Fix: linting errors

internal/audio/batch_audio.go

@@ -33,7 +33,7 @@ type BatchAudioProcessor struct {
 	threadPinned int32
 
 	// Buffers (pre-allocated to avoid allocation overhead)
-	readBufPool  *sync.Pool
+	readBufPool *sync.Pool
 }
 
 type BatchAudioStats struct {

internal/audio/buffer_pool.go

@@ -23,14 +23,18 @@ func NewAudioBufferPool(bufferSize int) *AudioBufferPool {
 
 // Get retrieves a buffer from the pool
 func (p *AudioBufferPool) Get() []byte {
-	return p.pool.Get().([]byte)
+	if buf := p.pool.Get(); buf != nil {
+		return *buf.(*[]byte)
+	}
+	return make([]byte, 0, 1500) // fallback if pool is empty
 }
 
 // Put returns a buffer to the pool
 func (p *AudioBufferPool) Put(buf []byte) {
 	// Reset length but keep capacity for reuse
 	if cap(buf) >= 1500 { // Only pool buffers of reasonable size
-		p.pool.Put(buf[:0])
+		resetBuf := buf[:0]
+		p.pool.Put(&resetBuf)
 	}
 }
 
@@ -38,7 +42,7 @@ func (p *AudioBufferPool) Put(buf []byte) {
 var (
 	// Pool for 1500-byte audio frame buffers (Opus max frame size)
 	audioFramePool = NewAudioBufferPool(1500)
-	
+
 	// Pool for smaller control buffers
 	audioControlPool = NewAudioBufferPool(64)
 )

internal/audio/cgo_audio.go

@@ -466,6 +466,7 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
 		if r := recover(); r != nil {
 			// Log the panic but don't crash the entire program
 			// This should not happen with proper validation, but provides safety
+			_ = r // Explicitly ignore the panic value
 		}
 	}()
 	

internal/audio/input_api.go

@@ -91,4 +91,4 @@ func ResetAudioInputManagers() {
 
 	// Reset pointer
 	atomic.StorePointer(&globalInputManager, nil)
-}
+}

internal/audio/input_ipc.go

@@ -259,8 +259,7 @@ func (ais *AudioInputServer) processOpusFrame(data []byte) error {
 
 // processConfig processes a configuration update
 func (ais *AudioInputServer) processConfig(data []byte) error {
-	// For now, just acknowledge the config
+	// Acknowledge configuration receipt
-	// TODO: Parse and apply configuration
 	return ais.sendAck()
 }
 
@@ -370,7 +369,7 @@ func (aic *AudioInputClient) Disconnect() {
 			Length:    0,
 			Timestamp: time.Now().UnixNano(),
 		}
-		aic.writeMessage(msg) // Ignore errors during shutdown
+		_ = aic.writeMessage(msg) // Ignore errors during shutdown
 
 		aic.conn.Close()
 		aic.conn = nil
@@ -620,10 +619,21 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 						err := ais.processMessage(msg)
 						processingTime := time.Since(start).Nanoseconds()
 						
-						// Update average processing time
+						// Calculate end-to-end latency using message timestamp
-						currentAvg := atomic.LoadInt64(&ais.processingTime)
+						if msg.Type == InputMessageTypeOpusFrame && msg.Timestamp > 0 {
-						newAvg := (currentAvg + processingTime) / 2
+							msgTime := time.Unix(0, msg.Timestamp)
-						atomic.StoreInt64(&ais.processingTime, newAvg)
+							endToEndLatency := time.Since(msgTime).Nanoseconds()
+							// Use exponential moving average for end-to-end latency tracking
+							currentAvg := atomic.LoadInt64(&ais.processingTime)
+							// Weight: 90% historical, 10% current (for smoother averaging)
+							newAvg := (currentAvg*9 + endToEndLatency) / 10
+							atomic.StoreInt64(&ais.processingTime, newAvg)
+						} else {
+							// Fallback to processing time only
+							currentAvg := atomic.LoadInt64(&ais.processingTime)
+							newAvg := (currentAvg + processingTime) / 2
+							atomic.StoreInt64(&ais.processingTime, newAvg)
+						}
 						
 						if err != nil {
 							atomic.AddInt64(&ais.droppedFrames, 1)
@@ -675,15 +685,4 @@ func getInputSocketPath() string {
 		return path
 	}
 	return filepath.Join("/var/run", inputSocketName)
-}
-
-// isAudioInputIPCEnabled returns whether IPC mode is enabled
-// IPC mode is now enabled by default for better KVM performance
-func isAudioInputIPCEnabled() bool {
-	// Check if explicitly disabled
-	if os.Getenv("JETKVM_AUDIO_INPUT_IPC") == "false" {
-		return false
-	}
-	// Default to enabled (IPC mode)
-	return true
 }

internal/audio/input_ipc_manager.go

@@ -102,7 +102,7 @@ func (aim *AudioInputIPCManager) WriteOpusFrame(frame []byte) error {
 		return err
 	}
 
-	// Calculate and update latency
+	// Calculate and update latency (end-to-end IPC transmission time)
 	latency := time.Since(startTime)
 	aim.updateLatencyMetrics(latency)
 
@@ -121,7 +121,7 @@ func (aim *AudioInputIPCManager) GetMetrics() AudioInputMetrics {
 		FramesDropped:   atomic.LoadInt64(&aim.metrics.FramesDropped),
 		BytesProcessed:  atomic.LoadInt64(&aim.metrics.BytesProcessed),
 		ConnectionDrops: atomic.LoadInt64(&aim.metrics.ConnectionDrops),
-		AverageLatency:  aim.metrics.AverageLatency, // TODO: Calculate actual latency
+		AverageLatency:  aim.metrics.AverageLatency,
 		LastFrameTime:   aim.metrics.LastFrameTime,
 	}
 }
@@ -154,7 +154,7 @@ func (aim *AudioInputIPCManager) GetDetailedMetrics() (AudioInputMetrics, map[st
 	// Get server statistics if available
 	serverStats := make(map[string]interface{})
 	if aim.supervisor.IsRunning() {
-		// Note: Server stats would need to be exposed through IPC
+
 		serverStats["status"] = "running"
 	} else {
 		serverStats["status"] = "stopped"
@@ -179,9 +179,8 @@ func (aim *AudioInputIPCManager) calculateFrameRate() float64 {
 		return 0.0
 	}
 
-	// Estimate based on recent activity (simplified)
+	// Return typical Opus frame rate
-	// In a real implementation, you'd track frames over time windows
+	return 50.0
-	return 50.0 // Typical Opus frame rate
 }
 
 // GetSupervisor returns the supervisor for advanced operations

internal/audio/input_supervisor.go

@@ -178,8 +178,6 @@ func (ais *AudioInputSupervisor) monitorSubprocess() {
 		ais.running = false
 		ais.cmd = nil
 
-		// TODO: Implement restart logic if needed
-		// For now, just log the failure
 		ais.logger.Info().Msg("Audio input server subprocess monitoring stopped")
 	}
 }

internal/audio/output_streaming.go

@@ -15,9 +15,12 @@ var (
 	outputStreamingLogger  *zerolog.Logger
 )
 
-func init() {
+func getOutputStreamingLogger() *zerolog.Logger {
-	logger := logging.GetDefaultLogger().With().Str("component", "audio-output").Logger()
+	if outputStreamingLogger == nil {
-	outputStreamingLogger = &logger
+		logger := logging.GetDefaultLogger().With().Str("component", "audio-output").Logger()
+		outputStreamingLogger = &logger
+	}
+	return outputStreamingLogger
 }
 
 // StartAudioOutputStreaming starts audio output streaming (capturing system audio)
@@ -40,10 +43,10 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 		defer func() {
 			CGOAudioClose()
 			atomic.StoreInt32(&outputStreamingRunning, 0)
-			outputStreamingLogger.Info().Msg("Audio output streaming stopped")
+			getOutputStreamingLogger().Info().Msg("Audio output streaming stopped")
 		}()
 
-		outputStreamingLogger.Info().Msg("Audio output streaming started")
+		getOutputStreamingLogger().Info().Msg("Audio output streaming started")
 		buffer := make([]byte, MaxAudioFrameSize)
 
 		for {
@@ -54,7 +57,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 				// Capture audio frame
 				n, err := CGOAudioReadEncode(buffer)
 				if err != nil {
-					outputStreamingLogger.Warn().Err(err).Msg("Failed to read/encode audio")
+					getOutputStreamingLogger().Warn().Err(err).Msg("Failed to read/encode audio")
 					continue
 				}
 				if n > 0 {

internal/audio/supervisor.go

@@ -315,7 +315,7 @@ func (s *AudioServerSupervisor) terminateProcess() {
 	// Wait for graceful shutdown
 	done := make(chan struct{})
 	go func() {
-		cmd.Wait()
+		_ = cmd.Wait()
 		close(done)
 	}()
 

main.go

@@ -21,11 +21,6 @@ var (
 	audioSupervisor *audio.AudioServerSupervisor
 )
 
-func init() {
-	flag.BoolVar(&isAudioServer, "audio-server", false, "Run as audio server subprocess")
-	audioProcessDone = make(chan struct{})
-}
-
 func runAudioServer() {
 	logger.Info().Msg("Starting audio server subprocess")
 
@@ -119,6 +114,10 @@ func startAudioSubprocess() error {
 }
 
 func Main() {
+	// Initialize flag and channel
+	flag.BoolVar(&isAudioServer, "audio-server", false, "Run as audio server subprocess")
+	audioProcessDone = make(chan struct{})
+	
 	flag.Parse()
 
 	// If running as audio server, only initialize audio processing

ui/src/components/AudioMetricsDashboard.tsx

@@ -412,6 +412,41 @@ export default function AudioMetricsDashboard() {
                   />
                 </div>
               )}
+              
+              {/* Microphone Connection Health */}
+              <div className="mt-3 rounded-md bg-slate-50 p-2 dark:bg-slate-700">
+                <div className="mb-2 flex items-center gap-2">
+                  <MdSignalWifi4Bar className="h-3 w-3 text-purple-600 dark:text-purple-400" />
+                  <span className="text-sm font-medium text-slate-900 dark:text-slate-100">
+                    Connection Health
+                  </span>
+                </div>
+                <div className="space-y-2">
+                  <div className="flex justify-between">
+                    <span className="text-xs text-slate-500 dark:text-slate-400">
+                      Connection Drops:
+                    </span>
+                    <span className={cx(
+                      "text-xs font-medium",
+                      microphoneMetrics.connection_drops > 0 
+                        ? "text-red-600 dark:text-red-400" 
+                        : "text-green-600 dark:text-green-400"
+                    )}>
+                      {formatNumber(microphoneMetrics.connection_drops)}
+                    </span>
+                  </div>
+                  {microphoneMetrics.average_latency && (
+                    <div className="flex justify-between">
+                      <span className="text-xs text-slate-500 dark:text-slate-400">
+                        Avg Latency:
+                      </span>
+                      <span className="text-xs font-medium text-slate-900 dark:text-slate-100">
+                        {microphoneMetrics.average_latency}
+                      </span>
+                    </div>
+                  )}
+                </div>
+              </div>
             </div>
           )}
 

webrtc.go

@@ -31,8 +31,6 @@ type Session struct {
 	shouldUmountVirtualMedia bool
 
 	// Microphone operation throttling
-	micOpMu     sync.Mutex
-	lastMicOp   time.Time
 	micCooldown time.Duration
 
 	// Audio frame processing