webrtc: speed up gathering of incoming tracks

internal/protocols/webrtc/peer_connection.go

@@ -10,6 +10,7 @@
 
 	"github.com/pion/ice/v2"
 	"github.com/pion/interceptor"
+	"github.com/pion/sdp/v3"
 	"github.com/pion/webrtc/v3"
 
 	"github.com/bluenviron/mediamtx/internal/conf"
@@ -29,6 +30,37 @@
 	return false
 }
 
+// TracksAreValid checks whether tracks in the SDP are valid
+func TracksAreValid(medias []*sdp.MediaDescription) error {
+	videoTrack := false
+	audioTrack := false
+
+	for _, media := range medias {
+		switch media.MediaName.Media {
+		case "video":
+			if videoTrack {
+				return fmt.Errorf("only a single video and a single audio track are supported")
+			}
+			videoTrack = true
+
+		case "audio":
+			if audioTrack {
+				return fmt.Errorf("only a single video and a single audio track are supported")
+			}
+			audioTrack = true
+
+		default:
+			return fmt.Errorf("unsupported media '%s'", media.MediaName.Media)
+		}
+	}
+
+	if !videoTrack && !audioTrack {
+		return fmt.Errorf("no valid tracks count")
+	}
+
+	return nil
+}
+
 type trackRecvPair struct {
 	track    *webrtc.TrackRemote
 	receiver *webrtc.RTPReceiver
@@ -334,10 +366,12 @@
 }
 
 // GatherIncomingTracks gathers incoming tracks.
-func (co *PeerConnection) GatherIncomingTracks(
-	ctx context.Context,
-	maxCount int,
-) ([]*IncomingTrack, error) {
+func (co *PeerConnection) GatherIncomingTracks(ctx context.Context) ([]*IncomingTrack, error) {
+	var sdp sdp.SessionDescription
+	sdp.Unmarshal([]byte(co.wr.RemoteDescription().SDP)) //nolint:errcheck
+
+	maxTrackCount := len(sdp.MediaDescriptions)
+
 	var tracks []*IncomingTrack
 
 	t := time.NewTimer(time.Duration(co.TrackGatherTimeout))
@@ -346,7 +380,7 @@
 	for {
 		select {
 		case <-t.C:
-			if maxCount == 0 && len(tracks) != 0 {
+			if len(tracks) != 0 {
 				return tracks, nil
 			}
 			return nil, fmt.Errorf("deadline exceeded while waiting tracks")
@@ -358,7 +392,7 @@
 			}
 			tracks = append(tracks, track)
 
-			if len(tracks) == maxCount || len(tracks) >= 2 {
+			if len(tracks) >= maxTrackCount {
 				return tracks, nil
 			}
 

internal/protocols/webrtc/peer_connection_test.go

@@ -284,7 +284,7 @@ func TestPeerConnectionPublishRead(t *testing.T) {
 			})
 			require.NoError(t, err)
 
-			inc, err := pc2.GatherIncomingTracks(context.Background(), 1)
+			inc, err := pc2.GatherIncomingTracks(context.Background())
 			require.NoError(t, err)
 
 			require.Equal(t, ca.out, inc[0].Format())

internal/protocols/webrtc/whip_client.go

@@ -169,8 +169,7 @@
 		return nil, err
 	}
 
-	// check that there are at most two tracks
-	_, err = TrackCount(sdp.MediaDescriptions)
+	err = TracksAreValid(sdp.MediaDescriptions)
 	if err != nil {
 		c.deleteSession(context.Background()) //nolint:errcheck
 		c.pc.Close()
@@ -210,7 +209,7 @@
 		}
 	}
 
-	tracks, err := c.pc.GatherIncomingTracks(ctx, 0)
+	tracks, err := c.pc.GatherIncomingTracks(ctx)
 	if err != nil {
 		c.deleteSession(context.Background()) //nolint:errcheck
 		c.pc.Close()

internal/servers/webrtc/session.go

@@ -461,7 +461,7 @@ func (s *session) runPublish() (int, error) {
 		return http.StatusBadRequest, err
 	}
 
-	trackCount, err := webrtc.TrackCount(sdp.MediaDescriptions)
+	err = webrtc.TracksAreValid(sdp.MediaDescriptions)
 	if err != nil {
 		// RFC draft-ietf-wish-whip
 		// if the number of audio and or video
@@ -489,7 +489,7 @@ func (s *session) runPublish() (int, error) {
 	s.pc = pc
 	s.mutex.Unlock()
 
-	tracks, err := pc.GatherIncomingTracks(s.ctx, trackCount)
+	tracks, err := pc.GatherIncomingTracks(s.ctx)
 	if err != nil {
 		return 0, err
 	}