Files
rtsp-simple-server/internal/protocols/webrtc/incoming_track.go

319 lines
6.9 KiB
Go

package webrtc
import (
"time"
"github.com/bluenviron/gortsplib/v4/pkg/liberrors"
"github.com/bluenviron/gortsplib/v4/pkg/rtpreorderer"
"github.com/pion/rtcp"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"github.com/bluenviron/mediamtx/internal/logger"
)
const (
keyFrameInterval = 2 * time.Second
mimeTypeMultiopus = "audio/multiopus"
mimeTypeL16 = "audio/L16"
)
var incomingVideoCodecs = []webrtc.RTPCodecParameters{
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeAV1,
ClockRate: 90000,
SDPFmtpLine: "profile=1",
},
PayloadType: 96,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeAV1,
ClockRate: 90000,
},
PayloadType: 97,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=3",
},
PayloadType: 98,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=2",
},
PayloadType: 99,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=1",
},
PayloadType: 100,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP9,
ClockRate: 90000,
SDPFmtpLine: "profile-id=0",
},
PayloadType: 101,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeVP8,
ClockRate: 90000,
},
PayloadType: 102,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH265,
ClockRate: 90000,
SDPFmtpLine: "level-id=93;profile-id=2;tier-flag=0;tx-mode=SRST",
},
PayloadType: 103,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH265,
ClockRate: 90000,
SDPFmtpLine: "level-id=93;profile-id=1;tier-flag=0;tx-mode=SRST",
},
PayloadType: 104,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
},
PayloadType: 105,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
},
PayloadType: 106,
},
}
var incomingAudioCodecs = []webrtc.RTPCodecParameters{
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 3,
SDPFmtpLine: "channel_mapping=0,2,1;num_streams=2;coupled_streams=1",
},
PayloadType: 112,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 4,
SDPFmtpLine: "channel_mapping=0,1,2,3;num_streams=2;coupled_streams=2",
},
PayloadType: 113,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 5,
SDPFmtpLine: "channel_mapping=0,4,1,2,3;num_streams=3;coupled_streams=2",
},
PayloadType: 114,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 6,
SDPFmtpLine: "channel_mapping=0,4,1,2,3,5;num_streams=4;coupled_streams=2",
},
PayloadType: 115,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 7,
SDPFmtpLine: "channel_mapping=0,4,1,2,3,5,6;num_streams=4;coupled_streams=4",
},
PayloadType: 116,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeMultiopus,
ClockRate: 48000,
Channels: 8,
SDPFmtpLine: "channel_mapping=0,6,1,4,5,2,3,7;num_streams=5;coupled_streams=4",
},
PayloadType: 117,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeOpus,
ClockRate: 48000,
Channels: 2,
SDPFmtpLine: "minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1",
},
PayloadType: 111,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeG722,
ClockRate: 8000,
},
PayloadType: 9,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: 8000,
Channels: 2,
},
PayloadType: 118,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: 8000,
Channels: 2,
},
PayloadType: 119,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: 8000,
},
PayloadType: 0,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: 8000,
},
PayloadType: 8,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeL16,
ClockRate: 8000,
Channels: 2,
},
PayloadType: 120,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeL16,
ClockRate: 16000,
Channels: 2,
},
PayloadType: 121,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: mimeTypeL16,
ClockRate: 48000,
Channels: 2,
},
PayloadType: 122,
},
}
// IncomingTrack is an incoming track.
type IncomingTrack struct {
OnPacketRTP func(*rtp.Packet)
track *webrtc.TrackRemote
receiver *webrtc.RTPReceiver
writeRTCP func([]rtcp.Packet) error
log logger.Writer
}
func (t *IncomingTrack) initialize() {
t.OnPacketRTP = func(*rtp.Packet) {}
}
// ClockRate returns the clock rate. Needed by rtptime.GlobalDecoder
func (t *IncomingTrack) ClockRate() int {
return int(t.track.Codec().ClockRate)
}
// PTSEqualsDTS returns whether PTS equals DTS. Needed by rtptime.GlobalDecoder
func (*IncomingTrack) PTSEqualsDTS(*rtp.Packet) bool {
return true
}
func (t *IncomingTrack) start() {
// read incoming RTCP packets to make interceptors work
go func() {
buf := make([]byte, 1500)
for {
_, _, err := t.receiver.Read(buf)
if err != nil {
return
}
}
}()
// send period key frame requests
if t.track.Kind() == webrtc.RTPCodecTypeVideo {
go func() {
keyframeTicker := time.NewTicker(keyFrameInterval)
defer keyframeTicker.Stop()
for range keyframeTicker.C {
err := t.writeRTCP([]rtcp.Packet{
&rtcp.PictureLossIndication{
MediaSSRC: uint32(t.track.SSRC()),
},
})
if err != nil {
return
}
}
}()
}
// read incoming RTP packets
go func() {
reorderer := rtpreorderer.New()
for {
pkt, _, err := t.track.ReadRTP()
if err != nil {
return
}
packets, lost := reorderer.Process(pkt)
if lost != 0 {
t.log.Log(logger.Warn, (liberrors.ErrClientRTPPacketsLost{Lost: lost}).Error())
// do not return
}
for _, pkt := range packets {
// sometimes Chrome sends empty RTP packets. ignore them.
if len(pkt.Payload) == 0 {
continue
}
t.OnPacketRTP(pkt)
}
}
}()
}