Files
go2rtc/pkg/webrtc/consumer.go
2022-09-21 22:28:59 +03:00

139 lines
3.1 KiB
Go

package webrtc
import (
"encoding/json"
"github.com/AlexxIT/go2rtc/pkg/h264"
"github.com/AlexxIT/go2rtc/pkg/h265"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
)
// Consumer
func (c *Conn) GetMedias() []*streamer.Media {
return c.medias
}
func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
switch track.Direction {
// send our track to WebRTC consumer
case streamer.DirectionSendonly:
codec := track.Codec
// webrtc.codecParametersFuzzySearch
caps := webrtc.RTPCodecCapability{
MimeType: MimeType(codec),
Channels: codec.Channels,
ClockRate: codec.ClockRate,
}
if codec.Name == streamer.CodecH264 {
// don't know if this really neccessary
// I have tested multiple browsers and H264 profile has no effect on anything
caps.SDPFmtpLine = "packetization-mode=1;profile-level-id=42e01f"
}
// important to use same streamID so JS will automatically
// join two tracks as one source/stream
trackLocal, err := webrtc.NewTrackLocalStaticRTP(
caps, caps.MimeType[:5], "go2rtc",
)
if err != nil {
return nil
}
if _, err = c.Conn.AddTrack(trackLocal); err != nil {
return nil
}
push := func(packet *rtp.Packet) error {
c.send += packet.MarshalSize()
return trackLocal.WriteRTP(packet)
}
switch codec.Name {
case streamer.CodecH264:
wrapper := h264.RTPPay(1200)
push = wrapper(push)
if h264.IsAVC(codec) {
wrapper = h264.RepairAVC(track)
} else {
wrapper = h264.RTPDepay(track)
}
push = wrapper(push)
case streamer.CodecH265:
// SafariPay because it is the only browser in the world
// that supports WebRTC + H265
wrapper := h265.SafariPay(1200)
push = wrapper(push)
wrapper = h265.RTPDepay(track)
push = wrapper(push)
}
track = track.Bind(push)
c.tracks = append(c.tracks, track)
return track
// receive track from WebRTC consumer (microphone, backchannel, two way audio)
case streamer.DirectionRecvonly:
for _, tr := range c.Conn.GetTransceivers() {
if tr.Mid() != media.MID {
continue
}
codec := track.Codec
caps := webrtc.RTPCodecCapability{
MimeType: MimeType(codec),
ClockRate: codec.ClockRate,
Channels: codec.Channels,
}
codecs := []webrtc.RTPCodecParameters{
{RTPCodecCapability: caps},
}
if err := tr.SetCodecPreferences(codecs); err != nil {
return nil
}
c.tracks = append(c.tracks, track)
return track
}
}
panic("wrong direction")
}
//
func (c *Conn) Push(msg interface{}) {
if msg := msg.(*streamer.Message); msg != nil {
if msg.Type == MsgTypeCandidate {
_ = c.Conn.AddICECandidate(webrtc.ICECandidateInit{
Candidate: msg.Value.(string),
})
}
}
}
func (c *Conn) MarshalJSON() ([]byte, error) {
v := map[string]interface{}{
streamer.JSONType: "WebRTC server consumer",
streamer.JSONRemoteAddr: c.remote(),
}
if c.receive > 0 {
v[streamer.JSONReceive] = c.receive
}
if c.send > 0 {
v[streamer.JSONSend] = c.send
}
if c.UserAgent != "" {
v[streamer.JSONUserAgent] = c.UserAgent
}
return json.Marshal(v)
}