Merge Brandon McGinty's UDP and reliability work

Merge the full history from bmmcginty/barnard-stormdragon-ai-fork dev, including UDP audio transport, jitter handling, concurrency fixes, and regression coverage.

Co-authored-by: Brandon McGinty <git@bmcginty.us>

README.md

@@ -140,6 +140,35 @@ If you modify the config file while Barnard is running, your changes may be over
 You can set username and defaultserver in your config file, and they will be used if none is specified when launching barnard.
 (Note that the default username (an empty string) and the default server name (localhost:64738) have been the defaults for barnard up to this  point, and have been left that way for compatibility.)
 
+## Audio Packet Duration
+
+Barnard sends 10 ms audio packets by default. On a slow or unstable connection,
+using larger packets can reduce packet overhead and make short dropouts less
+noticeable, at the cost of additional voice latency. Start Barnard with one of
+the supported durations:
+
+```sh
+barnard --audio-interval 20
+```
+
+Supported values are `10`, `20`, `40`, and `60` milliseconds. Try `20` ms
+first; use `40` ms only if the connection remains unreliable.
+
+## Incoming Audio Jitter Buffer
+
+Barnard holds 40 ms of audio separately for each speaker before starting
+playback. This prevents brief delayed UDP packets from draining OpenAL's audio
+queue, which otherwise produces clicks or pops. To adjust this tradeoff between
+resilience and added incoming latency:
+
+```sh
+barnard --jitter-buffer 60
+```
+
+Supported values are `0`, `20`, `40` (default), and `60` milliseconds. Try
+`60` ms for a lossy or jittery connection. Use `0` only when minimizing latency
+is more important than avoiding playback underruns.
+
 ## Audio Devices
 
 You can set the default input and output devices in the config file as well.

client_notification_test.go

@@ -70,6 +70,38 @@ func TestNotificationExpansionIsSinglePassAndNotifyDoesNotBlock(t *testing.T) {
 	}
 }
 
+func TestAudioIntervalDuration(t *testing.T) {
+	for _, milliseconds := range []int{10, 20, 40, 60} {
+		got, err := audioIntervalDuration(milliseconds)
+		if err != nil {
+			t.Errorf("audioIntervalDuration(%d): %v", milliseconds, err)
+			continue
+		}
+		if got != time.Duration(milliseconds)*time.Millisecond {
+			t.Errorf("audioIntervalDuration(%d) = %v", milliseconds, got)
+		}
+	}
+	if _, err := audioIntervalDuration(30); err == nil {
+		t.Fatal("audioIntervalDuration accepted unsupported duration")
+	}
+}
+
+func TestJitterBufferDuration(t *testing.T) {
+	for _, milliseconds := range []int{0, 20, 40, 60} {
+		got, err := jitterBufferDuration(milliseconds)
+		if err != nil {
+			t.Errorf("jitterBufferDuration(%d): %v", milliseconds, err)
+			continue
+		}
+		if got != time.Duration(milliseconds)*time.Millisecond {
+			t.Errorf("jitterBufferDuration(%d) = %v", milliseconds, got)
+		}
+	}
+	if _, err := jitterBufferDuration(10); err == nil {
+		t.Fatal("jitterBufferDuration accepted unsupported duration")
+	}
+}
+
 func TestServerAddressDefaultsPortWithoutBreakingIPv6(t *testing.T) {
 	for input, want := range map[string]string{
 		"server":          "server:64738",

gumble/gumble/audio.go

@@ -92,6 +92,10 @@ type AudioPacket struct {
 
 	AudioBuffer
 
+	// Terminator marks the final packet in a talk burst. Audio listeners use
+	// it to discard ordering state before the sender starts a new burst.
+	Terminator bool
+
 	HasPosition      bool
 	X, Y, Z          float32
 	VolumeAdjustment float32

gumble/gumble/client.go

@@ -3,6 +3,7 @@ package gumble
 import (
 	"crypto/tls"
 	"errors"
+	"fmt"
 	"math"
 	"net"
 	"runtime"
@@ -101,6 +102,19 @@ func Dial(config *Config) (*Client, error) {
 	return DialWithDialer(new(net.Dialer), config, nil)
 }
 
+// tlsServerName returns the hostname portion of a Mumble server address for
+// TLS certificate verification and SNI.
+func tlsServerName(address string) (string, error) {
+	host, _, err := net.SplitHostPort(address)
+	if err != nil {
+		return "", fmt.Errorf("gumble: derive TLS server name from %q: %w", address, err)
+	}
+	if host == "" {
+		return "", fmt.Errorf("gumble: derive TLS server name from %q: empty host", address)
+	}
+	return host, nil
+}
+
 // DialWithDialer connects to the Mumble server at the address given in config.
 //
 // The function returns after the connection has been established, the initial
@@ -120,6 +134,23 @@ func DialWithDialer(dialer *net.Dialer, config *Config, tlsConfig *tls.Config) (
 	if err != nil {
 		return nil, err
 	}
+
+	// tls.Client cannot infer a server name from an already-open connection.
+	// Clone the caller's configuration before deriving it so reconnects and
+	// concurrent clients do not mutate a shared configuration.
+	if tlsConfig == nil {
+		tlsConfig = &tls.Config{}
+	} else {
+		tlsConfig = tlsConfig.Clone()
+	}
+	if tlsConfig.ServerName == "" {
+		serverName, err := tlsServerName(config.Address)
+		if err != nil {
+			rawConn.Close()
+			return nil, err
+		}
+		tlsConfig.ServerName = serverName
+	}
 	conn := tls.Client(rawConn, tlsConfig)
 	// net.Dialer.Timeout covers only the TCP dial. Apply the same bounded
 	// deadline to TLS negotiation so a peer that accepts but never responds

gumble/gumble/client_tls_test.go

@@ -0,0 +1,28 @@
+package gumble
+
+import "testing"
+
+func TestTLSServerNameUsesAddressHost(t *testing.T) {
+	for _, test := range []struct {
+		address string
+		want    string
+	}{
+		{"mumble.example:64738", "mumble.example"},
+		{"[2001:db8::1]:64738", "2001:db8::1"},
+	} {
+		got, err := tlsServerName(test.address)
+		if err != nil {
+			t.Errorf("tlsServerName(%q): %v", test.address, err)
+			continue
+		}
+		if got != test.want {
+			t.Errorf("tlsServerName(%q) = %q, want %q", test.address, got, test.want)
+		}
+	}
+}
+
+func TestTLSServerNameRejectsAddressWithoutHost(t *testing.T) {
+	if _, err := tlsServerName(":64738"); err == nil {
+		t.Fatal("tlsServerName accepted an empty host")
+	}
+}

gumble/gumble/config.go

@@ -25,6 +25,9 @@ type Config struct {
 	AudioInterval time.Duration
 	// AudioDataBytes is the number of bytes that an audio frame can use.
 	AudioDataBytes int
+	// IncomingAudioBuffer is the amount of per-speaker audio retained before
+	// playback starts, absorbing jitter in incoming UDP packet delivery.
+	IncomingAudioBuffer time.Duration
 
 	// DisableUDP forces all audio to use the TCP tunnel instead of UDP.
 	DisableUDP bool
@@ -41,6 +44,7 @@ func NewConfig() *Config {
 		Buffers:             8,
 		AudioInterval:       AudioDefaultInterval,
 		AudioDataBytes:      AudioDefaultDataBytes,
+		IncomingAudioBuffer: 40 * time.Millisecond,
 	}
 }
 
@@ -54,6 +58,9 @@ func (c *Config) Validate() error {
 	if c.AudioDataBytes <= 0 {
 		return fmt.Errorf("gumble: AudioDataBytes must be positive")
 	}
+	if c.IncomingAudioBuffer < 0 {
+		return fmt.Errorf("gumble: IncomingAudioBuffer must not be negative")
+	}
 	if c.Buffers <= 0 {
 		return fmt.Errorf("gumble: Buffers must be positive")
 	}

gumble/gumble/handlers.go

@@ -223,6 +223,11 @@ func (c *Client) handleUDPTunnel(buffer []byte) error {
 	}
 
 	c.dispatchAudio(user, &event)
+	if isFinal {
+		decoder.Reset()
+		user.audioSequenceValid = false
+		c.dispatchAudio(user, &AudioPacket{Client: c, Sender: user, Terminator: true})
+	}
 	return nil
 }
 

gumble/gumble/udp.go

@@ -59,7 +59,10 @@ func (c *Client) udpReadRoutine() {
 			return
 		}
 		packetCount++
-		if log.Enabled(log.LevelDebug) {
+		// A synchronous log write for every UDP datagram can itself make the
+		// reader fall behind and lose voice packets. Keep enough samples to
+		// diagnose framing while avoiding work on the audio hot path.
+		if log.Enabled(log.LevelDebug) && (packetCount <= 3 || packetCount%1000 == 0) {
 			log.Debug("UDP recv #%d: %d bytes from %s hex=%s",
 				packetCount, n, addr, hex.EncodeToString(buf[:n]))
 		}

gumble/gumble/udp15.go

@@ -292,10 +292,13 @@ func (cs *cryptState15) decrypt15(packet []byte) ([]byte, error) {
 			backupIV(cs.decryptIV[:])
 			restore = true
 		} else if ivByte > cs.decryptIV[0] && diff > 0 {
-			// We missed packets; catch up. Already handled above.
+			// We missed packets; move the low IV byte forward.
+			cs.decryptIV[0] = ivByte
 		} else if ivByte < cs.decryptIV[0] && diff > 0 {
-			// Wrapped forward; advance and catch up.
+			// We missed packets across a low-byte wrap. The IV's higher
-			advanceIV(cs.decryptIV[:])
+			// bytes must advance even though the received low byte is set
+			// below rather than incremented.
+			advanceIVHighBytes(cs.decryptIV[:])
 			cs.decryptIV[0] = ivByte
 		} else {
 			return nil, errors.New("gumble: OCB IV too far off")
@@ -340,6 +343,16 @@ func advanceIV(iv []byte) {
 	}
 }
 
+// advanceIVHighBytes advances all but the low IV byte as a little-endian integer.
+func advanceIVHighBytes(iv []byte) {
+	for i := 1; i < len(iv); i++ {
+		iv[i]++
+		if iv[i] != 0 {
+			break
+		}
+	}
+}
+
 // backupIV decrements a 16-byte IV as a little-endian integer.
 func backupIV(iv []byte) {
 	for i := 0; i < len(iv); i++ {
@@ -626,8 +639,10 @@ func (c *Client) WriteAudioUDP15(format byte, target uint32, sequence int64, dat
 		return false, err
 	}
 
+	if log.Enabled(log.LevelDebug) && (frameNum < 3 || frameNum%1000 == 0 || final) {
 		log.Debug("UDP15 send: frame=%d opus_len=%d enc_len=%d final=%v",
 			frameNum, len(data), len(encrypted), final)
+	}
 
 	_, err = udpConn.Write(encrypted)
 	if err != nil {
@@ -662,7 +677,9 @@ func (c *Client) HandleUDPPacket15(packet []byte, pktNum uint64) {
 		return
 	}
 
+	if log.Enabled(log.LevelDebug) && (pktNum <= 3 || pktNum%1000 == 0) {
 		log.Debug("UDP15 #%d: decrypt OK, plaintext_len=%d", pktNum, len(plaintext))
+	}
 	c.markUDPActive()
 
 	// Check type byte (0x00 = Audio, 0x01 = Ping)
@@ -735,6 +752,9 @@ func (c *Client) dispatchOpus15(pktNum uint64, session uint32, frameNum int64, o
 		decoder.Reset()
 		user.audioSequenceValid = false
 		user.audioFrameStep = 0
+		// The audio stream remains open between talk bursts. Deliver the
+		// terminator so listeners can reset their own packet ordering state.
+		c.dispatchAudio(user, &AudioPacket{Client: c, Sender: user, Terminator: true})
 		log.Info("UDP15 #%d: terminator for %s, decoder reset", pktNum, user.Name)
 		return
 	}
@@ -743,7 +763,7 @@ func (c *Client) dispatchOpus15(pktNum uint64, session uint32, frameNum int64, o
 		return
 	}
 
-	c.decodeAndDispatch(pktNum, user, decoder, frameNum, opusData, context, position, volumeAdjustment)
+	c.decodeAndDispatch(pktNum, user, decoder, frameNum, opusData, terminator, context, position, volumeAdjustment)
 }
 
 // handleLegacyUDPVoice parses the legacy UDPVoice format (type byte 0x80)
@@ -792,7 +812,7 @@ func (c *Client) handleLegacyUDPVoice(pktNum uint64, data []byte) {
 }
 
 // decodeAndDispatch decodes an Opus frame and dispatches PCM to audio listeners.
-func (c *Client) decodeAndDispatch(pktNum uint64, user *User, decoder AudioDecoder, frameNum int64, opusData []byte, context uint32, position *[3]float32, volumeAdjustment float32) {
+func (c *Client) decodeAndDispatch(pktNum uint64, user *User, decoder AudioDecoder, frameNum int64, opusData []byte, terminator bool, context uint32, position *[3]float32, volumeAdjustment float32) {
 	// Frame numbers are timestamps in 10 ms units, not packet counters. For
 	// example, a standard 20 ms Opus packet advances its frame number by two.
 	// Only generate PLC for complete missing packets; treating every timestamp
@@ -828,7 +848,9 @@ func (c *Client) decodeAndDispatch(pktNum uint64, user *User, decoder AudioDecod
 		return
 	}
 
+	if log.Enabled(log.LevelDebug) && (pktNum <= 3 || pktNum%1000 == 0) {
 		log.Debug("UDP15 #%d: Opus OK for %s, pcm_samples=%d", pktNum, user.Name, len(pcm))
+	}
 	user.audioSequence = frameNum
 	user.audioSequenceValid = true
 	user.audioFrameStep = audioFrameStep(len(pcm))
@@ -846,6 +868,12 @@ func (c *Client) decodeAndDispatch(pktNum uint64, user *User, decoder AudioDecod
 		event.X, event.Y, event.Z = position[0], position[1], position[2]
 	}
 	c.dispatchAudio(user, &event)
+	if terminator {
+		decoder.Reset()
+		user.audioSequenceValid = false
+		user.audioFrameStep = 0
+		c.dispatchAudio(user, &AudioPacket{Client: c, Sender: user, Terminator: true})
+	}
 }
 
 // missingAudioPackets returns the number of whole packets absent from a

gumble/gumble/udp15_terminator_test.go

@@ -0,0 +1,37 @@
+package gumble
+
+import "testing"
+
+type terminatorDecoder struct{ resets int }
+
+func (d *terminatorDecoder) ID() int                             { return audioCodecIDOpus }
+func (d *terminatorDecoder) Decode([]byte, int) ([]int16, error) { return nil, nil }
+func (d *terminatorDecoder) Reset()                              { d.resets++ }
+
+type terminatorListener struct{ packets chan *AudioPacket }
+
+func (l *terminatorListener) OnAudioStream(e *AudioStreamEvent) {
+	go func() { l.packets <- <-e.C }()
+}
+
+func TestUDP15EmptyTerminatorResetsAudioListeners(t *testing.T) {
+	decoder := &terminatorDecoder{}
+	listener := &terminatorListener{packets: make(chan *AudioPacket, 1)}
+	config := NewConfig()
+	config.AttachAudio(listener)
+	user := &User{Session: 1, Name: "speaker", decoder: decoder, audioSequenceValid: true}
+	client := &Client{Config: config, Users: Users{user.Session: user}}
+
+	client.dispatchOpus15(1, user.Session, 0, nil, true, 0, nil, 0)
+
+	packet := <-listener.packets
+	if !packet.Terminator {
+		t.Fatal("empty UDP terminator was not delivered to audio listeners")
+	}
+	if packet.AudioBuffer != nil {
+		t.Fatalf("terminator carried unexpected audio: %v", packet.AudioBuffer)
+	}
+	if decoder.resets != 1 || user.audioSequenceValid {
+		t.Fatalf("terminator did not reset decoder state: resets=%d valid=%v", decoder.resets, user.audioSequenceValid)
+	}
+}

gumble/gumble/udp15_test.go

@@ -32,6 +32,75 @@ func TestAdvanceIV(t *testing.T) {
 }
 
 // Regression coverage for the native IV carry path at the 255->256 wrap.
+func TestCryptState15DecryptsAfterMissedPackets(t *testing.T) {
+	key := mustDecodeHex("93360b0f86a926c4561563469026eb94")
+	nonce := mustDecodeHex("10000000000000000000000000000000")
+	out, in := &cryptState15{}, &cryptState15{}
+	if err := out.setup15(key, nonce, nonce); err != nil {
+		t.Fatal(err)
+	}
+	if err := in.setup15(key, nonce, nonce); err != nil {
+		t.Fatal(err)
+	}
+
+	first, err := out.encrypt15([]byte("first"))
+	if err != nil {
+		t.Fatal(err)
+	}
+	if _, err := in.decrypt15(first); err != nil {
+		t.Fatal(err)
+	}
+	for i := 0; i < 3; i++ {
+		if _, err := out.encrypt15([]byte("dropped")); err != nil {
+			t.Fatal(err)
+		}
+	}
+	last, err := out.encrypt15([]byte("after loss"))
+	if err != nil {
+		t.Fatal(err)
+	}
+	plain, err := in.decrypt15(last)
+	if err != nil || !bytes.Equal(plain, []byte("after loss")) {
+		t.Fatalf("decrypt after missed packets = %q, %v", plain, err)
+	}
+}
+
+func TestCryptState15DecryptsAfterMissedPacketsAcrossIVByteWrap(t *testing.T) {
+	key := mustDecodeHex("93360b0f86a926c4561563469026eb94")
+	nonce := mustDecodeHex("fa000000000000000000000000000000")
+	out, in := &cryptState15{}, &cryptState15{}
+	if err := out.setup15(key, nonce, nonce); err != nil {
+		t.Fatal(err)
+	}
+	if err := in.setup15(key, nonce, nonce); err != nil {
+		t.Fatal(err)
+	}
+
+	first, err := out.encrypt15([]byte("before wrap"))
+	if err != nil {
+		t.Fatal(err)
+	}
+	if _, err := in.decrypt15(first); err != nil {
+		t.Fatal(err)
+	}
+	for i := 0; i < 6; i++ {
+		if _, err := out.encrypt15([]byte("dropped")); err != nil {
+			t.Fatal(err)
+		}
+	}
+	last, err := out.encrypt15([]byte("after wrap"))
+	if err != nil {
+		t.Fatal(err)
+	}
+	plain, err := in.decrypt15(last)
+	if err != nil || !bytes.Equal(plain, []byte("after wrap")) {
+		t.Fatalf("decrypt after missed packets across IV wrap = %q, %v", plain, err)
+	}
+	if in.decryptIV[0] != 2 || in.decryptIV[1] != 1 {
+		t.Fatalf("unexpected IV after wrapped loss: %x", in.decryptIV[:2])
+	}
+}
+
 func TestCryptState15DecryptsAcrossIVByteWrap(t *testing.T) {
 	key := mustDecodeHex("93360b0f86a926c4561563469026eb94")
 	clientNonce := mustDecodeHex("ff000000000000000000000000000000")

gumble/gumbleopenal/stream.go

@@ -52,14 +52,22 @@ const recorderOutgoingSource uint32 = ^uint32(0)
 
 const (
 	maxBufferSize    = 11520 // Max frame size (2880) * bytes per stereo sample (4)
-	jitterMinPackets = 3
+	jitterMaxPackets = 50
-	jitterMaxPackets = 10
 )
 
-// jitterPlaybackReady holds the initial playout delay only once. Requiring
+// jitterPlaybackReady holds the requested initial playout delay only once.
-// the minimum on every packet drains and refills the renderer in bursts.
+// Requiring the delay on every packet drains and refills the renderer in bursts.
-func jitterPlaybackReady(started bool, buffered int) bool {
+func jitterPlaybackReady(started bool, buffered, target time.Duration) bool {
-	return started || buffered >= jitterMinPackets
+	return started || buffered >= target
+}
+
+func audioPacketDuration(packet *gumble.AudioPacket) time.Duration {
+	if packet == nil || len(packet.AudioBuffer) == 0 {
+		return 0
+	}
+	// Opus decoders deliver interleaved stereo PCM to this renderer.
+	frames := len(packet.AudioBuffer) / gumble.AudioChannels
+	return time.Duration(frames) * time.Second / gumble.AudioSampleRate
 }
 
 var (
@@ -484,8 +492,17 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 		// Jitter buffer: collects incoming packets, reorders by
 		// sequence number, and releases them after a small initial delay.
 		var jitterBuf []*gumble.AudioPacket
+		var jitterDuration time.Duration
 		var jitterNextSeq int64
 		var jitterInit, jitterStarted bool
+		var jitterDrainLogCounter, jitterAnomalyLogCounter int
+		resetJitter := func() {
+			jitterBuf = nil
+			jitterDuration = 0
+			jitterNextSeq = 0
+			jitterInit = false
+			jitterStarted = false
+		}
 
 		// insertSorted inserts a packet into the jitter buffer sorted
 		// by sequence number.
@@ -506,6 +523,7 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 			jitterBuf = append(jitterBuf, nil)
 			copy(jitterBuf[i+1:], jitterBuf[i:])
 			jitterBuf[i] = p
+			jitterDuration += audioPacketDuration(p)
 		}
 
 		// popNext removes and returns the packet with the expected next
@@ -516,6 +534,7 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 			}
 			p := jitterBuf[0]
 			jitterBuf = jitterBuf[1:]
+			jitterDuration -= audioPacketDuration(p)
 			// Frame numbers are Mumble timestamps in 10 ms units.
 			// Compute the actual step from the PCM sample count so we
 			// never skip a legitimate gap.
@@ -540,6 +559,14 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 		}
 
 		for packet := range e.C {
+			// A talk burst may restart its frame numbers from zero. Reset before
+			// testing local mute so an unmute cannot retain the previous burst's
+			// timestamp and discard the new burst as permanently late.
+			if packet.Terminator {
+				resetJitter()
+				continue
+			}
+
 			// Skip processing if user is locally muted
 			if e.User.LocallyMuted() {
 				continue
@@ -556,14 +583,13 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 
 			// Hold only the initial packets. Once playback starts, drain every
 			// ready packet so the renderer is fed continuously rather than in
-			// bursts of jitterMinPackets packets.
+			// bursts of packets.
-			if !jitterPlaybackReady(jitterStarted, len(jitterBuf)) {
+			if !jitterPlaybackReady(jitterStarted, jitterDuration, e.Client.Config.IncomingAudioBuffer) {
 				continue
 			}
 			jitterStarted = true
 
 			// Drain all packets that are ready (in sequence order)
-			drainedCount := 0
 			for {
 				pkt := popNext()
 				if pkt == nil {
@@ -571,16 +597,23 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 						if jitterBuf[0].Sequence < jitterNextSeq {
 							// Late or duplicate: discard so it doesn't
 							// permanently block the drain loop.
+							jitterAnomalyLogCounter++
+							if jitterAnomalyLogCounter <= 3 || jitterAnomalyLogCounter%1000 == 0 {
 								log.Debug("jitter: discarding late seq=%d for %s (next=%d buf=%d)",
 									jitterBuf[0].Sequence, e.User.Name, jitterNextSeq, len(jitterBuf))
+							}
+							jitterDuration -= audioPacketDuration(jitterBuf[0])
 							jitterBuf = jitterBuf[1:]
 							continue
 						}
 						if jitterBuf[0].Sequence > jitterNextSeq {
 							// Gap in sequence: skip ahead so we don't
 							// wait forever for a lost packet.
+							jitterAnomalyLogCounter++
+							if jitterAnomalyLogCounter <= 3 || jitterAnomalyLogCounter%1000 == 0 {
 								log.Debug("jitter: seq gap for %s, skipping from %d to %d (buf=%d)",
 									e.User.Name, jitterNextSeq, jitterBuf[0].Sequence, len(jitterBuf))
+							}
 							jitterNextSeq = jitterBuf[0].Sequence
 							continue
 						}
@@ -589,8 +622,8 @@ func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
 					}
 					break
 				}
-				drainedCount++
+				jitterDrainLogCounter++
-				if drainedCount <= 3 || drainedCount%50 == 0 {
+				if jitterDrainLogCounter <= 3 || jitterDrainLogCounter%1000 == 0 {
 					log.Debug("jitter: draining seq=%d for %s (buf=%d emptyBufs=%d)",
 						pkt.Sequence, e.User.Name, len(jitterBuf), len(emptyBufs))
 				}

gumble/gumbleopenal/stream_regression_test.go

@@ -4,8 +4,10 @@ import (
 	"errors"
 	"strings"
 	"testing"
+	"time"
 
 	"git.stormux.org/storm/barnard/gumble/go-openal/openal"
+	"git.stormux.org/storm/barnard/gumble/gumble"
 )
 
 // Regression: audio cleanup could send a final render command after Destroy
@@ -49,17 +51,24 @@ func TestStopSourceWaitsForWorker(t *testing.T) {
 }
 
 func TestJitterPlaybackDelayAppliesOnlyAtStartup(t *testing.T) {
-	if jitterPlaybackReady(false, jitterMinPackets-1) {
+	if jitterPlaybackReady(false, 20*time.Millisecond, 40*time.Millisecond) {
 		t.Fatal("jitter playback started before initial buffer filled")
 	}
-	if !jitterPlaybackReady(false, jitterMinPackets) {
+	if !jitterPlaybackReady(false, 40*time.Millisecond, 40*time.Millisecond) {
 		t.Fatal("jitter playback did not start after initial buffer filled")
 	}
-	if !jitterPlaybackReady(true, 1) {
+	if !jitterPlaybackReady(true, 0, 40*time.Millisecond) {
 		t.Fatal("jitter playback paused while refilling after startup")
 	}
 }
 
+func TestAudioPacketDurationUsesStereoFrameCount(t *testing.T) {
+	packet := &gumble.AudioPacket{AudioBuffer: make(gumble.AudioBuffer, 2*gumble.AudioDefaultFrameSize)}
+	if got := audioPacketDuration(packet); got != 10*time.Millisecond {
+		t.Fatalf("audioPacketDuration = %v, want 10ms", got)
+	}
+}
+
 func TestRenderRejectsWorkAfterShutdown(t *testing.T) {
 	s := &Stream{renderClosed: true}
 	called := false

main.go

@@ -11,6 +11,7 @@ import (
 	"net/http"
 	"os"
 	"strings"
+	"time"
 
 	barnlog "git.stormux.org/storm/barnard/log"
 
@@ -84,6 +85,8 @@ func main() {
 	configSet := false
 	certificateSet := false
 	buffers := flag.Int("buffers", 16, "number of audio buffers to use")
+	audioInterval := flag.Int("audio-interval", 10, "outgoing audio packet duration in ms (10, 20, 40, or 60)")
+	jitterBuffer := flag.Int("jitter-buffer", 40, "incoming per-user audio buffer in ms (0, 20, 40, or 60)")
 	profile := flag.Bool("profile", false, "add http server to serve profiles")
 	noiseSuppressionEnabled := flag.Bool("noise-suppression", false, "enable noise suppression for microphone input")
 	autoTransmit := flag.Bool("auto-transmit", false, "start transmitting immediately on connect")
@@ -94,6 +97,14 @@ func main() {
 	logFile := flag.String("logfile", "", "write logs to this file (logging is disabled when omitted)")
 
 	flag.Parse()
+	selectedAudioInterval, err := audioIntervalDuration(*audioInterval)
+	if err != nil {
+		handle_raw_error(err)
+	}
+	selectedJitterBuffer, err := jitterBufferDuration(*jitterBuffer)
+	if err != nil {
+		handle_raw_error(err)
+	}
 
 	// Set up logging
 	var level barnlog.Level
@@ -192,6 +203,8 @@ func main() {
 		NoiseSuppressor: noise.NewSuppressor(),
 	}
 	b.Config.Buffers = *buffers
+	b.Config.AudioInterval = selectedAudioInterval
+	b.Config.IncomingAudioBuffer = selectedJitterBuffer
 	b.Config.DisableUDP = *tcpOnly
 
 	b.Hotkeys = b.UserConfig.GetHotkeys()
@@ -238,6 +251,30 @@ func main() {
 	handle_error(&b)
 }
 
+// audioIntervalDuration converts the packet duration requested at startup to
+// one of the Opus durations supported by Mumble.
+func audioIntervalDuration(milliseconds int) (time.Duration, error) {
+	interval := time.Duration(milliseconds) * time.Millisecond
+	switch interval {
+	case 10 * time.Millisecond, 20 * time.Millisecond, 40 * time.Millisecond, 60 * time.Millisecond:
+		return interval, nil
+	default:
+		return 0, fmt.Errorf("audio interval must be 10, 20, 40, or 60 ms, got %d", milliseconds)
+	}
+}
+
+// jitterBufferDuration converts the requested incoming playout delay to a
+// supported duration. Zero starts playback without an initial safety buffer.
+func jitterBufferDuration(milliseconds int) (time.Duration, error) {
+	interval := time.Duration(milliseconds) * time.Millisecond
+	switch interval {
+	case 0, 20 * time.Millisecond, 40 * time.Millisecond, 60 * time.Millisecond:
+		return interval, nil
+	default:
+		return 0, fmt.Errorf("jitter buffer must be 0, 20, 40, or 60 ms, got %d", milliseconds)
+	}
+}
+
 // serverAddress adds Mumble's default port without corrupting an IPv6 literal.
 func serverAddress(address string) string {
 	if _, port, err := net.SplitHostPort(address); err == nil && port != "" {