6 +815,15 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
return
}
// Frame is complete — update per-stream FPS from PTS.
if okPTS {
if ft := g.fpsTrackers[g.VideoH265Index]; ft != nil {
if ptsFPS := ft.update(pts); ptsFPS > 0 && ptsFPS <= 120 {
g.Streams[g.VideoH265Index].FPS = ptsFPS
}
}
}
filteredAU = [][]byte{
{byte(h265.NALUType_AUD_NUT) << 1, 1, 0x50},
}
@@ -796,7 +884,11 @@ func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets
keyframeInterval := g.trackKeyframeInterval(isRandomAccess)
if isRandomAccess && keyframeInterval > 0 {
avgInterval := g.getAverageKeyframeInterval()
gopDuration := float64(keyframeInterval) / g.Streams[g.VideoH265Index].FPS
fps := g.Streams[g.VideoH265Index].FPS
if fps <= 0 {
fps = 25.0 // Default fallback FPS
}
gopDuration := float64(keyframeInterval) / fps
gopSize := int(avgInterval) // Store GOP size in a separate variable
g.Streams[g.VideoH265Index].GopSize = gopSize
log.Log.Debug(fmt.Sprintf("capture.golibrtsp.Start(%s): Keyframe interval=%d packets, Avg=%.1f, GOP=%.1fs, GOPSize=%d",
@@ -1179,10 +1271,11 @@ func WriteMPEG4Audio(forma *format.MPEG4Audio, aus [][]byte) ([]byte, error) {
// Initialize FPS calculation buffers
func (g *Golibrtsp) initFPSCalculation() {
g.frameBufferSize = 30 // Store last 30 frame intervals
g.frameTimeBuffer = make([]time.Duration, g.frameBufferSize)
g.frameBufferIndex = 0
g.lastFrameTime = time.Time{}
// Ensure the per-stream FPS trackers map exists. Individual trackers
// can be created lazily when a given stream index is first used.
if g.fpsTrackers == nil {
g.fpsTrackers = make(map[int8]*fpsTracker)
}
// Initialize I-frame interval tracking
g.keyframeBufferSize = 10 // Store last 10 keyframe intervals
@@ -1192,50 +1285,11 @@ func (g *Golibrtsp) initFPSCalculation() {
g.lastKeyframePacketCount = 0
}
// Calculate FPS from frame timestamps
func (g *Golibrtsp) calculateFPSFromTimestamps() float64 {
g.fpsMutex.Lock()
defer g.fpsMutex.Unlock()
if g.lastFrameTime.IsZero() {
g.lastFrameTime = time.Now()
return 0
}
now := time.Now()
interval := now.Sub(g.lastFrameTime)
g.lastFrameTime = now
// Store the interval
g.frameTimeBuffer[g.frameBufferIndex] = interval
g.frameBufferIndex = (g.frameBufferIndex + 1) % g.frameBufferSize
// Calculate average FPS from stored intervals
var totalInterval time.Duration
validSamples := 0
for _, interval := range g.frameTimeBuffer {
if interval > 0 {
totalInterval += interval
validSamples++
}
}
if validSamples == 0 {
return 0
}
avgInterval := totalInterval / time.Duration(validSamples)
if avgInterval == 0 {
return 0
}
return float64(time.Second) / float64(avgInterval)
}
// Get enhanced FPS information from SPS with fallback
// Get enhanced FPS information from SPS with fallback to PTS-based calculation.
// The PTS-based FPS is computed per completed frame via fpsTracker.update(),
// so by the time this is called we already have a good estimate.
func (g *Golibrtsp) getEnhancedFPS(sps *h264.SPS, streamIndex int8) float64 {
// First try to get FPS from SPS
// First try to get FPS from SPS VUI parameters
spsFPS := sps.FPS()
// Check if SPS FPS is reasonable (between 1 and 120 fps)
@@ -1244,11 +1298,13 @@ func (g *Golibrtsp) getEnhancedFPS(sps *h264.SPS, streamIndex int8) float64 {
return spsFPS
}
// Fallback to timestamp-based calculation
timestampFPS := g.calculateFPSFromTimestamps()
if timestampFPS > 0 && timestampFPS <= 120 {
log.Log.Debug(fmt.Sprintf("capture.golibrtsp.getEnhancedFPS(): Timestamp FPS: %.2f", timestampFPS))
return timestampFPS
// Fallback to PTS-based FPS (already calculated per-frame)
if ft := g.fpsTrackers[streamIndex]; ft != nil {
ptsFPS := ft.fps()
if ptsFPS > 0 && ptsFPS <= 120 {
log.Log.Debug(fmt.Sprintf("capture.golibrtsp.getEnhancedFPS(): PTS FPS: %.2f", ptsFPS))
return ptsFPS
}
}
// Return SPS FPS even if it seems unreasonable, or default

View File

@@ -49,6 +49,7 @@ type peerConnectionWrapper struct {
conn *pionWebRTC.PeerConnection
cancelCtx context.CancelFunc
done chan struct{}
closeOnce sync.Once
}
var globalConnectionManager = NewConnectionManager()
@@ -339,18 +340,20 @@ func InitializeWebRTCConnection(configuration *models.Configuration, communicati
switch connectionState {
case pionWebRTC.PeerConnectionStateDisconnected, pionWebRTC.PeerConnectionStateClosed:
count := globalConnectionManager.DecrementPeerCount()
log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Peer disconnected. Active peers: " + string(rune(count)))
wrapper.closeOnce.Do(func() {
count := globalConnectionManager.DecrementPeerCount()
log.Log.Info("webrtc.main.InitializeWebRTCConnection(): Peer disconnected. Active peers: " + string(rune(count)))
// Clean up resources
globalConnectionManager.CloseCandidateChannel(sessionKey)
// Clean up resources
globalConnectionManager.CloseCandidateChannel(sessionKey)
if err := peerConnection.Close(); err != nil {
log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error closing peer connection: " + err.Error())
}
if err := peerConnection.Close(); err != nil {
log.Log.Error("webrtc.main.InitializeWebRTCConnection(): error closing peer connection: " + err.Error())
}
globalConnectionManager.RemovePeerConnection(handshake.SessionID)
close(wrapper.done)
globalConnectionManager.RemovePeerConnection(handshake.SessionID)
close(wrapper.done)
})
case pionWebRTC.PeerConnectionStateConnected:
count := globalConnectionManager.IncrementPeerCount()