Improve play audio on RTSP backchannel

This commit is contained in:
Alex X
2024-05-19 11:56:33 +03:00
parent 50f9913c41
commit bc8295baee
3 changed files with 49 additions and 29 deletions
+7
View File
@@ -45,6 +45,13 @@
[video4linux2,v4l2 @ 0x7f7de7c58bc0] Compressed: mjpeg : Motion-JPEG : 640x480 160x120 176x144 320x176 320x240 352x288 432x240 544x288 640x360 752x416 800x448 800x600 864x480 960x544 960x720 1024x576 1184x656 1280x720 1280x960
```
## TTS
```yaml
streams:
tts: ffmpeg:#input=-re -f lavfi -i "flite=text='1 2 3 4 5 6 7 8 9 0'"#audio=pcma
```
## Useful links
- https://superuser.com/questions/564402/explanation-of-x264-tune
+10 -6
View File
@@ -2,6 +2,8 @@ package streams
import (
"errors"
"time"
"github.com/AlexxIT/go2rtc/pkg/core"
)
@@ -80,18 +82,20 @@ func (s *Stream) Play(source string) error {
s.AddInternalProducer(src)
s.AddInternalConsumer(cons)
go func() {
_ = src.Start()
_ = dst.Stop()
s.RemoveProducer(src)
}()
go func() {
_ = dst.Start()
_ = src.Stop()
s.RemoveInternalConsumer(cons)
}()
go func() {
_ = src.Start()
// little timeout before stop dst, so the buffer can be transferred
time.Sleep(time.Second)
_ = dst.Stop()
s.RemoveProducer(src)
}()
return nil
}
+32 -23
View File
@@ -74,19 +74,38 @@ func (c *Conn) AddTrack(media *core.Media, codec *core.Codec, track *core.Receiv
return nil
}
const (
startVideoBuf = 32 * 1024 // 32KB
startAudioBuf = 2 * 1024 // 2KB
maxBuf = 1024 * 1024 // 1MB
rtpHdr = 12 // basic RTP header size
intHdr = 4 // interleaved header size
)
func (c *Conn) packetWriter(codec *core.Codec, channel, payloadType uint8) core.HandlerFunc {
var buf []byte
var n int
video := codec.IsVideo()
if video {
buf = make([]byte, 32*1024) // 32KB
buf = make([]byte, startVideoBuf)
} else {
buf = make([]byte, 2*1024) // 2KB
buf = make([]byte, startAudioBuf)
}
flushBuf := func() {
if err := c.conn.SetWriteDeadline(time.Now().Add(Timeout)); err != nil {
return
}
//log.Printf("[rtsp] channel:%2d write_size:%6d buffer_size:%6d", channel, n, len(buf))
if _, err := c.conn.Write(buf[:n]); err == nil {
c.send += n
}
n = 0
}
handlerFunc := func(packet *rtp.Packet) {
if c.state == StateNone || !c.playOK {
if c.state == StateNone {
return
}
@@ -106,16 +125,13 @@ func (c *Conn) packetWriter(codec *core.Codec, channel, payloadType uint8) core.
packet.Marker = true // better to have marker on all audio packets
}
size := 12 + len(packet.Payload)
size := rtpHdr + len(packet.Payload)
if n+4+size > len(buf) {
if len(buf) < 1024*1024 {
buf = append(buf, make([]byte, len(buf))...)
if l := len(buf); n+intHdr+size > l {
if l < maxBuf {
buf = append(buf, make([]byte, l)...) // double buffer size
} else {
if _, err := c.conn.Write(buf[:n]); err == nil {
c.send += n
}
n = 0
flushBuf()
}
}
@@ -134,21 +150,14 @@ func (c *Conn) packetWriter(codec *core.Codec, channel, payloadType uint8) core.
n += 4 + size
if !packet.Marker {
return // collect continious video packets to buffer
}
if err := c.conn.SetWriteDeadline(time.Now().Add(Timeout)); err != nil {
if !packet.Marker || !c.playOK {
// collect continious video packets to buffer
// or wait OK for PLAY command for backchannel
//log.Printf("[rtsp] collecting buffer ok=%t", c.playOK)
return
}
//log.Printf("[rtsp] channel:%2d write_size:%6d buffer_size:%6d", channel, n, len(buf))
if _, err := c.conn.Write(buf[:n]); err == nil {
c.send += n
}
n = 0
flushBuf()
}
if !codec.IsRTP() {