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Refactor WriteToTrack to improve sample handling by using last processed audio and video samples, enhancing buffer duration calculation and streamlining packet processing.

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This commit is contained in:
Cédric Verstraeten
2025-06-06 14:36:19 +00:00
parent c4453bb8b3
commit bfde87f888
1 changed files with 25 additions and 27 deletions
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52
machinery/src/webrtc/main.go
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@@ -394,20 +394,17 @@ func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.C
var cursorError error
var pkt packets.Packet
var nextPkt packets.Packet
var lastAudioSample *pionMedia.Sample = nil
var lastVideoSample *pionMedia.Sample = nil
start := false
receivedKeyFrame := false
lastKeepAlive := "0"
peerCount := "0"
if cursorError == nil {
pkt, cursorError = livestreamCursor.ReadPacket()
}
for cursorError == nil {
nextPkt, cursorError = livestreamCursor.ReadPacket()
pkt, cursorError = livestreamCursor.ReadPacket()
//if config.Capture.ForwardWebRTC != "true" && peerConnectionCount == 0 {
// start = false
@@ -433,13 +430,11 @@ func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.C
if config.Capture.ForwardWebRTC == "true" && (hasTimedOut || hasNoPeers) {
start = false
receivedKeyFrame = false
pkt = nextPkt
continue
}
if len(pkt.Data) == 0 || pkt.Data == nil {
receivedKeyFrame = false
pkt = nextPkt
continue
}
@@ -447,7 +442,6 @@ func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.C
if pkt.IsKeyFrame {
receivedKeyFrame = true
} else {
pkt = nextPkt
continue
}
}
@@ -459,24 +453,27 @@ func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.C
if pkt.IsVideo {
// Calculate the difference
bufferDuration := nextPkt.Time - pkt.Time
// Start at the first keyframe
if pkt.IsKeyFrame {
start = true
}
if start {
bufferDurationCasted := time.Duration(bufferDuration) * time.Millisecond
sample := pionMedia.Sample{Data: pkt.Data, Duration: bufferDurationCasted, PacketTimestamp: uint32(pkt.Time)}
sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.TimeLegacy)}
//sample = pionMedia.Sample{Data: pkt.Data, Duration: time.Second}
if config.Capture.ForwardWebRTC == "true" {
// We will send the video to a remote peer
// TODO..
} else {
if err := videoTrack.WriteSample(sample); err != nil && err != io.ErrClosedPipe {
log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
if lastVideoSample != nil {
duration := sample.PacketTimestamp - lastVideoSample.PacketTimestamp
bufferDurationCasted := time.Duration(duration) * time.Millisecond
lastVideoSample.Duration = bufferDurationCasted
if err := videoTrack.WriteSample(*lastVideoSample); err != nil && err != io.ErrClosedPipe {
log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
}
}
lastVideoSample = &sample
}
}
} else if pkt.IsAudio {
@@ -491,22 +488,23 @@ func WriteToTrack(livestreamCursor *packets.QueueCursor, configuration *models.C
// and then encode it to PCM_MULAW.
// TODO..
//d := fdkaac.NewAacDecoder()
pkt = nextPkt
continue
}
// Calculate the difference
bufferDuration := nextPkt.Time - pkt.Time
// We will send the audio
bufferDurationCasted := time.Duration(bufferDuration) * time.Millisecond
sample := pionMedia.Sample{Data: pkt.Data, Duration: bufferDurationCasted, PacketTimestamp: uint32(pkt.Time)}
if err := audioTrack.WriteSample(sample); err != nil && err != io.ErrClosedPipe {
log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
}
}
sample := pionMedia.Sample{Data: pkt.Data, PacketTimestamp: uint32(pkt.TimeLegacy)}
pkt = nextPkt
if lastAudioSample != nil {
duration := sample.PacketTimestamp - lastAudioSample.PacketTimestamp
bufferDurationCasted := time.Duration(duration) * time.Millisecond
lastAudioSample.Duration = bufferDurationCasted
if err := audioTrack.WriteSample(*lastAudioSample); err != nil && err != io.ErrClosedPipe {
log.Log.Error("webrtc.main.WriteToTrack(): something went wrong while writing sample: " + err.Error())
}
}
lastAudioSample = &sample
}
}
}