audio_handlers.go

@@ -22,14 +22,6 @@ func initAudioControlService() {
 		audio.SetCurrentSessionCallback(func() audio.AudioTrackWriter {
 			return GetCurrentSessionAudioTrack()
 		})
-
-		// Set up callback for audio relay to replace WebRTC audio track
-		audio.SetTrackReplacementCallback(func(newTrack audio.AudioTrackWriter) error {
-			if track, ok := newTrack.(*webrtc.TrackLocalStaticSample); ok {
-				return ReplaceCurrentSessionAudioTrack(track)
-			}
-			return nil
-		})
 	}
 }
 
@@ -100,60 +92,6 @@ func ConnectRelayToCurrentSession() error {
 	return nil
 }
 
-// ReplaceCurrentSessionAudioTrack replaces the audio track in the current WebRTC session
-func ReplaceCurrentSessionAudioTrack(newTrack *webrtc.TrackLocalStaticSample) error {
-	if currentSession == nil {
-		return nil // No session to update
-	}
-
-	err := currentSession.ReplaceAudioTrack(newTrack)
-	if err != nil {
-		logger.Error().Err(err).Msg("failed to replace audio track in current session")
-		return err
-	}
-
-	logger.Info().Msg("successfully replaced audio track in current session")
-	return nil
-}
-
-// SetAudioQuality is a global helper to set audio output quality
-func SetAudioQuality(quality audio.AudioQuality) error {
-	initAudioControlService()
-	audioControlService.SetAudioQuality(quality)
-	return nil
-}
-
-// SetMicrophoneQuality is a global helper to set microphone quality
-func SetMicrophoneQuality(quality audio.AudioQuality) error {
-	initAudioControlService()
-	audioControlService.SetMicrophoneQuality(quality)
-	return nil
-}
-
-// GetAudioQualityPresets is a global helper to get available audio quality presets
-func GetAudioQualityPresets() map[audio.AudioQuality]audio.AudioConfig {
-	initAudioControlService()
-	return audioControlService.GetAudioQualityPresets()
-}
-
-// GetMicrophoneQualityPresets is a global helper to get available microphone quality presets
-func GetMicrophoneQualityPresets() map[audio.AudioQuality]audio.AudioConfig {
-	initAudioControlService()
-	return audioControlService.GetMicrophoneQualityPresets()
-}
-
-// GetCurrentAudioQuality is a global helper to get current audio quality configuration
-func GetCurrentAudioQuality() audio.AudioConfig {
-	initAudioControlService()
-	return audioControlService.GetCurrentAudioQuality()
-}
-
-// GetCurrentMicrophoneQuality is a global helper to get current microphone quality configuration
-func GetCurrentMicrophoneQuality() audio.AudioConfig {
-	initAudioControlService()
-	return audioControlService.GetCurrentMicrophoneQuality()
-}
-
 // handleAudioMute handles POST /audio/mute requests
 func handleAudioMute(c *gin.Context) {
 	type muteReq struct {
@@ -264,8 +202,10 @@ func handleAudioStatus(c *gin.Context) {
 
 // handleAudioQuality handles GET requests for audio quality presets
 func handleAudioQuality(c *gin.Context) {
-	presets := GetAudioQualityPresets()
+	initAudioControlService()
-	current := GetCurrentAudioQuality()
+
+	presets := audioControlService.GetAudioQualityPresets()
+	current := audioControlService.GetCurrentAudioQuality()
 
 	c.JSON(200, gin.H{
 		"presets": presets,
@@ -284,24 +224,16 @@ func handleSetAudioQuality(c *gin.Context) {
 		return
 	}
 
-	// Check if audio output is active before attempting quality change
+	initAudioControlService()
-	// This prevents race conditions where quality changes are attempted before initialization
-	if !IsAudioOutputActive() {
-		c.JSON(503, gin.H{"error": "audio output not active - please wait for initialization to complete"})
-		return
-	}
 
 	// Convert int to AudioQuality type
 	quality := audio.AudioQuality(req.Quality)
 
-	// Set the audio quality using global convenience function
+	// Set the audio quality
-	if err := SetAudioQuality(quality); err != nil {
+	audioControlService.SetAudioQuality(quality)
-		c.JSON(500, gin.H{"error": err.Error()})
-		return
-	}
 
 	// Return the updated configuration
-	current := GetCurrentAudioQuality()
+	current := audioControlService.GetCurrentAudioQuality()
 	c.JSON(200, gin.H{
 		"success": true,
 		"config":  current,
@@ -310,9 +242,9 @@ func handleSetAudioQuality(c *gin.Context) {
 
 // handleMicrophoneQuality handles GET requests for microphone quality presets
 func handleMicrophoneQuality(c *gin.Context) {
-	presets := GetMicrophoneQualityPresets()
+	initAudioControlService()
-	current := GetCurrentMicrophoneQuality()
+	presets := audioControlService.GetMicrophoneQualityPresets()
-
+	current := audioControlService.GetCurrentMicrophoneQuality()
 	c.JSON(200, gin.H{
 		"presets": presets,
 		"current": current,
@@ -326,22 +258,21 @@ func handleSetMicrophoneQuality(c *gin.Context) {
 	}
 
 	if err := c.ShouldBindJSON(&req); err != nil {
-		c.JSON(400, gin.H{"error": err.Error()})
+		c.JSON(http.StatusBadRequest, gin.H{"error": err.Error()})
 		return
 	}
 
+	initAudioControlService()
+
 	// Convert int to AudioQuality type
 	quality := audio.AudioQuality(req.Quality)
 
-	// Set the microphone quality using global convenience function
+	// Set the microphone quality
-	if err := SetMicrophoneQuality(quality); err != nil {
+	audioControlService.SetMicrophoneQuality(quality)
-		c.JSON(500, gin.H{"error": err.Error()})
-		return
-	}
 
 	// Return the updated configuration
-	current := GetCurrentMicrophoneQuality()
+	current := audioControlService.GetCurrentMicrophoneQuality()
-	c.JSON(200, gin.H{
+	c.JSON(http.StatusOK, gin.H{
 		"success": true,
 		"config":  current,
 	})

internal/audio/adaptive_buffer.go

@@ -57,25 +57,25 @@ type AdaptiveBufferConfig struct {
 func DefaultAdaptiveBufferConfig() AdaptiveBufferConfig {
 	return AdaptiveBufferConfig{
 		// Conservative buffer sizes for 256MB RAM constraint
-		MinBufferSize:     Config.AdaptiveMinBufferSize,
+		MinBufferSize:     GetConfig().AdaptiveMinBufferSize,
-		MaxBufferSize:     Config.AdaptiveMaxBufferSize,
+		MaxBufferSize:     GetConfig().AdaptiveMaxBufferSize,
-		DefaultBufferSize: Config.AdaptiveDefaultBufferSize,
+		DefaultBufferSize: GetConfig().AdaptiveDefaultBufferSize,
 
 		// CPU thresholds optimized for single-core ARM Cortex A7 under load
-		LowCPUThreshold:  Config.LowCPUThreshold * 100,  // Below 20% CPU
+		LowCPUThreshold:  GetConfig().LowCPUThreshold * 100,  // Below 20% CPU
-		HighCPUThreshold: Config.HighCPUThreshold * 100, // Above 60% CPU (lowered to be more responsive)
+		HighCPUThreshold: GetConfig().HighCPUThreshold * 100, // Above 60% CPU (lowered to be more responsive)
 
 		// Memory thresholds for 256MB total RAM
-		LowMemoryThreshold:  Config.LowMemoryThreshold * 100,  // Below 35% memory usage
+		LowMemoryThreshold:  GetConfig().LowMemoryThreshold * 100,  // Below 35% memory usage
-		HighMemoryThreshold: Config.HighMemoryThreshold * 100, // Above 75% memory usage (lowered for earlier response)
+		HighMemoryThreshold: GetConfig().HighMemoryThreshold * 100, // Above 75% memory usage (lowered for earlier response)
 
 		// Latency targets
-		TargetLatency: Config.AdaptiveBufferTargetLatency, // Target 20ms latency
+		TargetLatency: GetConfig().AdaptiveBufferTargetLatency, // Target 20ms latency
-		MaxLatency:    Config.MaxLatencyThreshold,         // Max acceptable latency
+		MaxLatency:    GetConfig().LatencyMonitorTarget,        // Max acceptable latency
 
 		// Adaptation settings
-		AdaptationInterval: Config.BufferUpdateInterval, // Check every 500ms
+		AdaptationInterval: GetConfig().BufferUpdateInterval, // Check every 500ms
-		SmoothingFactor:    Config.SmoothingFactor,      // Moderate responsiveness
+		SmoothingFactor:    GetConfig().SmoothingFactor,      // Moderate responsiveness
 	}
 }
 
@@ -89,8 +89,9 @@ type AdaptiveBufferManager struct {
 	systemMemoryPercent     int64 // System memory percentage * 100 (atomic)
 	adaptationCount         int64 // Metrics tracking (atomic)
 
-	config AdaptiveBufferConfig
+	config         AdaptiveBufferConfig
-	logger zerolog.Logger
+	logger         zerolog.Logger
+	processMonitor *ProcessMonitor
 
 	// Control channels
 	ctx    context.Context
@@ -118,10 +119,10 @@ func NewAdaptiveBufferManager(config AdaptiveBufferConfig) *AdaptiveBufferManage
 		currentOutputBufferSize: int64(config.DefaultBufferSize),
 		config:                  config,
 		logger:                  logger,
-
+		processMonitor:          GetProcessMonitor(),
-		ctx:            ctx,
+		ctx:                     ctx,
-		cancel:         cancel,
+		cancel:                  cancel,
-		lastAdaptation: time.Now(),
+		lastAdaptation:          time.Now(),
 	}
 }
 
@@ -151,42 +152,6 @@ func (abm *AdaptiveBufferManager) GetOutputBufferSize() int {
 
 // UpdateLatency updates the current latency measurement
 func (abm *AdaptiveBufferManager) UpdateLatency(latency time.Duration) {
-	// Use exponential moving average for latency tracking
-	// Weight: 90% historical, 10% current (for smoother averaging)
-	currentAvg := atomic.LoadInt64(&abm.averageLatency)
-	newLatencyNs := latency.Nanoseconds()
-
-	if currentAvg == 0 {
-		// First measurement
-		atomic.StoreInt64(&abm.averageLatency, newLatencyNs)
-	} else {
-		// Exponential moving average
-		newAvg := (currentAvg*9 + newLatencyNs) / 10
-		atomic.StoreInt64(&abm.averageLatency, newAvg)
-	}
-
-	// Log high latency warnings only for truly problematic latencies
-	// Use a more reasonable threshold: 10ms for audio processing is concerning
-	highLatencyThreshold := 10 * time.Millisecond
-	if latency > highLatencyThreshold {
-		abm.logger.Debug().
-			Dur("latency_ms", latency/time.Millisecond).
-			Dur("threshold_ms", highLatencyThreshold/time.Millisecond).
-			Msg("High audio processing latency detected")
-	}
-}
-
-// BoostBuffersForQualityChange immediately increases buffer sizes to handle quality change bursts
-// This bypasses the normal adaptive algorithm for emergency situations
-func (abm *AdaptiveBufferManager) BoostBuffersForQualityChange() {
-	// Immediately set buffers to maximum size to handle quality change frame bursts
-	maxSize := int64(abm.config.MaxBufferSize)
-	atomic.StoreInt64(&abm.currentInputBufferSize, maxSize)
-	atomic.StoreInt64(&abm.currentOutputBufferSize, maxSize)
-
-	abm.logger.Info().
-		Int("buffer_size", int(maxSize)).
-		Msg("Boosted buffers to maximum size for quality change")
 }
 
 // adaptationLoop is the main loop that adjusts buffer sizes
@@ -234,9 +199,30 @@ func (abm *AdaptiveBufferManager) adaptationLoop() {
 // The algorithm runs periodically and only applies changes when the adaptation interval
 // has elapsed, preventing excessive adjustments that could destabilize the audio pipeline.
 func (abm *AdaptiveBufferManager) adaptBufferSizes() {
-	// Use fixed system metrics for stability
+	// Collect current system metrics
-	systemCPU := 50.0    // Assume moderate CPU usage
+	metrics := abm.processMonitor.GetCurrentMetrics()
-	systemMemory := 60.0 // Assume moderate memory usage
+	if len(metrics) == 0 {
+		return // No metrics available
+	}
+
+	// Calculate system-wide CPU and memory usage
+	totalCPU := 0.0
+	totalMemory := 0.0
+	processCount := 0
+
+	for _, metric := range metrics {
+		totalCPU += metric.CPUPercent
+		totalMemory += metric.MemoryPercent
+		processCount++
+	}
+
+	if processCount == 0 {
+		return
+	}
+
+	// Store system metrics atomically
+	systemCPU := totalCPU                               // Total CPU across all monitored processes
+	systemMemory := totalMemory / float64(processCount) // Average memory usage
 
 	atomic.StoreInt64(&abm.systemCPUPercent, int64(systemCPU*100))
 	atomic.StoreInt64(&abm.systemMemoryPercent, int64(systemMemory*100))
@@ -251,7 +237,7 @@ func (abm *AdaptiveBufferManager) adaptBufferSizes() {
 	latencyFactor := abm.calculateLatencyFactor(currentLatency)
 
 	// Combine factors with weights (CPU has highest priority for KVM coexistence)
-	combinedFactor := Config.CPUMemoryWeight*cpuFactor + Config.MemoryWeight*memoryFactor + Config.LatencyWeight*latencyFactor
+	combinedFactor := GetConfig().CPUMemoryWeight*cpuFactor + GetConfig().MemoryWeight*memoryFactor + GetConfig().LatencyWeight*latencyFactor
 
 	// Apply adaptation with smoothing
 	currentInput := float64(atomic.LoadInt64(&abm.currentInputBufferSize))
@@ -415,8 +401,8 @@ func (abm *AdaptiveBufferManager) GetStats() map[string]interface{} {
 		"input_buffer_size":     abm.GetInputBufferSize(),
 		"output_buffer_size":    abm.GetOutputBufferSize(),
 		"average_latency_ms":    float64(atomic.LoadInt64(&abm.averageLatency)) / 1e6,
-		"system_cpu_percent":    float64(atomic.LoadInt64(&abm.systemCPUPercent)) / Config.PercentageMultiplier,
+		"system_cpu_percent":    float64(atomic.LoadInt64(&abm.systemCPUPercent)) / GetConfig().PercentageMultiplier,
-		"system_memory_percent": float64(atomic.LoadInt64(&abm.systemMemoryPercent)) / Config.PercentageMultiplier,
+		"system_memory_percent": float64(atomic.LoadInt64(&abm.systemMemoryPercent)) / GetConfig().PercentageMultiplier,
 		"adaptation_count":      atomic.LoadInt64(&abm.adaptationCount),
 		"last_adaptation":       lastAdaptation,
 	}

internal/audio/batch_audio.go

@@ -82,16 +82,20 @@ type batchWriteResult struct {
 
 // NewBatchAudioProcessor creates a new batch audio processor
 func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAudioProcessor {
+	// Get cached config to avoid GetConfig() calls
+	cache := GetCachedConfig()
+	cache.Update()
+
 	// Validate input parameters with minimal overhead
 	if batchSize <= 0 || batchSize > 1000 {
-		batchSize = Config.BatchProcessorFramesPerBatch
+		batchSize = cache.BatchProcessorFramesPerBatch
 	}
 	if batchDuration <= 0 {
-		batchDuration = Config.BatchProcessingDelay
+		batchDuration = cache.BatchProcessingDelay
 	}
 
 	// Use optimized queue sizes from configuration
-	queueSize := Config.BatchProcessorMaxQueueSize
+	queueSize := cache.BatchProcessorMaxQueueSize
 	if queueSize <= 0 {
 		queueSize = batchSize * 2 // Fallback to double batch size
 	}
@@ -100,7 +104,8 @@ func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAu
 	// Pre-allocate logger to avoid repeated allocations
 	logger := logging.GetDefaultLogger().With().Str("component", "batch-audio").Logger()
 
-	frameSize := Config.MinReadEncodeBuffer
+	// Pre-calculate frame size to avoid repeated GetConfig() calls
+	frameSize := cache.GetMinReadEncodeBuffer()
 	if frameSize == 0 {
 		frameSize = 1500 // Safe fallback
 	}
@@ -115,11 +120,13 @@ func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAu
 		writeQueue:    make(chan batchWriteRequest, queueSize),
 		readBufPool: &sync.Pool{
 			New: func() interface{} {
+				// Use pre-calculated frame size to avoid GetConfig() calls
 				return make([]byte, 0, frameSize)
 			},
 		},
 		writeBufPool: &sync.Pool{
 			New: func() interface{} {
+				// Use pre-calculated frame size to avoid GetConfig() calls
 				return make([]byte, 0, frameSize)
 			},
 		},
@@ -159,13 +166,17 @@ func (bap *BatchAudioProcessor) Stop() {
 	bap.cancel()
 
 	// Wait for processing to complete
-	time.Sleep(bap.batchDuration + Config.BatchProcessingDelay)
+	time.Sleep(bap.batchDuration + GetConfig().BatchProcessingDelay)
 
 	bap.logger.Info().Msg("batch audio processor stopped")
 }
 
 // BatchReadEncode performs batched audio read and encode operations
 func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
+	// Get cached config to avoid GetConfig() calls in hot path
+	cache := GetCachedConfig()
+	cache.Update()
+
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
 		// Only log validation errors in debug mode to reduce overhead
@@ -210,7 +221,7 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(Config.BatchProcessorTimeout):
+	case <-time.After(cache.BatchProcessingTimeout):
 		// Timeout, fallback to single operation
 		// Use sampling to reduce atomic operations overhead
 		if atomic.LoadInt64(&bap.stats.SingleReads)%10 == 0 {
@@ -224,6 +235,10 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 // BatchDecodeWrite performs batched audio decode and write operations
 // This is the legacy version that uses a single buffer
 func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
+	// Get cached config to avoid GetConfig() calls in hot path
+	cache := GetCachedConfig()
+	cache.Update()
+
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
 		// Only log validation errors in debug mode to reduce overhead
@@ -268,7 +283,7 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(Config.BatchProcessorTimeout):
+	case <-time.After(cache.BatchProcessingTimeout):
 		// Use sampling to reduce atomic operations overhead
 		if atomic.LoadInt64(&bap.stats.SingleWrites)%10 == 0 {
 			atomic.AddInt64(&bap.stats.SingleWrites, 10)
@@ -280,6 +295,10 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 
 // BatchDecodeWriteWithBuffers performs batched audio decode and write operations with separate opus and PCM buffers
 func (bap *BatchAudioProcessor) BatchDecodeWriteWithBuffers(opusData []byte, pcmBuffer []byte) (int, error) {
+	// Get cached config to avoid GetConfig() calls in hot path
+	cache := GetCachedConfig()
+	cache.Update()
+
 	// Validate buffers before processing
 	if len(opusData) == 0 {
 		return 0, fmt.Errorf("empty opus data buffer")
@@ -320,7 +339,7 @@ func (bap *BatchAudioProcessor) BatchDecodeWriteWithBuffers(opusData []byte, pcm
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(Config.BatchProcessorTimeout):
+	case <-time.After(cache.BatchProcessingTimeout):
 		atomic.AddInt64(&bap.stats.SingleWrites, 1)
 		atomic.AddInt64(&bap.stats.WriteFrames, 1)
 		// Use the optimized function with separate buffers
@@ -407,9 +426,11 @@ func (bap *BatchAudioProcessor) processBatchRead(batch []batchReadRequest) {
 		return
 	}
 
-	threadPinningThreshold := Config.BatchProcessorThreadPinningThreshold
+	// Get cached config once - avoid repeated calls
+	cache := GetCachedConfig()
+	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
 	if threadPinningThreshold == 0 {
-		threadPinningThreshold = Config.MinBatchSizeForThreadPinning // Fallback
+		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
 	}
 
 	// Only pin to OS thread for large batches to reduce thread contention
@@ -458,9 +479,11 @@ func (bap *BatchAudioProcessor) processBatchWrite(batch []batchWriteRequest) {
 		return
 	}
 
-	threadPinningThreshold := Config.BatchProcessorThreadPinningThreshold
+	// Get cached config to avoid GetConfig() calls in hot path
+	cache := GetCachedConfig()
+	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
 	if threadPinningThreshold == 0 {
-		threadPinningThreshold = Config.MinBatchSizeForThreadPinning // Fallback
+		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
 	}
 
 	// Only pin to OS thread for large batches to reduce thread contention
@@ -562,7 +585,11 @@ func GetBatchAudioProcessor() *BatchAudioProcessor {
 
 	// Initialize on first use
 	if atomic.CompareAndSwapInt32(&batchProcessorInitialized, 0, 1) {
-		processor := NewBatchAudioProcessor(Config.BatchProcessorFramesPerBatch, Config.BatchProcessorTimeout)
+		// Get cached config to avoid GetConfig() calls
+		cache := GetCachedConfig()
+		cache.Update()
+
+		processor := NewBatchAudioProcessor(cache.BatchProcessorFramesPerBatch, cache.BatchProcessorTimeout)
 		atomic.StorePointer(&globalBatchProcessor, unsafe.Pointer(processor))
 		return processor
 	}
@@ -574,7 +601,8 @@ func GetBatchAudioProcessor() *BatchAudioProcessor {
 	}
 
 	// Fallback: create a new processor (should rarely happen)
-	return NewBatchAudioProcessor(Config.BatchProcessorFramesPerBatch, Config.BatchProcessorTimeout)
+	config := GetConfig()
+	return NewBatchAudioProcessor(config.BatchProcessorFramesPerBatch, config.BatchProcessorTimeout)
 }
 
 // EnableBatchAudioProcessing enables the global batch processor

internal/audio/batch_reference.go

@@ -1,331 +0,0 @@
-//go:build cgo
-
-package audio
-
-import (
-	"errors"
-	"sync"
-	"sync/atomic"
-	"unsafe"
-)
-
-// BatchReferenceManager handles batch reference counting operations
-// to reduce atomic operation overhead for high-frequency frame operations
-type BatchReferenceManager struct {
-	// Batch operations queue
-	batchQueue chan batchRefOperation
-	workerPool chan struct{} // Worker pool semaphore
-	running    int32
-	wg         sync.WaitGroup
-
-	// Statistics
-	batchedOps   int64
-	singleOps    int64
-	batchSavings int64 // Number of atomic operations saved
-}
-
-type batchRefOperation struct {
-	frames    []*ZeroCopyAudioFrame
-	operation refOperationType
-	resultCh  chan batchRefResult
-}
-
-type refOperationType int
-
-const (
-	refOpAddRef refOperationType = iota
-	refOpRelease
-	refOpMixed // For operations with mixed AddRef/Release
-)
-
-// Errors
-var (
-	ErrUnsupportedOperation = errors.New("unsupported batch reference operation")
-)
-
-type batchRefResult struct {
-	finalReleases []bool // For Release operations, indicates which frames had final release
-	err           error
-}
-
-// Global batch reference manager
-var (
-	globalBatchRefManager *BatchReferenceManager
-	batchRefOnce          sync.Once
-)
-
-// GetBatchReferenceManager returns the global batch reference manager
-func GetBatchReferenceManager() *BatchReferenceManager {
-	batchRefOnce.Do(func() {
-		globalBatchRefManager = NewBatchReferenceManager()
-		globalBatchRefManager.Start()
-	})
-	return globalBatchRefManager
-}
-
-// NewBatchReferenceManager creates a new batch reference manager
-func NewBatchReferenceManager() *BatchReferenceManager {
-	return &BatchReferenceManager{
-		batchQueue: make(chan batchRefOperation, 256), // Buffered for high throughput
-		workerPool: make(chan struct{}, 4),            // 4 workers for parallel processing
-	}
-}
-
-// Start starts the batch reference manager workers
-func (brm *BatchReferenceManager) Start() {
-	if !atomic.CompareAndSwapInt32(&brm.running, 0, 1) {
-		return // Already running
-	}
-
-	// Start worker goroutines
-	for i := 0; i < cap(brm.workerPool); i++ {
-		brm.wg.Add(1)
-		go brm.worker()
-	}
-}
-
-// Stop stops the batch reference manager
-func (brm *BatchReferenceManager) Stop() {
-	if !atomic.CompareAndSwapInt32(&brm.running, 1, 0) {
-		return // Already stopped
-	}
-
-	close(brm.batchQueue)
-	brm.wg.Wait()
-}
-
-// worker processes batch reference operations
-func (brm *BatchReferenceManager) worker() {
-	defer brm.wg.Done()
-
-	for op := range brm.batchQueue {
-		brm.processBatchOperation(op)
-	}
-}
-
-// processBatchOperation processes a batch of reference operations
-func (brm *BatchReferenceManager) processBatchOperation(op batchRefOperation) {
-	result := batchRefResult{}
-
-	switch op.operation {
-	case refOpAddRef:
-		// Batch AddRef operations
-		for _, frame := range op.frames {
-			if frame != nil {
-				atomic.AddInt32(&frame.refCount, 1)
-			}
-		}
-		atomic.AddInt64(&brm.batchedOps, int64(len(op.frames)))
-		atomic.AddInt64(&brm.batchSavings, int64(len(op.frames)-1)) // Saved ops vs individual calls
-
-	case refOpRelease:
-		// Batch Release operations
-		result.finalReleases = make([]bool, len(op.frames))
-		for i, frame := range op.frames {
-			if frame != nil {
-				newCount := atomic.AddInt32(&frame.refCount, -1)
-				if newCount == 0 {
-					result.finalReleases[i] = true
-					// Return to pool if pooled
-					if frame.pooled {
-						globalZeroCopyPool.Put(frame)
-					}
-				}
-			}
-		}
-		atomic.AddInt64(&brm.batchedOps, int64(len(op.frames)))
-		atomic.AddInt64(&brm.batchSavings, int64(len(op.frames)-1))
-
-	case refOpMixed:
-		// Handle mixed operations (not implemented in this version)
-		result.err = ErrUnsupportedOperation
-	}
-
-	// Send result back
-	if op.resultCh != nil {
-		op.resultCh <- result
-		close(op.resultCh)
-	}
-}
-
-// BatchAddRef performs AddRef on multiple frames in a single batch
-func (brm *BatchReferenceManager) BatchAddRef(frames []*ZeroCopyAudioFrame) error {
-	if len(frames) == 0 {
-		return nil
-	}
-
-	// For small batches, use direct operations to avoid overhead
-	if len(frames) <= 2 {
-		for _, frame := range frames {
-			if frame != nil {
-				frame.AddRef()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return nil
-	}
-
-	// Use batch processing for larger sets
-	if atomic.LoadInt32(&brm.running) == 0 {
-		// Fallback to individual operations if batch manager not running
-		for _, frame := range frames {
-			if frame != nil {
-				frame.AddRef()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return nil
-	}
-
-	resultCh := make(chan batchRefResult, 1)
-	op := batchRefOperation{
-		frames:    frames,
-		operation: refOpAddRef,
-		resultCh:  resultCh,
-	}
-
-	select {
-	case brm.batchQueue <- op:
-		// Wait for completion
-		<-resultCh
-		return nil
-	default:
-		// Queue full, fallback to individual operations
-		for _, frame := range frames {
-			if frame != nil {
-				frame.AddRef()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return nil
-	}
-}
-
-// BatchRelease performs Release on multiple frames in a single batch
-// Returns a slice indicating which frames had their final reference released
-func (brm *BatchReferenceManager) BatchRelease(frames []*ZeroCopyAudioFrame) ([]bool, error) {
-	if len(frames) == 0 {
-		return nil, nil
-	}
-
-	// For small batches, use direct operations
-	if len(frames) <= 2 {
-		finalReleases := make([]bool, len(frames))
-		for i, frame := range frames {
-			if frame != nil {
-				finalReleases[i] = frame.Release()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return finalReleases, nil
-	}
-
-	// Use batch processing for larger sets
-	if atomic.LoadInt32(&brm.running) == 0 {
-		// Fallback to individual operations
-		finalReleases := make([]bool, len(frames))
-		for i, frame := range frames {
-			if frame != nil {
-				finalReleases[i] = frame.Release()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return finalReleases, nil
-	}
-
-	resultCh := make(chan batchRefResult, 1)
-	op := batchRefOperation{
-		frames:    frames,
-		operation: refOpRelease,
-		resultCh:  resultCh,
-	}
-
-	select {
-	case brm.batchQueue <- op:
-		// Wait for completion
-		result := <-resultCh
-		return result.finalReleases, result.err
-	default:
-		// Queue full, fallback to individual operations
-		finalReleases := make([]bool, len(frames))
-		for i, frame := range frames {
-			if frame != nil {
-				finalReleases[i] = frame.Release()
-			}
-		}
-		atomic.AddInt64(&brm.singleOps, int64(len(frames)))
-		return finalReleases, nil
-	}
-}
-
-// GetStats returns batch reference counting statistics
-func (brm *BatchReferenceManager) GetStats() (batchedOps, singleOps, savings int64) {
-	return atomic.LoadInt64(&brm.batchedOps),
-		atomic.LoadInt64(&brm.singleOps),
-		atomic.LoadInt64(&brm.batchSavings)
-}
-
-// Convenience functions for global batch reference manager
-
-// BatchAddRefFrames performs batch AddRef on multiple frames
-func BatchAddRefFrames(frames []*ZeroCopyAudioFrame) error {
-	return GetBatchReferenceManager().BatchAddRef(frames)
-}
-
-// BatchReleaseFrames performs batch Release on multiple frames
-func BatchReleaseFrames(frames []*ZeroCopyAudioFrame) ([]bool, error) {
-	return GetBatchReferenceManager().BatchRelease(frames)
-}
-
-// GetBatchReferenceStats returns global batch reference statistics
-func GetBatchReferenceStats() (batchedOps, singleOps, savings int64) {
-	return GetBatchReferenceManager().GetStats()
-}
-
-// ZeroCopyFrameSlice provides utilities for working with slices of zero-copy frames
-type ZeroCopyFrameSlice []*ZeroCopyAudioFrame
-
-// AddRefAll performs batch AddRef on all frames in the slice
-func (zfs ZeroCopyFrameSlice) AddRefAll() error {
-	return BatchAddRefFrames(zfs)
-}
-
-// ReleaseAll performs batch Release on all frames in the slice
-func (zfs ZeroCopyFrameSlice) ReleaseAll() ([]bool, error) {
-	return BatchReleaseFrames(zfs)
-}
-
-// FilterNonNil returns a new slice with only non-nil frames
-func (zfs ZeroCopyFrameSlice) FilterNonNil() ZeroCopyFrameSlice {
-	filtered := make(ZeroCopyFrameSlice, 0, len(zfs))
-	for _, frame := range zfs {
-		if frame != nil {
-			filtered = append(filtered, frame)
-		}
-	}
-	return filtered
-}
-
-// Len returns the number of frames in the slice
-func (zfs ZeroCopyFrameSlice) Len() int {
-	return len(zfs)
-}
-
-// Get returns the frame at the specified index
-func (zfs ZeroCopyFrameSlice) Get(index int) *ZeroCopyAudioFrame {
-	if index < 0 || index >= len(zfs) {
-		return nil
-	}
-	return zfs[index]
-}
-
-// UnsafePointers returns unsafe pointers for all frames (for CGO batch operations)
-func (zfs ZeroCopyFrameSlice) UnsafePointers() []unsafe.Pointer {
-	pointers := make([]unsafe.Pointer, len(zfs))
-	for i, frame := range zfs {
-		if frame != nil {
-			pointers[i] = frame.UnsafePointer()
-		}
-	}
-	return pointers
-}

internal/audio/batch_zero_copy.go

@@ -1,415 +0,0 @@
-//go:build cgo
-
-package audio
-
-import (
-	"sync"
-	"sync/atomic"
-	"time"
-)
-
-// BatchZeroCopyProcessor handles batch operations on zero-copy audio frames
-// with optimized reference counting and memory management
-type BatchZeroCopyProcessor struct {
-	// Configuration
-	maxBatchSize      int
-	batchTimeout      time.Duration
-	processingDelay   time.Duration
-	adaptiveThreshold float64
-
-	// Processing queues
-	readEncodeQueue  chan *batchZeroCopyRequest
-	decodeWriteQueue chan *batchZeroCopyRequest
-
-	// Worker management
-	workerPool chan struct{}
-	running    int32
-	wg         sync.WaitGroup
-
-	// Statistics
-	batchedFrames    int64
-	singleFrames     int64
-	batchSavings     int64
-	processingTimeUs int64
-	adaptiveHits     int64
-	adaptiveMisses   int64
-}
-
-type batchZeroCopyRequest struct {
-	frames    []*ZeroCopyAudioFrame
-	operation batchZeroCopyOperation
-	resultCh  chan batchZeroCopyResult
-	timestamp time.Time
-}
-
-type batchZeroCopyOperation int
-
-const (
-	batchOpReadEncode batchZeroCopyOperation = iota
-	batchOpDecodeWrite
-	batchOpMixed
-)
-
-type batchZeroCopyResult struct {
-	encodedData    [][]byte // For read-encode operations
-	processedCount int      // Number of successfully processed frames
-	err            error
-}
-
-// Global batch zero-copy processor
-var (
-	globalBatchZeroCopyProcessor *BatchZeroCopyProcessor
-	batchZeroCopyOnce            sync.Once
-)
-
-// GetBatchZeroCopyProcessor returns the global batch zero-copy processor
-func GetBatchZeroCopyProcessor() *BatchZeroCopyProcessor {
-	batchZeroCopyOnce.Do(func() {
-		globalBatchZeroCopyProcessor = NewBatchZeroCopyProcessor()
-		globalBatchZeroCopyProcessor.Start()
-	})
-	return globalBatchZeroCopyProcessor
-}
-
-// NewBatchZeroCopyProcessor creates a new batch zero-copy processor
-func NewBatchZeroCopyProcessor() *BatchZeroCopyProcessor {
-	cache := Config
-	return &BatchZeroCopyProcessor{
-		maxBatchSize:      cache.BatchProcessorFramesPerBatch,
-		batchTimeout:      cache.BatchProcessorTimeout,
-		processingDelay:   cache.BatchProcessingDelay,
-		adaptiveThreshold: cache.BatchProcessorAdaptiveThreshold,
-		readEncodeQueue:   make(chan *batchZeroCopyRequest, cache.BatchProcessorMaxQueueSize),
-		decodeWriteQueue:  make(chan *batchZeroCopyRequest, cache.BatchProcessorMaxQueueSize),
-		workerPool:        make(chan struct{}, 4), // 4 workers for parallel processing
-	}
-}
-
-// Start starts the batch zero-copy processor workers
-func (bzcp *BatchZeroCopyProcessor) Start() {
-	if !atomic.CompareAndSwapInt32(&bzcp.running, 0, 1) {
-		return // Already running
-	}
-
-	// Start worker goroutines for read-encode operations
-	for i := 0; i < cap(bzcp.workerPool)/2; i++ {
-		bzcp.wg.Add(1)
-		go bzcp.readEncodeWorker()
-	}
-
-	// Start worker goroutines for decode-write operations
-	for i := 0; i < cap(bzcp.workerPool)/2; i++ {
-		bzcp.wg.Add(1)
-		go bzcp.decodeWriteWorker()
-	}
-}
-
-// Stop stops the batch zero-copy processor
-func (bzcp *BatchZeroCopyProcessor) Stop() {
-	if !atomic.CompareAndSwapInt32(&bzcp.running, 1, 0) {
-		return // Already stopped
-	}
-
-	close(bzcp.readEncodeQueue)
-	close(bzcp.decodeWriteQueue)
-	bzcp.wg.Wait()
-}
-
-// readEncodeWorker processes batch read-encode operations
-func (bzcp *BatchZeroCopyProcessor) readEncodeWorker() {
-	defer bzcp.wg.Done()
-
-	for req := range bzcp.readEncodeQueue {
-		bzcp.processBatchReadEncode(req)
-	}
-}
-
-// decodeWriteWorker processes batch decode-write operations
-func (bzcp *BatchZeroCopyProcessor) decodeWriteWorker() {
-	defer bzcp.wg.Done()
-
-	for req := range bzcp.decodeWriteQueue {
-		bzcp.processBatchDecodeWrite(req)
-	}
-}
-
-// processBatchReadEncode processes a batch of read-encode operations
-func (bzcp *BatchZeroCopyProcessor) processBatchReadEncode(req *batchZeroCopyRequest) {
-	startTime := time.Now()
-	result := batchZeroCopyResult{}
-
-	// Batch AddRef all frames first
-	err := BatchAddRefFrames(req.frames)
-	if err != nil {
-		result.err = err
-		if req.resultCh != nil {
-			req.resultCh <- result
-			close(req.resultCh)
-		}
-		return
-	}
-
-	// Process frames using existing batch read-encode logic
-	encodedData, err := BatchReadEncode(len(req.frames))
-	if err != nil {
-		// Batch release frames on error
-		if _, releaseErr := BatchReleaseFrames(req.frames); releaseErr != nil {
-			// Log release error but preserve original error
-			_ = releaseErr
-		}
-		result.err = err
-	} else {
-		result.encodedData = encodedData
-		result.processedCount = len(encodedData)
-		// Batch release frames after successful processing
-		if _, releaseErr := BatchReleaseFrames(req.frames); releaseErr != nil {
-			// Log release error but don't fail the operation
-			_ = releaseErr
-		}
-	}
-
-	// Update statistics
-	atomic.AddInt64(&bzcp.batchedFrames, int64(len(req.frames)))
-	atomic.AddInt64(&bzcp.batchSavings, int64(len(req.frames)-1))
-	atomic.AddInt64(&bzcp.processingTimeUs, time.Since(startTime).Microseconds())
-
-	// Send result back
-	if req.resultCh != nil {
-		req.resultCh <- result
-		close(req.resultCh)
-	}
-}
-
-// processBatchDecodeWrite processes a batch of decode-write operations
-func (bzcp *BatchZeroCopyProcessor) processBatchDecodeWrite(req *batchZeroCopyRequest) {
-	startTime := time.Now()
-	result := batchZeroCopyResult{}
-
-	// Batch AddRef all frames first
-	err := BatchAddRefFrames(req.frames)
-	if err != nil {
-		result.err = err
-		if req.resultCh != nil {
-			req.resultCh <- result
-			close(req.resultCh)
-		}
-		return
-	}
-
-	// Extract data from zero-copy frames for batch processing
-	frameData := make([][]byte, len(req.frames))
-	for i, frame := range req.frames {
-		if frame != nil {
-			// Get data from zero-copy frame
-			frameData[i] = frame.Data()[:frame.Length()]
-		}
-	}
-
-	// Process frames using existing batch decode-write logic
-	err = BatchDecodeWrite(frameData)
-	if err != nil {
-		result.err = err
-	} else {
-		result.processedCount = len(req.frames)
-	}
-
-	// Batch release frames
-	if _, releaseErr := BatchReleaseFrames(req.frames); releaseErr != nil {
-		// Log release error but don't override processing error
-		_ = releaseErr
-	}
-
-	// Update statistics
-	atomic.AddInt64(&bzcp.batchedFrames, int64(len(req.frames)))
-	atomic.AddInt64(&bzcp.batchSavings, int64(len(req.frames)-1))
-	atomic.AddInt64(&bzcp.processingTimeUs, time.Since(startTime).Microseconds())
-
-	// Send result back
-	if req.resultCh != nil {
-		req.resultCh <- result
-		close(req.resultCh)
-	}
-}
-
-// BatchReadEncodeZeroCopy performs batch read-encode on zero-copy frames
-func (bzcp *BatchZeroCopyProcessor) BatchReadEncodeZeroCopy(frames []*ZeroCopyAudioFrame) ([][]byte, error) {
-	if len(frames) == 0 {
-		return nil, nil
-	}
-
-	// For small batches, use direct operations to avoid overhead
-	if len(frames) <= 2 {
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleReadEncode(frames)
-	}
-
-	// Use adaptive threshold to determine batch vs single processing
-	batchedFrames := atomic.LoadInt64(&bzcp.batchedFrames)
-	singleFrames := atomic.LoadInt64(&bzcp.singleFrames)
-	totalFrames := batchedFrames + singleFrames
-
-	if totalFrames > 100 { // Only apply adaptive logic after some samples
-		batchRatio := float64(batchedFrames) / float64(totalFrames)
-		if batchRatio < bzcp.adaptiveThreshold {
-			// Batch processing not effective, use single processing
-			atomic.AddInt64(&bzcp.adaptiveMisses, 1)
-			atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-			return bzcp.processSingleReadEncode(frames)
-		}
-		atomic.AddInt64(&bzcp.adaptiveHits, 1)
-	}
-
-	// Use batch processing
-	if atomic.LoadInt32(&bzcp.running) == 0 {
-		// Fallback to single processing if batch processor not running
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleReadEncode(frames)
-	}
-
-	resultCh := make(chan batchZeroCopyResult, 1)
-	req := &batchZeroCopyRequest{
-		frames:    frames,
-		operation: batchOpReadEncode,
-		resultCh:  resultCh,
-		timestamp: time.Now(),
-	}
-
-	select {
-	case bzcp.readEncodeQueue <- req:
-		// Wait for completion
-		result := <-resultCh
-		return result.encodedData, result.err
-	default:
-		// Queue full, fallback to single processing
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleReadEncode(frames)
-	}
-}
-
-// BatchDecodeWriteZeroCopy performs batch decode-write on zero-copy frames
-func (bzcp *BatchZeroCopyProcessor) BatchDecodeWriteZeroCopy(frames []*ZeroCopyAudioFrame) error {
-	if len(frames) == 0 {
-		return nil
-	}
-
-	// For small batches, use direct operations
-	if len(frames) <= 2 {
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleDecodeWrite(frames)
-	}
-
-	// Use adaptive threshold
-	batchedFrames := atomic.LoadInt64(&bzcp.batchedFrames)
-	singleFrames := atomic.LoadInt64(&bzcp.singleFrames)
-	totalFrames := batchedFrames + singleFrames
-
-	if totalFrames > 100 {
-		batchRatio := float64(batchedFrames) / float64(totalFrames)
-		if batchRatio < bzcp.adaptiveThreshold {
-			atomic.AddInt64(&bzcp.adaptiveMisses, 1)
-			atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-			return bzcp.processSingleDecodeWrite(frames)
-		}
-		atomic.AddInt64(&bzcp.adaptiveHits, 1)
-	}
-
-	// Use batch processing
-	if atomic.LoadInt32(&bzcp.running) == 0 {
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleDecodeWrite(frames)
-	}
-
-	resultCh := make(chan batchZeroCopyResult, 1)
-	req := &batchZeroCopyRequest{
-		frames:    frames,
-		operation: batchOpDecodeWrite,
-		resultCh:  resultCh,
-		timestamp: time.Now(),
-	}
-
-	select {
-	case bzcp.decodeWriteQueue <- req:
-		// Wait for completion
-		result := <-resultCh
-		return result.err
-	default:
-		// Queue full, fallback to single processing
-		atomic.AddInt64(&bzcp.singleFrames, int64(len(frames)))
-		return bzcp.processSingleDecodeWrite(frames)
-	}
-}
-
-// processSingleReadEncode processes frames individually for read-encode
-func (bzcp *BatchZeroCopyProcessor) processSingleReadEncode(frames []*ZeroCopyAudioFrame) ([][]byte, error) {
-	// Extract data and use existing batch processing
-	frameData := make([][]byte, 0, len(frames))
-	for _, frame := range frames {
-		if frame != nil {
-			frame.AddRef()
-			frameData = append(frameData, frame.Data()[:frame.Length()])
-		}
-	}
-
-	// Use existing batch read-encode
-	result, err := BatchReadEncode(len(frameData))
-
-	// Release frames
-	for _, frame := range frames {
-		if frame != nil {
-			frame.Release()
-		}
-	}
-
-	return result, err
-}
-
-// processSingleDecodeWrite processes frames individually for decode-write
-func (bzcp *BatchZeroCopyProcessor) processSingleDecodeWrite(frames []*ZeroCopyAudioFrame) error {
-	// Extract data and use existing batch processing
-	frameData := make([][]byte, 0, len(frames))
-	for _, frame := range frames {
-		if frame != nil {
-			frame.AddRef()
-			frameData = append(frameData, frame.Data()[:frame.Length()])
-		}
-	}
-
-	// Use existing batch decode-write
-	err := BatchDecodeWrite(frameData)
-
-	// Release frames
-	for _, frame := range frames {
-		if frame != nil {
-			frame.Release()
-		}
-	}
-
-	return err
-}
-
-// GetBatchZeroCopyStats returns batch zero-copy processing statistics
-func (bzcp *BatchZeroCopyProcessor) GetBatchZeroCopyStats() (batchedFrames, singleFrames, savings, processingTimeUs, adaptiveHits, adaptiveMisses int64) {
-	return atomic.LoadInt64(&bzcp.batchedFrames),
-		atomic.LoadInt64(&bzcp.singleFrames),
-		atomic.LoadInt64(&bzcp.batchSavings),
-		atomic.LoadInt64(&bzcp.processingTimeUs),
-		atomic.LoadInt64(&bzcp.adaptiveHits),
-		atomic.LoadInt64(&bzcp.adaptiveMisses)
-}
-
-// Convenience functions for global batch zero-copy processor
-
-// BatchReadEncodeZeroCopyFrames performs batch read-encode on zero-copy frames
-func BatchReadEncodeZeroCopyFrames(frames []*ZeroCopyAudioFrame) ([][]byte, error) {
-	return GetBatchZeroCopyProcessor().BatchReadEncodeZeroCopy(frames)
-}
-
-// BatchDecodeWriteZeroCopyFrames performs batch decode-write on zero-copy frames
-func BatchDecodeWriteZeroCopyFrames(frames []*ZeroCopyAudioFrame) error {
-	return GetBatchZeroCopyProcessor().BatchDecodeWriteZeroCopy(frames)
-}
-
-// GetGlobalBatchZeroCopyStats returns global batch zero-copy processing statistics
-func GetGlobalBatchZeroCopyStats() (batchedFrames, singleFrames, savings, processingTimeUs, adaptiveHits, adaptiveMisses int64) {
-	return GetBatchZeroCopyProcessor().GetBatchZeroCopyStats()
-}

internal/audio/cgo_audio.go

@@ -14,15 +14,12 @@ import (
 /*
 #cgo CFLAGS: -I$HOME/.jetkvm/audio-libs/alsa-lib-$ALSA_VERSION/include -I$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/include -I$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/celt
 #cgo LDFLAGS: -L$HOME/.jetkvm/audio-libs/alsa-lib-$ALSA_VERSION/src/.libs -lasound -L$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/.libs -lopus -lm -ldl -static
-
 #include <alsa/asoundlib.h>
 #include <opus.h>
-#include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
-#include <unistd.h>
 #include <errno.h>
-#include <sys/mman.h>
+#include <unistd.h>
 
 // C state for ALSA/Opus with safety flags
 static snd_pcm_t *pcm_handle = NULL;
@@ -30,33 +27,25 @@ static snd_pcm_t *pcm_playback_handle = NULL;
 static OpusEncoder *encoder = NULL;
 static OpusDecoder *decoder = NULL;
 // Opus encoder settings - initialized from Go configuration
-static int opus_bitrate = 96000;        // Will be set from Config.CGOOpusBitrate
+static int opus_bitrate = 96000;        // Will be set from GetConfig().CGOOpusBitrate
-static int opus_complexity = 3;         // Will be set from Config.CGOOpusComplexity
+static int opus_complexity = 3;         // Will be set from GetConfig().CGOOpusComplexity
-static int opus_vbr = 1;                // Will be set from Config.CGOOpusVBR
+static int opus_vbr = 1;                // Will be set from GetConfig().CGOOpusVBR
-static int opus_vbr_constraint = 1;     // Will be set from Config.CGOOpusVBRConstraint
+static int opus_vbr_constraint = 1;     // Will be set from GetConfig().CGOOpusVBRConstraint
-static int opus_signal_type = 3;        // Will be set from Config.CGOOpusSignalType
+static int opus_signal_type = 3;        // Will be set from GetConfig().CGOOpusSignalType
 static int opus_bandwidth = 1105;       // OPUS_BANDWIDTH_WIDEBAND for compatibility (was 1101)
-static int opus_dtx = 0;                // Will be set from Config.CGOOpusDTX
+static int opus_dtx = 0;                // Will be set from GetConfig().CGOOpusDTX
 static int opus_lsb_depth = 16;         // LSB depth for improved bit allocation on constrained hardware
-static int sample_rate = 48000;         // Will be set from Config.CGOSampleRate
+static int sample_rate = 48000;         // Will be set from GetConfig().CGOSampleRate
-static int channels = 2;                // Will be set from Config.CGOChannels
+static int channels = 2;                // Will be set from GetConfig().CGOChannels
-static int frame_size = 960;            // Will be set from Config.CGOFrameSize
+static int frame_size = 960;            // Will be set from GetConfig().CGOFrameSize
-static int max_packet_size = 1500;      // Will be set from Config.CGOMaxPacketSize
+static int max_packet_size = 1500;      // Will be set from GetConfig().CGOMaxPacketSize
-static int sleep_microseconds = 1000;   // Will be set from Config.CGOUsleepMicroseconds
+static int sleep_microseconds = 1000;   // Will be set from GetConfig().CGOUsleepMicroseconds
-static int max_attempts_global = 5;     // Will be set from Config.CGOMaxAttempts
+static int max_attempts_global = 5;     // Will be set from GetConfig().CGOMaxAttempts
-static int max_backoff_us_global = 500000; // Will be set from Config.CGOMaxBackoffMicroseconds
+static int max_backoff_us_global = 500000; // Will be set from GetConfig().CGOMaxBackoffMicroseconds
 // Hardware optimization flags for constrained environments
 static int use_mmap_access = 0;         // Disable MMAP for compatibility (was 1)
 static int optimized_buffer_size = 0;   // Disable optimized buffer sizing for stability (was 1)
 
-// C function declarations (implementations are below)
-int jetkvm_audio_init();
-void jetkvm_audio_close();
-int jetkvm_audio_read_encode(void *opus_buf);
-int jetkvm_audio_decode_write(void *opus_buf, int opus_size);
-int jetkvm_audio_playback_init();
-void jetkvm_audio_playback_close();
-
 // Function to update constants from Go configuration
 void update_audio_constants(int bitrate, int complexity, int vbr, int vbr_constraint,
                            int signal_type, int bandwidth, int dtx, int lsb_depth, int sr, int ch,
@@ -87,10 +76,8 @@ static volatile int playback_initialized = 0;
 // Function to dynamically update Opus encoder parameters
 int update_opus_encoder_params(int bitrate, int complexity, int vbr, int vbr_constraint,
                               int signal_type, int bandwidth, int dtx) {
-    // This function works for both audio input and output encoder parameters
+    if (!encoder || !capture_initialized) {
-    // Require either capture (output) or playback (input) initialization
+        return -1; // Encoder not initialized
-    if (!encoder || (!capture_initialized && !playback_initialized)) {
-        return -1; // Audio encoder not initialized
     }
 
     // Update the static variables
@@ -711,9 +698,9 @@ func cgoAudioInit() error {
 		C.int(cache.channels.Load()),
 		C.int(cache.frameSize.Load()),
 		C.int(cache.maxPacketSize.Load()),
-		C.int(Config.CGOUsleepMicroseconds),
+		C.int(GetConfig().CGOUsleepMicroseconds),
-		C.int(Config.CGOMaxAttempts),
+		C.int(GetConfig().CGOMaxAttempts),
-		C.int(Config.CGOMaxBackoffMicroseconds),
+		C.int(GetConfig().CGOMaxBackoffMicroseconds),
 	)
 
 	result := C.jetkvm_audio_init()
@@ -728,6 +715,7 @@ func cgoAudioClose() {
 }
 
 // AudioConfigCache provides a comprehensive caching system for audio configuration
+// to minimize GetConfig() calls in the hot path
 type AudioConfigCache struct {
 	// Atomic int64 fields MUST be first for ARM32 alignment (8-byte alignment required)
 	minFrameDuration         atomic.Int64 // Store as nanoseconds
@@ -816,50 +804,52 @@ func (c *AudioConfigCache) Update() {
 
 	// Double-check after acquiring lock
 	if !c.initialized.Load() || time.Since(c.lastUpdate) > c.cacheExpiry {
+		config := GetConfig() // Call GetConfig() only once
+
 		// Update atomic values for lock-free access - CGO values
-		c.minReadEncodeBuffer.Store(int32(Config.MinReadEncodeBuffer))
+		c.minReadEncodeBuffer.Store(int32(config.MinReadEncodeBuffer))
-		c.maxDecodeWriteBuffer.Store(int32(Config.MaxDecodeWriteBuffer))
+		c.maxDecodeWriteBuffer.Store(int32(config.MaxDecodeWriteBuffer))
-		c.maxPacketSize.Store(int32(Config.CGOMaxPacketSize))
+		c.maxPacketSize.Store(int32(config.CGOMaxPacketSize))
-		c.maxPCMBufferSize.Store(int32(Config.MaxPCMBufferSize))
+		c.maxPCMBufferSize.Store(int32(config.MaxPCMBufferSize))
-		c.opusBitrate.Store(int32(Config.CGOOpusBitrate))
+		c.opusBitrate.Store(int32(config.CGOOpusBitrate))
-		c.opusComplexity.Store(int32(Config.CGOOpusComplexity))
+		c.opusComplexity.Store(int32(config.CGOOpusComplexity))
-		c.opusVBR.Store(int32(Config.CGOOpusVBR))
+		c.opusVBR.Store(int32(config.CGOOpusVBR))
-		c.opusVBRConstraint.Store(int32(Config.CGOOpusVBRConstraint))
+		c.opusVBRConstraint.Store(int32(config.CGOOpusVBRConstraint))
-		c.opusSignalType.Store(int32(Config.CGOOpusSignalType))
+		c.opusSignalType.Store(int32(config.CGOOpusSignalType))
-		c.opusBandwidth.Store(int32(Config.CGOOpusBandwidth))
+		c.opusBandwidth.Store(int32(config.CGOOpusBandwidth))
-		c.opusDTX.Store(int32(Config.CGOOpusDTX))
+		c.opusDTX.Store(int32(config.CGOOpusDTX))
-		c.sampleRate.Store(int32(Config.CGOSampleRate))
+		c.sampleRate.Store(int32(config.CGOSampleRate))
-		c.channels.Store(int32(Config.CGOChannels))
+		c.channels.Store(int32(config.CGOChannels))
-		c.frameSize.Store(int32(Config.CGOFrameSize))
+		c.frameSize.Store(int32(config.CGOFrameSize))
 
 		// Update additional validation values
-		c.maxAudioFrameSize.Store(int32(Config.MaxAudioFrameSize))
+		c.maxAudioFrameSize.Store(int32(config.MaxAudioFrameSize))
-		c.maxChannels.Store(int32(Config.MaxChannels))
+		c.maxChannels.Store(int32(config.MaxChannels))
-		c.minFrameDuration.Store(int64(Config.MinFrameDuration))
+		c.minFrameDuration.Store(int64(config.MinFrameDuration))
-		c.maxFrameDuration.Store(int64(Config.MaxFrameDuration))
+		c.maxFrameDuration.Store(int64(config.MaxFrameDuration))
-		c.minOpusBitrate.Store(int32(Config.MinOpusBitrate))
+		c.minOpusBitrate.Store(int32(config.MinOpusBitrate))
-		c.maxOpusBitrate.Store(int32(Config.MaxOpusBitrate))
+		c.maxOpusBitrate.Store(int32(config.MaxOpusBitrate))
 
 		// Update batch processing related values
 		c.BatchProcessingTimeout = 100 * time.Millisecond // Fixed timeout for batch processing
-		c.BatchProcessorFramesPerBatch = Config.BatchProcessorFramesPerBatch
+		c.BatchProcessorFramesPerBatch = config.BatchProcessorFramesPerBatch
-		c.BatchProcessorTimeout = Config.BatchProcessorTimeout
+		c.BatchProcessorTimeout = config.BatchProcessorTimeout
-		c.BatchProcessingDelay = Config.BatchProcessingDelay
+		c.BatchProcessingDelay = config.BatchProcessingDelay
-		c.MinBatchSizeForThreadPinning = Config.MinBatchSizeForThreadPinning
+		c.MinBatchSizeForThreadPinning = config.MinBatchSizeForThreadPinning
-		c.BatchProcessorMaxQueueSize = Config.BatchProcessorMaxQueueSize
+		c.BatchProcessorMaxQueueSize = config.BatchProcessorMaxQueueSize
-		c.BatchProcessorAdaptiveThreshold = Config.BatchProcessorAdaptiveThreshold
+		c.BatchProcessorAdaptiveThreshold = config.BatchProcessorAdaptiveThreshold
-		c.BatchProcessorThreadPinningThreshold = Config.BatchProcessorThreadPinningThreshold
+		c.BatchProcessorThreadPinningThreshold = config.BatchProcessorThreadPinningThreshold
 
 		// Pre-allocate common errors
-		c.bufferTooSmallReadEncode = newBufferTooSmallError(0, Config.MinReadEncodeBuffer)
+		c.bufferTooSmallReadEncode = newBufferTooSmallError(0, config.MinReadEncodeBuffer)
-		c.bufferTooLargeDecodeWrite = newBufferTooLargeError(Config.MaxDecodeWriteBuffer+1, Config.MaxDecodeWriteBuffer)
+		c.bufferTooLargeDecodeWrite = newBufferTooLargeError(config.MaxDecodeWriteBuffer+1, config.MaxDecodeWriteBuffer)
 
 		c.lastUpdate = time.Now()
 		c.initialized.Store(true)
 
 		// Update the global validation cache as well
 		if cachedMaxFrameSize != 0 {
-			cachedMaxFrameSize = Config.MaxAudioFrameSize
+			cachedMaxFrameSize = config.MaxAudioFrameSize
 		}
 	}
 }
@@ -911,28 +901,46 @@ func updateCacheIfNeeded(cache *AudioConfigCache) {
 }
 
 func cgoAudioReadEncode(buf []byte) (int, error) {
-	// Minimal buffer validation - assume caller provides correct size
+	cache := GetCachedConfig()
-	if len(buf) == 0 {
+	updateCacheIfNeeded(cache)
-		return 0, errEmptyBuffer
+
+	// Fast validation with cached values - avoid lock with atomic access
+	minRequired := cache.GetMinReadEncodeBuffer()
+
+	// Buffer validation - use pre-allocated error for common case
+	if len(buf) < minRequired {
+		// Use pre-allocated error for common case, only create custom error for edge cases
+		if len(buf) > 0 {
+			return 0, newBufferTooSmallError(len(buf), minRequired)
+		}
+		return 0, cache.GetBufferTooSmallError()
 	}
 
-	// Direct CGO call - hotpath optimization
+	// Skip initialization check for now to avoid CGO compilation issues
+
+	// Direct CGO call with minimal overhead - unsafe.Pointer(&slice[0]) is safe for validated non-empty buffers
 	n := C.jetkvm_audio_read_encode(unsafe.Pointer(&buf[0]))
 
-	// Fast path for success
+	// Fast path for success case
 	if n > 0 {
 		return int(n), nil
 	}
 
-	// Error handling with static errors
+	// Handle error cases - use static error codes to reduce allocations
 	if n < 0 {
-		if n == -1 {
+		// Common error cases
+		switch n {
+		case -1:
 			return 0, errAudioInitFailed
+		case -2:
+			return 0, errAudioReadEncode
+		default:
+			return 0, newAudioReadEncodeError(int(n))
 		}
-		return 0, errAudioReadEncode
 	}
 
-	return 0, nil
+	// n == 0 case
+	return 0, nil // No data available
 }
 
 // Audio playback functions
@@ -954,25 +962,58 @@ func cgoAudioPlaybackClose() {
 	C.jetkvm_audio_playback_close()
 }
 
-func cgoAudioDecodeWrite(buf []byte) (int, error) {
+func cgoAudioDecodeWrite(buf []byte) (n int, err error) {
-	// Minimal validation - assume caller provides correct size
+	// Fast validation with AudioConfigCache
+	cache := GetCachedConfig()
+	// Only update cache if expired - avoid unnecessary overhead
+	// Use proper locking to avoid race condition
+	if cache.initialized.Load() {
+		cache.mutex.RLock()
+		cacheExpired := time.Since(cache.lastUpdate) > cache.cacheExpiry
+		cache.mutex.RUnlock()
+		if cacheExpired {
+			cache.Update()
+		}
+	} else {
+		cache.Update()
+	}
+
+	// Optimized buffer validation
 	if len(buf) == 0 {
 		return 0, errEmptyBuffer
 	}
 
-	// Direct CGO call - hotpath optimization
+	// Use cached max buffer size with atomic access
-	n := int(C.jetkvm_audio_decode_write(unsafe.Pointer(&buf[0]), C.int(len(buf))))
+	maxAllowed := cache.GetMaxDecodeWriteBuffer()
+	if len(buf) > maxAllowed {
+		// Use pre-allocated error for common case
+		if len(buf) == maxAllowed+1 {
+			return 0, cache.GetBufferTooLargeError()
+		}
+		return 0, newBufferTooLargeError(len(buf), maxAllowed)
+	}
 
-	// Fast path for success
+	// Direct CGO call with minimal overhead - unsafe.Pointer(&slice[0]) is safe for validated non-empty buffers
+	n = int(C.jetkvm_audio_decode_write(unsafe.Pointer(&buf[0]), C.int(len(buf))))
+
+	// Fast path for success case
 	if n >= 0 {
 		return n, nil
 	}
 
-	// Error handling with static errors
+	// Handle error cases with static error codes
-	if n == -1 {
+	switch n {
-		return 0, errAudioInitFailed
+	case -1:
+		n = 0
+		err = errAudioInitFailed
+	case -2:
+		n = 0
+		err = errAudioDecodeWrite
+	default:
+		n = 0
+		err = newAudioDecodeWriteError(n)
 	}
-	return 0, errAudioDecodeWrite
+	return
 }
 
 // updateOpusEncoderParams dynamically updates OPUS encoder parameters
@@ -995,7 +1036,7 @@ func updateOpusEncoderParams(bitrate, complexity, vbr, vbrConstraint, signalType
 // Buffer pool for reusing buffers in CGO functions
 var (
 	// Using SizedBufferPool for better memory management
-	// Track buffer pool usage
+	// Track buffer pool usage for monitoring
 	cgoBufferPoolGets atomic.Int64
 	cgoBufferPoolPuts atomic.Int64
 	// Batch processing statistics - only enabled in debug builds
@@ -1058,24 +1099,70 @@ func DecodeWriteWithPooledBuffer(data []byte) (int, error) {
 }
 
 // BatchReadEncode reads and encodes multiple audio frames in a single batch
-// with optimized zero-copy frame management and batch reference counting
 func BatchReadEncode(batchSize int) ([][]byte, error) {
-	// Simple batch processing without complex overhead
+	cache := GetCachedConfig()
-	frames := make([][]byte, 0, batchSize)
+	updateCacheIfNeeded(cache)
-	frameSize := 4096 // Fixed frame size for performance
 
+	// Calculate total buffer size needed for batch
+	frameSize := cache.GetMinReadEncodeBuffer()
+	totalSize := frameSize * batchSize
+
+	// Get a single large buffer for all frames
+	batchBuffer := GetBufferFromPool(totalSize)
+	defer ReturnBufferToPool(batchBuffer)
+
+	// Pre-allocate frame result buffers from pool to avoid allocations in loop
+	frameBuffers := make([][]byte, 0, batchSize)
 	for i := 0; i < batchSize; i++ {
-		buf := make([]byte, frameSize)
+		frameBuffers = append(frameBuffers, GetBufferFromPool(frameSize))
-		n, err := cgoAudioReadEncode(buf)
+	}
+	defer func() {
+		// Return all frame buffers to pool
+		for _, buf := range frameBuffers {
+			ReturnBufferToPool(buf)
+		}
+	}()
+
+	// Track batch processing statistics - only if enabled
+	var startTime time.Time
+	// Batch time tracking removed
+	trackTime := false
+	if trackTime {
+		startTime = time.Now()
+	}
+	batchProcessingCount.Add(1)
+
+	// Process frames in batch
+	frames := make([][]byte, 0, batchSize)
+	for i := 0; i < batchSize; i++ {
+		// Calculate offset for this frame in the batch buffer
+		offset := i * frameSize
+		frameBuf := batchBuffer[offset : offset+frameSize]
+
+		// Process this frame
+		n, err := cgoAudioReadEncode(frameBuf)
 		if err != nil {
+			// Return partial batch on error
 			if i > 0 {
-				return frames, nil // Return partial batch
+				batchFrameCount.Add(int64(i))
+				if trackTime {
+					batchProcessingTime.Add(time.Since(startTime).Microseconds())
+				}
+				return frames, nil
 			}
 			return nil, err
 		}
-		if n > 0 {
+
-			frames = append(frames, buf[:n])
+		// Reuse pre-allocated buffer instead of make([]byte, n)
-		}
+		frameCopy := frameBuffers[i][:n] // Slice to actual size
+		copy(frameCopy, frameBuf[:n])
+		frames = append(frames, frameCopy)
+	}
+
+	// Update statistics
+	batchFrameCount.Add(int64(len(frames)))
+	if trackTime {
+		batchProcessingTime.Add(time.Since(startTime).Microseconds())
 	}
 
 	return frames, nil
@@ -1083,39 +1170,12 @@ func BatchReadEncode(batchSize int) ([][]byte, error) {
 
 // BatchDecodeWrite decodes and writes multiple audio frames in a single batch
 // This reduces CGO call overhead by processing multiple frames at once
-// with optimized zero-copy frame management and batch reference counting
 func BatchDecodeWrite(frames [][]byte) error {
 	// Validate input
 	if len(frames) == 0 {
 		return nil
 	}
 
-	// Convert to zero-copy frames for optimized processing
-	zeroCopyFrames := make([]*ZeroCopyAudioFrame, 0, len(frames))
-	for _, frameData := range frames {
-		if len(frameData) > 0 {
-			frame := GetZeroCopyFrame()
-			frame.SetDataDirect(frameData) // Direct assignment without copy
-			zeroCopyFrames = append(zeroCopyFrames, frame)
-		}
-	}
-
-	// Use batch reference counting for efficient management
-	if len(zeroCopyFrames) > 0 {
-		// Batch AddRef all frames at once
-		err := BatchAddRefFrames(zeroCopyFrames)
-		if err != nil {
-			return err
-		}
-		// Ensure cleanup with batch release
-		defer func() {
-			if _, err := BatchReleaseFrames(zeroCopyFrames); err != nil {
-				// Log release error but don't fail the operation
-				_ = err
-			}
-		}()
-	}
-
 	// Get cached config
 	cache := GetCachedConfig()
 	// Only update cache if expired - avoid unnecessary overhead
@@ -1144,17 +1204,16 @@ func BatchDecodeWrite(frames [][]byte) error {
 	pcmBuffer := GetBufferFromPool(cache.GetMaxPCMBufferSize())
 	defer ReturnBufferToPool(pcmBuffer)
 
-	// Process each zero-copy frame with optimized batch processing
+	// Process each frame
 	frameCount := 0
-	for _, zcFrame := range zeroCopyFrames {
+	for _, frame := range frames {
-		// Get frame data from zero-copy frame
+		// Skip empty frames
-		frameData := zcFrame.Data()[:zcFrame.Length()]
+		if len(frame) == 0 {
-		if len(frameData) == 0 {
 			continue
 		}
 
 		// Process this frame using optimized implementation
-		_, err := CGOAudioDecodeWrite(frameData, pcmBuffer)
+		_, err := CGOAudioDecodeWrite(frame, pcmBuffer)
 		if err != nil {
 			// Update statistics before returning error
 			batchFrameCount.Add(int64(frameCount))

internal/audio/core_config.go

@@ -4,12 +4,12 @@ import "time"
 
 // GetMetricsUpdateInterval returns the current metrics update interval from centralized config
 func GetMetricsUpdateInterval() time.Duration {
-	return Config.MetricsUpdateInterval
+	return GetConfig().MetricsUpdateInterval
 }
 
 // SetMetricsUpdateInterval sets the metrics update interval in centralized config
 func SetMetricsUpdateInterval(interval time.Duration) {
-	config := Config
+	config := GetConfig()
 	config.MetricsUpdateInterval = interval
 	UpdateConfig(config)
 }

internal/audio/core_config_constants.go

@@ -117,6 +117,7 @@ type AudioConfigConstants struct {
 
 	// Buffer Management
 
+	PreallocSize                 int
 	MaxPoolSize                  int
 	MessagePoolSize              int
 	OptimalSocketBuffer          int
@@ -130,27 +131,27 @@ type AudioConfigConstants struct {
 	MinReadEncodeBuffer          int
 	MaxDecodeWriteBuffer         int
 	MinBatchSizeForThreadPinning int
-
+	GoroutineMonitorInterval     time.Duration
-	MagicNumber            uint32
+	MagicNumber                  uint32
-	MaxFrameSize           int
+	MaxFrameSize                 int
-	WriteTimeout           time.Duration
+	WriteTimeout                 time.Duration
-	HeaderSize             int
+	HeaderSize                   int
-	MetricsUpdateInterval  time.Duration
+	MetricsUpdateInterval        time.Duration
-	WarmupSamples          int
+	WarmupSamples                int
-	MetricsChannelBuffer   int
+	MetricsChannelBuffer         int
-	LatencyHistorySize     int
+	LatencyHistorySize           int
-	MaxCPUPercent          float64
+	MaxCPUPercent                float64
-	MinCPUPercent          float64
+	MinCPUPercent                float64
-	DefaultClockTicks      float64
+	DefaultClockTicks            float64
-	DefaultMemoryGB        int
+	DefaultMemoryGB              int
-	MaxWarmupSamples       int
+	MaxWarmupSamples             int
-	WarmupCPUSamples       int
+	WarmupCPUSamples             int
-	LogThrottleIntervalSec int
+	LogThrottleIntervalSec       int
-	MinValidClockTicks     int
+	MinValidClockTicks           int
-	MaxValidClockTicks     int
+	MaxValidClockTicks           int
-	CPUFactor              float64
+	CPUFactor                    float64
-	MemoryFactor           float64
+	MemoryFactor                 float64
-	LatencyFactor          float64
+	LatencyFactor                float64
 
 	// Adaptive Buffer Configuration
 	AdaptiveMinBufferSize     int // Minimum buffer size in frames for adaptive buffering
@@ -171,25 +172,28 @@ type AudioConfigConstants struct {
 	OutputSupervisorTimeout time.Duration // 5s
 	BatchProcessingDelay    time.Duration // 10ms
 
+	AdaptiveOptimizerStability time.Duration // 10s
+	LatencyMonitorTarget       time.Duration // 50ms
+
 	// Adaptive Buffer Configuration
 	// LowCPUThreshold defines CPU usage threshold for buffer size reduction.
 	LowCPUThreshold float64 // 20% CPU threshold for buffer optimization
 
 	// HighCPUThreshold defines CPU usage threshold for buffer size increase.
-	HighCPUThreshold            float64       // 60% CPU threshold
+	HighCPUThreshold               float64       // 60% CPU threshold
-	LowMemoryThreshold          float64       // 50% memory threshold
+	LowMemoryThreshold             float64       // 50% memory threshold
-	HighMemoryThreshold         float64       // 75% memory threshold
+	HighMemoryThreshold            float64       // 75% memory threshold
-	AdaptiveBufferTargetLatency time.Duration // 20ms target latency
+	AdaptiveBufferTargetLatency    time.Duration // 20ms target latency
-	CooldownPeriod              time.Duration // 30s cooldown period
+	CooldownPeriod                 time.Duration // 30s cooldown period
-	RollbackThreshold           time.Duration // 300ms rollback threshold
+	RollbackThreshold              time.Duration // 300ms rollback threshold
-
+	AdaptiveOptimizerLatencyTarget time.Duration // 50ms latency target
-	MaxLatencyThreshold         time.Duration // 200ms max latency
+	MaxLatencyThreshold            time.Duration // 200ms max latency
-	JitterThreshold             time.Duration // 20ms jitter threshold
+	JitterThreshold                time.Duration // 20ms jitter threshold
-	LatencyOptimizationInterval time.Duration // 5s optimization interval
+	LatencyOptimizationInterval    time.Duration // 5s optimization interval
-	LatencyAdaptiveThreshold    float64       // 0.8 adaptive threshold
+	LatencyAdaptiveThreshold       float64       // 0.8 adaptive threshold
-	MicContentionTimeout        time.Duration // 200ms contention timeout
+	MicContentionTimeout           time.Duration // 200ms contention timeout
-	PreallocPercentage          int           // 20% preallocation percentage
+	PreallocPercentage             int           // 20% preallocation percentage
-	BackoffStart                time.Duration // 50ms initial backoff
+	BackoffStart                   time.Duration // 50ms initial backoff
 
 	InputMagicNumber uint32 // Magic number for input IPC messages (0x4A4B4D49 "JKMI")
 
@@ -210,8 +214,29 @@ type AudioConfigConstants struct {
 	// CGO Audio Processing Constants
 	CGOUsleepMicroseconds int // Sleep duration for CGO usleep calls (1000μs)
 
-	CGOPCMBufferSize        int     // PCM buffer size for CGO audio processing
+	CGOPCMBufferSize            int     // PCM buffer size for CGO audio processing
-	CGONanosecondsPerSecond float64 // Nanoseconds per second conversion
+	CGONanosecondsPerSecond     float64 // Nanoseconds per second conversion
+	FrontendOperationDebounceMS int     // Frontend operation debounce delay
+	FrontendSyncDebounceMS      int     // Frontend sync debounce delay
+	FrontendSampleRate          int     // Frontend sample rate
+	FrontendRetryDelayMS        int     // Frontend retry delay
+	FrontendShortDelayMS        int     // Frontend short delay
+	FrontendLongDelayMS         int     // Frontend long delay
+	FrontendSyncDelayMS         int     // Frontend sync delay
+	FrontendMaxRetryAttempts    int     // Frontend max retry attempts
+	FrontendAudioLevelUpdateMS  int     // Frontend audio level update interval
+	FrontendFFTSize             int     // Frontend FFT size
+	FrontendAudioLevelMax       int     // Frontend max audio level
+	FrontendReconnectIntervalMS int     // Frontend reconnect interval
+	FrontendSubscriptionDelayMS int     // Frontend subscription delay
+	FrontendDebugIntervalMS     int     // Frontend debug interval
+
+	// Process Monitoring Constants
+	ProcessMonitorDefaultMemoryGB int     // Default memory size for fallback (4GB)
+	ProcessMonitorKBToBytes       int     // KB to bytes conversion factor (1024)
+	ProcessMonitorDefaultClockHz  float64 // Default system clock frequency (250.0 Hz)
+	ProcessMonitorFallbackClockHz float64 // Fallback clock frequency (1000.0 Hz)
+	ProcessMonitorTraditionalHz   float64 // Traditional system clock frequency (100.0 Hz)
 
 	// Batch Processing Constants
 	BatchProcessorFramesPerBatch         int           // Frames processed per batch (4)
@@ -247,21 +272,14 @@ type AudioConfigConstants struct {
 	LatencyPercentile50        int
 	LatencyPercentile95        int
 	LatencyPercentile99        int
-
+	BufferPoolMaxOperations    int
-	// Buffer Pool Configuration
+	HitRateCalculationBase     float64
-	BufferPoolDefaultSize     int     // Default buffer pool size when MaxPoolSize is invalid
+	MaxLatency                 time.Duration
-	BufferPoolControlSize     int     // Control buffer pool size
+	MinMetricsUpdateInterval   time.Duration
-	ZeroCopyPreallocSizeBytes int     // Zero-copy frame pool preallocation size in bytes
+	MaxMetricsUpdateInterval   time.Duration
-	ZeroCopyMinPreallocFrames int     // Minimum preallocated frames for zero-copy pool
+	MinSampleRate              int
-	BufferPoolHitRateBase     float64 // Base for hit rate percentage calculation
+	MaxSampleRate              int
-
+	MaxChannels                int
-	HitRateCalculationBase   float64
-	MaxLatency               time.Duration
-	MinMetricsUpdateInterval time.Duration
-	MaxMetricsUpdateInterval time.Duration
-	MinSampleRate            int
-	MaxSampleRate            int
-	MaxChannels              int
 
 	// CGO Constants
 	CGOMaxBackoffMicroseconds int // Maximum CGO backoff time (500ms)
@@ -295,22 +313,6 @@ type AudioConfigConstants struct {
 	AudioProcessorQueueSize  int
 	AudioReaderQueueSize     int
 	WorkerMaxIdleTime        time.Duration
-
-	// Connection Retry Configuration
-	MaxConnectionAttempts   int           // Maximum connection retry attempts
-	ConnectionRetryDelay    time.Duration // Initial connection retry delay
-	MaxConnectionRetryDelay time.Duration // Maximum connection retry delay
-	ConnectionBackoffFactor float64       // Connection retry backoff factor
-	ConnectionTimeoutDelay  time.Duration // Connection timeout for each attempt
-	ReconnectionInterval    time.Duration // Interval for automatic reconnection attempts
-	HealthCheckInterval     time.Duration // Health check interval for connections
-
-	// Quality Change Timeout Configuration
-	QualityChangeSupervisorTimeout time.Duration // Timeout for supervisor stop during quality changes
-	QualityChangeTickerInterval    time.Duration // Ticker interval for supervisor stop polling
-	QualityChangeSettleDelay       time.Duration // Delay for quality change to settle
-	QualityChangeRecoveryDelay     time.Duration // Delay before attempting recovery
-
 }
 
 // DefaultAudioConfig returns the default configuration constants
@@ -420,31 +422,31 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		MaxRestartDelay: 30 * time.Second, // Maximum delay for exponential backoff
 
 		// Buffer Management
-
+		PreallocSize:         1024 * 1024, // 1MB buffer preallocation
 		MaxPoolSize:          100,         // Maximum object pool size
-		MessagePoolSize:      1024,        // Significantly increased message pool for quality change bursts
+		MessagePoolSize:      256,         // Message pool size for IPC
 		OptimalSocketBuffer:  262144,      // 256KB optimal socket buffer
 		MaxSocketBuffer:      1048576,     // 1MB maximum socket buffer
 		MinSocketBuffer:      8192,        // 8KB minimum socket buffer
-		ChannelBufferSize:    2048,        // Significantly increased channel buffer for quality change bursts
+		ChannelBufferSize:    500,         // Inter-goroutine channel buffer size
 		AudioFramePoolSize:   1500,        // Audio frame object pool size
 		PageSize:             4096,        // Memory page size for alignment
-		InitialBufferFrames:  1000,        // Increased initial buffer size during startup
+		InitialBufferFrames:  500,         // Initial buffer size during startup
 		BytesToMBDivisor:     1024 * 1024, // Byte to megabyte conversion
 		MinReadEncodeBuffer:  1276,        // Minimum CGO read/encode buffer
 		MaxDecodeWriteBuffer: 4096,        // Maximum CGO decode/write buffer
 
-		// IPC Configuration - Balanced for stability
+		// IPC Configuration
-		MagicNumber:  0xDEADBEEF,              // IPC message validation header
+		MagicNumber:  0xDEADBEEF,             // IPC message validation header
-		MaxFrameSize: 4096,                    // Maximum audio frame size (4KB)
+		MaxFrameSize: 4096,                   // Maximum audio frame size (4KB)
-		WriteTimeout: 1000 * time.Millisecond, // Further increased timeout to handle quality change bursts
+		WriteTimeout: 100 * time.Millisecond, // IPC write operation timeout
-		HeaderSize:   8,                       // IPC message header size
+		HeaderSize:   8,                      // IPC message header size
 
-		// Monitoring and Metrics - Balanced for stability
+		// Monitoring and Metrics
-		MetricsUpdateInterval: 1000 * time.Millisecond, // Stable metrics collection frequency
+		MetricsUpdateInterval: 1000 * time.Millisecond, // Metrics collection frequency
-		WarmupSamples:         10,                      // Adequate warmup samples for accuracy
+		WarmupSamples:         10,                      // Warmup samples for metrics accuracy
-		MetricsChannelBuffer:  100,                     // Adequate metrics data channel buffer
+		MetricsChannelBuffer:  100,                     // Metrics data channel buffer size
-		LatencyHistorySize:    100,                     // Adequate latency measurements to keep
+		LatencyHistorySize:    100,                     // Number of latency measurements to keep
 
 		// Process Monitoring Constants
 		MaxCPUPercent:          100.0, // Maximum CPU percentage
@@ -468,50 +470,41 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		BackoffMultiplier:    2.0,                    // Exponential backoff multiplier
 		MaxConsecutiveErrors: 5,                      // Consecutive error threshold
 
-		// Connection Retry Configuration
+		// Timing Constants
-		MaxConnectionAttempts:   15,                    // Maximum connection retry attempts
+		DefaultSleepDuration:       100 * time.Millisecond, // Standard polling interval
-		ConnectionRetryDelay:    50 * time.Millisecond, // Initial connection retry delay
+		ShortSleepDuration:         10 * time.Millisecond,  // High-frequency polling
-		MaxConnectionRetryDelay: 2 * time.Second,       // Maximum connection retry delay
+		LongSleepDuration:          200 * time.Millisecond, // Background tasks
-		ConnectionBackoffFactor: 1.5,                   // Connection retry backoff factor
+		DefaultTickerInterval:      100 * time.Millisecond, // Periodic task interval
-		ConnectionTimeoutDelay:  5 * time.Second,       // Connection timeout for each attempt
+		BufferUpdateInterval:       500 * time.Millisecond, // Buffer status updates
-		ReconnectionInterval:    30 * time.Second,      // Interval for automatic reconnection attempts
+		InputSupervisorTimeout:     5 * time.Second,        // Input monitoring timeout
-		HealthCheckInterval:     10 * time.Second,      // Health check interval for connections
+		OutputSupervisorTimeout:    5 * time.Second,        // Output monitoring timeout
+		BatchProcessingDelay:       10 * time.Millisecond,  // Batch processing delay
+		AdaptiveOptimizerStability: 10 * time.Second,       // Adaptive stability period
 
-		// Quality Change Timeout Configuration
+		LatencyMonitorTarget: 50 * time.Millisecond, // Target latency for monitoring
-		QualityChangeSupervisorTimeout: 5 * time.Second,        // Timeout for supervisor stop during quality changes
-		QualityChangeTickerInterval:    100 * time.Millisecond, // Ticker interval for supervisor stop polling
-		QualityChangeSettleDelay:       2 * time.Second,        // Delay for quality change to settle
-		QualityChangeRecoveryDelay:     1 * time.Second,        // Delay before attempting recovery
 
-		// Timing Constants - Optimized for quality change stability
+		// Adaptive Buffer Configuration
-		DefaultSleepDuration:    100 * time.Millisecond, // Balanced polling interval
+		LowCPUThreshold:             0.20,
-		ShortSleepDuration:      10 * time.Millisecond,  // Balanced high-frequency polling
+		HighCPUThreshold:            0.60,
-		LongSleepDuration:       200 * time.Millisecond, // Balanced background task delay
+		LowMemoryThreshold:          0.50,
-		DefaultTickerInterval:   100 * time.Millisecond, // Balanced periodic task interval
+		HighMemoryThreshold:         0.75,
-		BufferUpdateInterval:    250 * time.Millisecond, // Faster buffer size update frequency
+		AdaptiveBufferTargetLatency: 20 * time.Millisecond,
-		InputSupervisorTimeout:  5 * time.Second,        // Input monitoring timeout
-		OutputSupervisorTimeout: 5 * time.Second,        // Output monitoring timeout
-		BatchProcessingDelay:    5 * time.Millisecond,   // Reduced batch processing delay
 
-		// Adaptive Buffer Configuration - Optimized for single-core RV1106G3
+		// Adaptive Buffer Size Configuration
-		LowCPUThreshold:             0.40, // Adjusted for single-core ARM system
+		AdaptiveMinBufferSize:     3,  // Minimum 3 frames for stability
-		HighCPUThreshold:            0.75, // Adjusted for single-core RV1106G3 (current load ~64%)
+		AdaptiveMaxBufferSize:     20, // Maximum 20 frames for high load
-		LowMemoryThreshold:          0.60,
+		AdaptiveDefaultBufferSize: 6,  // Balanced buffer size (6 frames)
-		HighMemoryThreshold:         0.85,                  // Adjusted for 200MB total memory system
-		AdaptiveBufferTargetLatency: 10 * time.Millisecond, // Aggressive target latency for responsiveness
 
-		// Adaptive Buffer Size Configuration - Optimized for quality change bursts
+		// Adaptive Optimizer Configuration
-		AdaptiveMinBufferSize:     256,  // Further increased minimum to prevent emergency mode
+		CooldownPeriod:                 30 * time.Second,
-		AdaptiveMaxBufferSize:     1024, // Much higher maximum for quality changes
+		RollbackThreshold:              300 * time.Millisecond,
-		AdaptiveDefaultBufferSize: 512,  // Higher default for stability during bursts
+		AdaptiveOptimizerLatencyTarget: 50 * time.Millisecond,
 
-		CooldownPeriod:    15 * time.Second,       // Reduced cooldown period
+		// Latency Monitor Configuration
-		RollbackThreshold: 200 * time.Millisecond, // Lower rollback threshold
+		MaxLatencyThreshold:         200 * time.Millisecond,
-
+		JitterThreshold:             20 * time.Millisecond,
-		MaxLatencyThreshold:         150 * time.Millisecond, // Lower max latency threshold
+		LatencyOptimizationInterval: 5 * time.Second,
-		JitterThreshold:             15 * time.Millisecond,  // Reduced jitter threshold
+		LatencyAdaptiveThreshold:    0.8,
-		LatencyOptimizationInterval: 3 * time.Second,        // More frequent optimization
-		LatencyAdaptiveThreshold:    0.7,                    // More aggressive adaptive threshold
 
 		// Microphone Contention Configuration
 		MicContentionTimeout: 200 * time.Millisecond,
@@ -539,25 +532,48 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		LatencyScalingFactor:    2.0, // Latency ratio scaling factor
 		OptimizerAggressiveness: 0.7, // Optimizer aggressiveness factor
 
-		// CGO Audio Processing Constants - Balanced for stability
+		// CGO Audio Processing Constants
-		CGOUsleepMicroseconds:   1000,         // 1000 microseconds (1ms) for stable CGO usleep calls
+		CGOUsleepMicroseconds:   1000,         // 1000 microseconds (1ms) for CGO usleep calls
 		CGOPCMBufferSize:        1920,         // 1920 samples for PCM buffer (max 2ch*960)
 		CGONanosecondsPerSecond: 1000000000.0, // 1000000000.0 for nanosecond conversions
 
-		// Batch Processing Constants - Optimized for quality change bursts
+		// Frontend Constants
-		BatchProcessorFramesPerBatch:         16,                    // Larger batches for quality changes
+		FrontendOperationDebounceMS: 1000,  // 1000ms debounce for frontend operations
-		BatchProcessorTimeout:                20 * time.Millisecond, // Longer timeout for bursts
+		FrontendSyncDebounceMS:      1000,  // 1000ms debounce for sync operations
-		BatchProcessorMaxQueueSize:           64,                    // Larger queue for quality changes
+		FrontendSampleRate:          48000, // 48000Hz sample rate for frontend audio
-		BatchProcessorAdaptiveThreshold:      0.6,                   // Lower threshold for faster adaptation
+		FrontendRetryDelayMS:        500,   // 500ms retry delay
-		BatchProcessorThreadPinningThreshold: 8,                     // Lower threshold for better performance
+		FrontendShortDelayMS:        200,   // 200ms short delay
+		FrontendLongDelayMS:         300,   // 300ms long delay
+		FrontendSyncDelayMS:         500,   // 500ms sync delay
+		FrontendMaxRetryAttempts:    3,     // 3 maximum retry attempts
+		FrontendAudioLevelUpdateMS:  100,   // 100ms audio level update interval
+		FrontendFFTSize:             256,   // 256 FFT size for audio analysis
+		FrontendAudioLevelMax:       100,   // 100 maximum audio level
+		FrontendReconnectIntervalMS: 3000,  // 3000ms reconnect interval
+		FrontendSubscriptionDelayMS: 100,   // 100ms subscription delay
+		FrontendDebugIntervalMS:     5000,  // 5000ms debug interval
 
-		// Output Streaming Constants - Balanced for stability
+		// Process Monitor Constants
-		OutputStreamingFrameIntervalMS: 20, // 20ms frame interval (50 FPS) for stability
+		ProcessMonitorDefaultMemoryGB: 4,      // 4GB default memory for fallback
+		ProcessMonitorKBToBytes:       1024,   // 1024 conversion factor
+		ProcessMonitorDefaultClockHz:  250.0,  // 250.0 Hz default for ARM systems
+		ProcessMonitorFallbackClockHz: 1000.0, // 1000.0 Hz fallback clock
+		ProcessMonitorTraditionalHz:   100.0,  // 100.0 Hz traditional clock
+
+		// Batch Processing Constants
+		BatchProcessorFramesPerBatch:         4,                    // 4 frames per batch
+		BatchProcessorTimeout:                5 * time.Millisecond, // 5ms timeout
+		BatchProcessorMaxQueueSize:           16,                   // 16 max queue size for balanced memory/performance
+		BatchProcessorAdaptiveThreshold:      0.8,                  // 0.8 threshold for adaptive batching (80% queue full)
+		BatchProcessorThreadPinningThreshold: 8,                    // 8 frames minimum for thread pinning optimization
+
+		// Output Streaming Constants
+		OutputStreamingFrameIntervalMS: 20, // 20ms frame interval (50 FPS)
 
 		// IPC Constants
 		IPCInitialBufferFrames: 500, // 500 frames for initial buffer
 
-		// Event Constants - Balanced for stability
+		// Event Constants
 		EventTimeoutSeconds:      2,                          // 2 seconds for event timeout
 		EventTimeFormatString:    "2006-01-02T15:04:05.000Z", // "2006-01-02T15:04:05.000Z" time format
 		EventSubscriptionDelayMS: 100,                        // 100ms subscription delay
@@ -569,7 +585,7 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		AudioReaderQueueSize:     32,               // 32 tasks queue size for reader pool
 		WorkerMaxIdleTime:        60 * time.Second, // 60s maximum idle time before worker termination
 
-		// Input Processing Constants - Balanced for stability
+		// Input Processing Constants
 		InputProcessingTimeoutMS: 10, // 10ms processing timeout threshold
 
 		// Adaptive Buffer Constants
@@ -598,15 +614,9 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		LatencyPercentile95:        95,   // 95th percentile calculation factor
 		LatencyPercentile99:        99,   // 99th percentile calculation factor
 
-		// Buffer Pool Configuration
-		BufferPoolDefaultSize:     64,          // Default buffer pool size when MaxPoolSize is invalid
-		BufferPoolControlSize:     512,         // Control buffer pool size
-		ZeroCopyPreallocSizeBytes: 1024 * 1024, // Zero-copy frame pool preallocation size in bytes (1MB)
-		ZeroCopyMinPreallocFrames: 1,           // Minimum preallocated frames for zero-copy pool
-		BufferPoolHitRateBase:     100.0,       // Base for hit rate percentage calculation
-
 		// Buffer Pool Efficiency Constants
-		HitRateCalculationBase: 100.0, // 100.0 base for hit rate percentage calculation
+		BufferPoolMaxOperations: 1000,  // 1000 operations for efficiency tracking
+		HitRateCalculationBase:  100.0, // 100.0 base for hit rate percentage calculation
 
 		// Validation Constants
 		MaxLatency:               500 * time.Millisecond, // 500ms maximum allowed latency
@@ -636,6 +646,8 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		MinFrameSize:       1,               // 1 byte minimum frame size (allow small frames)
 		FrameSizeTolerance: 512,             // 512 bytes frame size tolerance
 
+		// Removed device health monitoring configuration - functionality not used
+
 		// Latency Histogram Bucket Configuration
 		LatencyBucket10ms:  10 * time.Millisecond,  // 10ms latency bucket
 		LatencyBucket25ms:  25 * time.Millisecond,  // 25ms latency bucket
@@ -649,13 +661,16 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Batch Audio Processing Configuration
 		MinBatchSizeForThreadPinning: 5, // Minimum batch size to pin thread
 
+		// Goroutine Monitoring Configuration
+		GoroutineMonitorInterval: 30 * time.Second, // 30s monitoring interval
+
 		// Performance Configuration Flags - Production optimizations
 
 	}
 }
 
 // Global configuration instance
-var Config = DefaultAudioConfig()
+var audioConfigInstance = DefaultAudioConfig()
 
 // UpdateConfig allows runtime configuration updates
 func UpdateConfig(newConfig *AudioConfigConstants) {
@@ -667,12 +682,12 @@ func UpdateConfig(newConfig *AudioConfigConstants) {
 		return
 	}
 
-	Config = newConfig
+	audioConfigInstance = newConfig
 	logger := logging.GetDefaultLogger().With().Str("component", "AudioConfig").Logger()
 	logger.Info().Msg("Audio configuration updated successfully")
 }
 
 // GetConfig returns the current configuration
 func GetConfig() *AudioConfigConstants {
-	return Config
+	return audioConfigInstance
 }

internal/audio/core_handlers.go

@@ -29,9 +29,11 @@ func (s *AudioControlService) MuteAudio(muted bool) error {
 		supervisor := GetAudioOutputSupervisor()
 		if supervisor != nil {
 			supervisor.Stop()
+			s.logger.Info().Msg("audio output supervisor stopped")
 		}
 		StopAudioRelay()
 		SetAudioMuted(true)
+		s.logger.Info().Msg("audio output muted (subprocess and relay stopped)")
 	} else {
 		// Unmute: Start audio output subprocess and relay
 		if !s.sessionProvider.IsSessionActive() {
@@ -42,9 +44,10 @@ func (s *AudioControlService) MuteAudio(muted bool) error {
 		if supervisor != nil {
 			err := supervisor.Start()
 			if err != nil {
-				s.logger.Debug().Err(err).Msg("failed to start audio output supervisor")
+				s.logger.Error().Err(err).Msg("failed to start audio output supervisor during unmute")
 				return err
 			}
+			s.logger.Info().Msg("audio output supervisor started")
 		}
 
 		// Start audio relay

internal/audio/core_metrics.go

@@ -158,6 +158,78 @@ var (
 		},
 	)
 
+	// Audio subprocess process metrics
+	audioProcessCpuPercent = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_process_cpu_percent",
+			Help: "CPU usage percentage of audio output subprocess",
+		},
+	)
+
+	audioProcessMemoryPercent = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_process_memory_percent",
+			Help: "Memory usage percentage of audio output subprocess",
+		},
+	)
+
+	audioProcessMemoryRssBytes = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_process_memory_rss_bytes",
+			Help: "RSS memory usage in bytes of audio output subprocess",
+		},
+	)
+
+	audioProcessMemoryVmsBytes = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_process_memory_vms_bytes",
+			Help: "VMS memory usage in bytes of audio output subprocess",
+		},
+	)
+
+	audioProcessRunning = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_process_running",
+			Help: "Whether audio output subprocess is running (1=running, 0=stopped)",
+		},
+	)
+
+	// Microphone subprocess process metrics
+	microphoneProcessCpuPercent = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_microphone_process_cpu_percent",
+			Help: "CPU usage percentage of microphone input subprocess",
+		},
+	)
+
+	microphoneProcessMemoryPercent = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_microphone_process_memory_percent",
+			Help: "Memory usage percentage of microphone input subprocess",
+		},
+	)
+
+	microphoneProcessMemoryRssBytes = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_microphone_process_memory_rss_bytes",
+			Help: "RSS memory usage in bytes of microphone input subprocess",
+		},
+	)
+
+	microphoneProcessMemoryVmsBytes = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_microphone_process_memory_vms_bytes",
+			Help: "VMS memory usage in bytes of microphone input subprocess",
+		},
+	)
+
+	microphoneProcessRunning = promauto.NewGauge(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_microphone_process_running",
+			Help: "Whether microphone input subprocess is running (1=running, 0=stopped)",
+		},
+	)
+
 	// Device health metrics
 	// Removed device health metrics - functionality not used
 
@@ -374,6 +446,42 @@ func UpdateMicrophoneMetrics(metrics UnifiedAudioMetrics) {
 	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
 }
 
+// UpdateAudioProcessMetrics updates Prometheus metrics with audio subprocess data
+func UpdateAudioProcessMetrics(metrics ProcessMetrics, isRunning bool) {
+	metricsUpdateMutex.Lock()
+	defer metricsUpdateMutex.Unlock()
+
+	audioProcessCpuPercent.Set(metrics.CPUPercent)
+	audioProcessMemoryPercent.Set(metrics.MemoryPercent)
+	audioProcessMemoryRssBytes.Set(float64(metrics.MemoryRSS))
+	audioProcessMemoryVmsBytes.Set(float64(metrics.MemoryVMS))
+	if isRunning {
+		audioProcessRunning.Set(1)
+	} else {
+		audioProcessRunning.Set(0)
+	}
+
+	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
+}
+
+// UpdateMicrophoneProcessMetrics updates Prometheus metrics with microphone subprocess data
+func UpdateMicrophoneProcessMetrics(metrics ProcessMetrics, isRunning bool) {
+	metricsUpdateMutex.Lock()
+	defer metricsUpdateMutex.Unlock()
+
+	microphoneProcessCpuPercent.Set(metrics.CPUPercent)
+	microphoneProcessMemoryPercent.Set(metrics.MemoryPercent)
+	microphoneProcessMemoryRssBytes.Set(float64(metrics.MemoryRSS))
+	microphoneProcessMemoryVmsBytes.Set(float64(metrics.MemoryVMS))
+	if isRunning {
+		microphoneProcessRunning.Set(1)
+	} else {
+		microphoneProcessRunning.Set(0)
+	}
+
+	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
+}
+
 // UpdateAdaptiveBufferMetrics updates Prometheus metrics with adaptive buffer information
 func UpdateAdaptiveBufferMetrics(inputBufferSize, outputBufferSize int, cpuPercent, memoryPercent float64, adjustmentMade bool) {
 	metricsUpdateMutex.Lock()
@@ -406,7 +514,8 @@ func UpdateSocketBufferMetrics(component, bufferType string, size, utilization f
 	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
 }
 
-// UpdateDeviceHealthMetrics - Placeholder for future device health metrics
+// UpdateDeviceHealthMetrics - Device health monitoring functionality has been removed
+// This function is no longer used as device health monitoring is not implemented
 
 // UpdateMemoryMetrics updates memory metrics
 func UpdateMemoryMetrics() {

internal/audio/core_validation.go

@@ -55,11 +55,12 @@ func ValidateZeroCopyFrame(frame *ZeroCopyAudioFrame) error {
 	maxFrameSize := cachedMaxFrameSize
 	if maxFrameSize == 0 {
 		// Fallback: get from cache
-		cache := Config
+		cache := GetCachedConfig()
-		maxFrameSize = cache.MaxAudioFrameSize
+		maxFrameSize = int(cache.maxAudioFrameSize.Load())
 		if maxFrameSize == 0 {
-			// Last resort: use default
+			// Last resort: update cache
-			maxFrameSize = cache.MaxAudioFrameSize
+			cache.Update()
+			maxFrameSize = int(cache.maxAudioFrameSize.Load())
 		}
 		// Cache globally for next calls
 		cachedMaxFrameSize = maxFrameSize
@@ -72,15 +73,28 @@ func ValidateZeroCopyFrame(frame *ZeroCopyAudioFrame) error {
 }
 
 // ValidateBufferSize validates buffer size parameters with enhanced boundary checks
-// Optimized for minimal overhead in hotpath
+// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateBufferSize(size int) error {
 	if size <= 0 {
 		return fmt.Errorf("%w: buffer size %d must be positive", ErrInvalidBufferSize, size)
 	}
-	// Single boundary check using pre-cached value
+
-	if size > Config.SocketMaxBuffer {
+	// Fast path: Check against cached max frame size
+	cache := GetCachedConfig()
+	maxFrameSize := int(cache.maxAudioFrameSize.Load())
+
+	// Most common case: validating a buffer that's sized for audio frames
+	if maxFrameSize > 0 && size <= maxFrameSize {
+		return nil
+	}
+
+	// Slower path: full validation against SocketMaxBuffer
+	config := GetConfig()
+	// Use SocketMaxBuffer as the upper limit for general buffer validation
+	// This allows for socket buffers while still preventing extremely large allocations
+	if size > config.SocketMaxBuffer {
 		return fmt.Errorf("%w: buffer size %d exceeds maximum %d",
-			ErrInvalidBufferSize, size, Config.SocketMaxBuffer)
+			ErrInvalidBufferSize, size, config.SocketMaxBuffer)
 	}
 	return nil
 }
@@ -93,8 +107,8 @@ func ValidateLatency(latency time.Duration) error {
 	}
 
 	// Fast path: check against cached max latency
-	cache := Config
+	cache := GetCachedConfig()
-	maxLatency := time.Duration(cache.MaxLatency)
+	maxLatency := time.Duration(cache.maxLatency.Load())
 
 	// If we have a valid cached value, use it
 	if maxLatency > 0 {
@@ -110,14 +124,16 @@ func ValidateLatency(latency time.Duration) error {
 		return nil
 	}
 
+	// Slower path: full validation with GetConfig()
+	config := GetConfig()
 	minLatency := time.Millisecond // Minimum reasonable latency
 	if latency > 0 && latency < minLatency {
 		return fmt.Errorf("%w: latency %v below minimum %v",
 			ErrInvalidLatency, latency, minLatency)
 	}
-	if latency > Config.MaxLatency {
+	if latency > config.MaxLatency {
 		return fmt.Errorf("%w: latency %v exceeds maximum %v",
-			ErrInvalidLatency, latency, Config.MaxLatency)
+			ErrInvalidLatency, latency, config.MaxLatency)
 	}
 	return nil
 }
@@ -126,9 +142,9 @@ func ValidateLatency(latency time.Duration) error {
 // Optimized to use AudioConfigCache for frequently accessed values
 func ValidateMetricsInterval(interval time.Duration) error {
 	// Fast path: check against cached values
-	cache := Config
+	cache := GetCachedConfig()
-	minInterval := time.Duration(cache.MinMetricsUpdateInterval)
+	minInterval := time.Duration(cache.minMetricsUpdateInterval.Load())
-	maxInterval := time.Duration(cache.MaxMetricsUpdateInterval)
+	maxInterval := time.Duration(cache.maxMetricsUpdateInterval.Load())
 
 	// If we have valid cached values, use them
 	if minInterval > 0 && maxInterval > 0 {
@@ -143,8 +159,10 @@ func ValidateMetricsInterval(interval time.Duration) error {
 		return nil
 	}
 
-	minInterval = Config.MinMetricsUpdateInterval
+	// Slower path: full validation with GetConfig()
-	maxInterval = Config.MaxMetricsUpdateInterval
+	config := GetConfig()
+	minInterval = config.MinMetricsUpdateInterval
+	maxInterval = config.MaxMetricsUpdateInterval
 	if interval < minInterval {
 		return ErrInvalidMetricsInterval
 	}
@@ -166,7 +184,7 @@ func ValidateAdaptiveBufferConfig(minSize, maxSize, defaultSize int) error {
 		return ErrInvalidBufferSize
 	}
 	// Validate against global limits
-	maxBuffer := Config.SocketMaxBuffer
+	maxBuffer := GetConfig().SocketMaxBuffer
 	if maxSize > maxBuffer {
 		return ErrInvalidBufferSize
 	}
@@ -175,9 +193,11 @@ func ValidateAdaptiveBufferConfig(minSize, maxSize, defaultSize int) error {
 
 // ValidateInputIPCConfig validates input IPC configuration
 func ValidateInputIPCConfig(sampleRate, channels, frameSize int) error {
-	minSampleRate := Config.MinSampleRate
+	// Use config values
-	maxSampleRate := Config.MaxSampleRate
+	config := GetConfig()
-	maxChannels := Config.MaxChannels
+	minSampleRate := config.MinSampleRate
+	maxSampleRate := config.MaxSampleRate
+	maxChannels := config.MaxChannels
 	if sampleRate < minSampleRate || sampleRate > maxSampleRate {
 		return ErrInvalidSampleRate
 	}
@@ -192,9 +212,11 @@ func ValidateInputIPCConfig(sampleRate, channels, frameSize int) error {
 
 // ValidateOutputIPCConfig validates output IPC configuration
 func ValidateOutputIPCConfig(sampleRate, channels, frameSize int) error {
-	minSampleRate := Config.MinSampleRate
+	// Use config values
-	maxSampleRate := Config.MaxSampleRate
+	config := GetConfig()
-	maxChannels := Config.MaxChannels
+	minSampleRate := config.MinSampleRate
+	maxSampleRate := config.MaxSampleRate
+	maxChannels := config.MaxChannels
 	if sampleRate < minSampleRate || sampleRate > maxSampleRate {
 		return ErrInvalidSampleRate
 	}
@@ -207,51 +229,130 @@ func ValidateOutputIPCConfig(sampleRate, channels, frameSize int) error {
 	return nil
 }
 
+// ValidateLatencyConfig validates latency monitor configuration
+func ValidateLatencyConfig(config LatencyConfig) error {
+	if err := ValidateLatency(config.TargetLatency); err != nil {
+		return err
+	}
+	if err := ValidateLatency(config.MaxLatency); err != nil {
+		return err
+	}
+	if config.TargetLatency >= config.MaxLatency {
+		return ErrInvalidLatency
+	}
+	if err := ValidateMetricsInterval(config.OptimizationInterval); err != nil {
+		return err
+	}
+	if config.HistorySize <= 0 {
+		return ErrInvalidBufferSize
+	}
+	if config.JitterThreshold < 0 {
+		return ErrInvalidLatency
+	}
+	if config.AdaptiveThreshold < 0 || config.AdaptiveThreshold > 1.0 {
+		return ErrInvalidConfiguration
+	}
+	return nil
+}
+
 // ValidateSampleRate validates audio sample rate values
-// Optimized for minimal overhead in hotpath
+// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateSampleRate(sampleRate int) error {
 	if sampleRate <= 0 {
 		return fmt.Errorf("%w: sample rate %d must be positive", ErrInvalidSampleRate, sampleRate)
 	}
-	// Direct validation against valid rates
+
-	for _, rate := range Config.ValidSampleRates {
+	// Fast path: Check against cached sample rate first
+	cache := GetCachedConfig()
+	cachedRate := int(cache.sampleRate.Load())
+
+	// Most common case: validating against the current sample rate
+	if sampleRate == cachedRate {
+		return nil
+	}
+
+	// Slower path: check against all valid rates
+	config := GetConfig()
+	validRates := config.ValidSampleRates
+	for _, rate := range validRates {
 		if sampleRate == rate {
 			return nil
 		}
 	}
-	return fmt.Errorf("%w: sample rate %d not in valid rates %v",
+	return fmt.Errorf("%w: sample rate %d not in supported rates %v",
-		ErrInvalidSampleRate, sampleRate, Config.ValidSampleRates)
+		ErrInvalidSampleRate, sampleRate, validRates)
 }
 
 // ValidateChannelCount validates audio channel count
-// Optimized for minimal overhead in hotpath
+// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateChannelCount(channels int) error {
 	if channels <= 0 {
 		return fmt.Errorf("%w: channel count %d must be positive", ErrInvalidChannels, channels)
 	}
-	// Direct boundary check
+
-	if channels > Config.MaxChannels {
+	// Fast path: Check against cached channels first
+	cache := GetCachedConfig()
+	cachedChannels := int(cache.channels.Load())
+
+	// Most common case: validating against the current channel count
+	if channels == cachedChannels {
+		return nil
+	}
+
+	// Fast path: Check against cached max channels
+	cachedMaxChannels := int(cache.maxChannels.Load())
+	if cachedMaxChannels > 0 && channels <= cachedMaxChannels {
+		return nil
+	}
+
+	// Slow path: Update cache and validate
+	cache.Update()
+	updatedMaxChannels := int(cache.maxChannels.Load())
+	if channels > updatedMaxChannels {
 		return fmt.Errorf("%w: channel count %d exceeds maximum %d",
-			ErrInvalidChannels, channels, Config.MaxChannels)
+			ErrInvalidChannels, channels, updatedMaxChannels)
 	}
 	return nil
 }
 
 // ValidateBitrate validates audio bitrate values (expects kbps)
-// Optimized for minimal overhead in hotpath
+// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateBitrate(bitrate int) error {
 	if bitrate <= 0 {
 		return fmt.Errorf("%w: bitrate %d must be positive", ErrInvalidBitrate, bitrate)
 	}
-	// Direct boundary check with single conversion
+
-	bitrateInBps := bitrate * 1000
+	// Fast path: Check against cached bitrate values
-	if bitrateInBps < Config.MinOpusBitrate {
+	cache := GetCachedConfig()
-		return fmt.Errorf("%w: bitrate %d kbps (%d bps) below minimum %d bps",
+	minBitrate := int(cache.minOpusBitrate.Load())
-			ErrInvalidBitrate, bitrate, bitrateInBps, Config.MinOpusBitrate)
+	maxBitrate := int(cache.maxOpusBitrate.Load())
+
+	// If we have valid cached values, use them
+	if minBitrate > 0 && maxBitrate > 0 {
+		// Convert kbps to bps for comparison with config limits
+		bitrateInBps := bitrate * 1000
+		if bitrateInBps < minBitrate {
+			return fmt.Errorf("%w: bitrate %d kbps (%d bps) below minimum %d bps",
+				ErrInvalidBitrate, bitrate, bitrateInBps, minBitrate)
+		}
+		if bitrateInBps > maxBitrate {
+			return fmt.Errorf("%w: bitrate %d kbps (%d bps) exceeds maximum %d bps",
+				ErrInvalidBitrate, bitrate, bitrateInBps, maxBitrate)
+		}
+		return nil
 	}
-	if bitrateInBps > Config.MaxOpusBitrate {
+
+	// Slower path: full validation with GetConfig()
+	config := GetConfig()
+	// Convert kbps to bps for comparison with config limits
+	bitrateInBps := bitrate * 1000
+	if bitrateInBps < config.MinOpusBitrate {
+		return fmt.Errorf("%w: bitrate %d kbps (%d bps) below minimum %d bps",
+			ErrInvalidBitrate, bitrate, bitrateInBps, config.MinOpusBitrate)
+	}
+	if bitrateInBps > config.MaxOpusBitrate {
 		return fmt.Errorf("%w: bitrate %d kbps (%d bps) exceeds maximum %d bps",
-			ErrInvalidBitrate, bitrate, bitrateInBps, Config.MaxOpusBitrate)
+			ErrInvalidBitrate, bitrate, bitrateInBps, config.MaxOpusBitrate)
 	}
 	return nil
 }
@@ -264,11 +365,11 @@ func ValidateFrameDuration(duration time.Duration) error {
 	}
 
 	// Fast path: Check against cached frame size first
-	cache := Config
+	cache := GetCachedConfig()
 
 	// Convert frameSize (samples) to duration for comparison
-	cachedFrameSize := cache.FrameSize
+	cachedFrameSize := int(cache.frameSize.Load())
-	cachedSampleRate := cache.SampleRate
+	cachedSampleRate := int(cache.sampleRate.Load())
 
 	// Only do this calculation if we have valid cached values
 	if cachedFrameSize > 0 && cachedSampleRate > 0 {
@@ -281,8 +382,8 @@ func ValidateFrameDuration(duration time.Duration) error {
 	}
 
 	// Fast path: Check against cached min/max frame duration
-	cachedMinDuration := time.Duration(cache.MinFrameDuration)
+	cachedMinDuration := time.Duration(cache.minFrameDuration.Load())
-	cachedMaxDuration := time.Duration(cache.MaxFrameDuration)
+	cachedMaxDuration := time.Duration(cache.maxFrameDuration.Load())
 
 	if cachedMinDuration > 0 && cachedMaxDuration > 0 {
 		if duration < cachedMinDuration {
@@ -296,9 +397,10 @@ func ValidateFrameDuration(duration time.Duration) error {
 		return nil
 	}
 
-	// Slow path: Use current config values
+	// Slow path: Update cache and validate
-	updatedMinDuration := time.Duration(cache.MinFrameDuration)
+	cache.Update()
-	updatedMaxDuration := time.Duration(cache.MaxFrameDuration)
+	updatedMinDuration := time.Duration(cache.minFrameDuration.Load())
+	updatedMaxDuration := time.Duration(cache.maxFrameDuration.Load())
 
 	if duration < updatedMinDuration {
 		return fmt.Errorf("%w: frame duration %v below minimum %v",
@@ -315,11 +417,11 @@ func ValidateFrameDuration(duration time.Duration) error {
 // Uses optimized validation functions that leverage AudioConfigCache
 func ValidateAudioConfigComplete(config AudioConfig) error {
 	// Fast path: Check if all values match the current cached configuration
-	cache := Config
+	cache := GetCachedConfig()
-	cachedSampleRate := cache.SampleRate
+	cachedSampleRate := int(cache.sampleRate.Load())
-	cachedChannels := cache.Channels
+	cachedChannels := int(cache.channels.Load())
-	cachedBitrate := cache.OpusBitrate / 1000 // Convert from bps to kbps
+	cachedBitrate := int(cache.opusBitrate.Load()) / 1000 // Convert from bps to kbps
-	cachedFrameSize := cache.FrameSize
+	cachedFrameSize := int(cache.frameSize.Load())
 
 	// Only do this calculation if we have valid cached values
 	if cachedSampleRate > 0 && cachedChannels > 0 && cachedBitrate > 0 && cachedFrameSize > 0 {
@@ -363,11 +465,11 @@ func ValidateAudioConfigConstants(config *AudioConfigConstants) error {
 	}
 	// Validate configuration values if config is provided
 	if config != nil {
-		if Config.MaxFrameSize <= 0 {
+		if config.MaxFrameSize <= 0 {
-			return fmt.Errorf("invalid MaxFrameSize: %d", Config.MaxFrameSize)
+			return fmt.Errorf("invalid MaxFrameSize: %d", config.MaxFrameSize)
 		}
-		if Config.SampleRate <= 0 {
+		if config.SampleRate <= 0 {
-			return fmt.Errorf("invalid SampleRate: %d", Config.SampleRate)
+			return fmt.Errorf("invalid SampleRate: %d", config.SampleRate)
 		}
 	}
 	return nil
@@ -379,10 +481,11 @@ var cachedMaxFrameSize int
 // InitValidationCache initializes cached validation values with actual config
 func InitValidationCache() {
 	// Initialize the global cache variable for backward compatibility
-	cachedMaxFrameSize = Config.MaxAudioFrameSize
+	config := GetConfig()
+	cachedMaxFrameSize = config.MaxAudioFrameSize
 
-	// Initialize the global audio config cache
+	// Update the global audio config cache
-	cachedMaxFrameSize = Config.MaxAudioFrameSize
+	GetCachedConfig().Update()
 }
 
 // ValidateAudioFrame validates audio frame data with cached max size for performance
@@ -399,11 +502,12 @@ func ValidateAudioFrame(data []byte) error {
 	maxSize := cachedMaxFrameSize
 	if maxSize == 0 {
 		// Fallback: get from cache only if global cache not initialized
-		cache := Config
+		cache := GetCachedConfig()
-		maxSize = cache.MaxAudioFrameSize
+		maxSize = int(cache.maxAudioFrameSize.Load())
 		if maxSize == 0 {
-			// Last resort: get fresh value
+			// Last resort: update cache and get fresh value
-			maxSize = cache.MaxAudioFrameSize
+			cache.Update()
+			maxSize = int(cache.maxAudioFrameSize.Load())
 		}
 		// Cache the value globally for next calls
 		cachedMaxFrameSize = maxSize

internal/audio/goroutine_pool.go

@@ -65,42 +65,6 @@ func (p *GoroutinePool) Submit(task Task) bool {
 	}
 }
 
-// SubmitWithBackpressure adds a task to the pool with backpressure handling
-// Returns true if task was accepted, false if dropped due to backpressure
-func (p *GoroutinePool) SubmitWithBackpressure(task Task) bool {
-	select {
-	case <-p.shutdown:
-		return false // Pool is shutting down
-	case p.taskQueue <- task:
-		// Task accepted, ensure we have a worker to process it
-		p.ensureWorkerAvailable()
-		return true
-	default:
-		// Queue is full - apply backpressure
-		// Check if we're in a high-load situation
-		queueLen := len(p.taskQueue)
-		queueCap := cap(p.taskQueue)
-		workerCount := atomic.LoadInt64(&p.workerCount)
-
-		// If queue is >90% full and we're at max workers, drop the task
-		if queueLen > int(float64(queueCap)*0.9) && workerCount >= int64(p.maxWorkers) {
-			p.logger.Warn().Int("queue_len", queueLen).Int("queue_cap", queueCap).Msg("Dropping task due to backpressure")
-			return false
-		}
-
-		// Try one more time with a short timeout
-		select {
-		case p.taskQueue <- task:
-			p.ensureWorkerAvailable()
-			return true
-		case <-time.After(1 * time.Millisecond):
-			// Still can't submit after timeout - drop task
-			p.logger.Debug().Msg("Task dropped after backpressure timeout")
-			return false
-		}
-	}
-}
-
 // ensureWorkerAvailable makes sure at least one worker is available to process tasks
 func (p *GoroutinePool) ensureWorkerAvailable() {
 	// Check if we already have enough workers
@@ -196,7 +160,7 @@ func (p *GoroutinePool) supervisor() {
 			tasks := atomic.LoadInt64(&p.taskCount)
 			queueLen := len(p.taskQueue)
 
-			p.logger.Debug().
+			p.logger.Info().
 				Int64("workers", workers).
 				Int64("tasks_processed", tasks).
 				Int("queue_length", queueLen).
@@ -215,7 +179,7 @@ func (p *GoroutinePool) Shutdown(wait bool) {
 		if wait {
 			// Wait for all tasks to be processed
 			if len(p.taskQueue) > 0 {
-				p.logger.Debug().Int("remaining_tasks", len(p.taskQueue)).Msg("Waiting for tasks to complete")
+				p.logger.Info().Int("remaining_tasks", len(p.taskQueue)).Msg("Waiting for tasks to complete")
 			}
 
 			// Close the task queue to signal no more tasks
@@ -255,7 +219,7 @@ func GetAudioProcessorPool() *GoroutinePool {
 	}
 
 	globalAudioProcessorInitOnce.Do(func() {
-		config := Config
+		config := GetConfig()
 		newPool := NewGoroutinePool(
 			"audio-processor",
 			config.MaxAudioProcessorWorkers,
@@ -277,7 +241,7 @@ func GetAudioReaderPool() *GoroutinePool {
 	}
 
 	globalAudioReaderInitOnce.Do(func() {
-		config := Config
+		config := GetConfig()
 		newPool := NewGoroutinePool(
 			"audio-reader",
 			config.MaxAudioReaderWorkers,
@@ -301,16 +265,6 @@ func SubmitAudioReaderTask(task Task) bool {
 	return GetAudioReaderPool().Submit(task)
 }
 
-// SubmitAudioProcessorTaskWithBackpressure submits a task with backpressure handling
-func SubmitAudioProcessorTaskWithBackpressure(task Task) bool {
-	return GetAudioProcessorPool().SubmitWithBackpressure(task)
-}
-
-// SubmitAudioReaderTaskWithBackpressure submits a task with backpressure handling
-func SubmitAudioReaderTaskWithBackpressure(task Task) bool {
-	return GetAudioReaderPool().SubmitWithBackpressure(task)
-}
-
 // ShutdownAudioPools shuts down all audio goroutine pools
 func ShutdownAudioPools(wait bool) {
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-pools").Logger()

internal/audio/input_microphone_manager.go

@@ -108,13 +108,14 @@ func (aim *AudioInputManager) WriteOpusFrame(frame []byte) error {
 	processingTime := time.Since(startTime)
 
 	// Log high latency warnings
-	if processingTime > time.Duration(Config.InputProcessingTimeoutMS)*time.Millisecond {
+	if processingTime > time.Duration(GetConfig().InputProcessingTimeoutMS)*time.Millisecond {
 		latencyMs := float64(processingTime.Milliseconds())
 		aim.logger.Warn().
 			Float64("latency_ms", latencyMs).
 			Msg("High audio processing latency detected")
 
 		// Record latency for goroutine cleanup optimization
+		RecordAudioLatency(latencyMs)
 	}
 
 	if err != nil {
@@ -148,13 +149,14 @@ func (aim *AudioInputManager) WriteOpusFrameZeroCopy(frame *ZeroCopyAudioFrame)
 	processingTime := time.Since(startTime)
 
 	// Log high latency warnings
-	if processingTime > time.Duration(Config.InputProcessingTimeoutMS)*time.Millisecond {
+	if processingTime > time.Duration(GetConfig().InputProcessingTimeoutMS)*time.Millisecond {
 		latencyMs := float64(processingTime.Milliseconds())
 		aim.logger.Warn().
 			Float64("latency_ms", latencyMs).
 			Msg("High audio processing latency detected")
 
 		// Record latency for goroutine cleanup optimization
+		RecordAudioLatency(latencyMs)
 	}
 
 	if err != nil {

internal/audio/input_server_main.go

@@ -19,28 +19,6 @@ import (
 	"github.com/jetkvm/kvm/internal/logging"
 )
 
-// Global audio input server instance
-var globalAudioInputServer *AudioInputServer
-
-// GetGlobalAudioInputServer returns the global audio input server instance
-func GetGlobalAudioInputServer() *AudioInputServer {
-	return globalAudioInputServer
-}
-
-// ResetGlobalAudioInputServerStats resets the global audio input server stats
-func ResetGlobalAudioInputServerStats() {
-	if globalAudioInputServer != nil {
-		globalAudioInputServer.ResetServerStats()
-	}
-}
-
-// RecoverGlobalAudioInputServer attempts to recover from dropped frames
-func RecoverGlobalAudioInputServer() {
-	if globalAudioInputServer != nil {
-		globalAudioInputServer.RecoverFromDroppedFrames()
-	}
-}
-
 // getEnvInt reads an integer from environment variable with a default value
 
 // RunAudioInputServer runs the audio input server subprocess
@@ -55,6 +33,10 @@ func RunAudioInputServer() error {
 	// Initialize validation cache for optimal performance
 	InitValidationCache()
 
+	// Start adaptive buffer management for optimal performance
+	StartAdaptiveBuffering()
+	defer StopAdaptiveBuffering()
+
 	// Initialize CGO audio playback (optional for input server)
 	// This is used for audio loopback/monitoring features
 	err := CGOAudioPlaybackInit()
@@ -74,9 +56,6 @@ func RunAudioInputServer() error {
 	}
 	defer server.Close()
 
-	// Store globally for access by other functions
-	globalAudioInputServer = server
-
 	err = server.Start()
 	if err != nil {
 		logger.Error().Err(err).Msg("failed to start audio input server")
@@ -103,7 +82,7 @@ func RunAudioInputServer() error {
 	server.Stop()
 
 	// Give some time for cleanup
-	time.Sleep(Config.DefaultSleepDuration)
+	time.Sleep(GetConfig().DefaultSleepDuration)
 
 	return nil
 }

internal/audio/input_supervisor.go

@@ -73,7 +73,7 @@ func (ais *AudioInputSupervisor) supervisionLoop() {
 	// Configure supervision parameters (no restart for input supervisor)
 	config := SupervisionConfig{
 		ProcessType:        "audio input server",
-		Timeout:            Config.InputSupervisorTimeout,
+		Timeout:            GetConfig().InputSupervisorTimeout,
 		EnableRestart:      false, // Input supervisor doesn't restart
 		MaxRestartAttempts: 0,
 		RestartWindow:      0,
@@ -135,9 +135,10 @@ func (ais *AudioInputSupervisor) startProcess() error {
 	ais.logger.Info().Int("pid", ais.processPID).Strs("args", args).Strs("opus_env", ais.opusEnv).Msg("audio input server process started")
 
 	// Add process to monitoring
+	ais.processMonitor.AddProcess(ais.processPID, "audio-input-server")
 
-	// Connect client to the server synchronously to avoid race condition
+	// Connect client to the server
-	ais.connectClient()
+	go ais.connectClient()
 
 	return nil
 }
@@ -163,7 +164,7 @@ func (ais *AudioInputSupervisor) Stop() {
 	select {
 	case <-ais.processDone:
 		ais.logger.Info().Str("component", "audio-input-supervisor").Msg("component stopped gracefully")
-	case <-time.After(Config.InputSupervisorTimeout):
+	case <-time.After(GetConfig().InputSupervisorTimeout):
 		ais.logger.Warn().Str("component", "audio-input-supervisor").Msg("component did not stop gracefully, forcing termination")
 		ais.forceKillProcess("audio input server")
 	}
@@ -189,7 +190,7 @@ func (ais *AudioInputSupervisor) GetClient() *AudioInputClient {
 // connectClient attempts to connect the client to the server
 func (ais *AudioInputSupervisor) connectClient() {
 	// Wait briefly for the server to start and create socket
-	time.Sleep(Config.DefaultSleepDuration)
+	time.Sleep(GetConfig().DefaultSleepDuration)
 
 	// Additional small delay to ensure socket is ready after restart
 	time.Sleep(20 * time.Millisecond)

internal/audio/ipc_common.go

@@ -49,7 +49,7 @@ func NewGenericMessagePool(size int) *GenericMessagePool {
 	pool.preallocated = make([]*OptimizedMessage, pool.preallocSize)
 	for i := 0; i < pool.preallocSize; i++ {
 		pool.preallocated[i] = &OptimizedMessage{
-			data: make([]byte, 0, Config.MaxFrameSize),
+			data: make([]byte, 0, GetConfig().MaxFrameSize),
 		}
 	}
 
@@ -57,7 +57,7 @@ func NewGenericMessagePool(size int) *GenericMessagePool {
 	for i := 0; i < size-pool.preallocSize; i++ {
 		select {
 		case pool.pool <- &OptimizedMessage{
-			data: make([]byte, 0, Config.MaxFrameSize),
+			data: make([]byte, 0, GetConfig().MaxFrameSize),
 		}:
 		default:
 			break
@@ -89,7 +89,7 @@ func (mp *GenericMessagePool) Get() *OptimizedMessage {
 		// Pool empty, create new message
 		atomic.AddInt64(&mp.missCount, 1)
 		return &OptimizedMessage{
-			data: make([]byte, 0, Config.MaxFrameSize),
+			data: make([]byte, 0, GetConfig().MaxFrameSize),
 		}
 	}
 }
@@ -132,42 +132,6 @@ func (mp *GenericMessagePool) GetStats() (hitCount, missCount int64, hitRate flo
 	return hits, misses, hitRate
 }
 
-// Helper functions
-
-// EncodeMessageHeader encodes a message header into a byte slice
-func EncodeMessageHeader(magic uint32, msgType uint8, length uint32, timestamp int64) []byte {
-	header := make([]byte, 17)
-	binary.LittleEndian.PutUint32(header[0:4], magic)
-	header[4] = msgType
-	binary.LittleEndian.PutUint32(header[5:9], length)
-	binary.LittleEndian.PutUint64(header[9:17], uint64(timestamp))
-	return header
-}
-
-// EncodeAudioConfig encodes basic audio configuration to binary format
-func EncodeAudioConfig(sampleRate, channels, frameSize int) []byte {
-	data := make([]byte, 12) // 3 * int32
-	binary.LittleEndian.PutUint32(data[0:4], uint32(sampleRate))
-	binary.LittleEndian.PutUint32(data[4:8], uint32(channels))
-	binary.LittleEndian.PutUint32(data[8:12], uint32(frameSize))
-	return data
-}
-
-// EncodeOpusConfig encodes complete Opus configuration to binary format
-func EncodeOpusConfig(sampleRate, channels, frameSize, bitrate, complexity, vbr, signalType, bandwidth, dtx int) []byte {
-	data := make([]byte, 36) // 9 * int32
-	binary.LittleEndian.PutUint32(data[0:4], uint32(sampleRate))
-	binary.LittleEndian.PutUint32(data[4:8], uint32(channels))
-	binary.LittleEndian.PutUint32(data[8:12], uint32(frameSize))
-	binary.LittleEndian.PutUint32(data[12:16], uint32(bitrate))
-	binary.LittleEndian.PutUint32(data[16:20], uint32(complexity))
-	binary.LittleEndian.PutUint32(data[20:24], uint32(vbr))
-	binary.LittleEndian.PutUint32(data[24:28], uint32(signalType))
-	binary.LittleEndian.PutUint32(data[28:32], uint32(bandwidth))
-	binary.LittleEndian.PutUint32(data[32:36], uint32(dtx))
-	return data
-}
-
 // Common write message function
 func WriteIPCMessage(conn net.Conn, msg IPCMessage, pool *GenericMessagePool, droppedFramesCounter *int64) error {
 	if conn == nil {
@@ -179,11 +143,13 @@ func WriteIPCMessage(conn net.Conn, msg IPCMessage, pool *GenericMessagePool, dr
 	defer pool.Put(optMsg)
 
 	// Prepare header in pre-allocated buffer
-	header := EncodeMessageHeader(msg.GetMagic(), msg.GetType(), msg.GetLength(), msg.GetTimestamp())
+	binary.LittleEndian.PutUint32(optMsg.header[0:4], msg.GetMagic())
-	copy(optMsg.header[:], header)
+	optMsg.header[4] = msg.GetType()
+	binary.LittleEndian.PutUint32(optMsg.header[5:9], msg.GetLength())
+	binary.LittleEndian.PutUint64(optMsg.header[9:17], uint64(msg.GetTimestamp()))
 
 	// Set write deadline for timeout handling (more efficient than goroutines)
-	if deadline := time.Now().Add(Config.WriteTimeout); deadline.After(time.Now()) {
+	if deadline := time.Now().Add(GetConfig().WriteTimeout); deadline.After(time.Now()) {
 		if err := conn.SetWriteDeadline(deadline); err != nil {
 			// If we can't set deadline, proceed without it
 			// This maintains compatibility with connections that don't support deadlines

internal/audio/ipc_input.go

@@ -23,8 +23,8 @@ const (
 
 // Constants are now defined in unified_ipc.go
 var (
-	maxFrameSize    = Config.MaxFrameSize    // Maximum Opus frame size
+	maxFrameSize    = GetConfig().MaxFrameSize    // Maximum Opus frame size
-	messagePoolSize = Config.MessagePoolSize // Pre-allocated message pool size
+	messagePoolSize = GetConfig().MessagePoolSize // Pre-allocated message pool size
 )
 
 // Legacy aliases for backward compatibility
@@ -77,7 +77,7 @@ func initializeMessagePool() {
 	messagePoolInitOnce.Do(func() {
 		preallocSize := messagePoolSize / 4 // 25% pre-allocated for immediate use
 		globalMessagePool.preallocSize = preallocSize
-		globalMessagePool.maxPoolSize = messagePoolSize * Config.PoolGrowthMultiplier // Allow growth up to 2x
+		globalMessagePool.maxPoolSize = messagePoolSize * GetConfig().PoolGrowthMultiplier // Allow growth up to 2x
 		globalMessagePool.preallocated = make([]*OptimizedIPCMessage, 0, preallocSize)
 
 		// Pre-allocate messages for immediate use
@@ -191,10 +191,6 @@ type AudioInputServer struct {
 	stopChan    chan struct{}         // Stop signal for all goroutines
 	wg          sync.WaitGroup        // Wait group for goroutine coordination
 
-	// Channel resizing support
-	channelMutex   sync.RWMutex // Protects channel recreation
-	lastBufferSize int64        // Last known buffer size for change detection
-
 	// Socket buffer configuration
 	socketBufferConfig SocketBufferConfig
 }
@@ -231,15 +227,9 @@ func NewAudioInputServer() (*AudioInputServer, error) {
 		return nil, fmt.Errorf("failed to create unix socket after 3 attempts: %w", err)
 	}
 
-	// Get initial buffer size from config
+	// Get initial buffer size from adaptive buffer manager
-	initialBufferSize := int64(Config.AdaptiveDefaultBufferSize)
+	adaptiveManager := GetAdaptiveBufferManager()
-
+	initialBufferSize := int64(adaptiveManager.GetInputBufferSize())
-	// Ensure minimum buffer size to prevent immediate overflow
-	// Use at least 50 frames to handle burst traffic
-	minBufferSize := int64(50)
-	if initialBufferSize < minBufferSize {
-		initialBufferSize = minBufferSize
-	}
 
 	// Initialize socket buffer configuration
 	socketBufferConfig := DefaultSocketBufferConfig()
@@ -250,7 +240,6 @@ func NewAudioInputServer() (*AudioInputServer, error) {
 		processChan:        make(chan *InputIPCMessage, initialBufferSize),
 		stopChan:           make(chan struct{}),
 		bufferSize:         initialBufferSize,
-		lastBufferSize:     initialBufferSize,
 		socketBufferConfig: socketBufferConfig,
 	}, nil
 }
@@ -377,7 +366,7 @@ func (ais *AudioInputServer) handleConnection(conn net.Conn) {
 			if ais.conn == nil {
 				return
 			}
-			time.Sleep(Config.DefaultSleepDuration)
+			time.Sleep(GetConfig().DefaultSleepDuration)
 		}
 	}
 }
@@ -498,11 +487,11 @@ func (ais *AudioInputServer) processOpusFrame(data []byte) error {
 	}
 
 	// Get cached config once - avoid repeated calls and locking
-	cache := Config
+	cache := GetCachedConfig()
 	// Skip cache expiry check in hotpath - background updates handle this
 
 	// Get a PCM buffer from the pool for optimized decode-write
-	pcmBuffer := GetBufferFromPool(cache.MaxPCMBufferSize)
+	pcmBuffer := GetBufferFromPool(cache.GetMaxPCMBufferSize())
 	defer ReturnBufferToPool(pcmBuffer)
 
 	// Direct CGO call - avoid wrapper function overhead
@@ -645,9 +634,9 @@ func (aic *AudioInputClient) Connect() error {
 			return nil
 		}
 		// Exponential backoff starting from config
-		backoffStart := Config.BackoffStart
+		backoffStart := GetConfig().BackoffStart
 		delay := time.Duration(backoffStart.Nanoseconds()*(1<<uint(i/3))) * time.Nanosecond
-		maxDelay := Config.MaxRetryDelay
+		maxDelay := GetConfig().MaxRetryDelay
 		if delay > maxDelay {
 			delay = maxDelay
 		}
@@ -688,28 +677,32 @@ func (aic *AudioInputClient) Disconnect() {
 
 // SendFrame sends an Opus frame to the audio input server
 func (aic *AudioInputClient) SendFrame(frame []byte) error {
-	// Fast path validation
-	if len(frame) == 0 {
-		return nil
-	}
-
 	aic.mtx.Lock()
+	defer aic.mtx.Unlock()
+
 	if !aic.running || aic.conn == nil {
-		aic.mtx.Unlock()
+		return fmt.Errorf("not connected to audio input server")
-		return fmt.Errorf("not connected")
+	}
+
+	frameLen := len(frame)
+	if frameLen == 0 {
+		return nil // Empty frame, ignore
+	}
+
+	// Inline frame validation to reduce function call overhead
+	if frameLen > maxFrameSize {
+		return ErrFrameDataTooLarge
 	}
 
-	// Direct message creation without timestamp overhead
 	msg := &InputIPCMessage{
-		Magic:  inputMagicNumber,
+		Magic:     inputMagicNumber,
-		Type:   InputMessageTypeOpusFrame,
+		Type:      InputMessageTypeOpusFrame,
-		Length: uint32(len(frame)),
+		Length:    uint32(frameLen),
-		Data:   frame,
+		Timestamp: time.Now().UnixNano(),
+		Data:      frame,
 	}
 
-	err := aic.writeMessage(msg)
+	return aic.writeMessage(msg)
-	aic.mtx.Unlock()
-	return err
 }
 
 // SendFrameZeroCopy sends a zero-copy Opus frame to the audio input server
@@ -763,8 +756,11 @@ func (aic *AudioInputClient) SendConfig(config InputIPCConfig) error {
 		return fmt.Errorf("input configuration validation failed: %w", err)
 	}
 
-	// Serialize config using common function
+	// Serialize config (simple binary format)
-	data := EncodeAudioConfig(config.SampleRate, config.Channels, config.FrameSize)
+	data := make([]byte, 12) // 3 * int32
+	binary.LittleEndian.PutUint32(data[0:4], uint32(config.SampleRate))
+	binary.LittleEndian.PutUint32(data[4:8], uint32(config.Channels))
+	binary.LittleEndian.PutUint32(data[8:12], uint32(config.FrameSize))
 
 	msg := &InputIPCMessage{
 		Magic:     inputMagicNumber,
@@ -792,8 +788,17 @@ func (aic *AudioInputClient) SendOpusConfig(config InputIPCOpusConfig) error {
 			config.SampleRate, config.Channels, config.FrameSize, config.Bitrate)
 	}
 
-	// Serialize Opus configuration using common function
+	// Serialize Opus configuration (9 * int32 = 36 bytes)
-	data := EncodeOpusConfig(config.SampleRate, config.Channels, config.FrameSize, config.Bitrate, config.Complexity, config.VBR, config.SignalType, config.Bandwidth, config.DTX)
+	data := make([]byte, 36)
+	binary.LittleEndian.PutUint32(data[0:4], uint32(config.SampleRate))
+	binary.LittleEndian.PutUint32(data[4:8], uint32(config.Channels))
+	binary.LittleEndian.PutUint32(data[8:12], uint32(config.FrameSize))
+	binary.LittleEndian.PutUint32(data[12:16], uint32(config.Bitrate))
+	binary.LittleEndian.PutUint32(data[16:20], uint32(config.Complexity))
+	binary.LittleEndian.PutUint32(data[20:24], uint32(config.VBR))
+	binary.LittleEndian.PutUint32(data[24:28], uint32(config.SignalType))
+	binary.LittleEndian.PutUint32(data[28:32], uint32(config.Bandwidth))
+	binary.LittleEndian.PutUint32(data[32:36], uint32(config.DTX))
 
 	msg := &InputIPCMessage{
 		Magic:     inputMagicNumber,
@@ -861,28 +866,6 @@ func (aic *AudioInputClient) ResetStats() {
 	ResetFrameStats(&aic.totalFrames, &aic.droppedFrames)
 }
 
-// ResetServerStats resets server frame statistics
-func (ais *AudioInputServer) ResetServerStats() {
-	atomic.StoreInt64(&ais.totalFrames, 0)
-	atomic.StoreInt64(&ais.droppedFrames, 0)
-}
-
-// RecoverFromDroppedFrames attempts to recover when too many frames are dropped
-func (ais *AudioInputServer) RecoverFromDroppedFrames() {
-	total := atomic.LoadInt64(&ais.totalFrames)
-	dropped := atomic.LoadInt64(&ais.droppedFrames)
-
-	// If more than 50% of frames are dropped, attempt recovery
-	if total > 100 && dropped > total/2 {
-		logger := logging.GetDefaultLogger().With().Str("component", AudioInputServerComponent).Logger()
-		logger.Warn().Int64("total", total).Int64("dropped", dropped).Msg("high drop rate detected, attempting recovery")
-
-		// Reset stats and update buffer size from adaptive manager
-		ais.ResetServerStats()
-		ais.UpdateBufferSize()
-	}
-}
-
 // startReaderGoroutine starts the message reader using the goroutine pool
 func (ais *AudioInputServer) startReaderGoroutine() {
 	ais.wg.Add(1)
@@ -894,10 +877,10 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 		// Enhanced error tracking and recovery
 		var consecutiveErrors int
 		var lastErrorTime time.Time
-		maxConsecutiveErrors := Config.MaxConsecutiveErrors
+		maxConsecutiveErrors := GetConfig().MaxConsecutiveErrors
-		errorResetWindow := Config.RestartWindow // Use existing restart window
+		errorResetWindow := GetConfig().RestartWindow // Use existing restart window
-		baseBackoffDelay := Config.RetryDelay
+		baseBackoffDelay := GetConfig().RetryDelay
-		maxBackoffDelay := Config.MaxRetryDelay
+		maxBackoffDelay := GetConfig().MaxRetryDelay
 
 		logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
 
@@ -967,13 +950,9 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 				}
 			}
 
-			// Send to message channel with non-blocking write (use read lock for channel access)
+			// Send to message channel with non-blocking write
-			ais.channelMutex.RLock()
-			messageChan := ais.messageChan
-			ais.channelMutex.RUnlock()
-
 			select {
-			case messageChan <- msg:
+			case ais.messageChan <- msg:
 				atomic.AddInt64(&ais.totalFrames, 1)
 			default:
 				// Channel full, drop message
@@ -987,16 +966,16 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 		}
 	}
 
-	// Submit the reader task to the audio reader pool with backpressure
+	// Submit the reader task to the audio reader pool
 	logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
-	if !SubmitAudioReaderTaskWithBackpressure(readerTask) {
+	if !SubmitAudioReaderTask(readerTask) {
-		// Task was dropped due to backpressure - this is expected under high load
+		// If the pool is full or shutting down, fall back to direct goroutine creation
-		// Log at debug level to avoid spam, but track the drop
+		// Only log if warn level enabled - avoid sampling logic in critical path
-		logger.Debug().Msg("Audio reader task dropped due to backpressure")
+		if logger.GetLevel() <= zerolog.WarnLevel {
+			logger.Warn().Msg("Audio reader pool full or shutting down, falling back to direct goroutine creation")
+		}
 
-		// Don't fall back to unlimited goroutine creation
+		go readerTask()
-		// Instead, let the system recover naturally
-		ais.wg.Done() // Decrement the wait group since we're not starting the task
 	}
 }
 
@@ -1008,7 +987,7 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 	processorTask := func() {
 		// Only lock OS thread and set priority for high-load scenarios
 		// This reduces interference with input processing threads
-		config := Config
+		config := GetConfig()
 		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
 		if useThreadOptimizations {
@@ -1032,7 +1011,7 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 			select {
 			case <-ais.stopChan:
 				return
-			case msg := <-ais.getMessageChan():
+			case msg := <-ais.messageChan:
 				// Process message with error handling
 				start := time.Now()
 				err := ais.processMessageWithRecovery(msg, logger)
@@ -1053,10 +1032,9 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 					// If too many processing errors, drop frames more aggressively
 					if processingErrors >= maxProcessingErrors {
 						// Clear processing queue to recover
-						processChan := ais.getProcessChan()
+						for len(ais.processChan) > 0 {
-						for len(processChan) > 0 {
 							select {
-							case <-processChan:
+							case <-ais.processChan:
 								atomic.AddInt64(&ais.droppedFrames, 1)
 							default:
 								break
@@ -1079,16 +1057,13 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 		}
 	}
 
-	// Submit the processor task to the audio processor pool with backpressure
+	// Submit the processor task to the audio processor pool
 	logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
-	if !SubmitAudioProcessorTaskWithBackpressure(processorTask) {
+	if !SubmitAudioProcessorTask(processorTask) {
-		// Task was dropped due to backpressure - this is expected under high load
+		// If the pool is full or shutting down, fall back to direct goroutine creation
-		// Log at debug level to avoid spam, but track the drop
+		logger.Warn().Msg("Audio processor pool full or shutting down, falling back to direct goroutine creation")
-		logger.Debug().Msg("Audio processor task dropped due to backpressure")
 
-		// Don't fall back to unlimited goroutine creation
+		go processorTask()
-		// Instead, let the system recover naturally
-		ais.wg.Done() // Decrement the wait group since we're not starting the task
 	}
 }
 
@@ -1097,14 +1072,13 @@ func (ais *AudioInputServer) processMessageWithRecovery(msg *InputIPCMessage, lo
 	// Intelligent frame dropping: prioritize recent frames
 	if msg.Type == InputMessageTypeOpusFrame {
 		// Check if processing queue is getting full
-		processChan := ais.getProcessChan()
+		queueLen := len(ais.processChan)
-		queueLen := len(processChan)
 		bufferSize := int(atomic.LoadInt64(&ais.bufferSize))
 
 		if queueLen > bufferSize*3/4 {
 			// Drop oldest frames, keep newest
 			select {
-			case <-processChan: // Remove oldest
+			case <-ais.processChan: // Remove oldest
 				atomic.AddInt64(&ais.droppedFrames, 1)
 				logger.Debug().Msg("Dropped oldest frame to make room")
 			default:
@@ -1112,15 +1086,11 @@ func (ais *AudioInputServer) processMessageWithRecovery(msg *InputIPCMessage, lo
 		}
 	}
 
-	// Send to processing queue with timeout (use read lock for channel access)
+	// Send to processing queue with timeout
-	ais.channelMutex.RLock()
-	processChan := ais.processChan
-	ais.channelMutex.RUnlock()
-
 	select {
-	case processChan <- msg:
+	case ais.processChan <- msg:
 		return nil
-	case <-time.After(Config.WriteTimeout):
+	case <-time.After(GetConfig().WriteTimeout):
 		// Processing queue full and timeout reached, drop frame
 		atomic.AddInt64(&ais.droppedFrames, 1)
 		return fmt.Errorf("processing queue timeout")
@@ -1139,7 +1109,7 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 	monitorTask := func() {
 		// Monitor goroutine doesn't need thread locking for most scenarios
 		// Only use thread optimizations for high-throughput scenarios
-		config := Config
+		config := GetConfig()
 		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
 		if useThreadOptimizations {
@@ -1150,11 +1120,11 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 		}
 
 		defer ais.wg.Done()
-		ticker := time.NewTicker(Config.DefaultTickerInterval)
+		ticker := time.NewTicker(GetConfig().DefaultTickerInterval)
 		defer ticker.Stop()
 
 		// Buffer size update ticker (less frequent)
-		bufferUpdateTicker := time.NewTicker(Config.BufferUpdateInterval)
+		bufferUpdateTicker := time.NewTicker(GetConfig().BufferUpdateInterval)
 		defer bufferUpdateTicker.Stop()
 
 		for {
@@ -1165,7 +1135,7 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 				// Process frames from processing queue
 				for {
 					select {
-					case msg := <-ais.getProcessChan():
+					case msg := <-ais.processChan:
 						start := time.Now()
 						err := ais.processMessage(msg)
 						processingTime := time.Since(start)
@@ -1204,7 +1174,8 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 				// Check if we need to update buffer size
 				select {
 				case <-bufferUpdateTicker.C:
-					// Buffer size is now fixed from config
+					// Update buffer size from adaptive buffer manager
+					ais.UpdateBufferSize()
 				default:
 					// No buffer update needed
 				}
@@ -1212,16 +1183,13 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 		}
 	}
 
-	// Submit the monitor task to the audio processor pool with backpressure
+	// Submit the monitor task to the audio processor pool
 	logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
-	if !SubmitAudioProcessorTaskWithBackpressure(monitorTask) {
+	if !SubmitAudioProcessorTask(monitorTask) {
-		// Task was dropped due to backpressure - this is expected under high load
+		// If the pool is full or shutting down, fall back to direct goroutine creation
-		// Log at debug level to avoid spam, but track the drop
+		logger.Warn().Msg("Audio processor pool full or shutting down, falling back to direct goroutine creation")
-		logger.Debug().Msg("Audio monitor task dropped due to backpressure")
 
-		// Don't fall back to unlimited goroutine creation
+		go monitorTask()
-		// Instead, let the system recover naturally
-		ais.wg.Done() // Decrement the wait group since we're not starting the task
 	}
 }
 
@@ -1233,16 +1201,17 @@ func (ais *AudioInputServer) GetServerStats() (total, dropped int64, avgProcessi
 		atomic.LoadInt64(&ais.bufferSize)
 }
 
-// UpdateBufferSize updates the buffer size (now using fixed config values)
+// UpdateBufferSize updates the buffer size from adaptive buffer manager
 func (ais *AudioInputServer) UpdateBufferSize() {
-	// Buffer size is now fixed from config
+	adaptiveManager := GetAdaptiveBufferManager()
-	newSize := int64(Config.AdaptiveDefaultBufferSize)
+	newSize := int64(adaptiveManager.GetInputBufferSize())
 	atomic.StoreInt64(&ais.bufferSize, newSize)
 }
 
-// ReportLatency reports processing latency (now a no-op with fixed buffers)
+// ReportLatency reports processing latency to adaptive buffer manager
 func (ais *AudioInputServer) ReportLatency(latency time.Duration) {
-	// Latency reporting is now a no-op with fixed buffer sizes
+	adaptiveManager := GetAdaptiveBufferManager()
+	adaptiveManager.UpdateLatency(latency)
 }
 
 // GetMessagePoolStats returns detailed statistics about the message pool
@@ -1257,7 +1226,7 @@ func (mp *MessagePool) GetMessagePoolStats() MessagePoolStats {
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * Config.PercentageMultiplier
+		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
 	}
 
 	// Calculate channel pool size
@@ -1290,20 +1259,6 @@ func GetGlobalMessagePoolStats() MessagePoolStats {
 	return globalMessagePool.GetMessagePoolStats()
 }
 
-// getMessageChan safely returns the current message channel
-func (ais *AudioInputServer) getMessageChan() chan *InputIPCMessage {
-	ais.channelMutex.RLock()
-	defer ais.channelMutex.RUnlock()
-	return ais.messageChan
-}
-
-// getProcessChan safely returns the current process channel
-func (ais *AudioInputServer) getProcessChan() chan *InputIPCMessage {
-	ais.channelMutex.RLock()
-	defer ais.channelMutex.RUnlock()
-	return ais.processChan
-}
-
 // Helper functions
 
 // getInputSocketPath is now defined in unified_ipc.go

internal/audio/ipc_output.go

@@ -4,393 +4,67 @@ import (
 	"encoding/binary"
 	"fmt"
 	"io"
-	"net"
-	"sync"
 	"sync/atomic"
-	"time"
-
-	"github.com/jetkvm/kvm/internal/logging"
-	"github.com/rs/zerolog"
 )
 
 // Legacy aliases for backward compatibility
 type OutputIPCConfig = UnifiedIPCConfig
-type OutputIPCOpusConfig = UnifiedIPCOpusConfig
 type OutputMessageType = UnifiedMessageType
 type OutputIPCMessage = UnifiedIPCMessage
 
 // Legacy constants for backward compatibility
 const (
-	OutputMessageTypeOpusFrame  = MessageTypeOpusFrame
+	OutputMessageTypeOpusFrame = MessageTypeOpusFrame
-	OutputMessageTypeConfig     = MessageTypeConfig
+	OutputMessageTypeConfig    = MessageTypeConfig
-	OutputMessageTypeOpusConfig = MessageTypeOpusConfig
+	OutputMessageTypeStop      = MessageTypeStop
-	OutputMessageTypeStop       = MessageTypeStop
+	OutputMessageTypeHeartbeat = MessageTypeHeartbeat
-	OutputMessageTypeHeartbeat  = MessageTypeHeartbeat
+	OutputMessageTypeAck       = MessageTypeAck
-	OutputMessageTypeAck        = MessageTypeAck
 )
 
 // Methods are now inherited from UnifiedIPCMessage
 
 // Global shared message pool for output IPC client header reading
-var globalOutputClientMessagePool = NewGenericMessagePool(Config.OutputMessagePoolSize)
+var globalOutputClientMessagePool = NewGenericMessagePool(GetConfig().OutputMessagePoolSize)
 
-// AudioOutputServer provides audio output IPC functionality
+// AudioOutputServer is now an alias for UnifiedAudioServer
-type AudioOutputServer struct {
+type AudioOutputServer = UnifiedAudioServer
-	// Atomic counters
-	bufferSize    int64 // Current buffer size (atomic)
-	droppedFrames int64 // Dropped frames counter (atomic)
-	totalFrames   int64 // Total frames counter (atomic)
-
-	listener net.Listener
-	conn     net.Conn
-	mtx      sync.Mutex
-	running  bool
-	logger   zerolog.Logger
-
-	// Message channels
-	messageChan chan *OutputIPCMessage // Buffered channel for incoming messages
-	processChan chan *OutputIPCMessage // Buffered channel for processing queue
-	wg          sync.WaitGroup         // Wait group for goroutine coordination
-
-	// Configuration
-	socketPath  string
-	magicNumber uint32
-}
 
 func NewAudioOutputServer() (*AudioOutputServer, error) {
-	socketPath := getOutputSocketPath()
+	return NewUnifiedAudioServer(false) // false = output server
-	logger := logging.GetDefaultLogger().With().Str("component", "audio-output-server").Logger()
-
-	server := &AudioOutputServer{
-		socketPath:  socketPath,
-		magicNumber: Config.OutputMagicNumber,
-		logger:      logger,
-		messageChan: make(chan *OutputIPCMessage, Config.ChannelBufferSize),
-		processChan: make(chan *OutputIPCMessage, Config.ChannelBufferSize),
-	}
-
-	return server, nil
 }
 
+// Start method is now inherited from UnifiedAudioServer
+
+// acceptConnections method is now inherited from UnifiedAudioServer
+
+// startProcessorGoroutine method is now inherited from UnifiedAudioServer
+
+// Stop method is now inherited from UnifiedAudioServer
+
+// Close method is now inherited from UnifiedAudioServer
+
+// SendFrame method is now inherited from UnifiedAudioServer
+
 // GetServerStats returns server performance statistics
-// Start starts the audio output server
-func (s *AudioOutputServer) Start() error {
-	s.mtx.Lock()
-	defer s.mtx.Unlock()
-
-	if s.running {
-		return fmt.Errorf("audio output server is already running")
-	}
-
-	// Create Unix socket
-	listener, err := net.Listen("unix", s.socketPath)
-	if err != nil {
-		return fmt.Errorf("failed to create unix socket: %w", err)
-	}
-
-	s.listener = listener
-	s.running = true
-
-	// Start goroutines
-	s.wg.Add(1)
-	go s.acceptConnections()
-
-	s.logger.Info().Str("socket_path", s.socketPath).Msg("Audio output server started")
-	return nil
-}
-
-// Stop stops the audio output server
-func (s *AudioOutputServer) Stop() {
-	s.mtx.Lock()
-	defer s.mtx.Unlock()
-
-	if !s.running {
-		return
-	}
-
-	s.running = false
-
-	if s.listener != nil {
-		s.listener.Close()
-	}
-
-	if s.conn != nil {
-		s.conn.Close()
-	}
-
-	// Close channels
-	close(s.messageChan)
-	close(s.processChan)
-
-	s.wg.Wait()
-	s.logger.Info().Msg("Audio output server stopped")
-}
-
-// acceptConnections handles incoming connections
-func (s *AudioOutputServer) acceptConnections() {
-	defer s.wg.Done()
-
-	for s.running {
-		conn, err := s.listener.Accept()
-		if err != nil {
-			if s.running {
-				s.logger.Error().Err(err).Msg("Failed to accept connection")
-			}
-			return
-		}
-
-		s.mtx.Lock()
-		s.conn = conn
-		s.mtx.Unlock()
-
-		s.logger.Info().Msg("Client connected to audio output server")
-		// Start message processing for this connection
-		s.wg.Add(1)
-		go s.handleConnection(conn)
-	}
-}
-
-// handleConnection processes messages from a client connection
-func (s *AudioOutputServer) handleConnection(conn net.Conn) {
-	defer s.wg.Done()
-	defer conn.Close()
-
-	for s.running {
-		msg, err := s.readMessage(conn)
-		if err != nil {
-			if s.running {
-				s.logger.Error().Err(err).Msg("Failed to read message from client")
-			}
-			return
-		}
-
-		if err := s.processMessage(msg); err != nil {
-			s.logger.Error().Err(err).Msg("Failed to process message")
-		}
-	}
-}
-
-// readMessage reads a message from the connection
-func (s *AudioOutputServer) readMessage(conn net.Conn) (*OutputIPCMessage, error) {
-	header := make([]byte, 17)
-	if _, err := io.ReadFull(conn, header); err != nil {
-		return nil, fmt.Errorf("failed to read header: %w", err)
-	}
-
-	magic := binary.LittleEndian.Uint32(header[0:4])
-	if magic != s.magicNumber {
-		return nil, fmt.Errorf("invalid magic number: expected %d, got %d", s.magicNumber, magic)
-	}
-
-	msgType := OutputMessageType(header[4])
-	length := binary.LittleEndian.Uint32(header[5:9])
-	timestamp := int64(binary.LittleEndian.Uint64(header[9:17]))
-
-	var data []byte
-	if length > 0 {
-		data = make([]byte, length)
-		if _, err := io.ReadFull(conn, data); err != nil {
-			return nil, fmt.Errorf("failed to read data: %w", err)
-		}
-	}
-
-	return &OutputIPCMessage{
-		Magic:     magic,
-		Type:      msgType,
-		Length:    length,
-		Timestamp: timestamp,
-		Data:      data,
-	}, nil
-}
-
-// processMessage processes a received message
-func (s *AudioOutputServer) processMessage(msg *OutputIPCMessage) error {
-	switch msg.Type {
-	case OutputMessageTypeOpusConfig:
-		return s.processOpusConfig(msg.Data)
-	case OutputMessageTypeStop:
-		s.logger.Info().Msg("Received stop message")
-		return nil
-	case OutputMessageTypeHeartbeat:
-		s.logger.Debug().Msg("Received heartbeat")
-		return nil
-	default:
-		s.logger.Warn().Int("type", int(msg.Type)).Msg("Unknown message type")
-		return nil
-	}
-}
-
-// processOpusConfig processes Opus configuration updates
-func (s *AudioOutputServer) processOpusConfig(data []byte) error {
-	// Validate configuration data size (9 * int32 = 36 bytes)
-	if len(data) != 36 {
-		return fmt.Errorf("invalid Opus configuration data size: expected 36 bytes, got %d", len(data))
-	}
-
-	// Decode Opus configuration
-	config := OutputIPCOpusConfig{
-		SampleRate: int(binary.LittleEndian.Uint32(data[0:4])),
-		Channels:   int(binary.LittleEndian.Uint32(data[4:8])),
-		FrameSize:  int(binary.LittleEndian.Uint32(data[8:12])),
-		Bitrate:    int(binary.LittleEndian.Uint32(data[12:16])),
-		Complexity: int(binary.LittleEndian.Uint32(data[16:20])),
-		VBR:        int(binary.LittleEndian.Uint32(data[20:24])),
-		SignalType: int(binary.LittleEndian.Uint32(data[24:28])),
-		Bandwidth:  int(binary.LittleEndian.Uint32(data[28:32])),
-		DTX:        int(binary.LittleEndian.Uint32(data[32:36])),
-	}
-
-	s.logger.Info().Interface("config", config).Msg("Received Opus configuration update")
-
-	// Ensure we're running in the audio server subprocess
-	if !isAudioServerProcess() {
-		s.logger.Warn().Msg("Opus configuration update ignored - not running in audio server subprocess")
-		return nil
-	}
-
-	// Check if audio output streaming is currently active
-	if atomic.LoadInt32(&outputStreamingRunning) == 0 {
-		s.logger.Info().Msg("Audio output streaming not active, configuration will be applied when streaming starts")
-		return nil
-	}
-
-	// Ensure capture is initialized before updating encoder parameters
-	// The C function requires both encoder and capture_initialized to be true
-	if err := cgoAudioInit(); err != nil {
-		s.logger.Debug().Err(err).Msg("Audio capture already initialized or initialization failed")
-		// Continue anyway - capture may already be initialized
-	}
-
-	// Apply configuration using CGO function (only if audio system is running)
-	vbrConstraint := Config.CGOOpusVBRConstraint
-	if err := updateOpusEncoderParams(config.Bitrate, config.Complexity, config.VBR, vbrConstraint, config.SignalType, config.Bandwidth, config.DTX); err != nil {
-		s.logger.Error().Err(err).Msg("Failed to update Opus encoder parameters - encoder may not be initialized")
-		return err
-	}
-
-	s.logger.Info().Msg("Opus encoder parameters updated successfully")
-	return nil
-}
-
-// SendFrame sends an audio frame to the client
-func (s *AudioOutputServer) SendFrame(frame []byte) error {
-	s.mtx.Lock()
-	conn := s.conn
-	s.mtx.Unlock()
-
-	if conn == nil {
-		return fmt.Errorf("no client connected")
-	}
-
-	msg := &OutputIPCMessage{
-		Magic:     s.magicNumber,
-		Type:      OutputMessageTypeOpusFrame,
-		Length:    uint32(len(frame)),
-		Timestamp: time.Now().UnixNano(),
-		Data:      frame,
-	}
-
-	return s.writeMessage(conn, msg)
-}
-
-// writeMessage writes a message to the connection
-func (s *AudioOutputServer) writeMessage(conn net.Conn, msg *OutputIPCMessage) error {
-	header := EncodeMessageHeader(msg.Magic, uint8(msg.Type), msg.Length, msg.Timestamp)
-
-	if _, err := conn.Write(header); err != nil {
-		return fmt.Errorf("failed to write header: %w", err)
-	}
-
-	if msg.Length > 0 && msg.Data != nil {
-		if _, err := conn.Write(msg.Data); err != nil {
-			return fmt.Errorf("failed to write data: %w", err)
-		}
-	}
-
-	atomic.AddInt64(&s.totalFrames, 1)
-	return nil
-}
-
 func (s *AudioOutputServer) GetServerStats() (total, dropped int64, bufferSize int64) {
-	return atomic.LoadInt64(&s.totalFrames), atomic.LoadInt64(&s.droppedFrames), atomic.LoadInt64(&s.bufferSize)
+	stats := GetFrameStats(&s.totalFrames, &s.droppedFrames)
+	return stats.Total, stats.Dropped, atomic.LoadInt64(&s.bufferSize)
 }
 
-// AudioOutputClient provides audio output IPC client functionality
+// AudioOutputClient is now an alias for UnifiedAudioClient
-type AudioOutputClient struct {
+type AudioOutputClient = UnifiedAudioClient
-	// Atomic counters
-	droppedFrames int64 // Atomic counter for dropped frames
-	totalFrames   int64 // Atomic counter for total frames
-
-	conn        net.Conn
-	mtx         sync.Mutex
-	running     bool
-	logger      zerolog.Logger
-	socketPath  string
-	magicNumber uint32
-	bufferPool  *AudioBufferPool // Buffer pool for memory optimization
-
-	// Health monitoring
-	autoReconnect bool // Enable automatic reconnection
-}
 
 func NewAudioOutputClient() *AudioOutputClient {
-	socketPath := getOutputSocketPath()
+	return NewUnifiedAudioClient(false) // false = output client
-	logger := logging.GetDefaultLogger().With().Str("component", "audio-output-client").Logger()
-
-	return &AudioOutputClient{
-		socketPath:    socketPath,
-		magicNumber:   Config.OutputMagicNumber,
-		logger:        logger,
-		bufferPool:    NewAudioBufferPool(Config.MaxFrameSize),
-		autoReconnect: true,
-	}
 }
 
-// Connect connects to the audio output server
+// Connect method is now inherited from UnifiedAudioClient
-func (c *AudioOutputClient) Connect() error {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
 
-	if c.running {
+// Disconnect method is now inherited from UnifiedAudioClient
-		return fmt.Errorf("audio output client is already connected")
-	}
 
-	conn, err := net.Dial("unix", c.socketPath)
+// IsConnected method is now inherited from UnifiedAudioClient
-	if err != nil {
-		return fmt.Errorf("failed to connect to audio output server: %w", err)
-	}
 
-	c.conn = conn
+// Close method is now inherited from UnifiedAudioClient
-	c.running = true
-	c.logger.Info().Str("socket_path", c.socketPath).Msg("Connected to audio output server")
-	return nil
-}
-
-// Disconnect disconnects from the audio output server
-func (c *AudioOutputClient) Disconnect() {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
-
-	if !c.running {
-		return
-	}
-
-	c.running = false
-
-	if c.conn != nil {
-		c.conn.Close()
-		c.conn = nil
-	}
-
-	c.logger.Info().Msg("Disconnected from audio output server")
-}
-
-// IsConnected returns whether the client is connected
-func (c *AudioOutputClient) IsConnected() bool {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
-	return c.running && c.conn != nil
-}
 
 func (c *AudioOutputClient) ReceiveFrame() ([]byte, error) {
 	c.mtx.Lock()
@@ -421,7 +95,7 @@ func (c *AudioOutputClient) ReceiveFrame() ([]byte, error) {
 	}
 
 	size := binary.LittleEndian.Uint32(optMsg.header[5:9])
-	maxFrameSize := Config.OutputMaxFrameSize
+	maxFrameSize := GetConfig().OutputMaxFrameSize
 	if int(size) > maxFrameSize {
 		return nil, fmt.Errorf("received frame size validation failed: got %d bytes, maximum allowed %d bytes", size, maxFrameSize)
 	}
@@ -442,53 +116,6 @@ func (c *AudioOutputClient) ReceiveFrame() ([]byte, error) {
 	return frame, nil
 }
 
-// SendOpusConfig sends Opus configuration to the audio output server
-func (c *AudioOutputClient) SendOpusConfig(config OutputIPCOpusConfig) error {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
-
-	if !c.running || c.conn == nil {
-		return fmt.Errorf("not connected to audio output server")
-	}
-
-	// Validate configuration parameters
-	if config.SampleRate <= 0 || config.Channels <= 0 || config.FrameSize <= 0 || config.Bitrate <= 0 {
-		return fmt.Errorf("invalid Opus configuration: SampleRate=%d, Channels=%d, FrameSize=%d, Bitrate=%d",
-			config.SampleRate, config.Channels, config.FrameSize, config.Bitrate)
-	}
-
-	// Serialize Opus configuration using common function
-	data := EncodeOpusConfig(config.SampleRate, config.Channels, config.FrameSize, config.Bitrate, config.Complexity, config.VBR, config.SignalType, config.Bandwidth, config.DTX)
-
-	msg := &OutputIPCMessage{
-		Magic:     c.magicNumber,
-		Type:      OutputMessageTypeOpusConfig,
-		Length:    uint32(len(data)),
-		Timestamp: time.Now().UnixNano(),
-		Data:      data,
-	}
-
-	return c.writeMessage(msg)
-}
-
-// writeMessage writes a message to the connection
-func (c *AudioOutputClient) writeMessage(msg *OutputIPCMessage) error {
-	header := EncodeMessageHeader(msg.Magic, uint8(msg.Type), msg.Length, msg.Timestamp)
-
-	if _, err := c.conn.Write(header); err != nil {
-		return fmt.Errorf("failed to write header: %w", err)
-	}
-
-	if msg.Length > 0 && msg.Data != nil {
-		if _, err := c.conn.Write(msg.Data); err != nil {
-			return fmt.Errorf("failed to write data: %w", err)
-		}
-	}
-
-	atomic.AddInt64(&c.totalFrames, 1)
-	return nil
-}
-
 // GetClientStats returns client performance statistics
 func (c *AudioOutputClient) GetClientStats() (total, dropped int64) {
 	stats := GetFrameStats(&c.totalFrames, &c.droppedFrames)
@@ -496,4 +123,5 @@ func (c *AudioOutputClient) GetClientStats() (total, dropped int64) {
 }
 
 // Helper functions
-// getOutputSocketPath is defined in ipc_unified.go
+
+// getOutputSocketPath is now defined in unified_ipc.go

internal/audio/ipc_unified.go

@@ -4,11 +4,9 @@ import (
 	"encoding/binary"
 	"fmt"
 	"io"
-	"math"
 	"net"
 	"os"
 	"path/filepath"
-	"strings"
 	"sync"
 	"sync/atomic"
 	"time"
@@ -19,21 +17,13 @@ import (
 
 // Unified IPC constants
 var (
-	outputMagicNumber uint32 = Config.OutputMagicNumber // "JKOU" (JetKVM Output)
+	outputMagicNumber uint32 = GetConfig().OutputMagicNumber // "JKOU" (JetKVM Output)
-	inputMagicNumber  uint32 = Config.InputMagicNumber  // "JKMI" (JetKVM Microphone Input)
+	inputMagicNumber  uint32 = GetConfig().InputMagicNumber  // "JKMI" (JetKVM Microphone Input)
 	outputSocketName         = "audio_output.sock"
 	inputSocketName          = "audio_input.sock"
 	headerSize               = 17 // Fixed header size: 4+1+4+8 bytes
 )
 
-// Header buffer pool to reduce allocation overhead
-var headerBufferPool = sync.Pool{
-	New: func() interface{} {
-		buf := make([]byte, headerSize)
-		return &buf
-	},
-}
-
 // UnifiedMessageType represents the type of IPC message for both input and output
 type UnifiedMessageType uint8
 
@@ -99,6 +89,7 @@ type UnifiedIPCOpusConfig struct {
 // UnifiedAudioServer provides common functionality for both input and output servers
 type UnifiedAudioServer struct {
 	// Atomic counters for performance monitoring
+	bufferSize    int64 // Current buffer size (atomic)
 	droppedFrames int64 // Dropped frames counter (atomic)
 	totalFrames   int64 // Total frames counter (atomic)
 
@@ -117,6 +108,10 @@ type UnifiedAudioServer struct {
 	socketPath         string
 	magicNumber        uint32
 	socketBufferConfig SocketBufferConfig
+
+	// Performance monitoring
+	latencyMonitor    *LatencyMonitor
+	adaptiveOptimizer *AdaptiveOptimizer
 }
 
 // NewUnifiedAudioServer creates a new unified audio server
@@ -141,9 +136,11 @@ func NewUnifiedAudioServer(isInput bool) (*UnifiedAudioServer, error) {
 		logger:             logger,
 		socketPath:         socketPath,
 		magicNumber:        magicNumber,
-		messageChan:        make(chan *UnifiedIPCMessage, Config.ChannelBufferSize),
+		messageChan:        make(chan *UnifiedIPCMessage, GetConfig().ChannelBufferSize),
-		processChan:        make(chan *UnifiedIPCMessage, Config.ChannelBufferSize),
+		processChan:        make(chan *UnifiedIPCMessage, GetConfig().ChannelBufferSize),
 		socketBufferConfig: DefaultSocketBufferConfig(),
+		latencyMonitor:     nil,
+		adaptiveOptimizer:  nil,
 	}
 
 	return server, nil
@@ -158,38 +155,15 @@ func (s *UnifiedAudioServer) Start() error {
 		return fmt.Errorf("server already running")
 	}
 
-	// Remove existing socket file with retry logic
+	// Remove existing socket file
-	for i := 0; i < 3; i++ {
+	if err := os.Remove(s.socketPath); err != nil && !os.IsNotExist(err) {
-		if err := os.Remove(s.socketPath); err != nil && !os.IsNotExist(err) {
+		return fmt.Errorf("failed to remove existing socket: %w", err)
-			s.logger.Warn().Err(err).Int("attempt", i+1).Msg("failed to remove existing socket file, retrying")
-			time.Sleep(100 * time.Millisecond)
-			continue
-		}
-		break
-	}
-
-	// Create listener with retry on address already in use
-	var listener net.Listener
-	var err error
-	for i := 0; i < 3; i++ {
-		listener, err = net.Listen("unix", s.socketPath)
-		if err == nil {
-			break
-		}
-
-		// If address is still in use, try to remove socket file again
-		if strings.Contains(err.Error(), "address already in use") {
-			s.logger.Warn().Err(err).Int("attempt", i+1).Msg("socket address in use, attempting cleanup and retry")
-			os.Remove(s.socketPath)
-			time.Sleep(200 * time.Millisecond)
-			continue
-		}
-
-		return fmt.Errorf("failed to create unix socket: %w", err)
 	}
 
+	// Create listener
+	listener, err := net.Listen("unix", s.socketPath)
 	if err != nil {
-		return fmt.Errorf("failed to create unix socket after retries: %w", err)
+		return fmt.Errorf("failed to create listener: %w", err)
 	}
 
 	s.listener = listener
@@ -309,11 +283,8 @@ func (s *UnifiedAudioServer) startProcessorGoroutine() {
 
 // readMessage reads a message from the connection
 func (s *UnifiedAudioServer) readMessage(conn net.Conn) (*UnifiedIPCMessage, error) {
-	// Get header buffer from pool
+	// Read header
-	headerPtr := headerBufferPool.Get().(*[]byte)
+	header := make([]byte, headerSize)
-	header := *headerPtr
-	defer headerBufferPool.Put(headerPtr)
-
 	if _, err := io.ReadFull(conn, header); err != nil {
 		return nil, fmt.Errorf("failed to read header: %w", err)
 	}
@@ -329,7 +300,7 @@ func (s *UnifiedAudioServer) readMessage(conn net.Conn) (*UnifiedIPCMessage, err
 	timestamp := int64(binary.LittleEndian.Uint64(header[9:17]))
 
 	// Validate length
-	if length > uint32(Config.MaxFrameSize) {
+	if length > uint32(GetConfig().MaxFrameSize) {
 		return nil, fmt.Errorf("message too large: %d bytes", length)
 	}
 
@@ -357,10 +328,7 @@ func (s *UnifiedAudioServer) SendFrame(frame []byte) error {
 	defer s.mtx.Unlock()
 
 	if !s.running || s.conn == nil {
-		// Silently drop frames when no client is connected
+		return fmt.Errorf("no client connected")
-		// This prevents "no client connected" warnings during startup and quality changes
-		atomic.AddInt64(&s.droppedFrames, 1)
-		return nil // Return nil to avoid flooding logs with connection warnings
 	}
 
 	start := time.Now()
@@ -382,6 +350,10 @@ func (s *UnifiedAudioServer) SendFrame(frame []byte) error {
 	}
 
 	// Record latency for monitoring
+	if s.latencyMonitor != nil {
+		writeLatency := time.Since(start)
+		s.latencyMonitor.RecordLatency(writeLatency, "ipc_write")
+	}
 
 	atomic.AddInt64(&s.totalFrames, 1)
 	return nil
@@ -389,20 +361,21 @@ func (s *UnifiedAudioServer) SendFrame(frame []byte) error {
 
 // writeMessage writes a message to the connection
 func (s *UnifiedAudioServer) writeMessage(conn net.Conn, msg *UnifiedIPCMessage) error {
-	header := EncodeMessageHeader(msg.Magic, uint8(msg.Type), msg.Length, msg.Timestamp)
+	// Write header
+	header := make([]byte, headerSize)
+	binary.LittleEndian.PutUint32(header[0:4], msg.Magic)
+	header[4] = uint8(msg.Type)
+	binary.LittleEndian.PutUint32(header[5:9], msg.Length)
+	binary.LittleEndian.PutUint64(header[9:17], uint64(msg.Timestamp))
 
-	// Optimize: Use single write for header+data to reduce system calls
+	if _, err := conn.Write(header); err != nil {
+		return fmt.Errorf("failed to write header: %w", err)
+	}
+
+	// Write data if present
 	if msg.Length > 0 && msg.Data != nil {
-		// Pre-allocate combined buffer to avoid copying
+		if _, err := conn.Write(msg.Data); err != nil {
-		combined := make([]byte, len(header)+len(msg.Data))
+			return fmt.Errorf("failed to write data: %w", err)
-		copy(combined, header)
-		copy(combined[len(header):], msg.Data)
-		if _, err := conn.Write(combined); err != nil {
-			return fmt.Errorf("failed to write message: %w", err)
-		}
-	} else {
-		if _, err := conn.Write(header); err != nil {
-			return fmt.Errorf("failed to write header: %w", err)
 		}
 	}
 
@@ -411,7 +384,7 @@ func (s *UnifiedAudioServer) writeMessage(conn net.Conn, msg *UnifiedIPCMessage)
 
 // UnifiedAudioClient provides common functionality for both input and output clients
 type UnifiedAudioClient struct {
-	// Atomic counters for frame statistics
+	// Atomic fields first for ARM32 alignment
 	droppedFrames int64 // Atomic counter for dropped frames
 	totalFrames   int64 // Atomic counter for total frames
 
@@ -422,13 +395,6 @@ type UnifiedAudioClient struct {
 	socketPath  string
 	magicNumber uint32
 	bufferPool  *AudioBufferPool // Buffer pool for memory optimization
-
-	// Connection health monitoring
-	lastHealthCheck   time.Time
-	connectionErrors  int64 // Atomic counter for connection errors
-	autoReconnect     bool  // Enable automatic reconnection
-	healthCheckTicker *time.Ticker
-	stopHealthCheck   chan struct{}
 }
 
 // NewUnifiedAudioClient creates a new unified audio client
@@ -450,12 +416,10 @@ func NewUnifiedAudioClient(isInput bool) *UnifiedAudioClient {
 	logger := logging.GetDefaultLogger().With().Str("component", componentName).Logger()
 
 	return &UnifiedAudioClient{
-		logger:          logger,
+		logger:      logger,
-		socketPath:      socketPath,
+		socketPath:  socketPath,
-		magicNumber:     magicNumber,
+		magicNumber: magicNumber,
-		bufferPool:      NewAudioBufferPool(Config.MaxFrameSize),
+		bufferPool:  NewAudioBufferPool(GetConfig().MaxFrameSize),
-		autoReconnect:   true, // Enable automatic reconnection by default
-		stopHealthCheck: make(chan struct{}),
 	}
 }
 
@@ -475,46 +439,32 @@ func (c *UnifiedAudioClient) Connect() error {
 	}
 
 	// Try connecting multiple times as the server might not be ready
-	// Use configurable retry parameters for better control
+	// Reduced retry count and delay for faster startup
-	maxAttempts := Config.MaxConnectionAttempts
+	for i := 0; i < 10; i++ {
-	initialDelay := Config.ConnectionRetryDelay
+		conn, err := net.Dial("unix", c.socketPath)
-	maxDelay := Config.MaxConnectionRetryDelay
-	backoffFactor := Config.ConnectionBackoffFactor
-
-	for i := 0; i < maxAttempts; i++ {
-		// Set connection timeout for each attempt
-		conn, err := net.DialTimeout("unix", c.socketPath, Config.ConnectionTimeoutDelay)
 		if err == nil {
 			c.conn = conn
 			c.running = true
 			// Reset frame counters on successful connection
 			atomic.StoreInt64(&c.totalFrames, 0)
 			atomic.StoreInt64(&c.droppedFrames, 0)
-			atomic.StoreInt64(&c.connectionErrors, 0)
+			c.logger.Info().Str("socket_path", c.socketPath).Msg("Connected to server")
-			c.lastHealthCheck = time.Now()
-			// Start health check monitoring if auto-reconnect is enabled
-			if c.autoReconnect {
-				c.startHealthCheck()
-			}
-			c.logger.Info().Str("socket_path", c.socketPath).Int("attempt", i+1).Msg("Connected to server")
 			return nil
 		}
-
+		// Exponential backoff starting from config
-		// Log connection attempt failure
+		backoffStart := GetConfig().BackoffStart
-		c.logger.Debug().Err(err).Str("socket_path", c.socketPath).Int("attempt", i+1).Int("max_attempts", maxAttempts).Msg("Connection attempt failed")
+		delay := time.Duration(backoffStart.Nanoseconds()*(1<<uint(i/3))) * time.Nanosecond
-
+		maxDelay := GetConfig().MaxRetryDelay
-		// Don't sleep after the last attempt
+		if delay > maxDelay {
-		if i < maxAttempts-1 {
+			delay = maxDelay
-			// Calculate adaptive delay based on connection failure patterns
-			delay := c.calculateAdaptiveDelay(i, initialDelay, maxDelay, backoffFactor)
-			time.Sleep(delay)
 		}
+		time.Sleep(delay)
 	}
 
 	// Ensure clean state on connection failure
 	c.conn = nil
 	c.running = false
-	return fmt.Errorf("failed to connect to audio server after %d attempts", Config.MaxConnectionAttempts)
+	return fmt.Errorf("failed to connect to audio server after 10 attempts")
 }
 
 // Disconnect disconnects the client from the server
@@ -528,9 +478,6 @@ func (c *UnifiedAudioClient) Disconnect() {
 
 	c.running = false
 
-	// Stop health check monitoring
-	c.stopHealthCheckMonitoring()
-
 	if c.conn != nil {
 		c.conn.Close()
 		c.conn = nil
@@ -550,129 +497,14 @@ func (c *UnifiedAudioClient) IsConnected() bool {
 func (c *UnifiedAudioClient) GetFrameStats() (total, dropped int64) {
 	total = atomic.LoadInt64(&c.totalFrames)
 	dropped = atomic.LoadInt64(&c.droppedFrames)
-	return
+	return total, dropped
-}
-
-// startHealthCheck starts the connection health monitoring
-func (c *UnifiedAudioClient) startHealthCheck() {
-	if c.healthCheckTicker != nil {
-		c.healthCheckTicker.Stop()
-	}
-
-	c.healthCheckTicker = time.NewTicker(Config.HealthCheckInterval)
-	go func() {
-		for {
-			select {
-			case <-c.healthCheckTicker.C:
-				c.performHealthCheck()
-			case <-c.stopHealthCheck:
-				return
-			}
-		}
-	}()
-}
-
-// stopHealthCheckMonitoring stops the health check monitoring
-func (c *UnifiedAudioClient) stopHealthCheckMonitoring() {
-	if c.healthCheckTicker != nil {
-		c.healthCheckTicker.Stop()
-		c.healthCheckTicker = nil
-	}
-	select {
-	case c.stopHealthCheck <- struct{}{}:
-	default:
-	}
-}
-
-// performHealthCheck checks the connection health and attempts reconnection if needed
-func (c *UnifiedAudioClient) performHealthCheck() {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
-
-	if !c.running || c.conn == nil {
-		return
-	}
-
-	// Simple health check: try to get connection info
-	if tcpConn, ok := c.conn.(*net.UnixConn); ok {
-		if _, err := tcpConn.File(); err != nil {
-			// Connection is broken
-			atomic.AddInt64(&c.connectionErrors, 1)
-			c.logger.Warn().Err(err).Msg("Connection health check failed, attempting reconnection")
-
-			// Close the broken connection
-			c.conn.Close()
-			c.conn = nil
-			c.running = false
-
-			// Attempt reconnection
-			go func() {
-				time.Sleep(Config.ReconnectionInterval)
-				if err := c.Connect(); err != nil {
-					c.logger.Error().Err(err).Msg("Failed to reconnect during health check")
-				}
-			}()
-		}
-	}
-
-	c.lastHealthCheck = time.Now()
-}
-
-// SetAutoReconnect enables or disables automatic reconnection
-func (c *UnifiedAudioClient) SetAutoReconnect(enabled bool) {
-	c.mtx.Lock()
-	defer c.mtx.Unlock()
-
-	c.autoReconnect = enabled
-	if !enabled {
-		c.stopHealthCheckMonitoring()
-	} else if c.running {
-		c.startHealthCheck()
-	}
-}
-
-// GetConnectionErrors returns the number of connection errors
-func (c *UnifiedAudioClient) GetConnectionErrors() int64 {
-	return atomic.LoadInt64(&c.connectionErrors)
-}
-
-// calculateAdaptiveDelay calculates retry delay based on system load and failure patterns
-func (c *UnifiedAudioClient) calculateAdaptiveDelay(attempt int, initialDelay, maxDelay time.Duration, backoffFactor float64) time.Duration {
-	// Base exponential backoff
-	baseDelay := time.Duration(float64(initialDelay.Nanoseconds()) * math.Pow(backoffFactor, float64(attempt)))
-
-	// Get connection error history for adaptive adjustment
-	errorCount := atomic.LoadInt64(&c.connectionErrors)
-
-	// Adjust delay based on recent connection errors
-	// More errors = longer delays to avoid overwhelming the server
-	adaptiveFactor := 1.0
-	if errorCount > 5 {
-		adaptiveFactor = 1.5 // 50% longer delays after many errors
-	} else if errorCount > 10 {
-		adaptiveFactor = 2.0 // Double delays after excessive errors
-	}
-
-	// Apply adaptive factor
-	adaptiveDelay := time.Duration(float64(baseDelay.Nanoseconds()) * adaptiveFactor)
-
-	// Ensure we don't exceed maximum delay
-	if adaptiveDelay > maxDelay {
-		adaptiveDelay = maxDelay
-	}
-
-	// Add small random jitter to avoid thundering herd
-	jitter := time.Duration(float64(adaptiveDelay.Nanoseconds()) * 0.1 * (0.5 + float64(attempt%3)/6.0))
-	adaptiveDelay += jitter
-
-	return adaptiveDelay
 }
 
 // Helper functions for socket paths
 func getInputSocketPath() string {
-	return filepath.Join("/var/run", inputSocketName)
+	return filepath.Join(os.TempDir(), inputSocketName)
 }
 
 func getOutputSocketPath() string {
-	return filepath.Join("/var/run", outputSocketName)
+	return filepath.Join(os.TempDir(), outputSocketName)
 }

internal/audio/mgmt_base_supervisor.go

@@ -28,6 +28,7 @@ type BaseSupervisor struct {
 	processPID int
 
 	// Process monitoring
+	processMonitor *ProcessMonitor
 
 	// Exit tracking
 	lastExitCode int
@@ -44,10 +45,10 @@ type BaseSupervisor struct {
 func NewBaseSupervisor(componentName string) *BaseSupervisor {
 	logger := logging.GetDefaultLogger().With().Str("component", componentName).Logger()
 	return &BaseSupervisor{
-		logger: &logger,
+		logger:         &logger,
-
+		processMonitor: GetProcessMonitor(),
-		stopChan:    make(chan struct{}),
+		stopChan:       make(chan struct{}),
-		processDone: make(chan struct{}),
+		processDone:    make(chan struct{}),
 	}
 }
 
@@ -210,6 +211,7 @@ func (bs *BaseSupervisor) waitForProcessExit(processType string) {
 	bs.mutex.Unlock()
 
 	// Remove process from monitoring
+	bs.processMonitor.RemoveProcess(pid)
 
 	if exitCode != 0 {
 		bs.logger.Error().Int("pid", pid).Int("exit_code", exitCode).Msgf("%s process exited with error", processType)

internal/audio/mgmt_input_ipc_manager.go

@@ -32,7 +32,7 @@ type AudioInputIPCManager struct {
 // NewAudioInputIPCManager creates a new IPC-based audio input manager
 func NewAudioInputIPCManager() *AudioInputIPCManager {
 	return &AudioInputIPCManager{
-		supervisor: GetAudioInputSupervisor(), // Use global shared supervisor
+		supervisor: NewAudioInputSupervisor(),
 		logger:     logging.GetDefaultLogger().With().Str("component", AudioInputIPCComponent).Logger(),
 	}
 }
@@ -63,9 +63,9 @@ func (aim *AudioInputIPCManager) Start() error {
 	}
 
 	config := InputIPCConfig{
-		SampleRate: Config.InputIPCSampleRate,
+		SampleRate: GetConfig().InputIPCSampleRate,
-		Channels:   Config.InputIPCChannels,
+		Channels:   GetConfig().InputIPCChannels,
-		FrameSize:  Config.InputIPCFrameSize,
+		FrameSize:  GetConfig().InputIPCFrameSize,
 	}
 
 	// Validate configuration before using it
@@ -80,7 +80,7 @@ func (aim *AudioInputIPCManager) Start() error {
 	}
 
 	// Wait for subprocess readiness
-	time.Sleep(Config.LongSleepDuration)
+	time.Sleep(GetConfig().LongSleepDuration)
 
 	err = aim.supervisor.SendConfig(config)
 	if err != nil {

internal/audio/mgmt_output_ipc_manager.go

@@ -57,9 +57,9 @@ func (aom *AudioOutputIPCManager) Start() error {
 
 	// Send initial configuration
 	config := OutputIPCConfig{
-		SampleRate: Config.SampleRate,
+		SampleRate: GetConfig().SampleRate,
-		Channels:   Config.Channels,
+		Channels:   GetConfig().Channels,
-		FrameSize:  int(Config.AudioQualityMediumFrameSize.Milliseconds()),
+		FrameSize:  int(GetConfig().AudioQualityMediumFrameSize.Milliseconds()),
 	}
 
 	if err := aom.SendConfig(config); err != nil {

internal/audio/mic_contention.go

@@ -105,7 +105,7 @@ func GetMicrophoneContentionManager() *MicrophoneContentionManager {
 	}
 
 	if atomic.CompareAndSwapInt32(&micContentionInitialized, 0, 1) {
-		manager := NewMicrophoneContentionManager(Config.MicContentionTimeout)
+		manager := NewMicrophoneContentionManager(GetConfig().MicContentionTimeout)
 		atomic.StorePointer(&globalMicContentionManager, unsafe.Pointer(manager))
 		return manager
 	}
@@ -115,7 +115,7 @@ func GetMicrophoneContentionManager() *MicrophoneContentionManager {
 		return (*MicrophoneContentionManager)(ptr)
 	}
 
-	return NewMicrophoneContentionManager(Config.MicContentionTimeout)
+	return NewMicrophoneContentionManager(GetConfig().MicContentionTimeout)
 }
 
 func TryMicrophoneOperation() OperationResult {

internal/audio/monitor_adaptive_optimizer.go

@@ -0,0 +1,198 @@
+package audio
+
+import (
+	"context"
+	"sync"
+	"sync/atomic"
+	"time"
+
+	"github.com/rs/zerolog"
+)
+
+// AdaptiveOptimizer automatically adjusts audio parameters based on latency metrics
+type AdaptiveOptimizer struct {
+	// Atomic fields MUST be first for ARM32 alignment (int64 fields need 8-byte alignment)
+	optimizationCount int64 // Number of optimizations performed (atomic)
+	lastOptimization  int64 // Timestamp of last optimization (atomic)
+	optimizationLevel int64 // Current optimization level (0-10) (atomic)
+
+	latencyMonitor *LatencyMonitor
+	bufferManager  *AdaptiveBufferManager
+	logger         zerolog.Logger
+
+	// Control channels
+	ctx    context.Context
+	cancel context.CancelFunc
+	wg     sync.WaitGroup
+
+	// Configuration
+	config OptimizerConfig
+}
+
+// OptimizerConfig holds configuration for the adaptive optimizer
+type OptimizerConfig struct {
+	MaxOptimizationLevel int           // Maximum optimization level (0-10)
+	CooldownPeriod       time.Duration // Minimum time between optimizations
+	Aggressiveness       float64       // How aggressively to optimize (0.0-1.0)
+	RollbackThreshold    time.Duration // Latency threshold to rollback optimizations
+	StabilityPeriod      time.Duration // Time to wait for stability after optimization
+}
+
+// DefaultOptimizerConfig returns a sensible default configuration
+func DefaultOptimizerConfig() OptimizerConfig {
+	return OptimizerConfig{
+		MaxOptimizationLevel: 8,
+		CooldownPeriod:       GetConfig().CooldownPeriod,
+		Aggressiveness:       GetConfig().OptimizerAggressiveness,
+		RollbackThreshold:    GetConfig().RollbackThreshold,
+		StabilityPeriod:      GetConfig().AdaptiveOptimizerStability,
+	}
+}
+
+// NewAdaptiveOptimizer creates a new adaptive optimizer
+func NewAdaptiveOptimizer(latencyMonitor *LatencyMonitor, bufferManager *AdaptiveBufferManager, config OptimizerConfig, logger zerolog.Logger) *AdaptiveOptimizer {
+	ctx, cancel := context.WithCancel(context.Background())
+
+	optimizer := &AdaptiveOptimizer{
+		latencyMonitor: latencyMonitor,
+		bufferManager:  bufferManager,
+		config:         config,
+		logger:         logger.With().Str("component", "adaptive-optimizer").Logger(),
+		ctx:            ctx,
+		cancel:         cancel,
+	}
+
+	// Register as latency monitor callback
+	latencyMonitor.AddOptimizationCallback(optimizer.handleLatencyOptimization)
+
+	return optimizer
+}
+
+// Start begins the adaptive optimization process
+func (ao *AdaptiveOptimizer) Start() {
+	ao.wg.Add(1)
+	go ao.optimizationLoop()
+	ao.logger.Debug().Msg("adaptive optimizer started")
+}
+
+// Stop stops the adaptive optimizer
+func (ao *AdaptiveOptimizer) Stop() {
+	ao.cancel()
+	ao.wg.Wait()
+	ao.logger.Debug().Msg("adaptive optimizer stopped")
+}
+
+// initializeStrategies sets up the available optimization strategies
+
+// handleLatencyOptimization is called when latency optimization is needed
+func (ao *AdaptiveOptimizer) handleLatencyOptimization(metrics LatencyMetrics) error {
+	currentLevel := atomic.LoadInt64(&ao.optimizationLevel)
+	lastOpt := atomic.LoadInt64(&ao.lastOptimization)
+
+	// Check cooldown period
+	if time.Since(time.Unix(0, lastOpt)) < ao.config.CooldownPeriod {
+		return nil
+	}
+
+	// Determine if we need to increase or decrease optimization level
+	targetLevel := ao.calculateTargetOptimizationLevel(metrics)
+
+	if targetLevel > currentLevel {
+		return ao.increaseOptimization(int(targetLevel))
+	} else if targetLevel < currentLevel {
+		return ao.decreaseOptimization(int(targetLevel))
+	}
+
+	return nil
+}
+
+// calculateTargetOptimizationLevel determines the appropriate optimization level
+func (ao *AdaptiveOptimizer) calculateTargetOptimizationLevel(metrics LatencyMetrics) int64 {
+	// Base calculation on current latency vs target
+	latencyRatio := float64(metrics.Current) / float64(GetConfig().AdaptiveOptimizerLatencyTarget) // 50ms target
+
+	// Adjust based on trend
+	switch metrics.Trend {
+	case LatencyTrendIncreasing:
+		latencyRatio *= 1.2 // Be more aggressive
+	case LatencyTrendDecreasing:
+		latencyRatio *= 0.8 // Be less aggressive
+	case LatencyTrendVolatile:
+		latencyRatio *= 1.1 // Slightly more aggressive
+	}
+
+	// Apply aggressiveness factor
+	latencyRatio *= ao.config.Aggressiveness
+
+	// Convert to optimization level
+	targetLevel := int64(latencyRatio * GetConfig().LatencyScalingFactor) // Scale to 0-10 range
+	if targetLevel > int64(ao.config.MaxOptimizationLevel) {
+		targetLevel = int64(ao.config.MaxOptimizationLevel)
+	}
+	if targetLevel < 0 {
+		targetLevel = 0
+	}
+
+	return targetLevel
+}
+
+// increaseOptimization applies optimization strategies up to the target level
+func (ao *AdaptiveOptimizer) increaseOptimization(targetLevel int) error {
+	atomic.StoreInt64(&ao.optimizationLevel, int64(targetLevel))
+	atomic.StoreInt64(&ao.lastOptimization, time.Now().UnixNano())
+	atomic.AddInt64(&ao.optimizationCount, 1)
+
+	return nil
+}
+
+// decreaseOptimization rolls back optimization strategies to the target level
+func (ao *AdaptiveOptimizer) decreaseOptimization(targetLevel int) error {
+	atomic.StoreInt64(&ao.optimizationLevel, int64(targetLevel))
+	atomic.StoreInt64(&ao.lastOptimization, time.Now().UnixNano())
+
+	return nil
+}
+
+// optimizationLoop runs the main optimization monitoring loop
+func (ao *AdaptiveOptimizer) optimizationLoop() {
+	defer ao.wg.Done()
+
+	ticker := time.NewTicker(ao.config.StabilityPeriod)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-ao.ctx.Done():
+			return
+		case <-ticker.C:
+			ao.checkStability()
+		}
+	}
+}
+
+// checkStability monitors system stability and rolls back if needed
+func (ao *AdaptiveOptimizer) checkStability() {
+	metrics := ao.latencyMonitor.GetMetrics()
+
+	// Check if we need to rollback due to excessive latency
+	if metrics.Current > ao.config.RollbackThreshold {
+		currentLevel := int(atomic.LoadInt64(&ao.optimizationLevel))
+		if currentLevel > 0 {
+			ao.logger.Warn().Dur("current_latency", metrics.Current).Dur("threshold", ao.config.RollbackThreshold).Msg("rolling back optimizations due to excessive latency")
+			if err := ao.decreaseOptimization(currentLevel - 1); err != nil {
+				ao.logger.Error().Err(err).Msg("failed to decrease optimization level")
+			}
+		}
+	}
+}
+
+// GetOptimizationStats returns current optimization statistics
+func (ao *AdaptiveOptimizer) GetOptimizationStats() map[string]interface{} {
+	return map[string]interface{}{
+		"optimization_level": atomic.LoadInt64(&ao.optimizationLevel),
+		"optimization_count": atomic.LoadInt64(&ao.optimizationCount),
+		"last_optimization":  time.Unix(0, atomic.LoadInt64(&ao.lastOptimization)),
+	}
+}
+
+// Strategy implementation methods (stubs for now)

internal/audio/monitor_goroutine.go

@@ -0,0 +1,144 @@
+package audio
+
+import (
+	"runtime"
+	"sync/atomic"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+)
+
+// GoroutineMonitor tracks goroutine count and provides cleanup mechanisms
+type GoroutineMonitor struct {
+	baselineCount   int
+	peakCount       int
+	lastCount       int
+	monitorInterval time.Duration
+	lastCheck       time.Time
+	enabled         int32
+}
+
+// Global goroutine monitor instance
+var globalGoroutineMonitor *GoroutineMonitor
+
+// NewGoroutineMonitor creates a new goroutine monitor
+func NewGoroutineMonitor(monitorInterval time.Duration) *GoroutineMonitor {
+	if monitorInterval <= 0 {
+		monitorInterval = 30 * time.Second
+	}
+
+	// Get current goroutine count as baseline
+	baselineCount := runtime.NumGoroutine()
+
+	return &GoroutineMonitor{
+		baselineCount:   baselineCount,
+		peakCount:       baselineCount,
+		lastCount:       baselineCount,
+		monitorInterval: monitorInterval,
+		lastCheck:       time.Now(),
+	}
+}
+
+// Start begins goroutine monitoring
+func (gm *GoroutineMonitor) Start() {
+	if !atomic.CompareAndSwapInt32(&gm.enabled, 0, 1) {
+		return // Already running
+	}
+
+	go gm.monitorLoop()
+}
+
+// Stop stops goroutine monitoring
+func (gm *GoroutineMonitor) Stop() {
+	atomic.StoreInt32(&gm.enabled, 0)
+}
+
+// monitorLoop periodically checks goroutine count
+func (gm *GoroutineMonitor) monitorLoop() {
+	logger := logging.GetDefaultLogger().With().Str("component", "goroutine-monitor").Logger()
+	logger.Info().Int("baseline", gm.baselineCount).Msg("goroutine monitor started")
+
+	for atomic.LoadInt32(&gm.enabled) == 1 {
+		time.Sleep(gm.monitorInterval)
+		gm.checkGoroutineCount()
+	}
+
+	logger.Info().Msg("goroutine monitor stopped")
+}
+
+// checkGoroutineCount checks current goroutine count and logs if it exceeds thresholds
+func (gm *GoroutineMonitor) checkGoroutineCount() {
+	currentCount := runtime.NumGoroutine()
+	gm.lastCount = currentCount
+
+	// Update peak count if needed
+	if currentCount > gm.peakCount {
+		gm.peakCount = currentCount
+	}
+
+	// Calculate growth since baseline
+	growth := currentCount - gm.baselineCount
+	growthPercent := float64(growth) / float64(gm.baselineCount) * 100
+
+	// Log warning if growth exceeds thresholds
+	logger := logging.GetDefaultLogger().With().Str("component", "goroutine-monitor").Logger()
+
+	// Different log levels based on growth severity
+	if growthPercent > 30 {
+		// Severe growth - trigger cleanup
+		logger.Warn().Int("current", currentCount).Int("baseline", gm.baselineCount).
+			Int("growth", growth).Float64("growth_percent", growthPercent).
+			Msg("excessive goroutine growth detected - triggering cleanup")
+
+		// Force garbage collection to clean up unused resources
+		runtime.GC()
+
+		// Force cleanup of goroutine buffer cache
+		cleanupGoroutineCache()
+	} else if growthPercent > 20 {
+		// Moderate growth - just log warning
+		logger.Warn().Int("current", currentCount).Int("baseline", gm.baselineCount).
+			Int("growth", growth).Float64("growth_percent", growthPercent).
+			Msg("significant goroutine growth detected")
+	} else if growthPercent > 10 {
+		// Minor growth - log info
+		logger.Info().Int("current", currentCount).Int("baseline", gm.baselineCount).
+			Int("growth", growth).Float64("growth_percent", growthPercent).
+			Msg("goroutine growth detected")
+	}
+
+	// Update last check time
+	gm.lastCheck = time.Now()
+}
+
+// GetGoroutineStats returns current goroutine statistics
+func (gm *GoroutineMonitor) GetGoroutineStats() map[string]interface{} {
+	return map[string]interface{}{
+		"current_count":  gm.lastCount,
+		"baseline_count": gm.baselineCount,
+		"peak_count":     gm.peakCount,
+		"growth":         gm.lastCount - gm.baselineCount,
+		"growth_percent": float64(gm.lastCount-gm.baselineCount) / float64(gm.baselineCount) * 100,
+		"last_check":     gm.lastCheck,
+	}
+}
+
+// GetGoroutineMonitor returns the global goroutine monitor instance
+func GetGoroutineMonitor() *GoroutineMonitor {
+	if globalGoroutineMonitor == nil {
+		globalGoroutineMonitor = NewGoroutineMonitor(GetConfig().GoroutineMonitorInterval)
+	}
+	return globalGoroutineMonitor
+}
+
+// StartGoroutineMonitoring starts the global goroutine monitor
+func StartGoroutineMonitoring() {
+	// Goroutine monitoring disabled
+}
+
+// StopGoroutineMonitoring stops the global goroutine monitor
+func StopGoroutineMonitoring() {
+	if globalGoroutineMonitor != nil {
+		globalGoroutineMonitor.Stop()
+	}
+}

internal/audio/monitor_latency.go

@@ -0,0 +1,333 @@
+package audio
+
+import (
+	"context"
+	"sync"
+	"sync/atomic"
+	"time"
+
+	"github.com/rs/zerolog"
+)
+
+// LatencyMonitor tracks and optimizes audio latency in real-time
+type LatencyMonitor struct {
+	// Atomic fields MUST be first for ARM32 alignment (int64 fields need 8-byte alignment)
+	currentLatency    int64 // Current latency in nanoseconds (atomic)
+	averageLatency    int64 // Rolling average latency in nanoseconds (atomic)
+	minLatency        int64 // Minimum observed latency in nanoseconds (atomic)
+	maxLatency        int64 // Maximum observed latency in nanoseconds (atomic)
+	latencySamples    int64 // Number of latency samples collected (atomic)
+	jitterAccumulator int64 // Accumulated jitter for variance calculation (atomic)
+	lastOptimization  int64 // Timestamp of last optimization in nanoseconds (atomic)
+
+	config LatencyConfig
+	logger zerolog.Logger
+
+	// Control channels
+	ctx    context.Context
+	cancel context.CancelFunc
+	wg     sync.WaitGroup
+
+	// Optimization callbacks
+	optimizationCallbacks []OptimizationCallback
+	mutex                 sync.RWMutex
+
+	// Performance tracking
+	latencyHistory []LatencyMeasurement
+	historyMutex   sync.RWMutex
+}
+
+// LatencyConfig holds configuration for latency monitoring
+type LatencyConfig struct {
+	TargetLatency        time.Duration // Target latency to maintain
+	MaxLatency           time.Duration // Maximum acceptable latency
+	OptimizationInterval time.Duration // How often to run optimization
+	HistorySize          int           // Number of latency measurements to keep
+	JitterThreshold      time.Duration // Jitter threshold for optimization
+	AdaptiveThreshold    float64       // Threshold for adaptive adjustments (0.0-1.0)
+}
+
+// LatencyMeasurement represents a single latency measurement
+type LatencyMeasurement struct {
+	Timestamp time.Time
+	Latency   time.Duration
+	Jitter    time.Duration
+	Source    string // Source of the measurement (e.g., "input", "output", "processing")
+}
+
+// OptimizationCallback is called when latency optimization is triggered
+type OptimizationCallback func(metrics LatencyMetrics) error
+
+// LatencyMetrics provides comprehensive latency statistics
+type LatencyMetrics struct {
+	Current     time.Duration
+	Average     time.Duration
+	Min         time.Duration
+	Max         time.Duration
+	Jitter      time.Duration
+	SampleCount int64
+	Trend       LatencyTrend
+}
+
+// LatencyTrend indicates the direction of latency changes
+type LatencyTrend int
+
+const (
+	LatencyTrendStable LatencyTrend = iota
+	LatencyTrendIncreasing
+	LatencyTrendDecreasing
+	LatencyTrendVolatile
+)
+
+// DefaultLatencyConfig returns a sensible default configuration
+func DefaultLatencyConfig() LatencyConfig {
+	config := GetConfig()
+	return LatencyConfig{
+		TargetLatency:        config.LatencyMonitorTarget,
+		MaxLatency:           config.MaxLatencyThreshold,
+		OptimizationInterval: config.LatencyOptimizationInterval,
+		HistorySize:          config.LatencyHistorySize,
+		JitterThreshold:      config.JitterThreshold,
+		AdaptiveThreshold:    config.LatencyAdaptiveThreshold,
+	}
+}
+
+// NewLatencyMonitor creates a new latency monitoring system
+func NewLatencyMonitor(config LatencyConfig, logger zerolog.Logger) *LatencyMonitor {
+	// Validate latency configuration
+	if err := ValidateLatencyConfig(config); err != nil {
+		// Log validation error and use default configuration
+		logger.Error().Err(err).Msg("Invalid latency configuration provided, using defaults")
+		config = DefaultLatencyConfig()
+	}
+
+	ctx, cancel := context.WithCancel(context.Background())
+
+	return &LatencyMonitor{
+		config:         config,
+		logger:         logger.With().Str("component", "latency-monitor").Logger(),
+		ctx:            ctx,
+		cancel:         cancel,
+		latencyHistory: make([]LatencyMeasurement, 0, config.HistorySize),
+		minLatency:     int64(time.Hour), // Initialize to high value
+	}
+}
+
+// Start begins latency monitoring and optimization
+func (lm *LatencyMonitor) Start() {
+	lm.wg.Add(1)
+	go lm.monitoringLoop()
+}
+
+// Stop stops the latency monitor
+func (lm *LatencyMonitor) Stop() {
+	lm.cancel()
+	lm.wg.Wait()
+}
+
+// RecordLatency records a new latency measurement
+func (lm *LatencyMonitor) RecordLatency(latency time.Duration, source string) {
+	now := time.Now()
+	latencyNanos := latency.Nanoseconds()
+
+	// Update atomic counters
+	atomic.StoreInt64(&lm.currentLatency, latencyNanos)
+	atomic.AddInt64(&lm.latencySamples, 1)
+
+	// Update min/max
+	for {
+		oldMin := atomic.LoadInt64(&lm.minLatency)
+		if latencyNanos >= oldMin || atomic.CompareAndSwapInt64(&lm.minLatency, oldMin, latencyNanos) {
+			break
+		}
+	}
+
+	for {
+		oldMax := atomic.LoadInt64(&lm.maxLatency)
+		if latencyNanos <= oldMax || atomic.CompareAndSwapInt64(&lm.maxLatency, oldMax, latencyNanos) {
+			break
+		}
+	}
+
+	// Update rolling average using exponential moving average
+	oldAvg := atomic.LoadInt64(&lm.averageLatency)
+	newAvg := oldAvg + (latencyNanos-oldAvg)/10 // Alpha = 0.1
+	atomic.StoreInt64(&lm.averageLatency, newAvg)
+
+	// Calculate jitter (difference from average)
+	jitter := latencyNanos - newAvg
+	if jitter < 0 {
+		jitter = -jitter
+	}
+	atomic.AddInt64(&lm.jitterAccumulator, jitter)
+
+	// Store in history
+	lm.historyMutex.Lock()
+	measurement := LatencyMeasurement{
+		Timestamp: now,
+		Latency:   latency,
+		Jitter:    time.Duration(jitter),
+		Source:    source,
+	}
+
+	if len(lm.latencyHistory) >= lm.config.HistorySize {
+		// Remove oldest measurement
+		copy(lm.latencyHistory, lm.latencyHistory[1:])
+		lm.latencyHistory[len(lm.latencyHistory)-1] = measurement
+	} else {
+		lm.latencyHistory = append(lm.latencyHistory, measurement)
+	}
+	lm.historyMutex.Unlock()
+}
+
+// GetMetrics returns current latency metrics
+func (lm *LatencyMonitor) GetMetrics() LatencyMetrics {
+	current := atomic.LoadInt64(&lm.currentLatency)
+	average := atomic.LoadInt64(&lm.averageLatency)
+	min := atomic.LoadInt64(&lm.minLatency)
+	max := atomic.LoadInt64(&lm.maxLatency)
+	samples := atomic.LoadInt64(&lm.latencySamples)
+	jitterSum := atomic.LoadInt64(&lm.jitterAccumulator)
+
+	var jitter time.Duration
+	if samples > 0 {
+		jitter = time.Duration(jitterSum / samples)
+	}
+
+	return LatencyMetrics{
+		Current:     time.Duration(current),
+		Average:     time.Duration(average),
+		Min:         time.Duration(min),
+		Max:         time.Duration(max),
+		Jitter:      jitter,
+		SampleCount: samples,
+		Trend:       lm.calculateTrend(),
+	}
+}
+
+// AddOptimizationCallback adds a callback for latency optimization
+func (lm *LatencyMonitor) AddOptimizationCallback(callback OptimizationCallback) {
+	lm.mutex.Lock()
+	lm.optimizationCallbacks = append(lm.optimizationCallbacks, callback)
+	lm.mutex.Unlock()
+}
+
+// monitoringLoop runs the main monitoring and optimization loop
+func (lm *LatencyMonitor) monitoringLoop() {
+	defer lm.wg.Done()
+
+	ticker := time.NewTicker(lm.config.OptimizationInterval)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-lm.ctx.Done():
+			return
+		case <-ticker.C:
+			lm.runOptimization()
+		}
+	}
+}
+
+// runOptimization checks if optimization is needed and triggers callbacks with threshold validation.
+//
+// Validation Rules:
+//   - Current latency must not exceed MaxLatency (default: 200ms)
+//   - Average latency checked against adaptive threshold: TargetLatency * (1 + AdaptiveThreshold)
+//   - Jitter must not exceed JitterThreshold (default: 20ms)
+//   - All latency values must be non-negative durations
+//
+// Optimization Triggers:
+//   - Current latency > MaxLatency: Immediate optimization needed
+//   - Average latency > adaptive threshold: Gradual optimization needed
+//   - Jitter > JitterThreshold: Stability optimization needed
+//
+// Threshold Calculations:
+//   - Adaptive threshold = TargetLatency * (1.0 + AdaptiveThreshold)
+//   - Default: 50ms * (1.0 + 0.8) = 90ms adaptive threshold
+//   - Provides buffer above target before triggering optimization
+//
+// The function ensures real-time audio performance by monitoring multiple
+// latency metrics and triggering optimization callbacks when thresholds are exceeded.
+func (lm *LatencyMonitor) runOptimization() {
+	metrics := lm.GetMetrics()
+
+	// Check if optimization is needed
+	needsOptimization := false
+
+	// Check if current latency exceeds threshold
+	if metrics.Current > lm.config.MaxLatency {
+		needsOptimization = true
+		lm.logger.Warn().Dur("current_latency", metrics.Current).Dur("max_latency", lm.config.MaxLatency).Msg("latency exceeds maximum threshold")
+	}
+
+	// Check if average latency is above adaptive threshold
+	adaptiveThreshold := time.Duration(float64(lm.config.TargetLatency.Nanoseconds()) * (1.0 + lm.config.AdaptiveThreshold))
+	if metrics.Average > adaptiveThreshold {
+		needsOptimization = true
+	}
+
+	// Check if jitter is too high
+	if metrics.Jitter > lm.config.JitterThreshold {
+		needsOptimization = true
+	}
+
+	if needsOptimization {
+		atomic.StoreInt64(&lm.lastOptimization, time.Now().UnixNano())
+
+		// Run optimization callbacks
+		lm.mutex.RLock()
+		callbacks := make([]OptimizationCallback, len(lm.optimizationCallbacks))
+		copy(callbacks, lm.optimizationCallbacks)
+		lm.mutex.RUnlock()
+
+		for _, callback := range callbacks {
+			if err := callback(metrics); err != nil {
+				lm.logger.Error().Err(err).Msg("optimization callback failed")
+			}
+		}
+	}
+}
+
+// calculateTrend analyzes recent latency measurements to determine trend
+func (lm *LatencyMonitor) calculateTrend() LatencyTrend {
+	lm.historyMutex.RLock()
+	defer lm.historyMutex.RUnlock()
+
+	if len(lm.latencyHistory) < 10 {
+		return LatencyTrendStable
+	}
+
+	// Analyze last 10 measurements
+	recentMeasurements := lm.latencyHistory[len(lm.latencyHistory)-10:]
+
+	var increasing, decreasing int
+	for i := 1; i < len(recentMeasurements); i++ {
+		if recentMeasurements[i].Latency > recentMeasurements[i-1].Latency {
+			increasing++
+		} else if recentMeasurements[i].Latency < recentMeasurements[i-1].Latency {
+			decreasing++
+		}
+	}
+
+	// Determine trend based on direction changes
+	if increasing > 6 {
+		return LatencyTrendIncreasing
+	} else if decreasing > 6 {
+		return LatencyTrendDecreasing
+	} else if increasing+decreasing > 7 {
+		return LatencyTrendVolatile
+	}
+
+	return LatencyTrendStable
+}
+
+// GetLatencyHistory returns a copy of recent latency measurements
+func (lm *LatencyMonitor) GetLatencyHistory() []LatencyMeasurement {
+	lm.historyMutex.RLock()
+	defer lm.historyMutex.RUnlock()
+
+	history := make([]LatencyMeasurement, len(lm.latencyHistory))
+	copy(history, lm.latencyHistory)
+	return history
+}

internal/audio/monitor_process.go

@@ -0,0 +1,406 @@
+package audio
+
+import (
+	"bufio"
+	"fmt"
+	"os"
+	"strconv"
+	"strings"
+	"sync"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// Variables for process monitoring (using configuration)
+var (
+	// System constants
+	maxCPUPercent     = GetConfig().MaxCPUPercent
+	minCPUPercent     = GetConfig().MinCPUPercent
+	defaultClockTicks = GetConfig().DefaultClockTicks
+	defaultMemoryGB   = GetConfig().DefaultMemoryGB
+
+	// Monitoring thresholds
+	maxWarmupSamples = GetConfig().MaxWarmupSamples
+	warmupCPUSamples = GetConfig().WarmupCPUSamples
+
+	// Channel buffer size
+	metricsChannelBuffer = GetConfig().MetricsChannelBuffer
+
+	// Clock tick detection ranges
+	minValidClockTicks = float64(GetConfig().MinValidClockTicks)
+	maxValidClockTicks = float64(GetConfig().MaxValidClockTicks)
+)
+
+// Variables for process monitoring
+var (
+	pageSize = GetConfig().PageSize
+)
+
+// ProcessMetrics represents CPU and memory usage metrics for a process
+type ProcessMetrics struct {
+	PID           int       `json:"pid"`
+	CPUPercent    float64   `json:"cpu_percent"`
+	MemoryRSS     int64     `json:"memory_rss_bytes"`
+	MemoryVMS     int64     `json:"memory_vms_bytes"`
+	MemoryPercent float64   `json:"memory_percent"`
+	Timestamp     time.Time `json:"timestamp"`
+	ProcessName   string    `json:"process_name"`
+}
+
+type ProcessMonitor struct {
+	logger         zerolog.Logger
+	mutex          sync.RWMutex
+	monitoredPIDs  map[int]*processState
+	running        bool
+	stopChan       chan struct{}
+	metricsChan    chan ProcessMetrics
+	updateInterval time.Duration
+	totalMemory    int64
+	memoryOnce     sync.Once
+	clockTicks     float64
+	clockTicksOnce sync.Once
+}
+
+// processState tracks the state needed for CPU calculation
+type processState struct {
+	name          string
+	lastCPUTime   int64
+	lastSysTime   int64
+	lastUserTime  int64
+	lastSample    time.Time
+	warmupSamples int
+}
+
+// NewProcessMonitor creates a new process monitor
+func NewProcessMonitor() *ProcessMonitor {
+	return &ProcessMonitor{
+		logger:         logging.GetDefaultLogger().With().Str("component", "process-monitor").Logger(),
+		monitoredPIDs:  make(map[int]*processState),
+		stopChan:       make(chan struct{}),
+		metricsChan:    make(chan ProcessMetrics, metricsChannelBuffer),
+		updateInterval: GetMetricsUpdateInterval(),
+	}
+}
+
+// Start begins monitoring processes
+func (pm *ProcessMonitor) Start() {
+	pm.mutex.Lock()
+	defer pm.mutex.Unlock()
+
+	if pm.running {
+		return
+	}
+
+	pm.running = true
+	go pm.monitorLoop()
+	pm.logger.Debug().Msg("process monitor started")
+}
+
+// Stop stops monitoring processes
+func (pm *ProcessMonitor) Stop() {
+	pm.mutex.Lock()
+	defer pm.mutex.Unlock()
+
+	if !pm.running {
+		return
+	}
+
+	pm.running = false
+	close(pm.stopChan)
+	pm.logger.Debug().Msg("process monitor stopped")
+}
+
+// AddProcess adds a process to monitor
+func (pm *ProcessMonitor) AddProcess(pid int, name string) {
+	pm.mutex.Lock()
+	defer pm.mutex.Unlock()
+
+	pm.monitoredPIDs[pid] = &processState{
+		name:       name,
+		lastSample: time.Now(),
+	}
+	pm.logger.Info().Int("pid", pid).Str("name", name).Msg("Added process to monitor")
+}
+
+// RemoveProcess removes a process from monitoring
+func (pm *ProcessMonitor) RemoveProcess(pid int) {
+	pm.mutex.Lock()
+	defer pm.mutex.Unlock()
+
+	delete(pm.monitoredPIDs, pid)
+	pm.logger.Info().Int("pid", pid).Msg("Removed process from monitor")
+}
+
+// GetMetricsChan returns the channel for receiving metrics
+func (pm *ProcessMonitor) GetMetricsChan() <-chan ProcessMetrics {
+	return pm.metricsChan
+}
+
+// GetCurrentMetrics returns current metrics for all monitored processes
+func (pm *ProcessMonitor) GetCurrentMetrics() []ProcessMetrics {
+	pm.mutex.RLock()
+	defer pm.mutex.RUnlock()
+
+	var metrics []ProcessMetrics
+	for pid, state := range pm.monitoredPIDs {
+		if metric, err := pm.collectMetrics(pid, state); err == nil {
+			metrics = append(metrics, metric)
+		}
+	}
+	return metrics
+}
+
+// monitorLoop is the main monitoring loop
+func (pm *ProcessMonitor) monitorLoop() {
+	ticker := time.NewTicker(pm.updateInterval)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-pm.stopChan:
+			return
+		case <-ticker.C:
+			pm.collectAllMetrics()
+		}
+	}
+}
+
+func (pm *ProcessMonitor) collectAllMetrics() {
+	pm.mutex.RLock()
+	pidsToCheck := make([]int, 0, len(pm.monitoredPIDs))
+	states := make([]*processState, 0, len(pm.monitoredPIDs))
+	for pid, state := range pm.monitoredPIDs {
+		pidsToCheck = append(pidsToCheck, pid)
+		states = append(states, state)
+	}
+	pm.mutex.RUnlock()
+
+	deadPIDs := make([]int, 0)
+	for i, pid := range pidsToCheck {
+		if metric, err := pm.collectMetrics(pid, states[i]); err == nil {
+			select {
+			case pm.metricsChan <- metric:
+			default:
+			}
+		} else {
+			deadPIDs = append(deadPIDs, pid)
+		}
+	}
+
+	for _, pid := range deadPIDs {
+		pm.RemoveProcess(pid)
+	}
+}
+
+func (pm *ProcessMonitor) collectMetrics(pid int, state *processState) (ProcessMetrics, error) {
+	now := time.Now()
+	metric := ProcessMetrics{
+		PID:         pid,
+		Timestamp:   now,
+		ProcessName: state.name,
+	}
+
+	statPath := fmt.Sprintf("/proc/%d/stat", pid)
+	statData, err := os.ReadFile(statPath)
+	if err != nil {
+		return metric, fmt.Errorf("failed to read process statistics from /proc/%d/stat: %w", pid, err)
+	}
+
+	fields := strings.Fields(string(statData))
+	if len(fields) < 24 {
+		return metric, fmt.Errorf("invalid process stat format: expected at least 24 fields, got %d from /proc/%d/stat", len(fields), pid)
+	}
+
+	utime, _ := strconv.ParseInt(fields[13], 10, 64)
+	stime, _ := strconv.ParseInt(fields[14], 10, 64)
+	totalCPUTime := utime + stime
+
+	vsize, _ := strconv.ParseInt(fields[22], 10, 64)
+	rss, _ := strconv.ParseInt(fields[23], 10, 64)
+
+	metric.MemoryRSS = rss * int64(pageSize)
+	metric.MemoryVMS = vsize
+
+	// Calculate CPU percentage
+	metric.CPUPercent = pm.calculateCPUPercent(totalCPUTime, state, now)
+
+	// Increment warmup counter
+	if state.warmupSamples < maxWarmupSamples {
+		state.warmupSamples++
+	}
+
+	// Calculate memory percentage (RSS / total system memory)
+	if totalMem := pm.getTotalMemory(); totalMem > 0 {
+		metric.MemoryPercent = float64(metric.MemoryRSS) / float64(totalMem) * GetConfig().PercentageMultiplier
+	}
+
+	// Update state for next calculation
+	state.lastCPUTime = totalCPUTime
+	state.lastUserTime = utime
+	state.lastSysTime = stime
+	state.lastSample = now
+
+	return metric, nil
+}
+
+// calculateCPUPercent calculates CPU percentage for a process with validation and bounds checking.
+//
+// Validation Rules:
+//   - Returns 0.0 for first sample (no baseline for comparison)
+//   - Requires positive time delta between samples
+//   - Applies CPU percentage bounds: [MinCPUPercent, MaxCPUPercent]
+//   - Uses system clock ticks for accurate CPU time conversion
+//   - Validates clock ticks within range [MinValidClockTicks, MaxValidClockTicks]
+//
+// Bounds Applied:
+//   - CPU percentage clamped to [0.01%, 100.0%] (default values)
+//   - Clock ticks validated within [50, 1000] range (default values)
+//   - Time delta must be > 0 to prevent division by zero
+//
+// Warmup Behavior:
+//   - During warmup period (< WarmupCPUSamples), returns MinCPUPercent for idle processes
+//   - This indicates process is alive but not consuming significant CPU
+//
+// The function ensures accurate CPU percentage calculation while preventing
+// invalid measurements that could affect system monitoring and adaptive algorithms.
+func (pm *ProcessMonitor) calculateCPUPercent(totalCPUTime int64, state *processState, now time.Time) float64 {
+	if state.lastSample.IsZero() {
+		// First sample - initialize baseline
+		state.warmupSamples = 0
+		return 0.0
+	}
+
+	timeDelta := now.Sub(state.lastSample).Seconds()
+	cpuDelta := float64(totalCPUTime - state.lastCPUTime)
+
+	if timeDelta <= 0 {
+		return 0.0
+	}
+
+	if cpuDelta > 0 {
+		// Convert from clock ticks to seconds using actual system clock ticks
+		clockTicks := pm.getClockTicks()
+		cpuSeconds := cpuDelta / clockTicks
+		cpuPercent := (cpuSeconds / timeDelta) * GetConfig().PercentageMultiplier
+
+		// Apply bounds
+		if cpuPercent > maxCPUPercent {
+			cpuPercent = maxCPUPercent
+		}
+		if cpuPercent < minCPUPercent {
+			cpuPercent = minCPUPercent
+		}
+
+		return cpuPercent
+	}
+
+	// No CPU delta - process was idle
+	if state.warmupSamples < warmupCPUSamples {
+		// During warmup, provide a small non-zero value to indicate process is alive
+		return minCPUPercent
+	}
+
+	return 0.0
+}
+
+func (pm *ProcessMonitor) getClockTicks() float64 {
+	pm.clockTicksOnce.Do(func() {
+		// Try to detect actual clock ticks from kernel boot parameters or /proc/stat
+		if data, err := os.ReadFile("/proc/cmdline"); err == nil {
+			// Look for HZ parameter in kernel command line
+			cmdline := string(data)
+			if strings.Contains(cmdline, "HZ=") {
+				fields := strings.Fields(cmdline)
+				for _, field := range fields {
+					if strings.HasPrefix(field, "HZ=") {
+						if hz, err := strconv.ParseFloat(field[3:], 64); err == nil && hz > 0 {
+							pm.clockTicks = hz
+							return
+						}
+					}
+				}
+			}
+		}
+
+		// Try reading from /proc/timer_list for more accurate detection
+		if data, err := os.ReadFile("/proc/timer_list"); err == nil {
+			timer := string(data)
+			// Look for tick device frequency
+			lines := strings.Split(timer, "\n")
+			for _, line := range lines {
+				if strings.Contains(line, "tick_period:") {
+					fields := strings.Fields(line)
+					if len(fields) >= 2 {
+						if period, err := strconv.ParseInt(fields[1], 10, 64); err == nil && period > 0 {
+							// Convert nanoseconds to Hz
+							hz := GetConfig().CGONanosecondsPerSecond / float64(period)
+							if hz >= minValidClockTicks && hz <= maxValidClockTicks {
+								pm.clockTicks = hz
+								return
+							}
+						}
+					}
+				}
+			}
+		}
+
+		// Fallback: Most embedded ARM systems (like jetKVM) use 250 Hz or 1000 Hz
+		// rather than the traditional 100 Hz
+		pm.clockTicks = defaultClockTicks
+		pm.logger.Warn().Float64("clock_ticks", pm.clockTicks).Msg("Using fallback clock ticks value")
+
+		// Log successful detection for non-fallback values
+		if pm.clockTicks != defaultClockTicks {
+			pm.logger.Info().Float64("clock_ticks", pm.clockTicks).Msg("Detected system clock ticks")
+		}
+	})
+	return pm.clockTicks
+}
+
+func (pm *ProcessMonitor) getTotalMemory() int64 {
+	pm.memoryOnce.Do(func() {
+		file, err := os.Open("/proc/meminfo")
+		if err != nil {
+			pm.totalMemory = int64(defaultMemoryGB) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes)
+			return
+		}
+		defer file.Close()
+
+		scanner := bufio.NewScanner(file)
+		for scanner.Scan() {
+			line := scanner.Text()
+			if strings.HasPrefix(line, "MemTotal:") {
+				fields := strings.Fields(line)
+				if len(fields) >= 2 {
+					if kb, err := strconv.ParseInt(fields[1], 10, 64); err == nil {
+						pm.totalMemory = kb * int64(GetConfig().ProcessMonitorKBToBytes)
+						return
+					}
+				}
+				break
+			}
+		}
+		pm.totalMemory = int64(defaultMemoryGB) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes) // Fallback
+	})
+	return pm.totalMemory
+}
+
+// GetTotalMemory returns total system memory in bytes (public method)
+func (pm *ProcessMonitor) GetTotalMemory() int64 {
+	return pm.getTotalMemory()
+}
+
+// Global process monitor instance
+var globalProcessMonitor *ProcessMonitor
+var processMonitorOnce sync.Once
+
+// GetProcessMonitor returns the global process monitor instance
+func GetProcessMonitor() *ProcessMonitor {
+	processMonitorOnce.Do(func() {
+		globalProcessMonitor = NewProcessMonitor()
+		globalProcessMonitor.Start()
+	})
+	return globalProcessMonitor
+}

internal/audio/output_server_main.go

@@ -70,7 +70,7 @@ func RunAudioOutputServer() error {
 	StopNonBlockingAudioStreaming()
 
 	// Give some time for cleanup
-	time.Sleep(Config.DefaultSleepDuration)
+	time.Sleep(GetConfig().DefaultSleepDuration)
 
 	return nil
 }

internal/audio/output_streaming.go

@@ -48,6 +48,9 @@ func getOutputStreamingLogger() *zerolog.Logger {
 
 // StartAudioOutputStreaming starts audio output streaming (capturing system audio)
 func StartAudioOutputStreaming(send func([]byte)) error {
+	// Initialize audio monitoring (latency tracking and cache cleanup)
+	InitializeAudioMonitoring()
+
 	if !atomic.CompareAndSwapInt32(&outputStreamingRunning, 0, 1) {
 		return ErrAudioAlreadyRunning
 	}
@@ -81,9 +84,9 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 		buffer := make([]byte, GetMaxAudioFrameSize())
 
 		consecutiveErrors := 0
-		maxConsecutiveErrors := Config.MaxConsecutiveErrors
+		maxConsecutiveErrors := GetConfig().MaxConsecutiveErrors
-		errorBackoffDelay := Config.RetryDelay
+		errorBackoffDelay := GetConfig().RetryDelay
-		maxErrorBackoff := Config.MaxRetryDelay
+		maxErrorBackoff := GetConfig().MaxRetryDelay
 
 		for {
 			select {
@@ -120,18 +123,18 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 								Err(initErr).
 								Msg("Failed to reinitialize audio system")
 							// Exponential backoff for reinitialization failures
-							errorBackoffDelay = time.Duration(float64(errorBackoffDelay) * Config.BackoffMultiplier)
+							errorBackoffDelay = time.Duration(float64(errorBackoffDelay) * GetConfig().BackoffMultiplier)
 							if errorBackoffDelay > maxErrorBackoff {
 								errorBackoffDelay = maxErrorBackoff
 							}
 						} else {
 							getOutputStreamingLogger().Info().Msg("Audio system reinitialized successfully")
 							consecutiveErrors = 0
-							errorBackoffDelay = Config.RetryDelay // Reset backoff
+							errorBackoffDelay = GetConfig().RetryDelay // Reset backoff
 						}
 					} else {
 						// Brief delay for transient errors
-						time.Sleep(Config.ShortSleepDuration)
+						time.Sleep(GetConfig().ShortSleepDuration)
 					}
 					continue
 				}
@@ -139,7 +142,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 				// Success - reset error counters
 				if consecutiveErrors > 0 {
 					consecutiveErrors = 0
-					errorBackoffDelay = Config.RetryDelay
+					errorBackoffDelay = GetConfig().RetryDelay
 				}
 
 				if n > 0 {
@@ -161,7 +164,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 					RecordFrameReceived(n)
 				}
 				// Small delay to prevent busy waiting
-				time.Sleep(Config.ShortSleepDuration)
+				time.Sleep(GetConfig().ShortSleepDuration)
 			}
 		}
 	}()
@@ -182,6 +185,6 @@ func StopAudioOutputStreaming() {
 
 	// Wait for streaming to stop
 	for atomic.LoadInt32(&outputStreamingRunning) == 1 {
-		time.Sleep(Config.ShortSleepDuration)
+		time.Sleep(GetConfig().ShortSleepDuration)
 	}
 }

internal/audio/output_supervisor.go

@@ -19,19 +19,19 @@ const (
 
 // Restart configuration is now retrieved from centralized config
 func getMaxRestartAttempts() int {
-	return Config.MaxRestartAttempts
+	return GetConfig().MaxRestartAttempts
 }
 
 func getRestartWindow() time.Duration {
-	return Config.RestartWindow
+	return GetConfig().RestartWindow
 }
 
 func getRestartDelay() time.Duration {
-	return Config.RestartDelay
+	return GetConfig().RestartDelay
 }
 
 func getMaxRestartDelay() time.Duration {
-	return Config.MaxRestartDelay
+	return GetConfig().MaxRestartDelay
 }
 
 // AudioOutputSupervisor manages the audio output server subprocess lifecycle
@@ -125,12 +125,6 @@ func (s *AudioOutputSupervisor) Start() error {
 	// Start the supervision loop
 	go s.supervisionLoop()
 
-	// Establish IPC connection to subprocess after a brief delay
-	go func() {
-		time.Sleep(500 * time.Millisecond) // Wait for subprocess to start
-		s.connectClient()
-	}()
-
 	s.logger.Info().Str("component", AudioOutputSupervisorComponent).Msg("component started successfully")
 	return nil
 }
@@ -151,20 +145,11 @@ func (s *AudioOutputSupervisor) Stop() {
 	select {
 	case <-s.processDone:
 		s.logger.Info().Str("component", AudioOutputSupervisorComponent).Msg("component stopped gracefully")
-	case <-time.After(Config.OutputSupervisorTimeout):
+	case <-time.After(GetConfig().OutputSupervisorTimeout):
 		s.logger.Warn().Str("component", AudioOutputSupervisorComponent).Msg("component did not stop gracefully, forcing termination")
 		s.forceKillProcess("audio output server")
 	}
 
-	// Ensure socket file cleanup even if subprocess didn't clean up properly
-	// This prevents "address already in use" errors on restart
-	outputSocketPath := getOutputSocketPath()
-	if err := os.Remove(outputSocketPath); err != nil && !os.IsNotExist(err) {
-		s.logger.Warn().Err(err).Str("socket_path", outputSocketPath).Msg("failed to remove output socket file during supervisor stop")
-	} else if err == nil {
-		s.logger.Debug().Str("socket_path", outputSocketPath).Msg("cleaned up output socket file")
-	}
-
 	s.logger.Info().Str("component", AudioOutputSupervisorComponent).Msg("component stopped")
 }
 
@@ -173,7 +158,7 @@ func (s *AudioOutputSupervisor) supervisionLoop() {
 	// Configure supervision parameters
 	config := SupervisionConfig{
 		ProcessType:        "audio output server",
-		Timeout:            Config.OutputSupervisorTimeout,
+		Timeout:            GetConfig().OutputSupervisorTimeout,
 		EnableRestart:      true,
 		MaxRestartAttempts: getMaxRestartAttempts(),
 		RestartWindow:      getRestartWindow(),
@@ -228,6 +213,7 @@ func (s *AudioOutputSupervisor) startProcess() error {
 	s.logger.Info().Int("pid", s.processPID).Strs("args", args).Strs("opus_env", s.opusEnv).Msg("audio server process started")
 
 	// Add process to monitoring
+	s.processMonitor.AddProcess(s.processPID, "audio-output-server")
 
 	if s.onProcessStart != nil {
 		s.onProcessStart(s.processPID)
@@ -289,43 +275,3 @@ func (s *AudioOutputSupervisor) calculateRestartDelay() time.Duration {
 
 	return delay
 }
-
-// client holds the IPC client for communicating with the subprocess
-var outputClient *AudioOutputClient
-
-// IsConnected returns whether the supervisor has an active connection to the subprocess
-func (s *AudioOutputSupervisor) IsConnected() bool {
-	return outputClient != nil && outputClient.IsConnected()
-}
-
-// GetClient returns the IPC client for the subprocess
-func (s *AudioOutputSupervisor) GetClient() *AudioOutputClient {
-	return outputClient
-}
-
-// connectClient establishes connection to the audio output subprocess
-func (s *AudioOutputSupervisor) connectClient() {
-	if outputClient == nil {
-		outputClient = NewAudioOutputClient()
-	}
-
-	// Try to connect to the subprocess
-	if err := outputClient.Connect(); err != nil {
-		s.logger.Warn().Err(err).Msg("Failed to connect to audio output subprocess")
-	} else {
-		s.logger.Info().Msg("Connected to audio output subprocess")
-	}
-}
-
-// SendOpusConfig sends Opus configuration to the audio output subprocess
-func (s *AudioOutputSupervisor) SendOpusConfig(config OutputIPCOpusConfig) error {
-	if outputClient == nil {
-		return fmt.Errorf("client not initialized")
-	}
-
-	if !outputClient.IsConnected() {
-		return fmt.Errorf("client not connected")
-	}
-
-	return outputClient.SendOpusConfig(config)
-}

internal/audio/quality_presets.go

@@ -39,7 +39,7 @@ var (
 
 // MaxAudioFrameSize is now retrieved from centralized config
 func GetMaxAudioFrameSize() int {
-	return Config.MaxAudioFrameSize
+	return GetConfig().MaxAudioFrameSize
 }
 
 // AudioQuality represents different audio quality presets
@@ -74,17 +74,17 @@ type AudioMetrics struct {
 var (
 	currentConfig = AudioConfig{
 		Quality:    AudioQualityMedium,
-		Bitrate:    Config.AudioQualityMediumOutputBitrate,
+		Bitrate:    GetConfig().AudioQualityMediumOutputBitrate,
-		SampleRate: Config.SampleRate,
+		SampleRate: GetConfig().SampleRate,
-		Channels:   Config.Channels,
+		Channels:   GetConfig().Channels,
-		FrameSize:  Config.AudioQualityMediumFrameSize,
+		FrameSize:  GetConfig().AudioQualityMediumFrameSize,
 	}
 	currentMicrophoneConfig = AudioConfig{
 		Quality:    AudioQualityMedium,
-		Bitrate:    Config.AudioQualityMediumInputBitrate,
+		Bitrate:    GetConfig().AudioQualityMediumInputBitrate,
-		SampleRate: Config.SampleRate,
+		SampleRate: GetConfig().SampleRate,
 		Channels:   1,
-		FrameSize:  Config.AudioQualityMediumFrameSize,
+		FrameSize:  GetConfig().AudioQualityMediumFrameSize,
 	}
 	metrics AudioMetrics
 )
@@ -96,24 +96,24 @@ var qualityPresets = map[AudioQuality]struct {
 	frameSize                   time.Duration
 }{
 	AudioQualityLow: {
-		outputBitrate: Config.AudioQualityLowOutputBitrate, inputBitrate: Config.AudioQualityLowInputBitrate,
+		outputBitrate: GetConfig().AudioQualityLowOutputBitrate, inputBitrate: GetConfig().AudioQualityLowInputBitrate,
-		sampleRate: Config.AudioQualityLowSampleRate, channels: Config.AudioQualityLowChannels,
+		sampleRate: GetConfig().AudioQualityLowSampleRate, channels: GetConfig().AudioQualityLowChannels,
-		frameSize: Config.AudioQualityLowFrameSize,
+		frameSize: GetConfig().AudioQualityLowFrameSize,
 	},
 	AudioQualityMedium: {
-		outputBitrate: Config.AudioQualityMediumOutputBitrate, inputBitrate: Config.AudioQualityMediumInputBitrate,
+		outputBitrate: GetConfig().AudioQualityMediumOutputBitrate, inputBitrate: GetConfig().AudioQualityMediumInputBitrate,
-		sampleRate: Config.AudioQualityMediumSampleRate, channels: Config.AudioQualityMediumChannels,
+		sampleRate: GetConfig().AudioQualityMediumSampleRate, channels: GetConfig().AudioQualityMediumChannels,
-		frameSize: Config.AudioQualityMediumFrameSize,
+		frameSize: GetConfig().AudioQualityMediumFrameSize,
 	},
 	AudioQualityHigh: {
-		outputBitrate: Config.AudioQualityHighOutputBitrate, inputBitrate: Config.AudioQualityHighInputBitrate,
+		outputBitrate: GetConfig().AudioQualityHighOutputBitrate, inputBitrate: GetConfig().AudioQualityHighInputBitrate,
-		sampleRate: Config.SampleRate, channels: Config.AudioQualityHighChannels,
+		sampleRate: GetConfig().SampleRate, channels: GetConfig().AudioQualityHighChannels,
-		frameSize: Config.AudioQualityHighFrameSize,
+		frameSize: GetConfig().AudioQualityHighFrameSize,
 	},
 	AudioQualityUltra: {
-		outputBitrate: Config.AudioQualityUltraOutputBitrate, inputBitrate: Config.AudioQualityUltraInputBitrate,
+		outputBitrate: GetConfig().AudioQualityUltraOutputBitrate, inputBitrate: GetConfig().AudioQualityUltraInputBitrate,
-		sampleRate: Config.SampleRate, channels: Config.AudioQualityUltraChannels,
+		sampleRate: GetConfig().SampleRate, channels: GetConfig().AudioQualityUltraChannels,
-		frameSize: Config.AudioQualityUltraFrameSize,
+		frameSize: GetConfig().AudioQualityUltraFrameSize,
 	},
 }
 
@@ -142,7 +142,7 @@ func GetMicrophoneQualityPresets() map[AudioQuality]AudioConfig {
 			Bitrate: preset.inputBitrate,
 			SampleRate: func() int {
 				if quality == AudioQualityLow {
-					return Config.AudioQualityMicLowSampleRate
+					return GetConfig().AudioQualityMicLowSampleRate
 				}
 				return preset.sampleRate
 			}(),
@@ -172,84 +172,58 @@ func SetAudioQuality(quality AudioQuality) {
 		var complexity, vbr, signalType, bandwidth, dtx int
 		switch quality {
 		case AudioQualityLow:
-			complexity = Config.AudioQualityLowOpusComplexity
+			complexity = GetConfig().AudioQualityLowOpusComplexity
-			vbr = Config.AudioQualityLowOpusVBR
+			vbr = GetConfig().AudioQualityLowOpusVBR
-			signalType = Config.AudioQualityLowOpusSignalType
+			signalType = GetConfig().AudioQualityLowOpusSignalType
-			bandwidth = Config.AudioQualityLowOpusBandwidth
+			bandwidth = GetConfig().AudioQualityLowOpusBandwidth
-			dtx = Config.AudioQualityLowOpusDTX
+			dtx = GetConfig().AudioQualityLowOpusDTX
 		case AudioQualityMedium:
-			complexity = Config.AudioQualityMediumOpusComplexity
+			complexity = GetConfig().AudioQualityMediumOpusComplexity
-			vbr = Config.AudioQualityMediumOpusVBR
+			vbr = GetConfig().AudioQualityMediumOpusVBR
-			signalType = Config.AudioQualityMediumOpusSignalType
+			signalType = GetConfig().AudioQualityMediumOpusSignalType
-			bandwidth = Config.AudioQualityMediumOpusBandwidth
+			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
-			dtx = Config.AudioQualityMediumOpusDTX
+			dtx = GetConfig().AudioQualityMediumOpusDTX
 		case AudioQualityHigh:
-			complexity = Config.AudioQualityHighOpusComplexity
+			complexity = GetConfig().AudioQualityHighOpusComplexity
-			vbr = Config.AudioQualityHighOpusVBR
+			vbr = GetConfig().AudioQualityHighOpusVBR
-			signalType = Config.AudioQualityHighOpusSignalType
+			signalType = GetConfig().AudioQualityHighOpusSignalType
-			bandwidth = Config.AudioQualityHighOpusBandwidth
+			bandwidth = GetConfig().AudioQualityHighOpusBandwidth
-			dtx = Config.AudioQualityHighOpusDTX
+			dtx = GetConfig().AudioQualityHighOpusDTX
 		case AudioQualityUltra:
-			complexity = Config.AudioQualityUltraOpusComplexity
+			complexity = GetConfig().AudioQualityUltraOpusComplexity
-			vbr = Config.AudioQualityUltraOpusVBR
+			vbr = GetConfig().AudioQualityUltraOpusVBR
-			signalType = Config.AudioQualityUltraOpusSignalType
+			signalType = GetConfig().AudioQualityUltraOpusSignalType
-			bandwidth = Config.AudioQualityUltraOpusBandwidth
+			bandwidth = GetConfig().AudioQualityUltraOpusBandwidth
-			dtx = Config.AudioQualityUltraOpusDTX
+			dtx = GetConfig().AudioQualityUltraOpusDTX
 		default:
 			// Use medium quality as fallback
-			complexity = Config.AudioQualityMediumOpusComplexity
+			complexity = GetConfig().AudioQualityMediumOpusComplexity
-			vbr = Config.AudioQualityMediumOpusVBR
+			vbr = GetConfig().AudioQualityMediumOpusVBR
-			signalType = Config.AudioQualityMediumOpusSignalType
+			signalType = GetConfig().AudioQualityMediumOpusSignalType
-			bandwidth = Config.AudioQualityMediumOpusBandwidth
+			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
-			dtx = Config.AudioQualityMediumOpusDTX
+			dtx = GetConfig().AudioQualityMediumOpusDTX
 		}
 
-		// Update audio output subprocess configuration dynamically without restart
+		// Restart audio output subprocess with new OPUS configuration
-		logger := logging.GetDefaultLogger().With().Str("component", "audio").Logger()
-		logger.Info().Int("quality", int(quality)).Msg("updating audio output quality settings dynamically")
-
-		// Set new OPUS configuration for future restarts
 		if supervisor := GetAudioOutputSupervisor(); supervisor != nil {
+			logger := logging.GetDefaultLogger().With().Str("component", "audio").Logger()
+			logger.Info().Int("quality", int(quality)).Msg("restarting audio output subprocess with new quality settings")
+
+			// Set new OPUS configuration
 			supervisor.SetOpusConfig(config.Bitrate*1000, complexity, vbr, signalType, bandwidth, dtx)
 
-			// Send dynamic configuration update to running subprocess via IPC
+			// Stop current subprocess
-			if supervisor.IsConnected() {
+			supervisor.Stop()
-				// Convert AudioConfig to OutputIPCOpusConfig with complete Opus parameters
-				opusConfig := OutputIPCOpusConfig{
-					SampleRate: config.SampleRate,
-					Channels:   config.Channels,
-					FrameSize:  int(config.FrameSize.Milliseconds() * int64(config.SampleRate) / 1000), // Convert ms to samples
-					Bitrate:    config.Bitrate * 1000,                                                  // Convert kbps to bps
-					Complexity: complexity,
-					VBR:        vbr,
-					SignalType: signalType,
-					Bandwidth:  bandwidth,
-					DTX:        dtx,
-				}
 
-				logger.Info().Interface("opusConfig", opusConfig).Msg("sending Opus configuration to audio output subprocess")
+			// Start subprocess with new configuration
-				if err := supervisor.SendOpusConfig(opusConfig); err != nil {
+			if err := supervisor.Start(); err != nil {
-					logger.Warn().Err(err).Msg("failed to send dynamic Opus config update via IPC, falling back to subprocess restart")
+				logger.Error().Err(err).Msg("failed to restart audio output subprocess")
-					// Fallback to subprocess restart if IPC update fails
+			}
-					supervisor.Stop()
+		} else {
-					if err := supervisor.Start(); err != nil {
+			// Fallback to dynamic update if supervisor is not available
-						logger.Error().Err(err).Msg("failed to restart audio output subprocess after IPC update failure")
+			vbrConstraint := GetConfig().CGOOpusVBRConstraint
-					}
+			if err := updateOpusEncoderParams(config.Bitrate*1000, complexity, vbr, vbrConstraint, signalType, bandwidth, dtx); err != nil {
-				} else {
+				logging.GetDefaultLogger().Error().Err(err).Msg("Failed to update OPUS encoder parameters")
-					logger.Info().Msg("audio output quality updated dynamically via IPC")
-
-					// Reset audio output stats after config update
-					go func() {
-						time.Sleep(Config.QualityChangeSettleDelay) // Wait for quality change to settle
-						// Reset audio input server stats to clear persistent warnings
-						ResetGlobalAudioInputServerStats()
-						// Attempt recovery if there are still issues
-						time.Sleep(1 * time.Second)
-						RecoverGlobalAudioInputServer()
-					}()
-				}
-			} else {
-				logger.Info().Bool("supervisor_running", supervisor.IsRunning()).Msg("audio output subprocess not connected, configuration will apply on next start")
 			}
 		}
 	}
@@ -260,16 +234,6 @@ func GetAudioConfig() AudioConfig {
 	return currentConfig
 }
 
-// Simplified OPUS parameter lookup table
-var opusParams = map[AudioQuality]struct {
-	complexity, vbr, signalType, bandwidth, dtx int
-}{
-	AudioQualityLow:    {Config.AudioQualityLowOpusComplexity, Config.AudioQualityLowOpusVBR, Config.AudioQualityLowOpusSignalType, Config.AudioQualityLowOpusBandwidth, Config.AudioQualityLowOpusDTX},
-	AudioQualityMedium: {Config.AudioQualityMediumOpusComplexity, Config.AudioQualityMediumOpusVBR, Config.AudioQualityMediumOpusSignalType, Config.AudioQualityMediumOpusBandwidth, Config.AudioQualityMediumOpusDTX},
-	AudioQualityHigh:   {Config.AudioQualityHighOpusComplexity, Config.AudioQualityHighOpusVBR, Config.AudioQualityHighOpusSignalType, Config.AudioQualityHighOpusBandwidth, Config.AudioQualityHighOpusDTX},
-	AudioQualityUltra:  {Config.AudioQualityUltraOpusComplexity, Config.AudioQualityUltraOpusVBR, Config.AudioQualityUltraOpusSignalType, Config.AudioQualityUltraOpusBandwidth, Config.AudioQualityUltraOpusDTX},
-}
-
 // SetMicrophoneQuality updates the current microphone quality configuration
 func SetMicrophoneQuality(quality AudioQuality) {
 	// Validate audio quality parameter
@@ -284,32 +248,51 @@ func SetMicrophoneQuality(quality AudioQuality) {
 	if config, exists := presets[quality]; exists {
 		currentMicrophoneConfig = config
 
-		// Get OPUS parameters using lookup table
+		// Get OPUS parameters for the selected quality
-		params, exists := opusParams[quality]
+		var complexity, vbr, signalType, bandwidth, dtx int
-		if !exists {
+		switch quality {
-			// Fallback to medium quality
+		case AudioQualityLow:
-			params = opusParams[AudioQualityMedium]
+			complexity = GetConfig().AudioQualityLowOpusComplexity
+			vbr = GetConfig().AudioQualityLowOpusVBR
+			signalType = GetConfig().AudioQualityLowOpusSignalType
+			bandwidth = GetConfig().AudioQualityLowOpusBandwidth
+			dtx = GetConfig().AudioQualityLowOpusDTX
+		case AudioQualityMedium:
+			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			vbr = GetConfig().AudioQualityMediumOpusVBR
+			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			dtx = GetConfig().AudioQualityMediumOpusDTX
+		case AudioQualityHigh:
+			complexity = GetConfig().AudioQualityHighOpusComplexity
+			vbr = GetConfig().AudioQualityHighOpusVBR
+			signalType = GetConfig().AudioQualityHighOpusSignalType
+			bandwidth = GetConfig().AudioQualityHighOpusBandwidth
+			dtx = GetConfig().AudioQualityHighOpusDTX
+		case AudioQualityUltra:
+			complexity = GetConfig().AudioQualityUltraOpusComplexity
+			vbr = GetConfig().AudioQualityUltraOpusVBR
+			signalType = GetConfig().AudioQualityUltraOpusSignalType
+			bandwidth = GetConfig().AudioQualityUltraOpusBandwidth
+			dtx = GetConfig().AudioQualityUltraOpusDTX
+		default:
+			// Use medium quality as fallback
+			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			vbr = GetConfig().AudioQualityMediumOpusVBR
+			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			dtx = GetConfig().AudioQualityMediumOpusDTX
 		}
 
 		// Update audio input subprocess configuration dynamically without restart
-		logger := logging.GetDefaultLogger().With().Str("component", "audio").Logger()
-
-		// Set new OPUS configuration for future restarts
 		if supervisor := GetAudioInputSupervisor(); supervisor != nil {
-			supervisor.SetOpusConfig(config.Bitrate*1000, params.complexity, params.vbr, params.signalType, params.bandwidth, params.dtx)
+			logger := logging.GetDefaultLogger().With().Str("component", "audio").Logger()
+			logger.Info().Int("quality", int(quality)).Msg("updating audio input subprocess quality settings dynamically")
 
-			// Check if microphone is active but IPC control is broken
+			// Set new OPUS configuration for future restarts
-			inputManager := getAudioInputManager()
+			supervisor.SetOpusConfig(config.Bitrate*1000, complexity, vbr, signalType, bandwidth, dtx)
-			if inputManager.IsRunning() && !supervisor.IsConnected() {
-				// Reconnect the IPC control channel
-				supervisor.Stop()
-				time.Sleep(50 * time.Millisecond)
-				if err := supervisor.Start(); err != nil {
-					logger.Debug().Err(err).Msg("failed to reconnect IPC control channel")
-				}
-			}
 
-			// Send dynamic configuration update to running subprocess via IPC
+			// Send dynamic configuration update to running subprocess
 			if supervisor.IsConnected() {
 				// Convert AudioConfig to InputIPCOpusConfig with complete Opus parameters
 				opusConfig := InputIPCOpusConfig{
@@ -317,32 +300,23 @@ func SetMicrophoneQuality(quality AudioQuality) {
 					Channels:   config.Channels,
 					FrameSize:  int(config.FrameSize.Milliseconds() * int64(config.SampleRate) / 1000), // Convert ms to samples
 					Bitrate:    config.Bitrate * 1000,                                                  // Convert kbps to bps
-					Complexity: params.complexity,
+					Complexity: complexity,
-					VBR:        params.vbr,
+					VBR:        vbr,
-					SignalType: params.signalType,
+					SignalType: signalType,
-					Bandwidth:  params.bandwidth,
+					Bandwidth:  bandwidth,
-					DTX:        params.dtx,
+					DTX:        dtx,
 				}
 
+				logger.Info().Interface("opusConfig", opusConfig).Msg("sending Opus configuration to audio input subprocess")
 				if err := supervisor.SendOpusConfig(opusConfig); err != nil {
-					logger.Debug().Err(err).Msg("failed to send dynamic Opus config update via IPC")
+					logger.Warn().Err(err).Msg("failed to send dynamic Opus config update, subprocess may need restart")
-					// Fallback to subprocess restart if IPC update fails
+					// Fallback to restart if dynamic update fails
 					supervisor.Stop()
 					if err := supervisor.Start(); err != nil {
-						logger.Error().Err(err).Msg("failed to restart audio input subprocess after IPC update failure")
+						logger.Error().Err(err).Msg("failed to restart audio input subprocess after config update failure")
 					}
 				} else {
-					logger.Info().Msg("audio input quality updated dynamically via IPC")
+					logger.Info().Msg("audio input quality updated dynamically with complete Opus configuration")
-
-					// Reset audio input stats after config update
-					go func() {
-						time.Sleep(Config.QualityChangeSettleDelay) // Wait for quality change to settle
-						// Reset audio input server stats to clear persistent warnings
-						ResetGlobalAudioInputServerStats()
-						// Attempt recovery if microphone is still having issues
-						time.Sleep(1 * time.Second)
-						RecoverGlobalAudioInputServer()
-					}()
 				}
 			} else {
 				logger.Info().Bool("supervisor_running", supervisor.IsRunning()).Msg("audio input subprocess not connected, configuration will apply on next start")

internal/audio/relay_api.go

@@ -2,9 +2,7 @@ package audio
 
 import (
 	"errors"
-	"fmt"
 	"sync"
-	"time"
 )
 
 // Global relay instance for the main process
@@ -30,34 +28,13 @@ func StartAudioRelay(audioTrack AudioTrackWriter) error {
 	// Get current audio config
 	config := GetAudioConfig()
 
-	// Retry starting the relay with exponential backoff
+	// Start the relay (audioTrack can be nil initially)
-	// This handles cases where the subprocess hasn't created its socket yet
+	if err := relay.Start(audioTrack, config); err != nil {
-	maxAttempts := 5
+		return err
-	baseDelay := 200 * time.Millisecond
-	maxDelay := 2 * time.Second
-
-	var lastErr error
-	for i := 0; i < maxAttempts; i++ {
-		if err := relay.Start(audioTrack, config); err != nil {
-			lastErr = err
-			if i < maxAttempts-1 {
-				// Calculate exponential backoff delay
-				delay := time.Duration(float64(baseDelay) * (1.5 * float64(i+1)))
-				if delay > maxDelay {
-					delay = maxDelay
-				}
-				time.Sleep(delay)
-				continue
-			}
-			return fmt.Errorf("failed to start audio relay after %d attempts: %w", maxAttempts, lastErr)
-		}
-
-		// Success
-		globalRelay = relay
-		return nil
 	}
 
-	return fmt.Errorf("failed to start audio relay after %d attempts: %w", maxAttempts, lastErr)
+	globalRelay = relay
+	return nil
 }
 
 // StopAudioRelay stops the audio relay system
@@ -112,93 +89,37 @@ func IsAudioRelayRunning() bool {
 }
 
 // UpdateAudioRelayTrack updates the WebRTC audio track for the relay
-// This function is refactored to prevent mutex deadlocks during quality changes
 func UpdateAudioRelayTrack(audioTrack AudioTrackWriter) error {
-	var needsCallback bool
-	var callbackFunc TrackReplacementCallback
-
-	// Critical section: minimize time holding the mutex
 	relayMutex.Lock()
+	defer relayMutex.Unlock()
+
 	if globalRelay == nil {
 		// No relay running, start one with the provided track
 		relay := NewAudioRelay()
 		config := GetAudioConfig()
 		if err := relay.Start(audioTrack, config); err != nil {
-			relayMutex.Unlock()
 			return err
 		}
 		globalRelay = relay
-	} else {
+		return nil
-		// Update the track in the existing relay
-		globalRelay.UpdateTrack(audioTrack)
-	}
-
-	// Capture callback state while holding mutex
-	needsCallback = trackReplacementCallback != nil
-	if needsCallback {
-		callbackFunc = trackReplacementCallback
-	}
-	relayMutex.Unlock()
-
-	// Execute callback outside of mutex to prevent deadlock
-	if needsCallback && callbackFunc != nil {
-		// Use goroutine with timeout to prevent blocking
-		done := make(chan error, 1)
-		go func() {
-			done <- callbackFunc(audioTrack)
-		}()
-
-		// Wait for callback with timeout
-		select {
-		case err := <-done:
-			if err != nil {
-				// Log error but don't fail the relay operation
-				// The relay can still work even if WebRTC track replacement fails
-				_ = err // Suppress linter warning
-			}
-		case <-time.After(5 * time.Second):
-			// Timeout: log warning but continue
-			// This prevents indefinite blocking during quality changes
-			_ = fmt.Errorf("track replacement callback timed out")
-		}
 	}
 
+	// Update the track in the existing relay
+	globalRelay.UpdateTrack(audioTrack)
 	return nil
 }
 
 // CurrentSessionCallback is a function type for getting the current session's audio track
 type CurrentSessionCallback func() AudioTrackWriter
 
-// TrackReplacementCallback is a function type for replacing the WebRTC audio track
-type TrackReplacementCallback func(AudioTrackWriter) error
-
 // currentSessionCallback holds the callback function to get the current session's audio track
 var currentSessionCallback CurrentSessionCallback
 
-// trackReplacementCallback holds the callback function to replace the WebRTC audio track
-var trackReplacementCallback TrackReplacementCallback
-
 // SetCurrentSessionCallback sets the callback function to get the current session's audio track
 func SetCurrentSessionCallback(callback CurrentSessionCallback) {
 	currentSessionCallback = callback
 }
 
-// SetTrackReplacementCallback sets the callback function to replace the WebRTC audio track
-func SetTrackReplacementCallback(callback TrackReplacementCallback) {
-	trackReplacementCallback = callback
-}
-
-// UpdateAudioRelayTrackAsync performs async track update to prevent blocking
-// This is used during WebRTC session creation to avoid deadlocks
-func UpdateAudioRelayTrackAsync(audioTrack AudioTrackWriter) {
-	go func() {
-		if err := UpdateAudioRelayTrack(audioTrack); err != nil {
-			// Log error but don't block session creation
-			_ = err // Suppress linter warning
-		}
-	}()
-}
-
 // connectRelayToCurrentSession connects the audio relay to the current WebRTC session's audio track
 // This is used when restarting the relay during unmute operations
 func connectRelayToCurrentSession() error {

internal/audio/socket_buffer.go

@@ -18,8 +18,8 @@ type SocketBufferConfig struct {
 // DefaultSocketBufferConfig returns the default socket buffer configuration
 func DefaultSocketBufferConfig() SocketBufferConfig {
 	return SocketBufferConfig{
-		SendBufferSize: Config.SocketOptimalBuffer,
+		SendBufferSize: GetConfig().SocketOptimalBuffer,
-		RecvBufferSize: Config.SocketOptimalBuffer,
+		RecvBufferSize: GetConfig().SocketOptimalBuffer,
 		Enabled:        true,
 	}
 }
@@ -27,8 +27,8 @@ func DefaultSocketBufferConfig() SocketBufferConfig {
 // HighLoadSocketBufferConfig returns configuration for high-load scenarios
 func HighLoadSocketBufferConfig() SocketBufferConfig {
 	return SocketBufferConfig{
-		SendBufferSize: Config.SocketMaxBuffer,
+		SendBufferSize: GetConfig().SocketMaxBuffer,
-		RecvBufferSize: Config.SocketMaxBuffer,
+		RecvBufferSize: GetConfig().SocketMaxBuffer,
 		Enabled:        true,
 	}
 }
@@ -123,8 +123,8 @@ func ValidateSocketBufferConfig(config SocketBufferConfig) error {
 		return nil
 	}
 
-	minBuffer := Config.SocketMinBuffer
+	minBuffer := GetConfig().SocketMinBuffer
-	maxBuffer := Config.SocketMaxBuffer
+	maxBuffer := GetConfig().SocketMaxBuffer
 
 	if config.SendBufferSize < minBuffer {
 		return fmt.Errorf("send buffer size validation failed: got %d bytes, minimum required %d bytes (configured range: %d-%d)",

internal/audio/util_buffer_pool.go

@@ -4,138 +4,765 @@
 package audio
 
 import (
+	"runtime"
+	"sort"
+	"sync"
 	"sync/atomic"
+	"time"
+	"unsafe"
 )
 
-// AudioBufferPool provides a simple buffer pool for audio processing
+// AudioLatencyInfo holds simplified latency information for cleanup decisions
-type AudioBufferPool struct {
+type AudioLatencyInfo struct {
-	// Atomic counters
+	LatencyMs float64
-	hitCount  int64 // Pool hit counter (atomic)
+	Timestamp time.Time
-	missCount int64 // Pool miss counter (atomic)
-
-	// Pool configuration
-	bufferSize int
-	pool       chan []byte
-	maxSize    int
 }
 
-// NewAudioBufferPool creates a new simple audio buffer pool
+// Global latency tracking
-func NewAudioBufferPool(bufferSize int) *AudioBufferPool {
+var (
-	maxSize := Config.MaxPoolSize
+	currentAudioLatency        = AudioLatencyInfo{}
-	if maxSize <= 0 {
+	currentAudioLatencyLock    sync.RWMutex
-		maxSize = Config.BufferPoolDefaultSize
+	audioMonitoringInitialized int32 // Atomic flag to track initialization
+)
+
+// InitializeAudioMonitoring starts the background goroutines for latency tracking and cache cleanup
+// This is safe to call multiple times as it will only initialize once
+func InitializeAudioMonitoring() {
+	// Use atomic CAS to ensure we only initialize once
+	if atomic.CompareAndSwapInt32(&audioMonitoringInitialized, 0, 1) {
+		// Start the latency recorder
+		startLatencyRecorder()
+
+		// Start the cleanup goroutine
+		startCleanupGoroutine()
+	}
+}
+
+// latencyChannel is used for non-blocking latency recording
+var latencyChannel = make(chan float64, 10)
+
+// startLatencyRecorder starts the latency recorder goroutine
+// This should be called during package initialization
+func startLatencyRecorder() {
+	go latencyRecorderLoop()
+}
+
+// latencyRecorderLoop processes latency recordings in the background
+func latencyRecorderLoop() {
+	for latencyMs := range latencyChannel {
+		currentAudioLatencyLock.Lock()
+		currentAudioLatency = AudioLatencyInfo{
+			LatencyMs: latencyMs,
+			Timestamp: time.Now(),
+		}
+		currentAudioLatencyLock.Unlock()
+	}
+}
+
+// RecordAudioLatency records the current audio processing latency
+// This is called from the audio input manager when latency is measured
+// It is non-blocking to ensure zero overhead in the critical audio path
+func RecordAudioLatency(latencyMs float64) {
+	// Non-blocking send - if channel is full, we drop the update
+	select {
+	case latencyChannel <- latencyMs:
+		// Successfully sent
+	default:
+		// Channel full, drop this update to avoid blocking the audio path
+	}
+}
+
+// GetAudioLatencyMetrics returns the current audio latency information
+// Returns nil if no latency data is available or if it's too old
+func GetAudioLatencyMetrics() *AudioLatencyInfo {
+	currentAudioLatencyLock.RLock()
+	defer currentAudioLatencyLock.RUnlock()
+
+	// Check if we have valid latency data
+	if currentAudioLatency.Timestamp.IsZero() {
+		return nil
 	}
 
-	pool := &AudioBufferPool{
+	// Check if the data is too old (more than 5 seconds)
-		bufferSize: bufferSize,
+	if time.Since(currentAudioLatency.Timestamp) > 5*time.Second {
-		pool:       make(chan []byte, maxSize),
+		return nil
-		maxSize:    maxSize,
 	}
 
-	// Pre-populate the pool
+	return &AudioLatencyInfo{
-	for i := 0; i < maxSize/2; i++ {
+		LatencyMs: currentAudioLatency.LatencyMs,
-		buf := make([]byte, bufferSize)
+		Timestamp: currentAudioLatency.Timestamp,
+	}
+}
+
+// Enhanced lock-free buffer cache for per-goroutine optimization
+type lockFreeBufferCache struct {
+	buffers [8]*[]byte // Increased from 4 to 8 buffers per goroutine cache for better hit rates
+}
+
+const (
+	// Enhanced cache configuration for per-goroutine optimization
+	cacheSize = 8                // Increased from 4 to 8 buffers per goroutine cache for better hit rates
+	cacheTTL  = 10 * time.Second // Increased from 5s to 10s for better cache retention
+	// Additional cache constants for enhanced performance
+	maxCacheEntries      = 256              // Maximum number of goroutine cache entries to prevent memory bloat
+	cacheCleanupInterval = 30 * time.Second // How often to clean up stale cache entries
+	cacheWarmupThreshold = 50               // Number of requests before enabling cache warmup
+	cacheHitRateTarget   = 0.85             // Target cache hit rate for optimization
+)
+
+// TTL tracking for goroutine cache entries
+type cacheEntry struct {
+	cache      *lockFreeBufferCache
+	lastAccess int64 // Unix timestamp of last access
+	gid        int64 // Goroutine ID for better tracking
+}
+
+// Per-goroutine buffer cache using goroutine-local storage
+var goroutineBufferCache = make(map[int64]*lockFreeBufferCache)
+var goroutineCacheMutex sync.RWMutex
+var lastCleanupTime int64        // Unix timestamp of last cleanup
+const maxCacheSize = 500         // Maximum number of goroutine caches (reduced from 1000)
+const cleanupInterval int64 = 30 // Cleanup interval in seconds (30 seconds, reduced from 60)
+const bufferTTL int64 = 60       // Time-to-live for cached buffers in seconds (1 minute, reduced from 2)
+
+// getGoroutineID extracts goroutine ID from runtime stack for cache key
+func getGoroutineID() int64 {
+	b := make([]byte, 64)
+	b = b[:runtime.Stack(b, false)]
+	// Parse "goroutine 123 [running]:" format
+	for i := 10; i < len(b); i++ {
+		if b[i] == ' ' {
+			id := int64(0)
+			for j := 10; j < i; j++ {
+				if b[j] >= '0' && b[j] <= '9' {
+					id = id*10 + int64(b[j]-'0')
+				}
+			}
+			return id
+		}
+	}
+	return 0
+}
+
+// Map of goroutine ID to cache entry with TTL tracking
+var goroutineCacheWithTTL = make(map[int64]*cacheEntry)
+
+// cleanupChannel is used for asynchronous cleanup requests
+var cleanupChannel = make(chan struct{}, 1)
+
+// startCleanupGoroutine starts the cleanup goroutine
+// This should be called during package initialization
+func startCleanupGoroutine() {
+	go cleanupLoop()
+}
+
+// cleanupLoop processes cleanup requests in the background
+func cleanupLoop() {
+	ticker := time.NewTicker(10 * time.Second)
+	defer ticker.Stop()
+
+	for {
 		select {
-		case pool.pool <- buf:
+		case <-cleanupChannel:
-		default:
+			// Received explicit cleanup request
-			break
+			performCleanup(true)
+		case <-ticker.C:
+			// Regular cleanup check
+			performCleanup(false)
+		}
+	}
+}
+
+// requestCleanup signals the cleanup goroutine to perform a cleanup
+// This is non-blocking and can be called from the critical path
+func requestCleanup() {
+	select {
+	case cleanupChannel <- struct{}{}:
+		// Successfully requested cleanup
+	default:
+		// Channel full, cleanup already pending
+	}
+}
+
+// performCleanup does the actual cache cleanup work
+// This runs in a dedicated goroutine, not in the critical path
+func performCleanup(forced bool) {
+	now := time.Now().Unix()
+	lastCleanup := atomic.LoadInt64(&lastCleanupTime)
+
+	// Check if we're in a high-latency situation
+	isHighLatency := false
+	latencyMetrics := GetAudioLatencyMetrics()
+	if latencyMetrics != nil && latencyMetrics.LatencyMs > 10.0 {
+		// Under high latency, be more aggressive with cleanup
+		isHighLatency = true
+	}
+
+	// Only cleanup if enough time has passed (less time if high latency) or if forced
+	interval := cleanupInterval
+	if isHighLatency {
+		interval = cleanupInterval / 2 // More frequent cleanup under high latency
+	}
+
+	if !forced && now-lastCleanup < interval {
+		return
+	}
+
+	// Try to acquire cleanup lock atomically
+	if !atomic.CompareAndSwapInt64(&lastCleanupTime, lastCleanup, now) {
+		return // Another goroutine is already cleaning up
+	}
+
+	// Perform the actual cleanup
+	doCleanupGoroutineCache()
+}
+
+// cleanupGoroutineCache triggers an asynchronous cleanup of the goroutine cache
+// This is safe to call from the critical path as it's non-blocking
+func cleanupGoroutineCache() {
+	// Request asynchronous cleanup
+	requestCleanup()
+}
+
+// The actual cleanup implementation that runs in the background goroutine
+func doCleanupGoroutineCache() {
+	// Get current time for TTL calculations
+	now := time.Now().Unix()
+
+	// Check if we're in a high-latency situation
+	isHighLatency := false
+	latencyMetrics := GetAudioLatencyMetrics()
+	if latencyMetrics != nil && latencyMetrics.LatencyMs > 10.0 {
+		// Under high latency, be more aggressive with cleanup
+		isHighLatency = true
+	}
+
+	goroutineCacheMutex.Lock()
+	defer goroutineCacheMutex.Unlock()
+
+	// Convert old cache format to new TTL-based format if needed
+	if len(goroutineCacheWithTTL) == 0 && len(goroutineBufferCache) > 0 {
+		for gid, cache := range goroutineBufferCache {
+			goroutineCacheWithTTL[gid] = &cacheEntry{
+				cache:      cache,
+				lastAccess: now,
+				gid:        gid,
+			}
+		}
+		// Clear old cache to free memory
+		goroutineBufferCache = make(map[int64]*lockFreeBufferCache)
+	}
+
+	// Enhanced cleanup with size limits and better TTL management
+	entriesToRemove := make([]int64, 0)
+	ttl := bufferTTL
+	if isHighLatency {
+		// Under high latency, use a much shorter TTL
+		ttl = bufferTTL / 4
+	}
+
+	// Remove entries older than enhanced TTL
+	for gid, entry := range goroutineCacheWithTTL {
+		// Both now and entry.lastAccess are int64, so this comparison is safe
+		if now-entry.lastAccess > ttl {
+			entriesToRemove = append(entriesToRemove, gid)
 		}
 	}
 
-	return pool
+	// If we have too many cache entries, remove the oldest ones
+	if len(goroutineCacheWithTTL) > maxCacheEntries {
+		// Sort by last access time and remove oldest entries
+		type cacheEntryWithGID struct {
+			gid        int64
+			lastAccess int64
+		}
+		entries := make([]cacheEntryWithGID, 0, len(goroutineCacheWithTTL))
+		for gid, entry := range goroutineCacheWithTTL {
+			entries = append(entries, cacheEntryWithGID{gid: gid, lastAccess: entry.lastAccess})
+		}
+		// Sort by last access time (oldest first)
+		sort.Slice(entries, func(i, j int) bool {
+			return entries[i].lastAccess < entries[j].lastAccess
+		})
+		// Mark oldest entries for removal
+		excessCount := len(goroutineCacheWithTTL) - maxCacheEntries
+		for i := 0; i < excessCount && i < len(entries); i++ {
+			entriesToRemove = append(entriesToRemove, entries[i].gid)
+		}
+	}
+
+	// If cache is still too large after TTL cleanup, remove oldest entries
+	// Under high latency, use a more aggressive target size
+	targetSize := maxCacheSize
+	targetReduction := maxCacheSize / 2
+
+	if isHighLatency {
+		// Under high latency, target a much smaller cache size
+		targetSize = maxCacheSize / 4
+		targetReduction = maxCacheSize / 8
+	}
+
+	if len(goroutineCacheWithTTL) > targetSize {
+		// Find oldest entries
+		type ageEntry struct {
+			gid        int64
+			lastAccess int64
+		}
+		oldestEntries := make([]ageEntry, 0, len(goroutineCacheWithTTL))
+		for gid, entry := range goroutineCacheWithTTL {
+			oldestEntries = append(oldestEntries, ageEntry{gid, entry.lastAccess})
+		}
+
+		// Sort by lastAccess (oldest first)
+		sort.Slice(oldestEntries, func(i, j int) bool {
+			return oldestEntries[i].lastAccess < oldestEntries[j].lastAccess
+		})
+
+		// Remove oldest entries to get down to target reduction size
+		toRemove := len(goroutineCacheWithTTL) - targetReduction
+		for i := 0; i < toRemove && i < len(oldestEntries); i++ {
+			entriesToRemove = append(entriesToRemove, oldestEntries[i].gid)
+		}
+	}
+
+	// Remove marked entries and return their buffers to the pool
+	for _, gid := range entriesToRemove {
+		if entry, exists := goroutineCacheWithTTL[gid]; exists {
+			// Return buffers to main pool before removing entry
+			for i, buf := range entry.cache.buffers {
+				if buf != nil {
+					// Clear the buffer slot atomically
+					entry.cache.buffers[i] = nil
+				}
+			}
+			delete(goroutineCacheWithTTL, gid)
+		}
+	}
+}
+
+type AudioBufferPool struct {
+	// Atomic fields MUST be first for ARM32 alignment (int64 fields need 8-byte alignment)
+	currentSize int64 // Current pool size (atomic)
+	hitCount    int64 // Pool hit counter (atomic)
+	missCount   int64 // Pool miss counter (atomic)
+
+	// Other fields
+	pool        sync.Pool
+	bufferSize  int
+	maxPoolSize int
+	mutex       sync.RWMutex
+	// Memory optimization fields
+	preallocated []*[]byte // Pre-allocated buffers for immediate use
+	preallocSize int       // Number of pre-allocated buffers
+}
+
+func NewAudioBufferPool(bufferSize int) *AudioBufferPool {
+	// Validate buffer size parameter
+	if err := ValidateBufferSize(bufferSize); err != nil {
+		// Use default value on validation error
+		bufferSize = GetConfig().AudioFramePoolSize
+	}
+
+	// Enhanced preallocation strategy based on buffer size and system capacity
+	var preallocSize int
+	if bufferSize <= GetConfig().AudioFramePoolSize {
+		// For smaller pools, use enhanced preallocation (40% instead of 20%)
+		preallocSize = GetConfig().PreallocPercentage * 2
+	} else {
+		// For larger pools, use standard enhanced preallocation (30% instead of 10%)
+		preallocSize = (GetConfig().PreallocPercentage * 3) / 2
+	}
+
+	// Ensure minimum preallocation for better performance
+	minPrealloc := 50 // Minimum 50 buffers for startup performance
+	if preallocSize < minPrealloc {
+		preallocSize = minPrealloc
+	}
+
+	// Pre-allocate with exact capacity to avoid slice growth
+	preallocated := make([]*[]byte, 0, preallocSize)
+
+	// Pre-allocate buffers with optimized capacity
+	for i := 0; i < preallocSize; i++ {
+		// Use exact buffer size to prevent over-allocation
+		buf := make([]byte, 0, bufferSize)
+		preallocated = append(preallocated, &buf)
+	}
+
+	return &AudioBufferPool{
+		bufferSize:   bufferSize,
+		maxPoolSize:  GetConfig().MaxPoolSize * 2, // Double the max pool size for better buffering
+		preallocated: preallocated,
+		preallocSize: preallocSize,
+		pool: sync.Pool{
+			New: func() interface{} {
+				// Allocate exact size to minimize memory waste
+				buf := make([]byte, 0, bufferSize)
+				return &buf
+			},
+		},
+	}
 }
 
-// Get retrieves a buffer from the pool
 func (p *AudioBufferPool) Get() []byte {
-	select {
+	// Skip cleanup trigger in hotpath - cleanup runs in background
-	case buf := <-p.pool:
+	// cleanupGoroutineCache() - moved to background goroutine
+
+	// Fast path: Try lock-free per-goroutine cache first
+	gid := getGoroutineID()
+	goroutineCacheMutex.RLock()
+	cacheEntry, exists := goroutineCacheWithTTL[gid]
+	goroutineCacheMutex.RUnlock()
+
+	if exists && cacheEntry != nil && cacheEntry.cache != nil {
+		// Try to get buffer from lock-free cache
+		cache := cacheEntry.cache
+		for i := 0; i < len(cache.buffers); i++ {
+			bufPtr := (*unsafe.Pointer)(unsafe.Pointer(&cache.buffers[i]))
+			buf := (*[]byte)(atomic.LoadPointer(bufPtr))
+			if buf != nil && atomic.CompareAndSwapPointer(bufPtr, unsafe.Pointer(buf), nil) {
+				// Direct hit count update to avoid sampling complexity in critical path
+				atomic.AddInt64(&p.hitCount, 1)
+				*buf = (*buf)[:0]
+				return *buf
+			}
+		}
+		// Update access time only after cache miss to reduce overhead
+		cacheEntry.lastAccess = time.Now().Unix()
+	}
+
+	// Fallback: Try pre-allocated pool with mutex
+	p.mutex.Lock()
+	if len(p.preallocated) > 0 {
+		lastIdx := len(p.preallocated) - 1
+		buf := p.preallocated[lastIdx]
+		p.preallocated = p.preallocated[:lastIdx]
+		p.mutex.Unlock()
+		// Direct hit count update to avoid sampling complexity in critical path
 		atomic.AddInt64(&p.hitCount, 1)
-		return buf[:0] // Reset length but keep capacity
+		*buf = (*buf)[:0]
-	default:
+		return *buf
-		atomic.AddInt64(&p.missCount, 1)
-		return make([]byte, 0, p.bufferSize)
 	}
+	p.mutex.Unlock()
+
+	// Try sync.Pool next
+	if poolBuf := p.pool.Get(); poolBuf != nil {
+		buf := poolBuf.(*[]byte)
+		// Direct hit count update to avoid sampling complexity in critical path
+		atomic.AddInt64(&p.hitCount, 1)
+		atomic.AddInt64(&p.currentSize, -1)
+		// Fast capacity check - most buffers should be correct size
+		if cap(*buf) >= p.bufferSize {
+			*buf = (*buf)[:0]
+			return *buf
+		}
+		// Buffer too small, fall through to allocation
+	}
+
+	// Pool miss - allocate new buffer with exact capacity
+	// Direct miss count update to avoid sampling complexity in critical path
+	atomic.AddInt64(&p.missCount, 1)
+	return make([]byte, 0, p.bufferSize)
 }
 
-// Put returns a buffer to the pool
 func (p *AudioBufferPool) Put(buf []byte) {
-	if buf == nil || cap(buf) != p.bufferSize {
+	// Fast validation - reject buffers that are too small or too large
-		return // Invalid buffer
+	bufCap := cap(buf)
+	if bufCap < p.bufferSize || bufCap > p.bufferSize*2 {
+		return // Buffer size mismatch, don't pool it to prevent memory bloat
 	}
 
-	// Reset the buffer
+	// Enhanced buffer clearing - only clear if buffer contains sensitive data
-	buf = buf[:0]
+	// For audio buffers, we can skip clearing for performance unless needed
-
+	// This reduces CPU overhead significantly
-	// Try to return to pool
+	var resetBuf []byte
-	select {
+	if cap(buf) > p.bufferSize {
-	case p.pool <- buf:
+		// If capacity is larger than expected, create a new properly sized buffer
-		// Successfully returned to pool
+		resetBuf = make([]byte, 0, p.bufferSize)
-	default:
+	} else {
-		// Pool is full, discard buffer
+		// Reset length but keep capacity for reuse efficiency
+		resetBuf = buf[:0]
 	}
+
+	// Fast path: Try to put in lock-free per-goroutine cache
+	gid := getGoroutineID()
+	goroutineCacheMutex.RLock()
+	entryWithTTL, exists := goroutineCacheWithTTL[gid]
+	goroutineCacheMutex.RUnlock()
+
+	var cache *lockFreeBufferCache
+	if exists && entryWithTTL != nil {
+		cache = entryWithTTL.cache
+		// Update access time only when we successfully use the cache
+	} else {
+		// Create new cache for this goroutine
+		cache = &lockFreeBufferCache{}
+		now := time.Now().Unix()
+		goroutineCacheMutex.Lock()
+		goroutineCacheWithTTL[gid] = &cacheEntry{
+			cache:      cache,
+			lastAccess: now,
+			gid:        gid,
+		}
+		goroutineCacheMutex.Unlock()
+	}
+
+	if cache != nil {
+		// Try to store in lock-free cache
+		for i := 0; i < len(cache.buffers); i++ {
+			bufPtr := (*unsafe.Pointer)(unsafe.Pointer(&cache.buffers[i]))
+			if atomic.CompareAndSwapPointer(bufPtr, nil, unsafe.Pointer(&resetBuf)) {
+				// Update access time only on successful cache
+				if exists && entryWithTTL != nil {
+					entryWithTTL.lastAccess = time.Now().Unix()
+				}
+				return // Successfully cached
+			}
+		}
+	}
+
+	// Fallback: Try to return to pre-allocated pool for fastest reuse
+	p.mutex.Lock()
+	if len(p.preallocated) < p.preallocSize {
+		p.preallocated = append(p.preallocated, &resetBuf)
+		p.mutex.Unlock()
+		return
+	}
+	p.mutex.Unlock()
+
+	// Check sync.Pool size limit to prevent excessive memory usage
+	if atomic.LoadInt64(&p.currentSize) >= int64(p.maxPoolSize) {
+		return // Pool is full, let GC handle this buffer
+	}
+
+	// Return to sync.Pool and update counter atomically
+	p.pool.Put(&resetBuf)
+	atomic.AddInt64(&p.currentSize, 1)
 }
 
-// GetStats returns pool statistics
+// Enhanced global buffer pools for different audio frame types with improved sizing
-func (p *AudioBufferPool) GetStats() AudioBufferPoolStats {
+var (
+	// Main audio frame pool with enhanced capacity
+	audioFramePool = NewAudioBufferPool(GetConfig().AudioFramePoolSize)
+	// Control message pool with enhanced capacity for better throughput
+	audioControlPool = NewAudioBufferPool(512) // Increased from GetConfig().OutputHeaderSize to 512 for better control message handling
+)
+
+func GetAudioFrameBuffer() []byte {
+	return audioFramePool.Get()
+}
+
+func PutAudioFrameBuffer(buf []byte) {
+	audioFramePool.Put(buf)
+}
+
+func GetAudioControlBuffer() []byte {
+	return audioControlPool.Get()
+}
+
+func PutAudioControlBuffer(buf []byte) {
+	audioControlPool.Put(buf)
+}
+
+// GetPoolStats returns detailed statistics about this buffer pool
+func (p *AudioBufferPool) GetPoolStats() AudioBufferPoolDetailedStats {
+	p.mutex.RLock()
+	preallocatedCount := len(p.preallocated)
+	currentSize := p.currentSize
+	p.mutex.RUnlock()
+
 	hitCount := atomic.LoadInt64(&p.hitCount)
 	missCount := atomic.LoadInt64(&p.missCount)
 	totalRequests := hitCount + missCount
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * Config.BufferPoolHitRateBase
+		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
 	}
 
+	return AudioBufferPoolDetailedStats{
+		BufferSize:        p.bufferSize,
+		MaxPoolSize:       p.maxPoolSize,
+		CurrentPoolSize:   currentSize,
+		PreallocatedCount: int64(preallocatedCount),
+		PreallocatedMax:   int64(p.preallocSize),
+		HitCount:          hitCount,
+		MissCount:         missCount,
+		HitRate:           hitRate,
+	}
+}
+
+// AudioBufferPoolDetailedStats provides detailed pool statistics
+type AudioBufferPoolDetailedStats struct {
+	BufferSize        int
+	MaxPoolSize       int
+	CurrentPoolSize   int64
+	PreallocatedCount int64
+	PreallocatedMax   int64
+	HitCount          int64
+	MissCount         int64
+	HitRate           float64 // Percentage
+	TotalBytes        int64   // Total memory usage in bytes
+	AverageBufferSize float64 // Average size of buffers in the pool
+}
+
+// GetAudioBufferPoolStats returns statistics about the audio buffer pools
+type AudioBufferPoolStats struct {
+	FramePoolSize   int64
+	FramePoolMax    int
+	ControlPoolSize int64
+	ControlPoolMax  int
+	// Enhanced statistics
+	FramePoolHitRate   float64
+	ControlPoolHitRate float64
+	FramePoolDetails   AudioBufferPoolDetailedStats
+	ControlPoolDetails AudioBufferPoolDetailedStats
+}
+
+func GetAudioBufferPoolStats() AudioBufferPoolStats {
+	audioFramePool.mutex.RLock()
+	frameSize := audioFramePool.currentSize
+	frameMax := audioFramePool.maxPoolSize
+	audioFramePool.mutex.RUnlock()
+
+	audioControlPool.mutex.RLock()
+	controlSize := audioControlPool.currentSize
+	controlMax := audioControlPool.maxPoolSize
+	audioControlPool.mutex.RUnlock()
+
+	// Get detailed statistics
+	frameDetails := audioFramePool.GetPoolStats()
+	controlDetails := audioControlPool.GetPoolStats()
+
 	return AudioBufferPoolStats{
-		BufferSize:  p.bufferSize,
+		FramePoolSize:      frameSize,
-		MaxPoolSize: p.maxSize,
+		FramePoolMax:       frameMax,
-		CurrentSize: int64(len(p.pool)),
+		ControlPoolSize:    controlSize,
-		HitCount:    hitCount,
+		ControlPoolMax:     controlMax,
-		MissCount:   missCount,
+		FramePoolHitRate:   frameDetails.HitRate,
-		HitRate:     hitRate,
+		ControlPoolHitRate: controlDetails.HitRate,
+		FramePoolDetails:   frameDetails,
+		ControlPoolDetails: controlDetails,
 	}
 }
 
-// AudioBufferPoolStats represents pool statistics
+// AdaptiveResize dynamically adjusts pool parameters based on performance metrics
-type AudioBufferPoolStats struct {
+func (p *AudioBufferPool) AdaptiveResize() {
-	BufferSize  int
+	hitCount := atomic.LoadInt64(&p.hitCount)
-	MaxPoolSize int
+	missCount := atomic.LoadInt64(&p.missCount)
-	CurrentSize int64
+	totalRequests := hitCount + missCount
-	HitCount    int64
-	MissCount   int64
-	HitRate     float64
-}
 
-// Global buffer pools
+	if totalRequests < 100 {
-var (
+		return // Not enough data for meaningful adaptation
-	audioFramePool   = NewAudioBufferPool(Config.AudioFramePoolSize)
+	}
-	audioControlPool = NewAudioBufferPool(Config.BufferPoolControlSize)
-)
 
-// GetAudioFrameBuffer gets a buffer for audio frames
+	hitRate := float64(hitCount) / float64(totalRequests)
-func GetAudioFrameBuffer() []byte {
+	currentSize := atomic.LoadInt64(&p.currentSize)
-	return audioFramePool.Get()
-}
 
-// PutAudioFrameBuffer returns a buffer to the frame pool
+	// If hit rate is low (< 80%), consider increasing pool size
-func PutAudioFrameBuffer(buf []byte) {
+	if hitRate < 0.8 && currentSize < int64(p.maxPoolSize) {
-	audioFramePool.Put(buf)
+		// Increase preallocation by 25% up to max pool size
-}
+		newPreallocSize := int(float64(len(p.preallocated)) * 1.25)
+		if newPreallocSize > p.maxPoolSize {
+			newPreallocSize = p.maxPoolSize
+		}
 
-// GetAudioControlBuffer gets a buffer for control messages
+		// Preallocate additional buffers
-func GetAudioControlBuffer() []byte {
+		for len(p.preallocated) < newPreallocSize {
-	return audioControlPool.Get()
+			buf := make([]byte, p.bufferSize)
-}
+			p.preallocated = append(p.preallocated, &buf)
+		}
+	}
 
-// PutAudioControlBuffer returns a buffer to the control pool
+	// If hit rate is very high (> 95%) and pool is large, consider shrinking
-func PutAudioControlBuffer(buf []byte) {
+	if hitRate > 0.95 && len(p.preallocated) > p.preallocSize {
-	audioControlPool.Put(buf)
+		// Reduce preallocation by 10% but not below original size
-}
+		newSize := int(float64(len(p.preallocated)) * 0.9)
+		if newSize < p.preallocSize {
+			newSize = p.preallocSize
+		}
 
-// GetAudioBufferPoolStats returns statistics for all pools
+		// Remove excess preallocated buffers
-func GetAudioBufferPoolStats() map[string]AudioBufferPoolStats {
+		if newSize < len(p.preallocated) {
-	return map[string]AudioBufferPoolStats{
+			p.preallocated = p.preallocated[:newSize]
-		"frame_pool":   audioFramePool.GetStats(),
+		}
-		"control_pool": audioControlPool.GetStats(),
+	}
+}
+
+// WarmupCache pre-populates goroutine-local caches for better initial performance
+func (p *AudioBufferPool) WarmupCache() {
+	// Only warmup if we have sufficient request history
+	hitCount := atomic.LoadInt64(&p.hitCount)
+	missCount := atomic.LoadInt64(&p.missCount)
+	totalRequests := hitCount + missCount
+
+	if totalRequests < int64(cacheWarmupThreshold) {
+		return
+	}
+
+	// Get or create cache for current goroutine
+	gid := getGoroutineID()
+	goroutineCacheMutex.RLock()
+	entryWithTTL, exists := goroutineCacheWithTTL[gid]
+	goroutineCacheMutex.RUnlock()
+
+	var cache *lockFreeBufferCache
+	if exists && entryWithTTL != nil {
+		cache = entryWithTTL.cache
+	} else {
+		// Create new cache for this goroutine
+		cache = &lockFreeBufferCache{}
+		now := time.Now().Unix()
+		goroutineCacheMutex.Lock()
+		goroutineCacheWithTTL[gid] = &cacheEntry{
+			cache:      cache,
+			lastAccess: now,
+			gid:        gid,
+		}
+		goroutineCacheMutex.Unlock()
+	}
+
+	if cache != nil {
+		// Fill cache to optimal level based on hit rate
+		hitRate := float64(hitCount) / float64(totalRequests)
+		optimalCacheSize := int(float64(cacheSize) * hitRate)
+		if optimalCacheSize < 2 {
+			optimalCacheSize = 2
+		}
+
+		// Pre-allocate buffers for cache
+		for i := 0; i < optimalCacheSize && i < len(cache.buffers); i++ {
+			if cache.buffers[i] == nil {
+				// Get buffer from main pool
+				buf := p.Get()
+				if len(buf) > 0 {
+					cache.buffers[i] = &buf
+				}
+			}
+		}
+	}
+}
+
+// OptimizeCache performs periodic cache optimization based on usage patterns
+func (p *AudioBufferPool) OptimizeCache() {
+	hitCount := atomic.LoadInt64(&p.hitCount)
+	missCount := atomic.LoadInt64(&p.missCount)
+	totalRequests := hitCount + missCount
+
+	if totalRequests < 100 {
+		return
+	}
+
+	hitRate := float64(hitCount) / float64(totalRequests)
+
+	// If hit rate is below target, trigger cache warmup
+	if hitRate < cacheHitRateTarget {
+		p.WarmupCache()
+	}
+
+	// Reset counters periodically to avoid overflow and get fresh metrics
+	if totalRequests > 10000 {
+		atomic.StoreInt64(&p.hitCount, hitCount/2)
+		atomic.StoreInt64(&p.missCount, missCount/2)
 	}
 }

internal/audio/util_env.go

@@ -21,12 +21,12 @@ func getEnvInt(key string, defaultValue int) int {
 // with fallback to default config values
 func parseOpusConfig() (bitrate, complexity, vbr, signalType, bandwidth, dtx int) {
 	// Read configuration from environment variables with config defaults
-	bitrate = getEnvInt("JETKVM_OPUS_BITRATE", Config.CGOOpusBitrate)
+	bitrate = getEnvInt("JETKVM_OPUS_BITRATE", GetConfig().CGOOpusBitrate)
-	complexity = getEnvInt("JETKVM_OPUS_COMPLEXITY", Config.CGOOpusComplexity)
+	complexity = getEnvInt("JETKVM_OPUS_COMPLEXITY", GetConfig().CGOOpusComplexity)
-	vbr = getEnvInt("JETKVM_OPUS_VBR", Config.CGOOpusVBR)
+	vbr = getEnvInt("JETKVM_OPUS_VBR", GetConfig().CGOOpusVBR)
-	signalType = getEnvInt("JETKVM_OPUS_SIGNAL_TYPE", Config.CGOOpusSignalType)
+	signalType = getEnvInt("JETKVM_OPUS_SIGNAL_TYPE", GetConfig().CGOOpusSignalType)
-	bandwidth = getEnvInt("JETKVM_OPUS_BANDWIDTH", Config.CGOOpusBandwidth)
+	bandwidth = getEnvInt("JETKVM_OPUS_BANDWIDTH", GetConfig().CGOOpusBandwidth)
-	dtx = getEnvInt("JETKVM_OPUS_DTX", Config.CGOOpusDTX)
+	dtx = getEnvInt("JETKVM_OPUS_DTX", GetConfig().CGOOpusDTX)
 
 	return bitrate, complexity, vbr, signalType, bandwidth, dtx
 }
@@ -34,7 +34,7 @@ func parseOpusConfig() (bitrate, complexity, vbr, signalType, bandwidth, dtx int
 // applyOpusConfig applies OPUS configuration to the global config
 // with optional logging for the specified component
 func applyOpusConfig(bitrate, complexity, vbr, signalType, bandwidth, dtx int, component string, enableLogging bool) {
-	config := Config
+	config := GetConfig()
 	config.CGOOpusBitrate = bitrate
 	config.CGOOpusComplexity = complexity
 	config.CGOOpusVBR = vbr

internal/audio/webrtc_relay.go

@@ -5,7 +5,6 @@ import (
 	"fmt"
 	"reflect"
 	"sync"
-	"sync/atomic"
 	"time"
 
 	"github.com/jetkvm/kvm/internal/logging"
@@ -119,7 +118,9 @@ func (r *AudioRelay) IsMuted() bool {
 
 // GetStats returns relay statistics
 func (r *AudioRelay) GetStats() (framesRelayed, framesDropped int64) {
-	return atomic.LoadInt64(&r.framesRelayed), atomic.LoadInt64(&r.framesDropped)
+	r.mutex.RLock()
+	defer r.mutex.RUnlock()
+	return r.framesRelayed, r.framesDropped
 }
 
 // UpdateTrack updates the WebRTC audio track for the relay
@@ -131,43 +132,34 @@ func (r *AudioRelay) UpdateTrack(audioTrack AudioTrackWriter) {
 
 func (r *AudioRelay) relayLoop() {
 	defer r.wg.Done()
+	r.logger.Debug().Msg("Audio relay loop started")
 
-	var maxConsecutiveErrors = Config.MaxConsecutiveErrors
+	var maxConsecutiveErrors = GetConfig().MaxConsecutiveErrors
 	consecutiveErrors := 0
-	backoffDelay := time.Millisecond * 10
-	maxBackoff := time.Second * 5
 
 	for {
 		select {
 		case <-r.ctx.Done():
+			r.logger.Debug().Msg("audio relay loop stopping")
 			return
 		default:
 			frame, err := r.client.ReceiveFrame()
 			if err != nil {
 				consecutiveErrors++
+				r.logger.Error().Err(err).Int("consecutive_errors", consecutiveErrors).Msg("error reading frame from audio output server")
 				r.incrementDropped()
 
-				// Exponential backoff for stability
 				if consecutiveErrors >= maxConsecutiveErrors {
-					// Attempt reconnection
+					r.logger.Error().Int("consecutive_errors", consecutiveErrors).Int("max_errors", maxConsecutiveErrors).Msg("too many consecutive read errors, stopping audio relay")
-					if r.attemptReconnection() {
-						consecutiveErrors = 0
-						backoffDelay = time.Millisecond * 10
-						continue
-					}
 					return
 				}
-
+				time.Sleep(GetConfig().ShortSleepDuration)
-				time.Sleep(backoffDelay)
-				if backoffDelay < maxBackoff {
-					backoffDelay *= 2
-				}
 				continue
 			}
 
 			consecutiveErrors = 0
-			backoffDelay = time.Millisecond * 10
 			if err := r.forwardToWebRTC(frame); err != nil {
+				r.logger.Warn().Err(err).Msg("failed to forward frame to webrtc")
 				r.incrementDropped()
 			} else {
 				r.incrementRelayed()
@@ -226,24 +218,14 @@ func (r *AudioRelay) forwardToWebRTC(frame []byte) error {
 
 // incrementRelayed atomically increments the relayed frames counter
 func (r *AudioRelay) incrementRelayed() {
-	atomic.AddInt64(&r.framesRelayed, 1)
+	r.mutex.Lock()
+	r.framesRelayed++
+	r.mutex.Unlock()
 }
 
 // incrementDropped atomically increments the dropped frames counter
 func (r *AudioRelay) incrementDropped() {
-	atomic.AddInt64(&r.framesDropped, 1)
+	r.mutex.Lock()
-}
+	r.framesDropped++
-
+	r.mutex.Unlock()
-// attemptReconnection tries to reconnect the audio client for stability
-func (r *AudioRelay) attemptReconnection() bool {
-	if r.client == nil {
-		return false
-	}
-
-	// Disconnect and reconnect
-	r.client.Disconnect()
-	time.Sleep(time.Millisecond * 100)
-
-	err := r.client.Connect()
-	return err == nil
 }

internal/audio/websocket_events.go

@@ -224,7 +224,7 @@ func (aeb *AudioEventBroadcaster) sendToSubscriber(subscriber *AudioEventSubscri
 		return false
 	}
 
-	ctx, cancel := context.WithTimeout(subscriber.ctx, time.Duration(Config.EventTimeoutSeconds)*time.Second)
+	ctx, cancel := context.WithTimeout(subscriber.ctx, time.Duration(GetConfig().EventTimeoutSeconds)*time.Second)
 	defer cancel()
 
 	err := wsjson.Write(ctx, subscriber.conn, event)

internal/audio/zero_copy.go

@@ -98,16 +98,16 @@ type ZeroCopyFramePool struct {
 // NewZeroCopyFramePool creates a new zero-copy frame pool
 func NewZeroCopyFramePool(maxFrameSize int) *ZeroCopyFramePool {
 	// Pre-allocate frames for immediate availability
-	preallocSizeBytes := Config.ZeroCopyPreallocSizeBytes
+	preallocSizeBytes := GetConfig().PreallocSize
-	maxPoolSize := Config.MaxPoolSize // Limit total pool size
+	maxPoolSize := GetConfig().MaxPoolSize // Limit total pool size
 
 	// Calculate number of frames based on memory budget, not frame count
 	preallocFrameCount := preallocSizeBytes / maxFrameSize
 	if preallocFrameCount > maxPoolSize {
 		preallocFrameCount = maxPoolSize
 	}
-	if preallocFrameCount < Config.ZeroCopyMinPreallocFrames {
+	if preallocFrameCount < 1 {
-		preallocFrameCount = Config.ZeroCopyMinPreallocFrames
+		preallocFrameCount = 1 // Always preallocate at least one frame
 	}
 
 	preallocated := make([]*ZeroCopyAudioFrame, 0, preallocFrameCount)
@@ -147,7 +147,7 @@ func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
 		// If we've allocated too many frames, force pool reuse
 		frame := p.pool.Get().(*ZeroCopyAudioFrame)
 		frame.mutex.Lock()
-		atomic.StoreInt32(&frame.refCount, 1)
+		frame.refCount = 1
 		frame.length = 0
 		frame.data = frame.data[:0]
 		frame.mutex.Unlock()
@@ -163,12 +163,11 @@ func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
 		p.mutex.Unlock()
 
 		frame.mutex.Lock()
-		atomic.StoreInt32(&frame.refCount, 1)
+		frame.refCount = 1
 		frame.length = 0
 		frame.data = frame.data[:0]
 		frame.mutex.Unlock()
 
-		atomic.AddInt64(&p.hitCount, 1)
 		return frame
 	}
 	p.mutex.Unlock()
@@ -176,7 +175,7 @@ func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
 	// Try sync.Pool next and track allocation
 	frame := p.pool.Get().(*ZeroCopyAudioFrame)
 	frame.mutex.Lock()
-	atomic.StoreInt32(&frame.refCount, 1)
+	frame.refCount = 1
 	frame.length = 0
 	frame.data = frame.data[:0]
 	frame.mutex.Unlock()
@@ -192,34 +191,43 @@ func (p *ZeroCopyFramePool) Put(frame *ZeroCopyAudioFrame) {
 		return
 	}
 
-	// Reset frame state for reuse
 	frame.mutex.Lock()
-	atomic.StoreInt32(&frame.refCount, 0)
+	frame.refCount--
-	frame.length = 0
+	if frame.refCount <= 0 {
-	frame.data = frame.data[:0]
+		frame.refCount = 0
-	frame.mutex.Unlock()
+		frame.length = 0
+		frame.data = frame.data[:0]
+		frame.mutex.Unlock()
 
-	// First try to return to pre-allocated pool for fastest reuse
+		// First try to return to pre-allocated pool for fastest reuse
-	p.mutex.Lock()
+		p.mutex.Lock()
-	if len(p.preallocated) < p.preallocSize {
+		if len(p.preallocated) < p.preallocSize {
-		p.preallocated = append(p.preallocated, frame)
+			p.preallocated = append(p.preallocated, frame)
+			p.mutex.Unlock()
+			return
+		}
 		p.mutex.Unlock()
-		return
-	}
-	p.mutex.Unlock()
 
-	// Check pool size limit to prevent excessive memory usage
+		// Check pool size limit to prevent excessive memory usage
-	p.mutex.RLock()
+		p.mutex.RLock()
-	currentCount := atomic.LoadInt64(&p.counter)
+		currentCount := atomic.LoadInt64(&p.counter)
-	p.mutex.RUnlock()
+		p.mutex.RUnlock()
 
-	if currentCount >= int64(p.maxPoolSize) {
+		if currentCount >= int64(p.maxPoolSize) {
-		return // Pool is full, let GC handle this frame
+			return // Pool is full, let GC handle this frame
+		}
+
+		// Return to sync.Pool
+		p.pool.Put(frame)
+		// Metrics collection removed
+		if false {
+			atomic.AddInt64(&p.counter, 1)
+		}
+	} else {
+		frame.mutex.Unlock()
 	}
 
-	// Return to sync.Pool
+	// Metrics recording removed - granular metrics collector was unused
-	p.pool.Put(frame)
-	atomic.AddInt64(&p.counter, 1)
 }
 
 // Data returns the frame data as a slice (zero-copy view)
@@ -263,28 +271,18 @@ func (f *ZeroCopyAudioFrame) SetDataDirect(data []byte) {
 	f.pooled = false // Direct assignment means we can't pool this frame
 }
 
-// AddRef increments the reference count atomically
+// AddRef increments the reference count for shared access
 func (f *ZeroCopyAudioFrame) AddRef() {
-	atomic.AddInt32(&f.refCount, 1)
+	f.mutex.Lock()
+	f.refCount++
+	f.mutex.Unlock()
 }
 
-// Release decrements the reference count atomically
+// Release decrements the reference count
-// Returns true if this was the final reference
+func (f *ZeroCopyAudioFrame) Release() {
-func (f *ZeroCopyAudioFrame) Release() bool {
+	f.mutex.Lock()
-	newCount := atomic.AddInt32(&f.refCount, -1)
+	f.refCount--
-	if newCount == 0 {
+	f.mutex.Unlock()
-		// Final reference released, return to pool if pooled
-		if f.pooled {
-			globalZeroCopyPool.Put(f)
-		}
-		return true
-	}
-	return false
-}
-
-// RefCount returns the current reference count atomically
-func (f *ZeroCopyAudioFrame) RefCount() int32 {
-	return atomic.LoadInt32(&f.refCount)
 }
 
 // Length returns the current data length
@@ -327,7 +325,7 @@ func (p *ZeroCopyFramePool) GetZeroCopyPoolStats() ZeroCopyFramePoolStats {
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * Config.PercentageMultiplier
+		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
 	}
 
 	return ZeroCopyFramePoolStats{

main.go

@@ -35,6 +35,12 @@ func startAudioSubprocess() error {
 	// Initialize validation cache for optimal performance
 	audio.InitValidationCache()
 
+	// Start adaptive buffer management for optimal performance
+	audio.StartAdaptiveBuffering()
+
+	// Start goroutine monitoring to detect and prevent leaks
+	audio.StartGoroutineMonitoring()
+
 	// Enable batch audio processing to reduce CGO call overhead
 	if err := audio.EnableBatchAudioProcessing(); err != nil {
 		logger.Warn().Err(err).Msg("failed to enable batch audio processing")
@@ -52,7 +58,7 @@ func startAudioSubprocess() error {
 	audio.SetAudioInputSupervisor(audioInputSupervisor)
 
 	// Set default OPUS configuration for audio input supervisor (low quality for single-core RV1106)
-	config := audio.Config
+	config := audio.GetConfig()
 	audioInputSupervisor.SetOpusConfig(
 		config.AudioQualityLowInputBitrate*1000, // Convert kbps to bps
 		config.AudioQualityLowOpusComplexity,
@@ -71,32 +77,19 @@ func startAudioSubprocess() error {
 		func(pid int) {
 			logger.Info().Int("pid", pid).Msg("audio server process started")
 
-			// Wait for audio output server to be fully ready before starting relay
+			// Start audio relay system for main process
-			// This prevents "no client connected" errors during quality changes
+			// If there's an active WebRTC session, use its audio track
-			go func() {
+			var audioTrack *webrtc.TrackLocalStaticSample
-				// Give the audio output server time to initialize and start listening
+			if currentSession != nil && currentSession.AudioTrack != nil {
-				// Increased delay to reduce frame drops during connection establishment
+				audioTrack = currentSession.AudioTrack
-				time.Sleep(1 * time.Second)
+				logger.Info().Msg("restarting audio relay with existing WebRTC audio track")
+			} else {
+				logger.Info().Msg("starting audio relay without WebRTC track (will be updated when session is created)")
+			}
 
-				// Start audio relay system for main process
+			if err := audio.StartAudioRelay(audioTrack); err != nil {
-				// If there's an active WebRTC session, use its audio track
+				logger.Error().Err(err).Msg("failed to start audio relay")
-				var audioTrack *webrtc.TrackLocalStaticSample
+			}
-				if currentSession != nil && currentSession.AudioTrack != nil {
-					audioTrack = currentSession.AudioTrack
-					logger.Info().Msg("restarting audio relay with existing WebRTC audio track")
-				} else {
-					logger.Info().Msg("starting audio relay without WebRTC track (will be updated when session is created)")
-				}
-
-				if err := audio.StartAudioRelay(audioTrack); err != nil {
-					logger.Error().Err(err).Msg("failed to start audio relay")
-					// Retry once after additional delay if initial attempt fails
-					time.Sleep(1 * time.Second)
-					if err := audio.StartAudioRelay(audioTrack); err != nil {
-						logger.Error().Err(err).Msg("failed to start audio relay after retry")
-					}
-				}
-			}()
 		},
 		// onProcessExit
 		func(pid int, exitCode int, crashed bool) {
@@ -108,7 +101,10 @@ func startAudioSubprocess() error {
 
 			// Stop audio relay when process exits
 			audio.StopAudioRelay()
-
+			// Stop adaptive buffering
+			audio.StopAdaptiveBuffering()
+			// Stop goroutine monitoring
+			audio.StopGoroutineMonitoring()
 			// Disable batch audio processing
 			audio.DisableBatchAudioProcessing()
 		},

ui/src/components/CustomTooltip.tsx

@@ -1,25 +1,25 @@
 import Card from "@components/Card";
 
 export interface CustomTooltipProps {
-  payload: { payload: { date: number; metric: number }; unit: string }[];
+  payload: { payload: { date: number; stat: number }; unit: string }[];
 }
 
 export default function CustomTooltip({ payload }: CustomTooltipProps) {
   if (payload?.length) {
     const toolTipData = payload[0];
-    const { date, metric } = toolTipData.payload;
+    const { date, stat } = toolTipData.payload;
 
     return (
       <Card>
-        <div className="px-2 py-1.5 text-black dark:text-white">
+        <div className="p-2 text-black dark:text-white">
-          <div className="text-[13px] font-semibold">
+          <div className="font-semibold">
             {new Date(date * 1000).toLocaleTimeString()}
           </div>
           <div className="space-y-1">
             <div className="flex items-center gap-x-1">
               <div className="h-[2px] w-2 bg-blue-700" />
-              <span className="text-[13px]">
+              <span >
-                {metric} {toolTipData?.unit}
+                {stat} {toolTipData?.unit}
               </span>
             </div>
           </div>

ui/src/components/Header.tsx

@@ -103,7 +103,7 @@ export default function DashboardNavbar({
                     <hr className="h-[20px] w-px self-center border-none bg-slate-800/20 dark:bg-slate-300/20" />
                     <div className="relative inline-block text-left">
                       <Menu>
-                        <MenuButton as="div" className="h-full">
+                        <MenuButton className="h-full">
                           <Button className="flex h-full items-center gap-x-3 rounded-md border border-slate-800/20 bg-white px-2 py-1.5 dark:border-slate-600 dark:bg-slate-800 dark:text-white">
                             {picture ? (
                               <img

ui/src/components/KvmCard.tsx

@@ -100,12 +100,15 @@ export default function KvmCard({
             )}
           </div>
           <Menu as="div" className="relative inline-block text-left">
-            <MenuButton
+            <div>
-              as={Button}
+              <MenuButton
-              theme="light"
+                as={Button}
-              TrailingIcon={LuEllipsisVertical}
+                theme="light"
-              size="MD"
+                TrailingIcon={LuEllipsisVertical}
-            ></MenuButton>
+                size="MD"
+              ></MenuButton>
+            </div>
+
             <MenuItems
               transition
               className="data-closed:scale-95 data-closed:transform data-closed:opacity-0 data-enter:duration-100 data-leave:duration-75 data-enter:ease-out data-leave:ease-in"

ui/src/components/Metric.tsx

@@ -1,180 +0,0 @@
-/* eslint-disable react-refresh/only-export-components */
-import { ComponentProps } from "react";
-import { cva, cx } from "cva";
-
-import { someIterable } from "../utils";
-
-import { GridCard } from "./Card";
-import MetricsChart from "./MetricsChart";
-
-interface ChartPoint {
-  date: number;
-  metric: number | null;
-}
-
-interface MetricProps<T, K extends keyof T> {
-  title: string;
-  description: string;
-  stream?: Map<number, T>;
-  metric?: K;
-  data?: ChartPoint[];
-  gate?: Map<number, unknown>;
-  supported?: boolean;
-  map?: (p: { date: number; metric: number | null }) => ChartPoint;
-  domain?: [number, number];
-  unit: string;
-  heightClassName?: string;
-  referenceValue?: number;
-  badge?: ComponentProps<typeof MetricHeader>["badge"];
-  badgeTheme?: ComponentProps<typeof MetricHeader>["badgeTheme"];
-}
-
-/**
- * Creates a chart array from a metrics map and a metric name.
- *
- * @param metrics - Expected to be ordered from oldest to newest.
- * @param metricName - Name of the metric to create a chart array for.
- */
-export function createChartArray<T, K extends keyof T>(
-  metrics: Map<number, T>,
-  metricName: K,
-) {
-  const result: { date: number; metric: number | null }[] = [];
-  const iter = metrics.entries();
-  let next = iter.next() as IteratorResult<[number, T]>;
-  const nowSeconds = Math.floor(Date.now() / 1000);
-
-  // We want 120 data points, in the chart.
-  const firstDate = Math.min(next.value?.[0] ?? nowSeconds, nowSeconds - 120);
-
-  for (let t = firstDate; t < nowSeconds; t++) {
-    while (!next.done && next.value[0] < t) next = iter.next();
-    const has = !next.done && next.value[0] === t;
-
-    let metric = null;
-    if (has) metric = next.value[1][metricName] as number;
-    result.push({ date: t, metric });
-
-    if (has) next = iter.next();
-  }
-
-  return result;
-}
-
-function computeReferenceValue(points: ChartPoint[]): number | undefined {
-  const values = points
-    .filter(p => p.metric != null && Number.isFinite(p.metric))
-    .map(p => Number(p.metric));
-
-  if (values.length === 0) return undefined;
-
-  const sum = values.reduce((acc, v) => acc + v, 0);
-  const mean = sum / values.length;
-  return Math.round(mean);
-}
-
-const theme = {
-  light:
-    "bg-white text-black border border-slate-800/20 dark:border dark:border-slate-700 dark:bg-slate-800 dark:text-slate-300",
-  danger: "bg-red-500 dark:border-red-700 dark:bg-red-800 dark:text-red-50",
-  primary: "bg-blue-500 dark:border-blue-700 dark:bg-blue-800 dark:text-blue-50",
-};
-
-interface SettingsItemProps {
-  readonly title: string;
-  readonly description: string | React.ReactNode;
-  readonly badge?: string;
-  readonly className?: string;
-  readonly children?: React.ReactNode;
-  readonly badgeTheme?: keyof typeof theme;
-}
-
-export function MetricHeader(props: SettingsItemProps) {
-  const { title, description, badge } = props;
-  const badgeVariants = cva({ variants: { theme: theme } });
-
-  return (
-    <div className="space-y-0.5">
-      <div className="flex items-center gap-x-2">
-        <div className="flex w-full items-center justify-between text-base font-semibold text-black dark:text-white">
-          {title}
-          {badge && (
-            <span
-              className={cx(
-                "ml-2 rounded-sm px-2 py-1 font-mono text-[10px] leading-none font-medium",
-                badgeVariants({ theme: props.badgeTheme ?? "light" }),
-              )}
-            >
-              {badge}
-            </span>
-          )}
-        </div>
-      </div>
-      <div className="text-sm text-slate-700 dark:text-slate-300">{description}</div>
-    </div>
-  );
-}
-
-export function Metric<T, K extends keyof T>({
-  title,
-  description,
-  stream,
-  metric,
-  data,
-  gate,
-  supported,
-  map,
-  domain = [0, 600],
-  unit = "",
-  heightClassName = "h-[127px]",
-  badge,
-  badgeTheme,
-}: MetricProps<T, K>) {
-  const ready = gate ? gate.size > 0 : stream ? stream.size > 0 : true;
-  const supportedFinal =
-    supported ??
-    (stream && metric ? someIterable(stream, ([, s]) => s[metric] !== undefined) : true);
-
-  // Either we let the consumer provide their own chartArray, or we create one from the stream and metric.
-  const raw = data ?? ((stream && metric && createChartArray(stream, metric)) || []);
-
-  // If the consumer provides a map function, we apply it to the raw data.
-  const dataFinal: ChartPoint[] = map ? raw.map(map) : raw;
-
-  // Compute the average value of the metric.
-  const referenceValue = computeReferenceValue(dataFinal);
-
-  return (
-    <div className="space-y-2">
-      <MetricHeader
-        title={title}
-        description={description}
-        badge={badge}
-        badgeTheme={badgeTheme}
-      />
-
-      <GridCard>
-        <div
-          className={`flex ${heightClassName} w-full items-center justify-center text-sm text-slate-500`}
-        >
-          {!ready ? (
-            <div className="flex flex-col items-center space-y-1">
-              <p className="text-slate-700">Waiting for data...</p>
-            </div>
-          ) : supportedFinal ? (
-            <MetricsChart
-              data={dataFinal}
-              domain={domain}
-              unit={unit}
-              referenceValue={referenceValue}
-            />
-          ) : (
-            <div className="flex flex-col items-center space-y-1">
-              <p className="text-black">Metric not supported</p>
-            </div>
-          )}
-        </div>
-      </GridCard>
-    </div>
-  );
-}

ui/src/components/StatChart.tsx

@@ -12,13 +12,13 @@ import {
 
 import CustomTooltip, { CustomTooltipProps } from "@components/CustomTooltip";
 
-export default function MetricsChart({
+export default function StatChart({
   data,
   domain,
   unit,
   referenceValue,
 }: {
-  data: { date: number; metric: number | null | undefined }[];
+  data: { date: number; stat: number | null | undefined }[];
   domain?: [string | number, string | number];
   unit?: string;
   referenceValue?: number;
@@ -33,7 +33,7 @@ export default function MetricsChart({
           strokeLinecap="butt"
           stroke="rgba(30, 41, 59, 0.1)"
         />
-        {referenceValue !== undefined && (
+        {referenceValue && (
           <ReferenceLine
             y={referenceValue}
             strokeDasharray="3 3"
@@ -64,7 +64,7 @@ export default function MetricsChart({
             .map(x => x.date)}
         />
         <YAxis
-          dataKey="metric"
+          dataKey="stat"
           axisLine={false}
           orientation="right"
           tick={{
@@ -73,7 +73,6 @@ export default function MetricsChart({
             fill: "rgba(107, 114, 128, 1)",
           }}
           padding={{ top: 0, bottom: 0 }}
-          allowDecimals
           tickLine={false}
           unit={unit}
           domain={domain || ["auto", "auto"]}
@@ -88,7 +87,7 @@ export default function MetricsChart({
         <Line
           type="monotone"
           isAnimationActive={false}
-          dataKey="metric"
+          dataKey="stat"
           stroke="rgb(29 78 216)"
           strokeLinecap="round"
           strokeWidth={2}

ui/src/components/WebRTCVideo.tsx

@@ -345,13 +345,8 @@ export default function WebRTCVideo({ microphone }: WebRTCVideoProps) {
 
       peerConnection.addEventListener(
         "track",
-        (_e: RTCTrackEvent) => {
+        (e: RTCTrackEvent) => {
-          // The combined MediaStream is now managed in the main component
+          addStreamToVideoElm(e.streams[0]);
-          // We'll use the mediaStream from the store instead of individual track streams
-          const { mediaStream } = useRTCStore.getState();
-          if (mediaStream) {
-            addStreamToVideoElm(mediaStream);
-          }
         },
         { signal },
       );

ui/src/components/sidebar/connectionStats.tsx

@@ -1,40 +1,74 @@
 import { useInterval } from "usehooks-ts";
 
 import SidebarHeader from "@/components/SidebarHeader";
+import { GridCard } from "@/components/Card";
 import { useRTCStore, useUiStore } from "@/hooks/stores";
-import { someIterable } from "@/utils";
+import StatChart from "@/components/StatChart";
 
-import { createChartArray, Metric } from "../Metric";
+function createChartArray<T, K extends keyof T>(
-import { SettingsSectionHeader } from "../SettingsSectionHeader";
+  stream: Map<number, T>,
+  metric: K,
+): { date: number; stat: T[K] | null }[] {
+  const stat = Array.from(stream).map(([key, stats]) => {
+    return { date: key, stat: stats[metric] };
+  });
+
+  // Sort the dates to ensure they are in chronological order
+  const sortedStat = stat.map(x => x.date).sort((a, b) => a - b);
+
+  // Determine the earliest statistic date
+  const earliestStat = sortedStat[0];
+
+  // Current time in seconds since the Unix epoch
+  const now = Math.floor(Date.now() / 1000);
+
+  // Determine the starting point for the chart data
+  const firstChartDate = earliestStat ? Math.min(earliestStat, now - 120) : now - 120;
+
+  // Generate the chart array for the range between 'firstChartDate' and 'now'
+  return Array.from({ length: now - firstChartDate }, (_, i) => {
+    const currentDate = firstChartDate + i;
+    return {
+      date: currentDate,
+      // Find the statistic for 'currentDate', or use the last known statistic if none exists for that date
+      stat: stat.find(x => x.date === currentDate)?.stat ?? null,
+    };
+  });
+}
 
 export default function ConnectionStatsSidebar() {
   const { sidebarView, setSidebarView } = useUiStore();
   const {
-    mediaStream,
+     mediaStream,
-    peerConnection,
+     peerConnection,
-    inboundRtpStats: inboundVideoRtpStats,
+     inboundRtpStats,
-    appendInboundRtpStats: appendInboundVideoRtpStats,
+     appendInboundRtpStats,
-    candidatePairStats: iceCandidatePairStats,
+     candidatePairStats,
-    appendCandidatePairStats,
+     appendCandidatePairStats,
-    appendLocalCandidateStats,
+     appendLocalCandidateStats,
-    appendRemoteCandidateStats,
+     appendRemoteCandidateStats,
-    appendDiskDataChannelStats,
+     appendDiskDataChannelStats,
   } = useRTCStore();
 
+  function isMetricSupported<T, K extends keyof T>(
+    stream: Map<number, T>,
+    metric: K,
+  ): boolean {
+    return Array.from(stream).some(([, stat]) => stat[metric] !== undefined);
+  }
+
   useInterval(function collectWebRTCStats() {
     (async () => {
       if (!mediaStream) return;
-
       const videoTrack = mediaStream.getVideoTracks()[0];
       if (!videoTrack) return;
-
       const stats = await peerConnection?.getStats();
       let successfulLocalCandidateId: string | null = null;
       let successfulRemoteCandidateId: string | null = null;
 
       stats?.forEach(report => {
-        if (report.type === "inbound-rtp" && report.kind === "video") {
+        if (report.type === "inbound-rtp") {
-          appendInboundVideoRtpStats(report);
+          appendInboundRtpStats(report);
         } else if (report.type === "candidate-pair" && report.nominated) {
           if (report.state === "succeeded") {
             successfulLocalCandidateId = report.localCandidateId;
@@ -57,133 +91,144 @@ export default function ConnectionStatsSidebar() {
     })();
   }, 500);
 
-  const jitterBufferDelay = createChartArray(inboundVideoRtpStats, "jitterBufferDelay");
-  const jitterBufferEmittedCount = createChartArray(
-    inboundVideoRtpStats,
-    "jitterBufferEmittedCount",
-  );
-
-  const jitterBufferAvgDelayData = jitterBufferDelay.map((d, idx) => {
-    if (idx === 0) return { date: d.date, metric: null };
-    const prevDelay = jitterBufferDelay[idx - 1]?.metric as number | null | undefined;
-    const currDelay = d.metric as number | null | undefined;
-    const prevCountEmitted =
-      (jitterBufferEmittedCount[idx - 1]?.metric as number | null | undefined) ?? null;
-    const currCountEmitted =
-      (jitterBufferEmittedCount[idx]?.metric as number | null | undefined) ?? null;
-
-    if (
-      prevDelay == null ||
-      currDelay == null ||
-      prevCountEmitted == null ||
-      currCountEmitted == null
-    ) {
-      return { date: d.date, metric: null };
-    }
-
-    const deltaDelay = currDelay - prevDelay;
-    const deltaEmitted = currCountEmitted - prevCountEmitted;
-
-    // Guard counter resets or no emitted frames
-    if (deltaDelay < 0 || deltaEmitted <= 0) {
-      return { date: d.date, metric: null };
-    }
-
-    const valueMs = Math.round((deltaDelay / deltaEmitted) * 1000);
-    return { date: d.date, metric: valueMs };
-  });
-
   return (
     <div className="grid h-full grid-rows-(--grid-headerBody) shadow-xs">
       <SidebarHeader title="Connection Stats" setSidebarView={setSidebarView} />
       <div className="h-full space-y-4 overflow-y-scroll bg-white px-4 py-2 pb-8 dark:bg-slate-900">
         <div className="space-y-4">
+          {/*
+            The entire sidebar component is always rendered, with a display none when not visible
+            The charts below, need a height and width, otherwise they throw. So simply don't render them unless the thing is visible
+          */}
           {sidebarView === "connection-stats" && (
-            <div className="space-y-8">
+            <div className="space-y-4">
-              {/* Connection Group */}
+              <div className="space-y-2">
-              <div className="space-y-3">
+                <div>
-                <SettingsSectionHeader
+                  <h2 className="text-lg font-semibold text-black dark:text-white">
-                  title="Connection"
+                    Packets Lost
-                  description="The connection between the client and the JetKVM."
+                  </h2>
-                />
+                  <p className="text-sm text-slate-700 dark:text-slate-300">
-                <Metric
+                    Number of data packets lost during transmission.
-                  title="Round-Trip Time"
+                  </p>
-                  description="Round-trip time for the active ICE candidate pair between peers."
+                </div>
-                  stream={iceCandidatePairStats}
+                <GridCard>
-                  metric="currentRoundTripTime"
+                  <div className="flex h-[127px] w-full items-center justify-center text-sm text-slate-500">
-                  map={x => ({
+                    {inboundRtpStats.size === 0 ? (
-                    date: x.date,
+                      <div className="flex flex-col items-center space-y-1">
-                    metric: x.metric != null ? Math.round(x.metric * 1000) : null,
+                        <p className="text-slate-700">Waiting for data...</p>
-                  })}
+                      </div>
-                  domain={[0, 600]}
+                    ) : isMetricSupported(inboundRtpStats, "packetsLost") ? (
-                  unit=" ms"
+                      <StatChart
-                />
+                        data={createChartArray(inboundRtpStats, "packetsLost")}
+                        domain={[0, 100]}
+                        unit=" packets"
+                      />
+                    ) : (
+                      <div className="flex flex-col items-center space-y-1">
+                        <p className="text-black">Metric not supported</p>
+                      </div>
+                    )}
+                  </div>
+                </GridCard>
               </div>
-
+              <div className="space-y-2">
-              {/* Video Group */}
+                <div>
-              <div className="space-y-3">
+                  <h2 className="text-lg font-semibold text-black dark:text-white">
-                <SettingsSectionHeader
+                    Round-Trip Time
-                  title="Video"
+                  </h2>
-                  description="The video stream from the JetKVM to the client."
+                  <p className="text-sm text-slate-700 dark:text-slate-300">
-                />
+                    Time taken for data to travel from source to destination and back
-
+                  </p>
-                {/* RTP Jitter */}
+                </div>
-                <Metric
+                <GridCard>
-                  title="Network Stability"
+                  <div className="flex h-[127px] w-full items-center justify-center text-sm text-slate-500">
-                  badge="Jitter"
+                    {inboundRtpStats.size === 0 ? (
-                  badgeTheme="light"
+                      <div className="flex flex-col items-center space-y-1">
-                  description="How steady the flow of inbound video packets is across the network."
+                        <p className="text-slate-700">Waiting for data...</p>
-                  stream={inboundVideoRtpStats}
+                      </div>
-                  metric="jitter"
+                    ) : isMetricSupported(candidatePairStats, "currentRoundTripTime") ? (
-                  map={x => ({
+                      <StatChart
-                    date: x.date,
+                        data={createChartArray(
-                    metric: x.metric != null ? Math.round(x.metric * 1000) : null,
+                          candidatePairStats,
-                  })}
+                          "currentRoundTripTime",
-                  domain={[0, 10]}
+                        ).map(x => {
-                  unit=" ms"
+                          return {
-                />
+                            date: x.date,
-
+                            stat: x.stat ? Math.round(x.stat * 1000) : null,
-                {/* Playback Delay */}
+                          };
-                <Metric
+                        })}
-                  title="Playback Delay"
+                        domain={[0, 600]}
-                  description="Delay added by the jitter buffer to smooth playback when frames arrive unevenly."
+                        unit=" ms"
-                  badge="Jitter Buffer Avg. Delay"
+                      />
-                  badgeTheme="light"
+                    ) : (
-                  data={jitterBufferAvgDelayData}
+                      <div className="flex flex-col items-center space-y-1">
-                  gate={inboundVideoRtpStats}
+                        <p className="text-black">Metric not supported</p>
-                  supported={
+                      </div>
-                    someIterable(
+                    )}
-                      inboundVideoRtpStats,
+                  </div>
-                      ([, x]) => x.jitterBufferDelay != null,
+                </GridCard>
-                    ) &&
+              </div>
-                    someIterable(
+              <div className="space-y-2">
-                      inboundVideoRtpStats,
+                <div>
-                      ([, x]) => x.jitterBufferEmittedCount != null,
+                  <h2 className="text-lg font-semibold text-black dark:text-white">
-                    )
+                    Jitter
-                  }
+                  </h2>
-                  domain={[0, 30]}
+                  <p className="text-sm text-slate-700 dark:text-slate-300">
-                  unit=" ms"
+                    Variation in packet delay, affecting video smoothness.{" "}
-                />
+                  </p>
-
+                </div>
-                {/* Packets Lost */}
+                <GridCard>
-                <Metric
+                  <div className="flex h-[127px] w-full items-center justify-center text-sm text-slate-500">
-                  title="Packets Lost"
+                    {inboundRtpStats.size === 0 ? (
-                  description="Count of lost inbound video RTP packets."
+                      <div className="flex flex-col items-center space-y-1">
-                  stream={inboundVideoRtpStats}
+                        <p className="text-slate-700">Waiting for data...</p>
-                  metric="packetsLost"
+                      </div>
-                  domain={[0, 100]}
+                    ) : (
-                  unit=" packets"
+                      <StatChart
-                />
+                        data={createChartArray(inboundRtpStats, "jitter").map(x => {
-
+                          return {
-                {/* Frames Per Second */}
+                            date: x.date,
-                <Metric
+                            stat: x.stat ? Math.round(x.stat * 1000) : null,
-                  title="Frames per second"
+                          };
-                  description="Number of inbound video frames displayed per second."
+                        })}
-                  stream={inboundVideoRtpStats}
+                        domain={[0, 300]}
-                  metric="framesPerSecond"
+                        unit=" ms"
-                  domain={[0, 80]}
+                      />
-                  unit=" fps"
+                    )}
-                />
+                  </div>
+                </GridCard>
+              </div>
+              <div className="space-y-2">
+                <div>
+                  <h2 className="text-lg font-semibold text-black dark:text-white">
+                    Frames per second
+                  </h2>
+                  <p className="text-sm text-slate-700 dark:text-slate-300">
+                    Number of video frames displayed per second.
+                  </p>
+                </div>
+                <GridCard>
+                  <div className="flex h-[127px] w-full items-center justify-center text-sm text-slate-500">
+                    {inboundRtpStats.size === 0 ? (
+                      <div className="flex flex-col items-center space-y-1">
+                        <p className="text-slate-700">Waiting for data...</p>
+                      </div>
+                    ) : (
+                      <StatChart
+                        data={createChartArray(inboundRtpStats, "framesPerSecond").map(
+                          x => {
+                            return {
+                              date: x.date,
+                              stat: x.stat ? x.stat : null,
+                            };
+                          },
+                        )}
+                        domain={[0, 80]}
+                        unit=" fps"
+                      />
+                    )}
+                  </div>
+                </GridCard>
               </div>
             </div>
           )}

ui/src/hooks/stores.ts

@@ -355,10 +355,6 @@ export interface SettingsState {
   setVideoBrightness: (value: number) => void;
   videoContrast: number;
   setVideoContrast: (value: number) => void;
-
-  // Microphone persistence settings
-  microphoneWasEnabled: boolean;
-  setMicrophoneWasEnabled: (enabled: boolean) => void;
 }
 
 export const useSettingsStore = create(
@@ -404,10 +400,6 @@ export const useSettingsStore = create(
       setVideoBrightness: (value: number) => set({ videoBrightness: value }),
       videoContrast: 1.0,
       setVideoContrast: (value: number) => set({ videoContrast: value }),
-
-      // Microphone persistence settings
-      microphoneWasEnabled: false,
-      setMicrophoneWasEnabled: (enabled: boolean) => set({ microphoneWasEnabled: enabled }),
     }),
     {
       name: "settings",

ui/src/hooks/useMicrophone.ts

@@ -1,6 +1,6 @@
 import { useCallback, useEffect, useRef, useState } from "react";
 
-import { useRTCStore, useSettingsStore } from "@/hooks/stores";
+import { useRTCStore } from "@/hooks/stores";
 import api from "@/api";
 import { devLog, devInfo, devWarn, devError, devOnly } from "@/utils/debug";
 import { AUDIO_CONFIG } from "@/config/constants";
@@ -23,8 +23,6 @@ export function useMicrophone() {
     setMicrophoneMuted,
   } = useRTCStore();
 
-  const { microphoneWasEnabled, setMicrophoneWasEnabled } = useSettingsStore();
-
   const microphoneStreamRef = useRef<MediaStream | null>(null);
   
   // Loading states
@@ -63,7 +61,7 @@ export function useMicrophone() {
     // Cleaning up microphone stream
     
     if (microphoneStreamRef.current) {
-      microphoneStreamRef.current.getTracks().forEach((track: MediaStreamTrack) => {
+      microphoneStreamRef.current.getTracks().forEach(track => {
         track.stop();
       });
       microphoneStreamRef.current = null;
@@ -195,7 +193,7 @@ export function useMicrophone() {
         
         // Find the audio transceiver (should already exist with sendrecv direction)
         const transceivers = peerConnection.getTransceivers();
-        devLog("Available transceivers:", transceivers.map((t: RTCRtpTransceiver) => ({
+        devLog("Available transceivers:", transceivers.map(t => ({
           direction: t.direction,
           mid: t.mid,
           senderTrack: t.sender.track?.kind,
@@ -203,7 +201,7 @@ export function useMicrophone() {
         })));
         
         // Look for an audio transceiver that can send (has sendrecv or sendonly direction)
-        const audioTransceiver = transceivers.find((transceiver: RTCRtpTransceiver) => {
+        const audioTransceiver = transceivers.find(transceiver => {
           // Check if this transceiver is for audio and can send
           const canSend = transceiver.direction === 'sendrecv' || transceiver.direction === 'sendonly';
           
@@ -391,9 +389,6 @@ export function useMicrophone() {
       setMicrophoneActive(true);
       setMicrophoneMuted(false);
       
-      // Save microphone enabled state for auto-restore on page reload
-      setMicrophoneWasEnabled(true);
-      
       devLog("Microphone state set to active. Verifying state:", {
         streamInRef: !!microphoneStreamRef.current,
         streamInStore: !!microphoneStream,
@@ -452,7 +447,7 @@ export function useMicrophone() {
       setIsStarting(false);
       return { success: false, error: micError };
     }
-  }, [peerConnection, setMicrophoneStream, setMicrophoneSender, setMicrophoneActive, setMicrophoneMuted, setMicrophoneWasEnabled, stopMicrophoneStream, isMicrophoneActive, isMicrophoneMuted, microphoneStream, isStarting, isStopping, isToggling]);
+  }, [peerConnection, setMicrophoneStream, setMicrophoneSender, setMicrophoneActive, setMicrophoneMuted, stopMicrophoneStream, isMicrophoneActive, isMicrophoneMuted, microphoneStream, isStarting, isStopping, isToggling]);
 
 
 
@@ -481,9 +476,6 @@ export function useMicrophone() {
       setMicrophoneActive(false);
       setMicrophoneMuted(false);
 
-      // Save microphone disabled state for persistence
-      setMicrophoneWasEnabled(false);
-
       // Sync state after stopping to ensure consistency (with longer delay)
       setTimeout(() => syncMicrophoneState(), 500);
 
@@ -500,7 +492,7 @@ export function useMicrophone() {
         }
       };
     }
-  }, [stopMicrophoneStream, syncMicrophoneState, setMicrophoneActive, setMicrophoneMuted, setMicrophoneWasEnabled, isStarting, isStopping, isToggling]);
+  }, [stopMicrophoneStream, syncMicrophoneState, setMicrophoneActive, setMicrophoneMuted, isStarting, isStopping, isToggling]);
 
   // Toggle microphone mute
   const toggleMicrophoneMute = useCallback(async (): Promise<{ success: boolean; error?: MicrophoneError }> => {
@@ -568,7 +560,7 @@ export function useMicrophone() {
       const newMutedState = !isMicrophoneMuted;
       
       // Mute/unmute the audio track
-      audioTracks.forEach((track: MediaStreamTrack) => {
+      audioTracks.forEach(track => {
         track.enabled = !newMutedState;
         devLog(`Audio track ${track.id} enabled: ${track.enabled}`);
       });
@@ -615,30 +607,10 @@ export function useMicrophone() {
 
 
 
-  // Sync state on mount and auto-restore microphone if it was enabled before page reload
+  // Sync state on mount
   useEffect(() => {
-    const autoRestoreMicrophone = async () => {
+    syncMicrophoneState();
-      // First sync the current state
+  }, [syncMicrophoneState]);
-      await syncMicrophoneState();
-      
-      // If microphone was enabled before page reload and is not currently active, restore it
-      if (microphoneWasEnabled && !isMicrophoneActive && peerConnection) {
-        devLog("Auto-restoring microphone after page reload");
-        try {
-          const result = await startMicrophone();
-          if (result.success) {
-            devInfo("Microphone auto-restored successfully after page reload");
-          } else {
-            devWarn("Failed to auto-restore microphone:", result.error);
-          }
-        } catch (error) {
-          devWarn("Error during microphone auto-restoration:", error);
-        }
-      }
-    };
-    
-    autoRestoreMicrophone();
-  }, [syncMicrophoneState, microphoneWasEnabled, isMicrophoneActive, peerConnection, startMicrophone]);
 
   // Cleanup on unmount - use ref to avoid dependency on stopMicrophoneStream
   useEffect(() => {
@@ -647,7 +619,7 @@ export function useMicrophone() {
       const stream = microphoneStreamRef.current;
       if (stream) {
         devLog("Cleanup: stopping microphone stream on unmount");
-        stream.getAudioTracks().forEach((track: MediaStreamTrack) => {
+        stream.getAudioTracks().forEach(track => {
           track.stop();
           devLog(`Cleanup: stopped audio track ${track.id}`);
         });

ui/src/main.tsx

@@ -116,7 +116,6 @@ if (isOnDevice) {
       path: "/",
       errorElement: <ErrorBoundary />,
       element: <DeviceRoute />,
-      HydrateFallback: () => <div className="p-4">Loading...</div>,
       loader: DeviceRoute.loader,
       children: [
         {

ui/src/routes/devices.$id.mount.tsx

@@ -355,7 +355,7 @@ function UrlView({
   const popularImages = [
     {
       name: "Ubuntu 24.04 LTS",
-      url: "https://releases.ubuntu.com/24.04.3/ubuntu-24.04.3-desktop-amd64.iso",
+      url: "https://releases.ubuntu.com/24.04.2/ubuntu-24.04.2-desktop-amd64.iso",
       icon: UbuntuIcon,
     },
     {
@@ -369,8 +369,8 @@ function UrlView({
       icon: DebianIcon,
     },
     {
-      name: "Fedora 42",
+      name: "Fedora 41",
-      url: "https://download.fedoraproject.org/pub/fedora/linux/releases/42/Workstation/x86_64/iso/Fedora-Workstation-Live-42-1.1.x86_64.iso",
+      url: "https://download.fedoraproject.org/pub/fedora/linux/releases/41/Workstation/x86_64/iso/Fedora-Workstation-Live-x86_64-41-1.4.iso",
       icon: FedoraIcon,
     },
     {
@@ -385,7 +385,7 @@ function UrlView({
     },
     {
       name: "Arch Linux",
-      url: "https://archlinux.doridian.net/iso/latest/archlinux-x86_64.iso",
+      url: "https://archlinux.doridian.net/iso/2025.02.01/archlinux-2025.02.01-x86_64.iso",
       icon: ArchIcon,
     },
     {

ui/src/routes/devices.$id.settings.video.tsx

@@ -1,16 +1,15 @@
-import { useEffect, useState } from "react";
+import { useState, useEffect } from "react";
 
 import { Button } from "@/components/Button";
 import { TextAreaWithLabel } from "@/components/TextArea";
 import { JsonRpcResponse, useJsonRpc } from "@/hooks/useJsonRpc";
 import { SettingsPageHeader } from "@components/SettingsPageheader";
 import { useSettingsStore } from "@/hooks/stores";
-import { SelectMenuBasic } from "@components/SelectMenuBasic";
+
-import Fieldset from "@components/Fieldset";
+import notifications from "../notifications";
-import notifications from "@/notifications";
+import { SelectMenuBasic } from "../components/SelectMenuBasic";
 
 import { SettingsItem } from "./devices.$id.settings";
-
 const defaultEdid =
   "00ffffffffffff0052620188008888881c150103800000780a0dc9a05747982712484c00000001010101010101010101010101010101023a801871382d40582c4500c48e2100001e011d007251d01e206e285500c48e2100001e000000fc00543734392d6648443732300a20000000fd00147801ff1d000a202020202020017b";
 const edids = [
@@ -51,27 +50,21 @@ export default function SettingsVideoRoute() {
   const [streamQuality, setStreamQuality] = useState("1");
   const [customEdidValue, setCustomEdidValue] = useState<string | null>(null);
   const [edid, setEdid] = useState<string | null>(null);
-  const [edidLoading, setEdidLoading] = useState(false);
 
   // Video enhancement settings from store
   const {
-    videoSaturation,
+    videoSaturation, setVideoSaturation,
-    setVideoSaturation,
+    videoBrightness, setVideoBrightness,
-    videoBrightness,
+    videoContrast, setVideoContrast
-    setVideoBrightness,
-    videoContrast,
-    setVideoContrast,
   } = useSettingsStore();
 
   useEffect(() => {
-    setEdidLoading(true);
     send("getStreamQualityFactor", {}, (resp: JsonRpcResponse) => {
       if ("error" in resp) return;
       setStreamQuality(String(resp.result));
     });
 
     send("getEDID", {}, (resp: JsonRpcResponse) => {
-      setEdidLoading(false);
       if ("error" in resp) {
         notifications.error(`Failed to get EDID: ${resp.error.data || "Unknown error"}`);
         return;
@@ -96,36 +89,28 @@ export default function SettingsVideoRoute() {
   }, [send]);
 
   const handleStreamQualityChange = (factor: string) => {
-    send(
+    send("setStreamQualityFactor", { factor: Number(factor) }, (resp: JsonRpcResponse) => {
-      "setStreamQualityFactor",
+      if ("error" in resp) {
-      { factor: Number(factor) },
+        notifications.error(
-      (resp: JsonRpcResponse) => {
+          `Failed to set stream quality: ${resp.error.data || "Unknown error"}`,
-        if ("error" in resp) {
-          notifications.error(
-            `Failed to set stream quality: ${resp.error.data || "Unknown error"}`,
-          );
-          return;
-        }
-
-        notifications.success(
-          `Stream quality set to ${streamQualityOptions.find(x => x.value === factor)?.label}`,
         );
-        setStreamQuality(factor);
+        return;
-      },
+      }
-    );
+
+      notifications.success(`Stream quality set to ${streamQualityOptions.find(x => x.value === factor)?.label}`);
+      setStreamQuality(factor);
+    });
   };
 
   const handleEDIDChange = (newEdid: string) => {
-    setEdidLoading(true);
     send("setEDID", { edid: newEdid }, (resp: JsonRpcResponse) => {
-      setEdidLoading(false);
       if ("error" in resp) {
         notifications.error(`Failed to set EDID: ${resp.error.data || "Unknown error"}`);
         return;
       }
 
       notifications.success(
-        `EDID set successfully to ${edids.find(x => x.value === newEdid)?.label ?? "the custom EDID"}`,
+        `EDID set successfully to ${edids.find(x => x.value === newEdid)?.label}`,
       );
       // Update the EDID value in the UI
       setEdid(newEdid);
@@ -173,7 +158,7 @@ export default function SettingsVideoRoute() {
                   step="0.1"
                   value={videoSaturation}
                   onChange={e => setVideoSaturation(parseFloat(e.target.value))}
-                  className="h-2 w-32 cursor-pointer appearance-none rounded-lg bg-gray-200 dark:bg-gray-700"
+                  className="w-32 h-2 bg-gray-200 rounded-lg appearance-none cursor-pointer dark:bg-gray-700"
                 />
               </SettingsItem>
 
@@ -188,7 +173,7 @@ export default function SettingsVideoRoute() {
                   step="0.1"
                   value={videoBrightness}
                   onChange={e => setVideoBrightness(parseFloat(e.target.value))}
-                  className="h-2 w-32 cursor-pointer appearance-none rounded-lg bg-gray-200 dark:bg-gray-700"
+                  className="w-32 h-2 bg-gray-200 rounded-lg appearance-none cursor-pointer dark:bg-gray-700"
                 />
               </SettingsItem>
 
@@ -203,7 +188,7 @@ export default function SettingsVideoRoute() {
                   step="0.1"
                   value={videoContrast}
                   onChange={e => setVideoContrast(parseFloat(e.target.value))}
-                  className="h-2 w-32 cursor-pointer appearance-none rounded-lg bg-gray-200 dark:bg-gray-700"
+                  className="w-32 h-2 bg-gray-200 rounded-lg appearance-none cursor-pointer dark:bg-gray-700"
                 />
               </SettingsItem>
 
@@ -220,64 +205,60 @@ export default function SettingsVideoRoute() {
                 />
               </div>
             </div>
-            <Fieldset disabled={edidLoading} className="space-y-2">
+
-              <SettingsItem
+            <SettingsItem
-                title="EDID"
+              title="EDID"
-                description="Adjust the EDID settings for the display"
+              description="Adjust the EDID settings for the display"
-                loading={edidLoading}
+            >
-              >
+              <SelectMenuBasic
-                <SelectMenuBasic
+                size="SM"
-                  size="SM"
+                label=""
-                  label=""
+                fullWidth
-                  fullWidth
+                value={customEdidValue ? "custom" : edid || "asd"}
-                  value={customEdidValue ? "custom" : edid || "asd"}
+                onChange={e => {
-                  onChange={e => {
+                  if (e.target.value === "custom") {
-                    if (e.target.value === "custom") {
+                    setEdid("custom");
-                      setEdid("custom");
+                    setCustomEdidValue("");
-                      setCustomEdidValue("");
+                  } else {
-                    } else {
+                    setCustomEdidValue(null);
-                      setCustomEdidValue(null);
+                    handleEDIDChange(e.target.value as string);
-                      handleEDIDChange(e.target.value as string);
+                  }
-                    }
+                }}
-                  }}
+                options={[...edids, { value: "custom", label: "Custom" }]}
-                  options={[...edids, { value: "custom", label: "Custom" }]}
+              />
+            </SettingsItem>
+            {customEdidValue !== null && (
+              <>
+                <SettingsItem
+                  title="Custom EDID"
+                  description="EDID details video mode compatibility. Default settings works in most cases, but unique UEFI/BIOS might need adjustments."
                 />
-              </SettingsItem>
+                <TextAreaWithLabel
-              {customEdidValue !== null && (
+                  label="EDID File"
-                <>
+                  placeholder="00F..."
-                  <SettingsItem
+                  rows={3}
-                    title="Custom EDID"
+                  value={customEdidValue}
-                    description="EDID details video mode compatibility. Default settings works in most cases, but unique UEFI/BIOS might need adjustments."
+                  onChange={e => setCustomEdidValue(e.target.value)}
+                />
+                <div className="flex justify-start gap-x-2">
+                  <Button
+                    size="SM"
+                    theme="primary"
+                    text="Set Custom EDID"
+                    onClick={() => handleEDIDChange(customEdidValue)}
                   />
-                  <TextAreaWithLabel
+                  <Button
-                    label="EDID File"
+                    size="SM"
-                    placeholder="00F..."
+                    theme="light"
-                    rows={3}
+                    text="Restore to default"
-                    value={customEdidValue}
+                    onClick={() => {
-                    onChange={e => setCustomEdidValue(e.target.value)}
+                      setCustomEdidValue(null);
+                      handleEDIDChange(defaultEdid);
+                    }}
                   />
-                  <div className="flex justify-start gap-x-2">
+                </div>
-                    <Button
+              </>
-                      size="SM"
+            )}
-                      theme="primary"
-                      text="Set Custom EDID"
-                      loading={edidLoading}
-                      onClick={() => handleEDIDChange(customEdidValue)}
-                    />
-                    <Button
-                      size="SM"
-                      theme="light"
-                      text="Restore to default"
-                      loading={edidLoading}
-                      onClick={() => {
-                        setCustomEdidValue(null);
-                        handleEDIDChange(defaultEdid);
-                      }}
-                    />
-                  </div>
-                </>
-              )}
-            </Fieldset>
           </div>
         </div>
       </div>

ui/src/routes/devices.$id.tsx

@@ -54,7 +54,6 @@ import { useDeviceUiNavigation } from "@/hooks/useAppNavigation";
 import { FeatureFlagProvider } from "@/providers/FeatureFlagProvider";
 import { DeviceStatus } from "@routes/welcome-local";
 import { SystemVersionInfo } from "@routes/devices.$id.settings.general.update";
-import audioQualityService from "@/services/audioQualityService";
 
 interface LocalLoaderResp {
   authMode: "password" | "noPassword" | null;
@@ -475,27 +474,8 @@ export default function KvmIdRoute() {
       }
     };
 
-    pc.ontrack = function (event: RTCTrackEvent) {
+    pc.ontrack = function (event) {
-      // Handle separate MediaStreams for audio and video tracks
+      setMediaStream(event.streams[0]);
-      const track = event.track;
-      const streams = event.streams;
-      
-      if (streams && streams.length > 0) {
-        // Get existing MediaStream or create a new one
-        const existingStream = useRTCStore.getState().mediaStream;
-        let combinedStream: MediaStream;
-        
-        if (existingStream) {
-          combinedStream = existingStream;
-          // Add the new track to the existing stream
-          combinedStream.addTrack(track);
-        } else {
-          // Create a new MediaStream with the track
-          combinedStream = new MediaStream([track]);
-        }
-        
-        setMediaStream(combinedStream);
-      }
     };
 
     setTransceiver(pc.addTransceiver("video", { direction: "recvonly" }));
@@ -553,11 +533,6 @@ export default function KvmIdRoute() {
     };
   }, [clearCandidatePairStats, clearInboundRtpStats, setPeerConnection, setSidebarView]);
 
-  // Register callback with audioQualityService
-  useEffect(() => {
-    audioQualityService.setReconnectionCallback(setupPeerConnection);
-  }, [setupPeerConnection]);
-
   // TURN server usage detection
   useEffect(() => {
     if (peerConnectionState !== "connected") return;

ui/src/services/audioQualityService.ts

@@ -24,7 +24,6 @@ class AudioQualityService {
     2: 'High',
     3: 'Ultra'
   };
-  private reconnectionCallback: (() => Promise<void>) | null = null;
 
   /**
    * Fetch audio quality presets from the backend
@@ -97,34 +96,12 @@ class AudioQualityService {
   }
 
   /**
-   * Set reconnection callback for WebRTC reset
+   * Set audio quality
-   */
-  setReconnectionCallback(callback: () => Promise<void>): void {
-    this.reconnectionCallback = callback;
-  }
-
-  /**
-   * Trigger audio track replacement using backend's track replacement mechanism
-   */
-  private async replaceAudioTrack(): Promise<void> {
-    if (this.reconnectionCallback) {
-      await this.reconnectionCallback();
-    }
-  }
-
-  /**
-   * Set audio quality with track replacement
    */
   async setAudioQuality(quality: number): Promise<boolean> {
     try {
       const response = await api.POST('/audio/quality', { quality });
-      
+      return response.ok;
-      if (!response.ok) {
-        return false;
-      }
-      
-      await this.replaceAudioTrack();
-      return true;
     } catch (error) {
       console.error('Failed to set audio quality:', error);
       return false;

ui/src/utils.ts

@@ -94,17 +94,6 @@ export const formatters = {
   },
 };
 
-export function someIterable<T>(
-  iterable: Iterable<T>,
-  predicate: (item: T) => boolean,
-): boolean {
-  for (const item of iterable) {
-    if (predicate(item)) return true;
-  }
-
-  return false;
-}
-
 export const VIDEO = new Blob(
   [
     new Uint8Array([

webrtc.go

@@ -4,7 +4,6 @@ import (
 	"context"
 	"encoding/base64"
 	"encoding/json"
-	"fmt"
 	"net"
 	"runtime"
 	"strings"
@@ -25,7 +24,6 @@ type Session struct {
 	peerConnection           *webrtc.PeerConnection
 	VideoTrack               *webrtc.TrackLocalStaticSample
 	AudioTrack               *webrtc.TrackLocalStaticSample
-	AudioRtpSender           *webrtc.RTPSender
 	ControlChannel           *webrtc.DataChannel
 	RPCChannel               *webrtc.DataChannel
 	HidChannel               *webrtc.DataChannel
@@ -233,21 +231,22 @@ func newSession(config SessionConfig) (*Session, error) {
 		}
 	})
 
-	session.VideoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, "video", "kvm-video")
+	session.VideoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, "video", "kvm")
 	if err != nil {
 		scopedLogger.Warn().Err(err).Msg("Failed to create VideoTrack")
 		return nil, err
 	}
 
-	session.AudioTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "kvm-audio")
+	session.AudioTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "kvm")
 	if err != nil {
 		scopedLogger.Warn().Err(err).Msg("Failed to add VideoTrack to PeerConnection")
 		return nil, err
 	}
 
-	// Update the audio relay with the new WebRTC audio track asynchronously
+	// Update the audio relay with the new WebRTC audio track
-	// This prevents blocking during session creation and avoids mutex deadlocks
+	if err := audio.UpdateAudioRelayTrack(session.AudioTrack); err != nil {
-	audio.UpdateAudioRelayTrackAsync(session.AudioTrack)
+		scopedLogger.Warn().Err(err).Msg("Failed to update audio relay track")
+	}
 
 	videoRtpSender, err := peerConnection.AddTrack(session.VideoTrack)
 	if err != nil {
@@ -262,7 +261,6 @@ func newSession(config SessionConfig) (*Session, error) {
 		return nil, err
 	}
 	audioRtpSender := audioTransceiver.Sender()
-	session.AudioRtpSender = audioRtpSender
 
 	// Handle incoming audio track (microphone from browser)
 	peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
@@ -412,22 +410,6 @@ func (s *Session) stopAudioProcessor() {
 	s.audioWg.Wait()
 }
 
-// ReplaceAudioTrack replaces the current audio track with a new one
-func (s *Session) ReplaceAudioTrack(newTrack *webrtc.TrackLocalStaticSample) error {
-	if s.AudioRtpSender == nil {
-		return fmt.Errorf("audio RTP sender not available")
-	}
-
-	// Replace the track using the RTP sender
-	if err := s.AudioRtpSender.ReplaceTrack(newTrack); err != nil {
-		return fmt.Errorf("failed to replace audio track: %w", err)
-	}
-
-	// Update the session's audio track reference
-	s.AudioTrack = newTrack
-	return nil
-}
-
 func drainRtpSender(rtpSender *webrtc.RTPSender) {
 	// Lock to OS thread to isolate RTCP processing
 	runtime.LockOSThread()