Scale Opus bitrate to audio frame duration

fix.txt

@@ -176,7 +176,7 @@ Priority 2: protocol and data correctness
     terminators, positional data, and volume adjustment. Test real UDP
     fallback behavior too.
 
-22. Opus bitrate calculation assumes a 10 ms interval
+[x] 22. Opus bitrate calculation assumes a 10 ms interval
     File: gumble/opus/opus.go
     bitrate is maxDataBytes * 8 * 100. For permitted 20/40/60 ms intervals it
     is 2x/4x/6x too high. Calculate bits per frame divided by the actual

gumble/opus/opus.go

@@ -31,6 +31,7 @@ func (*generator) NewEncoder() gumble.AudioEncoder {
 	e, _ := opus.NewEncoder(gumble.AudioSampleRate, VoiceChannels, opus.AppVoIP)
 	return &Encoder{
 		Encoder:  e,
+		channels: VoiceChannels,
 	}
 }
 
@@ -40,6 +41,7 @@ func NewStereoEncoder() gumble.AudioEncoder {
 	e, _ := opus.NewEncoder(gumble.AudioSampleRate, gumble.AudioChannels, opus.AppAudio)
 	return &Encoder{
 		Encoder:  e,
+		channels: gumble.AudioChannels,
 	}
 }
 
@@ -56,18 +58,15 @@ func (*generator) NewDecoder() gumble.AudioDecoder {
 // encoder
 type Encoder struct {
 	*opus.Encoder
+	channels int
 }
 
 func (*Encoder) ID() int {
 	return ID
 }
 
-func (e *Encoder) Encode(pcm []int16, _, maxDataBytes int) ([]byte, error) {
-	// Set the encoder bitrate to match the available bandwidth per frame.
-	// The bitrate is: maxDataBytes * 8 (bits per byte) * 100 (10ms frames per second).
-	// Opus encodes best when the bitrate target is set properly rather than
-	// relying on truncation, which can produce incomplete or corrupt frames.
-	bitrate := maxDataBytes * 8 * 100
+func (e *Encoder) Encode(pcm []int16, frameSamples, maxDataBytes int) ([]byte, error) {
+	bitrate := encoderBitrate(maxDataBytes, frameSamples, e.channels)
 	if bitrate < 8000 {
 		bitrate = 8000 // Opus minimum viable bitrate for voice
 	}
@@ -81,6 +80,14 @@ func (e *Encoder) Encode(pcm []int16, _, maxDataBytes int) ([]byte, error) {
 	return buf[:n], nil
 }
 
+// encoderBitrate converts a per-frame packet budget to bits per second.
+func encoderBitrate(maxDataBytes, frameSamples, channels int) int {
+	if frameSamples <= 0 || channels <= 0 {
+		return 8000
+	}
+	return maxDataBytes * 8 * gumble.AudioSampleRate * channels / frameSamples
+}
+
 func (e *Encoder) Reset() {
 	_ = e.Encoder.Reset()
 }

gumble/opus/opus_test.go

@@ -0,0 +1,16 @@
+package opus
+
+import "testing"
+
+// Regression: the encoder always assumed ten millisecond frames, causing the
+// bitrate for 20/40/60 ms packets to be 2/4/6 times their actual budget.
+func TestEncoderBitrateUsesFrameDuration(t *testing.T) {
+	const budget = 100
+	for _, frameSamples := range []int{480, 960, 1920, 2880} {
+		got := encoderBitrate(budget, frameSamples, 1)
+		want := budget * 8 * 48000 / frameSamples
+		if got != want {
+			t.Fatalf("%d samples: got %d, want %d", frameSamples, got, want)
+		}
+	}
+}