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rtsp server: generate RTCP sender reports automatically; stop routing RTCP packets

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This commit is contained in:
aler9 2022-03-24 15:38:54 +01:00
parent a6986e9fa4
commit f53b316c0d
13 changed files with 8 additions and 160 deletions
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15
internal/core/rtmp_source.go
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@ -11,11 +11,9 @@ import (
"github.com/aler9/gortsplib/pkg/rtpaac"
"github.com/aler9/gortsplib/pkg/rtph264"
"github.com/notedit/rtmp/av"
"github.com/pion/rtp/v2"
"github.com/aler9/rtsp-simple-server/internal/conf"
"github.com/aler9/rtsp-simple-server/internal/logger"
"github.com/aler9/rtsp-simple-server/internal/rtcpsenderset"
"github.com/aler9/rtsp-simple-server/internal/rtmp"
)
@ -166,15 +164,6 @@ func (s *rtmpSource) runInner() bool {
defer func() {
s.parent.onSourceStaticSetNotReady(pathSourceStaticSetNotReadyReq{source: s})
}()
rtcpSenders := rtcpsenderset.New(tracks, res.stream.writePacketRTCP)
defer rtcpSenders.Close()
onPacketRTP := func(trackID int, pkt *rtp.Packet) {
rtcpSenders.OnPacketRTP(trackID, pkt)
res.stream.writePacketRTP(trackID, pkt)
}
for {
conn.SetReadDeadline(time.Now().Add(time.Duration(s.readTimeout)))
pkt, err := conn.ReadPacket()
@ -211,7 +200,7 @@ func (s *rtmpSource) runInner() bool {
}
for _, pkt := range pkts {
onPacketRTP(videoTrackID, pkt)
res.stream.writePacketRTP(videoTrackID, pkt)
}
case av.AAC:
@ -225,7 +214,7 @@ func (s *rtmpSource) runInner() bool {
}
for _, pkt := range pkts {
onPacketRTP(audioTrackID, pkt)
res.stream.writePacketRTP(audioTrackID, pkt)
}
}
}