Initial commit

pkg/webrtc/streamer.go

@@ -0,0 +1,125 @@
+package webrtc
+
+import (
+	"encoding/json"
+	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"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)
+		}
+
+		if codec.Name == streamer.CodecH264 {
+			wrapper := h264.RTPPay(1200)
+			push = wrapper(push)
+
+			if codec.PayloadType != 255 {
+				wrapper = h264.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)
+}