cloud.go

@@ -77,6 +77,23 @@ var (
 		},
 		[]string{"type", "source"},
 	)
+	metricConnectionPingDuration = promauto.NewHistogramVec(
+		prometheus.HistogramOpts{
+			Name: "jetkvm_connection_ping_duration_seconds",
+			Help: "The duration of the ping response",
+			Buckets: []float64{
+				0.1, 0.5, 1, 10,
+			},
+		},
+		[]string{"type", "source"},
+	)
+	metricConnectionTotalPingSentCount = promauto.NewCounterVec(
+		prometheus.CounterOpts{
+			Name: "jetkvm_connection_ping_sent_total",
+			Help: "The total number of pings sent to the connection",
+		},
+		[]string{"type", "source"},
+	)
 	metricConnectionTotalPingReceivedCount = promauto.NewCounterVec(
 		prometheus.CounterOpts{
 			Name: "jetkvm_connection_ping_received_total",
@@ -84,6 +101,13 @@ var (
 		},
 		[]string{"type", "source"},
 	)
+	metricConnectionSessionRequestCount = promauto.NewCounterVec(
+		prometheus.CounterOpts{
+			Name: "jetkvm_connection_session_requests_total",
+			Help: "The total number of session requests received",
+		},
+		[]string{"type", "source"},
+	)
 	metricConnectionSessionRequestDuration = promauto.NewHistogramVec(
 		prometheus.HistogramOpts{
 			Name: "jetkvm_connection_session_request_duration_seconds",

input_rpc_test.go

@@ -0,0 +1,560 @@
+package kvm
+
+import (
+	"testing"
+
+	"github.com/stretchr/testify/assert"
+)
+
+// Test validateFloat64Param function
+func TestValidateFloat64Param(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		paramName   string
+		methodName  string
+		min         float64
+		max         float64
+		expected    float64
+		expectError bool
+	}{
+		{
+			name:        "valid parameter",
+			params:      map[string]interface{}{"test": 50.0},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    50.0,
+			expectError: false,
+		},
+		{
+			name:        "parameter at minimum boundary",
+			params:      map[string]interface{}{"test": 0.0},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    0.0,
+			expectError: false,
+		},
+		{
+			name:        "parameter at maximum boundary",
+			params:      map[string]interface{}{"test": 100.0},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    100.0,
+			expectError: false,
+		},
+		{
+			name:        "parameter below minimum",
+			params:      map[string]interface{}{"test": -1.0},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    0,
+			expectError: true,
+		},
+		{
+			name:        "parameter above maximum",
+			params:      map[string]interface{}{"test": 101.0},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    0,
+			expectError: true,
+		},
+		{
+			name:        "wrong parameter type",
+			params:      map[string]interface{}{"test": "not a number"},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    0,
+			expectError: true,
+		},
+		{
+			name:        "missing parameter",
+			params:      map[string]interface{}{},
+			paramName:   "test",
+			methodName:  "testMethod",
+			min:         0,
+			max:         100,
+			expected:    0,
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			result, err := validateFloat64Param(tt.params, tt.paramName, tt.methodName, tt.min, tt.max)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+				assert.Equal(t, tt.expected, result)
+			}
+		})
+	}
+}
+
+// Test validateKeysArray function
+func TestValidateKeysArray(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		methodName  string
+		expected    []uint8
+		expectError bool
+	}{
+		{
+			name:        "valid keys array",
+			params:      map[string]interface{}{"keys": []interface{}{65.0, 66.0, 67.0}},
+			methodName:  "testMethod",
+			expected:    []uint8{65, 66, 67},
+			expectError: false,
+		},
+		{
+			name:        "empty keys array",
+			params:      map[string]interface{}{"keys": []interface{}{}},
+			methodName:  "testMethod",
+			expected:    []uint8{},
+			expectError: false,
+		},
+		{
+			name:        "maximum keys array",
+			params:      map[string]interface{}{"keys": []interface{}{1.0, 2.0, 3.0, 4.0, 5.0, 6.0}},
+			methodName:  "testMethod",
+			expected:    []uint8{1, 2, 3, 4, 5, 6},
+			expectError: false,
+		},
+		{
+			name:        "too many keys",
+			params:      map[string]interface{}{"keys": []interface{}{1.0, 2.0, 3.0, 4.0, 5.0, 6.0, 7.0}},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+		{
+			name:        "invalid key type",
+			params:      map[string]interface{}{"keys": []interface{}{"not a number"}},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+		{
+			name:        "key value out of range (negative)",
+			params:      map[string]interface{}{"keys": []interface{}{-1.0}},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+		{
+			name:        "key value out of range (too high)",
+			params:      map[string]interface{}{"keys": []interface{}{256.0}},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+		{
+			name:        "wrong parameter type",
+			params:      map[string]interface{}{"keys": "not an array"},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+		{
+			name:        "missing keys parameter",
+			params:      map[string]interface{}{},
+			methodName:  "testMethod",
+			expected:    nil,
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			result, err := validateKeysArray(tt.params, tt.methodName)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+				assert.Equal(t, tt.expected, result)
+			}
+		})
+	}
+}
+
+// Test handleKeyboardReportDirect function
+func TestHandleKeyboardReportDirect(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		expectError bool
+	}{
+		{
+			name: "valid keyboard report",
+			params: map[string]interface{}{
+				"modifier": 2.0,                       // Shift key
+				"keys":     []interface{}{65.0, 66.0}, // A, B keys
+			},
+			expectError: false,
+		},
+		{
+			name: "empty keys array",
+			params: map[string]interface{}{
+				"modifier": 0.0,
+				"keys":     []interface{}{},
+			},
+			expectError: false,
+		},
+		{
+			name: "invalid modifier",
+			params: map[string]interface{}{
+				"modifier": 256.0, // Out of range
+				"keys":     []interface{}{65.0},
+			},
+			expectError: true,
+		},
+		{
+			name: "invalid keys",
+			params: map[string]interface{}{
+				"modifier": 0.0,
+				"keys":     []interface{}{1.0, 2.0, 3.0, 4.0, 5.0, 6.0, 7.0}, // Too many keys
+			},
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			_, err := handleKeyboardReportDirect(tt.params)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+			}
+		})
+	}
+}
+
+// Test handleAbsMouseReportDirect function
+func TestHandleAbsMouseReportDirect(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		expectError bool
+	}{
+		{
+			name: "valid absolute mouse report",
+			params: map[string]interface{}{
+				"x":       1000.0,
+				"y":       500.0,
+				"buttons": 1.0, // Left button
+			},
+			expectError: false,
+		},
+		{
+			name: "boundary values",
+			params: map[string]interface{}{
+				"x":       0.0,
+				"y":       32767.0,
+				"buttons": 255.0,
+			},
+			expectError: false,
+		},
+		{
+			name: "invalid x coordinate",
+			params: map[string]interface{}{
+				"x":       -1.0, // Out of range
+				"y":       500.0,
+				"buttons": 0.0,
+			},
+			expectError: true,
+		},
+		{
+			name: "invalid y coordinate",
+			params: map[string]interface{}{
+				"x":       1000.0,
+				"y":       32768.0, // Out of range
+				"buttons": 0.0,
+			},
+			expectError: true,
+		},
+		{
+			name: "invalid buttons",
+			params: map[string]interface{}{
+				"x":       1000.0,
+				"y":       500.0,
+				"buttons": 256.0, // Out of range
+			},
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			_, err := handleAbsMouseReportDirect(tt.params)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+			}
+		})
+	}
+}
+
+// Test handleRelMouseReportDirect function
+func TestHandleRelMouseReportDirect(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		expectError bool
+	}{
+		{
+			name: "valid relative mouse report",
+			params: map[string]interface{}{
+				"dx":      10.0,
+				"dy":      -5.0,
+				"buttons": 2.0, // Right button
+			},
+			expectError: false,
+		},
+		{
+			name: "boundary values",
+			params: map[string]interface{}{
+				"dx":      -127.0,
+				"dy":      127.0,
+				"buttons": 0.0,
+			},
+			expectError: false,
+		},
+		{
+			name: "invalid dx",
+			params: map[string]interface{}{
+				"dx":      -128.0, // Out of range
+				"dy":      0.0,
+				"buttons": 0.0,
+			},
+			expectError: true,
+		},
+		{
+			name: "invalid dy",
+			params: map[string]interface{}{
+				"dx":      0.0,
+				"dy":      128.0, // Out of range
+				"buttons": 0.0,
+			},
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			_, err := handleRelMouseReportDirect(tt.params)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+			}
+		})
+	}
+}
+
+// Test handleWheelReportDirect function
+func TestHandleWheelReportDirect(t *testing.T) {
+	tests := []struct {
+		name        string
+		params      map[string]interface{}
+		expectError bool
+	}{
+		{
+			name: "valid wheel report",
+			params: map[string]interface{}{
+				"wheelY": 3.0,
+			},
+			expectError: false,
+		},
+		{
+			name: "boundary values",
+			params: map[string]interface{}{
+				"wheelY": -127.0,
+			},
+			expectError: false,
+		},
+		{
+			name: "invalid wheelY",
+			params: map[string]interface{}{
+				"wheelY": 128.0, // Out of range
+			},
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			_, err := handleWheelReportDirect(tt.params)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+			}
+		})
+	}
+}
+
+// Test handleInputRPCDirect function
+func TestHandleInputRPCDirect(t *testing.T) {
+	tests := []struct {
+		name        string
+		method      string
+		params      map[string]interface{}
+		expectError bool
+	}{
+		{
+			name:   "keyboard report",
+			method: "keyboardReport",
+			params: map[string]interface{}{
+				"modifier": 0.0,
+				"keys":     []interface{}{65.0},
+			},
+			expectError: false,
+		},
+		{
+			name:   "absolute mouse report",
+			method: "absMouseReport",
+			params: map[string]interface{}{
+				"x":       1000.0,
+				"y":       500.0,
+				"buttons": 1.0,
+			},
+			expectError: false,
+		},
+		{
+			name:   "relative mouse report",
+			method: "relMouseReport",
+			params: map[string]interface{}{
+				"dx":      10.0,
+				"dy":      -5.0,
+				"buttons": 2.0,
+			},
+			expectError: false,
+		},
+		{
+			name:   "wheel report",
+			method: "wheelReport",
+			params: map[string]interface{}{
+				"wheelY": 3.0,
+			},
+			expectError: false,
+		},
+		{
+			name:        "unknown method",
+			method:      "unknownMethod",
+			params:      map[string]interface{}{},
+			expectError: true,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			_, err := handleInputRPCDirect(tt.method, tt.params)
+			if tt.expectError {
+				assert.Error(t, err)
+			} else {
+				assert.NoError(t, err)
+			}
+		})
+	}
+}
+
+// Test isInputMethod function
+func TestIsInputMethod(t *testing.T) {
+	tests := []struct {
+		name     string
+		method   string
+		expected bool
+	}{
+		{
+			name:     "keyboard report method",
+			method:   "keyboardReport",
+			expected: true,
+		},
+		{
+			name:     "absolute mouse report method",
+			method:   "absMouseReport",
+			expected: true,
+		},
+		{
+			name:     "relative mouse report method",
+			method:   "relMouseReport",
+			expected: true,
+		},
+		{
+			name:     "wheel report method",
+			method:   "wheelReport",
+			expected: true,
+		},
+		{
+			name:     "non-input method",
+			method:   "someOtherMethod",
+			expected: false,
+		},
+		{
+			name:     "empty method",
+			method:   "",
+			expected: false,
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			result := isInputMethod(tt.method)
+			assert.Equal(t, tt.expected, result)
+		})
+	}
+}
+
+// Benchmark tests to verify performance improvements
+func BenchmarkValidateFloat64Param(b *testing.B) {
+	params := map[string]interface{}{"test": 50.0}
+	b.ResetTimer()
+	for i := 0; i < b.N; i++ {
+		_, _ = validateFloat64Param(params, "test", "benchmarkMethod", 0, 100)
+	}
+}
+
+func BenchmarkValidateKeysArray(b *testing.B) {
+	params := map[string]interface{}{"keys": []interface{}{65.0, 66.0, 67.0}}
+	b.ResetTimer()
+	for i := 0; i < b.N; i++ {
+		_, _ = validateKeysArray(params, "benchmarkMethod")
+	}
+}
+
+func BenchmarkHandleKeyboardReportDirect(b *testing.B) {
+	params := map[string]interface{}{
+		"modifier": 2.0,
+		"keys":     []interface{}{65.0, 66.0},
+	}
+	b.ResetTimer()
+	for i := 0; i < b.N; i++ {
+		_, _ = handleKeyboardReportDirect(params)
+	}
+}
+
+func BenchmarkHandleInputRPCDirect(b *testing.B) {
+	params := map[string]interface{}{
+		"modifier": 2.0,
+		"keys":     []interface{}{65.0, 66.0},
+	}
+	b.ResetTimer()
+	for i := 0; i < b.N; i++ {
+		_, _ = handleInputRPCDirect("keyboardReport", params)
+	}
+}

internal/audio/adaptive_buffer.go

@@ -152,6 +152,17 @@ func (abm *AdaptiveBufferManager) GetOutputBufferSize() int {
 
 // UpdateLatency updates the current latency measurement
 func (abm *AdaptiveBufferManager) UpdateLatency(latency time.Duration) {
+	// Use exponential moving average for latency
+	currentAvg := atomic.LoadInt64(&abm.averageLatency)
+	newLatency := latency.Nanoseconds()
+
+	if currentAvg == 0 {
+		atomic.StoreInt64(&abm.averageLatency, newLatency)
+	} else {
+		// Exponential moving average: 70% historical, 30% current
+		newAvg := int64(float64(currentAvg)*GetConfig().HistoricalWeight + float64(newLatency)*GetConfig().CurrentWeight)
+		atomic.StoreInt64(&abm.averageLatency, newAvg)
+	}
 }
 
 // adaptationLoop is the main loop that adjusts buffer sizes

internal/audio/api.go

@@ -84,21 +84,3 @@ func GetAudioInputSupervisor() *AudioInputSupervisor {
 	}
 	return (*AudioInputSupervisor)(ptr)
 }
-
-// PrewarmAudioInputSubprocess starts an audio input subprocess in advance to reduce activation latency
-func PrewarmAudioInputSubprocess() error {
-	supervisor := GetAudioInputSupervisor()
-	if supervisor == nil {
-		return nil // No supervisor available, skip prewarming
-	}
-	return supervisor.PrewarmSubprocess()
-}
-
-// IsAudioInputSubprocessPrewarmed returns whether an audio input subprocess is prewarmed and ready
-func IsAudioInputSubprocessPrewarmed() bool {
-	supervisor := GetAudioInputSupervisor()
-	if supervisor == nil {
-		return false
-	}
-	return supervisor.IsPrewarmed()
-}

internal/audio/audio.go

@@ -23,6 +23,8 @@
 //	SetAudioQuality(AudioQualityHigh)
 //
 //	// Audio output will automatically start when frames are received
+//	metrics := GetAudioMetrics()
+//	fmt.Printf("Latency: %v, Frames: %d\n", metrics.AverageLatency, metrics.FramesReceived)
 package audio
 
 import (
@@ -330,60 +332,42 @@ func GetMicrophoneConfig() AudioConfig {
 	return currentMicrophoneConfig
 }
 
-// Batched metrics to reduce atomic operations frequency
+// GetAudioMetrics returns current audio metrics
-var (
+func GetAudioMetrics() AudioMetrics {
-	batchedFramesReceived  int64
+	// Get base metrics
-	batchedBytesProcessed  int64
+	framesReceived := atomic.LoadInt64(&metrics.FramesReceived)
-	batchedFramesDropped   int64
+	framesDropped := atomic.LoadInt64(&metrics.FramesDropped)
-	batchedConnectionDrops int64
 
-	lastFlushTime int64 // Unix timestamp in nanoseconds
+	// If audio relay is running, use relay stats instead
-)
+	if IsAudioRelayRunning() {
+		relayReceived, relayDropped := GetAudioRelayStats()
+		framesReceived = relayReceived
+		framesDropped = relayDropped
+	}
 
-// RecordFrameReceived increments the frames received counter with batched updates
+	return AudioMetrics{
+		FramesReceived:  framesReceived,
+		FramesDropped:   framesDropped,
+		BytesProcessed:  atomic.LoadInt64(&metrics.BytesProcessed),
+		LastFrameTime:   metrics.LastFrameTime,
+		ConnectionDrops: atomic.LoadInt64(&metrics.ConnectionDrops),
+		AverageLatency:  metrics.AverageLatency,
+	}
+}
+
+// RecordFrameReceived increments the frames received counter
 func RecordFrameReceived(bytes int) {
-	// Use local batching to reduce atomic operations frequency
+	atomic.AddInt64(&metrics.FramesReceived, 1)
-	atomic.AddInt64(&batchedBytesProcessed, int64(bytes))
+	atomic.AddInt64(&metrics.BytesProcessed, int64(bytes))
-
-	// Update timestamp immediately for accurate tracking
 	metrics.LastFrameTime = time.Now()
 }
 
-// RecordFrameDropped increments the frames dropped counter with batched updates
+// RecordFrameDropped increments the frames dropped counter
 func RecordFrameDropped() {
+	atomic.AddInt64(&metrics.FramesDropped, 1)
 }
 
-// RecordConnectionDrop increments the connection drops counter with batched updates
+// RecordConnectionDrop increments the connection drops counter
 func RecordConnectionDrop() {
-}
+	atomic.AddInt64(&metrics.ConnectionDrops, 1)
-
-// flushBatchedMetrics flushes accumulated metrics to the main counters
-func flushBatchedMetrics() {
-	// Atomically move batched metrics to main metrics
-	framesReceived := atomic.SwapInt64(&batchedFramesReceived, 0)
-	bytesProcessed := atomic.SwapInt64(&batchedBytesProcessed, 0)
-	framesDropped := atomic.SwapInt64(&batchedFramesDropped, 0)
-	connectionDrops := atomic.SwapInt64(&batchedConnectionDrops, 0)
-
-	// Update main metrics if we have any batched data
-	if framesReceived > 0 {
-		atomic.AddInt64(&metrics.FramesReceived, framesReceived)
-	}
-	if bytesProcessed > 0 {
-		atomic.AddInt64(&metrics.BytesProcessed, bytesProcessed)
-	}
-	if framesDropped > 0 {
-		atomic.AddInt64(&metrics.FramesDropped, framesDropped)
-	}
-	if connectionDrops > 0 {
-		atomic.AddInt64(&metrics.ConnectionDrops, connectionDrops)
-	}
-
-	// Update last flush time
-	atomic.StoreInt64(&lastFlushTime, time.Now().UnixNano())
-}
-
-// FlushPendingMetrics forces a flush of all batched metrics
-func FlushPendingMetrics() {
-	flushBatchedMetrics()
 }

internal/audio/audio_quality_edge_cases_test.go

@@ -0,0 +1,317 @@
+//go:build cgo
+// +build cgo
+
+package audio
+
+import (
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestAudioQualityEdgeCases tests edge cases for audio quality functions
+// These tests ensure the recent validation removal doesn't introduce regressions
+func TestAudioQualityEdgeCases(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"AudioQualityBoundaryValues", testAudioQualityBoundaryValues},
+		{"MicrophoneQualityBoundaryValues", testMicrophoneQualityBoundaryValues},
+		{"AudioQualityPresetsConsistency", testAudioQualityPresetsConsistency},
+		{"MicrophoneQualityPresetsConsistency", testMicrophoneQualityPresetsConsistency},
+		{"QualitySettingsThreadSafety", testQualitySettingsThreadSafety},
+		{"QualityPresetsImmutability", testQualityPresetsImmutability},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testAudioQualityBoundaryValues tests boundary values for audio quality
+func testAudioQualityBoundaryValues(t *testing.T) {
+	// Test minimum valid quality (0)
+	originalConfig := GetAudioConfig()
+	SetAudioQuality(AudioQualityLow)
+	assert.Equal(t, AudioQualityLow, GetAudioConfig().Quality, "Should accept minimum quality value")
+
+	// Test maximum valid quality (3)
+	SetAudioQuality(AudioQualityUltra)
+	assert.Equal(t, AudioQualityUltra, GetAudioConfig().Quality, "Should accept maximum quality value")
+
+	// Test that quality settings work correctly
+	SetAudioQuality(AudioQualityMedium)
+	currentConfig := GetAudioConfig()
+	assert.Equal(t, AudioQualityMedium, currentConfig.Quality, "Should set medium quality")
+	t.Logf("Medium quality config: %+v", currentConfig)
+
+	SetAudioQuality(AudioQualityHigh)
+	currentConfig = GetAudioConfig()
+	assert.Equal(t, AudioQualityHigh, currentConfig.Quality, "Should set high quality")
+	t.Logf("High quality config: %+v", currentConfig)
+
+	// Restore original quality
+	SetAudioQuality(originalConfig.Quality)
+}
+
+// testMicrophoneQualityBoundaryValues tests boundary values for microphone quality
+func testMicrophoneQualityBoundaryValues(t *testing.T) {
+	// Test minimum valid quality
+	originalConfig := GetMicrophoneConfig()
+	SetMicrophoneQuality(AudioQualityLow)
+	assert.Equal(t, AudioQualityLow, GetMicrophoneConfig().Quality, "Should accept minimum microphone quality value")
+
+	// Test maximum valid quality
+	SetMicrophoneQuality(AudioQualityUltra)
+	assert.Equal(t, AudioQualityUltra, GetMicrophoneConfig().Quality, "Should accept maximum microphone quality value")
+
+	// Test that quality settings work correctly
+	SetMicrophoneQuality(AudioQualityMedium)
+	currentConfig := GetMicrophoneConfig()
+	assert.Equal(t, AudioQualityMedium, currentConfig.Quality, "Should set medium microphone quality")
+	t.Logf("Medium microphone quality config: %+v", currentConfig)
+
+	SetMicrophoneQuality(AudioQualityHigh)
+	currentConfig = GetMicrophoneConfig()
+	assert.Equal(t, AudioQualityHigh, currentConfig.Quality, "Should set high microphone quality")
+	t.Logf("High microphone quality config: %+v", currentConfig)
+
+	// Restore original quality
+	SetMicrophoneQuality(originalConfig.Quality)
+}
+
+// testAudioQualityPresetsConsistency tests consistency of audio quality presets
+func testAudioQualityPresetsConsistency(t *testing.T) {
+	presets := GetAudioQualityPresets()
+	require.NotNil(t, presets, "Audio quality presets should not be nil")
+	require.NotEmpty(t, presets, "Audio quality presets should not be empty")
+
+	// Verify presets have expected structure
+	for i, preset := range presets {
+		t.Logf("Audio preset %d: %+v", i, preset)
+
+		// Each preset should have reasonable values
+		assert.GreaterOrEqual(t, preset.Bitrate, 0, "Bitrate should be non-negative")
+		assert.Greater(t, preset.SampleRate, 0, "Sample rate should be positive")
+		assert.Greater(t, preset.Channels, 0, "Channels should be positive")
+	}
+
+	// Test that presets are accessible by valid quality levels
+	qualityLevels := []AudioQuality{AudioQualityLow, AudioQualityMedium, AudioQualityHigh, AudioQualityUltra}
+	for _, quality := range qualityLevels {
+		preset, exists := presets[quality]
+		assert.True(t, exists, "Preset should exist for quality %v", quality)
+		assert.Greater(t, preset.Bitrate, 0, "Preset bitrate should be positive for quality %v", quality)
+	}
+}
+
+// testMicrophoneQualityPresetsConsistency tests consistency of microphone quality presets
+func testMicrophoneQualityPresetsConsistency(t *testing.T) {
+	presets := GetMicrophoneQualityPresets()
+	require.NotNil(t, presets, "Microphone quality presets should not be nil")
+	require.NotEmpty(t, presets, "Microphone quality presets should not be empty")
+
+	// Verify presets have expected structure
+	for i, preset := range presets {
+		t.Logf("Microphone preset %d: %+v", i, preset)
+
+		// Each preset should have reasonable values
+		assert.GreaterOrEqual(t, preset.Bitrate, 0, "Bitrate should be non-negative")
+		assert.Greater(t, preset.SampleRate, 0, "Sample rate should be positive")
+		assert.Greater(t, preset.Channels, 0, "Channels should be positive")
+	}
+
+	// Test that presets are accessible by valid quality levels
+	qualityLevels := []AudioQuality{AudioQualityLow, AudioQualityMedium, AudioQualityHigh, AudioQualityUltra}
+	for _, quality := range qualityLevels {
+		preset, exists := presets[quality]
+		assert.True(t, exists, "Microphone preset should exist for quality %v", quality)
+		assert.Greater(t, preset.Bitrate, 0, "Microphone preset bitrate should be positive for quality %v", quality)
+	}
+}
+
+// testQualitySettingsThreadSafety tests thread safety of quality settings
+func testQualitySettingsThreadSafety(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping thread safety test in short mode")
+	}
+
+	originalAudioConfig := GetAudioConfig()
+	originalMicConfig := GetMicrophoneConfig()
+
+	// Test concurrent access to quality settings
+	const numGoroutines = 50
+	const numOperations = 100
+
+	done := make(chan bool, numGoroutines*2)
+
+	// Audio quality goroutines
+	for i := 0; i < numGoroutines; i++ {
+		go func(id int) {
+			for j := 0; j < numOperations; j++ {
+				// Cycle through valid quality values
+				qualityIndex := j % 4
+				var quality AudioQuality
+				switch qualityIndex {
+				case 0:
+					quality = AudioQualityLow
+				case 1:
+					quality = AudioQualityMedium
+				case 2:
+					quality = AudioQualityHigh
+				case 3:
+					quality = AudioQualityUltra
+				}
+				SetAudioQuality(quality)
+				_ = GetAudioConfig()
+			}
+			done <- true
+		}(i)
+	}
+
+	// Microphone quality goroutines
+	for i := 0; i < numGoroutines; i++ {
+		go func(id int) {
+			for j := 0; j < numOperations; j++ {
+				// Cycle through valid quality values
+				qualityIndex := j % 4
+				var quality AudioQuality
+				switch qualityIndex {
+				case 0:
+					quality = AudioQualityLow
+				case 1:
+					quality = AudioQualityMedium
+				case 2:
+					quality = AudioQualityHigh
+				case 3:
+					quality = AudioQualityUltra
+				}
+				SetMicrophoneQuality(quality)
+				_ = GetMicrophoneConfig()
+			}
+			done <- true
+		}(i)
+	}
+
+	// Wait for all goroutines to complete
+	for i := 0; i < numGoroutines*2; i++ {
+		<-done
+	}
+
+	// Verify system is still functional
+	SetAudioQuality(AudioQualityHigh)
+	assert.Equal(t, AudioQualityHigh, GetAudioConfig().Quality, "Audio quality should be settable after concurrent access")
+
+	SetMicrophoneQuality(AudioQualityMedium)
+	assert.Equal(t, AudioQualityMedium, GetMicrophoneConfig().Quality, "Microphone quality should be settable after concurrent access")
+
+	// Restore original values
+	SetAudioQuality(originalAudioConfig.Quality)
+	SetMicrophoneQuality(originalMicConfig.Quality)
+}
+
+// testQualityPresetsImmutability tests that quality presets are not accidentally modified
+func testQualityPresetsImmutability(t *testing.T) {
+	// Get presets multiple times and verify they're consistent
+	presets1 := GetAudioQualityPresets()
+	presets2 := GetAudioQualityPresets()
+
+	require.Equal(t, len(presets1), len(presets2), "Preset count should be consistent")
+
+	// Verify each preset is identical
+	for quality := range presets1 {
+		assert.Equal(t, presets1[quality].Bitrate, presets2[quality].Bitrate,
+			"Preset %v bitrate should be consistent", quality)
+		assert.Equal(t, presets1[quality].SampleRate, presets2[quality].SampleRate,
+			"Preset %v sample rate should be consistent", quality)
+		assert.Equal(t, presets1[quality].Channels, presets2[quality].Channels,
+			"Preset %v channels should be consistent", quality)
+	}
+
+	// Test microphone presets as well
+	micPresets1 := GetMicrophoneQualityPresets()
+	micPresets2 := GetMicrophoneQualityPresets()
+
+	require.Equal(t, len(micPresets1), len(micPresets2), "Microphone preset count should be consistent")
+
+	for quality := range micPresets1 {
+		assert.Equal(t, micPresets1[quality].Bitrate, micPresets2[quality].Bitrate,
+			"Microphone preset %v bitrate should be consistent", quality)
+		assert.Equal(t, micPresets1[quality].SampleRate, micPresets2[quality].SampleRate,
+			"Microphone preset %v sample rate should be consistent", quality)
+		assert.Equal(t, micPresets1[quality].Channels, micPresets2[quality].Channels,
+			"Microphone preset %v channels should be consistent", quality)
+	}
+}
+
+// TestQualityValidationRemovalRegression tests that validation removal doesn't cause regressions
+func TestQualityValidationRemovalRegression(t *testing.T) {
+	// This test ensures that removing validation from GET endpoints doesn't break functionality
+
+	// Test that presets are still accessible
+	audioPresets := GetAudioQualityPresets()
+	assert.NotNil(t, audioPresets, "Audio presets should be accessible after validation removal")
+	assert.NotEmpty(t, audioPresets, "Audio presets should not be empty")
+
+	micPresets := GetMicrophoneQualityPresets()
+	assert.NotNil(t, micPresets, "Microphone presets should be accessible after validation removal")
+	assert.NotEmpty(t, micPresets, "Microphone presets should not be empty")
+
+	// Test that quality getters still work
+	audioConfig := GetAudioConfig()
+	assert.GreaterOrEqual(t, int(audioConfig.Quality), 0, "Audio quality should be non-negative")
+
+	micConfig := GetMicrophoneConfig()
+	assert.GreaterOrEqual(t, int(micConfig.Quality), 0, "Microphone quality should be non-negative")
+
+	// Test that setters still work (for valid values)
+	originalAudio := GetAudioConfig()
+	originalMic := GetMicrophoneConfig()
+
+	SetAudioQuality(AudioQualityMedium)
+	assert.Equal(t, AudioQualityMedium, GetAudioConfig().Quality, "Audio quality setter should work")
+
+	SetMicrophoneQuality(AudioQualityHigh)
+	assert.Equal(t, AudioQualityHigh, GetMicrophoneConfig().Quality, "Microphone quality setter should work")
+
+	// Restore original values
+	SetAudioQuality(originalAudio.Quality)
+	SetMicrophoneQuality(originalMic.Quality)
+}
+
+// TestPerformanceAfterValidationRemoval tests that performance improved after validation removal
+func TestPerformanceAfterValidationRemoval(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping performance test in short mode")
+	}
+
+	// Benchmark preset access (should be faster without validation)
+	const iterations = 10000
+
+	// Time audio preset access
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = GetAudioQualityPresets()
+	}
+	audioDuration := time.Since(start)
+
+	// Time microphone preset access
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = GetMicrophoneQualityPresets()
+	}
+	micDuration := time.Since(start)
+
+	t.Logf("Audio presets access time for %d iterations: %v", iterations, audioDuration)
+	t.Logf("Microphone presets access time for %d iterations: %v", iterations, micDuration)
+
+	// Verify reasonable performance (should complete quickly without validation overhead)
+	maxExpectedDuration := time.Second // Very generous limit
+	assert.Less(t, audioDuration, maxExpectedDuration, "Audio preset access should be fast")
+	assert.Less(t, micDuration, maxExpectedDuration, "Microphone preset access should be fast")
+}

internal/audio/audio_test.go

@@ -0,0 +1,366 @@
+package audio
+
+import (
+	"context"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+
+	"github.com/jetkvm/kvm/internal/usbgadget"
+)
+
+// Unit tests for the audio package
+
+func TestAudioQuality(t *testing.T) {
+	tests := []struct {
+		name     string
+		quality  AudioQuality
+		expected string
+	}{
+		{"Low Quality", AudioQualityLow, "low"},
+		{"Medium Quality", AudioQualityMedium, "medium"},
+		{"High Quality", AudioQualityHigh, "high"},
+		{"Ultra Quality", AudioQualityUltra, "ultra"},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			// Test quality setting
+			SetAudioQuality(tt.quality)
+			config := GetAudioConfig()
+			assert.Equal(t, tt.quality, config.Quality)
+			assert.Greater(t, config.Bitrate, 0)
+			assert.Greater(t, config.SampleRate, 0)
+			assert.Greater(t, config.Channels, 0)
+			assert.Greater(t, config.FrameSize, time.Duration(0))
+		})
+	}
+}
+
+func TestMicrophoneQuality(t *testing.T) {
+	tests := []struct {
+		name    string
+		quality AudioQuality
+	}{
+		{"Low Quality", AudioQualityLow},
+		{"Medium Quality", AudioQualityMedium},
+		{"High Quality", AudioQualityHigh},
+		{"Ultra Quality", AudioQualityUltra},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			// Test microphone quality setting
+			SetMicrophoneQuality(tt.quality)
+			config := GetMicrophoneConfig()
+			assert.Equal(t, tt.quality, config.Quality)
+			assert.Equal(t, 1, config.Channels) // Microphone is always mono
+			assert.Greater(t, config.Bitrate, 0)
+			assert.Greater(t, config.SampleRate, 0)
+		})
+	}
+}
+
+func TestAudioQualityPresets(t *testing.T) {
+	presets := GetAudioQualityPresets()
+	require.NotEmpty(t, presets)
+
+	// Test that all quality levels have presets
+	for quality := AudioQualityLow; quality <= AudioQualityUltra; quality++ {
+		config, exists := presets[quality]
+		require.True(t, exists, "Preset should exist for quality %d", quality)
+		assert.Equal(t, quality, config.Quality)
+		assert.Greater(t, config.Bitrate, 0)
+		assert.Greater(t, config.SampleRate, 0)
+		assert.Greater(t, config.Channels, 0)
+		assert.Greater(t, config.FrameSize, time.Duration(0))
+	}
+
+	// Test that higher quality has higher bitrate
+	lowConfig := presets[AudioQualityLow]
+	mediumConfig := presets[AudioQualityMedium]
+	highConfig := presets[AudioQualityHigh]
+	ultraConfig := presets[AudioQualityUltra]
+
+	assert.Less(t, lowConfig.Bitrate, mediumConfig.Bitrate)
+	assert.Less(t, mediumConfig.Bitrate, highConfig.Bitrate)
+	assert.Less(t, highConfig.Bitrate, ultraConfig.Bitrate)
+}
+
+func TestMicrophoneQualityPresets(t *testing.T) {
+	presets := GetMicrophoneQualityPresets()
+	require.NotEmpty(t, presets)
+
+	// Test that all quality levels have presets
+	for quality := AudioQualityLow; quality <= AudioQualityUltra; quality++ {
+		config, exists := presets[quality]
+		require.True(t, exists, "Microphone preset should exist for quality %d", quality)
+		assert.Equal(t, quality, config.Quality)
+		assert.Equal(t, 1, config.Channels) // Always mono
+		assert.Greater(t, config.Bitrate, 0)
+		assert.Greater(t, config.SampleRate, 0)
+	}
+}
+
+func TestAudioMetrics(t *testing.T) {
+	// Test initial metrics
+	metrics := GetAudioMetrics()
+	assert.GreaterOrEqual(t, metrics.FramesReceived, int64(0))
+	assert.GreaterOrEqual(t, metrics.FramesDropped, int64(0))
+	assert.GreaterOrEqual(t, metrics.BytesProcessed, int64(0))
+	assert.GreaterOrEqual(t, metrics.ConnectionDrops, int64(0))
+
+	// Test recording metrics
+	RecordFrameReceived(1024)
+	metrics = GetAudioMetrics()
+	assert.Greater(t, metrics.BytesProcessed, int64(0))
+	assert.Greater(t, metrics.FramesReceived, int64(0))
+
+	RecordFrameDropped()
+	metrics = GetAudioMetrics()
+	assert.Greater(t, metrics.FramesDropped, int64(0))
+
+	RecordConnectionDrop()
+	metrics = GetAudioMetrics()
+	assert.Greater(t, metrics.ConnectionDrops, int64(0))
+}
+
+func TestMaxAudioFrameSize(t *testing.T) {
+	frameSize := GetMaxAudioFrameSize()
+	assert.Greater(t, frameSize, 0)
+	assert.Equal(t, GetConfig().MaxAudioFrameSize, frameSize)
+}
+
+func TestMetricsUpdateInterval(t *testing.T) {
+	// Test getting current interval
+	interval := GetMetricsUpdateInterval()
+	assert.Greater(t, interval, time.Duration(0))
+
+	// Test setting new interval
+	newInterval := 2 * time.Second
+	SetMetricsUpdateInterval(newInterval)
+	updatedInterval := GetMetricsUpdateInterval()
+	assert.Equal(t, newInterval, updatedInterval)
+}
+
+func TestAudioConfigConsistency(t *testing.T) {
+	// Test that setting audio quality updates the config consistently
+	for quality := AudioQualityLow; quality <= AudioQualityUltra; quality++ {
+		SetAudioQuality(quality)
+		config := GetAudioConfig()
+		presets := GetAudioQualityPresets()
+		expectedConfig := presets[quality]
+
+		assert.Equal(t, expectedConfig.Quality, config.Quality)
+		assert.Equal(t, expectedConfig.Bitrate, config.Bitrate)
+		assert.Equal(t, expectedConfig.SampleRate, config.SampleRate)
+		assert.Equal(t, expectedConfig.Channels, config.Channels)
+		assert.Equal(t, expectedConfig.FrameSize, config.FrameSize)
+	}
+}
+
+func TestMicrophoneConfigConsistency(t *testing.T) {
+	// Test that setting microphone quality updates the config consistently
+	for quality := AudioQualityLow; quality <= AudioQualityUltra; quality++ {
+		SetMicrophoneQuality(quality)
+		config := GetMicrophoneConfig()
+		presets := GetMicrophoneQualityPresets()
+		expectedConfig := presets[quality]
+
+		assert.Equal(t, expectedConfig.Quality, config.Quality)
+		assert.Equal(t, expectedConfig.Bitrate, config.Bitrate)
+		assert.Equal(t, expectedConfig.SampleRate, config.SampleRate)
+		assert.Equal(t, expectedConfig.Channels, config.Channels)
+		assert.Equal(t, expectedConfig.FrameSize, config.FrameSize)
+	}
+}
+
+// Benchmark tests
+func BenchmarkGetAudioConfig(b *testing.B) {
+	for i := 0; i < b.N; i++ {
+		_ = GetAudioConfig()
+	}
+}
+
+func BenchmarkGetAudioMetrics(b *testing.B) {
+	for i := 0; i < b.N; i++ {
+		_ = GetAudioMetrics()
+	}
+}
+
+func BenchmarkRecordFrameReceived(b *testing.B) {
+	for i := 0; i < b.N; i++ {
+		RecordFrameReceived(1024)
+	}
+}
+
+func BenchmarkSetAudioQuality(b *testing.B) {
+	qualities := []AudioQuality{AudioQualityLow, AudioQualityMedium, AudioQualityHigh, AudioQualityUltra}
+	b.ResetTimer()
+
+	for i := 0; i < b.N; i++ {
+		SetAudioQuality(qualities[i%len(qualities)])
+	}
+}
+
+// TestAudioUsbGadgetIntegration tests audio functionality with USB gadget reconfiguration
+// This test simulates the production scenario where audio devices are enabled/disabled
+// through USB gadget configuration changes
+func TestAudioUsbGadgetIntegration(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping integration test in short mode")
+	}
+
+	tests := []struct {
+		name                string
+		initialAudioEnabled bool
+		newAudioEnabled     bool
+		expectedTransition  string
+	}{
+		{
+			name:                "EnableAudio",
+			initialAudioEnabled: false,
+			newAudioEnabled:     true,
+			expectedTransition:  "disabled_to_enabled",
+		},
+		{
+			name:                "DisableAudio",
+			initialAudioEnabled: true,
+			newAudioEnabled:     false,
+			expectedTransition:  "enabled_to_disabled",
+		},
+		{
+			name:                "NoChange",
+			initialAudioEnabled: true,
+			newAudioEnabled:     true,
+			expectedTransition:  "no_change",
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			ctx, cancel := context.WithTimeout(context.Background(), 30*time.Second)
+			defer cancel()
+
+			// Simulate initial USB device configuration
+			initialDevices := &usbgadget.Devices{
+				Keyboard:      true,
+				AbsoluteMouse: true,
+				RelativeMouse: true,
+				MassStorage:   true,
+				Audio:         tt.initialAudioEnabled,
+			}
+
+			// Simulate new USB device configuration
+			newDevices := &usbgadget.Devices{
+				Keyboard:      true,
+				AbsoluteMouse: true,
+				RelativeMouse: true,
+				MassStorage:   true,
+				Audio:         tt.newAudioEnabled,
+			}
+
+			// Test audio configuration validation
+			err := validateAudioDeviceConfiguration(tt.newAudioEnabled)
+			assert.NoError(t, err, "Audio configuration should be valid")
+
+			// Test audio state transition simulation
+			transition := simulateAudioStateTransition(ctx, initialDevices, newDevices)
+			assert.Equal(t, tt.expectedTransition, transition, "Audio state transition should match expected")
+
+			// Test that audio configuration is consistent after transition
+			if tt.newAudioEnabled {
+				config := GetAudioConfig()
+				assert.Greater(t, config.Bitrate, 0, "Audio bitrate should be positive when enabled")
+				assert.Greater(t, config.SampleRate, 0, "Audio sample rate should be positive when enabled")
+			}
+		})
+	}
+}
+
+// validateAudioDeviceConfiguration simulates the audio validation that happens in production
+func validateAudioDeviceConfiguration(enabled bool) error {
+	if !enabled {
+		return nil // No validation needed when disabled
+	}
+
+	// Simulate audio device availability checks
+	// In production, this would check for ALSA devices, audio hardware, etc.
+	config := GetAudioConfig()
+	if config.Bitrate <= 0 {
+		return assert.AnError
+	}
+	if config.SampleRate <= 0 {
+		return assert.AnError
+	}
+
+	return nil
+}
+
+// simulateAudioStateTransition simulates the audio process management during USB reconfiguration
+func simulateAudioStateTransition(ctx context.Context, initial, new *usbgadget.Devices) string {
+	previousAudioEnabled := initial.Audio
+	newAudioEnabled := new.Audio
+
+	if previousAudioEnabled == newAudioEnabled {
+		return "no_change"
+	}
+
+	if !newAudioEnabled {
+		// Simulate stopping audio processes
+		// In production, this would stop AudioInputManager and audioSupervisor
+		time.Sleep(10 * time.Millisecond) // Simulate process stop time
+		return "enabled_to_disabled"
+	}
+
+	if newAudioEnabled {
+		// Simulate starting audio processes after USB reconfiguration
+		// In production, this would start audioSupervisor and broadcast events
+		time.Sleep(10 * time.Millisecond) // Simulate process start time
+		return "disabled_to_enabled"
+	}
+
+	return "unknown"
+}
+
+// TestAudioUsbGadgetTimeout tests that audio operations don't timeout during USB reconfiguration
+func TestAudioUsbGadgetTimeout(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping timeout test in short mode")
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 5*time.Second)
+	defer cancel()
+
+	// Test that audio configuration changes complete within reasonable time
+	start := time.Now()
+
+	// Simulate multiple rapid USB device configuration changes
+	for i := 0; i < 10; i++ {
+		audioEnabled := i%2 == 0
+		devices := &usbgadget.Devices{
+			Keyboard:      true,
+			AbsoluteMouse: true,
+			RelativeMouse: true,
+			MassStorage:   true,
+			Audio:         audioEnabled,
+		}
+
+		err := validateAudioDeviceConfiguration(devices.Audio)
+		assert.NoError(t, err, "Audio validation should not fail")
+
+		// Ensure we don't timeout
+		select {
+		case <-ctx.Done():
+			t.Fatal("Audio configuration test timed out")
+		default:
+			// Continue
+		}
+	}
+
+	elapsed := time.Since(start)
+	t.Logf("Audio USB gadget configuration test completed in %v", elapsed)
+	assert.Less(t, elapsed, 3*time.Second, "Audio configuration should complete quickly")
+}

internal/audio/base_manager.go

@@ -23,7 +23,6 @@ type BaseAudioMetrics struct {
 
 // BaseAudioManager provides common functionality for audio managers
 type BaseAudioManager struct {
-	// Core metrics and state
 	metrics BaseAudioMetrics
 	logger  zerolog.Logger
 	running int32
@@ -59,12 +58,6 @@ func (bam *BaseAudioManager) resetMetrics() {
 	bam.metrics.AverageLatency = 0
 }
 
-// flushPendingMetrics is now a no-op since we use direct atomic updates
-func (bam *BaseAudioManager) flushPendingMetrics() {
-	// No-op: metrics are now updated directly without local buffering
-	// This function is kept for API compatibility
-}
-
 // getBaseMetrics returns a copy of the base metrics
 func (bam *BaseAudioManager) getBaseMetrics() BaseAudioMetrics {
 	return BaseAudioMetrics{
@@ -77,16 +70,28 @@ func (bam *BaseAudioManager) getBaseMetrics() BaseAudioMetrics {
 	}
 }
 
-// recordFrameProcessed records a processed frame with simplified tracking
+// recordFrameProcessed records a processed frame
 func (bam *BaseAudioManager) recordFrameProcessed(bytes int) {
+	atomic.AddInt64(&bam.metrics.FramesProcessed, 1)
+	atomic.AddInt64(&bam.metrics.BytesProcessed, int64(bytes))
+	bam.metrics.LastFrameTime = time.Now()
 }
 
-// recordFrameDropped records a dropped frame with simplified tracking
+// recordFrameDropped records a dropped frame
 func (bam *BaseAudioManager) recordFrameDropped() {
+	atomic.AddInt64(&bam.metrics.FramesDropped, 1)
 }
 
 // updateLatency updates the average latency
 func (bam *BaseAudioManager) updateLatency(latency time.Duration) {
+	// Simple moving average - could be enhanced with more sophisticated algorithms
+	currentAvg := bam.metrics.AverageLatency
+	if currentAvg == 0 {
+		bam.metrics.AverageLatency = latency
+	} else {
+		// Weighted average: 90% old + 10% new
+		bam.metrics.AverageLatency = time.Duration(float64(currentAvg)*0.9 + float64(latency)*0.1)
+	}
 }
 
 // logComponentStart logs component start with consistent format

internal/audio/base_supervisor.go

@@ -71,6 +71,45 @@ func (bs *BaseSupervisor) GetLastExitInfo() (exitCode int, exitTime time.Time) {
 	return bs.lastExitCode, bs.lastExitTime
 }
 
+// GetProcessMetrics returns process metrics if available
+func (bs *BaseSupervisor) GetProcessMetrics() *ProcessMetrics {
+	bs.mutex.RLock()
+	defer bs.mutex.RUnlock()
+
+	if bs.cmd == nil || bs.cmd.Process == nil {
+		return &ProcessMetrics{
+			PID:           0,
+			CPUPercent:    0.0,
+			MemoryRSS:     0,
+			MemoryVMS:     0,
+			MemoryPercent: 0.0,
+			Timestamp:     time.Now(),
+			ProcessName:   "audio-server",
+		}
+	}
+
+	pid := bs.cmd.Process.Pid
+	if bs.processMonitor != nil {
+		metrics := bs.processMonitor.GetCurrentMetrics()
+		for _, metric := range metrics {
+			if metric.PID == pid {
+				return &metric
+			}
+		}
+	}
+
+	// Return default metrics if process not found in monitor
+	return &ProcessMetrics{
+		PID:           pid,
+		CPUPercent:    0.0,
+		MemoryRSS:     0,
+		MemoryVMS:     0,
+		MemoryPercent: 0.0,
+		Timestamp:     time.Now(),
+		ProcessName:   "audio-server",
+	}
+}
+
 // logSupervisorStart logs supervisor start event
 func (bs *BaseSupervisor) logSupervisorStart() {
 	bs.logger.Info().Msg("Supervisor starting")

internal/audio/batch_audio.go

@@ -94,12 +94,6 @@ func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAu
 		batchDuration = cache.BatchProcessingDelay
 	}
 
-	// Use optimized queue sizes from configuration
-	queueSize := cache.BatchProcessorMaxQueueSize
-	if queueSize <= 0 {
-		queueSize = batchSize * 2 // Fallback to double batch size
-	}
-
 	ctx, cancel := context.WithCancel(context.Background())
 	// Pre-allocate logger to avoid repeated allocations
 	logger := logging.GetDefaultLogger().With().Str("component", "batch-audio").Logger()
@@ -116,8 +110,8 @@ func NewBatchAudioProcessor(batchSize int, batchDuration time.Duration) *BatchAu
 		logger:        &logger,
 		batchSize:     batchSize,
 		batchDuration: batchDuration,
-		readQueue:     make(chan batchReadRequest, queueSize),
+		readQueue:     make(chan batchReadRequest, batchSize*2),
-		writeQueue:    make(chan batchWriteRequest, queueSize),
+		writeQueue:    make(chan batchWriteRequest, batchSize*2),
 		readBufPool: &sync.Pool{
 			New: func() interface{} {
 				// Use pre-calculated frame size to avoid GetConfig() calls
@@ -179,20 +173,14 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
-		// Only log validation errors in debug mode to reduce overhead
+		bap.logger.Debug().Err(err).Msg("invalid buffer for batch processing")
-		if bap.logger.GetLevel() <= zerolog.DebugLevel {
-			bap.logger.Debug().Err(err).Msg("invalid buffer for batch processing")
-		}
 		return 0, err
 	}
 
 	if !bap.IsRunning() {
 		// Fallback to single operation if batch processor is not running
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleReads, 1)
-		if atomic.LoadInt64(&bap.stats.SingleReads)%10 == 0 {
+		atomic.AddInt64(&bap.stats.SingleFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleReads, 10)
-			atomic.AddInt64(&bap.stats.SingleFrames, 10)
-		}
 		return CGOAudioReadEncode(buffer)
 	}
 
@@ -209,11 +197,8 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 		// Successfully queued
 	default:
 		// Queue is full, fallback to single operation
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleReads, 1)
-		if atomic.LoadInt64(&bap.stats.SingleReads)%10 == 0 {
+		atomic.AddInt64(&bap.stats.SingleFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleReads, 10)
-			atomic.AddInt64(&bap.stats.SingleFrames, 10)
-		}
 		return CGOAudioReadEncode(buffer)
 	}
 
@@ -223,11 +208,8 @@ func (bap *BatchAudioProcessor) BatchReadEncode(buffer []byte) (int, error) {
 		return result.length, result.err
 	case <-time.After(cache.BatchProcessingTimeout):
 		// Timeout, fallback to single operation
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleReads, 1)
-		if atomic.LoadInt64(&bap.stats.SingleReads)%10 == 0 {
+		atomic.AddInt64(&bap.stats.SingleFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleReads, 10)
-			atomic.AddInt64(&bap.stats.SingleFrames, 10)
-		}
 		return CGOAudioReadEncode(buffer)
 	}
 }
@@ -241,20 +223,14 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 
 	// Validate buffer before processing
 	if err := ValidateBufferSize(len(buffer)); err != nil {
-		// Only log validation errors in debug mode to reduce overhead
+		bap.logger.Debug().Err(err).Msg("invalid buffer for batch processing")
-		if bap.logger.GetLevel() <= zerolog.DebugLevel {
-			bap.logger.Debug().Err(err).Msg("invalid buffer for batch processing")
-		}
 		return 0, err
 	}
 
 	if !bap.IsRunning() {
 		// Fallback to single operation if batch processor is not running
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleWrites, 1)
-		if atomic.LoadInt64(&bap.stats.SingleWrites)%10 == 0 {
+		atomic.AddInt64(&bap.stats.WriteFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleWrites, 10)
-			atomic.AddInt64(&bap.stats.WriteFrames, 10)
-		}
 		return CGOAudioDecodeWriteLegacy(buffer)
 	}
 
@@ -271,11 +247,8 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 		// Successfully queued
 	default:
 		// Queue is full, fall back to single operation
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleWrites, 1)
-		if atomic.LoadInt64(&bap.stats.SingleWrites)%10 == 0 {
+		atomic.AddInt64(&bap.stats.WriteFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleWrites, 10)
-			atomic.AddInt64(&bap.stats.WriteFrames, 10)
-		}
 		return CGOAudioDecodeWriteLegacy(buffer)
 	}
 
@@ -284,11 +257,8 @@ func (bap *BatchAudioProcessor) BatchDecodeWrite(buffer []byte) (int, error) {
 	case result := <-resultChan:
 		return result.length, result.err
 	case <-time.After(cache.BatchProcessingTimeout):
-		// Use sampling to reduce atomic operations overhead
+		atomic.AddInt64(&bap.stats.SingleWrites, 1)
-		if atomic.LoadInt64(&bap.stats.SingleWrites)%10 == 0 {
+		atomic.AddInt64(&bap.stats.WriteFrames, 1)
-			atomic.AddInt64(&bap.stats.SingleWrites, 10)
-			atomic.AddInt64(&bap.stats.WriteFrames, 10)
-		}
 		return CGOAudioDecodeWriteLegacy(buffer)
 	}
 }
@@ -428,21 +398,19 @@ func (bap *BatchAudioProcessor) processBatchRead(batch []batchReadRequest) {
 
 	// Get cached config once - avoid repeated calls
 	cache := GetCachedConfig()
-	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
+	minBatchSize := cache.MinBatchSizeForThreadPinning
-	if threadPinningThreshold == 0 {
-		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
-	}
 
 	// Only pin to OS thread for large batches to reduce thread contention
 	var start time.Time
 	threadWasPinned := false
-	if batchSize >= threadPinningThreshold && atomic.CompareAndSwapInt32(&bap.threadPinned, 0, 1) {
+	if batchSize >= minBatchSize && atomic.CompareAndSwapInt32(&bap.threadPinned, 0, 1) {
 		start = time.Now()
 		threadWasPinned = true
 		runtime.LockOSThread()
+		// Skip priority setting for better performance - audio threads already have good priority
 	}
 
-	// Batch stats updates to reduce atomic operations (update once per batch instead of per frame)
+	// Update stats efficiently
 	atomic.AddInt64(&bap.stats.BatchedReads, 1)
 	atomic.AddInt64(&bap.stats.BatchedFrames, int64(batchSize))
 	if batchSize > 1 {
@@ -469,7 +437,6 @@ func (bap *BatchAudioProcessor) processBatchRead(batch []batchReadRequest) {
 		bap.stats.OSThreadPinTime += time.Since(start)
 	}
 
-	// Update timestamp only once per batch instead of per frame
 	bap.stats.LastBatchTime = time.Now()
 }
 
@@ -481,14 +448,10 @@ func (bap *BatchAudioProcessor) processBatchWrite(batch []batchWriteRequest) {
 
 	// Get cached config to avoid GetConfig() calls in hot path
 	cache := GetCachedConfig()
-	threadPinningThreshold := cache.BatchProcessorThreadPinningThreshold
-	if threadPinningThreshold == 0 {
-		threadPinningThreshold = cache.MinBatchSizeForThreadPinning // Fallback
-	}
 
 	// Only pin to OS thread for large batches to reduce thread contention
 	start := time.Now()
-	shouldPinThread := len(batch) >= threadPinningThreshold
+	shouldPinThread := len(batch) >= cache.MinBatchSizeForThreadPinning
 
 	// Track if we pinned the thread in this call
 	threadWasPinned := false
@@ -497,7 +460,8 @@ func (bap *BatchAudioProcessor) processBatchWrite(batch []batchWriteRequest) {
 		threadWasPinned = true
 		runtime.LockOSThread()
 
-		// Priority scheduler not implemented - using default thread priority
+		// Set high priority for batch audio processing - skip logging in hotpath
+		_ = SetAudioThreadPriority()
 	}
 
 	batchSize := len(batch)
@@ -510,7 +474,8 @@ func (bap *BatchAudioProcessor) processBatchWrite(batch []batchWriteRequest) {
 	// Add deferred function to release thread lock if we pinned it
 	if threadWasPinned {
 		defer func() {
-			// Priority scheduler not implemented - using default thread priority
+			// Skip logging in hotpath for performance
+			_ = ResetThreadPriority()
 			runtime.UnlockOSThread()
 			atomic.StoreInt32(&bap.writePinned, 0)
 			bap.stats.WriteThreadTime += time.Since(start)

internal/audio/buffer_pool.go

@@ -367,7 +367,6 @@ func (p *AudioBufferPool) Get() []byte {
 			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
@@ -384,7 +383,6 @@ func (p *AudioBufferPool) Get() []byte {
 		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)
 		*buf = (*buf)[:0]
 		return *buf
@@ -394,7 +392,6 @@ func (p *AudioBufferPool) Get() []byte {
 	// 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
@@ -406,7 +403,6 @@ func (p *AudioBufferPool) Get() []byte {
 	}
 
 	// 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)
 }

internal/audio/cgo_audio.go

@@ -712,17 +712,7 @@ 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)
+	// Atomic fields for lock-free access to frequently used values
-	minFrameDuration         atomic.Int64 // Store as nanoseconds
-	maxFrameDuration         atomic.Int64 // Store as nanoseconds
-	maxLatency               atomic.Int64 // Store as nanoseconds
-	minMetricsUpdateInterval atomic.Int64 // Store as nanoseconds
-	maxMetricsUpdateInterval atomic.Int64 // Store as nanoseconds
-	restartWindow            atomic.Int64 // Store as nanoseconds
-	restartDelay             atomic.Int64 // Store as nanoseconds
-	maxRestartDelay          atomic.Int64 // Store as nanoseconds
-
-	// Atomic int32 fields for lock-free access to frequently used values
 	minReadEncodeBuffer  atomic.Int32
 	maxDecodeWriteBuffer atomic.Int32
 	maxPacketSize        atomic.Int32
@@ -741,24 +731,17 @@ type AudioConfigCache struct {
 	// Additional cached values for validation functions
 	maxAudioFrameSize atomic.Int32
 	maxChannels       atomic.Int32
+	minFrameDuration  atomic.Int64 // Store as nanoseconds
+	maxFrameDuration  atomic.Int64 // Store as nanoseconds
 	minOpusBitrate    atomic.Int32
 	maxOpusBitrate    atomic.Int32
 
-	// Socket and buffer configuration values
-	socketMaxBuffer          atomic.Int32
-	socketMinBuffer          atomic.Int32
-	inputProcessingTimeoutMS atomic.Int32
-	maxRestartAttempts       atomic.Int32
-
 	// Batch processing related values
-	BatchProcessingTimeout               time.Duration
+	BatchProcessingTimeout       time.Duration
-	BatchProcessorFramesPerBatch         int
+	BatchProcessorFramesPerBatch int
-	BatchProcessorTimeout                time.Duration
+	BatchProcessorTimeout        time.Duration
-	BatchProcessingDelay                 time.Duration
+	BatchProcessingDelay         time.Duration
-	MinBatchSizeForThreadPinning         int
+	MinBatchSizeForThreadPinning int
-	BatchProcessorMaxQueueSize           int
-	BatchProcessorAdaptiveThreshold      float64
-	BatchProcessorThreadPinningThreshold int
 
 	// Mutex for updating the cache
 	mutex       sync.RWMutex
@@ -831,9 +814,6 @@ func (c *AudioConfigCache) Update() {
 		c.BatchProcessorTimeout = config.BatchProcessorTimeout
 		c.BatchProcessingDelay = config.BatchProcessingDelay
 		c.MinBatchSizeForThreadPinning = config.MinBatchSizeForThreadPinning
-		c.BatchProcessorMaxQueueSize = config.BatchProcessorMaxQueueSize
-		c.BatchProcessorAdaptiveThreshold = config.BatchProcessorAdaptiveThreshold
-		c.BatchProcessorThreadPinningThreshold = config.BatchProcessorThreadPinningThreshold
 
 		// Pre-allocate common errors
 		c.bufferTooSmallReadEncode = newBufferTooSmallError(0, config.MinReadEncodeBuffer)
@@ -1038,7 +1018,8 @@ var (
 	batchProcessingCount atomic.Int64
 	batchFrameCount      atomic.Int64
 	batchProcessingTime  atomic.Int64
-	// Batch time tracking removed
+	// Flag to control time tracking overhead
+	enableBatchTimeTracking atomic.Bool
 )
 
 // GetBufferFromPool gets a buffer from the pool with at least the specified capacity
@@ -1243,8 +1224,7 @@ func BatchReadEncode(batchSize int) ([][]byte, error) {
 
 	// Track batch processing statistics - only if enabled
 	var startTime time.Time
-	// Batch time tracking removed
+	trackTime := enableBatchTimeTracking.Load()
-	trackTime := false
 	if trackTime {
 		startTime = time.Now()
 	}
@@ -1311,8 +1291,7 @@ func BatchDecodeWrite(frames [][]byte) error {
 
 	// Track batch processing statistics - only if enabled
 	var startTime time.Time
-	// Batch time tracking removed
+	trackTime := enableBatchTimeTracking.Load()
-	trackTime := false
 	if trackTime {
 		startTime = time.Now()
 	}

internal/audio/cgo_audio_stub.go

@@ -0,0 +1,48 @@
+//go:build !cgo
+
+package audio
+
+import "errors"
+
+// Stub implementations for linting (no CGO dependencies)
+
+func cgoAudioInit() error {
+	return errors.New("audio not available in lint mode")
+}
+
+func cgoAudioClose() {
+	// No-op
+}
+
+func cgoAudioReadEncode(buf []byte) (int, error) {
+	return 0, errors.New("audio not available in lint mode")
+}
+
+func cgoAudioPlaybackInit() error {
+	return errors.New("audio not available in lint mode")
+}
+
+func cgoAudioPlaybackClose() {
+	// No-op
+}
+
+func cgoAudioDecodeWrite(buf []byte) (int, error) {
+	return 0, errors.New("audio not available in lint mode")
+}
+
+// cgoAudioDecodeWriteWithBuffers is a stub implementation for the optimized decode-write function
+func cgoAudioDecodeWriteWithBuffers(opusData []byte, pcmBuffer []byte) (int, error) {
+	return 0, errors.New("audio not available in lint mode")
+}
+
+// Uppercase aliases for external API compatibility
+
+var (
+	CGOAudioInit           = cgoAudioInit
+	CGOAudioClose          = cgoAudioClose
+	CGOAudioReadEncode     = cgoAudioReadEncode
+	CGOAudioPlaybackInit   = cgoAudioPlaybackInit
+	CGOAudioPlaybackClose  = cgoAudioPlaybackClose
+	CGOAudioDecodeWriteLegacy = cgoAudioDecodeWrite
+	CGOAudioDecodeWrite    = cgoAudioDecodeWriteWithBuffers
+)

internal/audio/config_constants.go

@@ -224,8 +224,51 @@ type AudioConfigConstants struct {
 	// Used in: process_monitor.go for configuring thread scheduling behavior
 	// Impact: Controls how audio threads are scheduled by the Linux kernel
 
-	// Removed unused scheduling policy constants and RT priority values
+	// SchedNormal defines normal (CFS) scheduling policy.
-	// The priority scheduler is not implemented - functions are called but don't exist
+	// Used in: process_monitor.go for non-critical audio threads
+	// Impact: Standard time-sharing scheduling, may cause audio latency under load.
+	// Value 0 corresponds to SCHED_NORMAL in Linux kernel.
+	SchedNormal int
+
+	// SchedFIFO defines First-In-First-Out real-time scheduling policy.
+	// Used in: process_monitor.go for critical audio threads requiring deterministic timing
+	// Impact: Provides real-time scheduling but may starve other processes if misused.
+	// Value 1 corresponds to SCHED_FIFO in Linux kernel.
+	SchedFIFO int
+
+	// SchedRR defines Round-Robin real-time scheduling policy.
+	// Used in: process_monitor.go for real-time threads that should share CPU time
+	// Impact: Real-time scheduling with time slicing, balances determinism and fairness.
+	// Value 2 corresponds to SCHED_RR in Linux kernel.
+	SchedRR int
+
+	// Real-time Priority Levels - Priority values for real-time audio thread scheduling
+	// Used in: process_monitor.go for setting thread priorities
+	// Impact: Higher priorities get more CPU time but may affect system responsiveness
+
+	// RTAudioHighPriority defines highest priority for critical audio threads.
+	// Used in: process_monitor.go for time-critical audio processing (encoding/decoding)
+	// Impact: Ensures audio threads get CPU time but may impact system responsiveness.
+	// Default 80 provides high priority without completely starving other processes.
+	RTAudioHighPriority int
+
+	// RTAudioMediumPriority defines medium priority for important audio threads.
+	// Used in: process_monitor.go for audio I/O and buffering operations
+	// Impact: Good priority for audio operations while maintaining system balance.
+	// Default 60 provides elevated priority for audio without extreme impact.
+	RTAudioMediumPriority int
+
+	// RTAudioLowPriority defines low priority for background audio threads.
+	// Used in: process_monitor.go for audio monitoring and metrics collection
+	// Impact: Ensures audio background tasks run without impacting critical operations.
+	// Default 40 provides some priority elevation while remaining background.
+	RTAudioLowPriority int
+
+	// RTNormalPriority defines normal priority (no real-time scheduling).
+	// Used in: process_monitor.go for non-critical audio threads
+	// Impact: Standard scheduling priority, no special real-time guarantees.
+	// Default 0 uses normal kernel scheduling without real-time privileges.
+	RTNormalPriority int
 
 	// Process Management - Configuration for audio process lifecycle management
 	// Used in: supervisor.go for managing audio process restarts and recovery
@@ -859,12 +902,6 @@ type AudioConfigConstants struct {
 	// Default 200ms provides reasonable wait time for microphone access.
 	MicContentionTimeout time.Duration // 200ms contention timeout
 
-	// Subprocess Pre-warming Configuration
-	// Used in: input_supervisor.go for reducing microphone activation latency
-	// Impact: Pre-warms audio input subprocess during startup to eliminate cold start delay
-	// Default true enables pre-warming for optimal user experience
-	EnableSubprocessPrewarming bool // Enable subprocess pre-warming (default: true)
-
 	// Priority Scheduler Configuration - Settings for process priority management
 	// Used in: priority_scheduler.go for system priority control
 	// Impact: Controls valid range for process priority adjustments
@@ -1169,24 +1206,6 @@ type AudioConfigConstants struct {
 	// Default 5ms provides quick batch processing with reasonable timeout.
 	BatchProcessorTimeout time.Duration
 
-	// BatchProcessorMaxQueueSize defines maximum queue size for batch operations.
-	// Used in: batch_audio.go for queue size control
-	// Impact: Larger queues reduce blocking but increase memory usage.
-	// Default 16 provides good balance between memory and performance.
-	BatchProcessorMaxQueueSize int
-
-	// BatchProcessorAdaptiveThreshold defines threshold for adaptive batch sizing.
-	// Used in: batch_audio.go for dynamic batch size adjustment
-	// Impact: Lower thresholds enable more aggressive batching.
-	// Default 0.8 enables batching when 80% of queue is full.
-	BatchProcessorAdaptiveThreshold float64
-
-	// BatchProcessorThreadPinningThreshold defines minimum batch size for thread pinning.
-	// Used in: batch_audio.go for OS thread pinning optimization
-	// Impact: Higher thresholds reduce thread pinning overhead.
-	// Default 8 frames enables pinning for larger batches only.
-	BatchProcessorThreadPinningThreshold int
-
 	// Output Streaming Constants - Configuration for audio output streaming
 	// Used in: output_streaming.go for output stream timing control
 	// Impact: Controls output streaming frame rate and timing
@@ -1483,7 +1502,36 @@ type AudioConfigConstants struct {
 	// Default 512 bytes accommodates typical encoding variations.
 	FrameSizeTolerance int
 
-	// Removed device health monitoring configuration - functionality not used
+	// Device Health Monitoring Configuration
+	// Used in: device_health.go for proactive device monitoring and recovery
+	// Impact: Controls health check frequency and recovery thresholds
+
+	// HealthCheckIntervalMS defines interval between device health checks in milliseconds.
+	// Used in: DeviceHealthMonitor for periodic health assessment
+	// Impact: Lower values provide faster detection but increase CPU usage.
+	// Default 5000ms (5s) provides good balance between responsiveness and overhead.
+	HealthCheckIntervalMS int
+
+	// HealthRecoveryThreshold defines number of consecutive successful operations
+	// required to mark a device as healthy after being unhealthy.
+	// Used in: DeviceHealthMonitor for recovery state management
+	// Impact: Higher values prevent premature recovery declarations.
+	// Default 3 consecutive successes ensures stable recovery.
+	HealthRecoveryThreshold int
+
+	// HealthLatencyThresholdMS defines maximum acceptable latency in milliseconds
+	// before considering a device unhealthy.
+	// Used in: DeviceHealthMonitor for latency-based health assessment
+	// Impact: Lower values trigger recovery sooner but may cause false positives.
+	// Default 100ms provides reasonable threshold for real-time audio.
+	HealthLatencyThresholdMS int
+
+	// HealthErrorRateLimit defines maximum error rate (0.0-1.0) before
+	// considering a device unhealthy.
+	// Used in: DeviceHealthMonitor for error rate assessment
+	// Impact: Lower values trigger recovery sooner for error-prone devices.
+	// Default 0.1 (10%) allows some transient errors while detecting problems.
+	HealthErrorRateLimit float64
 
 	// Latency Histogram Bucket Configuration
 	// Used in: LatencyHistogram for granular latency measurement buckets
@@ -1770,6 +1818,47 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Used in: Non-critical audio processing tasks
 		// Impact: Provides standard scheduling suitable for non-critical tasks.
 		// Default 0 (SCHED_NORMAL) for standard time-sharing scheduling.
+		SchedNormal: 0,
+
+		// SchedFIFO defines real-time first-in-first-out scheduling policy.
+		// Used in: Critical audio processing requiring deterministic timing
+		// Impact: Provides deterministic scheduling for latency-critical operations.
+		// Default 1 (SCHED_FIFO) for real-time first-in-first-out scheduling.
+		SchedFIFO: 1,
+
+		// SchedRR defines real-time round-robin scheduling policy.
+		// Used in: Balanced real-time processing with time slicing
+		// Impact: Provides real-time scheduling with balanced time slicing.
+		// Default 2 (SCHED_RR) for real-time round-robin scheduling.
+		SchedRR: 2,
+
+		// Real-time Priority Levels - Configuration for process priorities
+		// Used in: Process priority management and CPU scheduling
+		// Impact: Controls priority hierarchy for audio system components
+
+		// RTAudioHighPriority defines highest priority for audio processing.
+		// Used in: Latency-critical audio operations and CPU priority assignment
+		// Impact: Ensures highest CPU priority without starving system processes.
+		// Default 80 provides highest priority for latency-critical operations.
+		RTAudioHighPriority: 80,
+
+		// RTAudioMediumPriority defines medium priority for audio tasks.
+		// Used in: Important audio tasks requiring elevated priority
+		// Impact: Provides elevated priority while allowing higher priority operations.
+		// Default 60 balances importance with system operation priority.
+		RTAudioMediumPriority: 60,
+
+		// RTAudioLowPriority defines low priority for audio tasks.
+		// Used in: Audio tasks needing responsiveness but not latency-critical
+		// Impact: Provides moderate real-time priority for responsive tasks.
+		// Default 40 ensures responsiveness without being latency-critical.
+		RTAudioLowPriority: 40,
+
+		// RTNormalPriority defines normal scheduling priority.
+		// Used in: Non-real-time audio processing tasks
+		// Impact: Provides standard priority for non-real-time operations.
+		// Default 0 represents normal scheduling priority.
+		RTNormalPriority: 0,
 
 		// Process Management - Configuration for process restart and recovery
 		// Used in: Process monitoring and failure recovery systems
@@ -2058,17 +2147,17 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Used in: process management, thread scheduling for audio processing
 		// Impact: Controls CPU scheduling priority for audio threads
 
-		// AudioHighPriority defines highest priority for critical audio threads (5).
+		// AudioHighPriority defines highest priority for critical audio threads (-10).
 		// Used in: Real-time audio processing threads, encoder/decoder threads
-		// Impact: Ensures audio threads get CPU time but prioritizes mouse input
+		// Impact: Ensures audio threads get CPU time before other processes
-		// Modified to 5 to prevent mouse lag on single-core RV1106
+		// Default -10 provides high priority without requiring root privileges
-		AudioHighPriority: 5,
+		AudioHighPriority: -10,
 
-		// AudioMediumPriority defines medium priority for important audio threads (10).
+		// AudioMediumPriority defines medium priority for important audio threads (-5).
 		// Used in: Audio buffer management, IPC communication threads
 		// Impact: Balances audio performance with system responsiveness
-		// Modified to 10 to prioritize mouse input on single-core RV1106
+		// Default -5 ensures good performance while allowing other critical tasks
-		AudioMediumPriority: 10,
+		AudioMediumPriority: -5,
 
 		// AudioLowPriority defines low priority for non-critical audio threads (0).
 		// Used in: Metrics collection, logging, cleanup tasks
@@ -2082,11 +2171,11 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Default 0 represents normal Linux process priority
 		NormalPriority: 0,
 
-		// NiceValue defines default nice value for audio processes (5).
+		// NiceValue defines default nice value for audio processes (-10).
 		// Used in: Process creation, priority adjustment for audio components
-		// Impact: Ensures audio processes don't interfere with mouse input
+		// Impact: Improves audio process scheduling without requiring special privileges
-		// Modified to 5 to prioritize mouse input on single-core RV1106
+		// Default -10 provides better scheduling while remaining accessible to non-root users
-		NiceValue: 5,
+		NiceValue: -10,
 
 		// Error Handling - Configuration for robust error recovery and retry logic
 		// Used in: Throughout audio pipeline for handling transient failures
@@ -2254,9 +2343,6 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Microphone Contention Configuration
 		MicContentionTimeout: 200 * time.Millisecond,
 
-		// Subprocess Pre-warming Configuration
-		EnableSubprocessPrewarming: true,
-
 		// Priority Scheduler Configuration
 		MinNiceValue: -20,
 		MaxNiceValue: 19,
@@ -2336,11 +2422,8 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		ProcessMonitorTraditionalHz:   100.0,  // 100.0 Hz traditional clock
 
 		// Batch Processing Constants
-		BatchProcessorFramesPerBatch:         4,                    // 4 frames per batch
+		BatchProcessorFramesPerBatch: 4,                    // 4 frames per batch
-		BatchProcessorTimeout:                5 * time.Millisecond, // 5ms timeout
+		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)
@@ -2421,7 +2504,11 @@ 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
+		// Device Health Monitoring Configuration
+		HealthCheckIntervalMS:    5000, // 5000ms (5s) health check interval
+		HealthRecoveryThreshold:  3,    // 3 consecutive successes for recovery
+		HealthLatencyThresholdMS: 100,  // 100ms latency threshold for health
+		HealthErrorRateLimit:     0.1,  // 10% error rate limit for health
 
 		// Latency Histogram Bucket Configuration
 		LatencyBucket10ms:  10 * time.Millisecond,  // 10ms latency bucket
@@ -2438,9 +2525,6 @@ func DefaultAudioConfig() *AudioConfigConstants {
 
 		// Goroutine Monitoring Configuration
 		GoroutineMonitorInterval: 30 * time.Second, // 30s monitoring interval
-
-		// Performance Configuration Flags - Production optimizations
-
 	}
 }
 

internal/audio/device_health.go

@@ -0,0 +1,514 @@
+package audio
+
+import (
+	"context"
+	"fmt"
+	"sync"
+	"sync/atomic"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// DeviceHealthStatus represents the health status of an audio device
+type DeviceHealthStatus int
+
+const (
+	DeviceHealthUnknown DeviceHealthStatus = iota
+	DeviceHealthHealthy
+	DeviceHealthDegraded
+	DeviceHealthFailing
+	DeviceHealthCritical
+)
+
+func (s DeviceHealthStatus) String() string {
+	switch s {
+	case DeviceHealthHealthy:
+		return "healthy"
+	case DeviceHealthDegraded:
+		return "degraded"
+	case DeviceHealthFailing:
+		return "failing"
+	case DeviceHealthCritical:
+		return "critical"
+	default:
+		return "unknown"
+	}
+}
+
+// DeviceHealthMetrics tracks health-related metrics for audio devices
+type DeviceHealthMetrics struct {
+	// Error tracking
+	ConsecutiveErrors int64     `json:"consecutive_errors"`
+	TotalErrors       int64     `json:"total_errors"`
+	LastErrorTime     time.Time `json:"last_error_time"`
+	ErrorRate         float64   `json:"error_rate"` // errors per minute
+
+	// Performance metrics
+	AverageLatency time.Duration `json:"average_latency"`
+	MaxLatency     time.Duration `json:"max_latency"`
+	LatencySpikes  int64         `json:"latency_spikes"`
+	Underruns      int64         `json:"underruns"`
+	Overruns       int64         `json:"overruns"`
+
+	// Device availability
+	LastSuccessfulOp     time.Time `json:"last_successful_op"`
+	DeviceDisconnects    int64     `json:"device_disconnects"`
+	RecoveryAttempts     int64     `json:"recovery_attempts"`
+	SuccessfulRecoveries int64     `json:"successful_recoveries"`
+
+	// Health assessment
+	CurrentStatus     DeviceHealthStatus `json:"current_status"`
+	StatusLastChanged time.Time          `json:"status_last_changed"`
+	HealthScore       float64            `json:"health_score"` // 0.0 to 1.0
+}
+
+// DeviceHealthMonitor monitors the health of audio devices and triggers recovery
+type DeviceHealthMonitor struct {
+	// Atomic fields first for ARM32 alignment
+	running           int32
+	monitoringEnabled int32
+
+	// Configuration
+	checkInterval     time.Duration
+	recoveryThreshold int
+	latencyThreshold  time.Duration
+	errorRateLimit    float64 // max errors per minute
+
+	// State tracking
+	captureMetrics  *DeviceHealthMetrics
+	playbackMetrics *DeviceHealthMetrics
+	mutex           sync.RWMutex
+
+	// Control channels
+	ctx      context.Context
+	cancel   context.CancelFunc
+	stopChan chan struct{}
+	doneChan chan struct{}
+
+	// Recovery callbacks
+	recoveryCallbacks map[string]func() error
+	callbackMutex     sync.RWMutex
+
+	// Logging
+	logger zerolog.Logger
+	config *AudioConfigConstants
+}
+
+// NewDeviceHealthMonitor creates a new device health monitor
+func NewDeviceHealthMonitor() *DeviceHealthMonitor {
+	ctx, cancel := context.WithCancel(context.Background())
+	config := GetConfig()
+
+	return &DeviceHealthMonitor{
+		checkInterval:     time.Duration(config.HealthCheckIntervalMS) * time.Millisecond,
+		recoveryThreshold: config.HealthRecoveryThreshold,
+		latencyThreshold:  time.Duration(config.HealthLatencyThresholdMS) * time.Millisecond,
+		errorRateLimit:    config.HealthErrorRateLimit,
+		captureMetrics: &DeviceHealthMetrics{
+			CurrentStatus: DeviceHealthUnknown,
+			HealthScore:   1.0,
+		},
+		playbackMetrics: &DeviceHealthMetrics{
+			CurrentStatus: DeviceHealthUnknown,
+			HealthScore:   1.0,
+		},
+		ctx:               ctx,
+		cancel:            cancel,
+		stopChan:          make(chan struct{}),
+		doneChan:          make(chan struct{}),
+		recoveryCallbacks: make(map[string]func() error),
+		logger:            logging.GetDefaultLogger().With().Str("component", "device-health-monitor").Logger(),
+		config:            config,
+	}
+}
+
+// Start begins health monitoring
+func (dhm *DeviceHealthMonitor) Start() error {
+	if !atomic.CompareAndSwapInt32(&dhm.running, 0, 1) {
+		return fmt.Errorf("device health monitor already running")
+	}
+
+	dhm.logger.Debug().Msg("device health monitor starting")
+	atomic.StoreInt32(&dhm.monitoringEnabled, 1)
+
+	go dhm.monitoringLoop()
+	return nil
+}
+
+// Stop stops health monitoring
+func (dhm *DeviceHealthMonitor) Stop() {
+	if !atomic.CompareAndSwapInt32(&dhm.running, 1, 0) {
+		return
+	}
+
+	dhm.logger.Debug().Msg("device health monitor stopping")
+	atomic.StoreInt32(&dhm.monitoringEnabled, 0)
+
+	close(dhm.stopChan)
+	dhm.cancel()
+
+	// Wait for monitoring loop to finish
+	select {
+	case <-dhm.doneChan:
+		dhm.logger.Debug().Msg("device health monitor stopped")
+	case <-time.After(time.Duration(dhm.config.SupervisorTimeout)):
+		dhm.logger.Warn().Msg("device health monitor stop timeout")
+	}
+}
+
+// RegisterRecoveryCallback registers a recovery function for a specific component
+func (dhm *DeviceHealthMonitor) RegisterRecoveryCallback(component string, callback func() error) {
+	dhm.callbackMutex.Lock()
+	defer dhm.callbackMutex.Unlock()
+	dhm.recoveryCallbacks[component] = callback
+	dhm.logger.Debug().Str("component", component).Msg("registered recovery callback")
+}
+
+// RecordError records an error for health tracking
+func (dhm *DeviceHealthMonitor) RecordError(deviceType string, err error) {
+	if atomic.LoadInt32(&dhm.monitoringEnabled) == 0 {
+		return
+	}
+
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	var metrics *DeviceHealthMetrics
+	switch deviceType {
+	case "capture":
+		metrics = dhm.captureMetrics
+	case "playback":
+		metrics = dhm.playbackMetrics
+	default:
+		dhm.logger.Warn().Str("device_type", deviceType).Msg("unknown device type for error recording")
+		return
+	}
+
+	atomic.AddInt64(&metrics.ConsecutiveErrors, 1)
+	atomic.AddInt64(&metrics.TotalErrors, 1)
+	metrics.LastErrorTime = time.Now()
+
+	// Update error rate (errors per minute)
+	if !metrics.LastErrorTime.IsZero() {
+		timeSinceFirst := time.Since(metrics.LastErrorTime)
+		if timeSinceFirst > 0 {
+			metrics.ErrorRate = float64(metrics.TotalErrors) / timeSinceFirst.Minutes()
+		}
+	}
+
+	dhm.logger.Debug().
+		Str("device_type", deviceType).
+		Err(err).
+		Int64("consecutive_errors", metrics.ConsecutiveErrors).
+		Float64("error_rate", metrics.ErrorRate).
+		Msg("recorded device error")
+
+	// Trigger immediate health assessment
+	dhm.assessDeviceHealth(deviceType, metrics)
+}
+
+// RecordSuccess records a successful operation
+func (dhm *DeviceHealthMonitor) RecordSuccess(deviceType string) {
+	if atomic.LoadInt32(&dhm.monitoringEnabled) == 0 {
+		return
+	}
+
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	var metrics *DeviceHealthMetrics
+	switch deviceType {
+	case "capture":
+		metrics = dhm.captureMetrics
+	case "playback":
+		metrics = dhm.playbackMetrics
+	default:
+		return
+	}
+
+	// Reset consecutive errors on success
+	atomic.StoreInt64(&metrics.ConsecutiveErrors, 0)
+	metrics.LastSuccessfulOp = time.Now()
+
+	// Improve health score gradually
+	if metrics.HealthScore < 1.0 {
+		metrics.HealthScore = min(1.0, metrics.HealthScore+0.1)
+	}
+}
+
+// RecordLatency records operation latency for health assessment
+func (dhm *DeviceHealthMonitor) RecordLatency(deviceType string, latency time.Duration) {
+	if atomic.LoadInt32(&dhm.monitoringEnabled) == 0 {
+		return
+	}
+
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	var metrics *DeviceHealthMetrics
+	switch deviceType {
+	case "capture":
+		metrics = dhm.captureMetrics
+	case "playback":
+		metrics = dhm.playbackMetrics
+	default:
+		return
+	}
+
+	// Update latency metrics
+	if metrics.AverageLatency == 0 {
+		metrics.AverageLatency = latency
+	} else {
+		// Exponential moving average
+		metrics.AverageLatency = time.Duration(float64(metrics.AverageLatency)*0.9 + float64(latency)*0.1)
+	}
+
+	if latency > metrics.MaxLatency {
+		metrics.MaxLatency = latency
+	}
+
+	// Track latency spikes
+	if latency > dhm.latencyThreshold {
+		atomic.AddInt64(&metrics.LatencySpikes, 1)
+	}
+}
+
+// RecordUnderrun records an audio underrun event
+func (dhm *DeviceHealthMonitor) RecordUnderrun(deviceType string) {
+	if atomic.LoadInt32(&dhm.monitoringEnabled) == 0 {
+		return
+	}
+
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	var metrics *DeviceHealthMetrics
+	switch deviceType {
+	case "capture":
+		metrics = dhm.captureMetrics
+	case "playback":
+		metrics = dhm.playbackMetrics
+	default:
+		return
+	}
+
+	atomic.AddInt64(&metrics.Underruns, 1)
+	dhm.logger.Debug().Str("device_type", deviceType).Msg("recorded audio underrun")
+}
+
+// RecordOverrun records an audio overrun event
+func (dhm *DeviceHealthMonitor) RecordOverrun(deviceType string) {
+	if atomic.LoadInt32(&dhm.monitoringEnabled) == 0 {
+		return
+	}
+
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	var metrics *DeviceHealthMetrics
+	switch deviceType {
+	case "capture":
+		metrics = dhm.captureMetrics
+	case "playback":
+		metrics = dhm.playbackMetrics
+	default:
+		return
+	}
+
+	atomic.AddInt64(&metrics.Overruns, 1)
+	dhm.logger.Debug().Str("device_type", deviceType).Msg("recorded audio overrun")
+}
+
+// GetHealthMetrics returns current health metrics
+func (dhm *DeviceHealthMonitor) GetHealthMetrics() (capture, playback DeviceHealthMetrics) {
+	dhm.mutex.RLock()
+	defer dhm.mutex.RUnlock()
+	return *dhm.captureMetrics, *dhm.playbackMetrics
+}
+
+// monitoringLoop runs the main health monitoring loop
+func (dhm *DeviceHealthMonitor) monitoringLoop() {
+	defer close(dhm.doneChan)
+
+	ticker := time.NewTicker(dhm.checkInterval)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-dhm.stopChan:
+			return
+		case <-dhm.ctx.Done():
+			return
+		case <-ticker.C:
+			dhm.performHealthCheck()
+		}
+	}
+}
+
+// performHealthCheck performs a comprehensive health check
+func (dhm *DeviceHealthMonitor) performHealthCheck() {
+	dhm.mutex.Lock()
+	defer dhm.mutex.Unlock()
+
+	// Assess health for both devices
+	dhm.assessDeviceHealth("capture", dhm.captureMetrics)
+	dhm.assessDeviceHealth("playback", dhm.playbackMetrics)
+
+	// Check if recovery is needed
+	dhm.checkRecoveryNeeded("capture", dhm.captureMetrics)
+	dhm.checkRecoveryNeeded("playback", dhm.playbackMetrics)
+}
+
+// assessDeviceHealth assesses the health status of a device
+func (dhm *DeviceHealthMonitor) assessDeviceHealth(deviceType string, metrics *DeviceHealthMetrics) {
+	previousStatus := metrics.CurrentStatus
+	newStatus := dhm.calculateHealthStatus(metrics)
+
+	if newStatus != previousStatus {
+		metrics.CurrentStatus = newStatus
+		metrics.StatusLastChanged = time.Now()
+		dhm.logger.Info().
+			Str("device_type", deviceType).
+			Str("previous_status", previousStatus.String()).
+			Str("new_status", newStatus.String()).
+			Float64("health_score", metrics.HealthScore).
+			Msg("device health status changed")
+	}
+
+	// Update health score
+	metrics.HealthScore = dhm.calculateHealthScore(metrics)
+}
+
+// calculateHealthStatus determines health status based on metrics
+func (dhm *DeviceHealthMonitor) calculateHealthStatus(metrics *DeviceHealthMetrics) DeviceHealthStatus {
+	consecutiveErrors := atomic.LoadInt64(&metrics.ConsecutiveErrors)
+	totalErrors := atomic.LoadInt64(&metrics.TotalErrors)
+
+	// Critical: Too many consecutive errors or device disconnected recently
+	if consecutiveErrors >= int64(dhm.recoveryThreshold) {
+		return DeviceHealthCritical
+	}
+
+	// Critical: No successful operations in a long time
+	if !metrics.LastSuccessfulOp.IsZero() && time.Since(metrics.LastSuccessfulOp) > time.Duration(dhm.config.SupervisorTimeout) {
+		return DeviceHealthCritical
+	}
+
+	// Failing: High error rate or frequent latency spikes
+	if metrics.ErrorRate > dhm.errorRateLimit || atomic.LoadInt64(&metrics.LatencySpikes) > int64(dhm.config.MaxDroppedFrames) {
+		return DeviceHealthFailing
+	}
+
+	// Degraded: Some errors or performance issues
+	if consecutiveErrors > 0 || totalErrors > int64(dhm.config.MaxDroppedFrames/2) || metrics.AverageLatency > dhm.latencyThreshold {
+		return DeviceHealthDegraded
+	}
+
+	// Healthy: No significant issues
+	return DeviceHealthHealthy
+}
+
+// calculateHealthScore calculates a numeric health score (0.0 to 1.0)
+func (dhm *DeviceHealthMonitor) calculateHealthScore(metrics *DeviceHealthMetrics) float64 {
+	score := 1.0
+
+	// Penalize consecutive errors
+	consecutiveErrors := atomic.LoadInt64(&metrics.ConsecutiveErrors)
+	if consecutiveErrors > 0 {
+		score -= float64(consecutiveErrors) * 0.1
+	}
+
+	// Penalize high error rate
+	if metrics.ErrorRate > 0 {
+		score -= min(0.5, metrics.ErrorRate/dhm.errorRateLimit*0.5)
+	}
+
+	// Penalize high latency
+	if metrics.AverageLatency > dhm.latencyThreshold {
+		excess := float64(metrics.AverageLatency-dhm.latencyThreshold) / float64(dhm.latencyThreshold)
+		score -= min(0.3, excess*0.3)
+	}
+
+	// Penalize underruns/overruns
+	underruns := atomic.LoadInt64(&metrics.Underruns)
+	overruns := atomic.LoadInt64(&metrics.Overruns)
+	if underruns+overruns > 0 {
+		score -= min(0.2, float64(underruns+overruns)*0.01)
+	}
+
+	return max(0.0, score)
+}
+
+// checkRecoveryNeeded checks if recovery is needed and triggers it
+func (dhm *DeviceHealthMonitor) checkRecoveryNeeded(deviceType string, metrics *DeviceHealthMetrics) {
+	if metrics.CurrentStatus == DeviceHealthCritical {
+		dhm.triggerRecovery(deviceType, metrics)
+	}
+}
+
+// triggerRecovery triggers recovery for a device
+func (dhm *DeviceHealthMonitor) triggerRecovery(deviceType string, metrics *DeviceHealthMetrics) {
+	atomic.AddInt64(&metrics.RecoveryAttempts, 1)
+
+	dhm.logger.Warn().
+		Str("device_type", deviceType).
+		Str("status", metrics.CurrentStatus.String()).
+		Int64("consecutive_errors", atomic.LoadInt64(&metrics.ConsecutiveErrors)).
+		Float64("error_rate", metrics.ErrorRate).
+		Msg("triggering device recovery")
+
+	// Try registered recovery callbacks
+	dhm.callbackMutex.RLock()
+	defer dhm.callbackMutex.RUnlock()
+
+	for component, callback := range dhm.recoveryCallbacks {
+		if callback != nil {
+			go func(comp string, cb func() error) {
+				if err := cb(); err != nil {
+					dhm.logger.Error().
+						Str("component", comp).
+						Str("device_type", deviceType).
+						Err(err).
+						Msg("recovery callback failed")
+				} else {
+					atomic.AddInt64(&metrics.SuccessfulRecoveries, 1)
+					dhm.logger.Info().
+						Str("component", comp).
+						Str("device_type", deviceType).
+						Msg("recovery callback succeeded")
+				}
+			}(component, callback)
+		}
+	}
+}
+
+// Global device health monitor instance
+var (
+	globalDeviceHealthMonitor *DeviceHealthMonitor
+	deviceHealthOnce          sync.Once
+)
+
+// GetDeviceHealthMonitor returns the global device health monitor
+func GetDeviceHealthMonitor() *DeviceHealthMonitor {
+	deviceHealthOnce.Do(func() {
+		globalDeviceHealthMonitor = NewDeviceHealthMonitor()
+	})
+	return globalDeviceHealthMonitor
+}
+
+// Helper functions for min/max
+func min(a, b float64) float64 {
+	if a < b {
+		return a
+	}
+	return b
+}
+
+func max(a, b float64) float64 {
+	if a > b {
+		return a
+	}
+	return b
+}

internal/audio/events.go

@@ -2,6 +2,7 @@ package audio
 
 import (
 	"context"
+	"fmt"
 	"strings"
 	"sync"
 	"time"
@@ -16,9 +17,13 @@ import (
 type AudioEventType string
 
 const (
-	AudioEventMuteChanged     AudioEventType = "audio-mute-changed"
+	AudioEventMuteChanged       AudioEventType = "audio-mute-changed"
-	AudioEventMicrophoneState AudioEventType = "microphone-state-changed"
+	AudioEventMetricsUpdate     AudioEventType = "audio-metrics-update"
-	AudioEventDeviceChanged   AudioEventType = "audio-device-changed"
+	AudioEventMicrophoneState   AudioEventType = "microphone-state-changed"
+	AudioEventMicrophoneMetrics AudioEventType = "microphone-metrics-update"
+	AudioEventProcessMetrics    AudioEventType = "audio-process-metrics"
+	AudioEventMicProcessMetrics AudioEventType = "microphone-process-metrics"
+	AudioEventDeviceChanged     AudioEventType = "audio-device-changed"
 )
 
 // AudioEvent represents a WebSocket audio event
@@ -32,12 +37,43 @@ type AudioMuteData struct {
 	Muted bool `json:"muted"`
 }
 
+// AudioMetricsData represents audio metrics data
+type AudioMetricsData struct {
+	FramesReceived  int64  `json:"frames_received"`
+	FramesDropped   int64  `json:"frames_dropped"`
+	BytesProcessed  int64  `json:"bytes_processed"`
+	LastFrameTime   string `json:"last_frame_time"`
+	ConnectionDrops int64  `json:"connection_drops"`
+	AverageLatency  string `json:"average_latency"`
+}
+
 // MicrophoneStateData represents microphone state data
 type MicrophoneStateData struct {
 	Running       bool `json:"running"`
 	SessionActive bool `json:"session_active"`
 }
 
+// MicrophoneMetricsData represents microphone metrics data
+type MicrophoneMetricsData struct {
+	FramesSent      int64  `json:"frames_sent"`
+	FramesDropped   int64  `json:"frames_dropped"`
+	BytesProcessed  int64  `json:"bytes_processed"`
+	LastFrameTime   string `json:"last_frame_time"`
+	ConnectionDrops int64  `json:"connection_drops"`
+	AverageLatency  string `json:"average_latency"`
+}
+
+// ProcessMetricsData represents process metrics data for WebSocket events
+type ProcessMetricsData struct {
+	PID           int     `json:"pid"`
+	CPUPercent    float64 `json:"cpu_percent"`
+	MemoryRSS     int64   `json:"memory_rss"`
+	MemoryVMS     int64   `json:"memory_vms"`
+	MemoryPercent float64 `json:"memory_percent"`
+	Running       bool    `json:"running"`
+	ProcessName   string  `json:"process_name"`
+}
+
 // AudioDeviceChangedData represents audio device configuration change data
 type AudioDeviceChangedData struct {
 	Enabled bool   `json:"enabled"`
@@ -70,6 +106,12 @@ func initializeBroadcaster() {
 		subscribers: make(map[string]*AudioEventSubscriber),
 		logger:      &l,
 	}
+
+	// Start metrics broadcasting goroutine
+	go audioEventBroadcaster.startMetricsBroadcasting()
+
+	// Start granular metrics logging with same interval as metrics broadcasting
+	// StartGranularMetricsLogging(GetMetricsUpdateInterval()) // Disabled to reduce log pollution
 }
 
 // InitializeAudioEventBroadcaster initializes the global audio event broadcaster
@@ -176,6 +218,90 @@ func (aeb *AudioEventBroadcaster) sendInitialState(connectionID string) {
 		},
 	}
 	aeb.sendToSubscriber(subscriber, micStateEvent)
+
+	// Send current metrics
+	aeb.sendCurrentMetrics(subscriber)
+}
+
+// convertAudioMetricsToEventDataWithLatencyMs converts internal audio metrics to AudioMetricsData with millisecond latency formatting
+func convertAudioMetricsToEventDataWithLatencyMs(metrics AudioMetrics) AudioMetricsData {
+	return AudioMetricsData{
+		FramesReceived:  metrics.FramesReceived,
+		FramesDropped:   metrics.FramesDropped,
+		BytesProcessed:  metrics.BytesProcessed,
+		LastFrameTime:   metrics.LastFrameTime.Format(GetConfig().EventTimeFormatString),
+		ConnectionDrops: metrics.ConnectionDrops,
+		AverageLatency:  fmt.Sprintf("%.1fms", float64(metrics.AverageLatency.Nanoseconds())/1e6),
+	}
+}
+
+// convertAudioInputMetricsToEventDataWithLatencyMs converts internal audio input metrics to MicrophoneMetricsData with millisecond latency formatting
+func convertAudioInputMetricsToEventDataWithLatencyMs(metrics AudioInputMetrics) MicrophoneMetricsData {
+	return MicrophoneMetricsData{
+		FramesSent:      metrics.FramesSent,
+		FramesDropped:   metrics.FramesDropped,
+		BytesProcessed:  metrics.BytesProcessed,
+		LastFrameTime:   metrics.LastFrameTime.Format(GetConfig().EventTimeFormatString),
+		ConnectionDrops: metrics.ConnectionDrops,
+		AverageLatency:  fmt.Sprintf("%.1fms", float64(metrics.AverageLatency.Nanoseconds())/1e6),
+	}
+}
+
+// convertProcessMetricsToEventData converts internal process metrics to ProcessMetricsData for events
+func convertProcessMetricsToEventData(metrics ProcessMetrics, running bool) ProcessMetricsData {
+	return ProcessMetricsData{
+		PID:           metrics.PID,
+		CPUPercent:    metrics.CPUPercent,
+		MemoryRSS:     metrics.MemoryRSS,
+		MemoryVMS:     metrics.MemoryVMS,
+		MemoryPercent: metrics.MemoryPercent,
+		Running:       running,
+		ProcessName:   metrics.ProcessName,
+	}
+}
+
+// createProcessMetricsData creates ProcessMetricsData from ProcessMetrics with running status
+func createProcessMetricsData(metrics *ProcessMetrics, running bool, processName string) ProcessMetricsData {
+	if metrics == nil {
+		return ProcessMetricsData{
+			PID:           0,
+			CPUPercent:    0.0,
+			MemoryRSS:     0,
+			MemoryVMS:     0,
+			MemoryPercent: 0.0,
+			Running:       false,
+			ProcessName:   processName,
+		}
+	}
+	return ProcessMetricsData{
+		PID:           metrics.PID,
+		CPUPercent:    metrics.CPUPercent,
+		MemoryRSS:     metrics.MemoryRSS,
+		MemoryVMS:     metrics.MemoryVMS,
+		MemoryPercent: metrics.MemoryPercent,
+		Running:       running,
+		ProcessName:   metrics.ProcessName,
+	}
+}
+
+// getInactiveProcessMetrics returns ProcessMetricsData for an inactive audio input process
+func getInactiveProcessMetrics() ProcessMetricsData {
+	return createProcessMetricsData(nil, false, "audio-input-server")
+}
+
+// getActiveAudioInputSupervisor safely retrieves the audio input supervisor if session is active
+func getActiveAudioInputSupervisor() *AudioInputSupervisor {
+	sessionProvider := GetSessionProvider()
+	if !sessionProvider.IsSessionActive() {
+		return nil
+	}
+
+	inputManager := sessionProvider.GetAudioInputManager()
+	if inputManager == nil {
+		return nil
+	}
+
+	return inputManager.GetSupervisor()
 }
 
 // createAudioEvent creates an AudioEvent
@@ -186,6 +312,122 @@ func createAudioEvent(eventType AudioEventType, data interface{}) AudioEvent {
 	}
 }
 
+func (aeb *AudioEventBroadcaster) getMicrophoneProcessMetrics() ProcessMetricsData {
+	inputSupervisor := getActiveAudioInputSupervisor()
+	if inputSupervisor == nil {
+		return getInactiveProcessMetrics()
+	}
+
+	processMetrics := inputSupervisor.GetProcessMetrics()
+	if processMetrics == nil {
+		return getInactiveProcessMetrics()
+	}
+
+	// If process is running but CPU is 0%, it means we're waiting for the second sample
+	// to calculate CPU percentage. Return metrics with correct running status.
+	if inputSupervisor.IsRunning() && processMetrics.CPUPercent == 0.0 {
+		return createProcessMetricsData(processMetrics, true, processMetrics.ProcessName)
+	}
+
+	// Subprocess is running, return actual metrics
+	return createProcessMetricsData(processMetrics, inputSupervisor.IsRunning(), processMetrics.ProcessName)
+}
+
+// sendCurrentMetrics sends current audio and microphone metrics to a subscriber
+func (aeb *AudioEventBroadcaster) sendCurrentMetrics(subscriber *AudioEventSubscriber) {
+	// Send audio metrics
+	audioMetrics := GetAudioMetrics()
+	audioMetricsEvent := createAudioEvent(AudioEventMetricsUpdate, convertAudioMetricsToEventDataWithLatencyMs(audioMetrics))
+	aeb.sendToSubscriber(subscriber, audioMetricsEvent)
+
+	// Send audio process metrics
+	if outputSupervisor := GetAudioOutputSupervisor(); outputSupervisor != nil {
+		if processMetrics := outputSupervisor.GetProcessMetrics(); processMetrics != nil {
+			audioProcessEvent := createAudioEvent(AudioEventProcessMetrics, convertProcessMetricsToEventData(*processMetrics, outputSupervisor.IsRunning()))
+			aeb.sendToSubscriber(subscriber, audioProcessEvent)
+		}
+	}
+
+	// Send microphone metrics using session provider
+	sessionProvider := GetSessionProvider()
+	if sessionProvider.IsSessionActive() {
+		if inputManager := sessionProvider.GetAudioInputManager(); inputManager != nil {
+			micMetrics := inputManager.GetMetrics()
+			micMetricsEvent := createAudioEvent(AudioEventMicrophoneMetrics, convertAudioInputMetricsToEventDataWithLatencyMs(micMetrics))
+			aeb.sendToSubscriber(subscriber, micMetricsEvent)
+		}
+	}
+
+	// Send microphone process metrics (always send, even when subprocess is not running)
+	micProcessEvent := createAudioEvent(AudioEventMicProcessMetrics, aeb.getMicrophoneProcessMetrics())
+	aeb.sendToSubscriber(subscriber, micProcessEvent)
+}
+
+// startMetricsBroadcasting starts a goroutine that periodically broadcasts metrics
+func (aeb *AudioEventBroadcaster) startMetricsBroadcasting() {
+	// Use centralized interval to match process monitor frequency for synchronized metrics
+	ticker := time.NewTicker(GetMetricsUpdateInterval())
+	defer ticker.Stop()
+
+	for range ticker.C {
+		aeb.mutex.RLock()
+		subscriberCount := len(aeb.subscribers)
+
+		// Early exit if no subscribers to save CPU
+		if subscriberCount == 0 {
+			aeb.mutex.RUnlock()
+			continue
+		}
+
+		// Create a copy for safe iteration
+		subscribersCopy := make([]*AudioEventSubscriber, 0, subscriberCount)
+		for _, sub := range aeb.subscribers {
+			subscribersCopy = append(subscribersCopy, sub)
+		}
+		aeb.mutex.RUnlock()
+
+		// Pre-check for cancelled contexts to avoid unnecessary work
+		activeSubscribers := 0
+		for _, sub := range subscribersCopy {
+			if sub.ctx.Err() == nil {
+				activeSubscribers++
+			}
+		}
+
+		// Skip metrics gathering if no active subscribers
+		if activeSubscribers == 0 {
+			continue
+		}
+
+		// Broadcast audio metrics
+		audioMetrics := GetAudioMetrics()
+		audioMetricsEvent := createAudioEvent(AudioEventMetricsUpdate, convertAudioMetricsToEventDataWithLatencyMs(audioMetrics))
+		aeb.broadcast(audioMetricsEvent)
+
+		// Broadcast microphone metrics if available using session provider
+		sessionProvider := GetSessionProvider()
+		if sessionProvider.IsSessionActive() {
+			if inputManager := sessionProvider.GetAudioInputManager(); inputManager != nil {
+				micMetrics := inputManager.GetMetrics()
+				micMetricsEvent := createAudioEvent(AudioEventMicrophoneMetrics, convertAudioInputMetricsToEventDataWithLatencyMs(micMetrics))
+				aeb.broadcast(micMetricsEvent)
+			}
+		}
+
+		// Broadcast audio process metrics
+		if outputSupervisor := GetAudioOutputSupervisor(); outputSupervisor != nil {
+			if processMetrics := outputSupervisor.GetProcessMetrics(); processMetrics != nil {
+				audioProcessEvent := createAudioEvent(AudioEventProcessMetrics, convertProcessMetricsToEventData(*processMetrics, outputSupervisor.IsRunning()))
+				aeb.broadcast(audioProcessEvent)
+			}
+		}
+
+		// Broadcast microphone process metrics (always broadcast, even when subprocess is not running)
+		micProcessEvent := createAudioEvent(AudioEventMicProcessMetrics, aeb.getMicrophoneProcessMetrics())
+		aeb.broadcast(micProcessEvent)
+	}
+}
+
 // broadcast sends an event to all subscribers
 func (aeb *AudioEventBroadcaster) broadcast(event AudioEvent) {
 	aeb.mutex.RLock()

internal/audio/goroutine_monitor.go

@@ -133,7 +133,8 @@ func GetGoroutineMonitor() *GoroutineMonitor {
 
 // StartGoroutineMonitoring starts the global goroutine monitor
 func StartGoroutineMonitoring() {
-	// Goroutine monitoring disabled
+	monitor := GetGoroutineMonitor()
+	monitor.Start()
 }
 
 // StopGoroutineMonitoring stops the global goroutine monitor

internal/audio/granular_metrics.go

@@ -0,0 +1,263 @@
+package audio
+
+import (
+	"sync"
+	"sync/atomic"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// LatencyPercentiles holds calculated percentile values
+type LatencyPercentiles struct {
+	P50 time.Duration `json:"p50"`
+	P95 time.Duration `json:"p95"`
+	P99 time.Duration `json:"p99"`
+	Min time.Duration `json:"min"`
+	Max time.Duration `json:"max"`
+	Avg time.Duration `json:"avg"`
+}
+
+// BufferPoolEfficiencyMetrics tracks detailed buffer pool performance
+type BufferPoolEfficiencyMetrics struct {
+	// Pool utilization metrics
+	HitRate           float64 `json:"hit_rate"`
+	MissRate          float64 `json:"miss_rate"`
+	UtilizationRate   float64 `json:"utilization_rate"`
+	FragmentationRate float64 `json:"fragmentation_rate"`
+
+	// Memory efficiency metrics
+	MemoryEfficiency   float64 `json:"memory_efficiency"`
+	AllocationOverhead float64 `json:"allocation_overhead"`
+	ReuseEffectiveness float64 `json:"reuse_effectiveness"`
+
+	// Performance metrics
+	AverageGetLatency time.Duration `json:"average_get_latency"`
+	AveragePutLatency time.Duration `json:"average_put_latency"`
+	Throughput        float64       `json:"throughput"` // Operations per second
+}
+
+// GranularMetricsCollector aggregates all granular metrics
+type GranularMetricsCollector struct {
+	// Buffer pool efficiency tracking
+	framePoolMetrics   *BufferPoolEfficiencyTracker
+	controlPoolMetrics *BufferPoolEfficiencyTracker
+	zeroCopyMetrics    *BufferPoolEfficiencyTracker
+
+	mutex  sync.RWMutex
+	logger zerolog.Logger
+}
+
+// BufferPoolEfficiencyTracker tracks detailed efficiency metrics for a buffer pool
+type BufferPoolEfficiencyTracker struct {
+	// Atomic counters
+	getOperations   int64 // Total get operations (atomic)
+	putOperations   int64 // Total put operations (atomic)
+	getLatencySum   int64 // Sum of get latencies in nanoseconds (atomic)
+	putLatencySum   int64 // Sum of put latencies in nanoseconds (atomic)
+	allocationBytes int64 // Total bytes allocated (atomic)
+	reuseCount      int64 // Number of successful reuses (atomic)
+
+	// Recent operation times for throughput calculation
+	recentOps []time.Time
+	opsMutex  sync.RWMutex
+
+	poolName string
+	logger   zerolog.Logger
+}
+
+// NewBufferPoolEfficiencyTracker creates a new efficiency tracker
+func NewBufferPoolEfficiencyTracker(poolName string, logger zerolog.Logger) *BufferPoolEfficiencyTracker {
+	return &BufferPoolEfficiencyTracker{
+		recentOps: make([]time.Time, 0, 1000), // Track last 1000 operations
+		poolName:  poolName,
+		logger:    logger,
+	}
+}
+
+// RecordGetOperation records a buffer get operation with its latency
+func (bpet *BufferPoolEfficiencyTracker) RecordGetOperation(latency time.Duration, wasHit bool) {
+	atomic.AddInt64(&bpet.getOperations, 1)
+	atomic.AddInt64(&bpet.getLatencySum, latency.Nanoseconds())
+
+	if wasHit {
+		atomic.AddInt64(&bpet.reuseCount, 1)
+	}
+
+	// Record operation time for throughput calculation
+	bpet.opsMutex.Lock()
+	now := time.Now()
+	if len(bpet.recentOps) >= 1000 {
+		bpet.recentOps = bpet.recentOps[1:]
+	}
+	bpet.recentOps = append(bpet.recentOps, now)
+	bpet.opsMutex.Unlock()
+}
+
+// RecordPutOperation records a buffer put operation with its latency
+func (bpet *BufferPoolEfficiencyTracker) RecordPutOperation(latency time.Duration, bufferSize int) {
+	atomic.AddInt64(&bpet.putOperations, 1)
+	atomic.AddInt64(&bpet.putLatencySum, latency.Nanoseconds())
+	atomic.AddInt64(&bpet.allocationBytes, int64(bufferSize))
+}
+
+// GetEfficiencyMetrics calculates current efficiency metrics
+func (bpet *BufferPoolEfficiencyTracker) GetEfficiencyMetrics() BufferPoolEfficiencyMetrics {
+	getOps := atomic.LoadInt64(&bpet.getOperations)
+	putOps := atomic.LoadInt64(&bpet.putOperations)
+	reuseCount := atomic.LoadInt64(&bpet.reuseCount)
+	getLatencySum := atomic.LoadInt64(&bpet.getLatencySum)
+	putLatencySum := atomic.LoadInt64(&bpet.putLatencySum)
+	allocationBytes := atomic.LoadInt64(&bpet.allocationBytes)
+
+	var hitRate, missRate, avgGetLatency, avgPutLatency float64
+	var throughput float64
+
+	if getOps > 0 {
+		hitRate = float64(reuseCount) / float64(getOps) * 100
+		missRate = 100 - hitRate
+		avgGetLatency = float64(getLatencySum) / float64(getOps)
+	}
+
+	if putOps > 0 {
+		avgPutLatency = float64(putLatencySum) / float64(putOps)
+	}
+
+	// Calculate throughput from recent operations
+	bpet.opsMutex.RLock()
+	if len(bpet.recentOps) > 1 {
+		timeSpan := bpet.recentOps[len(bpet.recentOps)-1].Sub(bpet.recentOps[0])
+		if timeSpan > 0 {
+			throughput = float64(len(bpet.recentOps)) / timeSpan.Seconds()
+		}
+	}
+	bpet.opsMutex.RUnlock()
+
+	// Calculate efficiency metrics
+	utilizationRate := hitRate  // Simplified: hit rate as utilization
+	memoryEfficiency := hitRate // Simplified: reuse rate as memory efficiency
+	reuseEffectiveness := hitRate
+
+	// Calculate fragmentation (simplified as inverse of hit rate)
+	fragmentationRate := missRate
+
+	// Calculate allocation overhead (simplified)
+	allocationOverhead := float64(0)
+	if getOps > 0 && allocationBytes > 0 {
+		allocationOverhead = float64(allocationBytes) / float64(getOps)
+	}
+
+	return BufferPoolEfficiencyMetrics{
+		HitRate:            hitRate,
+		MissRate:           missRate,
+		UtilizationRate:    utilizationRate,
+		FragmentationRate:  fragmentationRate,
+		MemoryEfficiency:   memoryEfficiency,
+		AllocationOverhead: allocationOverhead,
+		ReuseEffectiveness: reuseEffectiveness,
+		AverageGetLatency:  time.Duration(avgGetLatency),
+		AveragePutLatency:  time.Duration(avgPutLatency),
+		Throughput:         throughput,
+	}
+}
+
+// NewGranularMetricsCollector creates a new granular metrics collector
+func NewGranularMetricsCollector(logger zerolog.Logger) *GranularMetricsCollector {
+	return &GranularMetricsCollector{
+		framePoolMetrics:   NewBufferPoolEfficiencyTracker("frame_pool", logger.With().Str("pool", "frame").Logger()),
+		controlPoolMetrics: NewBufferPoolEfficiencyTracker("control_pool", logger.With().Str("pool", "control").Logger()),
+		zeroCopyMetrics:    NewBufferPoolEfficiencyTracker("zero_copy_pool", logger.With().Str("pool", "zero_copy").Logger()),
+		logger:             logger,
+	}
+}
+
+// RecordFramePoolOperation records frame pool operations
+func (gmc *GranularMetricsCollector) RecordFramePoolGet(latency time.Duration, wasHit bool) {
+	gmc.framePoolMetrics.RecordGetOperation(latency, wasHit)
+}
+
+func (gmc *GranularMetricsCollector) RecordFramePoolPut(latency time.Duration, bufferSize int) {
+	gmc.framePoolMetrics.RecordPutOperation(latency, bufferSize)
+}
+
+// RecordControlPoolOperation records control pool operations
+func (gmc *GranularMetricsCollector) RecordControlPoolGet(latency time.Duration, wasHit bool) {
+	gmc.controlPoolMetrics.RecordGetOperation(latency, wasHit)
+}
+
+func (gmc *GranularMetricsCollector) RecordControlPoolPut(latency time.Duration, bufferSize int) {
+	gmc.controlPoolMetrics.RecordPutOperation(latency, bufferSize)
+}
+
+// RecordZeroCopyOperation records zero-copy pool operations
+func (gmc *GranularMetricsCollector) RecordZeroCopyGet(latency time.Duration, wasHit bool) {
+	gmc.zeroCopyMetrics.RecordGetOperation(latency, wasHit)
+}
+
+func (gmc *GranularMetricsCollector) RecordZeroCopyPut(latency time.Duration, bufferSize int) {
+	gmc.zeroCopyMetrics.RecordPutOperation(latency, bufferSize)
+}
+
+// GetBufferPoolEfficiency returns efficiency metrics for all buffer pools
+func (gmc *GranularMetricsCollector) GetBufferPoolEfficiency() map[string]BufferPoolEfficiencyMetrics {
+	gmc.mutex.RLock()
+	defer gmc.mutex.RUnlock()
+
+	return map[string]BufferPoolEfficiencyMetrics{
+		"frame_pool":     gmc.framePoolMetrics.GetEfficiencyMetrics(),
+		"control_pool":   gmc.controlPoolMetrics.GetEfficiencyMetrics(),
+		"zero_copy_pool": gmc.zeroCopyMetrics.GetEfficiencyMetrics(),
+	}
+}
+
+// LogGranularMetrics logs comprehensive granular metrics
+func (gmc *GranularMetricsCollector) LogGranularMetrics() {
+	bufferEfficiency := gmc.GetBufferPoolEfficiency()
+
+	// Log buffer pool efficiency
+	for poolName, efficiency := range bufferEfficiency {
+		gmc.logger.Info().
+			Str("pool", poolName).
+			Float64("hit_rate", efficiency.HitRate).
+			Float64("miss_rate", efficiency.MissRate).
+			Float64("utilization_rate", efficiency.UtilizationRate).
+			Float64("memory_efficiency", efficiency.MemoryEfficiency).
+			Dur("avg_get_latency", efficiency.AverageGetLatency).
+			Dur("avg_put_latency", efficiency.AveragePutLatency).
+			Float64("throughput", efficiency.Throughput).
+			Msg("Buffer pool efficiency metrics")
+	}
+}
+
+// Global granular metrics collector instance
+var (
+	granularMetricsCollector *GranularMetricsCollector
+	granularMetricsOnce      sync.Once
+)
+
+// GetGranularMetricsCollector returns the global granular metrics collector
+func GetGranularMetricsCollector() *GranularMetricsCollector {
+	granularMetricsOnce.Do(func() {
+		logger := logging.GetDefaultLogger().With().Str("component", "granular-metrics").Logger()
+		granularMetricsCollector = NewGranularMetricsCollector(logger)
+	})
+	return granularMetricsCollector
+}
+
+// StartGranularMetricsLogging starts periodic granular metrics logging
+func StartGranularMetricsLogging(interval time.Duration) {
+	collector := GetGranularMetricsCollector()
+	logger := collector.logger
+
+	logger.Info().Dur("interval", interval).Msg("Starting granular metrics logging")
+
+	go func() {
+		ticker := time.NewTicker(interval)
+		defer ticker.Stop()
+
+		for range ticker.C {
+			collector.LogGranularMetrics()
+		}
+	}()
+}

internal/audio/granular_metrics_test.go

@@ -0,0 +1,100 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestGranularMetricsCollector tests the GranularMetricsCollector functionality
+func TestGranularMetricsCollector(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"GetGranularMetricsCollector", testGetGranularMetricsCollector},
+		{"ConcurrentCollectorAccess", testConcurrentCollectorAccess},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testGetGranularMetricsCollector tests singleton behavior
+func testGetGranularMetricsCollector(t *testing.T) {
+	collector1 := GetGranularMetricsCollector()
+	collector2 := GetGranularMetricsCollector()
+
+	require.NotNil(t, collector1)
+	require.NotNil(t, collector2)
+	assert.Same(t, collector1, collector2, "Should return the same singleton instance")
+}
+
+// testConcurrentCollectorAccess tests thread safety of the collector
+func testConcurrentCollectorAccess(t *testing.T) {
+	collector := GetGranularMetricsCollector()
+	require.NotNil(t, collector)
+
+	const numGoroutines = 10
+	const operationsPerGoroutine = 50
+
+	var wg sync.WaitGroup
+	wg.Add(numGoroutines)
+
+	// Concurrent buffer pool operations
+	for i := 0; i < numGoroutines; i++ {
+		go func(id int) {
+			defer wg.Done()
+			for j := 0; j < operationsPerGoroutine; j++ {
+				// Test buffer pool operations
+				latency := time.Duration(id*operationsPerGoroutine+j) * time.Microsecond
+				collector.RecordFramePoolGet(latency, true)
+				collector.RecordFramePoolPut(latency, 1024)
+			}
+		}(i)
+	}
+
+	wg.Wait()
+
+	// Verify collector is still functional
+	efficiency := collector.GetBufferPoolEfficiency()
+	assert.NotNil(t, efficiency)
+}
+
+func BenchmarkGranularMetricsCollector(b *testing.B) {
+	collector := GetGranularMetricsCollector()
+
+	b.Run("RecordFramePoolGet", func(b *testing.B) {
+		latency := 5 * time.Millisecond
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			collector.RecordFramePoolGet(latency, true)
+		}
+	})
+
+	b.Run("RecordFramePoolPut", func(b *testing.B) {
+		latency := 5 * time.Millisecond
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			collector.RecordFramePoolPut(latency, 1024)
+		}
+	})
+
+	b.Run("GetBufferPoolEfficiency", func(b *testing.B) {
+		// Pre-populate with some data
+		for i := 0; i < 100; i++ {
+			collector.RecordFramePoolGet(time.Duration(i)*time.Microsecond, true)
+			collector.RecordFramePoolPut(time.Duration(i)*time.Microsecond, 1024)
+		}
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = collector.GetBufferPoolEfficiency()
+		}
+	})
+}

internal/audio/input.go

@@ -65,9 +65,6 @@ func (aim *AudioInputManager) Stop() {
 
 	aim.logComponentStop(AudioInputManagerComponent)
 
-	// Flush any pending sampled metrics before stopping
-	aim.flushPendingMetrics()
-
 	// Stop the IPC-based audio input
 	aim.ipcManager.Stop()
 
@@ -109,8 +106,12 @@ func (aim *AudioInputManager) WriteOpusFrame(frame []byte) error {
 	}
 
 	if err != nil {
+		atomic.AddInt64(&aim.metrics.FramesDropped, 1)
 		return err
 	}
+
+	// Update metrics
+	atomic.AddInt64(&aim.framesSent, 1)
 	aim.recordFrameProcessed(len(frame))
 	aim.updateLatency(processingTime)
 
@@ -195,6 +196,26 @@ func (aim *AudioInputManager) GetComprehensiveMetrics() map[string]interface{} {
 	return comprehensiveMetrics
 }
 
+// LogPerformanceStats logs current performance statistics
+func (aim *AudioInputManager) LogPerformanceStats() {
+	metrics := aim.GetComprehensiveMetrics()
+
+	managerStats := metrics["manager"].(map[string]interface{})
+	ipcStats := metrics["ipc"].(map[string]interface{})
+	detailedStats := metrics["detailed"].(map[string]interface{})
+
+	aim.logger.Info().
+		Int64("manager_frames_sent", managerStats["frames_sent"].(int64)).
+		Int64("manager_frames_dropped", managerStats["frames_dropped"].(int64)).
+		Float64("manager_latency_ms", managerStats["average_latency_ms"].(float64)).
+		Int64("ipc_frames_sent", ipcStats["frames_sent"].(int64)).
+		Int64("ipc_frames_dropped", ipcStats["frames_dropped"].(int64)).
+		Float64("ipc_latency_ms", ipcStats["average_latency_ms"].(float64)).
+		Float64("client_drop_rate", detailedStats["client_drop_rate"].(float64)).
+		Float64("frames_per_second", detailedStats["frames_per_second"].(float64)).
+		Msg("Audio input performance metrics")
+}
+
 // IsRunning returns whether the audio input manager is running
 // This checks both the internal state and existing system processes
 func (aim *AudioInputManager) IsRunning() bool {

internal/audio/input_ipc.go

@@ -292,6 +292,9 @@ func (ais *AudioInputServer) Start() error {
 	// Submit the connection acceptor to the audio reader pool
 	if !SubmitAudioReaderTask(ais.acceptConnections) {
 		// If the pool is full or shutting down, fall back to direct goroutine creation
+		logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
+		logger.Warn().Msg("Audio reader pool full or shutting down, falling back to direct goroutine creation")
+
 		go ais.acceptConnections()
 	}
 
@@ -366,6 +369,9 @@ func (ais *AudioInputServer) acceptConnections() {
 		// Handle this connection using the goroutine pool
 		if !SubmitAudioReaderTask(func() { ais.handleConnection(conn) }) {
 			// If the pool is full or shutting down, fall back to direct goroutine creation
+			logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
+			logger.Warn().Msg("Audio reader pool full or shutting down, falling back to direct goroutine creation")
+
 			go ais.handleConnection(conn)
 		}
 	}
@@ -936,12 +942,9 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 
 						// If too many consecutive errors, close connection to force reconnect
 						if consecutiveErrors >= maxConsecutiveErrors {
-							// Only log critical errors to reduce hotpath overhead
+							logger.Error().
-							if logger.GetLevel() <= zerolog.ErrorLevel {
+								Int("consecutive_errors", consecutiveErrors).
-								logger.Error().
+								Msg("Too many consecutive read errors, closing connection")
-									Int("consecutive_errors", consecutiveErrors).
-									Msg("Too many consecutive read errors, closing connection")
-							}
 
 							ais.mtx.Lock()
 							if ais.conn != nil {
@@ -958,10 +961,7 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 					// Reset error counter on successful read
 					if consecutiveErrors > 0 {
 						consecutiveErrors = 0
-						// Only log recovery info if debug level enabled to reduce overhead
+						logger.Info().Msg("Input connection recovered")
-						if logger.GetLevel() <= zerolog.InfoLevel {
-							logger.Info().Msg("Input connection recovered")
-						}
 					}
 
 					// Send to message channel with non-blocking write
@@ -971,11 +971,7 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 					default:
 						// Channel full, drop message
 						atomic.AddInt64(&ais.droppedFrames, 1)
-						// Avoid sampling logic in critical path - only log if warn level enabled
+						logger.Warn().Msg("Message channel full, dropping frame")
-						if logger.GetLevel() <= zerolog.WarnLevel {
-							droppedCount := atomic.LoadInt64(&ais.droppedFrames)
-							logger.Warn().Int64("total_dropped", droppedCount).Msg("Message channel full, dropping frame")
-						}
 					}
 				} else {
 					// No connection, wait briefly before checking again
@@ -989,10 +985,7 @@ func (ais *AudioInputServer) startReaderGoroutine() {
 	logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
 	if !SubmitAudioReaderTask(readerTask) {
 		// If the pool is full or shutting down, fall back to direct goroutine creation
-		// Only log if warn level enabled - avoid sampling logic in critical path
+		logger.Warn().Msg("Audio reader pool full or shutting down, falling back to direct goroutine creation")
-		if logger.GetLevel() <= zerolog.WarnLevel {
-			logger.Warn().Msg("Audio reader pool full or shutting down, falling back to direct goroutine creation")
-		}
 
 		go readerTask()
 	}
@@ -1004,17 +997,12 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 
 	// Create a processor task that will run in the goroutine pool
 	processorTask := func() {
-		// Only lock OS thread and set priority for high-load scenarios
+		runtime.LockOSThread()
-		// This reduces interference with input processing threads
+		defer runtime.UnlockOSThread()
-		config := GetConfig()
-		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
-		if useThreadOptimizations {
+		// Set high priority for audio processing - skip logging in hotpath
-			runtime.LockOSThread()
+		_ = SetAudioThreadPriority()
-			defer runtime.UnlockOSThread()
+		defer func() { _ = ResetThreadPriority() }()
-
-			// Priority scheduler not implemented - using default thread priority
-		}
 
 		// Create logger for this goroutine
 		logger := logging.GetDefaultLogger().With().Str("component", AudioInputServerComponent).Logger()
@@ -1022,8 +1010,8 @@ func (ais *AudioInputServer) startProcessorGoroutine() {
 		// Enhanced error tracking for processing
 		var processingErrors int
 		var lastProcessingError time.Time
-		maxProcessingErrors := config.MaxConsecutiveErrors
+		maxProcessingErrors := GetConfig().MaxConsecutiveErrors
-		errorResetWindow := config.RestartWindow
+		errorResetWindow := GetConfig().RestartWindow
 
 		defer ais.wg.Done()
 		for {
@@ -1126,17 +1114,19 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
 
 	// Create a monitor task that will run in the goroutine pool
 	monitorTask := func() {
-		// Monitor goroutine doesn't need thread locking for most scenarios
+		runtime.LockOSThread()
-		// Only use thread optimizations for high-throughput scenarios
+		defer runtime.UnlockOSThread()
-		config := GetConfig()
-		useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
 
-		if useThreadOptimizations {
+		// Set I/O priority for monitoring
-			runtime.LockOSThread()
+		logger := logging.GetDefaultLogger().With().Str("component", AudioInputClientComponent).Logger()
-			defer runtime.UnlockOSThread()
+		if err := SetAudioIOThreadPriority(); err != nil {
-
+			logger.Warn().Err(err).Msg("Failed to set audio I/O priority")
-			// Priority scheduler not implemented - using default thread priority
 		}
+		defer func() {
+			if err := ResetThreadPriority(); err != nil {
+				logger.Warn().Err(err).Msg("Failed to reset thread priority")
+			}
+		}()
 
 		defer ais.wg.Done()
 		ticker := time.NewTicker(GetConfig().DefaultTickerInterval)

internal/audio/input_ipc_manager_test.go

@@ -0,0 +1,277 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestAudioInputIPCManager tests the AudioInputIPCManager component
+func TestAudioInputIPCManager(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"Start", testAudioInputIPCManagerStart},
+		{"Stop", testAudioInputIPCManagerStop},
+		{"StartStop", testAudioInputIPCManagerStartStop},
+		{"IsRunning", testAudioInputIPCManagerIsRunning},
+		{"IsReady", testAudioInputIPCManagerIsReady},
+		{"GetMetrics", testAudioInputIPCManagerGetMetrics},
+		{"ConcurrentOperations", testAudioInputIPCManagerConcurrent},
+		{"MultipleStarts", testAudioInputIPCManagerMultipleStarts},
+		{"MultipleStops", testAudioInputIPCManagerMultipleStops},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+func testAudioInputIPCManagerStart(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Test initial state
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+
+	// Test start
+	err := manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioInputIPCManagerStop(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Start first
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Test stop
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+}
+
+func testAudioInputIPCManagerStartStop(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Test multiple start/stop cycles
+	for i := 0; i < 3; i++ {
+		// Start
+		err := manager.Start()
+		assert.NoError(t, err)
+		assert.True(t, manager.IsRunning())
+
+		// Stop
+		manager.Stop()
+		assert.False(t, manager.IsRunning())
+	}
+}
+
+func testAudioInputIPCManagerIsRunning(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Initially not running
+	assert.False(t, manager.IsRunning())
+
+	// Start and check
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Stop and check
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+func testAudioInputIPCManagerIsReady(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Initially not ready
+	assert.False(t, manager.IsReady())
+
+	// Start and check ready state
+	err := manager.Start()
+	require.NoError(t, err)
+
+	// Give some time for initialization
+	time.Sleep(100 * time.Millisecond)
+
+	// Stop
+	manager.Stop()
+	assert.False(t, manager.IsReady())
+}
+
+func testAudioInputIPCManagerGetMetrics(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Test metrics when not running
+	metrics := manager.GetMetrics()
+	assert.NotNil(t, metrics)
+
+	// Start and test metrics
+	err := manager.Start()
+	require.NoError(t, err)
+
+	metrics = manager.GetMetrics()
+	assert.NotNil(t, metrics)
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioInputIPCManagerConcurrent(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	var wg sync.WaitGroup
+	const numGoroutines = 10
+
+	// Test concurrent starts
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			manager.Start()
+		}()
+	}
+	wg.Wait()
+
+	// Should be running
+	assert.True(t, manager.IsRunning())
+
+	// Test concurrent stops
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			manager.Stop()
+		}()
+	}
+	wg.Wait()
+
+	// Should be stopped
+	assert.False(t, manager.IsRunning())
+}
+
+func testAudioInputIPCManagerMultipleStarts(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// First start should succeed
+	err := manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Subsequent starts should be no-op
+	err = manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	err = manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioInputIPCManagerMultipleStops(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Start first
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// First stop should work
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+
+	// Subsequent stops should be no-op
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+// TestAudioInputIPCMetrics tests the AudioInputMetrics functionality
+func TestAudioInputIPCMetrics(t *testing.T) {
+	metrics := &AudioInputMetrics{}
+
+	// Test initial state
+	assert.Equal(t, int64(0), metrics.FramesSent)
+	assert.Equal(t, int64(0), metrics.FramesDropped)
+	assert.Equal(t, int64(0), metrics.BytesProcessed)
+	assert.Equal(t, int64(0), metrics.ConnectionDrops)
+	assert.Equal(t, time.Duration(0), metrics.AverageLatency)
+	assert.True(t, metrics.LastFrameTime.IsZero())
+
+	// Test field assignment
+	metrics.FramesSent = 50
+	metrics.FramesDropped = 2
+	metrics.BytesProcessed = 512
+	metrics.ConnectionDrops = 1
+	metrics.AverageLatency = 5 * time.Millisecond
+	metrics.LastFrameTime = time.Now()
+
+	// Verify assignments
+	assert.Equal(t, int64(50), metrics.FramesSent)
+	assert.Equal(t, int64(2), metrics.FramesDropped)
+	assert.Equal(t, int64(512), metrics.BytesProcessed)
+	assert.Equal(t, int64(1), metrics.ConnectionDrops)
+	assert.Equal(t, 5*time.Millisecond, metrics.AverageLatency)
+	assert.False(t, metrics.LastFrameTime.IsZero())
+}
+
+// BenchmarkAudioInputIPCManager benchmarks the AudioInputIPCManager operations
+func BenchmarkAudioInputIPCManager(b *testing.B) {
+	b.Run("Start", func(b *testing.B) {
+		for i := 0; i < b.N; i++ {
+			manager := NewAudioInputIPCManager()
+			manager.Start()
+			manager.Stop()
+		}
+	})
+
+	b.Run("IsRunning", func(b *testing.B) {
+		manager := NewAudioInputIPCManager()
+		manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			manager.IsRunning()
+		}
+	})
+
+	b.Run("GetMetrics", func(b *testing.B) {
+		manager := NewAudioInputIPCManager()
+		manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			manager.GetMetrics()
+		}
+	})
+}

internal/audio/input_supervisor.go

@@ -21,10 +21,6 @@ type AudioInputSupervisor struct {
 
 	// Environment variables for OPUS configuration
 	opusEnv []string
-
-	// Pre-warming state
-	prewarmed   bool
-	prewarmTime time.Time
 }
 
 // NewAudioInputSupervisor creates a new audio input supervisor
@@ -52,73 +48,6 @@ func (ais *AudioInputSupervisor) SetOpusConfig(bitrate, complexity, vbr, signalT
 	}
 }
 
-// PrewarmSubprocess starts a subprocess in advance to reduce activation latency
-func (ais *AudioInputSupervisor) PrewarmSubprocess() error {
-	ais.mutex.Lock()
-	defer ais.mutex.Unlock()
-
-	// Don't prewarm if already running or prewarmed
-	if ais.IsRunning() || ais.prewarmed {
-		return nil
-	}
-
-	// Check for existing audio input server process first
-	if existingPID, err := ais.findExistingAudioInputProcess(); err == nil {
-		ais.logger.Info().Int("existing_pid", existingPID).Msg("Found existing audio input server process for prewarming")
-		ais.prewarmed = true
-		ais.prewarmTime = time.Now()
-		return nil
-	}
-
-	// Create context for subprocess management
-	ais.createContext()
-
-	// Get current executable path
-	execPath, err := os.Executable()
-	if err != nil {
-		return fmt.Errorf("failed to get executable path: %w", err)
-	}
-
-	// Build command arguments (only subprocess flag)
-	args := []string{"--audio-input-server"}
-
-	// Create command for audio input server subprocess
-	cmd := exec.CommandContext(ais.ctx, execPath, args...)
-
-	// Set environment variables for IPC and OPUS configuration
-	env := append(os.Environ(), "JETKVM_AUDIO_INPUT_IPC=true") // Enable IPC mode
-	env = append(env, ais.opusEnv...)                          // Add OPUS configuration
-	cmd.Env = env
-
-	// Set process group to allow clean termination
-	cmd.SysProcAttr = &syscall.SysProcAttr{
-		Setpgid: true,
-	}
-
-	ais.cmd = cmd
-
-	// Start the subprocess
-	err = cmd.Start()
-	if err != nil {
-		ais.cancelContext()
-		return fmt.Errorf("failed to prewarm audio input server process: %w", err)
-	}
-
-	ais.logger.Info().Int("pid", cmd.Process.Pid).Strs("args", args).Strs("opus_env", ais.opusEnv).Msg("Audio input server subprocess prewarmed")
-
-	// Add process to monitoring
-	ais.processMonitor.AddProcess(cmd.Process.Pid, "audio-input-server")
-
-	// Monitor the subprocess in a goroutine
-	go ais.monitorSubprocess()
-
-	// Mark as prewarmed
-	ais.prewarmed = true
-	ais.prewarmTime = time.Now()
-
-	return nil
-}
-
 // Start starts the audio input server subprocess
 func (ais *AudioInputSupervisor) Start() error {
 	ais.mutex.Lock()
@@ -131,16 +60,6 @@ func (ais *AudioInputSupervisor) Start() error {
 		return fmt.Errorf("audio input supervisor already running")
 	}
 
-	// Use prewarmed subprocess if available
-	if ais.prewarmed && ais.cmd != nil && ais.cmd.Process != nil {
-		ais.logger.Info().Int("pid", ais.cmd.Process.Pid).Dur("prewarm_age", time.Since(ais.prewarmTime)).Msg("Using prewarmed audio input server subprocess")
-		ais.setRunning(true)
-		ais.prewarmed = false // Reset prewarmed state
-		// Connect client to the server
-		go ais.connectClient()
-		return nil
-	}
-
 	// Check for existing audio input server process
 	if existingPID, err := ais.findExistingAudioInputProcess(); err == nil {
 		ais.logger.Info().Int("existing_pid", existingPID).Msg("Found existing audio input server process, connecting to it")
@@ -201,31 +120,11 @@ func (ais *AudioInputSupervisor) Start() error {
 	return nil
 }
 
-// IsPrewarmed returns whether a subprocess is prewarmed and ready
-func (ais *AudioInputSupervisor) IsPrewarmed() bool {
-	ais.mutex.RLock()
-	defer ais.mutex.RUnlock()
-	return ais.prewarmed
-}
-
-// GetPrewarmAge returns how long ago the subprocess was prewarmed
-func (ais *AudioInputSupervisor) GetPrewarmAge() time.Duration {
-	ais.mutex.RLock()
-	defer ais.mutex.RUnlock()
-	if !ais.prewarmed {
-		return 0
-	}
-	return time.Since(ais.prewarmTime)
-}
-
 // Stop stops the audio input server subprocess
 func (ais *AudioInputSupervisor) Stop() {
 	ais.mutex.Lock()
 	defer ais.mutex.Unlock()
 
-	// Reset prewarmed state
-	ais.prewarmed = false
-
 	if !ais.IsRunning() {
 		return
 	}
@@ -319,6 +218,13 @@ func (ais *AudioInputSupervisor) GetClient() *AudioInputClient {
 	return ais.client
 }
 
+// GetProcessMetrics returns current process metrics with audio-input-server name
+func (ais *AudioInputSupervisor) GetProcessMetrics() *ProcessMetrics {
+	metrics := ais.BaseSupervisor.GetProcessMetrics()
+	metrics.ProcessName = "audio-input-server"
+	return metrics
+}
+
 // monitorSubprocess monitors the subprocess and handles unexpected exits
 func (ais *AudioInputSupervisor) monitorSubprocess() {
 	if ais.cmd == nil || ais.cmd.Process == nil {

internal/audio/input_test.go

@@ -0,0 +1,244 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+func TestNewAudioInputManager(t *testing.T) {
+	manager := NewAudioInputManager()
+	assert.NotNil(t, manager)
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+}
+
+func TestAudioInputManagerStart(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test successful start
+	err := manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Test starting already running manager
+	err = manager.Start()
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "already running")
+
+	// Cleanup
+	manager.Stop()
+}
+
+func TestAudioInputManagerStop(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test stopping non-running manager
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+
+	// Start and then stop
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+func TestAudioInputManagerIsRunning(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test initial state
+	assert.False(t, manager.IsRunning())
+
+	// Test after start
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Test after stop
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+func TestAudioInputManagerIsReady(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test initial state
+	assert.False(t, manager.IsReady())
+
+	// Start manager
+	err := manager.Start()
+	require.NoError(t, err)
+
+	// Give some time for initialization
+	time.Sleep(100 * time.Millisecond)
+
+	// Test ready state (may vary based on implementation)
+	// Just ensure the method doesn't panic
+	_ = manager.IsReady()
+
+	// Cleanup
+	manager.Stop()
+}
+
+func TestAudioInputManagerGetMetrics(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test metrics when not running
+	metrics := manager.GetMetrics()
+	assert.NotNil(t, metrics)
+	assert.Equal(t, int64(0), metrics.FramesSent)
+	assert.Equal(t, int64(0), metrics.FramesDropped)
+	assert.Equal(t, int64(0), metrics.BytesProcessed)
+	assert.Equal(t, int64(0), metrics.ConnectionDrops)
+
+	// Start and test metrics
+	err := manager.Start()
+	require.NoError(t, err)
+
+	metrics = manager.GetMetrics()
+	assert.NotNil(t, metrics)
+	assert.GreaterOrEqual(t, metrics.FramesSent, int64(0))
+	assert.GreaterOrEqual(t, metrics.FramesDropped, int64(0))
+	assert.GreaterOrEqual(t, metrics.BytesProcessed, int64(0))
+	assert.GreaterOrEqual(t, metrics.ConnectionDrops, int64(0))
+
+	// Cleanup
+	manager.Stop()
+}
+
+func TestAudioInputManagerConcurrentOperations(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	var wg sync.WaitGroup
+
+	// Test concurrent start/stop operations
+	for i := 0; i < 10; i++ {
+		wg.Add(2)
+		go func() {
+			defer wg.Done()
+			_ = manager.Start()
+		}()
+		go func() {
+			defer wg.Done()
+			manager.Stop()
+		}()
+	}
+
+	// Test concurrent metric access
+	for i := 0; i < 5; i++ {
+		wg.Add(1)
+		go func() {
+			defer wg.Done()
+			_ = manager.GetMetrics()
+		}()
+	}
+
+	// Test concurrent status checks
+	for i := 0; i < 5; i++ {
+		wg.Add(2)
+		go func() {
+			defer wg.Done()
+			_ = manager.IsRunning()
+		}()
+		go func() {
+			defer wg.Done()
+			_ = manager.IsReady()
+		}()
+	}
+
+	wg.Wait()
+
+	// Cleanup
+	manager.Stop()
+}
+
+func TestAudioInputManagerMultipleStartStop(t *testing.T) {
+	manager := NewAudioInputManager()
+	require.NotNil(t, manager)
+
+	// Test multiple start/stop cycles
+	for i := 0; i < 5; i++ {
+		err := manager.Start()
+		assert.NoError(t, err)
+		assert.True(t, manager.IsRunning())
+
+		manager.Stop()
+		assert.False(t, manager.IsRunning())
+	}
+}
+
+func TestAudioInputMetrics(t *testing.T) {
+	metrics := &AudioInputMetrics{
+		BaseAudioMetrics: BaseAudioMetrics{
+			FramesProcessed: 100,
+			FramesDropped:   5,
+			BytesProcessed:  1024,
+			ConnectionDrops: 2,
+			AverageLatency:  time.Millisecond * 10,
+			LastFrameTime:   time.Now(),
+		},
+		FramesSent: 100,
+	}
+
+	assert.Equal(t, int64(100), metrics.FramesSent)
+	assert.Equal(t, int64(5), metrics.FramesDropped)
+	assert.Equal(t, int64(1024), metrics.BytesProcessed)
+	assert.Equal(t, int64(2), metrics.ConnectionDrops)
+	assert.Equal(t, time.Millisecond*10, metrics.AverageLatency)
+	assert.False(t, metrics.LastFrameTime.IsZero())
+}
+
+// Benchmark tests
+func BenchmarkAudioInputManager(b *testing.B) {
+	manager := NewAudioInputManager()
+
+	b.Run("Start", func(b *testing.B) {
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = manager.Start()
+			manager.Stop()
+		}
+	})
+
+	b.Run("GetMetrics", func(b *testing.B) {
+		_ = manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = manager.GetMetrics()
+		}
+	})
+
+	b.Run("IsRunning", func(b *testing.B) {
+		_ = manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = manager.IsRunning()
+		}
+	})
+
+	b.Run("IsReady", func(b *testing.B) {
+		_ = manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = manager.IsReady()
+		}
+	})
+}

internal/audio/integration_test.go

@@ -0,0 +1,320 @@
+//go:build integration
+// +build integration
+
+package audio
+
+import (
+	"context"
+	"fmt"
+	"net"
+	"os"
+	"path/filepath"
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestIPCCommunication tests the IPC communication between audio components
+func TestIPCCommunication(t *testing.T) {
+	tests := []struct {
+		name        string
+		testFunc    func(t *testing.T)
+		description string
+	}{
+		{
+			name:        "AudioOutputIPC",
+			testFunc:    testAudioOutputIPC,
+			description: "Test audio output IPC server and client communication",
+		},
+		{
+			name:        "AudioInputIPC",
+			testFunc:    testAudioInputIPC,
+			description: "Test audio input IPC server and client communication",
+		},
+		{
+			name:        "IPCReconnection",
+			testFunc:    testIPCReconnection,
+			description: "Test IPC reconnection after connection loss",
+		},
+		{
+			name:        "IPCConcurrency",
+			testFunc:    testIPCConcurrency,
+			description: "Test concurrent IPC operations",
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			t.Logf("Running test: %s - %s", tt.name, tt.description)
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testAudioOutputIPC tests the audio output IPC communication
+func testAudioOutputIPC(t *testing.T) {
+	tempDir := t.TempDir()
+	socketPath := filepath.Join(tempDir, "test_audio_output.sock")
+
+	// Create a test IPC server
+	server := &AudioIPCServer{
+		socketPath: socketPath,
+		logger:     getTestLogger(),
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 30*time.Second)
+	defer cancel()
+
+	// Start server in goroutine
+	var serverErr error
+	var wg sync.WaitGroup
+	wg.Add(1)
+	go func() {
+		defer wg.Done()
+		serverErr = server.Start(ctx)
+	}()
+
+	// Wait for server to start
+	time.Sleep(100 * time.Millisecond)
+
+	// Test client connection
+	conn, err := net.Dial("unix", socketPath)
+	require.NoError(t, err, "Failed to connect to IPC server")
+	defer conn.Close()
+
+	// Test sending a frame message
+	testFrame := []byte("test audio frame data")
+	msg := &OutputMessage{
+		Type:      OutputMessageTypeOpusFrame,
+		Timestamp: time.Now().UnixNano(),
+		Data:      testFrame,
+	}
+
+	err = writeOutputMessage(conn, msg)
+	require.NoError(t, err, "Failed to write message to IPC")
+
+	// Test heartbeat
+	heartbeatMsg := &OutputMessage{
+		Type:      OutputMessageTypeHeartbeat,
+		Timestamp: time.Now().UnixNano(),
+	}
+
+	err = writeOutputMessage(conn, heartbeatMsg)
+	require.NoError(t, err, "Failed to send heartbeat")
+
+	// Clean shutdown
+	cancel()
+	wg.Wait()
+
+	if serverErr != nil && serverErr != context.Canceled {
+		t.Errorf("Server error: %v", serverErr)
+	}
+}
+
+// testAudioInputIPC tests the audio input IPC communication
+func testAudioInputIPC(t *testing.T) {
+	tempDir := t.TempDir()
+	socketPath := filepath.Join(tempDir, "test_audio_input.sock")
+
+	// Create a test input IPC server
+	server := &AudioInputIPCServer{
+		socketPath: socketPath,
+		logger:     getTestLogger(),
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 30*time.Second)
+	defer cancel()
+
+	// Start server
+	var serverErr error
+	var wg sync.WaitGroup
+	wg.Add(1)
+	go func() {
+		defer wg.Done()
+		serverErr = server.Start(ctx)
+	}()
+
+	// Wait for server to start
+	time.Sleep(100 * time.Millisecond)
+
+	// Test client connection
+	conn, err := net.Dial("unix", socketPath)
+	require.NoError(t, err, "Failed to connect to input IPC server")
+	defer conn.Close()
+
+	// Test sending input frame
+	testInputFrame := []byte("test microphone data")
+	inputMsg := &InputMessage{
+		Type:      InputMessageTypeOpusFrame,
+		Timestamp: time.Now().UnixNano(),
+		Data:      testInputFrame,
+	}
+
+	err = writeInputMessage(conn, inputMsg)
+	require.NoError(t, err, "Failed to write input message")
+
+	// Test configuration message
+	configMsg := &InputMessage{
+		Type:      InputMessageTypeConfig,
+		Timestamp: time.Now().UnixNano(),
+		Data:      []byte("quality=medium"),
+	}
+
+	err = writeInputMessage(conn, configMsg)
+	require.NoError(t, err, "Failed to send config message")
+
+	// Clean shutdown
+	cancel()
+	wg.Wait()
+
+	if serverErr != nil && serverErr != context.Canceled {
+		t.Errorf("Input server error: %v", serverErr)
+	}
+}
+
+// testIPCReconnection tests IPC reconnection scenarios
+func testIPCReconnection(t *testing.T) {
+	tempDir := t.TempDir()
+	socketPath := filepath.Join(tempDir, "test_reconnect.sock")
+
+	// Create server
+	server := &AudioIPCServer{
+		socketPath: socketPath,
+		logger:     getTestLogger(),
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 45*time.Second)
+	defer cancel()
+
+	// Start server
+	var wg sync.WaitGroup
+	wg.Add(1)
+	go func() {
+		defer wg.Done()
+		server.Start(ctx)
+	}()
+
+	time.Sleep(100 * time.Millisecond)
+
+	// First connection
+	conn1, err := net.Dial("unix", socketPath)
+	require.NoError(t, err, "Failed initial connection")
+
+	// Send a message
+	msg := &OutputMessage{
+		Type:      OutputMessageTypeOpusFrame,
+		Timestamp: time.Now().UnixNano(),
+		Data:      []byte("test data 1"),
+	}
+	err = writeOutputMessage(conn1, msg)
+	require.NoError(t, err, "Failed to send first message")
+
+	// Close connection to simulate disconnect
+	conn1.Close()
+	time.Sleep(200 * time.Millisecond)
+
+	// Reconnect
+	conn2, err := net.Dial("unix", socketPath)
+	require.NoError(t, err, "Failed to reconnect")
+	defer conn2.Close()
+
+	// Send another message after reconnection
+	msg2 := &OutputMessage{
+		Type:      OutputMessageTypeOpusFrame,
+		Timestamp: time.Now().UnixNano(),
+		Data:      []byte("test data 2"),
+	}
+	err = writeOutputMessage(conn2, msg2)
+	require.NoError(t, err, "Failed to send message after reconnection")
+
+	cancel()
+	wg.Wait()
+}
+
+// testIPCConcurrency tests concurrent IPC operations
+func testIPCConcurrency(t *testing.T) {
+	tempDir := t.TempDir()
+	socketPath := filepath.Join(tempDir, "test_concurrent.sock")
+
+	server := &AudioIPCServer{
+		socketPath: socketPath,
+		logger:     getTestLogger(),
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 60*time.Second)
+	defer cancel()
+
+	// Start server
+	var wg sync.WaitGroup
+	wg.Add(1)
+	go func() {
+		defer wg.Done()
+		server.Start(ctx)
+	}()
+
+	time.Sleep(100 * time.Millisecond)
+
+	// Create multiple concurrent connections
+	numClients := 5
+	messagesPerClient := 10
+
+	var clientWg sync.WaitGroup
+	for i := 0; i < numClients; i++ {
+		clientWg.Add(1)
+		go func(clientID int) {
+			defer clientWg.Done()
+
+			conn, err := net.Dial("unix", socketPath)
+			if err != nil {
+				t.Errorf("Client %d failed to connect: %v", clientID, err)
+				return
+			}
+			defer conn.Close()
+
+			// Send multiple messages
+			for j := 0; j < messagesPerClient; j++ {
+				msg := &OutputMessage{
+					Type:      OutputMessageTypeOpusFrame,
+					Timestamp: time.Now().UnixNano(),
+					Data:      []byte(fmt.Sprintf("client_%d_msg_%d", clientID, j)),
+				}
+
+				if err := writeOutputMessage(conn, msg); err != nil {
+					t.Errorf("Client %d failed to send message %d: %v", clientID, j, err)
+					return
+				}
+
+				// Small delay between messages
+				time.Sleep(10 * time.Millisecond)
+			}
+		}(i)
+	}
+
+	clientWg.Wait()
+	cancel()
+	wg.Wait()
+}
+
+// Helper function to get a test logger
+func getTestLogger() zerolog.Logger {
+	return zerolog.New(os.Stdout).With().Timestamp().Logger()
+}
+
+// Helper functions for message writing (simplified versions)
+func writeOutputMessage(conn net.Conn, msg *OutputMessage) error {
+	// This is a simplified version for testing
+	// In real implementation, this would use the actual protocol
+	data := fmt.Sprintf("%d:%d:%s", msg.Type, msg.Timestamp, string(msg.Data))
+	_, err := conn.Write([]byte(data))
+	return err
+}
+
+func writeInputMessage(conn net.Conn, msg *InputMessage) error {
+	// This is a simplified version for testing
+	data := fmt.Sprintf("%d:%d:%s", msg.Type, msg.Timestamp, string(msg.Data))
+	_, err := conn.Write([]byte(data))
+	return err
+}

internal/audio/latency_profiler.go

@@ -0,0 +1,535 @@
+package audio
+
+import (
+	"context"
+	"fmt"
+	"runtime"
+	"sync"
+	"sync/atomic"
+	"time"
+	"unsafe"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// LatencyProfiler provides comprehensive end-to-end audio latency profiling
+// with nanosecond precision across the entire WebRTC->IPC->CGO->ALSA pipeline
+type LatencyProfiler struct {
+	// Atomic counters for thread-safe access (MUST be first for ARM32 alignment)
+	totalMeasurements       int64 // Total number of measurements taken
+	webrtcLatencySum        int64 // Sum of WebRTC processing latencies (nanoseconds)
+	ipcLatencySum           int64 // Sum of IPC communication latencies (nanoseconds)
+	cgoLatencySum           int64 // Sum of CGO call latencies (nanoseconds)
+	alsaLatencySum          int64 // Sum of ALSA device latencies (nanoseconds)
+	endToEndLatencySum      int64 // Sum of complete end-to-end latencies (nanoseconds)
+	validationLatencySum    int64 // Sum of validation overhead (nanoseconds)
+	serializationLatencySum int64 // Sum of serialization overhead (nanoseconds)
+
+	// Peak latency tracking
+	maxWebrtcLatency   int64 // Maximum WebRTC latency observed (nanoseconds)
+	maxIpcLatency      int64 // Maximum IPC latency observed (nanoseconds)
+	maxCgoLatency      int64 // Maximum CGO latency observed (nanoseconds)
+	maxAlsaLatency     int64 // Maximum ALSA latency observed (nanoseconds)
+	maxEndToEndLatency int64 // Maximum end-to-end latency observed (nanoseconds)
+
+	// Configuration and control
+	config  LatencyProfilerConfig
+	logger  zerolog.Logger
+	ctx     context.Context
+	cancel  context.CancelFunc
+	running int32 // Atomic flag for profiler state
+	enabled int32 // Atomic flag for measurement collection
+
+	// Detailed measurement storage
+	measurements     []DetailedLatencyMeasurement
+	measurementMutex sync.RWMutex
+	measurementIndex int
+
+	// High-resolution timing
+	timeSource func() int64 // Nanosecond precision time source
+}
+
+// LatencyProfilerConfig defines profiler configuration
+type LatencyProfilerConfig struct {
+	MaxMeasurements     int           // Maximum measurements to store in memory
+	SamplingRate        float64       // Sampling rate (0.0-1.0, 1.0 = profile every frame)
+	ReportingInterval   time.Duration // How often to log profiling reports
+	ThresholdWarning    time.Duration // Latency threshold for warnings
+	ThresholdCritical   time.Duration // Latency threshold for critical alerts
+	EnableDetailedTrace bool          // Enable detailed per-component tracing
+	EnableHistogram     bool          // Enable latency histogram collection
+}
+
+// DetailedLatencyMeasurement captures comprehensive latency breakdown
+type DetailedLatencyMeasurement struct {
+	Timestamp            time.Time     // When the measurement was taken
+	FrameID              uint64        // Unique frame identifier for tracing
+	WebRTCLatency        time.Duration // WebRTC processing time
+	IPCLatency           time.Duration // IPC communication time
+	CGOLatency           time.Duration // CGO call overhead
+	ALSALatency          time.Duration // ALSA device processing time
+	ValidationLatency    time.Duration // Frame validation overhead
+	SerializationLatency time.Duration // Data serialization overhead
+	EndToEndLatency      time.Duration // Complete pipeline latency
+	Source               string        // Source component (input/output)
+	FrameSize            int           // Size of the audio frame in bytes
+	CPUUsage             float64       // CPU usage at time of measurement
+	MemoryUsage          uint64        // Memory usage at time of measurement
+}
+
+// LatencyProfileReport contains aggregated profiling results
+type LatencyProfileReport struct {
+	TotalMeasurements int64         // Total measurements taken
+	TimeRange         time.Duration // Time span of measurements
+
+	// Average latencies
+	AvgWebRTCLatency        time.Duration
+	AvgIPCLatency           time.Duration
+	AvgCGOLatency           time.Duration
+	AvgALSALatency          time.Duration
+	AvgEndToEndLatency      time.Duration
+	AvgValidationLatency    time.Duration
+	AvgSerializationLatency time.Duration
+
+	// Peak latencies
+	MaxWebRTCLatency   time.Duration
+	MaxIPCLatency      time.Duration
+	MaxCGOLatency      time.Duration
+	MaxALSALatency     time.Duration
+	MaxEndToEndLatency time.Duration
+
+	// Performance analysis
+	BottleneckComponent string         // Component with highest average latency
+	LatencyDistribution map[string]int // Histogram of latency ranges
+	Throughput          float64        // Frames per second processed
+}
+
+// FrameLatencyTracker tracks latency for a single audio frame through the pipeline
+type FrameLatencyTracker struct {
+	frameID                uint64
+	startTime              int64 // Nanosecond timestamp
+	webrtcStartTime        int64
+	ipcStartTime           int64
+	cgoStartTime           int64
+	alsaStartTime          int64
+	validationStartTime    int64
+	serializationStartTime int64
+	frameSize              int
+	source                 string
+}
+
+// Global profiler instance
+var (
+	globalLatencyProfiler unsafe.Pointer // *LatencyProfiler
+	profilerInitialized   int32
+)
+
+// DefaultLatencyProfilerConfig returns default profiler configuration
+func DefaultLatencyProfilerConfig() LatencyProfilerConfig {
+	return LatencyProfilerConfig{
+		MaxMeasurements:     10000,
+		SamplingRate:        0.1, // Profile 10% of frames to minimize overhead
+		ReportingInterval:   30 * time.Second,
+		ThresholdWarning:    50 * time.Millisecond,
+		ThresholdCritical:   100 * time.Millisecond,
+		EnableDetailedTrace: false, // Disabled by default for performance
+		EnableHistogram:     true,
+	}
+}
+
+// NewLatencyProfiler creates a new latency profiler
+func NewLatencyProfiler(config LatencyProfilerConfig) *LatencyProfiler {
+	ctx, cancel := context.WithCancel(context.Background())
+	logger := logging.GetDefaultLogger().With().Str("component", "latency-profiler").Logger()
+
+	// Validate configuration
+	if config.MaxMeasurements <= 0 {
+		config.MaxMeasurements = 10000
+	}
+	if config.SamplingRate < 0.0 || config.SamplingRate > 1.0 {
+		config.SamplingRate = 0.1
+	}
+	if config.ReportingInterval <= 0 {
+		config.ReportingInterval = 30 * time.Second
+	}
+
+	profiler := &LatencyProfiler{
+		config:       config,
+		logger:       logger,
+		ctx:          ctx,
+		cancel:       cancel,
+		measurements: make([]DetailedLatencyMeasurement, config.MaxMeasurements),
+		timeSource:   func() int64 { return time.Now().UnixNano() },
+	}
+
+	// Initialize peak latencies to zero
+	atomic.StoreInt64(&profiler.maxWebrtcLatency, 0)
+	atomic.StoreInt64(&profiler.maxIpcLatency, 0)
+	atomic.StoreInt64(&profiler.maxCgoLatency, 0)
+	atomic.StoreInt64(&profiler.maxAlsaLatency, 0)
+	atomic.StoreInt64(&profiler.maxEndToEndLatency, 0)
+
+	return profiler
+}
+
+// Start begins latency profiling
+func (lp *LatencyProfiler) Start() error {
+	if !atomic.CompareAndSwapInt32(&lp.running, 0, 1) {
+		return fmt.Errorf("latency profiler already running")
+	}
+
+	// Enable measurement collection
+	atomic.StoreInt32(&lp.enabled, 1)
+
+	// Start reporting goroutine
+	go lp.reportingLoop()
+
+	lp.logger.Info().Float64("sampling_rate", lp.config.SamplingRate).Msg("latency profiler started")
+	return nil
+}
+
+// Stop stops latency profiling
+func (lp *LatencyProfiler) Stop() {
+	if !atomic.CompareAndSwapInt32(&lp.running, 1, 0) {
+		return
+	}
+
+	// Disable measurement collection
+	atomic.StoreInt32(&lp.enabled, 0)
+
+	// Cancel context to stop reporting
+	lp.cancel()
+
+	lp.logger.Info().Msg("latency profiler stopped")
+}
+
+// IsEnabled returns whether profiling is currently enabled
+func (lp *LatencyProfiler) IsEnabled() bool {
+	return atomic.LoadInt32(&lp.enabled) == 1
+}
+
+// StartFrameTracking begins tracking latency for a new audio frame
+func (lp *LatencyProfiler) StartFrameTracking(frameID uint64, frameSize int, source string) *FrameLatencyTracker {
+	if !lp.IsEnabled() {
+		return nil
+	}
+
+	// Apply sampling rate to reduce profiling overhead
+	if lp.config.SamplingRate < 1.0 {
+		// Simple sampling based on frame ID
+		if float64(frameID%100)/100.0 > lp.config.SamplingRate {
+			return nil
+		}
+	}
+
+	now := lp.timeSource()
+	return &FrameLatencyTracker{
+		frameID:   frameID,
+		startTime: now,
+		frameSize: frameSize,
+		source:    source,
+	}
+}
+
+// TrackWebRTCStart marks the start of WebRTC processing
+func (tracker *FrameLatencyTracker) TrackWebRTCStart() {
+	if tracker != nil {
+		tracker.webrtcStartTime = time.Now().UnixNano()
+	}
+}
+
+// TrackIPCStart marks the start of IPC communication
+func (tracker *FrameLatencyTracker) TrackIPCStart() {
+	if tracker != nil {
+		tracker.ipcStartTime = time.Now().UnixNano()
+	}
+}
+
+// TrackCGOStart marks the start of CGO processing
+func (tracker *FrameLatencyTracker) TrackCGOStart() {
+	if tracker != nil {
+		tracker.cgoStartTime = time.Now().UnixNano()
+	}
+}
+
+// TrackALSAStart marks the start of ALSA device processing
+func (tracker *FrameLatencyTracker) TrackALSAStart() {
+	if tracker != nil {
+		tracker.alsaStartTime = time.Now().UnixNano()
+	}
+}
+
+// TrackValidationStart marks the start of frame validation
+func (tracker *FrameLatencyTracker) TrackValidationStart() {
+	if tracker != nil {
+		tracker.validationStartTime = time.Now().UnixNano()
+	}
+}
+
+// TrackSerializationStart marks the start of data serialization
+func (tracker *FrameLatencyTracker) TrackSerializationStart() {
+	if tracker != nil {
+		tracker.serializationStartTime = time.Now().UnixNano()
+	}
+}
+
+// FinishTracking completes frame tracking and records the measurement
+func (lp *LatencyProfiler) FinishTracking(tracker *FrameLatencyTracker) {
+	if tracker == nil || !lp.IsEnabled() {
+		return
+	}
+
+	endTime := lp.timeSource()
+
+	// Calculate component latencies
+	var webrtcLatency, ipcLatency, cgoLatency, alsaLatency, validationLatency, serializationLatency time.Duration
+
+	if tracker.webrtcStartTime > 0 {
+		webrtcLatency = time.Duration(tracker.ipcStartTime - tracker.webrtcStartTime)
+	}
+	if tracker.ipcStartTime > 0 {
+		ipcLatency = time.Duration(tracker.cgoStartTime - tracker.ipcStartTime)
+	}
+	if tracker.cgoStartTime > 0 {
+		cgoLatency = time.Duration(tracker.alsaStartTime - tracker.cgoStartTime)
+	}
+	if tracker.alsaStartTime > 0 {
+		alsaLatency = time.Duration(endTime - tracker.alsaStartTime)
+	}
+	if tracker.validationStartTime > 0 {
+		validationLatency = time.Duration(tracker.ipcStartTime - tracker.validationStartTime)
+	}
+	if tracker.serializationStartTime > 0 {
+		serializationLatency = time.Duration(tracker.cgoStartTime - tracker.serializationStartTime)
+	}
+
+	endToEndLatency := time.Duration(endTime - tracker.startTime)
+
+	// Update atomic counters
+	atomic.AddInt64(&lp.totalMeasurements, 1)
+	atomic.AddInt64(&lp.webrtcLatencySum, webrtcLatency.Nanoseconds())
+	atomic.AddInt64(&lp.ipcLatencySum, ipcLatency.Nanoseconds())
+	atomic.AddInt64(&lp.cgoLatencySum, cgoLatency.Nanoseconds())
+	atomic.AddInt64(&lp.alsaLatencySum, alsaLatency.Nanoseconds())
+	atomic.AddInt64(&lp.endToEndLatencySum, endToEndLatency.Nanoseconds())
+	atomic.AddInt64(&lp.validationLatencySum, validationLatency.Nanoseconds())
+	atomic.AddInt64(&lp.serializationLatencySum, serializationLatency.Nanoseconds())
+
+	// Update peak latencies
+	lp.updatePeakLatency(&lp.maxWebrtcLatency, webrtcLatency.Nanoseconds())
+	lp.updatePeakLatency(&lp.maxIpcLatency, ipcLatency.Nanoseconds())
+	lp.updatePeakLatency(&lp.maxCgoLatency, cgoLatency.Nanoseconds())
+	lp.updatePeakLatency(&lp.maxAlsaLatency, alsaLatency.Nanoseconds())
+	lp.updatePeakLatency(&lp.maxEndToEndLatency, endToEndLatency.Nanoseconds())
+
+	// Store detailed measurement if enabled
+	if lp.config.EnableDetailedTrace {
+		lp.storeMeasurement(DetailedLatencyMeasurement{
+			Timestamp:            time.Now(),
+			FrameID:              tracker.frameID,
+			WebRTCLatency:        webrtcLatency,
+			IPCLatency:           ipcLatency,
+			CGOLatency:           cgoLatency,
+			ALSALatency:          alsaLatency,
+			ValidationLatency:    validationLatency,
+			SerializationLatency: serializationLatency,
+			EndToEndLatency:      endToEndLatency,
+			Source:               tracker.source,
+			FrameSize:            tracker.frameSize,
+			CPUUsage:             lp.getCurrentCPUUsage(),
+			MemoryUsage:          lp.getCurrentMemoryUsage(),
+		})
+	}
+
+	// Check for threshold violations
+	if endToEndLatency > lp.config.ThresholdCritical {
+		lp.logger.Error().Dur("latency", endToEndLatency).Uint64("frame_id", tracker.frameID).
+			Str("source", tracker.source).Msg("critical latency threshold exceeded")
+	} else if endToEndLatency > lp.config.ThresholdWarning {
+		lp.logger.Warn().Dur("latency", endToEndLatency).Uint64("frame_id", tracker.frameID).
+			Str("source", tracker.source).Msg("warning latency threshold exceeded")
+	}
+}
+
+// updatePeakLatency atomically updates peak latency if new value is higher
+func (lp *LatencyProfiler) updatePeakLatency(peakPtr *int64, newLatency int64) {
+	for {
+		current := atomic.LoadInt64(peakPtr)
+		if newLatency <= current || atomic.CompareAndSwapInt64(peakPtr, current, newLatency) {
+			break
+		}
+	}
+}
+
+// storeMeasurement stores a detailed measurement in the circular buffer
+func (lp *LatencyProfiler) storeMeasurement(measurement DetailedLatencyMeasurement) {
+	lp.measurementMutex.Lock()
+	defer lp.measurementMutex.Unlock()
+
+	lp.measurements[lp.measurementIndex] = measurement
+	lp.measurementIndex = (lp.measurementIndex + 1) % len(lp.measurements)
+}
+
+// GetReport generates a comprehensive latency profiling report
+func (lp *LatencyProfiler) GetReport() LatencyProfileReport {
+	totalMeasurements := atomic.LoadInt64(&lp.totalMeasurements)
+	if totalMeasurements == 0 {
+		return LatencyProfileReport{}
+	}
+
+	// Calculate averages
+	avgWebRTC := time.Duration(atomic.LoadInt64(&lp.webrtcLatencySum) / totalMeasurements)
+	avgIPC := time.Duration(atomic.LoadInt64(&lp.ipcLatencySum) / totalMeasurements)
+	avgCGO := time.Duration(atomic.LoadInt64(&lp.cgoLatencySum) / totalMeasurements)
+	avgALSA := time.Duration(atomic.LoadInt64(&lp.alsaLatencySum) / totalMeasurements)
+	avgEndToEnd := time.Duration(atomic.LoadInt64(&lp.endToEndLatencySum) / totalMeasurements)
+	avgValidation := time.Duration(atomic.LoadInt64(&lp.validationLatencySum) / totalMeasurements)
+	avgSerialization := time.Duration(atomic.LoadInt64(&lp.serializationLatencySum) / totalMeasurements)
+
+	// Get peak latencies
+	maxWebRTC := time.Duration(atomic.LoadInt64(&lp.maxWebrtcLatency))
+	maxIPC := time.Duration(atomic.LoadInt64(&lp.maxIpcLatency))
+	maxCGO := time.Duration(atomic.LoadInt64(&lp.maxCgoLatency))
+	maxALSA := time.Duration(atomic.LoadInt64(&lp.maxAlsaLatency))
+	maxEndToEnd := time.Duration(atomic.LoadInt64(&lp.maxEndToEndLatency))
+
+	// Determine bottleneck component
+	bottleneck := "WebRTC"
+	maxAvg := avgWebRTC
+	if avgIPC > maxAvg {
+		bottleneck = "IPC"
+		maxAvg = avgIPC
+	}
+	if avgCGO > maxAvg {
+		bottleneck = "CGO"
+		maxAvg = avgCGO
+	}
+	if avgALSA > maxAvg {
+		bottleneck = "ALSA"
+	}
+
+	return LatencyProfileReport{
+		TotalMeasurements:       totalMeasurements,
+		AvgWebRTCLatency:        avgWebRTC,
+		AvgIPCLatency:           avgIPC,
+		AvgCGOLatency:           avgCGO,
+		AvgALSALatency:          avgALSA,
+		AvgEndToEndLatency:      avgEndToEnd,
+		AvgValidationLatency:    avgValidation,
+		AvgSerializationLatency: avgSerialization,
+		MaxWebRTCLatency:        maxWebRTC,
+		MaxIPCLatency:           maxIPC,
+		MaxCGOLatency:           maxCGO,
+		MaxALSALatency:          maxALSA,
+		MaxEndToEndLatency:      maxEndToEnd,
+		BottleneckComponent:     bottleneck,
+	}
+}
+
+// reportingLoop periodically logs profiling reports
+func (lp *LatencyProfiler) reportingLoop() {
+	ticker := time.NewTicker(lp.config.ReportingInterval)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-lp.ctx.Done():
+			return
+		case <-ticker.C:
+			report := lp.GetReport()
+			if report.TotalMeasurements > 0 {
+				lp.logReport(report)
+			}
+		}
+	}
+}
+
+// logReport logs a comprehensive profiling report
+func (lp *LatencyProfiler) logReport(report LatencyProfileReport) {
+	lp.logger.Info().
+		Int64("total_measurements", report.TotalMeasurements).
+		Dur("avg_webrtc_latency", report.AvgWebRTCLatency).
+		Dur("avg_ipc_latency", report.AvgIPCLatency).
+		Dur("avg_cgo_latency", report.AvgCGOLatency).
+		Dur("avg_alsa_latency", report.AvgALSALatency).
+		Dur("avg_end_to_end_latency", report.AvgEndToEndLatency).
+		Dur("avg_validation_latency", report.AvgValidationLatency).
+		Dur("avg_serialization_latency", report.AvgSerializationLatency).
+		Dur("max_webrtc_latency", report.MaxWebRTCLatency).
+		Dur("max_ipc_latency", report.MaxIPCLatency).
+		Dur("max_cgo_latency", report.MaxCGOLatency).
+		Dur("max_alsa_latency", report.MaxALSALatency).
+		Dur("max_end_to_end_latency", report.MaxEndToEndLatency).
+		Str("bottleneck_component", report.BottleneckComponent).
+		Msg("latency profiling report")
+}
+
+// getCurrentCPUUsage returns current CPU usage percentage
+func (lp *LatencyProfiler) getCurrentCPUUsage() float64 {
+	// Simplified CPU usage - in production, this would use more sophisticated monitoring
+	var m runtime.MemStats
+	runtime.ReadMemStats(&m)
+	return float64(runtime.NumGoroutine()) / 100.0 // Rough approximation
+}
+
+// getCurrentMemoryUsage returns current memory usage in bytes
+func (lp *LatencyProfiler) getCurrentMemoryUsage() uint64 {
+	var m runtime.MemStats
+	runtime.ReadMemStats(&m)
+	return m.Alloc
+}
+
+// GetGlobalLatencyProfiler returns the global latency profiler instance
+func GetGlobalLatencyProfiler() *LatencyProfiler {
+	ptr := atomic.LoadPointer(&globalLatencyProfiler)
+	if ptr != nil {
+		return (*LatencyProfiler)(ptr)
+	}
+
+	// Initialize on first use
+	if atomic.CompareAndSwapInt32(&profilerInitialized, 0, 1) {
+		config := DefaultLatencyProfilerConfig()
+		profiler := NewLatencyProfiler(config)
+		atomic.StorePointer(&globalLatencyProfiler, unsafe.Pointer(profiler))
+		return profiler
+	}
+
+	// Another goroutine initialized it, try again
+	ptr = atomic.LoadPointer(&globalLatencyProfiler)
+	if ptr != nil {
+		return (*LatencyProfiler)(ptr)
+	}
+
+	// Fallback: create a new profiler
+	config := DefaultLatencyProfilerConfig()
+	return NewLatencyProfiler(config)
+}
+
+// EnableLatencyProfiling enables the global latency profiler
+func EnableLatencyProfiling() error {
+	profiler := GetGlobalLatencyProfiler()
+	return profiler.Start()
+}
+
+// DisableLatencyProfiling disables the global latency profiler
+func DisableLatencyProfiling() {
+	ptr := atomic.LoadPointer(&globalLatencyProfiler)
+	if ptr != nil {
+		profiler := (*LatencyProfiler)(ptr)
+		profiler.Stop()
+	}
+}
+
+// ProfileFrameLatency is a convenience function to profile a single frame's latency
+func ProfileFrameLatency(frameID uint64, frameSize int, source string, fn func(*FrameLatencyTracker)) {
+	profiler := GetGlobalLatencyProfiler()
+	if !profiler.IsEnabled() {
+		fn(nil)
+		return
+	}
+
+	tracker := profiler.StartFrameTracking(frameID, frameSize, source)
+	defer profiler.FinishTracking(tracker)
+	fn(tracker)
+}

internal/audio/logging_standards.go

@@ -0,0 +1,323 @@
+package audio
+
+import (
+	"time"
+
+	"github.com/rs/zerolog"
+)
+
+// AudioLoggerStandards provides standardized logging patterns for audio components
+type AudioLoggerStandards struct {
+	logger    zerolog.Logger
+	component string
+}
+
+// NewAudioLogger creates a new standardized logger for an audio component
+func NewAudioLogger(logger zerolog.Logger, component string) *AudioLoggerStandards {
+	return &AudioLoggerStandards{
+		logger:    logger.With().Str("component", component).Logger(),
+		component: component,
+	}
+}
+
+// Component Lifecycle Logging
+
+// LogComponentStarting logs component initialization start
+func (als *AudioLoggerStandards) LogComponentStarting() {
+	als.logger.Debug().Msg("starting component")
+}
+
+// LogComponentStarted logs successful component start
+func (als *AudioLoggerStandards) LogComponentStarted() {
+	als.logger.Debug().Msg("component started successfully")
+}
+
+// LogComponentStopping logs component shutdown start
+func (als *AudioLoggerStandards) LogComponentStopping() {
+	als.logger.Debug().Msg("stopping component")
+}
+
+// LogComponentStopped logs successful component stop
+func (als *AudioLoggerStandards) LogComponentStopped() {
+	als.logger.Debug().Msg("component stopped")
+}
+
+// LogComponentReady logs component ready state
+func (als *AudioLoggerStandards) LogComponentReady() {
+	als.logger.Info().Msg("component ready")
+}
+
+// Error Logging with Context
+
+// LogError logs a general error with context
+func (als *AudioLoggerStandards) LogError(err error, msg string) {
+	als.logger.Error().Err(err).Msg(msg)
+}
+
+// LogErrorWithContext logs an error with additional context fields
+func (als *AudioLoggerStandards) LogErrorWithContext(err error, msg string, fields map[string]interface{}) {
+	event := als.logger.Error().Err(err)
+	for key, value := range fields {
+		event = event.Interface(key, value)
+	}
+	event.Msg(msg)
+}
+
+// LogValidationError logs validation failures with specific context
+func (als *AudioLoggerStandards) LogValidationError(err error, validationType string, value interface{}) {
+	als.logger.Error().Err(err).
+		Str("validation_type", validationType).
+		Interface("invalid_value", value).
+		Msg("validation failed")
+}
+
+// LogConnectionError logs connection-related errors
+func (als *AudioLoggerStandards) LogConnectionError(err error, endpoint string, retryCount int) {
+	als.logger.Error().Err(err).
+		Str("endpoint", endpoint).
+		Int("retry_count", retryCount).
+		Msg("connection failed")
+}
+
+// LogProcessError logs process-related errors with PID context
+func (als *AudioLoggerStandards) LogProcessError(err error, pid int, msg string) {
+	als.logger.Error().Err(err).
+		Int("pid", pid).
+		Msg(msg)
+}
+
+// Performance and Metrics Logging
+
+// LogPerformanceMetrics logs standardized performance metrics
+func (als *AudioLoggerStandards) LogPerformanceMetrics(metrics map[string]interface{}) {
+	event := als.logger.Info()
+	for key, value := range metrics {
+		event = event.Interface(key, value)
+	}
+	event.Msg("performance metrics")
+}
+
+// LogLatencyMetrics logs latency-specific metrics
+func (als *AudioLoggerStandards) LogLatencyMetrics(current, average, max time.Duration, jitter time.Duration) {
+	als.logger.Info().
+		Dur("current_latency", current).
+		Dur("average_latency", average).
+		Dur("max_latency", max).
+		Dur("jitter", jitter).
+		Msg("latency metrics")
+}
+
+// LogFrameMetrics logs frame processing metrics
+func (als *AudioLoggerStandards) LogFrameMetrics(processed, dropped int64, rate float64) {
+	als.logger.Info().
+		Int64("frames_processed", processed).
+		Int64("frames_dropped", dropped).
+		Float64("processing_rate", rate).
+		Msg("frame processing metrics")
+}
+
+// LogBufferMetrics logs buffer utilization metrics
+func (als *AudioLoggerStandards) LogBufferMetrics(size, used, peak int, utilizationPercent float64) {
+	als.logger.Info().
+		Int("buffer_size", size).
+		Int("buffer_used", used).
+		Int("buffer_peak", peak).
+		Float64("utilization_percent", utilizationPercent).
+		Msg("buffer metrics")
+}
+
+// Warning Logging
+
+// LogWarning logs a general warning
+func (als *AudioLoggerStandards) LogWarning(msg string) {
+	als.logger.Warn().Msg(msg)
+}
+
+// LogWarningWithError logs a warning with error context
+func (als *AudioLoggerStandards) LogWarningWithError(err error, msg string) {
+	als.logger.Warn().Err(err).Msg(msg)
+}
+
+// LogThresholdWarning logs warnings when thresholds are exceeded
+func (als *AudioLoggerStandards) LogThresholdWarning(metric string, current, threshold interface{}, msg string) {
+	als.logger.Warn().
+		Str("metric", metric).
+		Interface("current_value", current).
+		Interface("threshold", threshold).
+		Msg(msg)
+}
+
+// LogRetryWarning logs retry attempts with context
+func (als *AudioLoggerStandards) LogRetryWarning(operation string, attempt, maxAttempts int, delay time.Duration) {
+	als.logger.Warn().
+		Str("operation", operation).
+		Int("attempt", attempt).
+		Int("max_attempts", maxAttempts).
+		Dur("retry_delay", delay).
+		Msg("retrying operation")
+}
+
+// LogRecoveryWarning logs recovery from error conditions
+func (als *AudioLoggerStandards) LogRecoveryWarning(condition string, duration time.Duration) {
+	als.logger.Warn().
+		Str("condition", condition).
+		Dur("recovery_time", duration).
+		Msg("recovered from error condition")
+}
+
+// Debug and Trace Logging
+
+// LogDebug logs debug information
+func (als *AudioLoggerStandards) LogDebug(msg string) {
+	als.logger.Debug().Msg(msg)
+}
+
+// LogDebugWithFields logs debug information with structured fields
+func (als *AudioLoggerStandards) LogDebugWithFields(msg string, fields map[string]interface{}) {
+	event := als.logger.Debug()
+	for key, value := range fields {
+		event = event.Interface(key, value)
+	}
+	event.Msg(msg)
+}
+
+// LogOperationTrace logs operation tracing for debugging
+func (als *AudioLoggerStandards) LogOperationTrace(operation string, duration time.Duration, success bool) {
+	als.logger.Debug().
+		Str("operation", operation).
+		Dur("duration", duration).
+		Bool("success", success).
+		Msg("operation trace")
+}
+
+// LogDataFlow logs data flow for debugging
+func (als *AudioLoggerStandards) LogDataFlow(source, destination string, bytes int, frameCount int) {
+	als.logger.Debug().
+		Str("source", source).
+		Str("destination", destination).
+		Int("bytes", bytes).
+		Int("frame_count", frameCount).
+		Msg("data flow")
+}
+
+// Configuration and State Logging
+
+// LogConfigurationChange logs configuration updates
+func (als *AudioLoggerStandards) LogConfigurationChange(configType string, oldValue, newValue interface{}) {
+	als.logger.Info().
+		Str("config_type", configType).
+		Interface("old_value", oldValue).
+		Interface("new_value", newValue).
+		Msg("configuration changed")
+}
+
+// LogStateTransition logs component state changes
+func (als *AudioLoggerStandards) LogStateTransition(fromState, toState string, reason string) {
+	als.logger.Info().
+		Str("from_state", fromState).
+		Str("to_state", toState).
+		Str("reason", reason).
+		Msg("state transition")
+}
+
+// LogResourceAllocation logs resource allocation/deallocation
+func (als *AudioLoggerStandards) LogResourceAllocation(resourceType string, allocated bool, amount interface{}) {
+	level := als.logger.Debug()
+	if allocated {
+		level.Str("action", "allocated")
+	} else {
+		level.Str("action", "deallocated")
+	}
+	level.Str("resource_type", resourceType).
+		Interface("amount", amount).
+		Msg("resource allocation")
+}
+
+// Network and IPC Logging
+
+// LogConnectionEvent logs connection lifecycle events
+func (als *AudioLoggerStandards) LogConnectionEvent(event, endpoint string, connectionID string) {
+	als.logger.Info().
+		Str("event", event).
+		Str("endpoint", endpoint).
+		Str("connection_id", connectionID).
+		Msg("connection event")
+}
+
+// LogIPCEvent logs IPC communication events
+func (als *AudioLoggerStandards) LogIPCEvent(event, socketPath string, bytes int) {
+	als.logger.Debug().
+		Str("event", event).
+		Str("socket_path", socketPath).
+		Int("bytes", bytes).
+		Msg("IPC event")
+}
+
+// LogNetworkStats logs network statistics
+func (als *AudioLoggerStandards) LogNetworkStats(sent, received int64, latency time.Duration, packetLoss float64) {
+	als.logger.Info().
+		Int64("bytes_sent", sent).
+		Int64("bytes_received", received).
+		Dur("network_latency", latency).
+		Float64("packet_loss_percent", packetLoss).
+		Msg("network statistics")
+}
+
+// Process and System Logging
+
+// LogProcessEvent logs process lifecycle events
+func (als *AudioLoggerStandards) LogProcessEvent(event string, pid int, exitCode *int) {
+	event_log := als.logger.Info().
+		Str("event", event).
+		Int("pid", pid)
+	if exitCode != nil {
+		event_log = event_log.Int("exit_code", *exitCode)
+	}
+	event_log.Msg("process event")
+}
+
+// LogSystemResource logs system resource usage
+func (als *AudioLoggerStandards) LogSystemResource(cpuPercent, memoryMB float64, goroutines int) {
+	als.logger.Info().
+		Float64("cpu_percent", cpuPercent).
+		Float64("memory_mb", memoryMB).
+		Int("goroutines", goroutines).
+		Msg("system resources")
+}
+
+// LogPriorityChange logs thread priority changes
+func (als *AudioLoggerStandards) LogPriorityChange(tid, oldPriority, newPriority int, policy string) {
+	als.logger.Debug().
+		Int("tid", tid).
+		Int("old_priority", oldPriority).
+		Int("new_priority", newPriority).
+		Str("policy", policy).
+		Msg("thread priority changed")
+}
+
+// Utility Functions
+
+// GetLogger returns the underlying zerolog.Logger for advanced usage
+func (als *AudioLoggerStandards) GetLogger() zerolog.Logger {
+	return als.logger
+}
+
+// WithFields returns a new logger with additional persistent fields
+func (als *AudioLoggerStandards) WithFields(fields map[string]interface{}) *AudioLoggerStandards {
+	event := als.logger.With()
+	for key, value := range fields {
+		event = event.Interface(key, value)
+	}
+	return &AudioLoggerStandards{
+		logger:    event.Logger(),
+		component: als.component,
+	}
+}
+
+// WithSubComponent creates a logger for a sub-component
+func (als *AudioLoggerStandards) WithSubComponent(subComponent string) *AudioLoggerStandards {
+	return &AudioLoggerStandards{
+		logger:    als.logger.With().Str("sub_component", subComponent).Logger(),
+		component: als.component + "." + subComponent,
+	}
+}

internal/audio/memory_metrics.go

@@ -0,0 +1,201 @@
+package audio
+
+import (
+	"encoding/json"
+	"net/http"
+	"runtime"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// MemoryMetrics provides comprehensive memory allocation statistics
+type MemoryMetrics struct {
+	// Runtime memory statistics
+	RuntimeStats RuntimeMemoryStats `json:"runtime_stats"`
+	// Audio buffer pool statistics
+	BufferPools AudioBufferPoolStats `json:"buffer_pools"`
+	// Zero-copy frame pool statistics
+	ZeroCopyPool ZeroCopyFramePoolStats `json:"zero_copy_pool"`
+	// Message pool statistics
+	MessagePool MessagePoolStats `json:"message_pool"`
+	// Batch processor statistics
+	BatchProcessor BatchProcessorMemoryStats `json:"batch_processor,omitempty"`
+	// Collection timestamp
+	Timestamp time.Time `json:"timestamp"`
+}
+
+// RuntimeMemoryStats provides Go runtime memory statistics
+type RuntimeMemoryStats struct {
+	Alloc         uint64  `json:"alloc"`           // Bytes allocated and not yet freed
+	TotalAlloc    uint64  `json:"total_alloc"`     // Total bytes allocated (cumulative)
+	Sys           uint64  `json:"sys"`             // Total bytes obtained from OS
+	Lookups       uint64  `json:"lookups"`         // Number of pointer lookups
+	Mallocs       uint64  `json:"mallocs"`         // Number of mallocs
+	Frees         uint64  `json:"frees"`           // Number of frees
+	HeapAlloc     uint64  `json:"heap_alloc"`      // Bytes allocated and not yet freed (heap)
+	HeapSys       uint64  `json:"heap_sys"`        // Bytes obtained from OS for heap
+	HeapIdle      uint64  `json:"heap_idle"`       // Bytes in idle spans
+	HeapInuse     uint64  `json:"heap_inuse"`      // Bytes in non-idle spans
+	HeapReleased  uint64  `json:"heap_released"`   // Bytes released to OS
+	HeapObjects   uint64  `json:"heap_objects"`    // Total number of allocated objects
+	StackInuse    uint64  `json:"stack_inuse"`     // Bytes used by stack spans
+	StackSys      uint64  `json:"stack_sys"`       // Bytes obtained from OS for stack
+	MSpanInuse    uint64  `json:"mspan_inuse"`     // Bytes used by mspan structures
+	MSpanSys      uint64  `json:"mspan_sys"`       // Bytes obtained from OS for mspan
+	MCacheInuse   uint64  `json:"mcache_inuse"`    // Bytes used by mcache structures
+	MCacheSys     uint64  `json:"mcache_sys"`      // Bytes obtained from OS for mcache
+	BuckHashSys   uint64  `json:"buck_hash_sys"`   // Bytes used by profiling bucket hash table
+	GCSys         uint64  `json:"gc_sys"`          // Bytes used for garbage collection metadata
+	OtherSys      uint64  `json:"other_sys"`       // Bytes used for other system allocations
+	NextGC        uint64  `json:"next_gc"`         // Target heap size for next GC
+	LastGC        uint64  `json:"last_gc"`         // Time of last GC (nanoseconds since epoch)
+	PauseTotalNs  uint64  `json:"pause_total_ns"`  // Total GC pause time
+	NumGC         uint32  `json:"num_gc"`          // Number of completed GC cycles
+	NumForcedGC   uint32  `json:"num_forced_gc"`   // Number of forced GC cycles
+	GCCPUFraction float64 `json:"gc_cpu_fraction"` // Fraction of CPU time used by GC
+}
+
+// BatchProcessorMemoryStats provides batch processor memory statistics
+type BatchProcessorMemoryStats struct {
+	Initialized bool                         `json:"initialized"`
+	Running     bool                         `json:"running"`
+	Stats       BatchAudioStats              `json:"stats"`
+	BufferPool  AudioBufferPoolDetailedStats `json:"buffer_pool,omitempty"`
+}
+
+// GetBatchAudioProcessor is defined in batch_audio.go
+// BatchAudioStats is defined in batch_audio.go
+
+var memoryMetricsLogger *zerolog.Logger
+
+func getMemoryMetricsLogger() *zerolog.Logger {
+	if memoryMetricsLogger == nil {
+		logger := logging.GetDefaultLogger().With().Str("component", "memory-metrics").Logger()
+		memoryMetricsLogger = &logger
+	}
+	return memoryMetricsLogger
+}
+
+// CollectMemoryMetrics gathers comprehensive memory allocation statistics
+func CollectMemoryMetrics() MemoryMetrics {
+	// Collect runtime memory statistics
+	var m runtime.MemStats
+	runtime.ReadMemStats(&m)
+
+	runtimeStats := RuntimeMemoryStats{
+		Alloc:         m.Alloc,
+		TotalAlloc:    m.TotalAlloc,
+		Sys:           m.Sys,
+		Lookups:       m.Lookups,
+		Mallocs:       m.Mallocs,
+		Frees:         m.Frees,
+		HeapAlloc:     m.HeapAlloc,
+		HeapSys:       m.HeapSys,
+		HeapIdle:      m.HeapIdle,
+		HeapInuse:     m.HeapInuse,
+		HeapReleased:  m.HeapReleased,
+		HeapObjects:   m.HeapObjects,
+		StackInuse:    m.StackInuse,
+		StackSys:      m.StackSys,
+		MSpanInuse:    m.MSpanInuse,
+		MSpanSys:      m.MSpanSys,
+		MCacheInuse:   m.MCacheInuse,
+		MCacheSys:     m.MCacheSys,
+		BuckHashSys:   m.BuckHashSys,
+		GCSys:         m.GCSys,
+		OtherSys:      m.OtherSys,
+		NextGC:        m.NextGC,
+		LastGC:        m.LastGC,
+		PauseTotalNs:  m.PauseTotalNs,
+		NumGC:         m.NumGC,
+		NumForcedGC:   m.NumForcedGC,
+		GCCPUFraction: m.GCCPUFraction,
+	}
+
+	// Collect audio buffer pool statistics
+	bufferPoolStats := GetAudioBufferPoolStats()
+
+	// Collect zero-copy frame pool statistics
+	zeroCopyStats := GetGlobalZeroCopyPoolStats()
+
+	// Collect message pool statistics
+	messagePoolStats := GetGlobalMessagePoolStats()
+
+	// Collect batch processor statistics if available
+	var batchStats BatchProcessorMemoryStats
+	if processor := GetBatchAudioProcessor(); processor != nil {
+		batchStats.Initialized = true
+		batchStats.Running = processor.IsRunning()
+		batchStats.Stats = processor.GetStats()
+		// Note: BatchAudioProcessor uses sync.Pool, detailed stats not available
+	}
+
+	return MemoryMetrics{
+		RuntimeStats:   runtimeStats,
+		BufferPools:    bufferPoolStats,
+		ZeroCopyPool:   zeroCopyStats,
+		MessagePool:    messagePoolStats,
+		BatchProcessor: batchStats,
+		Timestamp:      time.Now(),
+	}
+}
+
+// HandleMemoryMetrics provides an HTTP handler for memory metrics
+func HandleMemoryMetrics(w http.ResponseWriter, r *http.Request) {
+	logger := getMemoryMetricsLogger()
+
+	if r.Method != http.MethodGet {
+		http.Error(w, "Method not allowed", http.StatusMethodNotAllowed)
+		return
+	}
+
+	metrics := CollectMemoryMetrics()
+
+	w.Header().Set("Content-Type", "application/json")
+	w.Header().Set("Cache-Control", "no-cache")
+
+	encoder := json.NewEncoder(w)
+	encoder.SetIndent("", "  ")
+
+	if err := encoder.Encode(metrics); err != nil {
+		logger.Error().Err(err).Msg("failed to encode memory metrics")
+		http.Error(w, "Internal server error", http.StatusInternalServerError)
+		return
+	}
+
+	logger.Debug().Msg("memory metrics served")
+}
+
+// LogMemoryMetrics logs current memory metrics for debugging
+func LogMemoryMetrics() {
+	logger := getMemoryMetricsLogger()
+	metrics := CollectMemoryMetrics()
+
+	logger.Info().
+		Uint64("heap_alloc_mb", metrics.RuntimeStats.HeapAlloc/uint64(GetConfig().BytesToMBDivisor)).
+		Uint64("heap_sys_mb", metrics.RuntimeStats.HeapSys/uint64(GetConfig().BytesToMBDivisor)).
+		Uint64("heap_objects", metrics.RuntimeStats.HeapObjects).
+		Uint32("num_gc", metrics.RuntimeStats.NumGC).
+		Float64("gc_cpu_fraction", metrics.RuntimeStats.GCCPUFraction).
+		Float64("buffer_pool_hit_rate", metrics.BufferPools.FramePoolHitRate).
+		Float64("zero_copy_hit_rate", metrics.ZeroCopyPool.HitRate).
+		Float64("message_pool_hit_rate", metrics.MessagePool.HitRate).
+		Msg("memory metrics snapshot")
+}
+
+// StartMemoryMetricsLogging starts periodic memory metrics logging
+func StartMemoryMetricsLogging(interval time.Duration) {
+	logger := getMemoryMetricsLogger()
+	logger.Debug().Dur("interval", interval).Msg("memory metrics logging started")
+
+	go func() {
+		ticker := time.NewTicker(interval)
+		defer ticker.Stop()
+
+		for range ticker.C {
+			LogMemoryMetrics()
+		}
+	}()
+}

internal/audio/metrics.go

@@ -288,7 +288,45 @@ var (
 	)
 
 	// Device health metrics
-	// Removed device health metrics - functionality not used
+	deviceHealthStatus = promauto.NewGaugeVec(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_device_health_status",
+			Help: "Current device health status (0=Healthy, 1=Degraded, 2=Failing, 3=Critical)",
+		},
+		[]string{"device_type"}, // device_type: capture, playback
+	)
+
+	deviceHealthScore = promauto.NewGaugeVec(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_device_health_score",
+			Help: "Device health score (0.0-1.0, higher is better)",
+		},
+		[]string{"device_type"}, // device_type: capture, playback
+	)
+
+	deviceConsecutiveErrors = promauto.NewGaugeVec(
+		prometheus.GaugeOpts{
+			Name: "jetkvm_audio_device_consecutive_errors",
+			Help: "Number of consecutive errors for device",
+		},
+		[]string{"device_type"}, // device_type: capture, playback
+	)
+
+	deviceTotalErrors = promauto.NewCounterVec(
+		prometheus.CounterOpts{
+			Name: "jetkvm_audio_device_total_errors",
+			Help: "Total number of errors for device",
+		},
+		[]string{"device_type"}, // device_type: capture, playback
+	)
+
+	deviceLatencySpikes = promauto.NewCounterVec(
+		prometheus.CounterOpts{
+			Name: "jetkvm_audio_device_latency_spikes_total",
+			Help: "Total number of latency spikes for device",
+		},
+		[]string{"device_type"}, // device_type: capture, playback
+	)
 
 	// Memory metrics
 	memoryHeapAllocBytes = promauto.NewGauge(
@@ -398,7 +436,11 @@ var (
 	micBytesProcessedValue    int64
 	micConnectionDropsValue   int64
 
-	// Atomic counters for device health metrics - functionality removed, no longer used
+	// Atomic counters for device health metrics
+	deviceCaptureErrorsValue  int64
+	devicePlaybackErrorsValue int64
+	deviceCaptureSpikesValue  int64
+	devicePlaybackSpikesValue int64
 
 	// Atomic counter for memory GC
 	memoryGCCountValue uint32
@@ -597,8 +639,34 @@ func UpdateSocketBufferMetrics(component, bufferType string, size, utilization f
 	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
 }
 
-// UpdateDeviceHealthMetrics - Device health monitoring functionality has been removed
+// UpdateDeviceHealthMetrics updates device health metrics
-// This function is no longer used as device health monitoring is not implemented
+func UpdateDeviceHealthMetrics(deviceType string, status int, healthScore float64, consecutiveErrors, totalErrors, latencySpikes int64) {
+	metricsUpdateMutex.Lock()
+	defer metricsUpdateMutex.Unlock()
+
+	deviceHealthStatus.WithLabelValues(deviceType).Set(float64(status))
+	deviceHealthScore.WithLabelValues(deviceType).Set(healthScore)
+	deviceConsecutiveErrors.WithLabelValues(deviceType).Set(float64(consecutiveErrors))
+
+	// Update error counters with delta calculation
+	var prevErrors, prevSpikes int64
+	if deviceType == "capture" {
+		prevErrors = atomic.SwapInt64(&deviceCaptureErrorsValue, totalErrors)
+		prevSpikes = atomic.SwapInt64(&deviceCaptureSpikesValue, latencySpikes)
+	} else {
+		prevErrors = atomic.SwapInt64(&devicePlaybackErrorsValue, totalErrors)
+		prevSpikes = atomic.SwapInt64(&devicePlaybackSpikesValue, latencySpikes)
+	}
+
+	if prevErrors > 0 && totalErrors > prevErrors {
+		deviceTotalErrors.WithLabelValues(deviceType).Add(float64(totalErrors - prevErrors))
+	}
+	if prevSpikes > 0 && latencySpikes > prevSpikes {
+		deviceLatencySpikes.WithLabelValues(deviceType).Add(float64(latencySpikes - prevSpikes))
+	}
+
+	atomic.StoreInt64(&lastMetricsUpdate, time.Now().Unix())
+}
 
 // UpdateMemoryMetrics updates memory metrics
 func UpdateMemoryMetrics() {

internal/audio/naming_standards.go

@@ -97,6 +97,7 @@ type AudioSupervisorInterface interface {
 	Stop() error
 	IsRunning() bool
 	GetProcessPID() int
+	GetProcessMetrics() *ProcessMetrics
 }
 
 type AudioServerInterface interface {

internal/audio/output_manager.go

@@ -145,6 +145,20 @@ func (aom *AudioOutputManager) GetComprehensiveMetrics() map[string]interface{}
 	return comprehensiveMetrics
 }
 
+// LogPerformanceStats logs current performance statistics
+func (aom *AudioOutputManager) LogPerformanceStats() {
+	metrics := aom.GetMetrics()
+	aom.logger.Info().
+		Int64("frames_received", metrics.FramesReceived).
+		Int64("frames_dropped", metrics.FramesDropped).
+		Int64("bytes_processed", metrics.BytesProcessed).
+		Int64("connection_drops", metrics.ConnectionDrops).
+		Float64("average_latency_ms", float64(metrics.AverageLatency.Nanoseconds())/1e6).
+		Bool("running", aom.IsRunning()).
+		Bool("ready", aom.IsReady()).
+		Msg("Audio output manager performance stats")
+}
+
 // GetStreamer returns the streamer for advanced operations
 func (aom *AudioOutputManager) GetStreamer() *AudioOutputStreamer {
 	return aom.streamer

internal/audio/output_manager_test.go

@@ -0,0 +1,277 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestAudioOutputManager tests the AudioOutputManager component
+func TestAudioOutputManager(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"Start", testAudioOutputManagerStart},
+		{"Stop", testAudioOutputManagerStop},
+		{"StartStop", testAudioOutputManagerStartStop},
+		{"IsRunning", testAudioOutputManagerIsRunning},
+		{"IsReady", testAudioOutputManagerIsReady},
+		{"GetMetrics", testAudioOutputManagerGetMetrics},
+		{"ConcurrentOperations", testAudioOutputManagerConcurrent},
+		{"MultipleStarts", testAudioOutputManagerMultipleStarts},
+		{"MultipleStops", testAudioOutputManagerMultipleStops},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+func testAudioOutputManagerStart(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Test initial state
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+
+	// Test start
+	err := manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioOutputManagerStop(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Start first
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Test stop
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+}
+
+func testAudioOutputManagerStartStop(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Test multiple start/stop cycles
+	for i := 0; i < 3; i++ {
+		// Start
+		err := manager.Start()
+		assert.NoError(t, err)
+		assert.True(t, manager.IsRunning())
+
+		// Stop
+		manager.Stop()
+		assert.False(t, manager.IsRunning())
+	}
+}
+
+func testAudioOutputManagerIsRunning(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Initially not running
+	assert.False(t, manager.IsRunning())
+
+	// Start and check
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Stop and check
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+func testAudioOutputManagerIsReady(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Initially not ready
+	assert.False(t, manager.IsReady())
+
+	// Start and check ready state
+	err := manager.Start()
+	require.NoError(t, err)
+
+	// Give some time for initialization
+	time.Sleep(100 * time.Millisecond)
+
+	// Stop
+	manager.Stop()
+	assert.False(t, manager.IsReady())
+}
+
+func testAudioOutputManagerGetMetrics(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Test metrics when not running
+	metrics := manager.GetMetrics()
+	assert.NotNil(t, metrics)
+
+	// Start and test metrics
+	err := manager.Start()
+	require.NoError(t, err)
+
+	metrics = manager.GetMetrics()
+	assert.NotNil(t, metrics)
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioOutputManagerConcurrent(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	var wg sync.WaitGroup
+	const numGoroutines = 10
+
+	// Test concurrent starts
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			manager.Start()
+		}()
+	}
+	wg.Wait()
+
+	// Should be running
+	assert.True(t, manager.IsRunning())
+
+	// Test concurrent stops
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			manager.Stop()
+		}()
+	}
+	wg.Wait()
+
+	// Should be stopped
+	assert.False(t, manager.IsRunning())
+}
+
+func testAudioOutputManagerMultipleStarts(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// First start should succeed
+	err := manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Subsequent starts should be no-op
+	err = manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	err = manager.Start()
+	assert.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// Cleanup
+	manager.Stop()
+}
+
+func testAudioOutputManagerMultipleStops(t *testing.T) {
+	manager := NewAudioOutputManager()
+	require.NotNil(t, manager)
+
+	// Start first
+	err := manager.Start()
+	require.NoError(t, err)
+	assert.True(t, manager.IsRunning())
+
+	// First stop should work
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+
+	// Subsequent stops should be no-op
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}
+
+// TestAudioOutputMetrics tests the AudioOutputMetrics functionality
+func TestAudioOutputMetrics(t *testing.T) {
+	metrics := &AudioOutputMetrics{}
+
+	// Test initial state
+	assert.Equal(t, int64(0), metrics.FramesReceived)
+	assert.Equal(t, int64(0), metrics.FramesDropped)
+	assert.Equal(t, int64(0), metrics.BytesProcessed)
+	assert.Equal(t, int64(0), metrics.ConnectionDrops)
+	assert.Equal(t, time.Duration(0), metrics.AverageLatency)
+	assert.True(t, metrics.LastFrameTime.IsZero())
+
+	// Test field assignment
+	metrics.FramesReceived = 100
+	metrics.FramesDropped = 5
+	metrics.BytesProcessed = 1024
+	metrics.ConnectionDrops = 2
+	metrics.AverageLatency = 10 * time.Millisecond
+	metrics.LastFrameTime = time.Now()
+
+	// Verify assignments
+	assert.Equal(t, int64(100), metrics.FramesReceived)
+	assert.Equal(t, int64(5), metrics.FramesDropped)
+	assert.Equal(t, int64(1024), metrics.BytesProcessed)
+	assert.Equal(t, int64(2), metrics.ConnectionDrops)
+	assert.Equal(t, 10*time.Millisecond, metrics.AverageLatency)
+	assert.False(t, metrics.LastFrameTime.IsZero())
+}
+
+// BenchmarkAudioOutputManager benchmarks the AudioOutputManager operations
+func BenchmarkAudioOutputManager(b *testing.B) {
+	b.Run("Start", func(b *testing.B) {
+		for i := 0; i < b.N; i++ {
+			manager := NewAudioOutputManager()
+			manager.Start()
+			manager.Stop()
+		}
+	})
+
+	b.Run("IsRunning", func(b *testing.B) {
+		manager := NewAudioOutputManager()
+		manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			manager.IsRunning()
+		}
+	})
+
+	b.Run("GetMetrics", func(b *testing.B) {
+		manager := NewAudioOutputManager()
+		manager.Start()
+		defer manager.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			manager.GetMetrics()
+		}
+	})
+}

internal/audio/output_streaming.go

@@ -18,15 +18,12 @@ import (
 
 // AudioOutputStreamer manages high-performance audio output streaming
 type AudioOutputStreamer struct {
-	// Atomic int64 fields MUST be first for ARM32 alignment (8-byte alignment required)
+	// Performance metrics (atomic operations for thread safety)
 	processedFrames int64 // Total processed frames counter (atomic)
 	droppedFrames   int64 // Dropped frames counter (atomic)
 	processingTime  int64 // Average processing time in nanoseconds (atomic)
 	lastStatsTime   int64 // Last statistics update time (atomic)
 
-	// Other fields after atomic int64 fields
-	sampleRate int32 // Sample every N frames (default: 10)
-
 	client     *AudioOutputClient
 	bufferPool *AudioBufferPool
 	ctx        context.Context
@@ -73,7 +70,6 @@ func NewAudioOutputStreamer() (*AudioOutputStreamer, error) {
 		processingChan: make(chan []byte, GetConfig().ChannelBufferSize), // Large buffer for smooth processing
 		statsInterval:  GetConfig().StatsUpdateInterval,                  // Statistics interval from config
 		lastStatsTime:  time.Now().UnixNano(),
-		sampleRate:     10, // Update metrics every 10 frames to reduce atomic ops
 	}, nil
 }
 
@@ -112,9 +108,6 @@ func (s *AudioOutputStreamer) Stop() {
 	s.running = false
 	s.cancel()
 
-	// Flush any pending sampled metrics before stopping
-	s.flushPendingMetrics()
-
 	// Close processing channel to signal goroutines (only if not already closed)
 	if !s.chanClosed {
 		close(s.processingChan)
@@ -132,14 +125,9 @@ func (s *AudioOutputStreamer) Stop() {
 func (s *AudioOutputStreamer) streamLoop() {
 	defer s.wg.Done()
 
-	// Only pin to OS thread for high-throughput scenarios to reduce scheduler interference
+	// Pin goroutine to OS thread for consistent performance
-	config := GetConfig()
+	runtime.LockOSThread()
-	useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
+	defer runtime.UnlockOSThread()
-
-	if useThreadOptimizations {
-		runtime.LockOSThread()
-		defer runtime.UnlockOSThread()
-	}
 
 	// Adaptive timing for frame reading
 	frameInterval := time.Duration(GetConfig().OutputStreamingFrameIntervalMS) * time.Millisecond // 50 FPS base rate
@@ -200,15 +188,19 @@ func (s *AudioOutputStreamer) streamLoop() {
 func (s *AudioOutputStreamer) processingLoop() {
 	defer s.wg.Done()
 
-	// Only use thread optimizations for high-throughput scenarios
+	// Pin goroutine to OS thread for consistent performance
-	config := GetConfig()
+	runtime.LockOSThread()
-	useThreadOptimizations := config.MaxAudioProcessorWorkers > 8
+	defer runtime.UnlockOSThread()
 
-	if useThreadOptimizations {
+	// Set high priority for audio output processing
-		// Pin goroutine to OS thread for consistent performance
+	if err := SetAudioThreadPriority(); err != nil {
-		runtime.LockOSThread()
+		getOutputStreamingLogger().Warn().Err(err).Msg("Failed to set audio output processing priority")
-		defer runtime.UnlockOSThread()
 	}
+	defer func() {
+		if err := ResetThreadPriority(); err != nil {
+			getOutputStreamingLogger().Warn().Err(err).Msg("Failed to reset thread priority")
+		}
+	}()
 
 	for frameData := range s.processingChan {
 		// Process frame and return buffer to pool after processing
@@ -217,23 +209,15 @@ func (s *AudioOutputStreamer) processingLoop() {
 
 			if _, err := s.client.ReceiveFrame(); err != nil {
 				if s.client.IsConnected() {
-					// Sample logging to reduce overhead - log every 50th error
+					getOutputStreamingLogger().Warn().Err(err).Msg("Error reading audio frame from output server")
-					if atomic.LoadInt64(&s.droppedFrames)%50 == 0 && getOutputStreamingLogger().GetLevel() <= zerolog.WarnLevel {
+					atomic.AddInt64(&s.droppedFrames, 1)
-						getOutputStreamingLogger().Warn().Err(err).Msg("Error reading audio frame from output server")
-					}
-					s.recordFrameDropped()
 				}
 				// Try to reconnect if disconnected
 				if !s.client.IsConnected() {
 					if err := s.client.Connect(); err != nil {
-						// Only log reconnection failures if warn level enabled
+						getOutputStreamingLogger().Warn().Err(err).Msg("Failed to reconnect")
-						if getOutputStreamingLogger().GetLevel() <= zerolog.WarnLevel {
-							getOutputStreamingLogger().Warn().Err(err).Msg("Failed to reconnect")
-						}
 					}
 				}
-			} else {
-				s.recordFrameProcessed()
 			}
 		}()
 	}
@@ -274,23 +258,8 @@ func (s *AudioOutputStreamer) reportStatistics() {
 	}
 }
 
-// recordFrameProcessed records a processed frame with sampling optimization
+// GetStats returns streaming statistics
-func (s *AudioOutputStreamer) recordFrameProcessed() {
-}
-
-// recordFrameDropped records a dropped frame with sampling optimization
-func (s *AudioOutputStreamer) recordFrameDropped() {
-}
-
-// flushPendingMetrics flushes any pending sampled metrics to atomic counters
-func (s *AudioOutputStreamer) flushPendingMetrics() {
-}
-
-// GetStats returns streaming statistics with pending metrics flushed
 func (s *AudioOutputStreamer) GetStats() (processed, dropped int64, avgProcessingTime time.Duration) {
-	// Flush pending metrics for accurate reading
-	s.flushPendingMetrics()
-
 	processed = atomic.LoadInt64(&s.processedFrames)
 	dropped = atomic.LoadInt64(&s.droppedFrames)
 	processingTimeNs := atomic.LoadInt64(&s.processingTime)
@@ -300,9 +269,6 @@ func (s *AudioOutputStreamer) GetStats() (processed, dropped int64, avgProcessin
 
 // GetDetailedStats returns comprehensive streaming statistics
 func (s *AudioOutputStreamer) GetDetailedStats() map[string]interface{} {
-	// Flush pending metrics for accurate reading
-	s.flushPendingMetrics()
-
 	processed := atomic.LoadInt64(&s.processedFrames)
 	dropped := atomic.LoadInt64(&s.droppedFrames)
 	processingTime := atomic.LoadInt64(&s.processingTime)

internal/audio/output_streaming_test.go

@@ -0,0 +1,341 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestAudioOutputStreamer tests the AudioOutputStreamer component
+func TestAudioOutputStreamer(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"NewAudioOutputStreamer", testNewAudioOutputStreamer},
+		{"Start", testAudioOutputStreamerStart},
+		{"Stop", testAudioOutputStreamerStop},
+		{"StartStop", testAudioOutputStreamerStartStop},
+		{"GetStats", testAudioOutputStreamerGetStats},
+		{"GetDetailedStats", testAudioOutputStreamerGetDetailedStats},
+		{"UpdateBatchSize", testAudioOutputStreamerUpdateBatchSize},
+		{"ReportLatency", testAudioOutputStreamerReportLatency},
+		{"ConcurrentOperations", testAudioOutputStreamerConcurrent},
+		{"MultipleStarts", testAudioOutputStreamerMultipleStarts},
+		{"MultipleStops", testAudioOutputStreamerMultipleStops},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+func testNewAudioOutputStreamer(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		// If creation fails due to missing dependencies, skip the test
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test initial state
+	processed, dropped, avgTime := streamer.GetStats()
+	assert.GreaterOrEqual(t, processed, int64(0))
+	assert.GreaterOrEqual(t, dropped, int64(0))
+	assert.GreaterOrEqual(t, avgTime, time.Duration(0))
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerStart(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test start
+	err = streamer.Start()
+	assert.NoError(t, err)
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerStop(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Start first
+	err = streamer.Start()
+	require.NoError(t, err)
+
+	// Test stop
+	streamer.Stop()
+
+	// Multiple stops should be safe
+	streamer.Stop()
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerStartStop(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test multiple start/stop cycles
+	for i := 0; i < 3; i++ {
+		// Start
+		err = streamer.Start()
+		assert.NoError(t, err)
+
+		// Stop
+		streamer.Stop()
+	}
+}
+
+func testAudioOutputStreamerGetStats(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test stats when not running
+	processed, dropped, avgTime := streamer.GetStats()
+	assert.Equal(t, int64(0), processed)
+	assert.Equal(t, int64(0), dropped)
+	assert.GreaterOrEqual(t, avgTime, time.Duration(0))
+
+	// Start and test stats
+	err = streamer.Start()
+	require.NoError(t, err)
+
+	processed, dropped, avgTime = streamer.GetStats()
+	assert.GreaterOrEqual(t, processed, int64(0))
+	assert.GreaterOrEqual(t, dropped, int64(0))
+	assert.GreaterOrEqual(t, avgTime, time.Duration(0))
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerGetDetailedStats(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test detailed stats
+	stats := streamer.GetDetailedStats()
+	assert.NotNil(t, stats)
+	assert.Contains(t, stats, "processed_frames")
+	assert.Contains(t, stats, "dropped_frames")
+	assert.Contains(t, stats, "batch_size")
+	assert.Contains(t, stats, "connected")
+	assert.Equal(t, int64(0), stats["processed_frames"])
+	assert.Equal(t, int64(0), stats["dropped_frames"])
+
+	// Start and test detailed stats
+	err = streamer.Start()
+	require.NoError(t, err)
+
+	stats = streamer.GetDetailedStats()
+	assert.NotNil(t, stats)
+	assert.Contains(t, stats, "processed_frames")
+	assert.Contains(t, stats, "dropped_frames")
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerUpdateBatchSize(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test updating batch size (no parameters, uses adaptive manager)
+	streamer.UpdateBatchSize()
+	streamer.UpdateBatchSize()
+	streamer.UpdateBatchSize()
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerReportLatency(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Test reporting latency
+	streamer.ReportLatency(10 * time.Millisecond)
+	streamer.ReportLatency(5 * time.Millisecond)
+	streamer.ReportLatency(15 * time.Millisecond)
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerConcurrent(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	var wg sync.WaitGroup
+	const numGoroutines = 10
+
+	// Test concurrent starts
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			streamer.Start()
+		}()
+	}
+	wg.Wait()
+
+	// Test concurrent operations
+	wg.Add(numGoroutines * 3)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			streamer.GetStats()
+		}()
+		go func() {
+			defer wg.Done()
+			streamer.UpdateBatchSize()
+		}()
+		go func() {
+			defer wg.Done()
+			streamer.ReportLatency(10 * time.Millisecond)
+		}()
+	}
+	wg.Wait()
+
+	// Test concurrent stops
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			streamer.Stop()
+		}()
+	}
+	wg.Wait()
+}
+
+func testAudioOutputStreamerMultipleStarts(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// First start should succeed
+	err = streamer.Start()
+	assert.NoError(t, err)
+
+	// Subsequent starts should return error
+	err = streamer.Start()
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "already running")
+
+	err = streamer.Start()
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "already running")
+
+	// Cleanup
+	streamer.Stop()
+}
+
+func testAudioOutputStreamerMultipleStops(t *testing.T) {
+	streamer, err := NewAudioOutputStreamer()
+	if err != nil {
+		t.Skipf("Skipping test due to missing dependencies: %v", err)
+		return
+	}
+	require.NotNil(t, streamer)
+
+	// Start first
+	err = streamer.Start()
+	require.NoError(t, err)
+
+	// Multiple stops should be safe
+	streamer.Stop()
+	streamer.Stop()
+	streamer.Stop()
+}
+
+// BenchmarkAudioOutputStreamer benchmarks the AudioOutputStreamer operations
+func BenchmarkAudioOutputStreamer(b *testing.B) {
+	b.Run("GetStats", func(b *testing.B) {
+		streamer, err := NewAudioOutputStreamer()
+		if err != nil {
+			b.Skipf("Skipping benchmark due to missing dependencies: %v", err)
+			return
+		}
+		defer streamer.Stop()
+
+		streamer.Start()
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			streamer.GetStats()
+		}
+	})
+
+	b.Run("UpdateBatchSize", func(b *testing.B) {
+		streamer, err := NewAudioOutputStreamer()
+		if err != nil {
+			b.Skipf("Skipping benchmark due to missing dependencies: %v", err)
+			return
+		}
+		defer streamer.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			streamer.UpdateBatchSize()
+		}
+	})
+
+	b.Run("ReportLatency", func(b *testing.B) {
+		streamer, err := NewAudioOutputStreamer()
+		if err != nil {
+			b.Skipf("Skipping benchmark due to missing dependencies: %v", err)
+			return
+		}
+		defer streamer.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			streamer.ReportLatency(10 * time.Millisecond)
+		}
+	})
+}

internal/audio/performance_critical_test.go

@@ -0,0 +1,393 @@
+//go:build cgo
+// +build cgo
+
+package audio
+
+import (
+	"runtime"
+	"sync"
+	"sync/atomic"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestPerformanceCriticalPaths tests the most frequently executed code paths
+// to ensure they remain efficient and don't interfere with KVM functionality
+func TestPerformanceCriticalPaths(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping performance tests in short mode")
+	}
+
+	// Initialize validation cache for performance testing
+	InitValidationCache()
+
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"AudioFrameProcessingLatency", testAudioFrameProcessingLatency},
+		{"MetricsUpdateOverhead", testMetricsUpdateOverhead},
+		{"ConfigurationAccessSpeed", testConfigurationAccessSpeed},
+		{"ValidationFunctionSpeed", testValidationFunctionSpeed},
+		{"MemoryAllocationPatterns", testMemoryAllocationPatterns},
+		{"ConcurrentAccessPerformance", testConcurrentAccessPerformance},
+		{"BufferPoolEfficiency", testBufferPoolEfficiency},
+		{"AtomicOperationOverhead", testAtomicOperationOverhead},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testAudioFrameProcessingLatency tests the latency of audio frame processing
+// This is the most critical path that must not interfere with KVM
+func testAudioFrameProcessingLatency(t *testing.T) {
+	const (
+		frameCount         = 1000
+		maxLatencyPerFrame = 100 * time.Microsecond // Very strict requirement
+	)
+
+	// Create test frame data
+	frameData := make([]byte, 1920) // Typical frame size
+	for i := range frameData {
+		frameData[i] = byte(i % 256)
+	}
+
+	// Measure frame processing latency
+	start := time.Now()
+	for i := 0; i < frameCount; i++ {
+		// Simulate the critical path: validation + metrics update
+		err := ValidateAudioFrame(frameData)
+		require.NoError(t, err)
+
+		// Record frame received (atomic operation)
+		RecordFrameReceived(len(frameData))
+	}
+	elapsed := time.Since(start)
+
+	avgLatencyPerFrame := elapsed / frameCount
+	t.Logf("Average frame processing latency: %v", avgLatencyPerFrame)
+
+	// Ensure frame processing is fast enough to not interfere with KVM
+	assert.Less(t, avgLatencyPerFrame, maxLatencyPerFrame,
+		"Frame processing latency %v exceeds maximum %v - may interfere with KVM",
+		avgLatencyPerFrame, maxLatencyPerFrame)
+
+	// Ensure total processing time is reasonable
+	maxTotalTime := 50 * time.Millisecond
+	assert.Less(t, elapsed, maxTotalTime,
+		"Total processing time %v exceeds maximum %v", elapsed, maxTotalTime)
+}
+
+// testMetricsUpdateOverhead tests the overhead of metrics updates
+func testMetricsUpdateOverhead(t *testing.T) {
+	const iterations = 10000
+
+	// Test RecordFrameReceived performance
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		RecordFrameReceived(1024)
+	}
+	recordLatency := time.Since(start) / iterations
+
+	// Test GetAudioMetrics performance
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = GetAudioMetrics()
+	}
+	getLatency := time.Since(start) / iterations
+
+	t.Logf("RecordFrameReceived latency: %v", recordLatency)
+	t.Logf("GetAudioMetrics latency: %v", getLatency)
+
+	// Metrics operations should be optimized for JetKVM's ARM Cortex-A7 @ 1GHz
+	// With 256MB RAM, we need to be conservative with performance expectations
+	assert.Less(t, recordLatency, 50*time.Microsecond, "RecordFrameReceived too slow")
+	assert.Less(t, getLatency, 20*time.Microsecond, "GetAudioMetrics too slow")
+}
+
+// testConfigurationAccessSpeed tests configuration access performance
+func testConfigurationAccessSpeed(t *testing.T) {
+	const iterations = 10000
+
+	// Test GetAudioConfig performance
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = GetAudioConfig()
+	}
+	configLatency := time.Since(start) / iterations
+
+	// Test GetConfig performance
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = GetConfig()
+	}
+	constantsLatency := time.Since(start) / iterations
+
+	t.Logf("GetAudioConfig latency: %v", configLatency)
+	t.Logf("GetConfig latency: %v", constantsLatency)
+
+	// Configuration access should be very fast
+	assert.Less(t, configLatency, 100*time.Nanosecond, "GetAudioConfig too slow")
+	assert.Less(t, constantsLatency, 100*time.Nanosecond, "GetConfig too slow")
+}
+
+// testValidationFunctionSpeed tests validation function performance
+func testValidationFunctionSpeed(t *testing.T) {
+	const iterations = 10000
+	frameData := make([]byte, 1920)
+
+	// Test ValidateAudioFrame (most critical)
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		err := ValidateAudioFrame(frameData)
+		require.NoError(t, err)
+	}
+	fastValidationLatency := time.Since(start) / iterations
+
+	// Test ValidateAudioQuality
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		err := ValidateAudioQuality(AudioQualityMedium)
+		require.NoError(t, err)
+	}
+	qualityValidationLatency := time.Since(start) / iterations
+
+	// Test ValidateBufferSize
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		err := ValidateBufferSize(1024)
+		require.NoError(t, err)
+	}
+	bufferValidationLatency := time.Since(start) / iterations
+
+	t.Logf("ValidateAudioFrame latency: %v", fastValidationLatency)
+	t.Logf("ValidateAudioQuality latency: %v", qualityValidationLatency)
+	t.Logf("ValidateBufferSize latency: %v", bufferValidationLatency)
+
+	// Validation functions optimized for ARM Cortex-A7 single core @ 1GHz
+	// Conservative thresholds to ensure KVM functionality isn't impacted
+	assert.Less(t, fastValidationLatency, 100*time.Microsecond, "ValidateAudioFrame too slow")
+	assert.Less(t, qualityValidationLatency, 50*time.Microsecond, "ValidateAudioQuality too slow")
+	assert.Less(t, bufferValidationLatency, 50*time.Microsecond, "ValidateBufferSize too slow")
+}
+
+// testMemoryAllocationPatterns tests memory allocation efficiency
+func testMemoryAllocationPatterns(t *testing.T) {
+	// Test that frequent operations don't cause excessive allocations
+	var m1, m2 runtime.MemStats
+	runtime.GC()
+	runtime.ReadMemStats(&m1)
+
+	// Perform operations that should minimize allocations
+	for i := 0; i < 1000; i++ {
+		_ = GetAudioConfig()
+		_ = GetAudioMetrics()
+		RecordFrameReceived(1024)
+		_ = ValidateAudioQuality(AudioQualityMedium)
+	}
+
+	runtime.GC()
+	runtime.ReadMemStats(&m2)
+
+	allocations := m2.Mallocs - m1.Mallocs
+	t.Logf("Memory allocations for 1000 operations: %d", allocations)
+
+	// Should have minimal allocations for these hot path operations
+	assert.Less(t, allocations, uint64(100), "Too many memory allocations in hot path")
+}
+
+// testConcurrentAccessPerformance tests performance under concurrent access
+func testConcurrentAccessPerformance(t *testing.T) {
+	const (
+		numGoroutines          = 10
+		operationsPerGoroutine = 1000
+	)
+
+	var wg sync.WaitGroup
+	start := time.Now()
+
+	// Launch concurrent goroutines performing audio operations
+	for i := 0; i < numGoroutines; i++ {
+		wg.Add(1)
+		go func() {
+			defer wg.Done()
+			frameData := make([]byte, 1920)
+
+			for j := 0; j < operationsPerGoroutine; j++ {
+				// Simulate concurrent audio processing
+				_ = ValidateAudioFrame(frameData)
+				RecordFrameReceived(len(frameData))
+				_ = GetAudioMetrics()
+				_ = GetAudioConfig()
+			}
+		}()
+	}
+
+	wg.Wait()
+	elapsed := time.Since(start)
+
+	totalOperations := numGoroutines * operationsPerGoroutine * 4 // 4 operations per iteration
+	avgLatency := elapsed / time.Duration(totalOperations)
+
+	t.Logf("Concurrent access: %d operations in %v (avg: %v per operation)",
+		totalOperations, elapsed, avgLatency)
+
+	// Concurrent access should not significantly degrade performance
+	assert.Less(t, avgLatency, 1*time.Microsecond, "Concurrent access too slow")
+}
+
+// testBufferPoolEfficiency tests buffer pool performance
+func testBufferPoolEfficiency(t *testing.T) {
+	// Test buffer acquisition and release performance
+	const iterations = 1000
+
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		// Simulate buffer pool usage (if available)
+		buffer := make([]byte, 1920) // Fallback to allocation
+		_ = buffer
+		// In real implementation, this would be pool.Get() and pool.Put()
+	}
+	elapsed := time.Since(start)
+
+	avgLatency := elapsed / iterations
+	t.Logf("Buffer allocation latency: %v per buffer", avgLatency)
+
+	// Buffer operations should be fast
+	assert.Less(t, avgLatency, 1*time.Microsecond, "Buffer allocation too slow")
+}
+
+// testAtomicOperationOverhead tests atomic operation performance
+func testAtomicOperationOverhead(t *testing.T) {
+	const iterations = 10000
+	var counter int64
+
+	// Test atomic increment performance
+	start := time.Now()
+	for i := 0; i < iterations; i++ {
+		atomic.AddInt64(&counter, 1)
+	}
+	atomicLatency := time.Since(start) / iterations
+
+	// Test atomic load performance
+	start = time.Now()
+	for i := 0; i < iterations; i++ {
+		_ = atomic.LoadInt64(&counter)
+	}
+	loadLatency := time.Since(start) / iterations
+
+	t.Logf("Atomic add latency: %v", atomicLatency)
+	t.Logf("Atomic load latency: %v", loadLatency)
+
+	// Atomic operations on ARM Cortex-A7 - realistic expectations
+	assert.Less(t, atomicLatency, 1*time.Microsecond, "Atomic add too slow")
+	assert.Less(t, loadLatency, 500*time.Nanosecond, "Atomic load too slow")
+}
+
+// TestRegressionDetection tests for performance regressions
+func TestRegressionDetection(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping regression test in short mode")
+	}
+
+	// Baseline performance expectations
+	baselines := map[string]time.Duration{
+		"frame_processing": 100 * time.Microsecond,
+		"metrics_update":   500 * time.Nanosecond,
+		"config_access":    100 * time.Nanosecond,
+		"validation":       200 * time.Nanosecond,
+	}
+
+	// Test frame processing
+	frameData := make([]byte, 1920)
+	start := time.Now()
+	for i := 0; i < 100; i++ {
+		_ = ValidateAudioFrame(frameData)
+		RecordFrameReceived(len(frameData))
+	}
+	frameProcessingTime := time.Since(start) / 100
+
+	// Test metrics update
+	start = time.Now()
+	for i := 0; i < 1000; i++ {
+		RecordFrameReceived(1024)
+	}
+	metricsUpdateTime := time.Since(start) / 1000
+
+	// Test config access
+	start = time.Now()
+	for i := 0; i < 1000; i++ {
+		_ = GetAudioConfig()
+	}
+	configAccessTime := time.Since(start) / 1000
+
+	// Test validation
+	start = time.Now()
+	for i := 0; i < 1000; i++ {
+		_ = ValidateAudioQuality(AudioQualityMedium)
+	}
+	validationTime := time.Since(start) / 1000
+
+	// Performance regression thresholds for JetKVM hardware:
+	// - ARM Cortex-A7 @ 1GHz single core
+	// - 256MB DDR3L RAM
+	// - Must not interfere with primary KVM functionality
+	assert.Less(t, frameProcessingTime, baselines["frame_processing"],
+		"Frame processing regression: %v > %v", frameProcessingTime, baselines["frame_processing"])
+	assert.Less(t, metricsUpdateTime, 100*time.Microsecond,
+		"Metrics update regression: %v > 100μs", metricsUpdateTime)
+	assert.Less(t, configAccessTime, 10*time.Microsecond,
+		"Config access regression: %v > 10μs", configAccessTime)
+	assert.Less(t, validationTime, 10*time.Microsecond,
+		"Validation regression: %v > 10μs", validationTime)
+
+	t.Logf("Performance results:")
+	t.Logf("  Frame processing: %v (baseline: %v)", frameProcessingTime, baselines["frame_processing"])
+	t.Logf("  Metrics update: %v (baseline: %v)", metricsUpdateTime, baselines["metrics_update"])
+	t.Logf("  Config access: %v (baseline: %v)", configAccessTime, baselines["config_access"])
+	t.Logf("  Validation: %v (baseline: %v)", validationTime, baselines["validation"])
+}
+
+// TestMemoryLeakDetection tests for memory leaks in critical paths
+func TestMemoryLeakDetection(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping memory leak test in short mode")
+	}
+
+	var m1, m2 runtime.MemStats
+
+	// Baseline measurement
+	runtime.GC()
+	runtime.ReadMemStats(&m1)
+
+	// Perform many operations that should not leak memory
+	for cycle := 0; cycle < 10; cycle++ {
+		for i := 0; i < 1000; i++ {
+			frameData := make([]byte, 1920)
+			_ = ValidateAudioFrame(frameData)
+			RecordFrameReceived(len(frameData))
+			_ = GetAudioMetrics()
+			_ = GetAudioConfig()
+		}
+		// Force garbage collection between cycles
+		runtime.GC()
+	}
+
+	// Final measurement
+	runtime.GC()
+	runtime.ReadMemStats(&m2)
+
+	memoryGrowth := int64(m2.Alloc) - int64(m1.Alloc)
+	t.Logf("Memory growth after 10,000 operations: %d bytes", memoryGrowth)
+
+	// Memory growth should be minimal (less than 1MB)
+	assert.Less(t, memoryGrowth, int64(1024*1024),
+		"Excessive memory growth detected: %d bytes", memoryGrowth)
+}

internal/audio/priority_scheduler.go

@@ -0,0 +1,168 @@
+//go:build linux
+
+package audio
+
+import (
+	"runtime"
+	"syscall"
+	"unsafe"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// SchedParam represents scheduling parameters for Linux
+type SchedParam struct {
+	Priority int32
+}
+
+// getPriorityConstants returns priority levels from centralized config
+func getPriorityConstants() (audioHigh, audioMedium, audioLow, normal int) {
+	config := GetConfig()
+	return config.AudioHighPriority, config.AudioMediumPriority, config.AudioLowPriority, config.NormalPriority
+}
+
+// getSchedulingPolicies returns scheduling policies from centralized config
+func getSchedulingPolicies() (schedNormal, schedFIFO, schedRR int) {
+	config := GetConfig()
+	return config.SchedNormal, config.SchedFIFO, config.SchedRR
+}
+
+// PriorityScheduler manages thread priorities for audio processing
+type PriorityScheduler struct {
+	logger  zerolog.Logger
+	enabled bool
+}
+
+// NewPriorityScheduler creates a new priority scheduler
+func NewPriorityScheduler() *PriorityScheduler {
+	return &PriorityScheduler{
+		logger:  logging.GetDefaultLogger().With().Str("component", "priority-scheduler").Logger(),
+		enabled: true,
+	}
+}
+
+// SetThreadPriority sets the priority of the current thread
+func (ps *PriorityScheduler) SetThreadPriority(priority int, policy int) error {
+	if !ps.enabled {
+		return nil
+	}
+
+	// Lock to OS thread to ensure we're setting priority for the right thread
+	runtime.LockOSThread()
+
+	// Get current thread ID
+	tid := syscall.Gettid()
+
+	// Set scheduling parameters
+	param := &SchedParam{
+		Priority: int32(priority),
+	}
+
+	// Use syscall to set scheduler
+	_, _, errno := syscall.Syscall(syscall.SYS_SCHED_SETSCHEDULER,
+		uintptr(tid),
+		uintptr(policy),
+		uintptr(unsafe.Pointer(param)))
+
+	if errno != 0 {
+		// If we can't set real-time priority, try nice value instead
+		schedNormal, _, _ := getSchedulingPolicies()
+		if policy != schedNormal {
+			ps.logger.Warn().Int("errno", int(errno)).Msg("failed to set real-time priority, falling back to nice")
+			return ps.setNicePriority(priority)
+		}
+		return errno
+	}
+
+	ps.logger.Debug().Int("tid", tid).Int("priority", priority).Int("policy", policy).Msg("thread priority set")
+	return nil
+}
+
+// setNicePriority sets nice value as fallback when real-time scheduling is not available
+func (ps *PriorityScheduler) setNicePriority(rtPriority int) error {
+	// Convert real-time priority to nice value (inverse relationship)
+	// RT priority 80 -> nice -10, RT priority 40 -> nice 0
+	niceValue := (40 - rtPriority) / 4
+	if niceValue < GetConfig().MinNiceValue {
+		niceValue = GetConfig().MinNiceValue
+	}
+	if niceValue > GetConfig().MaxNiceValue {
+		niceValue = GetConfig().MaxNiceValue
+	}
+
+	err := syscall.Setpriority(syscall.PRIO_PROCESS, 0, niceValue)
+	if err != nil {
+		ps.logger.Warn().Err(err).Int("nice", niceValue).Msg("failed to set nice priority")
+		return err
+	}
+
+	ps.logger.Debug().Int("nice", niceValue).Msg("nice priority set as fallback")
+	return nil
+}
+
+// SetAudioProcessingPriority sets high priority for audio processing threads
+func (ps *PriorityScheduler) SetAudioProcessingPriority() error {
+	audioHigh, _, _, _ := getPriorityConstants()
+	_, schedFIFO, _ := getSchedulingPolicies()
+	return ps.SetThreadPriority(audioHigh, schedFIFO)
+}
+
+// SetAudioIOPriority sets medium priority for audio I/O threads
+func (ps *PriorityScheduler) SetAudioIOPriority() error {
+	_, audioMedium, _, _ := getPriorityConstants()
+	_, schedFIFO, _ := getSchedulingPolicies()
+	return ps.SetThreadPriority(audioMedium, schedFIFO)
+}
+
+// SetAudioBackgroundPriority sets low priority for background audio tasks
+func (ps *PriorityScheduler) SetAudioBackgroundPriority() error {
+	_, _, audioLow, _ := getPriorityConstants()
+	_, schedFIFO, _ := getSchedulingPolicies()
+	return ps.SetThreadPriority(audioLow, schedFIFO)
+}
+
+// ResetPriority resets thread to normal scheduling
+func (ps *PriorityScheduler) ResetPriority() error {
+	_, _, _, normal := getPriorityConstants()
+	schedNormal, _, _ := getSchedulingPolicies()
+	return ps.SetThreadPriority(normal, schedNormal)
+}
+
+// Disable disables priority scheduling (useful for testing or fallback)
+func (ps *PriorityScheduler) Disable() {
+	ps.enabled = false
+	ps.logger.Debug().Msg("priority scheduling disabled")
+}
+
+// Enable enables priority scheduling
+func (ps *PriorityScheduler) Enable() {
+	ps.enabled = true
+	ps.logger.Debug().Msg("priority scheduling enabled")
+}
+
+// Global priority scheduler instance
+var globalPriorityScheduler *PriorityScheduler
+
+// GetPriorityScheduler returns the global priority scheduler instance
+func GetPriorityScheduler() *PriorityScheduler {
+	if globalPriorityScheduler == nil {
+		globalPriorityScheduler = NewPriorityScheduler()
+	}
+	return globalPriorityScheduler
+}
+
+// SetAudioThreadPriority is a convenience function to set audio processing priority
+func SetAudioThreadPriority() error {
+	return GetPriorityScheduler().SetAudioProcessingPriority()
+}
+
+// SetAudioIOThreadPriority is a convenience function to set audio I/O priority
+func SetAudioIOThreadPriority() error {
+	return GetPriorityScheduler().SetAudioIOPriority()
+}
+
+// ResetThreadPriority is a convenience function to reset thread priority
+func ResetThreadPriority() error {
+	return GetPriorityScheduler().ResetPriority()
+}

internal/audio/regression_test.go

@@ -0,0 +1,362 @@
+//go:build cgo
+// +build cgo
+
+package audio
+
+import (
+	"fmt"
+	"net"
+	"os"
+	"sync"
+	"sync/atomic"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestRegressionScenarios tests critical edge cases and error conditions
+// that could cause system instability in production
+func TestRegressionScenarios(t *testing.T) {
+	tests := []struct {
+		name        string
+		testFunc    func(t *testing.T)
+		description string
+	}{
+		{
+			name:        "IPCConnectionFailure",
+			testFunc:    testIPCConnectionFailureRecovery,
+			description: "Test IPC connection failure and recovery scenarios",
+		},
+		{
+			name:        "BufferOverflow",
+			testFunc:    testBufferOverflowHandling,
+			description: "Test buffer overflow protection and recovery",
+		},
+		{
+			name:        "SupervisorRapidRestart",
+			testFunc:    testSupervisorRapidRestartScenario,
+			description: "Test supervisor behavior under rapid restart conditions",
+		},
+		{
+			name:        "ConcurrentStartStop",
+			testFunc:    testConcurrentStartStopOperations,
+			description: "Test concurrent start/stop operations for race conditions",
+		},
+		{
+			name:        "MemoryLeakPrevention",
+			testFunc:    testMemoryLeakPrevention,
+			description: "Test memory leak prevention in long-running scenarios",
+		},
+		{
+			name:        "ConfigValidationEdgeCases",
+			testFunc:    testConfigValidationEdgeCases,
+			description: "Test configuration validation with edge case values",
+		},
+		{
+			name:        "AtomicOperationConsistency",
+			testFunc:    testAtomicOperationConsistency,
+			description: "Test atomic operations consistency under high concurrency",
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			t.Logf("Running regression test: %s - %s", tt.name, tt.description)
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testIPCConnectionFailureRecovery tests IPC connection failure scenarios
+func testIPCConnectionFailureRecovery(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Test start with no IPC server available (should handle gracefully)
+	err := manager.Start()
+	// Should not panic or crash, may return error depending on implementation
+	if err != nil {
+		t.Logf("Expected error when no IPC server available: %v", err)
+	}
+
+	// Test that manager can recover after IPC becomes available
+	if manager.IsRunning() {
+		manager.Stop()
+	}
+
+	// Verify clean state after failure
+	assert.False(t, manager.IsRunning())
+	assert.False(t, manager.IsReady())
+}
+
+// testBufferOverflowHandling tests buffer overflow protection
+func testBufferOverflowHandling(t *testing.T) {
+	// Test with extremely large buffer sizes
+	extremelyLargeSize := 1024 * 1024 * 100 // 100MB
+	err := ValidateBufferSize(extremelyLargeSize)
+	assert.Error(t, err, "Should reject extremely large buffer sizes")
+
+	// Test with negative buffer sizes
+	err = ValidateBufferSize(-1)
+	assert.Error(t, err, "Should reject negative buffer sizes")
+
+	// Test with zero buffer size
+	err = ValidateBufferSize(0)
+	assert.Error(t, err, "Should reject zero buffer size")
+
+	// Test with maximum valid buffer size
+	maxValidSize := GetConfig().SocketMaxBuffer
+	err = ValidateBufferSize(int(maxValidSize))
+	assert.NoError(t, err, "Should accept maximum valid buffer size")
+}
+
+// testSupervisorRapidRestartScenario tests supervisor under rapid restart conditions
+func testSupervisorRapidRestartScenario(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping rapid restart test in short mode")
+	}
+
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Perform rapid start/stop cycles to test for race conditions
+	for i := 0; i < 10; i++ {
+		err := supervisor.Start()
+		if err != nil {
+			t.Logf("Start attempt %d failed (expected in test environment): %v", i, err)
+		}
+
+		// Very short delay to stress test
+		time.Sleep(10 * time.Millisecond)
+
+		supervisor.Stop()
+		time.Sleep(10 * time.Millisecond)
+	}
+
+	// Verify supervisor is in clean state after rapid cycling
+	assert.False(t, supervisor.IsRunning())
+}
+
+// testConcurrentStartStopOperations tests concurrent operations for race conditions
+func testConcurrentStartStopOperations(t *testing.T) {
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	var wg sync.WaitGroup
+	const numGoroutines = 10
+
+	// Launch multiple goroutines trying to start/stop concurrently
+	for i := 0; i < numGoroutines; i++ {
+		wg.Add(2)
+
+		// Start goroutine
+		go func(id int) {
+			defer wg.Done()
+			err := manager.Start()
+			if err != nil {
+				t.Logf("Concurrent start %d: %v", id, err)
+			}
+		}(i)
+
+		// Stop goroutine
+		go func(id int) {
+			defer wg.Done()
+			time.Sleep(5 * time.Millisecond) // Small delay
+			manager.Stop()
+		}(i)
+	}
+
+	wg.Wait()
+
+	// Ensure final state is consistent
+	manager.Stop() // Final cleanup
+	assert.False(t, manager.IsRunning())
+}
+
+// testMemoryLeakPrevention tests for memory leaks in long-running scenarios
+func testMemoryLeakPrevention(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping memory leak test in short mode")
+	}
+
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Simulate long-running operation with periodic restarts
+	for cycle := 0; cycle < 5; cycle++ {
+		err := manager.Start()
+		if err != nil {
+			t.Logf("Start cycle %d failed (expected): %v", cycle, err)
+		}
+
+		// Simulate some activity
+		time.Sleep(100 * time.Millisecond)
+
+		// Get metrics to ensure they're not accumulating indefinitely
+		metrics := manager.GetMetrics()
+		assert.NotNil(t, metrics, "Metrics should be available")
+
+		manager.Stop()
+		time.Sleep(50 * time.Millisecond)
+	}
+
+	// Final verification
+	assert.False(t, manager.IsRunning())
+}
+
+// testConfigValidationEdgeCases tests configuration validation with edge cases
+func testConfigValidationEdgeCases(t *testing.T) {
+	// Test sample rate edge cases
+	testCases := []struct {
+		sampleRate  int
+		channels    int
+		frameSize   int
+		shouldPass  bool
+		description string
+	}{
+		{0, 2, 960, false, "zero sample rate"},
+		{-1, 2, 960, false, "negative sample rate"},
+		{1, 2, 960, false, "extremely low sample rate"},
+		{999999, 2, 960, false, "extremely high sample rate"},
+		{48000, 0, 960, false, "zero channels"},
+		{48000, -1, 960, false, "negative channels"},
+		{48000, 100, 960, false, "too many channels"},
+		{48000, 2, 0, false, "zero frame size"},
+		{48000, 2, -1, false, "negative frame size"},
+		{48000, 2, 999999, true, "extremely large frame size"},
+		{48000, 2, 960, true, "valid configuration"},
+		{44100, 1, 441, true, "valid mono configuration"},
+	}
+
+	for _, tc := range testCases {
+		t.Run(tc.description, func(t *testing.T) {
+			err := ValidateInputIPCConfig(tc.sampleRate, tc.channels, tc.frameSize)
+			if tc.shouldPass {
+				assert.NoError(t, err, "Should accept valid config: %s", tc.description)
+			} else {
+				assert.Error(t, err, "Should reject invalid config: %s", tc.description)
+			}
+		})
+	}
+}
+
+// testAtomicOperationConsistency tests atomic operations under high concurrency
+func testAtomicOperationConsistency(t *testing.T) {
+	var counter int64
+	var wg sync.WaitGroup
+	const numGoroutines = 100
+	const incrementsPerGoroutine = 1000
+
+	// Launch multiple goroutines performing atomic operations
+	for i := 0; i < numGoroutines; i++ {
+		wg.Add(1)
+		go func() {
+			defer wg.Done()
+			for j := 0; j < incrementsPerGoroutine; j++ {
+				atomic.AddInt64(&counter, 1)
+			}
+		}()
+	}
+
+	wg.Wait()
+
+	// Verify final count is correct
+	expected := int64(numGoroutines * incrementsPerGoroutine)
+	actual := atomic.LoadInt64(&counter)
+	assert.Equal(t, expected, actual, "Atomic operations should be consistent")
+}
+
+// TestErrorRecoveryScenarios tests various error recovery scenarios
+func TestErrorRecoveryScenarios(t *testing.T) {
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"NetworkConnectionLoss", testNetworkConnectionLossRecovery},
+		{"ProcessCrashRecovery", testProcessCrashRecovery},
+		{"ResourceExhaustionRecovery", testResourceExhaustionRecovery},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testNetworkConnectionLossRecovery tests recovery from network connection loss
+func testNetworkConnectionLossRecovery(t *testing.T) {
+	// Create a temporary socket that we can close to simulate connection loss
+	tempDir := t.TempDir()
+	socketPath := fmt.Sprintf("%s/test_recovery.sock", tempDir)
+
+	// Create and immediately close a socket to test connection failure
+	listener, err := net.Listen("unix", socketPath)
+	if err != nil {
+		t.Skipf("Cannot create test socket: %v", err)
+	}
+	listener.Close() // Close immediately to simulate connection loss
+
+	// Remove socket file to ensure connection will fail
+	os.Remove(socketPath)
+
+	// Test that components handle connection loss gracefully
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// This should handle the connection failure gracefully
+	err = manager.Start()
+	if err != nil {
+		t.Logf("Expected connection failure handled: %v", err)
+	}
+
+	// Cleanup
+	manager.Stop()
+}
+
+// testProcessCrashRecovery tests recovery from process crashes
+func testProcessCrashRecovery(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping process crash test in short mode")
+	}
+
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Start supervisor (will likely fail in test environment, but should handle gracefully)
+	err := supervisor.Start()
+	if err != nil {
+		t.Logf("Supervisor start failed as expected in test environment: %v", err)
+	}
+
+	// Verify supervisor can be stopped cleanly even after start failure
+	supervisor.Stop()
+	assert.False(t, supervisor.IsRunning())
+}
+
+// testResourceExhaustionRecovery tests recovery from resource exhaustion
+func testResourceExhaustionRecovery(t *testing.T) {
+	// Test with resource constraints
+	manager := NewAudioInputIPCManager()
+	require.NotNil(t, manager)
+
+	// Simulate resource exhaustion by rapid start/stop cycles
+	for i := 0; i < 20; i++ {
+		err := manager.Start()
+		if err != nil {
+			t.Logf("Resource exhaustion cycle %d: %v", i, err)
+		}
+		manager.Stop()
+		// No delay to stress test resource management
+	}
+
+	// Verify system can still function after resource stress
+	err := manager.Start()
+	if err != nil {
+		t.Logf("Final start after resource stress: %v", err)
+	}
+	manager.Stop()
+	assert.False(t, manager.IsRunning())
+}

internal/audio/supervisor.go

@@ -11,8 +11,6 @@ import (
 	"sync/atomic"
 	"syscall"
 	"time"
-
-	"github.com/rs/zerolog"
 )
 
 // Restart configuration is now retrieved from centralized config
@@ -150,6 +148,13 @@ func (s *AudioOutputSupervisor) Stop() {
 	s.logger.Info().Str("component", AudioOutputSupervisorComponent).Msg("component stopped")
 }
 
+// GetProcessMetrics returns current process metrics with audio-output-server name
+func (s *AudioOutputSupervisor) GetProcessMetrics() *ProcessMetrics {
+	metrics := s.BaseSupervisor.GetProcessMetrics()
+	metrics.ProcessName = "audio-output-server"
+	return metrics
+}
+
 // supervisionLoop is the main supervision loop
 func (s *AudioOutputSupervisor) supervisionLoop() {
 	defer func() {
@@ -175,25 +180,16 @@ func (s *AudioOutputSupervisor) supervisionLoop() {
 		default:
 			// Start or restart the process
 			if err := s.startProcess(); err != nil {
-				// Only log start errors if error level enabled to reduce overhead
+				s.logger.Error().Err(err).Msg("failed to start audio server process")
-				if s.logger.GetLevel() <= zerolog.ErrorLevel {
-					s.logger.Error().Err(err).Msg("failed to start audio server process")
-				}
 
 				// Check if we should attempt restart
 				if !s.shouldRestart() {
-					// Only log critical errors to reduce overhead
+					s.logger.Error().Msg("maximum restart attempts exceeded, stopping supervisor")
-					if s.logger.GetLevel() <= zerolog.ErrorLevel {
-						s.logger.Error().Msg("maximum restart attempts exceeded, stopping supervisor")
-					}
 					return
 				}
 
 				delay := s.calculateRestartDelay()
-				// Sample logging to reduce overhead - log every 5th restart attempt
+				s.logger.Warn().Dur("delay", delay).Msg("retrying process start after delay")
-				if len(s.restartAttempts)%5 == 0 && s.logger.GetLevel() <= zerolog.WarnLevel {
-					s.logger.Warn().Dur("delay", delay).Int("attempt", len(s.restartAttempts)).Msg("retrying process start after delay")
-				}
 
 				if s.onRestart != nil {
 					s.onRestart(len(s.restartAttempts), delay)

internal/audio/supervisor_test.go

@@ -0,0 +1,393 @@
+//go:build integration && cgo
+// +build integration,cgo
+
+package audio
+
+import (
+	"context"
+	"os"
+	"os/exec"
+	"sync"
+	"syscall"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestSupervisorRestart tests various supervisor restart scenarios
+func TestSupervisorRestart(t *testing.T) {
+	tests := []struct {
+		name        string
+		testFunc    func(t *testing.T)
+		description string
+	}{
+		{
+			name:        "BasicRestart",
+			testFunc:    testBasicSupervisorRestart,
+			description: "Test basic supervisor restart functionality",
+		},
+		{
+			name:        "ProcessCrashRestart",
+			testFunc:    testProcessCrashRestart,
+			description: "Test supervisor restart after process crash",
+		},
+		{
+			name:        "MaxRestartAttempts",
+			testFunc:    testMaxRestartAttempts,
+			description: "Test supervisor respects max restart attempts",
+		},
+		{
+			name:        "ExponentialBackoff",
+			testFunc:    testExponentialBackoff,
+			description: "Test supervisor exponential backoff behavior",
+		},
+		{
+			name:        "HealthMonitoring",
+			testFunc:    testHealthMonitoring,
+			description: "Test supervisor health monitoring",
+		},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			t.Logf("Running supervisor test: %s - %s", tt.name, tt.description)
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testBasicSupervisorRestart tests basic restart functionality
+func testBasicSupervisorRestart(t *testing.T) {
+	ctx, cancel := context.WithTimeout(context.Background(), 60*time.Second)
+	defer cancel()
+
+	// Create a mock supervisor with a simple test command
+	supervisor := &AudioInputSupervisor{
+		logger:          getTestLogger(),
+		maxRestarts:     3,
+		restartDelay:    100 * time.Millisecond,
+		healthCheckInterval: 200 * time.Millisecond,
+	}
+
+	// Use a simple command that will exit quickly for testing
+	testCmd := exec.CommandContext(ctx, "sleep", "0.5")
+	supervisor.cmd = testCmd
+
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	// Start supervisor
+	go func() {
+		defer wg.Done()
+		supervisor.Start(ctx)
+	}()
+
+	// Wait for initial process to start and exit
+	time.Sleep(1 * time.Second)
+
+	// Verify that supervisor attempted restart
+	assert.True(t, supervisor.GetRestartCount() > 0, "Supervisor should have attempted restart")
+
+	// Stop supervisor
+	cancel()
+	wg.Wait()
+}
+
+// testProcessCrashRestart tests restart after process crash
+func testProcessCrashRestart(t *testing.T) {
+	ctx, cancel := context.WithTimeout(context.Background(), 45*time.Second)
+	defer cancel()
+
+	supervisor := &AudioInputSupervisor{
+		logger:          getTestLogger(),
+		maxRestarts:     2,
+		restartDelay:    200 * time.Millisecond,
+		healthCheckInterval: 100 * time.Millisecond,
+	}
+
+	// Create a command that will crash (exit with non-zero code)
+	testCmd := exec.CommandContext(ctx, "sh", "-c", "sleep 0.2 && exit 1")
+	supervisor.cmd = testCmd
+
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	go func() {
+		defer wg.Done()
+		supervisor.Start(ctx)
+	}()
+
+	// Wait for process to crash and restart attempts
+	time.Sleep(2 * time.Second)
+
+	// Verify restart attempts were made
+	restartCount := supervisor.GetRestartCount()
+	assert.True(t, restartCount > 0, "Supervisor should have attempted restart after crash")
+	assert.True(t, restartCount <= 2, "Supervisor should not exceed max restart attempts")
+
+	cancel()
+	wg.Wait()
+}
+
+// testMaxRestartAttempts tests that supervisor respects max restart limit
+func testMaxRestartAttempts(t *testing.T) {
+	ctx, cancel := context.WithTimeout(context.Background(), 30*time.Second)
+	defer cancel()
+
+	maxRestarts := 3
+	supervisor := &AudioInputSupervisor{
+		logger:          getTestLogger(),
+		maxRestarts:     maxRestarts,
+		restartDelay:    50 * time.Millisecond,
+		healthCheckInterval: 50 * time.Millisecond,
+	}
+
+	// Command that immediately fails
+	testCmd := exec.CommandContext(ctx, "false") // 'false' command always exits with code 1
+	supervisor.cmd = testCmd
+
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	go func() {
+		defer wg.Done()
+		supervisor.Start(ctx)
+	}()
+
+	// Wait for all restart attempts to complete
+	time.Sleep(2 * time.Second)
+
+	// Verify that supervisor stopped after max attempts
+	restartCount := supervisor.GetRestartCount()
+	assert.Equal(t, maxRestarts, restartCount, "Supervisor should stop after max restart attempts")
+	assert.False(t, supervisor.IsRunning(), "Supervisor should not be running after max attempts")
+
+	cancel()
+	wg.Wait()
+}
+
+// testExponentialBackoff tests the exponential backoff behavior
+func testExponentialBackoff(t *testing.T) {
+	ctx, cancel := context.WithTimeout(context.Background(), 45*time.Second)
+	defer cancel()
+
+	supervisor := &AudioInputSupervisor{
+		logger:          getTestLogger(),
+		maxRestarts:     3,
+		restartDelay:    100 * time.Millisecond, // Base delay
+		healthCheckInterval: 50 * time.Millisecond,
+	}
+
+	// Command that fails immediately
+	testCmd := exec.CommandContext(ctx, "false")
+	supervisor.cmd = testCmd
+
+	var restartTimes []time.Time
+	var mu sync.Mutex
+
+	// Hook into restart events to measure timing
+	originalRestart := supervisor.restart
+	supervisor.restart = func() {
+		mu.Lock()
+		restartTimes = append(restartTimes, time.Now())
+		mu.Unlock()
+		if originalRestart != nil {
+			originalRestart()
+		}
+	}
+
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	go func() {
+		defer wg.Done()
+		supervisor.Start(ctx)
+	}()
+
+	// Wait for restart attempts
+	time.Sleep(3 * time.Second)
+
+	mu.Lock()
+	defer mu.Unlock()
+
+	// Verify exponential backoff (each delay should be longer than the previous)
+	if len(restartTimes) >= 2 {
+		for i := 1; i < len(restartTimes); i++ {
+			delay := restartTimes[i].Sub(restartTimes[i-1])
+			expectedMinDelay := time.Duration(i) * 100 * time.Millisecond
+			assert.True(t, delay >= expectedMinDelay,
+				"Restart delay should increase exponentially: attempt %d delay %v should be >= %v",
+				i, delay, expectedMinDelay)
+		}
+	}
+
+	cancel()
+	wg.Wait()
+}
+
+// testHealthMonitoring tests the health monitoring functionality
+func testHealthMonitoring(t *testing.T) {
+	ctx, cancel := context.WithTimeout(context.Background(), 30*time.Second)
+	defer cancel()
+
+	supervisor := &AudioInputSupervisor{
+		logger:          getTestLogger(),
+		maxRestarts:     2,
+		restartDelay:    100 * time.Millisecond,
+		healthCheckInterval: 50 * time.Millisecond,
+	}
+
+	// Command that runs for a while then exits
+	testCmd := exec.CommandContext(ctx, "sleep", "1")
+	supervisor.cmd = testCmd
+
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	go func() {
+		defer wg.Done()
+		supervisor.Start(ctx)
+	}()
+
+	// Initially should be running
+	time.Sleep(200 * time.Millisecond)
+	assert.True(t, supervisor.IsRunning(), "Supervisor should be running initially")
+
+	// Wait for process to exit and health check to detect it
+	time.Sleep(1.5 * time.Second)
+
+	// Should have detected process exit and attempted restart
+	assert.True(t, supervisor.GetRestartCount() > 0, "Health monitoring should detect process exit")
+
+	cancel()
+	wg.Wait()
+}
+
+// TestAudioInputSupervisorIntegration tests the actual AudioInputSupervisor
+func TestAudioInputSupervisorIntegration(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping integration test in short mode")
+	}
+
+	ctx, cancel := context.WithTimeout(context.Background(), 60*time.Second)
+	defer cancel()
+
+	// Create a real supervisor instance
+	supervisor := NewAudioInputSupervisor()
+	require.NotNil(t, supervisor, "Supervisor should be created")
+
+	// Test that supervisor can be started and stopped cleanly
+	var wg sync.WaitGroup
+	wg.Add(1)
+
+	go func() {
+		defer wg.Done()
+		// This will likely fail due to missing audio hardware in test environment,
+		// but we're testing the supervisor logic, not the audio functionality
+		supervisor.Start(ctx)
+	}()
+
+	// Let it run briefly
+	time.Sleep(500 * time.Millisecond)
+
+	// Stop the supervisor
+	cancel()
+	wg.Wait()
+
+	// Verify clean shutdown
+	assert.False(t, supervisor.IsRunning(), "Supervisor should not be running after context cancellation")
+}
+
+// Mock supervisor for testing (simplified version)
+type AudioInputSupervisor struct {
+	logger              zerolog.Logger
+	cmd                 *exec.Cmd
+	maxRestarts         int
+	restartDelay        time.Duration
+	healthCheckInterval time.Duration
+	restartCount        int
+	running             bool
+	mu                  sync.RWMutex
+	restart             func() // Hook for testing
+}
+
+func (s *AudioInputSupervisor) Start(ctx context.Context) error {
+	s.mu.Lock()
+	s.running = true
+	s.mu.Unlock()
+
+	for s.restartCount < s.maxRestarts {
+		select {
+		case <-ctx.Done():
+			s.mu.Lock()
+			s.running = false
+			s.mu.Unlock()
+			return ctx.Err()
+		default:
+		}
+
+		// Start process
+		if s.cmd != nil {
+			err := s.cmd.Start()
+			if err != nil {
+				s.logger.Error().Err(err).Msg("Failed to start process")
+				s.restartCount++
+				time.Sleep(s.getBackoffDelay())
+				continue
+			}
+
+			// Wait for process to exit
+			err = s.cmd.Wait()
+			if err != nil {
+				s.logger.Error().Err(err).Msg("Process exited with error")
+			}
+		}
+
+		s.restartCount++
+		if s.restart != nil {
+			s.restart()
+		}
+
+		if s.restartCount < s.maxRestarts {
+			time.Sleep(s.getBackoffDelay())
+		}
+	}
+
+	s.mu.Lock()
+	s.running = false
+	s.mu.Unlock()
+	return nil
+}
+
+func (s *AudioInputSupervisor) IsRunning() bool {
+	s.mu.RLock()
+	defer s.mu.RUnlock()
+	return s.running
+}
+
+func (s *AudioInputSupervisor) GetRestartCount() int {
+	s.mu.RLock()
+	defer s.mu.RUnlock()
+	return s.restartCount
+}
+
+func (s *AudioInputSupervisor) getBackoffDelay() time.Duration {
+	// Simple exponential backoff
+	multiplier := 1 << uint(s.restartCount)
+	if multiplier > 8 {
+		multiplier = 8 // Cap the multiplier
+	}
+	return s.restartDelay * time.Duration(multiplier)
+}
+
+// NewAudioInputSupervisor creates a new supervisor for testing
+func NewAudioInputSupervisor() *AudioInputSupervisor {
+	return &AudioInputSupervisor{
+		logger:              getTestLogger(),
+		maxRestarts:         getMaxRestartAttempts(),
+		restartDelay:        getInitialRestartDelay(),
+		healthCheckInterval: 1 * time.Second,
+	}
+}

internal/audio/supervisor_unit_test.go

@@ -0,0 +1,217 @@
+package audio
+
+import (
+	"sync"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+func TestNewAudioOutputSupervisor(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	assert.NotNil(t, supervisor)
+	assert.False(t, supervisor.IsRunning())
+}
+
+func TestAudioOutputSupervisorStart(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Test successful start
+	err := supervisor.Start()
+	assert.NoError(t, err)
+	assert.True(t, supervisor.IsRunning())
+
+	// Test starting already running supervisor
+	err = supervisor.Start()
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "already running")
+
+	// Cleanup
+	supervisor.Stop()
+}
+
+func TestAudioOutputSupervisorStop(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Test stopping non-running supervisor
+	supervisor.Stop()
+	assert.False(t, supervisor.IsRunning())
+
+	// Start and then stop
+	err := supervisor.Start()
+	require.NoError(t, err)
+	assert.True(t, supervisor.IsRunning())
+
+	supervisor.Stop()
+	assert.False(t, supervisor.IsRunning())
+}
+
+func TestAudioOutputSupervisorIsRunning(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Test initial state
+	assert.False(t, supervisor.IsRunning())
+
+	// Test after start
+	err := supervisor.Start()
+	require.NoError(t, err)
+	assert.True(t, supervisor.IsRunning())
+
+	// Test after stop
+	supervisor.Stop()
+	assert.False(t, supervisor.IsRunning())
+}
+
+func TestAudioOutputSupervisorGetProcessMetrics(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Test metrics when not running
+	metrics := supervisor.GetProcessMetrics()
+	assert.NotNil(t, metrics)
+
+	// Start and test metrics
+	err := supervisor.Start()
+	require.NoError(t, err)
+
+	metrics = supervisor.GetProcessMetrics()
+	assert.NotNil(t, metrics)
+
+	// Cleanup
+	supervisor.Stop()
+}
+
+func TestAudioOutputSupervisorConcurrentOperations(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	var wg sync.WaitGroup
+
+	// Test concurrent start/stop operations
+	for i := 0; i < 10; i++ {
+		wg.Add(2)
+		go func() {
+			defer wg.Done()
+			_ = supervisor.Start()
+		}()
+		go func() {
+			defer wg.Done()
+			supervisor.Stop()
+		}()
+	}
+
+	// Test concurrent metric access
+	for i := 0; i < 5; i++ {
+		wg.Add(1)
+		go func() {
+			defer wg.Done()
+			_ = supervisor.GetProcessMetrics()
+		}()
+	}
+
+	// Test concurrent status checks
+	for i := 0; i < 5; i++ {
+		wg.Add(1)
+		go func() {
+			defer wg.Done()
+			_ = supervisor.IsRunning()
+		}()
+	}
+
+	wg.Wait()
+
+	// Cleanup
+	supervisor.Stop()
+}
+
+func TestAudioOutputSupervisorMultipleStartStop(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Test multiple start/stop cycles
+	for i := 0; i < 5; i++ {
+		err := supervisor.Start()
+		assert.NoError(t, err)
+		assert.True(t, supervisor.IsRunning())
+
+		supervisor.Stop()
+		assert.False(t, supervisor.IsRunning())
+	}
+}
+
+func TestAudioOutputSupervisorHealthCheck(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Start supervisor
+	err := supervisor.Start()
+	require.NoError(t, err)
+
+	// Give some time for health monitoring to initialize
+	time.Sleep(100 * time.Millisecond)
+
+	// Test that supervisor is still running
+	assert.True(t, supervisor.IsRunning())
+
+	// Cleanup
+	supervisor.Stop()
+}
+
+func TestAudioOutputSupervisorProcessManagement(t *testing.T) {
+	supervisor := NewAudioOutputSupervisor()
+	require.NotNil(t, supervisor)
+
+	// Start supervisor
+	err := supervisor.Start()
+	require.NoError(t, err)
+
+	// Give some time for process management to initialize
+	time.Sleep(200 * time.Millisecond)
+
+	// Test that supervisor is managing processes
+	assert.True(t, supervisor.IsRunning())
+
+	// Cleanup
+	supervisor.Stop()
+
+	// Ensure supervisor stopped cleanly
+	assert.False(t, supervisor.IsRunning())
+}
+
+// Benchmark tests
+func BenchmarkAudioOutputSupervisor(b *testing.B) {
+	supervisor := NewAudioOutputSupervisor()
+
+	b.Run("Start", func(b *testing.B) {
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = supervisor.Start()
+			supervisor.Stop()
+		}
+	})
+
+	b.Run("GetProcessMetrics", func(b *testing.B) {
+		_ = supervisor.Start()
+		defer supervisor.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = supervisor.GetProcessMetrics()
+		}
+	})
+
+	b.Run("IsRunning", func(b *testing.B) {
+		_ = supervisor.Start()
+		defer supervisor.Stop()
+
+		b.ResetTimer()
+		for i := 0; i < b.N; i++ {
+			_ = supervisor.IsRunning()
+		}
+	})
+}

internal/audio/test_utils.go

@@ -0,0 +1,319 @@
+//go:build integration
+// +build integration
+
+package audio
+
+import (
+	"context"
+	"net"
+	"os"
+	"sync"
+	"time"
+
+	"github.com/jetkvm/kvm/internal/logging"
+	"github.com/rs/zerolog"
+)
+
+// Test utilities and mock implementations for integration tests
+
+// MockAudioIPCServer provides a mock IPC server for testing
+type AudioIPCServer struct {
+	socketPath string
+	logger     zerolog.Logger
+	listener   net.Listener
+	connections map[net.Conn]bool
+	mu         sync.RWMutex
+	running    bool
+}
+
+// Start starts the mock IPC server
+func (s *AudioIPCServer) Start(ctx context.Context) error {
+	// Remove existing socket file
+	os.Remove(s.socketPath)
+
+	listener, err := net.Listen("unix", s.socketPath)
+	if err != nil {
+		return err
+	}
+	s.listener = listener
+	s.connections = make(map[net.Conn]bool)
+
+	s.mu.Lock()
+	s.running = true
+	s.mu.Unlock()
+
+	go s.acceptConnections(ctx)
+
+	<-ctx.Done()
+	s.Stop()
+	return ctx.Err()
+}
+
+// Stop stops the mock IPC server
+func (s *AudioIPCServer) Stop() {
+	s.mu.Lock()
+	defer s.mu.Unlock()
+
+	if !s.running {
+		return
+	}
+
+	s.running = false
+
+	if s.listener != nil {
+		s.listener.Close()
+	}
+
+	// Close all connections
+	for conn := range s.connections {
+		conn.Close()
+	}
+
+	// Clean up socket file
+	os.Remove(s.socketPath)
+}
+
+// acceptConnections handles incoming connections
+func (s *AudioIPCServer) acceptConnections(ctx context.Context) {
+	for {
+		select {
+		case <-ctx.Done():
+			return
+		default:
+		}
+
+		conn, err := s.listener.Accept()
+		if err != nil {
+			select {
+			case <-ctx.Done():
+				return
+			default:
+				s.logger.Error().Err(err).Msg("Failed to accept connection")
+				continue
+			}
+		}
+
+		s.mu.Lock()
+		s.connections[conn] = true
+		s.mu.Unlock()
+
+		go s.handleConnection(ctx, conn)
+	}
+}
+
+// handleConnection handles a single connection
+func (s *AudioIPCServer) handleConnection(ctx context.Context, conn net.Conn) {
+	defer func() {
+		s.mu.Lock()
+		delete(s.connections, conn)
+		s.mu.Unlock()
+		conn.Close()
+	}()
+
+	buffer := make([]byte, 4096)
+	for {
+		select {
+		case <-ctx.Done():
+			return
+		default:
+		}
+
+		// Set read timeout
+		conn.SetReadDeadline(time.Now().Add(100 * time.Millisecond))
+		n, err := conn.Read(buffer)
+		if err != nil {
+			if netErr, ok := err.(net.Error); ok && netErr.Timeout() {
+				continue
+			}
+			return
+		}
+
+		// Process received data (for testing, we just log it)
+		s.logger.Debug().Int("bytes", n).Msg("Received data from client")
+	}
+}
+
+// AudioInputIPCServer provides a mock input IPC server
+type AudioInputIPCServer struct {
+	*AudioIPCServer
+}
+
+// Test message structures
+type OutputMessage struct {
+	Type      OutputMessageType
+	Timestamp int64
+	Data      []byte
+}
+
+type InputMessage struct {
+	Type      InputMessageType
+	Timestamp int64
+	Data      []byte
+}
+
+// Test configuration helpers
+func getTestConfig() *AudioConfigConstants {
+	return &AudioConfigConstants{
+		// Basic audio settings
+		SampleRate: 48000,
+		Channels:   2,
+		MaxAudioFrameSize: 4096,
+
+		// IPC settings
+		OutputMagicNumber: 0x4A4B4F55, // "JKOU"
+		InputMagicNumber:  0x4A4B4D49, // "JKMI"
+		WriteTimeout:      5 * time.Second,
+		HeaderSize:        17,
+		MaxFrameSize:      4096,
+		MessagePoolSize:   100,
+
+		// Supervisor settings
+		MaxRestartAttempts:    3,
+		InitialRestartDelay:   1 * time.Second,
+		MaxRestartDelay:       30 * time.Second,
+		HealthCheckInterval:   5 * time.Second,
+
+		// Quality presets
+		AudioQualityLowOutputBitrate:    32000,
+		AudioQualityMediumOutputBitrate: 96000,
+		AudioQualityHighOutputBitrate:   192000,
+		AudioQualityUltraOutputBitrate:  320000,
+
+		AudioQualityLowInputBitrate:    16000,
+		AudioQualityMediumInputBitrate: 64000,
+		AudioQualityHighInputBitrate:   128000,
+		AudioQualityUltraInputBitrate:  256000,
+
+		AudioQualityLowSampleRate:    24000,
+		AudioQualityMediumSampleRate: 48000,
+		AudioQualityHighSampleRate:   48000,
+		AudioQualityUltraSampleRate:  48000,
+
+		AudioQualityLowChannels:    1,
+		AudioQualityMediumChannels: 2,
+		AudioQualityHighChannels:   2,
+		AudioQualityUltraChannels:  2,
+
+		AudioQualityLowFrameSize:    20 * time.Millisecond,
+		AudioQualityMediumFrameSize: 20 * time.Millisecond,
+		AudioQualityHighFrameSize:   20 * time.Millisecond,
+		AudioQualityUltraFrameSize:  20 * time.Millisecond,
+
+		AudioQualityMicLowSampleRate: 16000,
+
+		// Metrics settings
+		MetricsUpdateInterval: 1 * time.Second,
+
+		// Latency settings
+		DefaultTargetLatencyMS:             50,
+		DefaultOptimizationIntervalSeconds: 5,
+		DefaultAdaptiveThreshold:           0.8,
+		DefaultStatsIntervalSeconds:        5,
+
+		// Buffer settings
+		DefaultBufferPoolSize:     100,
+		DefaultControlPoolSize:    50,
+		DefaultFramePoolSize:      200,
+		DefaultMaxPooledFrames:    500,
+		DefaultPoolCleanupInterval: 30 * time.Second,
+
+		// Process monitoring
+		MaxCPUPercent:        100.0,
+		MinCPUPercent:        0.0,
+		DefaultClockTicks:    100,
+		DefaultMemoryGB:      4.0,
+		MaxWarmupSamples:     10,
+		WarmupCPUSamples:     5,
+		MetricsChannelBuffer: 100,
+		MinValidClockTicks:   50,
+		MaxValidClockTicks:   1000,
+		PageSize:             4096,
+
+		// CGO settings (for cgo builds)
+		CGOOpusBitrate:       96000,
+		CGOOpusComplexity:    3,
+		CGOOpusVBR:           1,
+		CGOOpusVBRConstraint: 1,
+		CGOOpusSignalType:    3,
+		CGOOpusBandwidth:     1105,
+		CGOOpusDTX:           0,
+		CGOSampleRate:        48000,
+
+		// Batch processing
+		BatchProcessorFramesPerBatch: 10,
+		BatchProcessorTimeout:        100 * time.Millisecond,
+
+		// Granular metrics
+		GranularMetricsMaxSamples:    1000,
+		GranularMetricsLogInterval:   30 * time.Second,
+		GranularMetricsCleanupInterval: 5 * time.Minute,
+	}
+}
+
+// setupTestEnvironment sets up the test environment
+func setupTestEnvironment() {
+	// Use test configuration
+	UpdateConfig(getTestConfig())
+
+	// Initialize logging for tests
+	logging.SetLevel("debug")
+}
+
+// cleanupTestEnvironment cleans up after tests
+func cleanupTestEnvironment() {
+	// Reset to default configuration
+	UpdateConfig(DefaultAudioConfig())
+}
+
+// createTestLogger creates a logger for testing
+func createTestLogger(name string) zerolog.Logger {
+	return zerolog.New(os.Stdout).With().
+		Timestamp().
+		Str("component", name).
+		Str("test", "true").
+		Logger()
+}
+
+// waitForCondition waits for a condition to be true with timeout
+func waitForCondition(condition func() bool, timeout time.Duration, checkInterval time.Duration) bool {
+	timeout_timer := time.NewTimer(timeout)
+	defer timeout_timer.Stop()
+
+	ticker := time.NewTicker(checkInterval)
+	defer ticker.Stop()
+
+	for {
+		select {
+		case <-timeout_timer.C:
+			return false
+		case <-ticker.C:
+			if condition() {
+				return true
+			}
+		}
+	}
+}
+
+// TestHelper provides common test functionality
+type TestHelper struct {
+	tempDir string
+	logger  zerolog.Logger
+}
+
+// NewTestHelper creates a new test helper
+func NewTestHelper(tempDir string) *TestHelper {
+	return &TestHelper{
+		tempDir: tempDir,
+		logger:  createTestLogger("test-helper"),
+	}
+}
+
+// CreateTempSocket creates a temporary socket path
+func (h *TestHelper) CreateTempSocket(name string) string {
+	return filepath.Join(h.tempDir, name)
+}
+
+// GetLogger returns the test logger
+func (h *TestHelper) GetLogger() zerolog.Logger {
+	return h.logger
+}

internal/audio/validation.go

@@ -41,7 +41,6 @@ func ValidateAudioQuality(quality AudioQuality) error {
 }
 
 // ValidateZeroCopyFrame validates zero-copy audio frame
-// Optimized to use cached max frame size
 func ValidateZeroCopyFrame(frame *ZeroCopyAudioFrame) error {
 	if frame == nil {
 		return ErrInvalidFrameData
@@ -50,22 +49,8 @@ func ValidateZeroCopyFrame(frame *ZeroCopyAudioFrame) error {
 	if len(data) == 0 {
 		return ErrInvalidFrameData
 	}
-
+	// Use config value
-	// Fast path: use cached max frame size
+	maxFrameSize := GetConfig().MaxAudioFrameSize
-	maxFrameSize := cachedMaxFrameSize
-	if maxFrameSize == 0 {
-		// Fallback: get from cache
-		cache := GetCachedConfig()
-		maxFrameSize = int(cache.maxAudioFrameSize.Load())
-		if maxFrameSize == 0 {
-			// Last resort: update cache
-			cache.Update()
-			maxFrameSize = int(cache.maxAudioFrameSize.Load())
-		}
-		// Cache globally for next calls
-		cachedMaxFrameSize = maxFrameSize
-	}
-
 	if len(data) > maxFrameSize {
 		return ErrInvalidFrameSize
 	}
@@ -110,31 +95,10 @@ func ValidateThreadPriority(priority int) error {
 }
 
 // ValidateLatency validates latency duration values with reasonable bounds
-// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateLatency(latency time.Duration) error {
 	if latency < 0 {
 		return fmt.Errorf("%w: latency %v cannot be negative", ErrInvalidLatency, latency)
 	}
-
-	// Fast path: check against cached max latency
-	cache := GetCachedConfig()
-	maxLatency := time.Duration(cache.maxLatency.Load())
-
-	// If we have a valid cached value, use it
-	if maxLatency > 0 {
-		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 > maxLatency {
-			return fmt.Errorf("%w: latency %v exceeds maximum %v",
-				ErrInvalidLatency, latency, maxLatency)
-		}
-		return nil
-	}
-
-	// Slower path: full validation with GetConfig()
 	config := GetConfig()
 	minLatency := time.Millisecond // Minimum reasonable latency
 	if latency > 0 && latency < minLatency {
@@ -149,30 +113,11 @@ func ValidateLatency(latency time.Duration) error {
 }
 
 // ValidateMetricsInterval validates metrics update interval
-// Optimized to use AudioConfigCache for frequently accessed values
 func ValidateMetricsInterval(interval time.Duration) error {
-	// Fast path: check against cached values
+	// Use config values
-	cache := GetCachedConfig()
-	minInterval := time.Duration(cache.minMetricsUpdateInterval.Load())
-	maxInterval := time.Duration(cache.maxMetricsUpdateInterval.Load())
-
-	// If we have valid cached values, use them
-	if minInterval > 0 && maxInterval > 0 {
-		if interval < minInterval {
-			return fmt.Errorf("%w: interval %v below minimum %v",
-				ErrInvalidMetricsInterval, interval, minInterval)
-		}
-		if interval > maxInterval {
-			return fmt.Errorf("%w: interval %v exceeds maximum %v",
-				ErrInvalidMetricsInterval, interval, maxInterval)
-		}
-		return nil
-	}
-
-	// Slower path: full validation with GetConfig()
 	config := GetConfig()
-	minInterval = config.MinMetricsUpdateInterval
+	minInterval := config.MinMetricsUpdateInterval
-	maxInterval = config.MaxMetricsUpdateInterval
+	maxInterval := config.MaxMetricsUpdateInterval
 	if interval < minInterval {
 		return ErrInvalidMetricsInterval
 	}
@@ -309,18 +254,12 @@ func ValidateChannelCount(channels int) error {
 		return nil
 	}
 
-	// Fast path: Check against cached max channels
+	// Check against max channels - still using cache to avoid GetConfig()
-	cachedMaxChannels := int(cache.maxChannels.Load())
+	// Note: We don't have maxChannels in the cache yet, so we'll use GetConfig() for now
-	if cachedMaxChannels > 0 && channels <= cachedMaxChannels {
+	config := GetConfig()
-		return nil
+	if channels > config.MaxChannels {
-	}
-
-	// 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, updatedMaxChannels)
+			ErrInvalidChannels, channels, config.MaxChannels)
 	}
 	return nil
 }
@@ -392,34 +331,15 @@ func ValidateFrameDuration(duration time.Duration) error {
 		}
 	}
 
-	// Fast path: Check against cached min/max frame duration
+	// Slower path: full validation against min/max
-	cachedMinDuration := time.Duration(cache.minFrameDuration.Load())
+	config := GetConfig()
-	cachedMaxDuration := time.Duration(cache.maxFrameDuration.Load())
+	if duration < config.MinFrameDuration {
-
-	if cachedMinDuration > 0 && cachedMaxDuration > 0 {
-		if duration < cachedMinDuration {
-			return fmt.Errorf("%w: frame duration %v below minimum %v",
-				ErrInvalidFrameDuration, duration, cachedMinDuration)
-		}
-		if duration > cachedMaxDuration {
-			return fmt.Errorf("%w: frame duration %v exceeds maximum %v",
-				ErrInvalidFrameDuration, duration, cachedMaxDuration)
-		}
-		return nil
-	}
-
-	// Slow path: Update cache and validate
-	cache.Update()
-	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",
-			ErrInvalidFrameDuration, duration, updatedMinDuration)
+			ErrInvalidFrameDuration, duration, config.MinFrameDuration)
 	}
-	if duration > updatedMaxDuration {
+	if duration > config.MaxFrameDuration {
 		return fmt.Errorf("%w: frame duration %v exceeds maximum %v",
-			ErrInvalidFrameDuration, duration, updatedMaxDuration)
+			ErrInvalidFrameDuration, duration, config.MaxFrameDuration)
 	}
 	return nil
 }

internal/audio/validation_test.go

@@ -0,0 +1,541 @@
+//go:build cgo
+// +build cgo
+
+package audio
+
+import (
+	"fmt"
+	"testing"
+	"time"
+
+	"github.com/stretchr/testify/assert"
+	"github.com/stretchr/testify/require"
+)
+
+// TestValidationFunctions provides comprehensive testing of all validation functions
+// to ensure they catch breaking changes and regressions effectively
+func TestValidationFunctions(t *testing.T) {
+	// Initialize validation cache for testing
+	InitValidationCache()
+
+	tests := []struct {
+		name     string
+		testFunc func(t *testing.T)
+	}{
+		{"AudioQualityValidation", testAudioQualityValidation},
+		{"FrameDataValidation", testFrameDataValidation},
+		{"BufferSizeValidation", testBufferSizeValidation},
+		{"ThreadPriorityValidation", testThreadPriorityValidation},
+		{"LatencyValidation", testLatencyValidation},
+		{"MetricsIntervalValidation", testMetricsIntervalValidation},
+		{"SampleRateValidation", testSampleRateValidation},
+		{"ChannelCountValidation", testChannelCountValidation},
+		{"BitrateValidation", testBitrateValidation},
+		{"FrameDurationValidation", testFrameDurationValidation},
+		{"IPCConfigValidation", testIPCConfigValidation},
+		{"AdaptiveBufferConfigValidation", testAdaptiveBufferConfigValidation},
+		{"AudioConfigCompleteValidation", testAudioConfigCompleteValidation},
+		{"ZeroCopyFrameValidation", testZeroCopyFrameValidation},
+		{"AudioFrameFastValidation", testAudioFrameFastValidation},
+		{"ErrorWrappingValidation", testErrorWrappingValidation},
+	}
+
+	for _, tt := range tests {
+		t.Run(tt.name, func(t *testing.T) {
+			tt.testFunc(t)
+		})
+	}
+}
+
+// testAudioQualityValidation tests audio quality validation with boundary conditions
+func testAudioQualityValidation(t *testing.T) {
+	// Test valid quality levels
+	validQualities := []AudioQuality{AudioQualityLow, AudioQualityMedium, AudioQualityHigh, AudioQualityUltra}
+	for _, quality := range validQualities {
+		err := ValidateAudioQuality(quality)
+		assert.NoError(t, err, "Valid quality %d should pass validation", quality)
+	}
+
+	// Test invalid quality levels
+	invalidQualities := []AudioQuality{-1, 4, 100, -100}
+	for _, quality := range invalidQualities {
+		err := ValidateAudioQuality(quality)
+		assert.Error(t, err, "Invalid quality %d should fail validation", quality)
+		assert.Contains(t, err.Error(), "invalid audio quality level", "Error should mention audio quality")
+	}
+}
+
+// testFrameDataValidation tests frame data validation with various edge cases using modern validation
+func testFrameDataValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test empty data
+	err := ValidateAudioFrame([]byte{})
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "frame data is empty")
+
+	// Test data above maximum size
+	largeData := make([]byte, config.MaxAudioFrameSize+1)
+	err = ValidateAudioFrame(largeData)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid data
+	validData := make([]byte, 1000) // Within bounds
+	if len(validData) <= config.MaxAudioFrameSize {
+		err = ValidateAudioFrame(validData)
+		assert.NoError(t, err)
+	}
+}
+
+// testBufferSizeValidation tests buffer size validation
+func testBufferSizeValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test negative and zero sizes
+	invalidSizes := []int{-1, -100, 0}
+	for _, size := range invalidSizes {
+		err := ValidateBufferSize(size)
+		assert.Error(t, err, "Buffer size %d should be invalid", size)
+		assert.Contains(t, err.Error(), "must be positive")
+	}
+
+	// Test size exceeding maximum
+	err := ValidateBufferSize(config.SocketMaxBuffer + 1)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid sizes
+	validSizes := []int{1, 1024, 4096, config.SocketMaxBuffer}
+	for _, size := range validSizes {
+		err := ValidateBufferSize(size)
+		assert.NoError(t, err, "Buffer size %d should be valid", size)
+	}
+}
+
+// testThreadPriorityValidation tests thread priority validation
+func testThreadPriorityValidation(t *testing.T) {
+	// Test valid priorities
+	validPriorities := []int{-20, -10, 0, 10, 19}
+	for _, priority := range validPriorities {
+		err := ValidateThreadPriority(priority)
+		assert.NoError(t, err, "Priority %d should be valid", priority)
+	}
+
+	// Test invalid priorities
+	invalidPriorities := []int{-21, -100, 20, 100}
+	for _, priority := range invalidPriorities {
+		err := ValidateThreadPriority(priority)
+		assert.Error(t, err, "Priority %d should be invalid", priority)
+		assert.Contains(t, err.Error(), "outside valid range")
+	}
+}
+
+// testLatencyValidation tests latency validation
+func testLatencyValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test negative latency
+	err := ValidateLatency(-1 * time.Millisecond)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "cannot be negative")
+
+	// Test zero latency (should be valid)
+	err = ValidateLatency(0)
+	assert.NoError(t, err)
+
+	// Test very small positive latency
+	err = ValidateLatency(500 * time.Microsecond)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "below minimum")
+
+	// Test latency exceeding maximum
+	err = ValidateLatency(config.MaxLatency + time.Second)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid latencies
+	validLatencies := []time.Duration{
+		1 * time.Millisecond,
+		10 * time.Millisecond,
+		100 * time.Millisecond,
+		config.MaxLatency,
+	}
+	for _, latency := range validLatencies {
+		err := ValidateLatency(latency)
+		assert.NoError(t, err, "Latency %v should be valid", latency)
+	}
+}
+
+// testMetricsIntervalValidation tests metrics interval validation
+func testMetricsIntervalValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test interval below minimum
+	err := ValidateMetricsInterval(config.MinMetricsUpdateInterval - time.Millisecond)
+	assert.Error(t, err)
+
+	// Test interval above maximum
+	err = ValidateMetricsInterval(config.MaxMetricsUpdateInterval + time.Second)
+	assert.Error(t, err)
+
+	// Test valid intervals
+	validIntervals := []time.Duration{
+		config.MinMetricsUpdateInterval,
+		config.MaxMetricsUpdateInterval,
+		(config.MinMetricsUpdateInterval + config.MaxMetricsUpdateInterval) / 2,
+	}
+	for _, interval := range validIntervals {
+		err := ValidateMetricsInterval(interval)
+		assert.NoError(t, err, "Interval %v should be valid", interval)
+	}
+}
+
+// testSampleRateValidation tests sample rate validation
+func testSampleRateValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test negative and zero sample rates
+	invalidRates := []int{-1, -48000, 0}
+	for _, rate := range invalidRates {
+		err := ValidateSampleRate(rate)
+		assert.Error(t, err, "Sample rate %d should be invalid", rate)
+		assert.Contains(t, err.Error(), "must be positive")
+	}
+
+	// Test unsupported sample rates
+	unsupportedRates := []int{1000, 12345, 96001}
+	for _, rate := range unsupportedRates {
+		err := ValidateSampleRate(rate)
+		assert.Error(t, err, "Sample rate %d should be unsupported", rate)
+		assert.Contains(t, err.Error(), "not in supported rates")
+	}
+
+	// Test valid sample rates
+	for _, rate := range config.ValidSampleRates {
+		err := ValidateSampleRate(rate)
+		assert.NoError(t, err, "Sample rate %d should be valid", rate)
+	}
+}
+
+// testChannelCountValidation tests channel count validation
+func testChannelCountValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test invalid channel counts
+	invalidCounts := []int{-1, -10, 0}
+	for _, count := range invalidCounts {
+		err := ValidateChannelCount(count)
+		assert.Error(t, err, "Channel count %d should be invalid", count)
+		assert.Contains(t, err.Error(), "must be positive")
+	}
+
+	// Test channel count exceeding maximum
+	err := ValidateChannelCount(config.MaxChannels + 1)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid channel counts
+	validCounts := []int{1, 2, config.MaxChannels}
+	for _, count := range validCounts {
+		err := ValidateChannelCount(count)
+		assert.NoError(t, err, "Channel count %d should be valid", count)
+	}
+}
+
+// testBitrateValidation tests bitrate validation
+func testBitrateValidation(t *testing.T) {
+	// Test invalid bitrates
+	invalidBitrates := []int{-1, -1000, 0}
+	for _, bitrate := range invalidBitrates {
+		err := ValidateBitrate(bitrate)
+		assert.Error(t, err, "Bitrate %d should be invalid", bitrate)
+		assert.Contains(t, err.Error(), "must be positive")
+	}
+
+	// Test bitrate below minimum (in kbps)
+	err := ValidateBitrate(5) // 5 kbps = 5000 bps < 6000 bps minimum
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "below minimum")
+
+	// Test bitrate above maximum (in kbps)
+	err = ValidateBitrate(511) // 511 kbps = 511000 bps > 510000 bps maximum
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid bitrates (in kbps)
+	validBitrates := []int{
+		6,   // 6 kbps = 6000 bps (minimum)
+		64,  // Medium quality preset
+		128, // High quality preset
+		192, // Ultra quality preset
+		510, // 510 kbps = 510000 bps (maximum)
+	}
+	for _, bitrate := range validBitrates {
+		err := ValidateBitrate(bitrate)
+		assert.NoError(t, err, "Bitrate %d kbps should be valid", bitrate)
+	}
+}
+
+// testFrameDurationValidation tests frame duration validation
+func testFrameDurationValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test invalid durations
+	invalidDurations := []time.Duration{-1 * time.Millisecond, -1 * time.Second, 0}
+	for _, duration := range invalidDurations {
+		err := ValidateFrameDuration(duration)
+		assert.Error(t, err, "Duration %v should be invalid", duration)
+		assert.Contains(t, err.Error(), "must be positive")
+	}
+
+	// Test duration below minimum
+	err := ValidateFrameDuration(config.MinFrameDuration - time.Microsecond)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "below minimum")
+
+	// Test duration above maximum
+	err = ValidateFrameDuration(config.MaxFrameDuration + time.Second)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid durations
+	validDurations := []time.Duration{
+		config.MinFrameDuration,
+		config.MaxFrameDuration,
+		20 * time.Millisecond, // Common frame duration
+	}
+	for _, duration := range validDurations {
+		err := ValidateFrameDuration(duration)
+		assert.NoError(t, err, "Duration %v should be valid", duration)
+	}
+}
+
+// testIPCConfigValidation tests IPC configuration validation
+func testIPCConfigValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test invalid configurations for input IPC
+	invalidConfigs := []struct {
+		sampleRate, channels, frameSize int
+		description                     string
+	}{
+		{0, 2, 960, "zero sample rate"},
+		{48000, 0, 960, "zero channels"},
+		{48000, 2, 0, "zero frame size"},
+		{config.MinSampleRate - 1, 2, 960, "sample rate below minimum"},
+		{config.MaxSampleRate + 1, 2, 960, "sample rate above maximum"},
+		{48000, config.MaxChannels + 1, 960, "too many channels"},
+		{48000, -1, 960, "negative channels"},
+		{48000, 2, -1, "negative frame size"},
+	}
+
+	for _, tc := range invalidConfigs {
+		// Test input IPC validation
+		err := ValidateInputIPCConfig(tc.sampleRate, tc.channels, tc.frameSize)
+		assert.Error(t, err, "Input IPC config should be invalid: %s", tc.description)
+
+		// Test output IPC validation
+		err = ValidateOutputIPCConfig(tc.sampleRate, tc.channels, tc.frameSize)
+		assert.Error(t, err, "Output IPC config should be invalid: %s", tc.description)
+	}
+
+	// Test valid configuration
+	err := ValidateInputIPCConfig(48000, 2, 960)
+	assert.NoError(t, err)
+	err = ValidateOutputIPCConfig(48000, 2, 960)
+	assert.NoError(t, err)
+}
+
+// testAdaptiveBufferConfigValidation tests adaptive buffer configuration validation
+func testAdaptiveBufferConfigValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test invalid configurations
+	invalidConfigs := []struct {
+		minSize, maxSize, defaultSize int
+		description                   string
+	}{
+		{0, 1024, 512, "zero min size"},
+		{-1, 1024, 512, "negative min size"},
+		{512, 0, 256, "zero max size"},
+		{512, -1, 256, "negative max size"},
+		{512, 1024, 0, "zero default size"},
+		{512, 1024, -1, "negative default size"},
+		{1024, 512, 768, "min >= max"},
+		{512, 1024, 256, "default < min"},
+		{512, 1024, 2048, "default > max"},
+		{512, config.SocketMaxBuffer + 1, 1024, "max exceeds global limit"},
+	}
+
+	for _, tc := range invalidConfigs {
+		err := ValidateAdaptiveBufferConfig(tc.minSize, tc.maxSize, tc.defaultSize)
+		assert.Error(t, err, "Config should be invalid: %s", tc.description)
+	}
+
+	// Test valid configuration
+	err := ValidateAdaptiveBufferConfig(512, 4096, 1024)
+	assert.NoError(t, err)
+}
+
+// testAudioConfigCompleteValidation tests complete audio configuration validation
+func testAudioConfigCompleteValidation(t *testing.T) {
+	// Test valid configuration using actual preset values
+	validConfig := AudioConfig{
+		Quality:    AudioQualityMedium,
+		Bitrate:    64, // kbps - matches medium quality preset
+		SampleRate: 48000,
+		Channels:   2,
+		FrameSize:  20 * time.Millisecond,
+	}
+	err := ValidateAudioConfigComplete(validConfig)
+	assert.NoError(t, err)
+
+	// Test invalid quality
+	invalidQualityConfig := validConfig
+	invalidQualityConfig.Quality = AudioQuality(99)
+	err = ValidateAudioConfigComplete(invalidQualityConfig)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "quality validation failed")
+
+	// Test invalid bitrate
+	invalidBitrateConfig := validConfig
+	invalidBitrateConfig.Bitrate = -1
+	err = ValidateAudioConfigComplete(invalidBitrateConfig)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "bitrate validation failed")
+
+	// Test invalid sample rate
+	invalidSampleRateConfig := validConfig
+	invalidSampleRateConfig.SampleRate = 12345
+	err = ValidateAudioConfigComplete(invalidSampleRateConfig)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "sample rate validation failed")
+
+	// Test invalid channels
+	invalidChannelsConfig := validConfig
+	invalidChannelsConfig.Channels = 0
+	err = ValidateAudioConfigComplete(invalidChannelsConfig)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "channel count validation failed")
+
+	// Test invalid frame duration
+	invalidFrameDurationConfig := validConfig
+	invalidFrameDurationConfig.FrameSize = -1 * time.Millisecond
+	err = ValidateAudioConfigComplete(invalidFrameDurationConfig)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "frame duration validation failed")
+}
+
+// testZeroCopyFrameValidation tests zero-copy frame validation
+func testZeroCopyFrameValidation(t *testing.T) {
+	// Test nil frame
+	err := ValidateZeroCopyFrame(nil)
+	assert.Error(t, err)
+
+	// Note: We can't easily test ZeroCopyAudioFrame without creating actual instances
+	// This would require more complex setup, but the validation logic is tested
+}
+
+// testAudioFrameFastValidation tests fast audio frame validation
+func testAudioFrameFastValidation(t *testing.T) {
+	config := GetConfig()
+
+	// Test empty data
+	err := ValidateAudioFrame([]byte{})
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "frame data is empty")
+
+	// Test data exceeding maximum size
+	largeData := make([]byte, config.MaxAudioFrameSize+1)
+	err = ValidateAudioFrame(largeData)
+	assert.Error(t, err)
+	assert.Contains(t, err.Error(), "exceeds maximum")
+
+	// Test valid data
+	validData := make([]byte, 1000)
+	err = ValidateAudioFrame(validData)
+	assert.NoError(t, err)
+}
+
+// testErrorWrappingValidation tests error wrapping functionality
+func testErrorWrappingValidation(t *testing.T) {
+	// Test wrapping nil error
+	wrapped := WrapWithMetadata(nil, "component", "operation", map[string]interface{}{"key": "value"})
+	assert.Nil(t, wrapped)
+
+	// Test wrapping actual error
+	originalErr := assert.AnError
+	metadata := map[string]interface{}{
+		"frame_size": 1024,
+		"quality":    "high",
+	}
+	wrapped = WrapWithMetadata(originalErr, "audio", "decode", metadata)
+	require.NotNil(t, wrapped)
+	assert.Contains(t, wrapped.Error(), "audio.decode")
+	assert.Contains(t, wrapped.Error(), "assert.AnError")
+	assert.Contains(t, wrapped.Error(), "metadata")
+	assert.Contains(t, wrapped.Error(), "frame_size")
+	assert.Contains(t, wrapped.Error(), "quality")
+}
+
+// TestValidationIntegration tests validation functions working together
+func TestValidationIntegration(t *testing.T) {
+	// Test that validation functions work correctly with actual audio configurations
+	presets := GetAudioQualityPresets()
+	require.NotEmpty(t, presets)
+
+	for quality, config := range presets {
+		t.Run(fmt.Sprintf("Quality_%d", quality), func(t *testing.T) {
+			// Validate the preset configuration
+			err := ValidateAudioConfigComplete(config)
+			assert.NoError(t, err, "Preset configuration for quality %d should be valid", quality)
+
+			// Validate individual components
+			err = ValidateAudioQuality(config.Quality)
+			assert.NoError(t, err, "Quality should be valid")
+
+			err = ValidateBitrate(config.Bitrate)
+			assert.NoError(t, err, "Bitrate should be valid")
+
+			err = ValidateSampleRate(config.SampleRate)
+			assert.NoError(t, err, "Sample rate should be valid")
+
+			err = ValidateChannelCount(config.Channels)
+			assert.NoError(t, err, "Channel count should be valid")
+
+			err = ValidateFrameDuration(config.FrameSize)
+			assert.NoError(t, err, "Frame duration should be valid")
+		})
+	}
+}
+
+// TestValidationPerformance ensures validation functions are efficient
+func TestValidationPerformance(t *testing.T) {
+	if testing.Short() {
+		t.Skip("Skipping performance test in short mode")
+	}
+
+	// Initialize validation cache for performance testing
+	InitValidationCache()
+
+	// Test that validation functions complete quickly
+	start := time.Now()
+	iterations := 10000
+
+	for i := 0; i < iterations; i++ {
+		_ = ValidateAudioQuality(AudioQualityMedium)
+		_ = ValidateBufferSize(1024)
+		_ = ValidateChannelCount(2)
+		_ = ValidateSampleRate(48000)
+		_ = ValidateBitrate(96) // 96 kbps
+	}
+
+	elapsed := time.Since(start)
+	perIteration := elapsed / time.Duration(iterations)
+
+	// Performance expectations for JetKVM (ARM Cortex-A7 @ 1GHz, 256MB RAM)
+	// Audio processing must not interfere with primary KVM functionality
+	assert.Less(t, perIteration, 200*time.Microsecond, "Validation should not impact KVM performance")
+	t.Logf("Validation performance: %v per iteration", perIteration)
+}

internal/audio/zero_copy.go

@@ -3,6 +3,7 @@ package audio
 import (
 	"sync"
 	"sync/atomic"
+	"time"
 	"unsafe"
 )
 
@@ -141,23 +142,32 @@ func NewZeroCopyFramePool(maxFrameSize int) *ZeroCopyFramePool {
 
 // Get retrieves a zero-copy frame from the pool
 func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
+	start := time.Now()
+	var wasHit bool
+	defer func() {
+		latency := time.Since(start)
+		GetGranularMetricsCollector().RecordZeroCopyGet(latency, wasHit)
+	}()
+
 	// Memory guard: Track allocation count to prevent excessive memory usage
 	allocationCount := atomic.LoadInt64(&p.allocationCount)
 	if allocationCount > int64(p.maxPoolSize*2) {
 		// If we've allocated too many frames, force pool reuse
+		atomic.AddInt64(&p.missCount, 1)
+		wasHit = true // Pool reuse counts as hit
 		frame := p.pool.Get().(*ZeroCopyAudioFrame)
 		frame.mutex.Lock()
 		frame.refCount = 1
 		frame.length = 0
 		frame.data = frame.data[:0]
 		frame.mutex.Unlock()
-
 		return frame
 	}
 
 	// First try pre-allocated frames for fastest access
 	p.mutex.Lock()
 	if len(p.preallocated) > 0 {
+		wasHit = true
 		frame := p.preallocated[len(p.preallocated)-1]
 		p.preallocated = p.preallocated[:len(p.preallocated)-1]
 		p.mutex.Unlock()
@@ -168,11 +178,13 @@ func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
 		frame.data = frame.data[:0]
 		frame.mutex.Unlock()
 
+		atomic.AddInt64(&p.hitCount, 1)
 		return frame
 	}
 	p.mutex.Unlock()
 
 	// Try sync.Pool next and track allocation
+	atomic.AddInt64(&p.allocationCount, 1)
 	frame := p.pool.Get().(*ZeroCopyAudioFrame)
 	frame.mutex.Lock()
 	frame.refCount = 1
@@ -180,13 +192,18 @@ func (p *ZeroCopyFramePool) Get() *ZeroCopyAudioFrame {
 	frame.data = frame.data[:0]
 	frame.mutex.Unlock()
 
+	wasHit = true // Pool hit
 	atomic.AddInt64(&p.hitCount, 1)
-
 	return frame
 }
 
 // Put returns a zero-copy frame to the pool
 func (p *ZeroCopyFramePool) Put(frame *ZeroCopyAudioFrame) {
+	start := time.Now()
+	defer func() {
+		latency := time.Since(start)
+		GetGranularMetricsCollector().RecordZeroCopyPut(latency, frame.capacity)
+	}()
 	if frame == nil || !frame.pooled {
 		return
 	}
@@ -219,15 +236,10 @@ func (p *ZeroCopyFramePool) Put(frame *ZeroCopyAudioFrame) {
 
 		// Return to sync.Pool
 		p.pool.Put(frame)
-		// Metrics collection removed
+		atomic.AddInt64(&p.counter, 1)
-		if false {
-			atomic.AddInt64(&p.counter, 1)
-		}
 	} else {
 		frame.mutex.Unlock()
 	}
-
-	// Metrics recording removed - granular metrics collector was unused
 }
 
 // Data returns the frame data as a slice (zero-copy view)

main.go

@@ -68,17 +68,6 @@ func startAudioSubprocess() error {
 		config.AudioQualityLowOpusDTX,
 	)
 
-	// Pre-warm audio input subprocess to reduce activation latency (if enabled)
-	if config.EnableSubprocessPrewarming {
-		if err := audio.PrewarmAudioInputSubprocess(); err != nil {
-			logger.Warn().Err(err).Msg("failed to pre-warm audio input subprocess")
-		} else {
-			logger.Info().Msg("audio input subprocess pre-warmed successfully")
-		}
-	} else {
-		logger.Info().Msg("audio input subprocess pre-warming disabled by configuration")
-	}
-
 	// Note: Audio input supervisor is NOT started here - it will be started on-demand
 	// when the user activates microphone input through the UI
 

ui/src/components/ActionBar.tsx

@@ -360,7 +360,7 @@ export default function Actionbar({
                 checkIfStateChanged(open);
                 return (
                   <div className="mx-auto">
-                    <AudioControlPopover microphone={microphone} />
+                    <AudioControlPopover microphone={microphone} open={open} />
                   </div>
                 );
               }}

ui/src/components/AudioConfigDisplay.tsx

@@ -0,0 +1,38 @@
+import { cx } from "@/cva.config";
+
+interface AudioConfig {
+  Quality: number;
+  Bitrate: number;
+  SampleRate: number;
+  Channels: number;
+  FrameSize: string;
+}
+
+interface AudioConfigDisplayProps {
+  config: AudioConfig;
+  variant?: 'default' | 'success' | 'info';
+  className?: string;
+}
+
+const variantStyles = {
+  default: "bg-slate-50 text-slate-600 dark:bg-slate-700 dark:text-slate-400",
+  success: "bg-green-50 text-green-600 dark:bg-green-900/20 dark:text-green-400",
+  info: "bg-blue-50 text-blue-600 dark:bg-blue-900/20 dark:text-blue-400"
+};
+
+export function AudioConfigDisplay({ config, variant = 'default', className }: AudioConfigDisplayProps) {
+  return (
+    <div className={cx(
+      "rounded-md p-2 text-xs",
+      variantStyles[variant],
+      className
+    )}>
+      <div className="grid grid-cols-2 gap-1">
+        <span>Sample Rate: {config.SampleRate}Hz</span>
+        <span>Channels: {config.Channels}</span>
+        <span>Bitrate: {config.Bitrate}kbps</span>
+        <span>Frame: {config.FrameSize}</span>
+      </div>
+    </div>
+  );
+}

ui/src/components/AudioLevelMeter.tsx

@@ -0,0 +1,77 @@
+import React from 'react';
+import clsx from 'clsx';
+
+interface AudioLevelMeterProps {
+  level: number; // 0-100 percentage
+  isActive: boolean;
+  className?: string;
+  size?: 'sm' | 'md' | 'lg';
+  showLabel?: boolean;
+}
+
+export const AudioLevelMeter: React.FC<AudioLevelMeterProps> = ({
+  level,
+  isActive,
+  className,
+  size = 'md',
+  showLabel = true
+}) => {
+  const sizeClasses = {
+    sm: 'h-1',
+    md: 'h-2',
+    lg: 'h-3'
+  };
+
+  const getLevelColor = (level: number) => {
+    if (level < 20) return 'bg-green-500';
+    if (level < 60) return 'bg-yellow-500';
+    return 'bg-red-500';
+  };
+
+  const getTextColor = (level: number) => {
+    if (level < 20) return 'text-green-600 dark:text-green-400';
+    if (level < 60) return 'text-yellow-600 dark:text-yellow-400';
+    return 'text-red-600 dark:text-red-400';
+  };
+
+  return (
+    <div className={clsx('space-y-1', className)}>
+      {showLabel && (
+        <div className="flex justify-between text-xs">
+          <span className="text-slate-500 dark:text-slate-400">
+            Microphone Level
+          </span>
+          <span className={clsx(
+            'font-mono',
+            isActive ? getTextColor(level) : 'text-slate-400 dark:text-slate-500'
+          )}>
+            {isActive ? `${Math.round(level)}%` : 'No Signal'}
+          </span>
+        </div>
+      )}
+      
+      <div className={clsx(
+        'w-full rounded-full bg-slate-200 dark:bg-slate-700',
+        sizeClasses[size]
+      )}>
+        <div
+          className={clsx(
+            'rounded-full transition-all duration-150 ease-out',
+            sizeClasses[size],
+            isActive ? getLevelColor(level) : 'bg-slate-300 dark:bg-slate-600'
+          )}
+          style={{
+            width: isActive ? `${Math.min(100, Math.max(2, level))}%` : '0%'
+          }}
+        />
+      </div>
+      
+      {/* Peak indicators */}
+      <div className="flex justify-between text-xs text-slate-400 dark:text-slate-500">
+        <span>0%</span>
+        <span>50%</span>
+        <span>100%</span>
+      </div>
+    </div>
+  );
+};

ui/src/components/AudioMetricsDashboard.tsx

@@ -0,0 +1,880 @@
+import { useEffect, useState } from "react";
+import { MdGraphicEq, MdSignalWifi4Bar, MdError, MdMic } from "react-icons/md";
+import { LuActivity, LuClock, LuHardDrive, LuSettings, LuCpu, LuMemoryStick } from "react-icons/lu";
+
+import { AudioLevelMeter } from "@components/AudioLevelMeter";
+import StatChart from "@components/StatChart";
+import { cx } from "@/cva.config";
+import { useMicrophone } from "@/hooks/useMicrophone";
+import { useAudioLevel } from "@/hooks/useAudioLevel";
+import { useAudioEvents } from "@/hooks/useAudioEvents";
+import api from "@/api";
+import { AUDIO_CONFIG } from "@/config/constants";
+import audioQualityService from "@/services/audioQualityService";
+
+interface AudioMetrics {
+  frames_received: number;
+  frames_dropped: number;
+  bytes_processed: number;
+  last_frame_time: string;
+  connection_drops: number;
+  average_latency: string;
+}
+
+interface MicrophoneMetrics {
+  frames_sent: number;
+  frames_dropped: number;
+  bytes_processed: number;
+  last_frame_time: string;
+  connection_drops: number;
+  average_latency: string;
+}
+
+interface ProcessMetrics {
+  cpu_percent: number;
+  memory_percent: number;
+  memory_rss: number;
+  memory_vms: number;
+  running: boolean;
+}
+
+interface AudioConfig {
+  Quality: number;
+  Bitrate: number;
+  SampleRate: number;
+  Channels: number;
+  FrameSize: string;
+}
+
+// Quality labels will be managed by the audio quality service
+const getQualityLabels = () => audioQualityService.getQualityLabels();
+
+// Format percentage values to 2 decimal places
+function formatPercentage(value: number | null | undefined): string {
+  if (value === null || value === undefined || isNaN(value)) {
+    return "0.00%";
+  }
+  return `${value.toFixed(2)}%`;
+}
+
+function formatMemoryMB(rssBytes: number | null | undefined): string {
+  if (rssBytes === null || rssBytes === undefined || isNaN(rssBytes)) {
+    return "0.00 MB";
+  }
+  const mb = rssBytes / (1024 * 1024);
+  return `${mb.toFixed(2)} MB`;
+}
+
+// Default system memory estimate in MB (will be replaced by actual value from backend)
+const DEFAULT_SYSTEM_MEMORY_MB = 4096; // 4GB default
+
+// Create chart array similar to connectionStats.tsx
+function createChartArray<T, K extends keyof T>(
+  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 AudioMetricsDashboard() {
+  // System memory state
+  const [systemMemoryMB, setSystemMemoryMB] = useState(DEFAULT_SYSTEM_MEMORY_MB);
+
+  // Use WebSocket-based audio events for real-time updates
+  const { 
+    audioMetrics, 
+    microphoneMetrics: wsMicrophoneMetrics, 
+    audioProcessMetrics: wsAudioProcessMetrics,
+    microphoneProcessMetrics: wsMicrophoneProcessMetrics,
+    isConnected: wsConnected 
+  } = useAudioEvents();
+
+  // Fetch system memory information on component mount
+  useEffect(() => {
+    const fetchSystemMemory = async () => {
+      try {
+        const response = await api.GET('/system/memory');
+        const data = await response.json();
+        setSystemMemoryMB(data.total_memory_mb);
+      } catch {
+        // Failed to fetch system memory, using default
+      }
+    };
+    fetchSystemMemory();
+  }, []);
+
+  // Update historical data when WebSocket process metrics are received
+   useEffect(() => {
+     if (wsConnected && wsAudioProcessMetrics && wsAudioProcessMetrics.running) {
+       const now = Math.floor(Date.now() / 1000); // Convert to seconds for StatChart
+       // Validate that now is a valid number
+       if (isNaN(now)) return;
+       
+       const cpuStat = isNaN(wsAudioProcessMetrics.cpu_percent) ? null : wsAudioProcessMetrics.cpu_percent;
+       
+       setAudioCpuStats(prev => {
+         const newMap = new Map(prev);
+         newMap.set(now, { cpu_percent: cpuStat });
+         // Keep only last 120 seconds of data for memory management
+         const cutoff = now - 120;
+         for (const [key] of newMap) {
+           if (key < cutoff) newMap.delete(key);
+         }
+         return newMap;
+       });
+       
+       setAudioMemoryStats(prev => {
+         const newMap = new Map(prev);
+         const memoryRss = isNaN(wsAudioProcessMetrics.memory_rss) ? null : wsAudioProcessMetrics.memory_rss;
+         newMap.set(now, { memory_rss: memoryRss });
+         // Keep only last 120 seconds of data for memory management
+         const cutoff = now - 120;
+         for (const [key] of newMap) {
+           if (key < cutoff) newMap.delete(key);
+         }
+         return newMap;
+       });
+     }
+   }, [wsConnected, wsAudioProcessMetrics]);
+
+   useEffect(() => {
+     if (wsConnected && wsMicrophoneProcessMetrics) {
+       const now = Math.floor(Date.now() / 1000); // Convert to seconds for StatChart
+       // Validate that now is a valid number
+       if (isNaN(now)) return;
+       
+       const cpuStat = isNaN(wsMicrophoneProcessMetrics.cpu_percent) ? null : wsMicrophoneProcessMetrics.cpu_percent;
+       
+       setMicCpuStats(prev => {
+         const newMap = new Map(prev);
+         newMap.set(now, { cpu_percent: cpuStat });
+         // Keep only last 120 seconds of data for memory management
+         const cutoff = now - 120;
+         for (const [key] of newMap) {
+           if (key < cutoff) newMap.delete(key);
+         }
+         return newMap;
+       });
+       
+       setMicMemoryStats(prev => {
+         const newMap = new Map(prev);
+         const memoryRss = isNaN(wsMicrophoneProcessMetrics.memory_rss) ? null : wsMicrophoneProcessMetrics.memory_rss;
+         newMap.set(now, { memory_rss: memoryRss });
+         // Keep only last 120 seconds of data for memory management
+         const cutoff = now - 120;
+         for (const [key] of newMap) {
+           if (key < cutoff) newMap.delete(key);
+         }
+         return newMap;
+       });
+     }
+   }, [wsConnected, wsMicrophoneProcessMetrics]);
+  
+  // Fallback state for when WebSocket is not connected
+  const [fallbackMetrics, setFallbackMetrics] = useState<AudioMetrics | null>(null);
+  const [fallbackMicrophoneMetrics, setFallbackMicrophoneMetrics] = useState<MicrophoneMetrics | null>(null);
+  const [fallbackConnected, setFallbackConnected] = useState(false);
+  
+  // Process metrics state (fallback for when WebSocket is not connected)
+  const [fallbackAudioProcessMetrics, setFallbackAudioProcessMetrics] = useState<ProcessMetrics | null>(null);
+  const [fallbackMicrophoneProcessMetrics, setFallbackMicrophoneProcessMetrics] = useState<ProcessMetrics | null>(null);
+  
+  // Historical data for charts using Maps for better memory management
+  const [audioCpuStats, setAudioCpuStats] = useState<Map<number, { cpu_percent: number | null }>>(new Map());
+  const [audioMemoryStats, setAudioMemoryStats] = useState<Map<number, { memory_rss: number | null }>>(new Map());
+  const [micCpuStats, setMicCpuStats] = useState<Map<number, { cpu_percent: number | null }>>(new Map());
+  const [micMemoryStats, setMicMemoryStats] = useState<Map<number, { memory_rss: number | null }>>(new Map());
+  
+  // Configuration state (these don't change frequently, so we can load them once)
+  const [config, setConfig] = useState<AudioConfig | null>(null);
+  const [microphoneConfig, setMicrophoneConfig] = useState<AudioConfig | null>(null);
+  const [lastUpdate, setLastUpdate] = useState<Date>(new Date());
+  
+  // Use WebSocket data when available, fallback to polling data otherwise
+  const metrics = wsConnected && audioMetrics !== null ? audioMetrics : fallbackMetrics;
+  const microphoneMetrics = wsConnected && wsMicrophoneMetrics !== null ? wsMicrophoneMetrics : fallbackMicrophoneMetrics;
+  const audioProcessMetrics = wsConnected && wsAudioProcessMetrics !== null ? wsAudioProcessMetrics : fallbackAudioProcessMetrics;
+  const microphoneProcessMetrics = wsConnected && wsMicrophoneProcessMetrics !== null ? wsMicrophoneProcessMetrics : fallbackMicrophoneProcessMetrics;
+  const isConnected = wsConnected ? wsConnected : fallbackConnected;
+  
+  // Microphone state for audio level monitoring
+  const { isMicrophoneActive, isMicrophoneMuted, microphoneStream } = useMicrophone();
+  const { audioLevel, isAnalyzing } = useAudioLevel(
+  isMicrophoneActive ? microphoneStream : null,
+  {
+  enabled: isMicrophoneActive,
+  updateInterval: 120,
+  });
+
+  useEffect(() => {
+    // Load initial configuration (only once)
+    loadAudioConfig();
+    
+    // Set up fallback polling only when WebSocket is not connected
+    if (!wsConnected) {
+      loadAudioData();
+      const interval = setInterval(loadAudioData, 1000);
+      return () => clearInterval(interval);
+    }
+  }, [wsConnected]);
+
+  const loadAudioConfig = async () => {
+    try {
+      // Use centralized audio quality service
+      const { audio, microphone } = await audioQualityService.loadAllConfigurations();
+
+      if (audio) {
+        setConfig(audio.current);
+      }
+
+      if (microphone) {
+        setMicrophoneConfig(microphone.current);
+      }
+    } catch (error) {
+      console.error("Failed to load audio config:", error);
+    }
+  };
+
+  const loadAudioData = async () => {
+    try {
+      // Load metrics
+      const metricsResp = await api.GET("/audio/metrics");
+      if (metricsResp.ok) {
+        const metricsData = await metricsResp.json();
+        setFallbackMetrics(metricsData);
+        // Consider connected if API call succeeds, regardless of frame count
+        setFallbackConnected(true);
+        setLastUpdate(new Date());
+      } else {
+        setFallbackConnected(false);
+      }
+
+      // Load audio process metrics
+      try {
+        const audioProcessResp = await api.GET("/audio/process-metrics");
+        if (audioProcessResp.ok) {
+          const audioProcessData = await audioProcessResp.json();
+          setFallbackAudioProcessMetrics(audioProcessData);
+          
+          // Update historical data for charts (keep last 120 seconds)
+          if (audioProcessData.running) {
+            const now = Math.floor(Date.now() / 1000); // Convert to seconds for StatChart
+            // Validate that now is a valid number
+            if (isNaN(now)) return;
+            
+            const cpuStat = isNaN(audioProcessData.cpu_percent) ? null : audioProcessData.cpu_percent;
+            const memoryRss = isNaN(audioProcessData.memory_rss) ? null : audioProcessData.memory_rss;
+            
+            setAudioCpuStats(prev => {
+              const newMap = new Map(prev);
+              newMap.set(now, { cpu_percent: cpuStat });
+              // Keep only last 120 seconds of data for memory management
+              const cutoff = now - 120;
+              for (const [key] of newMap) {
+                if (key < cutoff) newMap.delete(key);
+              }
+              return newMap;
+            });
+            
+            setAudioMemoryStats(prev => {
+              const newMap = new Map(prev);
+              newMap.set(now, { memory_rss: memoryRss });
+              // Keep only last 120 seconds of data for memory management
+              const cutoff = now - 120;
+              for (const [key] of newMap) {
+                if (key < cutoff) newMap.delete(key);
+              }
+              return newMap;
+            });
+          }
+        }
+      } catch {
+        // Audio process metrics not available
+      }
+
+      // Load microphone metrics
+      try {
+        const micResp = await api.GET("/microphone/metrics");
+        if (micResp.ok) {
+          const micData = await micResp.json();
+          setFallbackMicrophoneMetrics(micData);
+        }
+      } catch {
+        // Microphone metrics might not be available, that's okay
+        // Microphone metrics not available
+      }
+
+      // Load microphone process metrics
+      try {
+        const micProcessResp = await api.GET("/microphone/process-metrics");
+        if (micProcessResp.ok) {
+          const micProcessData = await micProcessResp.json();
+          setFallbackMicrophoneProcessMetrics(micProcessData);
+          
+          // Update historical data for charts (keep last 120 seconds)
+          const now = Math.floor(Date.now() / 1000); // Convert to seconds for StatChart
+          // Validate that now is a valid number
+          if (isNaN(now)) return;
+          
+          const cpuStat = isNaN(micProcessData.cpu_percent) ? null : micProcessData.cpu_percent;
+          const memoryRss = isNaN(micProcessData.memory_rss) ? null : micProcessData.memory_rss;
+          
+          setMicCpuStats(prev => {
+            const newMap = new Map(prev);
+            newMap.set(now, { cpu_percent: cpuStat });
+            // Keep only last 120 seconds of data for memory management
+            const cutoff = now - 120;
+            for (const [key] of newMap) {
+              if (key < cutoff) newMap.delete(key);
+            }
+            return newMap;
+          });
+          
+          setMicMemoryStats(prev => {
+            const newMap = new Map(prev);
+            newMap.set(now, { memory_rss: memoryRss });
+            // Keep only last 120 seconds of data for memory management
+            const cutoff = now - 120;
+            for (const [key] of newMap) {
+              if (key < cutoff) newMap.delete(key);
+            }
+            return newMap;
+          });
+        }
+      } catch {
+        // Microphone process metrics not available
+      }
+    } catch (error) {
+      console.error("Failed to load audio data:", error);
+      setFallbackConnected(false);
+    }
+  };
+
+  const formatBytes = (bytes: number) => {
+    if (bytes === 0) return "0 B";
+    const k = 1024;
+    const sizes = ["B", "KB", "MB", "GB"];
+    const i = Math.floor(Math.log(bytes) / Math.log(k));
+    return parseFloat((bytes / Math.pow(k, i)).toFixed(2)) + " " + sizes[i];
+  };
+
+  const formatNumber = (num: number) => {
+    return new Intl.NumberFormat().format(num);
+  };
+
+  const getDropRate = () => {
+    if (!metrics || metrics.frames_received === 0) return 0;
+    return ((metrics.frames_dropped / metrics.frames_received) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER);
+  };
+
+
+
+
+
+  const getQualityColor = (quality: number) => {
+    switch (quality) {
+      case 0: return "text-yellow-600 dark:text-yellow-400";
+      case 1: return "text-blue-600 dark:text-blue-400";
+      case 2: return "text-green-600 dark:text-green-400";
+      case 3: return "text-purple-600 dark:text-purple-400";
+      default: return "text-slate-600 dark:text-slate-400";
+    }
+  };
+
+  return (
+    <div className="space-y-4">
+      {/* Header */}
+      <div className="flex items-center justify-between">
+        <div className="flex items-center gap-2">
+          <MdGraphicEq className="h-5 w-5 text-blue-600 dark:text-blue-400" />
+          <h3 className="text-lg font-semibold text-slate-900 dark:text-slate-100">
+            Audio Metrics
+          </h3>
+        </div>
+        <div className="flex items-center gap-2">
+          <div className={cx(
+            "h-2 w-2 rounded-full",
+            isConnected ? "bg-green-500" : "bg-red-500"
+          )} />
+          <span className="text-xs text-slate-500 dark:text-slate-400">
+            {isConnected ? "Active" : "Inactive"}
+          </span>
+        </div>
+      </div>
+
+      {/* Current Configuration */}
+      <div className="grid grid-cols-1 md:grid-cols-2 gap-4">
+        {config && (
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-2 flex items-center gap-2">
+              <LuSettings className="h-4 w-4 text-blue-600 dark:text-blue-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Audio Output Config
+              </span>
+            </div>
+            <div className="space-y-2 text-sm">
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Quality:</span>
+                <span className={cx("font-medium", getQualityColor(config.Quality))}>
+                  {getQualityLabels()[config.Quality]}
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Bitrate:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {config.Bitrate}kbps
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Sample Rate:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {config.SampleRate}Hz
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Channels:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {config.Channels}
+                </span>
+              </div>
+            </div>
+          </div>
+        )}
+
+        {microphoneConfig && (
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-2 flex items-center gap-2">
+              <MdMic className="h-4 w-4 text-green-600 dark:text-green-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Audio Input Config
+              </span>
+            </div>
+            <div className="space-y-2 text-sm">
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Quality:</span>
+                <span className={cx("font-medium", getQualityColor(microphoneConfig.Quality))}>
+                  {getQualityLabels()[microphoneConfig.Quality]}
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Bitrate:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {microphoneConfig.Bitrate}kbps
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Sample Rate:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {microphoneConfig.SampleRate}Hz
+                </span>
+              </div>
+              <div className="flex justify-between">
+                <span className="text-slate-500 dark:text-slate-400">Channels:</span>
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  {microphoneConfig.Channels}
+                </span>
+              </div>
+            </div>
+          </div>
+        )}
+      </div>
+
+
+
+      {/* Subprocess Resource Usage - Histogram View */}
+      <div className="grid grid-cols-1 md:grid-cols-2 gap-4">
+        {/* Audio Output Subprocess */}
+        {audioProcessMetrics && (
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-3 flex items-center gap-2">
+              <LuCpu className="h-4 w-4 text-blue-600 dark:text-blue-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Audio Output Process
+              </span>
+              <div className={cx(
+                "h-2 w-2 rounded-full ml-auto",
+                audioProcessMetrics.running ? "bg-green-500" : "bg-red-500"
+              )} />
+            </div>
+            <div className="space-y-4">
+              <div>
+                <h4 className="text-sm font-medium text-slate-900 dark:text-slate-100 mb-2">CPU Usage</h4>
+                <div className="h-24">
+                  <StatChart
+                    data={createChartArray(audioCpuStats, 'cpu_percent')}
+                    unit="%"
+                    domain={[0, 100]}
+                  />
+                </div>
+              </div>
+              <div>
+                <h4 className="text-sm font-medium text-slate-900 dark:text-slate-100 mb-2">Memory Usage</h4>
+                <div className="h-24">
+                  <StatChart 
+                    data={createChartArray(audioMemoryStats, 'memory_rss').map(item => ({
+                      date: item.date,
+                      stat: item.stat ? item.stat / (1024 * 1024) : null // Convert bytes to MB
+                    }))}
+                    unit="MB" 
+                    domain={[0, systemMemoryMB]} 
+                  />
+                </div>
+              </div>
+              <div className="grid grid-cols-2 gap-2 text-xs">
+                <div className="text-center p-2 bg-slate-50 dark:bg-slate-800 rounded">
+                  <div className="font-medium text-slate-900 dark:text-slate-100">
+                    {formatPercentage(audioProcessMetrics.cpu_percent)}
+                  </div>
+                  <div className="text-slate-500 dark:text-slate-400">CPU</div>
+                </div>
+                <div className="text-center p-2 bg-slate-50 dark:bg-slate-800 rounded">
+                  <div className="font-medium text-slate-900 dark:text-slate-100">
+                    {formatMemoryMB(audioProcessMetrics.memory_rss)}
+                  </div>
+                  <div className="text-slate-500 dark:text-slate-400">Memory</div>
+                </div>
+              </div>
+            </div>
+          </div>
+        )}
+
+        {/* Microphone Input Subprocess */}
+        {microphoneProcessMetrics && (
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-3 flex items-center gap-2">
+              <LuMemoryStick className="h-4 w-4 text-green-600 dark:text-green-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Microphone Input Process
+              </span>
+              <div className={cx(
+                "h-2 w-2 rounded-full ml-auto",
+                microphoneProcessMetrics.running ? "bg-green-500" : "bg-red-500"
+              )} />
+            </div>
+            <div className="space-y-4">
+              <div>
+                <h4 className="text-sm font-medium text-slate-900 dark:text-slate-100 mb-2">CPU Usage</h4>
+                <div className="h-24">
+                  <StatChart 
+                    data={createChartArray(micCpuStats, 'cpu_percent')} 
+                    unit="%" 
+                    domain={[0, 100]} 
+                  />
+                </div>
+              </div>
+              <div>
+                <h4 className="text-sm font-medium text-slate-900 dark:text-slate-100 mb-2">Memory Usage</h4>
+                <div className="h-24">
+                  <StatChart 
+                    data={createChartArray(micMemoryStats, 'memory_rss').map(item => ({
+                      date: item.date,
+                      stat: item.stat ? item.stat / (1024 * 1024) : null // Convert bytes to MB
+                    }))}
+                    unit="MB" 
+                    domain={[0, systemMemoryMB]} 
+                  />
+                </div>
+              </div>
+              <div className="grid grid-cols-2 gap-2 text-xs">
+                <div className="text-center p-2 bg-slate-50 dark:bg-slate-800 rounded">
+                  <div className="font-medium text-slate-900 dark:text-slate-100">
+                    {formatPercentage(microphoneProcessMetrics.cpu_percent)}
+                  </div>
+                  <div className="text-slate-500 dark:text-slate-400">CPU</div>
+                </div>
+                <div className="text-center p-2 bg-slate-50 dark:bg-slate-800 rounded">
+                  <div className="font-medium text-slate-900 dark:text-slate-100">
+                    {formatMemoryMB(microphoneProcessMetrics.memory_rss)}
+                  </div>
+                  <div className="text-slate-500 dark:text-slate-400">Memory</div>
+                </div>
+              </div>
+            </div>
+          </div>
+        )}
+      </div>
+
+      {/* Performance Metrics */}
+      {metrics && (
+        <div className="space-y-3">
+          {/* Audio Output Frames */}
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-2 flex items-center gap-2">
+              <LuActivity className="h-4 w-4 text-green-600 dark:text-green-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Audio Output
+              </span>
+            </div>
+            <div className="grid grid-cols-2 gap-3">
+              <div className="text-center">
+                <div className="text-2xl font-bold text-green-600 dark:text-green-400">
+                  {formatNumber(metrics.frames_received)}
+                </div>
+                <div className="text-xs text-slate-500 dark:text-slate-400">
+                  Frames Received
+                </div>
+              </div>
+              <div className="text-center">
+                <div className={cx(
+                  "text-2xl font-bold",
+                  metrics.frames_dropped > 0 
+                    ? "text-red-600 dark:text-red-400" 
+                    : "text-green-600 dark:text-green-400"
+                )}>
+                  {formatNumber(metrics.frames_dropped)}
+                </div>
+                <div className="text-xs text-slate-500 dark:text-slate-400">
+                  Frames Dropped
+                </div>
+              </div>
+            </div>
+            
+            {/* Drop Rate */}
+            <div className="mt-3 rounded-md bg-slate-50 p-2 dark:bg-slate-700">
+              <div className="flex items-center justify-between">
+                <span className="text-sm text-slate-600 dark:text-slate-400">
+                  Drop Rate
+                </span>
+                <span className={cx(
+                  "font-bold",
+                  getDropRate() > AUDIO_CONFIG.DROP_RATE_CRITICAL_THRESHOLD 
+                    ? "text-red-600 dark:text-red-400"
+                    : getDropRate() > AUDIO_CONFIG.DROP_RATE_WARNING_THRESHOLD
+                    ? "text-yellow-600 dark:text-yellow-400"
+                    : "text-green-600 dark:text-green-400"
+                )}>
+                  {getDropRate().toFixed(AUDIO_CONFIG.PERCENTAGE_DECIMAL_PLACES)}%
+                </span>
+              </div>
+              <div className="mt-1 h-2 w-full rounded-full bg-slate-200 dark:bg-slate-600">
+                <div 
+                  className={cx(
+                    "h-2 rounded-full transition-all duration-300",
+                    getDropRate() > AUDIO_CONFIG.DROP_RATE_CRITICAL_THRESHOLD 
+                      ? "bg-red-500"
+                      : getDropRate() > AUDIO_CONFIG.DROP_RATE_WARNING_THRESHOLD
+                      ? "bg-yellow-500"
+                      : "bg-green-500"
+                  )}
+                  style={{ width: `${Math.min(getDropRate(), AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE)}%` }}
+                />
+              </div>
+            </div>
+          </div>
+
+          {/* Microphone Input Metrics */}
+          {microphoneMetrics && (
+            <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+              <div className="mb-2 flex items-center gap-2">
+                <MdMic className="h-4 w-4 text-orange-600 dark:text-orange-400" />
+                <span className="font-medium text-slate-900 dark:text-slate-100">
+                  Microphone Input
+                </span>
+              </div>
+              <div className="grid grid-cols-2 gap-3">
+                <div className="text-center">
+                  <div className="text-2xl font-bold text-orange-600 dark:text-orange-400">
+                    {formatNumber(microphoneMetrics.frames_sent)}
+                  </div>
+                  <div className="text-xs text-slate-500 dark:text-slate-400">
+                    Frames Sent
+                  </div>
+                </div>
+                <div className="text-center">
+                  <div className={cx(
+                    "text-2xl font-bold",
+                    microphoneMetrics.frames_dropped > 0 
+                      ? "text-red-600 dark:text-red-400" 
+                      : "text-green-600 dark:text-green-400"
+                  )}>
+                    {formatNumber(microphoneMetrics.frames_dropped)}
+                  </div>
+                  <div className="text-xs text-slate-500 dark:text-slate-400">
+                    Frames Dropped
+                  </div>
+                </div>
+              </div>
+              
+              {/* Microphone Drop Rate */}
+              <div className="mt-3 rounded-md bg-slate-50 p-2 dark:bg-slate-700">
+                <div className="flex items-center justify-between">
+                  <span className="text-sm text-slate-600 dark:text-slate-400">
+                    Drop Rate
+                  </span>
+                  <span className={cx(
+                    "font-bold",
+                    (microphoneMetrics.frames_sent > 0 ? (microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER : 0) > AUDIO_CONFIG.DROP_RATE_CRITICAL_THRESHOLD 
+                      ? "text-red-600 dark:text-red-400"
+                      : (microphoneMetrics.frames_sent > 0 ? (microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER : 0) > AUDIO_CONFIG.DROP_RATE_WARNING_THRESHOLD
+                      ? "text-yellow-600 dark:text-yellow-400"
+                      : "text-green-600 dark:text-green-400"
+                  )}>
+                    {microphoneMetrics.frames_sent > 0 ? ((microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER).toFixed(AUDIO_CONFIG.PERCENTAGE_DECIMAL_PLACES) : "0.00"}%
+                  </span>
+                </div>
+                <div className="mt-1 h-2 w-full rounded-full bg-slate-200 dark:bg-slate-600">
+                  <div 
+                    className={cx(
+                      "h-2 rounded-full transition-all duration-300",
+                      (microphoneMetrics.frames_sent > 0 ? (microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER : 0) > AUDIO_CONFIG.DROP_RATE_CRITICAL_THRESHOLD 
+                        ? "bg-red-500"
+                        : (microphoneMetrics.frames_sent > 0 ? (microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER : 0) > AUDIO_CONFIG.DROP_RATE_WARNING_THRESHOLD
+                        ? "bg-yellow-500"
+                        : "bg-green-500"
+                    )}
+                    style={{ 
+                      width: `${Math.min(microphoneMetrics.frames_sent > 0 ? (microphoneMetrics.frames_dropped / microphoneMetrics.frames_sent) * AUDIO_CONFIG.PERCENTAGE_MULTIPLIER : 0, AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE)}%` 
+                    }}
+                  />
+                </div>
+              </div>
+              
+              {/* Microphone Audio Level */}
+              {isMicrophoneActive && (
+                <div className="mt-3 rounded-md bg-slate-50 p-2 dark:bg-slate-700">
+                  <AudioLevelMeter
+                    level={audioLevel}
+                    isActive={isMicrophoneActive && !isMicrophoneMuted && isAnalyzing}
+                    size="sm"
+                    showLabel={true}
+                  />
+                </div>
+              )}
+              
+              {/* Microphone Connection Health */}
+              <div className="mt-3 rounded-md bg-slate-50 p-2 dark:bg-slate-700">
+                <div className="mb-2 flex items-center gap-2">
+                  <MdSignalWifi4Bar className="h-3 w-3 text-purple-600 dark:text-purple-400" />
+                  <span className="text-sm font-medium text-slate-900 dark:text-slate-100">
+                    Connection Health
+                  </span>
+                </div>
+                <div className="space-y-2">
+                  <div className="flex justify-between">
+                    <span className="text-xs text-slate-500 dark:text-slate-400">
+                      Connection Drops:
+                    </span>
+                    <span className={cx(
+                      "text-xs font-medium",
+                      microphoneMetrics.connection_drops > 0 
+                        ? "text-red-600 dark:text-red-400" 
+                        : "text-green-600 dark:text-green-400"
+                    )}>
+                      {formatNumber(microphoneMetrics.connection_drops)}
+                    </span>
+                  </div>
+                  {microphoneMetrics.average_latency && (
+                    <div className="flex justify-between">
+                      <span className="text-xs text-slate-500 dark:text-slate-400">
+                        Avg Latency:
+                      </span>
+                      <span className="text-xs font-medium text-slate-900 dark:text-slate-100">
+                        {microphoneMetrics.average_latency}
+                      </span>
+                    </div>
+                  )}
+                </div>
+              </div>
+            </div>
+          )}
+
+          {/* Data Transfer */}
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-2 flex items-center gap-2">
+              <LuHardDrive className="h-4 w-4 text-blue-600 dark:text-blue-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Data Transfer
+              </span>
+            </div>
+            <div className="text-center">
+              <div className="text-2xl font-bold text-blue-600 dark:text-blue-400">
+                {formatBytes(metrics.bytes_processed)}
+              </div>
+              <div className="text-xs text-slate-500 dark:text-slate-400">
+                Total Processed
+              </div>
+            </div>
+          </div>
+
+          {/* Connection Health */}
+          <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-700">
+            <div className="mb-2 flex items-center gap-2">
+              <MdSignalWifi4Bar className="h-4 w-4 text-purple-600 dark:text-purple-400" />
+              <span className="font-medium text-slate-900 dark:text-slate-100">
+                Connection Health
+              </span>
+            </div>
+            <div className="space-y-2">
+              <div className="flex justify-between">
+                <span className="text-sm text-slate-500 dark:text-slate-400">
+                  Connection Drops:
+                </span>
+                <span className={cx(
+                  "font-medium",
+                  metrics.connection_drops > 0 
+                    ? "text-red-600 dark:text-red-400" 
+                    : "text-green-600 dark:text-green-400"
+                )}>
+                  {formatNumber(metrics.connection_drops)}
+                </span>
+              </div>
+              {metrics.average_latency && (
+                <div className="flex justify-between">
+                  <span className="text-sm text-slate-500 dark:text-slate-400">
+                    Avg Latency:
+                  </span>
+                  <span className="font-medium text-slate-900 dark:text-slate-100">
+                    {metrics.average_latency}
+                  </span>
+                </div>
+              )}
+            </div>
+          </div>
+        </div>
+      )}
+
+      {/* Last Update */}
+      <div className="flex items-center justify-center gap-2 text-xs text-slate-500 dark:text-slate-400">
+        <LuClock className="h-3 w-3" />
+        <span>Last updated: {lastUpdate.toLocaleTimeString()}</span>
+      </div>
+
+      {/* No Data State */}
+      {!metrics && (
+        <div className="flex flex-col items-center justify-center py-8 text-center">
+          <MdError className="h-12 w-12 text-slate-400 dark:text-slate-600" />
+          <h3 className="mt-2 text-sm font-medium text-slate-900 dark:text-slate-100">
+            No Audio Data
+          </h3>
+          <p className="mt-1 text-sm text-slate-500 dark:text-slate-400">
+            Audio metrics will appear when audio streaming is active.
+          </p>
+        </div>
+      )}
+    </div>
+  );
+}

ui/src/components/AudioStatusIndicator.tsx

@@ -0,0 +1,33 @@
+import { cx } from "@/cva.config";
+
+interface AudioMetrics {
+  frames_dropped: number;
+  // Add other metrics properties as needed
+}
+
+interface AudioStatusIndicatorProps {
+  metrics?: AudioMetrics;
+  label: string;
+  className?: string;
+}
+
+export function AudioStatusIndicator({ metrics, label, className }: AudioStatusIndicatorProps) {
+  const hasIssues = metrics && metrics.frames_dropped > 0;
+  
+  return (
+    <div className={cx(
+      "text-center p-2 bg-slate-50 dark:bg-slate-800 rounded",
+      className
+    )}>
+      <div className={cx(
+        "font-medium",
+        hasIssues
+          ? "text-red-600 dark:text-red-400" 
+          : "text-green-600 dark:text-green-400"
+      )}>
+        {hasIssues ? "Issues" : "Good"}
+      </div>
+      <div className="text-slate-500 dark:text-slate-400">{label}</div>
+    </div>
+  );
+}

ui/src/components/popovers/AudioControlPopover.tsx

@@ -1,9 +1,15 @@
 import { useEffect, useState } from "react";
 import { MdVolumeOff, MdVolumeUp, MdGraphicEq, MdMic, MdMicOff, MdRefresh } from "react-icons/md";
+import { LuActivity, LuSignal } from "react-icons/lu";
 
 import { Button } from "@components/Button";
+import { AudioLevelMeter } from "@components/AudioLevelMeter";
+import { AudioConfigDisplay } from "@components/AudioConfigDisplay";
+import { AudioStatusIndicator } from "@components/AudioStatusIndicator";
 import { cx } from "@/cva.config";
+import { useUiStore } from "@/hooks/stores";
 import { useAudioDevices } from "@/hooks/useAudioDevices";
+import { useAudioLevel } from "@/hooks/useAudioLevel";
 import { useAudioEvents } from "@/hooks/useAudioEvents";
 import api from "@/api";
 import notifications from "@/notifications";
@@ -43,9 +49,10 @@ const getQualityLabels = () => audioQualityService.getQualityLabels();
 
 interface AudioControlPopoverProps {
   microphone: MicrophoneHookReturn;
+  open?: boolean; // whether the popover is open (controls analysis)
 }
 
-export default function AudioControlPopover({ microphone }: AudioControlPopoverProps) {
+export default function AudioControlPopover({ microphone, open }: AudioControlPopoverProps) {
   const [currentConfig, setCurrentConfig] = useState<AudioConfig | null>(null);
   const [currentMicrophoneConfig, setCurrentMicrophoneConfig] = useState<AudioConfig | null>(null);
 
@@ -61,6 +68,8 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
   // Use WebSocket-based audio events for real-time updates
   const { 
     audioMuted, 
+    audioMetrics, 
+    microphoneMetrics, 
     isConnected: wsConnected 
   } = useAudioEvents();
   
@@ -70,6 +79,7 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
   const {
     isMicrophoneActive,
     isMicrophoneMuted,
+    microphoneStream,
     startMicrophone,
     stopMicrophone,
     toggleMicrophoneMute,
@@ -82,9 +92,16 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
   
   // Use WebSocket data exclusively - no polling fallback
   const isMuted = audioMuted ?? false;
+  const metrics = audioMetrics;
+  const micMetrics = microphoneMetrics;
   const isConnected = wsConnected;
   
-
+  // Audio level monitoring - enable only when popover is open and microphone is active to save resources
+  const analysisEnabled = (open ?? true) && isMicrophoneActive;
+  const { audioLevel, isAnalyzing } = useAudioLevel(analysisEnabled ? microphoneStream : null, {
+    enabled: analysisEnabled,
+    updateInterval: 120, // 8-10 fps to reduce CPU without losing UX quality
+  });
   
   // Audio devices
   const { 
@@ -99,7 +116,7 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
     refreshDevices 
   } = useAudioDevices();
   
-
+  const { toggleSidebarView } = useUiStore();
 
   // Load initial configurations once - cache to prevent repeated calls
   useEffect(() => {
@@ -358,7 +375,42 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
             </div>
           </div>
           
-
+          {/* Audio Level Meter */}
+          {isMicrophoneActive && (
+            <div className="rounded-lg bg-slate-50 p-3 dark:bg-slate-700">
+              <AudioLevelMeter
+                level={audioLevel}
+                isActive={isMicrophoneActive && !isMicrophoneMuted && isAnalyzing}
+                size="md"
+                showLabel={true}
+              />
+              {/* Debug information */}
+              <div className="mt-2 text-xs text-slate-500 dark:text-slate-400">
+                <div className="grid grid-cols-2 gap-1">
+                  <span>Stream: {microphoneStream ? '✓' : '✗'}</span>
+                  <span>Analyzing: {isAnalyzing ? '✓' : '✗'}</span>
+                  <span>Active: {isMicrophoneActive ? '✓' : '✗'}</span>
+                  <span>Muted: {isMicrophoneMuted ? '✓' : '✗'}</span>
+                </div>
+                {microphoneStream && (
+                  <div className="mt-1">
+                    Tracks: {microphoneStream.getAudioTracks().length}
+                    {microphoneStream.getAudioTracks().length > 0 && (
+                      <span className="ml-2">
+                        (Enabled: {microphoneStream.getAudioTracks().filter((t: MediaStreamTrack) => t.enabled).length})
+                      </span>
+                    )}
+                  </div>
+                )}
+                <button
+                  onClick={syncMicrophoneState}
+                  className="mt-1 text-blue-500 hover:text-blue-600 dark:text-blue-400 dark:hover:text-blue-300"
+                >
+                  Sync State
+                </button>
+              </div>
+            </div>
+          )}
         </div>
 
         {/* Device Selection */}
@@ -462,11 +514,10 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
             </div>
 
             {currentMicrophoneConfig && (
-              <div className="text-xs text-slate-600 dark:text-slate-400 mt-2">
+              <AudioConfigDisplay 
-                Quality: {currentMicrophoneConfig.Quality} | 
+                config={currentMicrophoneConfig} 
-                Bitrate: {currentMicrophoneConfig.Bitrate}kbps | 
+                variant="success" 
-                Sample Rate: {currentMicrophoneConfig.SampleRate}Hz
+              />
-              </div>
             )}
           </div>
         )}
@@ -500,16 +551,59 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
           </div>
 
           {currentConfig && (
-            <div className="text-xs text-slate-600 dark:text-slate-400 mt-2">
+            <AudioConfigDisplay 
-              Quality: {currentConfig.Quality} | 
+              config={currentConfig} 
-              Bitrate: {currentConfig.Bitrate}kbps | 
+              variant="default" 
-              Sample Rate: {currentConfig.SampleRate}Hz
+            />
+          )}
+        </div>
+
+        {/* Quick Status Summary */}
+        <div className="rounded-lg border border-slate-200 p-3 dark:border-slate-600">
+          <div className="flex items-center gap-2 mb-2">
+            <LuActivity className="h-4 w-4 text-slate-600 dark:text-slate-400" />
+            <span className="font-medium text-slate-900 dark:text-slate-100">
+              Quick Status
+            </span>
+          </div>
+          
+          {metrics ? (
+            <div className="grid grid-cols-2 gap-3 text-xs">
+              <AudioStatusIndicator 
+                metrics={metrics} 
+                label="Audio Output" 
+              />
+              
+              {micMetrics && (
+                <AudioStatusIndicator 
+                  metrics={micMetrics} 
+                  label="Microphone" 
+                />
+              )}
+            </div>
+          ) : (
+            <div className="text-center py-2">
+              <div className="text-sm text-slate-500 dark:text-slate-400">
+                No data available
+              </div>
             </div>
           )}
         </div>
 
-
+        {/* Audio Metrics Dashboard Button */}
-
+        <div className="pt-2 border-t border-slate-200 dark:border-slate-600">
+          <div className="flex justify-center">
+            <button
+              onClick={() => {
+                toggleSidebarView("audio-metrics");
+              }}
+              className="flex items-center gap-2 rounded-md border border-slate-200 bg-white px-4 py-2 text-sm font-medium text-slate-700 hover:bg-slate-50 dark:border-slate-600 dark:bg-slate-700 dark:text-slate-300 dark:hover:bg-slate-600 transition-colors"
+            >
+              <LuSignal className="h-4 w-4 text-blue-500" />
+              <span>View Full Audio Metrics</span>
+            </button>
+          </div>
+        </div>
       </div>
     </div>
   );

ui/src/components/sidebar/AudioMetricsSidebar.tsx

@@ -0,0 +1,16 @@
+import SidebarHeader from "@/components/SidebarHeader";
+import { useUiStore } from "@/hooks/stores";
+import AudioMetricsDashboard from "@/components/AudioMetricsDashboard";
+
+export default function AudioMetricsSidebar() {
+  const setSidebarView = useUiStore(state => state.setSidebarView);
+
+  return (
+    <>
+      <SidebarHeader title="Audio Metrics" setSidebarView={setSidebarView} />
+      <div className="h-full overflow-y-scroll bg-white px-4 py-2 pb-8 dark:bg-slate-900">
+        <AudioMetricsDashboard />
+      </div>
+    </>
+  );
+}

ui/src/hooks/stores.ts

@@ -40,7 +40,7 @@ const appendStatToMap = <T extends { timestamp: number }>(
 };
 
 // Constants and types
-export type AvailableSidebarViews = "connection-stats";
+export type AvailableSidebarViews = "connection-stats" | "audio-metrics";
 export type AvailableTerminalTypes = "kvm" | "serial" | "none";
 
 export interface User {

ui/src/hooks/useAudioEvents.ts

@@ -7,7 +7,11 @@ import { NETWORK_CONFIG } from '../config/constants';
 // Audio event types matching the backend
 export type AudioEventType = 
   | 'audio-mute-changed'
+  | 'audio-metrics-update'
   | 'microphone-state-changed'
+  | 'microphone-metrics-update'
+  | 'audio-process-metrics'
+  | 'microphone-process-metrics'
   | 'audio-device-changed';
 
 // Audio event data interfaces
@@ -15,11 +19,39 @@ export interface AudioMuteData {
   muted: boolean;
 }
 
+export interface AudioMetricsData {
+  frames_received: number;
+  frames_dropped: number;
+  bytes_processed: number;
+  last_frame_time: string;
+  connection_drops: number;
+  average_latency: string;
+}
+
 export interface MicrophoneStateData {
   running: boolean;
   session_active: boolean;
 }
 
+export interface MicrophoneMetricsData {
+  frames_sent: number;
+  frames_dropped: number;
+  bytes_processed: number;
+  last_frame_time: string;
+  connection_drops: number;
+  average_latency: string;
+}
+
+export interface ProcessMetricsData {
+  pid: number;
+  cpu_percent: number;
+  memory_rss: number;
+  memory_vms: number;
+  memory_percent: number;
+  running: boolean;
+  process_name: string;
+}
+
 export interface AudioDeviceChangedData {
   enabled: boolean;
   reason: string;
@@ -28,7 +60,7 @@ export interface AudioDeviceChangedData {
 // Audio event structure
 export interface AudioEvent {
   type: AudioEventType;
-  data: AudioMuteData | MicrophoneStateData | AudioDeviceChangedData;
+  data: AudioMuteData | AudioMetricsData | MicrophoneStateData | MicrophoneMetricsData | ProcessMetricsData | AudioDeviceChangedData;
 }
 
 // Hook return type
@@ -39,9 +71,15 @@ export interface UseAudioEventsReturn {
   
   // Audio state
   audioMuted: boolean | null;
+  audioMetrics: AudioMetricsData | null;
   
   // Microphone state
   microphoneState: MicrophoneStateData | null;
+  microphoneMetrics: MicrophoneMetricsData | null;
+  
+  // Process metrics
+  audioProcessMetrics: ProcessMetricsData | null;
+  microphoneProcessMetrics: ProcessMetricsData | null;
   
   // Device change events
   onAudioDeviceChanged?: (data: AudioDeviceChangedData) => void;
@@ -61,7 +99,11 @@ const globalSubscriptionState = {
 export function useAudioEvents(onAudioDeviceChanged?: (data: AudioDeviceChangedData) => void): UseAudioEventsReturn {
   // State for audio data
   const [audioMuted, setAudioMuted] = useState<boolean | null>(null);
+  const [audioMetrics, setAudioMetrics] = useState<AudioMetricsData | null>(null);
   const [microphoneState, setMicrophoneState] = useState<MicrophoneStateData | null>(null);
+  const [microphoneMetrics, setMicrophoneMetricsData] = useState<MicrophoneMetricsData | null>(null);
+  const [audioProcessMetrics, setAudioProcessMetrics] = useState<ProcessMetricsData | null>(null);
+  const [microphoneProcessMetrics, setMicrophoneProcessMetrics] = useState<ProcessMetricsData | null>(null);
   
   // Local subscription state
   const [isLocallySubscribed, setIsLocallySubscribed] = useState(false);
@@ -183,6 +225,12 @@ export function useAudioEvents(onAudioDeviceChanged?: (data: AudioDeviceChangedD
               break;
             }
               
+            case 'audio-metrics-update': {
+              const audioMetricsData = audioEvent.data as AudioMetricsData;
+              setAudioMetrics(audioMetricsData);
+              break;
+            }
+              
             case 'microphone-state-changed': {
               const micStateData = audioEvent.data as MicrophoneStateData;
               setMicrophoneState(micStateData);
@@ -190,6 +238,24 @@ export function useAudioEvents(onAudioDeviceChanged?: (data: AudioDeviceChangedD
               break;
             }
               
+            case 'microphone-metrics-update': {
+              const micMetricsData = audioEvent.data as MicrophoneMetricsData;
+              setMicrophoneMetricsData(micMetricsData);
+              break;
+            }
+              
+            case 'audio-process-metrics': {
+              const audioProcessData = audioEvent.data as ProcessMetricsData;
+              setAudioProcessMetrics(audioProcessData);
+              break;
+            }
+              
+            case 'microphone-process-metrics': {
+              const micProcessData = audioEvent.data as ProcessMetricsData;
+              setMicrophoneProcessMetrics(micProcessData);
+              break;
+            }
+              
             case 'audio-device-changed': {
               const deviceChangedData = audioEvent.data as AudioDeviceChangedData;
               // Audio device changed
@@ -254,9 +320,15 @@ export function useAudioEvents(onAudioDeviceChanged?: (data: AudioDeviceChangedD
     
     // Audio state
     audioMuted,
+    audioMetrics,
     
     // Microphone state
     microphoneState,
+    microphoneMetrics: microphoneMetrics,
+    
+    // Process metrics
+    audioProcessMetrics,
+    microphoneProcessMetrics,
     
     // Device change events
     onAudioDeviceChanged,

ui/src/hooks/useAudioLevel.ts

@@ -0,0 +1,136 @@
+import { useEffect, useRef, useState } from 'react';
+
+import { AUDIO_CONFIG } from '@/config/constants';
+
+interface AudioLevelHookResult {
+  audioLevel: number; // 0-100 percentage
+  isAnalyzing: boolean;
+}
+
+interface AudioLevelOptions {
+  enabled?: boolean; // Allow external control of analysis
+  updateInterval?: number; // Throttle updates (default from AUDIO_CONFIG)
+}
+
+export const useAudioLevel = (
+  stream: MediaStream | null, 
+  options: AudioLevelOptions = {}
+): AudioLevelHookResult => {
+  const { enabled = true, updateInterval = AUDIO_CONFIG.LEVEL_UPDATE_INTERVAL } = options;
+  
+  const [audioLevel, setAudioLevel] = useState(0);
+  const [isAnalyzing, setIsAnalyzing] = useState(false);
+  const audioContextRef = useRef<AudioContext | null>(null);
+  const analyserRef = useRef<AnalyserNode | null>(null);
+  const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);
+  const intervalRef = useRef<number | null>(null);
+  const lastUpdateTimeRef = useRef<number>(0);
+
+  useEffect(() => {
+    if (!stream || !enabled) {
+      // Clean up when stream is null or disabled
+      if (intervalRef.current !== null) {
+        clearInterval(intervalRef.current);
+        intervalRef.current = null;
+      }
+      if (sourceRef.current) {
+        sourceRef.current.disconnect();
+        sourceRef.current = null;
+      }
+      if (audioContextRef.current) {
+        audioContextRef.current.close();
+        audioContextRef.current = null;
+      }
+      analyserRef.current = null;
+      setIsAnalyzing(false);
+      setAudioLevel(0);
+      return;
+    }
+
+    const audioTracks = stream.getAudioTracks();
+    if (audioTracks.length === 0) {
+      setIsAnalyzing(false);
+      setAudioLevel(0);
+      return;
+    }
+
+    try {
+      // Create audio context and analyser
+      const audioContext = new (window.AudioContext || (window as Window & { webkitAudioContext?: typeof AudioContext }).webkitAudioContext)();
+      const analyser = audioContext.createAnalyser();
+      const source = audioContext.createMediaStreamSource(stream);
+
+      // Configure analyser - use smaller FFT for better performance
+      analyser.fftSize = AUDIO_CONFIG.FFT_SIZE;
+      analyser.smoothingTimeConstant = AUDIO_CONFIG.SMOOTHING_TIME_CONSTANT;
+      
+      // Connect nodes
+      source.connect(analyser);
+
+      // Store references
+      audioContextRef.current = audioContext;
+      analyserRef.current = analyser;
+      sourceRef.current = source;
+
+      const dataArray = new Uint8Array(analyser.frequencyBinCount);
+
+      const updateLevel = () => {
+        if (!analyserRef.current) return;
+
+        const now = performance.now();
+        
+        // Throttle updates to reduce CPU usage
+        if (now - lastUpdateTimeRef.current < updateInterval) {
+          return;
+        }
+        lastUpdateTimeRef.current = now;
+
+        analyserRef.current.getByteFrequencyData(dataArray);
+        
+        // Optimized RMS calculation - process only relevant frequency bands
+        let sum = 0;
+        const relevantBins = Math.min(dataArray.length, AUDIO_CONFIG.RELEVANT_FREQUENCY_BINS);
+        for (let i = 0; i < relevantBins; i++) {
+          const value = dataArray[i];
+          sum += value * value;
+        }
+        const rms = Math.sqrt(sum / relevantBins);
+        
+        // Convert to percentage (0-100) with better scaling
+        const level = Math.min(AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE, Math.max(0, (rms / AUDIO_CONFIG.RMS_SCALING_FACTOR) * AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE));
+        setAudioLevel(Math.round(level));
+      };
+
+      setIsAnalyzing(true);
+      
+      // Use setInterval instead of requestAnimationFrame for more predictable timing
+      intervalRef.current = window.setInterval(updateLevel, updateInterval);
+
+    } catch {
+      // Audio level analyzer creation failed - silently handle
+      setIsAnalyzing(false);
+      setAudioLevel(0);
+    }
+
+    // Cleanup function
+    return () => {
+      if (intervalRef.current !== null) {
+        clearInterval(intervalRef.current);
+        intervalRef.current = null;
+      }
+      if (sourceRef.current) {
+        sourceRef.current.disconnect();
+        sourceRef.current = null;
+      }
+      if (audioContextRef.current) {
+        audioContextRef.current.close();
+        audioContextRef.current = null;
+      }
+      analyserRef.current = null;
+      setIsAnalyzing(false);
+      setAudioLevel(0);
+    };
+  }, [stream, enabled, updateInterval]);
+
+  return { audioLevel, isAnalyzing };
+};

ui/src/hooks/useMicrophone.ts

@@ -3,7 +3,7 @@ import { useCallback, useEffect, useRef, useState } from "react";
 import { useRTCStore } from "@/hooks/stores";
 import api from "@/api";
 import { devLog, devInfo, devWarn, devError, devOnly } from "@/utils/debug";
-import { AUDIO_CONFIG } from "@/config/constants";
+import { NETWORK_CONFIG, AUDIO_CONFIG } from "@/config/constants";
 
 export interface MicrophoneError {
   type: 'permission' | 'device' | 'network' | 'unknown';
@@ -84,7 +84,53 @@ export function useMicrophone() {
     setMicrophoneMuted(false);
   }, [microphoneSender, peerConnection, setMicrophoneStream, setMicrophoneSender, setMicrophoneActive, setMicrophoneMuted]);
 
+  // Debug function to check current state (can be called from browser console)
+  const debugMicrophoneState = useCallback(() => {
+    const refStream = microphoneStreamRef.current;
+    const state = {
+      isMicrophoneActive,
+      isMicrophoneMuted,
+      streamInRef: !!refStream,
+      streamInStore: !!microphoneStream,
+      senderInStore: !!microphoneSender,
+      streamId: refStream?.id,
+      storeStreamId: microphoneStream?.id,
+      audioTracks: refStream?.getAudioTracks().length || 0,
+      storeAudioTracks: microphoneStream?.getAudioTracks().length || 0,
+      audioTrackDetails: refStream?.getAudioTracks().map(track => ({
+        id: track.id,
+        label: track.label,
+        enabled: track.enabled,
+        readyState: track.readyState,
+        muted: track.muted
+      })) || [],
+      peerConnectionState: peerConnection ? {
+        connectionState: peerConnection.connectionState,
+        iceConnectionState: peerConnection.iceConnectionState,
+        signalingState: peerConnection.signalingState
+      } : "No peer connection",
+      streamMatch: refStream === microphoneStream
+    };
+    devLog("Microphone Debug State:", state);
+    
+    // Also check if streams are active
+    if (refStream) {
+      devLog("Ref stream active tracks:", refStream.getAudioTracks().filter(t => t.readyState === 'live').length);
+    }
+    if (microphoneStream && microphoneStream !== refStream) {
+      devLog("Store stream active tracks:", microphoneStream.getAudioTracks().filter(t => t.readyState === 'live').length);
+    }
+    
+    return state;
+  }, [isMicrophoneActive, isMicrophoneMuted, microphoneStream, microphoneSender, peerConnection]);
 
+  // Make debug function available globally for console access
+  useEffect(() => {
+    (window as Window & { debugMicrophoneState?: () => unknown }).debugMicrophoneState = debugMicrophoneState;
+    return () => {
+      delete (window as Window & { debugMicrophoneState?: () => unknown }).debugMicrophoneState;
+    };
+  }, [debugMicrophoneState]);
 
   const lastSyncRef = useRef<number>(0);
   const isStartingRef = useRef<boolean>(false); // Track if we're in the middle of starting
@@ -449,7 +495,51 @@ export function useMicrophone() {
     }
   }, [peerConnection, setMicrophoneStream, setMicrophoneSender, setMicrophoneActive, setMicrophoneMuted, stopMicrophoneStream, isMicrophoneActive, isMicrophoneMuted, microphoneStream, isStarting, isStopping, isToggling]);
 
-
+  // Reset backend microphone state
+  const resetBackendMicrophoneState = useCallback(async (): Promise<boolean> => {
+    try {
+      devLog("Resetting backend microphone state...");
+      const response = await api.POST("/microphone/reset", {});
+      
+      if (response.ok) {
+        const data = await response.json();
+        devLog("Backend microphone reset successful:", data);
+        
+        // Update frontend state to match backend
+        setMicrophoneActive(false);
+        setMicrophoneMuted(false);
+        
+        // Clean up any orphaned streams
+        if (microphoneStreamRef.current) {
+          devLog("Cleaning up orphaned stream after reset");
+          await stopMicrophoneStream();
+        }
+        
+        // Wait a bit for everything to settle
+        await new Promise(resolve => setTimeout(resolve, 200));
+        
+        // Sync state to ensure consistency
+        await syncMicrophoneState();
+        
+        return true;
+      } else {
+        devError("Backend microphone reset failed:", response.status);
+        return false;
+      }
+    } catch (error) {
+      devWarn("Failed to reset backend microphone state:", error);
+      // Fallback to old method
+      try {
+        devLog("Trying fallback reset method...");
+        await api.POST("/microphone/stop", {});
+        await new Promise(resolve => setTimeout(resolve, 300));
+        return true;
+      } catch (fallbackError) {
+        devError("Fallback reset also failed:", fallbackError);
+        return false;
+      }
+    }
+  }, [setMicrophoneActive, setMicrophoneMuted, stopMicrophoneStream, syncMicrophoneState]);
 
   // Stop microphone
   const stopMicrophone = useCallback(async (): Promise<{ success: boolean; error?: MicrophoneError }> => {
@@ -589,9 +679,173 @@ export function useMicrophone() {
     }
   }, [microphoneStream, isMicrophoneActive, isMicrophoneMuted, setMicrophoneMuted, isStarting, isStopping, isToggling]);
 
+  // Function to check WebRTC audio transmission stats
+  const checkAudioTransmissionStats = useCallback(async () => {
+    if (!microphoneSender) {
+      devLog("No microphone sender available");
+      return null;
+    }
 
+    try {
+      const stats = await microphoneSender.getStats();
+      const audioStats: {
+        id: string;
+        type: string;
+        kind: string;
+        packetsSent?: number;
+        bytesSent?: number;
+        timestamp?: number;
+        ssrc?: number;
+      }[] = [];
+      
+      stats.forEach((report, id) => {
+        if (report.type === 'outbound-rtp' && report.kind === 'audio') {
+          audioStats.push({
+            id,
+            type: report.type,
+            kind: report.kind,
+            packetsSent: report.packetsSent,
+            bytesSent: report.bytesSent,
+            timestamp: report.timestamp,
+            ssrc: report.ssrc
+          });
+        }
+      });
+      
+      devLog("Audio transmission stats:", audioStats);
+      return audioStats;
+    } catch (error) {
+      devError("Failed to get audio transmission stats:", error);
+      return null;
+    }
+  }, [microphoneSender]);
 
-
+  // Comprehensive test function to diagnose microphone issues
+  const testMicrophoneAudio = useCallback(async () => {
+    devLog("=== MICROPHONE AUDIO TEST ===");
+    
+    // 1. Check if we have a stream
+    const stream = microphoneStreamRef.current;
+    if (!stream) {
+      devLog("❌ No microphone stream available");
+      return;
+    }
+    
+    devLog("✅ Microphone stream exists:", stream.id);
+    
+    // 2. Check audio tracks
+    const audioTracks = stream.getAudioTracks();
+    devLog("Audio tracks:", audioTracks.length);
+    
+    if (audioTracks.length === 0) {
+      devLog("❌ No audio tracks in stream");
+      return;
+    }
+    
+    const track = audioTracks[0];
+    devLog("✅ Audio track details:", {
+      id: track.id,
+      label: track.label,
+      enabled: track.enabled,
+      readyState: track.readyState,
+      muted: track.muted
+    });
+    
+    // 3. Test audio level detection manually
+    try {
+      const audioContext = new (window.AudioContext || (window as Window & { webkitAudioContext?: typeof AudioContext }).webkitAudioContext)();
+      const analyser = audioContext.createAnalyser();
+      const source = audioContext.createMediaStreamSource(stream);
+      
+      analyser.fftSize = AUDIO_CONFIG.ANALYSIS_FFT_SIZE;
+      source.connect(analyser);
+      
+      const dataArray = new Uint8Array(analyser.frequencyBinCount);
+      
+      devLog("🎤 Testing audio level detection for 5 seconds...");
+      devLog("Please speak into your microphone now!");
+      
+      let maxLevel = 0;
+      let sampleCount = 0;
+      
+      const testInterval = setInterval(() => {
+        analyser.getByteFrequencyData(dataArray);
+        
+        let sum = 0;
+        for (const value of dataArray) {
+          sum += value * value;
+        }
+        const rms = Math.sqrt(sum / dataArray.length);
+        const level = Math.min(AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE, (rms / AUDIO_CONFIG.LEVEL_SCALING_FACTOR) * AUDIO_CONFIG.MAX_LEVEL_PERCENTAGE);
+        
+        maxLevel = Math.max(maxLevel, level);
+        sampleCount++;
+        
+        if (sampleCount % 10 === 0) { // Log every 10th sample
+          devLog(`Audio level: ${level.toFixed(1)}% (max so far: ${maxLevel.toFixed(1)}%)`);
+        }
+      }, AUDIO_CONFIG.ANALYSIS_UPDATE_INTERVAL);
+      
+      setTimeout(() => {
+        clearInterval(testInterval);
+        source.disconnect();
+        audioContext.close();
+        
+        devLog("🎤 Audio test completed!");
+        devLog(`Maximum audio level detected: ${maxLevel.toFixed(1)}%`);
+        
+        if (maxLevel > 5) {
+          devLog("✅ Microphone is detecting audio!");
+        } else {
+          devLog("❌ No significant audio detected. Check microphone permissions and hardware.");
+        }
+      }, NETWORK_CONFIG.AUDIO_TEST_DURATION);
+      
+    } catch (error) {
+      devError("❌ Failed to test audio level:", error);
+    }
+    
+    // 4. Check WebRTC sender
+    if (microphoneSender) {
+      devLog("✅ WebRTC sender exists");
+      devLog("Sender track:", {
+        id: microphoneSender.track?.id,
+        kind: microphoneSender.track?.kind,
+        enabled: microphoneSender.track?.enabled,
+        readyState: microphoneSender.track?.readyState
+      });
+      
+      // Check if sender track matches stream track
+      if (microphoneSender.track === track) {
+        devLog("✅ Sender track matches stream track");
+      } else {
+        devLog("❌ Sender track does NOT match stream track");
+      }
+    } else {
+      devLog("❌ No WebRTC sender available");
+    }
+    
+    // 5. Check peer connection
+    if (peerConnection) {
+      devLog("✅ Peer connection exists");
+      devLog("Connection state:", peerConnection.connectionState);
+      devLog("ICE connection state:", peerConnection.iceConnectionState);
+      
+      const transceivers = peerConnection.getTransceivers();
+      const audioTransceivers = transceivers.filter(t => 
+        t.sender.track?.kind === 'audio' || t.receiver.track?.kind === 'audio'
+      );
+      
+      devLog("Audio transceivers:", audioTransceivers.map(t => ({
+        direction: t.direction,
+        senderTrack: t.sender.track?.id,
+        receiverTrack: t.receiver.track?.id
+      })));
+    } else {
+      devLog("❌ No peer connection available");
+    }
+    
+  }, [microphoneSender, peerConnection]);
 
   const startMicrophoneDebounced = useCallback((deviceId?: string) => {
     debouncedOperation(async () => {
@@ -605,7 +859,59 @@ export function useMicrophone() {
     }, "stop");
   }, [stopMicrophone, debouncedOperation]);
 
-
+  // Make debug functions available globally for console access
+  useEffect(() => {
+    (window as Window & { 
+      debugMicrophone?: () => unknown;
+      checkAudioStats?: () => unknown;
+      testMicrophoneAudio?: () => unknown;
+      resetBackendMicrophone?: () => unknown;
+    }).debugMicrophone = debugMicrophoneState;
+    (window as Window & { 
+      debugMicrophone?: () => unknown;
+      checkAudioStats?: () => unknown;
+      testMicrophoneAudio?: () => unknown;
+      resetBackendMicrophone?: () => unknown;
+    }).checkAudioStats = checkAudioTransmissionStats;
+    (window as Window & { 
+      debugMicrophone?: () => unknown;
+      checkAudioStats?: () => unknown;
+      testMicrophoneAudio?: () => unknown;
+      resetBackendMicrophone?: () => unknown;
+    }).testMicrophoneAudio = testMicrophoneAudio;
+    (window as Window & { 
+      debugMicrophone?: () => unknown;
+      checkAudioStats?: () => unknown;
+      testMicrophoneAudio?: () => unknown;
+      resetBackendMicrophone?: () => unknown;
+    }).resetBackendMicrophone = resetBackendMicrophoneState;
+    return () => {
+      delete (window as Window & { 
+        debugMicrophone?: () => unknown;
+        checkAudioStats?: () => unknown;
+        testMicrophoneAudio?: () => unknown;
+        resetBackendMicrophone?: () => unknown;
+      }).debugMicrophone;
+      delete (window as Window & { 
+        debugMicrophone?: () => unknown;
+        checkAudioStats?: () => unknown;
+        testMicrophoneAudio?: () => unknown;
+        resetBackendMicrophone?: () => unknown;
+      }).checkAudioStats;
+      delete (window as Window & { 
+        debugMicrophone?: () => unknown;
+        checkAudioStats?: () => unknown;
+        testMicrophoneAudio?: () => unknown;
+        resetBackendMicrophone?: () => unknown;
+      }).testMicrophoneAudio;
+      delete (window as Window & { 
+        debugMicrophone?: () => unknown;
+        checkAudioStats?: () => unknown;
+        testMicrophoneAudio?: () => unknown;
+        resetBackendMicrophone?: () => unknown;
+      }).resetBackendMicrophone;
+    };
+  }, [debugMicrophoneState, checkAudioTransmissionStats, testMicrophoneAudio, resetBackendMicrophoneState]);
 
   // Sync state on mount
   useEffect(() => {
@@ -635,7 +941,7 @@ export function useMicrophone() {
     startMicrophone,
     stopMicrophone,
     toggleMicrophoneMute,
-
+    debugMicrophoneState,
     // Expose debounced variants for UI handlers
     startMicrophoneDebounced,
     stopMicrophoneDebounced,

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

@@ -39,6 +39,7 @@ import WebRTCVideo from "@components/WebRTCVideo";
 import { checkAuth, isInCloud, isOnDevice } from "@/main";
 import DashboardNavbar from "@components/Header";
 import ConnectionStatsSidebar from "@/components/sidebar/connectionStats";
+import AudioMetricsSidebar from "@/components/sidebar/AudioMetricsSidebar";
 import { JsonRpcRequest, JsonRpcResponse, useJsonRpc } from "@/hooks/useJsonRpc";
 import Terminal from "@components/Terminal";
 import { CLOUD_API, DEVICE_API } from "@/ui.config";
@@ -924,7 +925,22 @@ function SidebarContainer(props: SidebarContainerProps) {
               <ConnectionStatsSidebar />
             </motion.div>
           )}
-
+          {sidebarView === "audio-metrics" && (
+            <motion.div
+              className="absolute inset-0"
+              initial={{ opacity: 0 }}
+              animate={{ opacity: 1 }}
+              exit={{ opacity: 0 }}
+              transition={{
+                duration: 0.5,
+                ease: "easeInOut",
+              }}
+            >
+              <div className="grid h-full grid-rows-(--grid-headerBody) shadow-xs">
+                <AudioMetricsSidebar />
+              </div>
+            </motion.div>
+          )}
         </AnimatePresence>
       </div>
     </div>

web.go

@@ -24,7 +24,8 @@ import (
 	"github.com/google/uuid"
 	"github.com/jetkvm/kvm/internal/logging"
 	"github.com/pion/webrtc/v4"
-
+	"github.com/prometheus/client_golang/prometheus"
+	"github.com/prometheus/client_golang/prometheus/promhttp"
 	"github.com/rs/zerolog"
 	"golang.org/x/crypto/bcrypt"
 )
@@ -102,6 +103,9 @@ func setupRouter() *gin.Engine {
 	// We use this to setup the device in the welcome page
 	r.POST("/device/setup", handleSetup)
 
+	// A Prometheus metrics endpoint.
+	r.GET("/metrics", gin.WrapH(promhttp.Handler()))
+
 	// Developer mode protected routes
 	developerModeRouter := r.Group("/developer/")
 	developerModeRouter.Use(basicAuthProtectedMiddleware(true))
@@ -207,6 +211,19 @@ func setupRouter() *gin.Engine {
 		})
 	})
 
+	protected.GET("/audio/metrics", func(c *gin.Context) {
+		registry := audio.GetMetricsRegistry()
+		metrics := registry.GetAudioMetrics()
+		c.JSON(200, gin.H{
+			"frames_received":  metrics.FramesReceived,
+			"frames_dropped":   metrics.FramesDropped,
+			"bytes_processed":  metrics.BytesProcessed,
+			"last_frame_time":  metrics.LastFrameTime,
+			"connection_drops": metrics.ConnectionDrops,
+			"average_latency":  fmt.Sprintf("%.1fms", float64(metrics.AverageLatency.Nanoseconds())/1e6),
+		})
+	})
+
 	protected.GET("/microphone/quality", func(c *gin.Context) {
 		config := audio.GetMicrophoneConfig()
 		presets := audio.GetMicrophoneQualityPresets()
@@ -382,6 +399,103 @@ func setupRouter() *gin.Engine {
 		})
 	})
 
+	protected.GET("/microphone/metrics", func(c *gin.Context) {
+		registry := audio.GetMetricsRegistry()
+		metrics := registry.GetAudioInputMetrics()
+		c.JSON(200, gin.H{
+			"frames_sent":      metrics.FramesSent,
+			"frames_dropped":   metrics.FramesDropped,
+			"bytes_processed":  metrics.BytesProcessed,
+			"last_frame_time":  metrics.LastFrameTime.Format("2006-01-02T15:04:05.000Z"),
+			"connection_drops": metrics.ConnectionDrops,
+			"average_latency":  fmt.Sprintf("%.1fms", float64(metrics.AverageLatency.Nanoseconds())/1e6),
+		})
+	})
+
+	// Audio subprocess process metrics endpoints
+	protected.GET("/audio/process-metrics", func(c *gin.Context) {
+		// Access the global audio supervisor from main.go
+		if audioSupervisor == nil {
+			c.JSON(200, gin.H{
+				"cpu_percent":    0.0,
+				"memory_percent": 0.0,
+				"memory_rss":     0,
+				"memory_vms":     0,
+				"running":        false,
+			})
+			return
+		}
+
+		metrics := audioSupervisor.GetProcessMetrics()
+		if metrics == nil {
+			c.JSON(200, gin.H{
+				"cpu_percent":    0.0,
+				"memory_percent": 0.0,
+				"memory_rss":     0,
+				"memory_vms":     0,
+				"running":        false,
+			})
+			return
+		}
+
+		c.JSON(200, gin.H{
+			"cpu_percent":    metrics.CPUPercent,
+			"memory_percent": metrics.MemoryPercent,
+			"memory_rss":     metrics.MemoryRSS,
+			"memory_vms":     metrics.MemoryVMS,
+			"running":        true,
+		})
+	})
+
+	// Audio memory allocation metrics endpoint
+	protected.GET("/audio/memory-metrics", gin.WrapF(audio.HandleMemoryMetrics))
+
+	protected.GET("/microphone/process-metrics", func(c *gin.Context) {
+		if currentSession == nil || currentSession.AudioInputManager == nil {
+			c.JSON(200, gin.H{
+				"cpu_percent":    0.0,
+				"memory_percent": 0.0,
+				"memory_rss":     0,
+				"memory_vms":     0,
+				"running":        false,
+			})
+			return
+		}
+
+		// Get the supervisor from the audio input manager
+		supervisor := currentSession.AudioInputManager.GetSupervisor()
+		if supervisor == nil {
+			c.JSON(200, gin.H{
+				"cpu_percent":    0.0,
+				"memory_percent": 0.0,
+				"memory_rss":     0,
+				"memory_vms":     0,
+				"running":        false,
+			})
+			return
+		}
+
+		metrics := supervisor.GetProcessMetrics()
+		if metrics == nil {
+			c.JSON(200, gin.H{
+				"cpu_percent":    0.0,
+				"memory_percent": 0.0,
+				"memory_rss":     0,
+				"memory_vms":     0,
+				"running":        false,
+			})
+			return
+		}
+
+		c.JSON(200, gin.H{
+			"cpu_percent":    metrics.CPUPercent,
+			"memory_percent": metrics.MemoryPercent,
+			"memory_rss":     metrics.MemoryRSS,
+			"memory_vms":     metrics.MemoryVMS,
+			"running":        true,
+		})
+	})
+
 	// System memory information endpoint
 	protected.GET("/system/memory", func(c *gin.Context) {
 		processMonitor := audio.GetProcessMonitor()
@@ -598,7 +712,11 @@ func handleWebRTCSignalWsMessages(
 				return
 			}
 
-			// Metrics collection disabled
+			// set the timer for the ping duration
+			timer := prometheus.NewTimer(prometheus.ObserverFunc(func(v float64) {
+				metricConnectionLastPingDuration.WithLabelValues(sourceType, source).Set(v)
+				metricConnectionPingDuration.WithLabelValues(sourceType, source).Observe(v)
+			}))
 
 			l.Trace().Msg("sending ping frame")
 			err := wsCon.Ping(runCtx)
@@ -609,9 +727,13 @@ func handleWebRTCSignalWsMessages(
 				return
 			}
 
-			// Metrics collection disabled
+			// dont use `defer` here because we want to observe the duration of the ping
+			duration := timer.ObserveDuration()
 
-			l.Trace().Msg("received pong frame")
+			metricConnectionTotalPingSentCount.WithLabelValues(sourceType, source).Inc()
+			metricConnectionLastPingTimestamp.WithLabelValues(sourceType, source).SetToCurrentTime()
+
+			l.Trace().Str("duration", duration.String()).Msg("received pong frame")
 		}
 	}()
 
@@ -657,7 +779,8 @@ func handleWebRTCSignalWsMessages(
 				return err
 			}
 
-			// Metrics collection disabled
+			metricConnectionTotalPingReceivedCount.WithLabelValues(sourceType, source).Inc()
+			metricConnectionLastPingReceivedTimestamp.WithLabelValues(sourceType, source).SetToCurrentTime()
 
 			continue
 		}
@@ -681,7 +804,8 @@ func handleWebRTCSignalWsMessages(
 				l.Info().Str("oidcGoogle", req.OidcGoogle).Msg("new session request with OIDC Google")
 			}
 
-			// Metrics collection disabled
+			metricConnectionSessionRequestCount.WithLabelValues(sourceType, source).Inc()
+			metricConnectionLastSessionRequestTimestamp.WithLabelValues(sourceType, source).SetToCurrentTime()
 			err = handleSessionRequest(runCtx, wsCon, req, isCloudConnection, source, &l)
 			if err != nil {
 				l.Warn().Str("error", err.Error()).Msg("error starting new session")