365 lines
7.2 KiB
Go
365 lines
7.2 KiB
Go
package core
|
|
|
|
import (
|
|
"context"
|
|
"fmt"
|
|
"time"
|
|
|
|
"github.com/bluenviron/gortsplib/v3/pkg/formats"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/formats/rtpav1"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/formats/rtph264"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/formats/rtpvp8"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/formats/rtpvp9"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/media"
|
|
"github.com/bluenviron/gortsplib/v3/pkg/ringbuffer"
|
|
"github.com/pion/webrtc/v3"
|
|
|
|
"github.com/bluenviron/mediamtx/internal/formatprocessor"
|
|
)
|
|
|
|
type webRTCOutgoingTrack struct {
|
|
sender *webrtc.RTPSender
|
|
media *media.Media
|
|
format formats.Format
|
|
track *webrtc.TrackLocalStaticRTP
|
|
cb func(formatprocessor.Unit) error
|
|
}
|
|
|
|
func newWebRTCOutgoingTrackVideo(medias media.Medias) (*webRTCOutgoingTrack, error) {
|
|
var av1Format *formats.AV1
|
|
videoMedia := medias.FindFormat(&av1Format)
|
|
|
|
if videoMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeAV1,
|
|
ClockRate: 90000,
|
|
},
|
|
"av1",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
encoder := &rtpav1.Encoder{
|
|
PayloadType: 105,
|
|
PayloadMaxSize: webrtcPayloadMaxSize,
|
|
}
|
|
encoder.Init()
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: videoMedia,
|
|
format: av1Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
tunit := unit.(*formatprocessor.UnitAV1)
|
|
|
|
if tunit.OBUs == nil {
|
|
return nil
|
|
}
|
|
|
|
packets, err := encoder.Encode(tunit.OBUs, tunit.PTS)
|
|
if err != nil {
|
|
return nil
|
|
}
|
|
|
|
for _, pkt := range packets {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
var vp9Format *formats.VP9
|
|
videoMedia = medias.FindFormat(&vp9Format)
|
|
|
|
if videoMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeVP9,
|
|
ClockRate: uint32(vp9Format.ClockRate()),
|
|
},
|
|
"vp9",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
encoder := &rtpvp9.Encoder{
|
|
PayloadType: 96,
|
|
PayloadMaxSize: webrtcPayloadMaxSize,
|
|
}
|
|
encoder.Init()
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: videoMedia,
|
|
format: vp9Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
tunit := unit.(*formatprocessor.UnitVP9)
|
|
|
|
if tunit.Frame == nil {
|
|
return nil
|
|
}
|
|
|
|
packets, err := encoder.Encode(tunit.Frame, tunit.PTS)
|
|
if err != nil {
|
|
return nil
|
|
}
|
|
|
|
for _, pkt := range packets {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
var vp8Format *formats.VP8
|
|
videoMedia = medias.FindFormat(&vp8Format)
|
|
|
|
if videoMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeVP8,
|
|
ClockRate: uint32(vp8Format.ClockRate()),
|
|
},
|
|
"vp8",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
encoder := &rtpvp8.Encoder{
|
|
PayloadType: 96,
|
|
PayloadMaxSize: webrtcPayloadMaxSize,
|
|
}
|
|
encoder.Init()
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: videoMedia,
|
|
format: vp8Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
tunit := unit.(*formatprocessor.UnitVP8)
|
|
|
|
if tunit.Frame == nil {
|
|
return nil
|
|
}
|
|
|
|
packets, err := encoder.Encode(tunit.Frame, tunit.PTS)
|
|
if err != nil {
|
|
return nil
|
|
}
|
|
|
|
for _, pkt := range packets {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
var h264Format *formats.H264
|
|
videoMedia = medias.FindFormat(&h264Format)
|
|
|
|
if videoMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeH264,
|
|
ClockRate: uint32(h264Format.ClockRate()),
|
|
},
|
|
"h264",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
encoder := &rtph264.Encoder{
|
|
PayloadType: 96,
|
|
PayloadMaxSize: webrtcPayloadMaxSize,
|
|
}
|
|
encoder.Init()
|
|
|
|
var lastPTS time.Duration
|
|
firstNALUReceived := false
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: videoMedia,
|
|
format: h264Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
tunit := unit.(*formatprocessor.UnitH264)
|
|
|
|
if tunit.AU == nil {
|
|
return nil
|
|
}
|
|
|
|
if !firstNALUReceived {
|
|
firstNALUReceived = true
|
|
lastPTS = tunit.PTS
|
|
} else {
|
|
if tunit.PTS < lastPTS {
|
|
return fmt.Errorf("WebRTC doesn't support H264 streams with B-frames")
|
|
}
|
|
lastPTS = tunit.PTS
|
|
}
|
|
|
|
packets, err := encoder.Encode(tunit.AU, tunit.PTS)
|
|
if err != nil {
|
|
return nil
|
|
}
|
|
|
|
for _, pkt := range packets {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
return nil, nil
|
|
}
|
|
|
|
func newWebRTCOutgoingTrackAudio(medias media.Medias) (*webRTCOutgoingTrack, error) {
|
|
var opusFormat *formats.Opus
|
|
audioMedia := medias.FindFormat(&opusFormat)
|
|
|
|
if audioMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeOpus,
|
|
ClockRate: uint32(opusFormat.ClockRate()),
|
|
Channels: 2,
|
|
},
|
|
"opus",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: audioMedia,
|
|
format: opusFormat,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
for _, pkt := range unit.GetRTPPackets() {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
var g722Format *formats.G722
|
|
audioMedia = medias.FindFormat(&g722Format)
|
|
|
|
if audioMedia != nil {
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: webrtc.MimeTypeG722,
|
|
ClockRate: uint32(g722Format.ClockRate()),
|
|
},
|
|
"g722",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: audioMedia,
|
|
format: g722Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
for _, pkt := range unit.GetRTPPackets() {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
var g711Format *formats.G711
|
|
audioMedia = medias.FindFormat(&g711Format)
|
|
|
|
if audioMedia != nil {
|
|
var mtyp string
|
|
if g711Format.MULaw {
|
|
mtyp = webrtc.MimeTypePCMU
|
|
} else {
|
|
mtyp = webrtc.MimeTypePCMA
|
|
}
|
|
|
|
webRTCTrak, err := webrtc.NewTrackLocalStaticRTP(
|
|
webrtc.RTPCodecCapability{
|
|
MimeType: mtyp,
|
|
ClockRate: uint32(g711Format.ClockRate()),
|
|
},
|
|
"g711",
|
|
webrtcStreamID,
|
|
)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
return &webRTCOutgoingTrack{
|
|
media: audioMedia,
|
|
format: g711Format,
|
|
track: webRTCTrak,
|
|
cb: func(unit formatprocessor.Unit) error {
|
|
for _, pkt := range unit.GetRTPPackets() {
|
|
webRTCTrak.WriteRTP(pkt)
|
|
}
|
|
|
|
return nil
|
|
},
|
|
}, nil
|
|
}
|
|
|
|
return nil, nil
|
|
}
|
|
|
|
func (t *webRTCOutgoingTrack) start(
|
|
ctx context.Context,
|
|
r reader,
|
|
stream *stream,
|
|
ringBuffer *ringbuffer.RingBuffer,
|
|
writeError chan error,
|
|
) {
|
|
// read incoming RTCP packets to make interceptors work
|
|
go func() {
|
|
buf := make([]byte, 1500)
|
|
for {
|
|
_, _, err := t.sender.Read(buf)
|
|
if err != nil {
|
|
return
|
|
}
|
|
}
|
|
}()
|
|
|
|
stream.readerAdd(r, t.media, t.format, func(unit formatprocessor.Unit) {
|
|
ringBuffer.Push(func() {
|
|
err := t.cb(unit)
|
|
if err != nil {
|
|
select {
|
|
case writeError <- err:
|
|
case <-ctx.Done():
|
|
}
|
|
}
|
|
})
|
|
})
|
|
}
|