fix deadlock cause underlying buffer full (#413)

* fix deadlock cause underlying buffer full

pkg/sfu/buffer/buffer.go

@@ -86,6 +86,7 @@ type Buffer struct {
 	onAudioLevel func(level uint8, durationMs uint32)
 	feedbackCB   func([]rtcp.Packet)
 	feedbackTWCC func(sn uint16, timeNS int64, marker bool)
+	callbackOps  chan func()
 
 	// logger
 	logger logger.Logger
@@ -105,10 +106,11 @@ type Options struct {
 // NewBuffer constructs a new Buffer
 func NewBuffer(ssrc uint32, vp, ap *sync.Pool) *Buffer {
 	b := &Buffer{
-		mediaSSRC: ssrc,
-		videoPool: vp,
-		audioPool: ap,
-		logger:    logger.Logger(logger.GetLogger()), // will be reset with correct context via SetLogger
+		mediaSSRC:   ssrc,
+		videoPool:   vp,
+		audioPool:   ap,
+		logger:      logger.Logger(logger.GetLogger()), // will be reset with correct context via SetLogger
+		callbackOps: make(chan func(), 50),
 	}
 	b.bitrate.Store(make([]int64, len(b.bitrateHelper)))
 	b.extPackets.SetMinCapacity(7)
@@ -122,6 +124,11 @@ func (b *Buffer) SetLogger(logger logger.Logger) {
 func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability, o Options) {
 	b.Lock()
 	defer b.Unlock()
+	if b.bound {
+		return
+	}
+
+	go b.doCallbacks()
 
 	b.clockRate = codec.ClockRate
 	b.maxBitrate = int64(o.MaxBitRate)
@@ -179,6 +186,12 @@ func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapabili
 	b.logger.Debugw("NewBuffer", "MaxBitRate", o.MaxBitRate)
 }
 
+func (b *Buffer) doCallbacks() {
+	for op := range b.callbackOps {
+		op()
+	}
+}
+
 // Write adds an RTP Packet, out of order, new packet may be arrived later
 func (b *Buffer) Write(pkt []byte) (n int, err error) {
 	b.Lock()
@@ -256,6 +269,7 @@ func (b *Buffer) Close() error {
 		}
 		b.closed.set(true)
 		b.onClose()
+		close(b.callbackOps)
 	})
 	return nil
 }
@@ -417,7 +431,11 @@ func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime int64) {
 	// for bandwidth estimation
 	if b.twcc {
 		if ext := p.GetExtension(b.twccExt); len(ext) > 1 {
-			b.feedbackTWCC(binary.BigEndian.Uint16(ext[0:2]), arrivalTime, p.Marker)
+			sn := binary.BigEndian.Uint16(ext[0:2])
+			marker := p.Marker
+			b.callbackOps <- func() {
+				b.feedbackTWCC(sn, arrivalTime, marker)
+			}
 		}
 	}
 
@@ -432,7 +450,9 @@ func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime int64) {
 				if (p.Timestamp - b.latestTSForAudioLevel) < (1 << 31) {
 					duration := (int64(p.Timestamp) - int64(b.latestTSForAudioLevel)) * 1e3 / int64(b.clockRate)
 					if duration > 0 {
-						b.onAudioLevel(ext.Level, uint32(duration))
+						b.callbackOps <- func() {
+							b.onAudioLevel(ext.Level, uint32(duration))
+						}
 					}
 
 					b.latestTSForAudioLevel = p.Timestamp
@@ -510,7 +530,10 @@ func (b *Buffer) doReports(arrivalTime int64) {
 	b.bitrate.Store(bitrates)
 
 	// RTCP reports
-	go b.feedbackCB(b.getRTCP())
+	pkts := b.getRTCP()
+	b.callbackOps <- func() {
+		b.feedbackCB(pkts)
+	}
 }
 
 func (b *Buffer) buildNACKPacket() ([]rtcp.Packet, int) {