Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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package core
import (
"fmt"
"time"
"github.com/aler9/gortsplib/v2/pkg/format"
"github.com/aler9/gortsplib/v2/pkg/formatdecenc/rtpsimpleaudio"
"github.com/pion/rtp"
)
type dataOpus struct {
rtpPackets []*rtp.Packet
ntp time.Time
pts time.Duration
au []byte
}
func (d *dataOpus) getRTPPackets() []*rtp.Packet {
return d.rtpPackets
}
func (d *dataOpus) getNTP() time.Time {
return d.ntp
}
type formatProcessorOpus struct {
format *format.Opus
encoder *rtpsimpleaudio.Encoder
decoder *rtpsimpleaudio.Decoder
}
func newFormatProcessorOpus(
forma *format.Opus,
allocateEncoder bool,
) (*formatProcessorOpus, error) {
t := &formatProcessorOpus{
format: forma,
}
if allocateEncoder {
t.encoder = forma.CreateEncoder()
}
return t, nil
}
func (t *formatProcessorOpus) generateRTPPackets(tdata *dataOpus) error {
pkt, err := t.encoder.Encode(tdata.au, tdata.pts)
if err != nil {
return err
}
tdata.rtpPackets = []*rtp.Packet{pkt}
return nil
}
func (t *formatProcessorOpus) process(dat data, hasNonRTSPReaders bool) error {
tdata := dat.(*dataOpus)
if tdata.rtpPackets != nil {
pkt := tdata.rtpPackets[0]
// remove padding
pkt.Header.Padding = false
pkt.PaddingSize = 0
if pkt.MarshalSize() > maxPacketSize {
return fmt.Errorf("payload size (%d) is greater than maximum allowed (%d)",
pkt.MarshalSize(), maxPacketSize)
}
// decode from RTP
if hasNonRTSPReaders {
if t.decoder == nil {
t.decoder = t.format.CreateDecoder()
}
au, pts, err := t.decoder.Decode(pkt)
if err != nil {
return err
}
tdata.au = au
tdata.pts = pts
}
// route packet as is
return nil
}
return t.generateRTPPackets(tdata)
}