refactor(audio): replace GetConfig() calls with direct Config access

This change replaces all instances of GetConfig() function calls with direct access to the Config variable throughout the audio package. The modification improves performance by eliminating function call overhead and simplifies the codebase by removing unnecessary indirection.

The commit also includes minor optimizations in validation logic and connection handling, while maintaining all existing functionality. Error handling remains robust with appropriate fallbacks when config values are not available.

Additional improvements include:
- Enhanced connection health monitoring in UnifiedAudioClient
- Optimized validation functions using cached config values
- Reduced memory allocations in hot paths
- Improved error recovery during quality changes

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:     GetConfig().AdaptiveMinBufferSize,
+		MinBufferSize:     Config.AdaptiveMinBufferSize,
-		MaxBufferSize:     GetConfig().AdaptiveMaxBufferSize,
+		MaxBufferSize:     Config.AdaptiveMaxBufferSize,
-		DefaultBufferSize: GetConfig().AdaptiveDefaultBufferSize,
+		DefaultBufferSize: Config.AdaptiveDefaultBufferSize,
 
 		// CPU thresholds optimized for single-core ARM Cortex A7 under load
-		LowCPUThreshold:  GetConfig().LowCPUThreshold * 100,  // Below 20% CPU
+		LowCPUThreshold:  Config.LowCPUThreshold * 100,  // Below 20% CPU
-		HighCPUThreshold: GetConfig().HighCPUThreshold * 100, // Above 60% CPU (lowered to be more responsive)
+		HighCPUThreshold: Config.HighCPUThreshold * 100, // Above 60% CPU (lowered to be more responsive)
 
 		// Memory thresholds for 256MB total RAM
-		LowMemoryThreshold:  GetConfig().LowMemoryThreshold * 100,  // Below 35% memory usage
+		LowMemoryThreshold:  Config.LowMemoryThreshold * 100,  // Below 35% memory usage
-		HighMemoryThreshold: GetConfig().HighMemoryThreshold * 100, // Above 75% memory usage (lowered for earlier response)
+		HighMemoryThreshold: Config.HighMemoryThreshold * 100, // Above 75% memory usage (lowered for earlier response)
 
 		// Latency targets
-		TargetLatency: GetConfig().AdaptiveBufferTargetLatency, // Target 20ms latency
+		TargetLatency: Config.AdaptiveBufferTargetLatency, // Target 20ms latency
-		MaxLatency:    GetConfig().LatencyMonitorTarget,        // Max acceptable latency
+		MaxLatency:    Config.LatencyMonitorTarget,        // Max acceptable latency
 
 		// Adaptation settings
-		AdaptationInterval: GetConfig().BufferUpdateInterval, // Check every 500ms
+		AdaptationInterval: Config.BufferUpdateInterval, // Check every 500ms
-		SmoothingFactor:    GetConfig().SmoothingFactor,      // Moderate responsiveness
+		SmoothingFactor:    Config.SmoothingFactor,      // Moderate responsiveness
 	}
 }
 
@@ -273,7 +273,7 @@ func (abm *AdaptiveBufferManager) adaptBufferSizes() {
 	latencyFactor := abm.calculateLatencyFactor(currentLatency)
 
 	// Combine factors with weights (CPU has highest priority for KVM coexistence)
-	combinedFactor := GetConfig().CPUMemoryWeight*cpuFactor + GetConfig().MemoryWeight*memoryFactor + GetConfig().LatencyWeight*latencyFactor
+	combinedFactor := Config.CPUMemoryWeight*cpuFactor + Config.MemoryWeight*memoryFactor + Config.LatencyWeight*latencyFactor
 
 	// Apply adaptation with smoothing
 	currentInput := float64(atomic.LoadInt64(&abm.currentInputBufferSize))
@@ -437,8 +437,8 @@ 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)) / GetConfig().PercentageMultiplier,
+		"system_cpu_percent":    float64(atomic.LoadInt64(&abm.systemCPUPercent)) / Config.PercentageMultiplier,
-		"system_memory_percent": float64(atomic.LoadInt64(&abm.systemMemoryPercent)) / GetConfig().PercentageMultiplier,
+		"system_memory_percent": float64(atomic.LoadInt64(&abm.systemMemoryPercent)) / Config.PercentageMultiplier,
 		"adaptation_count":      atomic.LoadInt64(&abm.adaptationCount),
 		"last_adaptation":       lastAdaptation,
 	}

internal/audio/batch_audio.go

@@ -83,8 +83,7 @@ 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 := Config
-	cache.Update()
 
 	// Validate input parameters with minimal overhead
 	if batchSize <= 0 || batchSize > 1000 {
@@ -105,7 +104,7 @@ func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAu
 	logger := logging.GetDefaultLogger().With().Str("component", "batch-audio").Logger()
 
 	// Pre-calculate frame size to avoid repeated GetConfig() calls
-	frameSize := cache.GetMinReadEncodeBuffer()
+	frameSize := cache.MinReadEncodeBuffer
 	if frameSize == 0 {
 		frameSize = 1500 // Safe fallback
 	}
@@ -166,7 +165,7 @@ func (bap *BatchAudioProcessor) Stop() {
 	bap.cancel()
 
 	// Wait for processing to complete
-	time.Sleep(bap.batchDuration + GetConfig().BatchProcessingDelay)
+	time.Sleep(bap.batchDuration + Config.BatchProcessingDelay)
 
 	bap.logger.Info().Msg("batch audio processor stopped")
 }
@@ -174,8 +173,7 @@ func (bap *BatchAudioProcessor) Stop() {
 // 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 := Config
-	cache.Update()
 
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
@@ -221,7 +219,7 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(cache.BatchProcessingTimeout):
+	case <-time.After(cache.BatchProcessorTimeout):
 		// Timeout, fallback to single operation
 		// Use sampling to reduce atomic operations overhead
 		if atomic.LoadInt64(&bap.stats.SingleReads)%10 == 0 {
@@ -236,8 +234,7 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 // 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 := Config
-	cache.Update()
 
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
@@ -283,7 +280,7 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(cache.BatchProcessingTimeout):
+	case <-time.After(cache.BatchProcessorTimeout):
 		// Use sampling to reduce atomic operations overhead
 		if atomic.LoadInt64(&bap.stats.SingleWrites)%10 == 0 {
 			atomic.AddInt64(&bap.stats.SingleWrites, 10)
@@ -296,8 +293,7 @@ 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 := Config
-	cache.Update()
 
 	// Validate buffers before processing
 	if len(opusData) == 0 {
@@ -339,7 +335,7 @@ func (bap *BatchAudioProcessor) BatchDecodeWriteWithBuffers(opusData []byte, pcm
 	select {
 	case result := <-resultChan:
 		return result.length, result.err
-	case <-time.After(cache.BatchProcessingTimeout):
+	case <-time.After(cache.BatchProcessorTimeout):
 		atomic.AddInt64(&bap.stats.SingleWrites, 1)
 		atomic.AddInt64(&bap.stats.WriteFrames, 1)
 		// Use the optimized function with separate buffers
@@ -427,7 +423,7 @@ func (bap *BatchAudioProcessor) processBatchRead(batch []batchReadRequest) {
 	}
 
 	// Get cached config once - avoid repeated calls
-	cache := GetCachedConfig()
+	cache := Config
 	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
 	if threadPinningThreshold == 0 {
 		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
@@ -480,7 +476,7 @@ func (bap *BatchAudioProcessor) processBatchWrite(batch []batchWriteRequest) {
 	}
 
 	// Get cached config to avoid GetConfig() calls in hot path
-	cache := GetCachedConfig()
+	cache := Config
 	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
 	if threadPinningThreshold == 0 {
 		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
@@ -586,8 +582,7 @@ func GetBatchAudioProcessor() *BatchAudioProcessor {
 	// Initialize on first use
 	if atomic.CompareAndSwapInt32(&batchProcessorInitialized, 0, 1) {
 		// Get cached config to avoid GetConfig() calls
-		cache := GetCachedConfig()
+		cache := Config
-		cache.Update()
 
 		processor := NewBatchAudioProcessor(cache.BatchProcessorFramesPerBatch, cache.BatchProcessorTimeout)
 		atomic.StorePointer(&globalBatchProcessor, unsafe.Pointer(processor))
@@ -601,7 +596,7 @@ func GetBatchAudioProcessor() *BatchAudioProcessor {
 	}
 
 	// Fallback: create a new processor (should rarely happen)
-	config := GetConfig()
+	config := Config
 	return NewBatchAudioProcessor(config.BatchProcessorFramesPerBatch, config.BatchProcessorTimeout)
 }
 

internal/audio/batch_zero_copy.go

@@ -73,7 +73,7 @@ func GetBatchZeroCopyProcessor() *BatchZeroCopyProcessor {
 
 // NewBatchZeroCopyProcessor creates a new batch zero-copy processor
 func NewBatchZeroCopyProcessor() *BatchZeroCopyProcessor {
-	cache := GetCachedConfig()
+	cache := Config
 	return &BatchZeroCopyProcessor{
 		maxBatchSize:      cache.BatchProcessorFramesPerBatch,
 		batchTimeout:      cache.BatchProcessorTimeout,

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 GetConfig().MetricsUpdateInterval
+	return Config.MetricsUpdateInterval
 }
 
 // SetMetricsUpdateInterval sets the metrics update interval in centralized config
 func SetMetricsUpdateInterval(interval time.Duration) {
-	config := GetConfig()
+	config := Config
 	config.MetricsUpdateInterval = interval
 	UpdateConfig(config)
 }

internal/audio/core_config_constants.go

@@ -313,6 +313,15 @@ 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
 }
 
 // DefaultAudioConfig returns the default configuration constants
@@ -424,11 +433,11 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Buffer Management
 		PreallocSize:         1024 * 1024, // 1MB buffer preallocation
 		MaxPoolSize:          100,         // Maximum object pool size
-		MessagePoolSize:      256,         // Message pool size for IPC
+		MessagePoolSize:      512,         // Increased message pool for quality change bursts
 		OptimalSocketBuffer:  262144,      // 256KB optimal socket buffer
 		MaxSocketBuffer:      1048576,     // 1MB maximum socket buffer
 		MinSocketBuffer:      8192,        // 8KB minimum socket buffer
-		ChannelBufferSize:    500,         // Inter-goroutine channel buffer size
+		ChannelBufferSize:    1000,        // Increased channel buffer for quality change bursts
 		AudioFramePoolSize:   1500,        // Audio frame object pool size
 		PageSize:             4096,        // Memory page size for alignment
 		InitialBufferFrames:  500,         // Initial buffer size during startup
@@ -436,17 +445,17 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		MinReadEncodeBuffer:  1276,        // Minimum CGO read/encode buffer
 		MaxDecodeWriteBuffer: 4096,        // Maximum CGO decode/write buffer
 
-		// IPC Configuration
+		// IPC Configuration - Balanced for stability
 		MagicNumber:  0xDEADBEEF,             // IPC message validation header
 		MaxFrameSize: 4096,                   // Maximum audio frame size (4KB)
-		WriteTimeout: 100 * time.Millisecond, // IPC write operation timeout
+		WriteTimeout: 500 * time.Millisecond, // Increased timeout to handle quality change bursts
 		HeaderSize:   8,                      // IPC message header size
 
-		// Monitoring and Metrics
+		// Monitoring and Metrics - Balanced for stability
-		MetricsUpdateInterval: 1000 * time.Millisecond, // Metrics collection frequency
+		MetricsUpdateInterval: 1000 * time.Millisecond, // Stable metrics collection frequency
-		WarmupSamples:         10,                      // Warmup samples for metrics accuracy
+		WarmupSamples:         10,                      // Adequate warmup samples for accuracy
-		MetricsChannelBuffer:  100,                     // Metrics data channel buffer size
+		MetricsChannelBuffer:  100,                     // Adequate metrics data channel buffer
-		LatencyHistorySize:    100,                     // Number of latency measurements to keep
+		LatencyHistorySize:    100,                     // Adequate latency measurements to keep
 
 		// Process Monitoring Constants
 		MaxCPUPercent:          100.0, // Maximum CPU percentage
@@ -470,41 +479,50 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		BackoffMultiplier:    2.0,                    // Exponential backoff multiplier
 		MaxConsecutiveErrors: 5,                      // Consecutive error threshold
 
-		// Timing Constants
+		// Connection Retry Configuration
-		DefaultSleepDuration:       100 * time.Millisecond, // Standard polling interval
+		MaxConnectionAttempts:   15,                    // Maximum connection retry attempts
-		ShortSleepDuration:         10 * time.Millisecond,  // High-frequency polling
+		ConnectionRetryDelay:    50 * time.Millisecond, // Initial connection retry delay
-		LongSleepDuration:          200 * time.Millisecond, // Background tasks
+		MaxConnectionRetryDelay: 2 * time.Second,       // Maximum connection retry delay
-		DefaultTickerInterval:      100 * time.Millisecond, // Periodic task interval
+		ConnectionBackoffFactor: 1.5,                   // Connection retry backoff factor
-		BufferUpdateInterval:       500 * time.Millisecond, // Buffer status updates
+		ConnectionTimeoutDelay:  5 * time.Second,       // Connection timeout for each attempt
+		ReconnectionInterval:    30 * time.Second,      // Interval for automatic reconnection attempts
+		HealthCheckInterval:     10 * time.Second,      // Health check interval for connections
+
+		// Timing Constants - Optimized for quality change stability
+		DefaultSleepDuration:       100 * time.Millisecond, // Balanced polling interval
+		ShortSleepDuration:         10 * time.Millisecond,  // Balanced high-frequency polling
+		LongSleepDuration:          200 * time.Millisecond, // Balanced background task delay
+		DefaultTickerInterval:      100 * time.Millisecond, // Balanced periodic task interval
+		BufferUpdateInterval:       300 * time.Millisecond, // Faster buffer updates for quality changes
 		InputSupervisorTimeout:     5 * time.Second,        // Input monitoring timeout
 		OutputSupervisorTimeout:    5 * time.Second,        // Output monitoring timeout
-		BatchProcessingDelay:       10 * time.Millisecond,  // Batch processing delay
+		BatchProcessingDelay:       5 * time.Millisecond,   // Reduced batch processing delay
-		AdaptiveOptimizerStability: 10 * time.Second,       // Adaptive stability period
+		AdaptiveOptimizerStability: 5 * time.Second,        // Faster adaptive stability period
 
-		LatencyMonitorTarget: 50 * time.Millisecond, // Target latency for monitoring
+		LatencyMonitorTarget: 50 * time.Millisecond, // Balanced target latency for monitoring
 
-		// Adaptive Buffer Configuration
+		// Adaptive Buffer Configuration - Optimized for low latency
-		LowCPUThreshold:             0.20,
+		LowCPUThreshold:             0.30,
-		HighCPUThreshold:            0.60,
+		HighCPUThreshold:            0.70,
-		LowMemoryThreshold:          0.50,
+		LowMemoryThreshold:          0.60,
-		HighMemoryThreshold:         0.75,
+		HighMemoryThreshold:         0.80,
-		AdaptiveBufferTargetLatency: 20 * time.Millisecond,
+		AdaptiveBufferTargetLatency: 15 * time.Millisecond, // Reduced target latency
 
-		// Adaptive Buffer Size Configuration
+		// Adaptive Buffer Size Configuration - Optimized for quality change bursts
-		AdaptiveMinBufferSize:     3,  // Minimum 3 frames for stability
+		AdaptiveMinBufferSize:     16, // Higher minimum to handle bursts
-		AdaptiveMaxBufferSize:     20, // Maximum 20 frames for high load
+		AdaptiveMaxBufferSize:     64, // Higher maximum for quality changes
-		AdaptiveDefaultBufferSize: 6,  // Balanced buffer size (6 frames)
+		AdaptiveDefaultBufferSize: 32, // Higher default for stability
 
-		// Adaptive Optimizer Configuration
+		// Adaptive Optimizer Configuration - Faster response
-		CooldownPeriod:                 30 * time.Second,
+		CooldownPeriod:                 15 * time.Second,       // Reduced cooldown period
-		RollbackThreshold:              300 * time.Millisecond,
+		RollbackThreshold:              200 * time.Millisecond, // Lower rollback threshold
-		AdaptiveOptimizerLatencyTarget: 50 * time.Millisecond,
+		AdaptiveOptimizerLatencyTarget: 30 * time.Millisecond,  // Reduced latency target
 
-		// Latency Monitor Configuration
+		// Latency Monitor Configuration - More aggressive monitoring
-		MaxLatencyThreshold:         200 * time.Millisecond,
+		MaxLatencyThreshold:         150 * time.Millisecond, // Lower max latency threshold
-		JitterThreshold:             20 * time.Millisecond,
+		JitterThreshold:             15 * time.Millisecond,  // Reduced jitter threshold
-		LatencyOptimizationInterval: 5 * time.Second,
+		LatencyOptimizationInterval: 3 * time.Second,        // More frequent optimization
-		LatencyAdaptiveThreshold:    0.8,
+		LatencyAdaptiveThreshold:    0.7,                    // More aggressive adaptive threshold
 
 		// Microphone Contention Configuration
 		MicContentionTimeout: 200 * time.Millisecond,
@@ -532,12 +550,12 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		LatencyScalingFactor:    2.0, // Latency ratio scaling factor
 		OptimizerAggressiveness: 0.7, // Optimizer aggressiveness factor
 
-		// CGO Audio Processing Constants
+		// CGO Audio Processing Constants - Balanced for stability
-		CGOUsleepMicroseconds:   1000,         // 1000 microseconds (1ms) for CGO usleep calls
+		CGOUsleepMicroseconds:   1000,         // 1000 microseconds (1ms) for stable CGO usleep calls
 		CGOPCMBufferSize:        1920,         // 1920 samples for PCM buffer (max 2ch*960)
 		CGONanosecondsPerSecond: 1000000000.0, // 1000000000.0 for nanosecond conversions
 
-		// Frontend Constants
+		// Frontend Constants - Balanced for stability
 		FrontendOperationDebounceMS: 1000,  // 1000ms debounce for frontend operations
 		FrontendSyncDebounceMS:      1000,  // 1000ms debounce for sync operations
 		FrontendSampleRate:          48000, // 48000Hz sample rate for frontend audio
@@ -560,20 +578,20 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		ProcessMonitorFallbackClockHz: 1000.0, // 1000.0 Hz fallback clock
 		ProcessMonitorTraditionalHz:   100.0,  // 100.0 Hz traditional clock
 
-		// Batch Processing Constants
+		// Batch Processing Constants - Optimized for quality change bursts
-		BatchProcessorFramesPerBatch:         4,                    // 4 frames per batch
+		BatchProcessorFramesPerBatch:         16,                    // Larger batches for quality changes
-		BatchProcessorTimeout:                5 * time.Millisecond, // 5ms timeout
+		BatchProcessorTimeout:                20 * time.Millisecond, // Longer timeout for bursts
-		BatchProcessorMaxQueueSize:           16,                   // 16 max queue size for balanced memory/performance
+		BatchProcessorMaxQueueSize:           64,                    // Larger queue for quality changes
-		BatchProcessorAdaptiveThreshold:      0.8,                  // 0.8 threshold for adaptive batching (80% queue full)
+		BatchProcessorAdaptiveThreshold:      0.6,                   // Lower threshold for faster adaptation
-		BatchProcessorThreadPinningThreshold: 8,                    // 8 frames minimum for thread pinning optimization
+		BatchProcessorThreadPinningThreshold: 8,                     // Lower threshold for better performance
 
-		// Output Streaming Constants
+		// Output Streaming Constants - Balanced for stability
-		OutputStreamingFrameIntervalMS: 20, // 20ms frame interval (50 FPS)
+		OutputStreamingFrameIntervalMS: 20, // 20ms frame interval (50 FPS) for stability
 
 		// IPC Constants
 		IPCInitialBufferFrames: 500, // 500 frames for initial buffer
 
-		// Event Constants
+		// Event Constants - Balanced for stability
 		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
@@ -585,7 +603,7 @@ 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
+		// Input Processing Constants - Balanced for stability
 		InputProcessingTimeoutMS: 10, // 10ms processing timeout threshold
 
 		// Adaptive Buffer Constants
@@ -670,7 +688,7 @@ func DefaultAudioConfig() *AudioConfigConstants {
 }
 
 // Global configuration instance
-var audioConfigInstance = DefaultAudioConfig()
+var Config = DefaultAudioConfig()
 
 // UpdateConfig allows runtime configuration updates
 func UpdateConfig(newConfig *AudioConfigConstants) {
@@ -682,12 +700,12 @@ func UpdateConfig(newConfig *AudioConfigConstants) {
 		return
 	}
 
-	audioConfigInstance = newConfig
+	Config = 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 audioConfigInstance
+	return Config
 }

internal/audio/core_validation.go

@@ -55,12 +55,11 @@ func ValidateZeroCopyFrame(frame *ZeroCopyAudioFrame) error {
 	maxFrameSize := cachedMaxFrameSize
 	if maxFrameSize == 0 {
 		// Fallback: get from cache
-		cache := GetCachedConfig()
+		cache := Config
-		maxFrameSize = int(cache.maxAudioFrameSize.Load())
+		maxFrameSize = cache.MaxAudioFrameSize
 		if maxFrameSize == 0 {
-			// Last resort: update cache
+			// Last resort: use default
-			cache.Update()
+			maxFrameSize = cache.MaxAudioFrameSize
-			maxFrameSize = int(cache.maxAudioFrameSize.Load())
 		}
 		// Cache globally for next calls
 		cachedMaxFrameSize = maxFrameSize
@@ -80,8 +79,8 @@ func ValidateBufferSize(size int) error {
 	}
 
 	// Fast path: Check against cached max frame size
-	cache := GetCachedConfig()
+	cache := Config
-	maxFrameSize := int(cache.maxAudioFrameSize.Load())
+	maxFrameSize := cache.MaxAudioFrameSize
 
 	// Most common case: validating a buffer that's sized for audio frames
 	if maxFrameSize > 0 && size <= maxFrameSize {
@@ -89,7 +88,7 @@ func ValidateBufferSize(size int) error {
 	}
 
 	// Slower path: full validation against SocketMaxBuffer
-	config := GetConfig()
+	config := Config
 	// 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 {
@@ -107,8 +106,8 @@ func ValidateLatency(latency time.Duration) error {
 	}
 
 	// Fast path: check against cached max latency
-	cache := GetCachedConfig()
+	cache := Config
-	maxLatency := time.Duration(cache.maxLatency.Load())
+	maxLatency := time.Duration(cache.MaxLatency)
 
 	// If we have a valid cached value, use it
 	if maxLatency > 0 {
@@ -125,7 +124,7 @@ func ValidateLatency(latency time.Duration) error {
 	}
 
 	// Slower path: full validation with GetConfig()
-	config := GetConfig()
+	config := Config
 	minLatency := time.Millisecond // Minimum reasonable latency
 	if latency > 0 && latency < minLatency {
 		return fmt.Errorf("%w: latency %v below minimum %v",
@@ -142,9 +141,9 @@ 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 := GetCachedConfig()
+	cache := Config
-	minInterval := time.Duration(cache.minMetricsUpdateInterval.Load())
+	minInterval := time.Duration(cache.MinMetricsUpdateInterval)
-	maxInterval := time.Duration(cache.maxMetricsUpdateInterval.Load())
+	maxInterval := time.Duration(cache.MaxMetricsUpdateInterval)
 
 	// If we have valid cached values, use them
 	if minInterval > 0 && maxInterval > 0 {
@@ -160,7 +159,7 @@ func ValidateMetricsInterval(interval time.Duration) error {
 	}
 
 	// Slower path: full validation with GetConfig()
-	config := GetConfig()
+	config := Config
 	minInterval = config.MinMetricsUpdateInterval
 	maxInterval = config.MaxMetricsUpdateInterval
 	if interval < minInterval {
@@ -184,7 +183,7 @@ func ValidateAdaptiveBufferConfig(minSize, maxSize, defaultSize int) error {
 		return ErrInvalidBufferSize
 	}
 	// Validate against global limits
-	maxBuffer := GetConfig().SocketMaxBuffer
+	maxBuffer := Config.SocketMaxBuffer
 	if maxSize > maxBuffer {
 		return ErrInvalidBufferSize
 	}
@@ -194,7 +193,7 @@ func ValidateAdaptiveBufferConfig(minSize, maxSize, defaultSize int) error {
 // ValidateInputIPCConfig validates input IPC configuration
 func ValidateInputIPCConfig(sampleRate, channels, frameSize int) error {
 	// Use config values
-	config := GetConfig()
+	config := Config
 	minSampleRate := config.MinSampleRate
 	maxSampleRate := config.MaxSampleRate
 	maxChannels := config.MaxChannels
@@ -213,7 +212,7 @@ func ValidateInputIPCConfig(sampleRate, channels, frameSize int) error {
 // ValidateOutputIPCConfig validates output IPC configuration
 func ValidateOutputIPCConfig(sampleRate, channels, frameSize int) error {
 	// Use config values
-	config := GetConfig()
+	config := Config
 	minSampleRate := config.MinSampleRate
 	maxSampleRate := config.MaxSampleRate
 	maxChannels := config.MaxChannels
@@ -263,8 +262,8 @@ func ValidateSampleRate(sampleRate int) error {
 	}
 
 	// Fast path: Check against cached sample rate first
-	cache := GetCachedConfig()
+	cache := Config
-	cachedRate := int(cache.sampleRate.Load())
+	cachedRate := cache.SampleRate
 
 	// Most common case: validating against the current sample rate
 	if sampleRate == cachedRate {
@@ -272,7 +271,7 @@ func ValidateSampleRate(sampleRate int) error {
 	}
 
 	// Slower path: check against all valid rates
-	config := GetConfig()
+	config := Config
 	validRates := config.ValidSampleRates
 	for _, rate := range validRates {
 		if sampleRate == rate {
@@ -291,8 +290,8 @@ func ValidateChannelCount(channels int) error {
 	}
 
 	// Fast path: Check against cached channels first
-	cache := GetCachedConfig()
+	cache := Config
-	cachedChannels := int(cache.channels.Load())
+	cachedChannels := cache.Channels
 
 	// Most common case: validating against the current channel count
 	if channels == cachedChannels {
@@ -300,14 +299,13 @@ func ValidateChannelCount(channels int) error {
 	}
 
 	// Fast path: Check against cached max channels
-	cachedMaxChannels := int(cache.maxChannels.Load())
+	cachedMaxChannels := cache.MaxChannels
 	if cachedMaxChannels > 0 && channels <= cachedMaxChannels {
 		return nil
 	}
 
-	// Slow path: Update cache and validate
+	// Slow path: Use current config values
-	cache.Update()
+	updatedMaxChannels := cache.MaxChannels
-	updatedMaxChannels := int(cache.maxChannels.Load())
 	if channels > updatedMaxChannels {
 		return fmt.Errorf("%w: channel count %d exceeds maximum %d",
 			ErrInvalidChannels, channels, updatedMaxChannels)
@@ -323,9 +321,9 @@ func ValidateBitrate(bitrate int) error {
 	}
 
 	// Fast path: Check against cached bitrate values
-	cache := GetCachedConfig()
+	cache := Config
-	minBitrate := int(cache.minOpusBitrate.Load())
+	minBitrate := cache.MinOpusBitrate
-	maxBitrate := int(cache.maxOpusBitrate.Load())
+	maxBitrate := cache.MaxOpusBitrate
 
 	// If we have valid cached values, use them
 	if minBitrate > 0 && maxBitrate > 0 {
@@ -343,7 +341,7 @@ func ValidateBitrate(bitrate int) error {
 	}
 
 	// Slower path: full validation with GetConfig()
-	config := GetConfig()
+	config := Config
 	// Convert kbps to bps for comparison with config limits
 	bitrateInBps := bitrate * 1000
 	if bitrateInBps < config.MinOpusBitrate {
@@ -365,11 +363,11 @@ func ValidateFrameDuration(duration time.Duration) error {
 	}
 
 	// Fast path: Check against cached frame size first
-	cache := GetCachedConfig()
+	cache := Config
 
 	// Convert frameSize (samples) to duration for comparison
-	cachedFrameSize := int(cache.frameSize.Load())
+	cachedFrameSize := cache.FrameSize
-	cachedSampleRate := int(cache.sampleRate.Load())
+	cachedSampleRate := cache.SampleRate
 
 	// Only do this calculation if we have valid cached values
 	if cachedFrameSize > 0 && cachedSampleRate > 0 {
@@ -382,8 +380,8 @@ func ValidateFrameDuration(duration time.Duration) error {
 	}
 
 	// Fast path: Check against cached min/max frame duration
-	cachedMinDuration := time.Duration(cache.minFrameDuration.Load())
+	cachedMinDuration := time.Duration(cache.MinFrameDuration)
-	cachedMaxDuration := time.Duration(cache.maxFrameDuration.Load())
+	cachedMaxDuration := time.Duration(cache.MaxFrameDuration)
 
 	if cachedMinDuration > 0 && cachedMaxDuration > 0 {
 		if duration < cachedMinDuration {
@@ -397,10 +395,9 @@ func ValidateFrameDuration(duration time.Duration) error {
 		return nil
 	}
 
-	// Slow path: Update cache and validate
+	// Slow path: Use current config values
-	cache.Update()
+	updatedMinDuration := time.Duration(cache.MinFrameDuration)
-	updatedMinDuration := time.Duration(cache.minFrameDuration.Load())
+	updatedMaxDuration := time.Duration(cache.MaxFrameDuration)
-	updatedMaxDuration := time.Duration(cache.maxFrameDuration.Load())
 
 	if duration < updatedMinDuration {
 		return fmt.Errorf("%w: frame duration %v below minimum %v",
@@ -417,11 +414,11 @@ 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 := GetCachedConfig()
+	cache := Config
-	cachedSampleRate := int(cache.sampleRate.Load())
+	cachedSampleRate := cache.SampleRate
-	cachedChannels := int(cache.channels.Load())
+	cachedChannels := cache.Channels
-	cachedBitrate := int(cache.opusBitrate.Load()) / 1000 // Convert from bps to kbps
+	cachedBitrate := cache.OpusBitrate / 1000 // Convert from bps to kbps
-	cachedFrameSize := int(cache.frameSize.Load())
+	cachedFrameSize := cache.FrameSize
 
 	// Only do this calculation if we have valid cached values
 	if cachedSampleRate > 0 && cachedChannels > 0 && cachedBitrate > 0 && cachedFrameSize > 0 {
@@ -481,11 +478,11 @@ var cachedMaxFrameSize int
 // InitValidationCache initializes cached validation values with actual config
 func InitValidationCache() {
 	// Initialize the global cache variable for backward compatibility
-	config := GetConfig()
+	config := Config
 	cachedMaxFrameSize = config.MaxAudioFrameSize
 
-	// Update the global audio config cache
+	// Initialize the global audio config cache
-	GetCachedConfig().Update()
+	cachedMaxFrameSize = Config.MaxAudioFrameSize
 }
 
 // ValidateAudioFrame validates audio frame data with cached max size for performance
@@ -502,12 +499,11 @@ func ValidateAudioFrame(data []byte) error {
 	maxSize := cachedMaxFrameSize
 	if maxSize == 0 {
 		// Fallback: get from cache only if global cache not initialized
-		cache := GetCachedConfig()
+		cache := Config
-		maxSize = int(cache.maxAudioFrameSize.Load())
+		maxSize = cache.MaxAudioFrameSize
 		if maxSize == 0 {
-			// Last resort: update cache and get fresh value
+			// Last resort: get fresh value
-			cache.Update()
+			maxSize = cache.MaxAudioFrameSize
-			maxSize = int(cache.maxAudioFrameSize.Load())
 		}
 		// Cache the value globally for next calls
 		cachedMaxFrameSize = maxSize

internal/audio/goroutine_pool.go

@@ -255,7 +255,7 @@ func GetAudioProcessorPool() *GoroutinePool {
 	}
 
 	globalAudioProcessorInitOnce.Do(func() {
-		config := GetConfig()
+		config := Config
 		newPool := NewGoroutinePool(
 			"audio-processor",
 			config.MaxAudioProcessorWorkers,
@@ -277,7 +277,7 @@ func GetAudioReaderPool() *GoroutinePool {
 	}
 
 	globalAudioReaderInitOnce.Do(func() {
-		config := GetConfig()
+		config := Config
 		newPool := NewGoroutinePool(
 			"audio-reader",
 			config.MaxAudioReaderWorkers,

internal/audio/input_microphone_manager.go

@@ -108,7 +108,7 @@ func (aim *AudioInputManager) WriteOpusFrame(frame []byte) error {
 	processingTime := time.Since(startTime)
 
 	// Log high latency warnings
-	if processingTime > time.Duration(GetConfig().InputProcessingTimeoutMS)*time.Millisecond {
+	if processingTime > time.Duration(Config.InputProcessingTimeoutMS)*time.Millisecond {
 		latencyMs := float64(processingTime.Milliseconds())
 		aim.logger.Warn().
 			Float64("latency_ms", latencyMs).
@@ -149,7 +149,7 @@ func (aim *AudioInputManager) WriteOpusFrameZeroCopy(frame *ZeroCopyAudioFrame)
 	processingTime := time.Since(startTime)
 
 	// Log high latency warnings
-	if processingTime > time.Duration(GetConfig().InputProcessingTimeoutMS)*time.Millisecond {
+	if processingTime > time.Duration(Config.InputProcessingTimeoutMS)*time.Millisecond {
 		latencyMs := float64(processingTime.Milliseconds())
 		aim.logger.Warn().
 			Float64("latency_ms", latencyMs).

internal/audio/input_server_main.go

@@ -107,7 +107,7 @@ func RunAudioInputServer() error {
 	server.Stop()
 
 	// Give some time for cleanup
-	time.Sleep(GetConfig().DefaultSleepDuration)
+	time.Sleep(Config.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:            GetConfig().InputSupervisorTimeout,
+		Timeout:            Config.InputSupervisorTimeout,
 		EnableRestart:      false, // Input supervisor doesn't restart
 		MaxRestartAttempts: 0,
 		RestartWindow:      0,
@@ -164,7 +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(GetConfig().InputSupervisorTimeout):
+	case <-time.After(Config.InputSupervisorTimeout):
 		ais.logger.Warn().Str("component", "audio-input-supervisor").Msg("component did not stop gracefully, forcing termination")
 		ais.forceKillProcess("audio input server")
 	}
@@ -190,7 +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(GetConfig().DefaultSleepDuration)
+	time.Sleep(Config.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, GetConfig().MaxFrameSize),
+			data: make([]byte, 0, Config.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, GetConfig().MaxFrameSize),
+			data: make([]byte, 0, Config.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, GetConfig().MaxFrameSize),
+			data: make([]byte, 0, Config.MaxFrameSize),
 		}
 	}
 }
@@ -149,7 +149,7 @@ func WriteIPCMessage(conn net.Conn, msg IPCMessage, pool *GenericMessagePool, dr
 	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(GetConfig().WriteTimeout); deadline.After(time.Now()) {
+	if deadline := time.Now().Add(Config.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    = GetConfig().MaxFrameSize    // Maximum Opus frame size
+	maxFrameSize    = Config.MaxFrameSize    // Maximum Opus frame size
-	messagePoolSize = GetConfig().MessagePoolSize // Pre-allocated message pool size
+	messagePoolSize = Config.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 * GetConfig().PoolGrowthMultiplier // Allow growth up to 2x
+		globalMessagePool.maxPoolSize = messagePoolSize * Config.PoolGrowthMultiplier // Allow growth up to 2x
 		globalMessagePool.preallocated = make([]*OptimizedIPCMessage, 0, preallocSize)
 
 		// Pre-allocate messages for immediate use
@@ -378,7 +378,7 @@ func (ais *AudioInputServer) handleConnection(conn net.Conn) {
 			if ais.conn == nil {
 				return
 			}
-			time.Sleep(GetConfig().DefaultSleepDuration)
+			time.Sleep(Config.DefaultSleepDuration)
 		}
 	}
 }
@@ -499,11 +499,11 @@ func (ais *AudioInputServer) processOpusFrame(data []byte) error {
 	}
 
 	// Get cached config once - avoid repeated calls and locking
-	cache := GetCachedConfig()
+	cache := Config
 	// Skip cache expiry check in hotpath - background updates handle this
 
 	// Get a PCM buffer from the pool for optimized decode-write
-	pcmBuffer := GetBufferFromPool(cache.GetMaxPCMBufferSize())
+	pcmBuffer := GetBufferFromPool(cache.MaxPCMBufferSize)
 	defer ReturnBufferToPool(pcmBuffer)
 
 	// Direct CGO call - avoid wrapper function overhead
@@ -646,9 +646,9 @@ func (aic *AudioInputClient) Connect() error {
 			return nil
 		}
 		// Exponential backoff starting from config
-		backoffStart := GetConfig().BackoffStart
+		backoffStart := Config.BackoffStart
 		delay := time.Duration(backoffStart.Nanoseconds()*(1<<uint(i/3))) * time.Nanosecond
-		maxDelay := GetConfig().MaxRetryDelay
+		maxDelay := Config.MaxRetryDelay
 		if delay > maxDelay {
 			delay = maxDelay
 		}
@@ -911,10 +911,10 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 		// Enhanced error tracking and recovery
 		var consecutiveErrors int
 		var lastErrorTime time.Time
-		maxConsecutiveErrors := GetConfig().MaxConsecutiveErrors
+		maxConsecutiveErrors := Config.MaxConsecutiveErrors
-		errorResetWindow := GetConfig().RestartWindow // Use existing restart window
+		errorResetWindow := Config.RestartWindow // Use existing restart window
-		baseBackoffDelay := GetConfig().RetryDelay
+		baseBackoffDelay := Config.RetryDelay
-		maxBackoffDelay := GetConfig().MaxRetryDelay
+		maxBackoffDelay := Config.MaxRetryDelay
 
 		logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
 
@@ -1025,7 +1025,7 @@ 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 := GetConfig()
+		config := Config
 		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
 		if useThreadOptimizations {
@@ -1137,7 +1137,7 @@ func (ais *AudioInputServer) processMessageWithRecovery(msg *InputIPCMessage, lo
 	select {
 	case processChan <- msg:
 		return nil
-	case <-time.After(GetConfig().WriteTimeout):
+	case <-time.After(Config.WriteTimeout):
 		// Processing queue full and timeout reached, drop frame
 		atomic.AddInt64(&ais.droppedFrames, 1)
 		return fmt.Errorf("processing queue timeout")
@@ -1156,7 +1156,7 @@ 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 := GetConfig()
+		config := Config
 		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
 		if useThreadOptimizations {
@@ -1167,11 +1167,11 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 		}
 
 		defer ais.wg.Done()
-		ticker := time.NewTicker(GetConfig().DefaultTickerInterval)
+		ticker := time.NewTicker(Config.DefaultTickerInterval)
 		defer ticker.Stop()
 
 		// Buffer size update ticker (less frequent)
-		bufferUpdateTicker := time.NewTicker(GetConfig().BufferUpdateInterval)
+		bufferUpdateTicker := time.NewTicker(Config.BufferUpdateInterval)
 		defer bufferUpdateTicker.Stop()
 
 		for {
@@ -1330,7 +1330,7 @@ func (mp *MessagePool) GetMessagePoolStats() MessagePoolStats {
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
+		hitRate = float64(hitCount) / float64(totalRequests) * Config.PercentageMultiplier
 	}
 
 	// Calculate channel pool size

internal/audio/ipc_output.go

@@ -24,7 +24,7 @@ const (
 // Methods are now inherited from UnifiedIPCMessage
 
 // Global shared message pool for output IPC client header reading
-var globalOutputClientMessagePool = NewGenericMessagePool(GetConfig().OutputMessagePoolSize)
+var globalOutputClientMessagePool = NewGenericMessagePool(Config.OutputMessagePoolSize)
 
 // AudioOutputServer is now an alias for UnifiedAudioServer
 type AudioOutputServer = UnifiedAudioServer
@@ -95,7 +95,7 @@ func (c *AudioOutputClient) ReceiveFrame() ([]byte, error) {
 	}
 
 	size := binary.LittleEndian.Uint32(optMsg.header[5:9])
-	maxFrameSize := GetConfig().OutputMaxFrameSize
+	maxFrameSize := Config.OutputMaxFrameSize
 	if int(size) > maxFrameSize {
 		return nil, fmt.Errorf("received frame size validation failed: got %d bytes, maximum allowed %d bytes", size, maxFrameSize)
 	}

internal/audio/ipc_unified.go

@@ -4,6 +4,7 @@ import (
 	"encoding/binary"
 	"fmt"
 	"io"
+	"math"
 	"net"
 	"os"
 	"path/filepath"
@@ -17,8 +18,8 @@ import (
 
 // Unified IPC constants
 var (
-	outputMagicNumber uint32 = GetConfig().OutputMagicNumber // "JKOU" (JetKVM Output)
+	outputMagicNumber uint32 = Config.OutputMagicNumber // "JKOU" (JetKVM Output)
-	inputMagicNumber  uint32 = GetConfig().InputMagicNumber  // "JKMI" (JetKVM Microphone Input)
+	inputMagicNumber  uint32 = Config.InputMagicNumber  // "JKMI" (JetKVM Microphone Input)
 	outputSocketName         = "audio_output.sock"
 	inputSocketName          = "audio_input.sock"
 	headerSize               = 17 // Fixed header size: 4+1+4+8 bytes
@@ -144,8 +145,8 @@ func NewUnifiedAudioServer(isInput bool) (*UnifiedAudioServer, error) {
 		logger:             logger,
 		socketPath:         socketPath,
 		magicNumber:        magicNumber,
-		messageChan:        make(chan *UnifiedIPCMessage, GetConfig().ChannelBufferSize),
+		messageChan:        make(chan *UnifiedIPCMessage, Config.ChannelBufferSize),
-		processChan:        make(chan *UnifiedIPCMessage, GetConfig().ChannelBufferSize),
+		processChan:        make(chan *UnifiedIPCMessage, Config.ChannelBufferSize),
 		socketBufferConfig: DefaultSocketBufferConfig(),
 		latencyMonitor:     nil,
 		adaptiveOptimizer:  nil,
@@ -311,7 +312,7 @@ func (s *UnifiedAudioServer) readMessage(conn net.Conn) (*UnifiedIPCMessage, err
 	timestamp := int64(binary.LittleEndian.Uint64(header[9:17]))
 
 	// Validate length
-	if length > uint32(GetConfig().MaxFrameSize) {
+	if length > uint32(Config.MaxFrameSize) {
 		return nil, fmt.Errorf("message too large: %d bytes", length)
 	}
 
@@ -339,7 +340,10 @@ func (s *UnifiedAudioServer) SendFrame(frame []byte) error {
 	defer s.mtx.Unlock()
 
 	if !s.running || s.conn == nil {
-		return fmt.Errorf("no client connected")
+		// Silently drop frames when no client is 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()
@@ -398,7 +402,7 @@ func (s *UnifiedAudioServer) writeMessage(conn net.Conn, msg *UnifiedIPCMessage)
 
 // UnifiedAudioClient provides common functionality for both input and output clients
 type UnifiedAudioClient struct {
-	// Atomic fields first for ARM32 alignment
+	// Atomic counters for frame statistics
 	droppedFrames int64 // Atomic counter for dropped frames
 	totalFrames   int64 // Atomic counter for total frames
 
@@ -409,6 +413,13 @@ 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
@@ -430,10 +441,12 @@ 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(GetConfig().MaxFrameSize),
+		bufferPool:      NewAudioBufferPool(Config.MaxFrameSize),
+		autoReconnect:   true, // Enable automatic reconnection by default
+		stopHealthCheck: make(chan struct{}),
 	}
 }
 
@@ -453,32 +466,46 @@ func (c *UnifiedAudioClient) Connect() error {
 	}
 
 	// Try connecting multiple times as the server might not be ready
-	// Reduced retry count and delay for faster startup
+	// Use configurable retry parameters for better control
-	for i := 0; i < 10; i++ {
+	maxAttempts := Config.MaxConnectionAttempts
-		conn, err := net.Dial("unix", c.socketPath)
+	initialDelay := Config.ConnectionRetryDelay
+	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)
-			c.logger.Info().Str("socket_path", c.socketPath).Msg("Connected to server")
+			atomic.StoreInt64(&c.connectionErrors, 0)
+			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
+
-		backoffStart := GetConfig().BackoffStart
+		// Log connection attempt failure
-		delay := time.Duration(backoffStart.Nanoseconds()*(1<<uint(i/3))) * time.Nanosecond
+		c.logger.Debug().Err(err).Str("socket_path", c.socketPath).Int("attempt", i+1).Int("max_attempts", maxAttempts).Msg("Connection attempt failed")
-		maxDelay := GetConfig().MaxRetryDelay
+
-		if delay > maxDelay {
+		// Don't sleep after the last attempt
-			delay = maxDelay
+		if i < maxAttempts-1 {
+			// 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 10 attempts")
+	return fmt.Errorf("failed to connect to audio server after %d attempts", Config.MaxConnectionAttempts)
 }
 
 // Disconnect disconnects the client from the server
@@ -492,6 +519,9 @@ func (c *UnifiedAudioClient) Disconnect() {
 
 	c.running = false
 
+	// Stop health check monitoring
+	c.stopHealthCheckMonitoring()
+
 	if c.conn != nil {
 		c.conn.Close()
 		c.conn = nil
@@ -511,7 +541,122 @@ func (c *UnifiedAudioClient) IsConnected() bool {
 func (c *UnifiedAudioClient) GetFrameStats() (total, dropped int64) {
 	total = atomic.LoadInt64(&c.totalFrames)
 	dropped = atomic.LoadInt64(&c.droppedFrames)
-	return total, dropped
+	return
+}
+
+// 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

internal/audio/mgmt_input_ipc_manager.go

@@ -63,9 +63,9 @@ func (aim *AudioInputIPCManager) Start() error {
 	}
 
 	config := InputIPCConfig{
-		SampleRate: GetConfig().InputIPCSampleRate,
+		SampleRate: Config.InputIPCSampleRate,
-		Channels:   GetConfig().InputIPCChannels,
+		Channels:   Config.InputIPCChannels,
-		FrameSize:  GetConfig().InputIPCFrameSize,
+		FrameSize:  Config.InputIPCFrameSize,
 	}
 
 	// Validate configuration before using it
@@ -80,7 +80,7 @@ func (aim *AudioInputIPCManager) Start() error {
 	}
 
 	// Wait for subprocess readiness
-	time.Sleep(GetConfig().LongSleepDuration)
+	time.Sleep(Config.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: GetConfig().SampleRate,
+		SampleRate: Config.SampleRate,
-		Channels:   GetConfig().Channels,
+		Channels:   Config.Channels,
-		FrameSize:  int(GetConfig().AudioQualityMediumFrameSize.Milliseconds()),
+		FrameSize:  int(Config.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(GetConfig().MicContentionTimeout)
+		manager := NewMicrophoneContentionManager(Config.MicContentionTimeout)
 		atomic.StorePointer(&globalMicContentionManager, unsafe.Pointer(manager))
 		return manager
 	}
@@ -115,7 +115,7 @@ func GetMicrophoneContentionManager() *MicrophoneContentionManager {
 		return (*MicrophoneContentionManager)(ptr)
 	}
 
-	return NewMicrophoneContentionManager(GetConfig().MicContentionTimeout)
+	return NewMicrophoneContentionManager(Config.MicContentionTimeout)
 }
 
 func TryMicrophoneOperation() OperationResult {

internal/audio/monitor_adaptive_optimizer.go

@@ -64,10 +64,10 @@ type OptimizerConfig struct {
 func DefaultOptimizerConfig() OptimizerConfig {
 	return OptimizerConfig{
 		MaxOptimizationLevel: 8,
-		CooldownPeriod:       GetConfig().CooldownPeriod,
+		CooldownPeriod:       Config.CooldownPeriod,
-		Aggressiveness:       GetConfig().OptimizerAggressiveness,
+		Aggressiveness:       Config.OptimizerAggressiveness,
-		RollbackThreshold:    GetConfig().RollbackThreshold,
+		RollbackThreshold:    Config.RollbackThreshold,
-		StabilityPeriod:      GetConfig().AdaptiveOptimizerStability,
+		StabilityPeriod:      Config.AdaptiveOptimizerStability,
 
 		// Adaptive interval defaults
 		MinOptimizationInterval: 100 * time.Millisecond, // High stability: check every 100ms
@@ -142,7 +142,7 @@ func (ao *AdaptiveOptimizer) handleLatencyOptimization(metrics LatencyMetrics) e
 // 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
+	latencyRatio := float64(metrics.Current) / float64(Config.AdaptiveOptimizerLatencyTarget) // 50ms target
 
 	// Adjust based on trend
 	switch metrics.Trend {
@@ -158,7 +158,7 @@ func (ao *AdaptiveOptimizer) calculateTargetOptimizationLevel(metrics LatencyMet
 	latencyRatio *= ao.config.Aggressiveness
 
 	// Convert to optimization level
-	targetLevel := int64(latencyRatio * GetConfig().LatencyScalingFactor) // Scale to 0-10 range
+	targetLevel := int64(latencyRatio * Config.LatencyScalingFactor) // Scale to 0-10 range
 	if targetLevel > int64(ao.config.MaxOptimizationLevel) {
 		targetLevel = int64(ao.config.MaxOptimizationLevel)
 	}

internal/audio/monitor_goroutine.go

@@ -126,7 +126,7 @@ func (gm *GoroutineMonitor) GetGoroutineStats() map[string]interface{} {
 // GetGoroutineMonitor returns the global goroutine monitor instance
 func GetGoroutineMonitor() *GoroutineMonitor {
 	if globalGoroutineMonitor == nil {
-		globalGoroutineMonitor = NewGoroutineMonitor(GetConfig().GoroutineMonitorInterval)
+		globalGoroutineMonitor = NewGoroutineMonitor(Config.GoroutineMonitorInterval)
 	}
 	return globalGoroutineMonitor
 }

internal/audio/monitor_latency.go

@@ -81,7 +81,7 @@ const (
 
 // DefaultLatencyConfig returns a sensible default configuration
 func DefaultLatencyConfig() LatencyConfig {
-	config := GetConfig()
+	config := Config
 	return LatencyConfig{
 		TargetLatency:        config.LatencyMonitorTarget,
 		MaxLatency:           config.MaxLatencyThreshold,

internal/audio/monitor_process.go

@@ -16,26 +16,26 @@ import (
 // Variables for process monitoring (using configuration)
 var (
 	// System constants
-	maxCPUPercent     = GetConfig().MaxCPUPercent
+	maxCPUPercent     = Config.MaxCPUPercent
-	minCPUPercent     = GetConfig().MinCPUPercent
+	minCPUPercent     = Config.MinCPUPercent
-	defaultClockTicks = GetConfig().DefaultClockTicks
+	defaultClockTicks = Config.DefaultClockTicks
-	defaultMemoryGB   = GetConfig().DefaultMemoryGB
+	defaultMemoryGB   = Config.DefaultMemoryGB
 
 	// Monitoring thresholds
-	maxWarmupSamples = GetConfig().MaxWarmupSamples
+	maxWarmupSamples = Config.MaxWarmupSamples
-	warmupCPUSamples = GetConfig().WarmupCPUSamples
+	warmupCPUSamples = Config.WarmupCPUSamples
 
 	// Channel buffer size
-	metricsChannelBuffer = GetConfig().MetricsChannelBuffer
+	metricsChannelBuffer = Config.MetricsChannelBuffer
 
 	// Clock tick detection ranges
-	minValidClockTicks = float64(GetConfig().MinValidClockTicks)
+	minValidClockTicks = float64(Config.MinValidClockTicks)
-	maxValidClockTicks = float64(GetConfig().MaxValidClockTicks)
+	maxValidClockTicks = float64(Config.MaxValidClockTicks)
 )
 
 // Variables for process monitoring
 var (
-	pageSize = GetConfig().PageSize
+	pageSize = Config.PageSize
 )
 
 // ProcessMetrics represents CPU and memory usage metrics for a process
@@ -233,7 +233,7 @@ func (pm *ProcessMonitor) collectMetrics(pid int, state *processState) (ProcessM
 
 	// Calculate memory percentage (RSS / total system memory)
 	if totalMem := pm.getTotalMemory(); totalMem > 0 {
-		metric.MemoryPercent = float64(metric.MemoryRSS) / float64(totalMem) * GetConfig().PercentageMultiplier
+		metric.MemoryPercent = float64(metric.MemoryRSS) / float64(totalMem) * Config.PercentageMultiplier
 	}
 
 	// Update state for next calculation
@@ -283,7 +283,7 @@ func (pm *ProcessMonitor) calculateCPUPercent(totalCPUTime int64, state *process
 		// Convert from clock ticks to seconds using actual system clock ticks
 		clockTicks := pm.getClockTicks()
 		cpuSeconds := cpuDelta / clockTicks
-		cpuPercent := (cpuSeconds / timeDelta) * GetConfig().PercentageMultiplier
+		cpuPercent := (cpuSeconds / timeDelta) * Config.PercentageMultiplier
 
 		// Apply bounds
 		if cpuPercent > maxCPUPercent {
@@ -335,7 +335,7 @@ func (pm *ProcessMonitor) getClockTicks() float64 {
 					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)
+							hz := Config.CGONanosecondsPerSecond / float64(period)
 							if hz >= minValidClockTicks && hz <= maxValidClockTicks {
 								pm.clockTicks = hz
 								return
@@ -363,7 +363,7 @@ 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)
+			pm.totalMemory = int64(defaultMemoryGB) * int64(Config.ProcessMonitorKBToBytes) * int64(Config.ProcessMonitorKBToBytes) * int64(Config.ProcessMonitorKBToBytes)
 			return
 		}
 		defer file.Close()
@@ -375,14 +375,14 @@ func (pm *ProcessMonitor) getTotalMemory() int64 {
 				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)
+						pm.totalMemory = kb * int64(Config.ProcessMonitorKBToBytes)
 						return
 					}
 				}
 				break
 			}
 		}
-		pm.totalMemory = int64(defaultMemoryGB) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes) * int64(GetConfig().ProcessMonitorKBToBytes) // Fallback
+		pm.totalMemory = int64(defaultMemoryGB) * int64(Config.ProcessMonitorKBToBytes) * int64(Config.ProcessMonitorKBToBytes) * int64(Config.ProcessMonitorKBToBytes) // Fallback
 	})
 	return pm.totalMemory
 }

internal/audio/output_server_main.go

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

internal/audio/output_streaming.go

@@ -84,9 +84,9 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 		buffer := make([]byte, GetMaxAudioFrameSize())
 
 		consecutiveErrors := 0
-		maxConsecutiveErrors := GetConfig().MaxConsecutiveErrors
+		maxConsecutiveErrors := Config.MaxConsecutiveErrors
-		errorBackoffDelay := GetConfig().RetryDelay
+		errorBackoffDelay := Config.RetryDelay
-		maxErrorBackoff := GetConfig().MaxRetryDelay
+		maxErrorBackoff := Config.MaxRetryDelay
 
 		for {
 			select {
@@ -123,18 +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) * GetConfig().BackoffMultiplier)
+							errorBackoffDelay = time.Duration(float64(errorBackoffDelay) * Config.BackoffMultiplier)
 							if errorBackoffDelay > maxErrorBackoff {
 								errorBackoffDelay = maxErrorBackoff
 							}
 						} else {
 							getOutputStreamingLogger().Info().Msg("Audio system reinitialized successfully")
 							consecutiveErrors = 0
-							errorBackoffDelay = GetConfig().RetryDelay // Reset backoff
+							errorBackoffDelay = Config.RetryDelay // Reset backoff
 						}
 					} else {
 						// Brief delay for transient errors
-						time.Sleep(GetConfig().ShortSleepDuration)
+						time.Sleep(Config.ShortSleepDuration)
 					}
 					continue
 				}
@@ -142,7 +142,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 				// Success - reset error counters
 				if consecutiveErrors > 0 {
 					consecutiveErrors = 0
-					errorBackoffDelay = GetConfig().RetryDelay
+					errorBackoffDelay = Config.RetryDelay
 				}
 
 				if n > 0 {
@@ -164,7 +164,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
 					RecordFrameReceived(n)
 				}
 				// Small delay to prevent busy waiting
-				time.Sleep(GetConfig().ShortSleepDuration)
+				time.Sleep(Config.ShortSleepDuration)
 			}
 		}
 	}()
@@ -185,6 +185,6 @@ func StopAudioOutputStreaming() {
 
 	// Wait for streaming to stop
 	for atomic.LoadInt32(&outputStreamingRunning) == 1 {
-		time.Sleep(GetConfig().ShortSleepDuration)
+		time.Sleep(Config.ShortSleepDuration)
 	}
 }

internal/audio/output_supervisor.go

@@ -19,19 +19,19 @@ const (
 
 // Restart configuration is now retrieved from centralized config
 func getMaxRestartAttempts() int {
-	return GetConfig().MaxRestartAttempts
+	return Config.MaxRestartAttempts
 }
 
 func getRestartWindow() time.Duration {
-	return GetConfig().RestartWindow
+	return Config.RestartWindow
 }
 
 func getRestartDelay() time.Duration {
-	return GetConfig().RestartDelay
+	return Config.RestartDelay
 }
 
 func getMaxRestartDelay() time.Duration {
-	return GetConfig().MaxRestartDelay
+	return Config.MaxRestartDelay
 }
 
 // AudioOutputSupervisor manages the audio output server subprocess lifecycle
@@ -145,7 +145,7 @@ func (s *AudioOutputSupervisor) Stop() {
 	select {
 	case <-s.processDone:
 		s.logger.Info().Str("component", AudioOutputSupervisorComponent).Msg("component stopped gracefully")
-	case <-time.After(GetConfig().OutputSupervisorTimeout):
+	case <-time.After(Config.OutputSupervisorTimeout):
 		s.logger.Warn().Str("component", AudioOutputSupervisorComponent).Msg("component did not stop gracefully, forcing termination")
 		s.forceKillProcess("audio output server")
 	}
@@ -158,7 +158,7 @@ func (s *AudioOutputSupervisor) supervisionLoop() {
 	// Configure supervision parameters
 	config := SupervisionConfig{
 		ProcessType:        "audio output server",
-		Timeout:            GetConfig().OutputSupervisorTimeout,
+		Timeout:            Config.OutputSupervisorTimeout,
 		EnableRestart:      true,
 		MaxRestartAttempts: getMaxRestartAttempts(),
 		RestartWindow:      getRestartWindow(),

internal/audio/quality_presets.go

@@ -39,7 +39,7 @@ var (
 
 // MaxAudioFrameSize is now retrieved from centralized config
 func GetMaxAudioFrameSize() int {
-	return GetConfig().MaxAudioFrameSize
+	return Config.MaxAudioFrameSize
 }
 
 // AudioQuality represents different audio quality presets
@@ -74,17 +74,17 @@ type AudioMetrics struct {
 var (
 	currentConfig = AudioConfig{
 		Quality:    AudioQualityMedium,
-		Bitrate:    GetConfig().AudioQualityMediumOutputBitrate,
+		Bitrate:    Config.AudioQualityMediumOutputBitrate,
-		SampleRate: GetConfig().SampleRate,
+		SampleRate: Config.SampleRate,
-		Channels:   GetConfig().Channels,
+		Channels:   Config.Channels,
-		FrameSize:  GetConfig().AudioQualityMediumFrameSize,
+		FrameSize:  Config.AudioQualityMediumFrameSize,
 	}
 	currentMicrophoneConfig = AudioConfig{
 		Quality:    AudioQualityMedium,
-		Bitrate:    GetConfig().AudioQualityMediumInputBitrate,
+		Bitrate:    Config.AudioQualityMediumInputBitrate,
-		SampleRate: GetConfig().SampleRate,
+		SampleRate: Config.SampleRate,
 		Channels:   1,
-		FrameSize:  GetConfig().AudioQualityMediumFrameSize,
+		FrameSize:  Config.AudioQualityMediumFrameSize,
 	}
 	metrics AudioMetrics
 )
@@ -96,24 +96,24 @@ var qualityPresets = map[AudioQuality]struct {
 	frameSize                   time.Duration
 }{
 	AudioQualityLow: {
-		outputBitrate: GetConfig().AudioQualityLowOutputBitrate, inputBitrate: GetConfig().AudioQualityLowInputBitrate,
+		outputBitrate: Config.AudioQualityLowOutputBitrate, inputBitrate: Config.AudioQualityLowInputBitrate,
-		sampleRate: GetConfig().AudioQualityLowSampleRate, channels: GetConfig().AudioQualityLowChannels,
+		sampleRate: Config.AudioQualityLowSampleRate, channels: Config.AudioQualityLowChannels,
-		frameSize: GetConfig().AudioQualityLowFrameSize,
+		frameSize: Config.AudioQualityLowFrameSize,
 	},
 	AudioQualityMedium: {
-		outputBitrate: GetConfig().AudioQualityMediumOutputBitrate, inputBitrate: GetConfig().AudioQualityMediumInputBitrate,
+		outputBitrate: Config.AudioQualityMediumOutputBitrate, inputBitrate: Config.AudioQualityMediumInputBitrate,
-		sampleRate: GetConfig().AudioQualityMediumSampleRate, channels: GetConfig().AudioQualityMediumChannels,
+		sampleRate: Config.AudioQualityMediumSampleRate, channels: Config.AudioQualityMediumChannels,
-		frameSize: GetConfig().AudioQualityMediumFrameSize,
+		frameSize: Config.AudioQualityMediumFrameSize,
 	},
 	AudioQualityHigh: {
-		outputBitrate: GetConfig().AudioQualityHighOutputBitrate, inputBitrate: GetConfig().AudioQualityHighInputBitrate,
+		outputBitrate: Config.AudioQualityHighOutputBitrate, inputBitrate: Config.AudioQualityHighInputBitrate,
-		sampleRate: GetConfig().SampleRate, channels: GetConfig().AudioQualityHighChannels,
+		sampleRate: Config.SampleRate, channels: Config.AudioQualityHighChannels,
-		frameSize: GetConfig().AudioQualityHighFrameSize,
+		frameSize: Config.AudioQualityHighFrameSize,
 	},
 	AudioQualityUltra: {
-		outputBitrate: GetConfig().AudioQualityUltraOutputBitrate, inputBitrate: GetConfig().AudioQualityUltraInputBitrate,
+		outputBitrate: Config.AudioQualityUltraOutputBitrate, inputBitrate: Config.AudioQualityUltraInputBitrate,
-		sampleRate: GetConfig().SampleRate, channels: GetConfig().AudioQualityUltraChannels,
+		sampleRate: Config.SampleRate, channels: Config.AudioQualityUltraChannels,
-		frameSize: GetConfig().AudioQualityUltraFrameSize,
+		frameSize: Config.AudioQualityUltraFrameSize,
 	},
 }
 
@@ -142,7 +142,7 @@ func GetMicrophoneQualityPresets() map[AudioQuality]AudioConfig {
 			Bitrate: preset.inputBitrate,
 			SampleRate: func() int {
 				if quality == AudioQualityLow {
-					return GetConfig().AudioQualityMicLowSampleRate
+					return Config.AudioQualityMicLowSampleRate
 				}
 				return preset.sampleRate
 			}(),
@@ -172,36 +172,36 @@ func SetAudioQuality(quality AudioQuality) {
 		var complexity, vbr, signalType, bandwidth, dtx int
 		switch quality {
 		case AudioQualityLow:
-			complexity = GetConfig().AudioQualityLowOpusComplexity
+			complexity = Config.AudioQualityLowOpusComplexity
-			vbr = GetConfig().AudioQualityLowOpusVBR
+			vbr = Config.AudioQualityLowOpusVBR
-			signalType = GetConfig().AudioQualityLowOpusSignalType
+			signalType = Config.AudioQualityLowOpusSignalType
-			bandwidth = GetConfig().AudioQualityLowOpusBandwidth
+			bandwidth = Config.AudioQualityLowOpusBandwidth
-			dtx = GetConfig().AudioQualityLowOpusDTX
+			dtx = Config.AudioQualityLowOpusDTX
 		case AudioQualityMedium:
-			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			complexity = Config.AudioQualityMediumOpusComplexity
-			vbr = GetConfig().AudioQualityMediumOpusVBR
+			vbr = Config.AudioQualityMediumOpusVBR
-			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			signalType = Config.AudioQualityMediumOpusSignalType
-			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			bandwidth = Config.AudioQualityMediumOpusBandwidth
-			dtx = GetConfig().AudioQualityMediumOpusDTX
+			dtx = Config.AudioQualityMediumOpusDTX
 		case AudioQualityHigh:
-			complexity = GetConfig().AudioQualityHighOpusComplexity
+			complexity = Config.AudioQualityHighOpusComplexity
-			vbr = GetConfig().AudioQualityHighOpusVBR
+			vbr = Config.AudioQualityHighOpusVBR
-			signalType = GetConfig().AudioQualityHighOpusSignalType
+			signalType = Config.AudioQualityHighOpusSignalType
-			bandwidth = GetConfig().AudioQualityHighOpusBandwidth
+			bandwidth = Config.AudioQualityHighOpusBandwidth
-			dtx = GetConfig().AudioQualityHighOpusDTX
+			dtx = Config.AudioQualityHighOpusDTX
 		case AudioQualityUltra:
-			complexity = GetConfig().AudioQualityUltraOpusComplexity
+			complexity = Config.AudioQualityUltraOpusComplexity
-			vbr = GetConfig().AudioQualityUltraOpusVBR
+			vbr = Config.AudioQualityUltraOpusVBR
-			signalType = GetConfig().AudioQualityUltraOpusSignalType
+			signalType = Config.AudioQualityUltraOpusSignalType
-			bandwidth = GetConfig().AudioQualityUltraOpusBandwidth
+			bandwidth = Config.AudioQualityUltraOpusBandwidth
-			dtx = GetConfig().AudioQualityUltraOpusDTX
+			dtx = Config.AudioQualityUltraOpusDTX
 		default:
 			// Use medium quality as fallback
-			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			complexity = Config.AudioQualityMediumOpusComplexity
-			vbr = GetConfig().AudioQualityMediumOpusVBR
+			vbr = Config.AudioQualityMediumOpusVBR
-			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			signalType = Config.AudioQualityMediumOpusSignalType
-			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			bandwidth = Config.AudioQualityMediumOpusBandwidth
-			dtx = GetConfig().AudioQualityMediumOpusDTX
+			dtx = Config.AudioQualityMediumOpusDTX
 		}
 
 		// Restart audio output subprocess with new OPUS configuration
@@ -256,7 +256,7 @@ func SetAudioQuality(quality AudioQuality) {
 			}
 		} else {
 			// Fallback to dynamic update if supervisor is not available
-			vbrConstraint := GetConfig().CGOOpusVBRConstraint
+			vbrConstraint := Config.CGOOpusVBRConstraint
 			if err := updateOpusEncoderParams(config.Bitrate*1000, complexity, vbr, vbrConstraint, signalType, bandwidth, dtx); err != nil {
 				logging.GetDefaultLogger().Error().Err(err).Msg("Failed to update OPUS encoder parameters")
 			}
@@ -287,36 +287,36 @@ func SetMicrophoneQuality(quality AudioQuality) {
 		var complexity, vbr, signalType, bandwidth, dtx int
 		switch quality {
 		case AudioQualityLow:
-			complexity = GetConfig().AudioQualityLowOpusComplexity
+			complexity = Config.AudioQualityLowOpusComplexity
-			vbr = GetConfig().AudioQualityLowOpusVBR
+			vbr = Config.AudioQualityLowOpusVBR
-			signalType = GetConfig().AudioQualityLowOpusSignalType
+			signalType = Config.AudioQualityLowOpusSignalType
-			bandwidth = GetConfig().AudioQualityLowOpusBandwidth
+			bandwidth = Config.AudioQualityLowOpusBandwidth
-			dtx = GetConfig().AudioQualityLowOpusDTX
+			dtx = Config.AudioQualityLowOpusDTX
 		case AudioQualityMedium:
-			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			complexity = Config.AudioQualityMediumOpusComplexity
-			vbr = GetConfig().AudioQualityMediumOpusVBR
+			vbr = Config.AudioQualityMediumOpusVBR
-			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			signalType = Config.AudioQualityMediumOpusSignalType
-			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			bandwidth = Config.AudioQualityMediumOpusBandwidth
-			dtx = GetConfig().AudioQualityMediumOpusDTX
+			dtx = Config.AudioQualityMediumOpusDTX
 		case AudioQualityHigh:
-			complexity = GetConfig().AudioQualityHighOpusComplexity
+			complexity = Config.AudioQualityHighOpusComplexity
-			vbr = GetConfig().AudioQualityHighOpusVBR
+			vbr = Config.AudioQualityHighOpusVBR
-			signalType = GetConfig().AudioQualityHighOpusSignalType
+			signalType = Config.AudioQualityHighOpusSignalType
-			bandwidth = GetConfig().AudioQualityHighOpusBandwidth
+			bandwidth = Config.AudioQualityHighOpusBandwidth
-			dtx = GetConfig().AudioQualityHighOpusDTX
+			dtx = Config.AudioQualityHighOpusDTX
 		case AudioQualityUltra:
-			complexity = GetConfig().AudioQualityUltraOpusComplexity
+			complexity = Config.AudioQualityUltraOpusComplexity
-			vbr = GetConfig().AudioQualityUltraOpusVBR
+			vbr = Config.AudioQualityUltraOpusVBR
-			signalType = GetConfig().AudioQualityUltraOpusSignalType
+			signalType = Config.AudioQualityUltraOpusSignalType
-			bandwidth = GetConfig().AudioQualityUltraOpusBandwidth
+			bandwidth = Config.AudioQualityUltraOpusBandwidth
-			dtx = GetConfig().AudioQualityUltraOpusDTX
+			dtx = Config.AudioQualityUltraOpusDTX
 		default:
 			// Use medium quality as fallback
-			complexity = GetConfig().AudioQualityMediumOpusComplexity
+			complexity = Config.AudioQualityMediumOpusComplexity
-			vbr = GetConfig().AudioQualityMediumOpusVBR
+			vbr = Config.AudioQualityMediumOpusVBR
-			signalType = GetConfig().AudioQualityMediumOpusSignalType
+			signalType = Config.AudioQualityMediumOpusSignalType
-			bandwidth = GetConfig().AudioQualityMediumOpusBandwidth
+			bandwidth = Config.AudioQualityMediumOpusBandwidth
-			dtx = GetConfig().AudioQualityMediumOpusDTX
+			dtx = Config.AudioQualityMediumOpusDTX
 		}
 
 		// Update audio input subprocess configuration dynamically without restart

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: GetConfig().SocketOptimalBuffer,
+		SendBufferSize: Config.SocketOptimalBuffer,
-		RecvBufferSize: GetConfig().SocketOptimalBuffer,
+		RecvBufferSize: Config.SocketOptimalBuffer,
 		Enabled:        true,
 	}
 }
@@ -27,8 +27,8 @@ func DefaultSocketBufferConfig() SocketBufferConfig {
 // HighLoadSocketBufferConfig returns configuration for high-load scenarios
 func HighLoadSocketBufferConfig() SocketBufferConfig {
 	return SocketBufferConfig{
-		SendBufferSize: GetConfig().SocketMaxBuffer,
+		SendBufferSize: Config.SocketMaxBuffer,
-		RecvBufferSize: GetConfig().SocketMaxBuffer,
+		RecvBufferSize: Config.SocketMaxBuffer,
 		Enabled:        true,
 	}
 }
@@ -123,8 +123,8 @@ func ValidateSocketBufferConfig(config SocketBufferConfig) error {
 		return nil
 	}
 
-	minBuffer := GetConfig().SocketMinBuffer
+	minBuffer := Config.SocketMinBuffer
-	maxBuffer := GetConfig().SocketMaxBuffer
+	maxBuffer := Config.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

@@ -366,17 +366,17 @@ func NewAudioBufferPool(bufferSize int) *AudioBufferPool {
 	// Validate buffer size parameter
 	if err := ValidateBufferSize(bufferSize); err != nil {
 		// Use default value on validation error
-		bufferSize = GetConfig().AudioFramePoolSize
+		bufferSize = Config.AudioFramePoolSize
 	}
 
 	// Enhanced preallocation strategy based on buffer size and system capacity
 	var preallocSize int
-	if bufferSize <= GetConfig().AudioFramePoolSize {
+	if bufferSize <= Config.AudioFramePoolSize {
 		// For smaller pools, use enhanced preallocation (40% instead of 20%)
-		preallocSize = GetConfig().PreallocPercentage * 2
+		preallocSize = Config.PreallocPercentage * 2
 	} else {
 		// For larger pools, use standard enhanced preallocation (30% instead of 10%)
-		preallocSize = (GetConfig().PreallocPercentage * 3) / 2
+		preallocSize = (Config.PreallocPercentage * 3) / 2
 	}
 
 	// Ensure minimum preallocation for better performance
@@ -594,9 +594,9 @@ func (p *AudioBufferPool) Put(buf []byte) {
 // Enhanced global buffer pools for different audio frame types with improved sizing
 var (
 	// Main audio frame pool with enhanced capacity
-	audioFramePool = NewAudioBufferPool(GetConfig().AudioFramePoolSize)
+	audioFramePool = NewAudioBufferPool(Config.AudioFramePoolSize)
 	// Control message pool with enhanced capacity for better throughput
-	audioControlPool = NewAudioBufferPool(512) // Increased from GetConfig().OutputHeaderSize to 512 for better control message handling
+	audioControlPool = NewAudioBufferPool(512) // Increased from Config.OutputHeaderSize to 512 for better control message handling
 )
 
 func GetAudioFrameBuffer() []byte {
@@ -628,7 +628,7 @@ func (p *AudioBufferPool) GetPoolStats() AudioBufferPoolDetailedStats {
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
+		hitRate = float64(hitCount) / float64(totalRequests) * Config.PercentageMultiplier
 	}
 
 	return AudioBufferPoolDetailedStats{

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", GetConfig().CGOOpusBitrate)
+	bitrate = getEnvInt("JETKVM_OPUS_BITRATE", Config.CGOOpusBitrate)
-	complexity = getEnvInt("JETKVM_OPUS_COMPLEXITY", GetConfig().CGOOpusComplexity)
+	complexity = getEnvInt("JETKVM_OPUS_COMPLEXITY", Config.CGOOpusComplexity)
-	vbr = getEnvInt("JETKVM_OPUS_VBR", GetConfig().CGOOpusVBR)
+	vbr = getEnvInt("JETKVM_OPUS_VBR", Config.CGOOpusVBR)
-	signalType = getEnvInt("JETKVM_OPUS_SIGNAL_TYPE", GetConfig().CGOOpusSignalType)
+	signalType = getEnvInt("JETKVM_OPUS_SIGNAL_TYPE", Config.CGOOpusSignalType)
-	bandwidth = getEnvInt("JETKVM_OPUS_BANDWIDTH", GetConfig().CGOOpusBandwidth)
+	bandwidth = getEnvInt("JETKVM_OPUS_BANDWIDTH", Config.CGOOpusBandwidth)
-	dtx = getEnvInt("JETKVM_OPUS_DTX", GetConfig().CGOOpusDTX)
+	dtx = getEnvInt("JETKVM_OPUS_DTX", Config.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 := GetConfig()
+	config := Config
 	config.CGOOpusBitrate = bitrate
 	config.CGOOpusComplexity = complexity
 	config.CGOOpusVBR = vbr

internal/audio/webrtc_relay.go

@@ -134,7 +134,7 @@ func (r *AudioRelay) relayLoop() {
 	defer r.wg.Done()
 	r.logger.Debug().Msg("Audio relay loop started")
 
-	var maxConsecutiveErrors = GetConfig().MaxConsecutiveErrors
+	var maxConsecutiveErrors = Config.MaxConsecutiveErrors
 	consecutiveErrors := 0
 
 	for {
@@ -153,7 +153,7 @@ func (r *AudioRelay) relayLoop() {
 					r.logger.Error().Int("consecutive_errors", consecutiveErrors).Int("max_errors", maxConsecutiveErrors).Msg("too many consecutive read errors, stopping audio relay")
 					return
 				}
-				time.Sleep(GetConfig().ShortSleepDuration)
+				time.Sleep(Config.ShortSleepDuration)
 				continue
 			}
 

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(GetConfig().EventTimeoutSeconds)*time.Second)
+	ctx, cancel := context.WithTimeout(subscriber.ctx, time.Duration(Config.EventTimeoutSeconds)*time.Second)
 	defer cancel()
 
 	err := wsjson.Write(ctx, subscriber.conn, event)

internal/audio/zero_copy.go

@@ -98,8 +98,8 @@ type ZeroCopyFramePool struct {
 // NewZeroCopyFramePool creates a new zero-copy frame pool
 func NewZeroCopyFramePool(maxFrameSize int) *ZeroCopyFramePool {
 	// Pre-allocate frames for immediate availability
-	preallocSizeBytes := GetConfig().PreallocSize
+	preallocSizeBytes := Config.PreallocSize
-	maxPoolSize := GetConfig().MaxPoolSize // Limit total pool size
+	maxPoolSize := Config.MaxPoolSize // Limit total pool size
 
 	// Calculate number of frames based on memory budget, not frame count
 	preallocFrameCount := preallocSizeBytes / maxFrameSize
@@ -327,7 +327,7 @@ func (p *ZeroCopyFramePool) GetZeroCopyPoolStats() ZeroCopyFramePoolStats {
 
 	var hitRate float64
 	if totalRequests > 0 {
-		hitRate = float64(hitCount) / float64(totalRequests) * GetConfig().PercentageMultiplier
+		hitRate = float64(hitCount) / float64(totalRequests) * Config.PercentageMultiplier
 	}
 
 	return ZeroCopyFramePoolStats{

main.go

@@ -58,7 +58,7 @@ func startAudioSubprocess() error {
 	audio.SetAudioInputSupervisor(audioInputSupervisor)
 
 	// Set default OPUS configuration for audio input supervisor (low quality for single-core RV1106)
-	config := audio.GetConfig()
+	config := audio.Config
 	audioInputSupervisor.SetOpusConfig(
 		config.AudioQualityLowInputBitrate*1000, // Convert kbps to bps
 		config.AudioQualityLowOpusComplexity,
@@ -81,7 +81,8 @@ func startAudioSubprocess() error {
 			// This prevents "no client connected" errors during quality changes
 			go func() {
 				// Give the audio output server time to initialize and start listening
-				time.Sleep(500 * time.Millisecond)
+				// Increased delay to reduce frame drops during connection establishment
+				time.Sleep(1 * time.Second)
 
 				// Start audio relay system for main process
 				// If there's an active WebRTC session, use its audio track