Fix audio quality and cleanup issues in audio implementation

- Fix OPUS_BANDWIDTH constant from 1104 to 1105 (fullband 20kHz vs superwideband 12kHz)
- Fix PacketLossPerc default from 0 to 20 to match C layer default
- Add early return in FEC recovery when pcm_frames <= 0 to avoid zero-frame ALSA writes
- Remove unused InputRelay fields (source, ctx, cancel) and unused NewInputRelay parameter
- Add async cleanup on timeout in SetAudioOutputEnabled/SetAudioInputEnabled

audio.go

@@ -160,7 +160,7 @@ func startInputAudioUnderMutex(alsaPlaybackDevice string) error {
 	}
 
 	newSource := audio.NewCgoInputSource(alsaPlaybackDevice, getAudioConfig())
-	newRelay := audio.NewInputRelay(&newSource)
+	newRelay := audio.NewInputRelay()
 
 	if err := newRelay.Start(); err != nil {
 		audioLogger.Error().Err(err).Str("alsaPlaybackDevice", alsaPlaybackDevice).Msg("Failed to start input relay")
@@ -275,6 +275,7 @@ func SetAudioOutputEnabled(enabled bool) error {
 		case <-time.After(5 * time.Second):
 			audioLogger.Error().Msg("Audio output start timed out after 5 seconds")
 			audioOutputEnabled.Store(false) // Revert state on timeout
+			go stopOutputAudio()            // Clean up any partial initialization asynchronously
 			return fmt.Errorf("audio output start timed out after 5 seconds")
 		}
 	}
@@ -307,6 +308,7 @@ func SetAudioInputEnabled(enabled bool) error {
 		case <-time.After(5 * time.Second):
 			audioLogger.Error().Msg("Audio input start timed out after 5 seconds")
 			audioInputEnabled.Store(false) // Revert state on timeout
+			go stopInputAudio()            // Clean up any partial initialization asynchronously
 			return fmt.Errorf("audio input start timed out after 5 seconds")
 		}
 	}

internal/audio/c/audio.c

@@ -88,7 +88,7 @@ static uint16_t max_packet_size = 1500;
 #define OPUS_VBR 1                    // Variable bitrate mode enabled
 #define OPUS_VBR_CONSTRAINT 1         // Constrained VBR maintains bitrate ceiling
 #define OPUS_SIGNAL_TYPE 3002         // OPUS_SIGNAL_MUSIC (optimized for music/audio content)
-#define OPUS_BANDWIDTH 1104           // OPUS_BANDWIDTH_FULLBAND (0-20kHz frequency range)
+#define OPUS_BANDWIDTH 1105           // OPUS_BANDWIDTH_FULLBAND (0-20kHz frequency range)
 #define OPUS_LSB_DEPTH 16             // 16-bit PCM sample depth (S16_LE format)
 
 static uint8_t opus_dtx_enabled = 1;
@@ -1068,11 +1068,16 @@ __attribute__((hot)) int jetkvm_audio_decode_write(void * __restrict__ opus_buf,
 			fprintf(stdout, "INFO: playback: FEC recovered %d frames\n", pcm_frames);
 			fflush(stdout);
 		} else {
-			fprintf(stderr, "WARN: playback: FEC returned 0 frames (silence)\n");
-			fflush(stderr);
+			pthread_mutex_unlock(&playback_mutex);
+			return 0;  // FEC returned no frames, nothing to write
 		}
 	}
 
+	if (__builtin_expect(pcm_frames <= 0, 0)) {
+		pthread_mutex_unlock(&playback_mutex);
+		return 0;  // Nothing to write
+	}
+
 retry_write:
 	if (__builtin_expect(atomic_load(&playback_stop_requested), 0)) {
 		pthread_mutex_unlock(&playback_mutex);

internal/audio/relay.go

@@ -142,21 +142,14 @@ func (r *OutputRelay) relayLoop() {
 }
 
 type InputRelay struct {
-	source  *AudioSource
-	ctx     context.Context
-	cancel  context.CancelFunc
 	logger  zerolog.Logger
 	running atomic.Bool
 }
 
-func NewInputRelay(source *AudioSource) *InputRelay {
-	ctx, cancel := context.WithCancel(context.Background())
+func NewInputRelay() *InputRelay {
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-input-relay").Logger()
 
 	return &InputRelay{
-		source: source,
-		ctx:    ctx,
-		cancel: cancel,
 		logger: logger,
 	}
 }
@@ -175,6 +168,5 @@ func (r *InputRelay) Stop() {
 		return
 	}
 
-	r.cancel()
 	r.logger.Debug().Msg("input relay stopped")
 }

internal/audio/source.go

@@ -20,7 +20,7 @@ func DefaultAudioConfig() AudioConfig {
 		BufferPeriods:  12,
 		DTXEnabled:     true,
 		FECEnabled:     true,
-		PacketLossPerc: 0,
+		PacketLossPerc: 20,
 	}
 }