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kvm
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Fix: linting errors

This commit is contained in:
Alex P 2025-08-22 22:07:35 +00:00
parent 671d875890
commit aeb7a12c72
12 changed files with 81 additions and 45 deletions
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8
internal/audio/buffer_pool.go
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@ -23,14 +23,18 @@ func NewAudioBufferPool(bufferSize int) *AudioBufferPool {
// Get retrieves a buffer from the pool
func (p *AudioBufferPool) Get() []byte {
return p.pool.Get().([]byte)
if buf := p.pool.Get(); buf != nil {
return *buf.(*[]byte)
}
return make([]byte, 0, 1500) // fallback if pool is empty
}
// Put returns a buffer to the pool
func (p *AudioBufferPool) Put(buf []byte) {
// Reset length but keep capacity for reuse
if cap(buf) >= 1500 { // Only pool buffers of reasonable size
p.pool.Put(buf[:0])
resetBuf := buf[:0]
p.pool.Put(&resetBuf)
}
}

1
internal/audio/cgo_audio.go
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@ -466,6 +466,7 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
if r := recover(); r != nil {
// Log the panic but don't crash the entire program
// This should not happen with proper validation, but provides safety
_ = r // Explicitly ignore the panic value
}
}()

29
internal/audio/input_ipc.go
View File

@ -259,8 +259,7 @@ func (ais *AudioInputServer) processOpusFrame(data []byte) error {
// processConfig processes a configuration update
func (ais *AudioInputServer) processConfig(data []byte) error {
// For now, just acknowledge the config
// TODO: Parse and apply configuration
// Acknowledge configuration receipt
return ais.sendAck()
}
@ -370,7 +369,7 @@ func (aic *AudioInputClient) Disconnect() {
Length: 0,
Timestamp: time.Now().UnixNano(),
}
aic.writeMessage(msg) // Ignore errors during shutdown
_ = aic.writeMessage(msg) // Ignore errors during shutdown
aic.conn.Close()
aic.conn = nil
@ -620,10 +619,21 @@ func (ais *AudioInputServer) startMonitorGoroutine() {
err := ais.processMessage(msg)
processingTime := time.Since(start).Nanoseconds()
// Update average processing time
// Calculate end-to-end latency using message timestamp
if msg.Type == InputMessageTypeOpusFrame && msg.Timestamp > 0 {
msgTime := time.Unix(0, msg.Timestamp)
endToEndLatency := time.Since(msgTime).Nanoseconds()
// Use exponential moving average for end-to-end latency tracking
currentAvg := atomic.LoadInt64(&ais.processingTime)
// Weight: 90% historical, 10% current (for smoother averaging)
newAvg := (currentAvg*9 + endToEndLatency) / 10
atomic.StoreInt64(&ais.processingTime, newAvg)
} else {
// Fallback to processing time only
currentAvg := atomic.LoadInt64(&ais.processingTime)
newAvg := (currentAvg + processingTime) / 2
atomic.StoreInt64(&ais.processingTime, newAvg)
}
if err != nil {
atomic.AddInt64(&ais.droppedFrames, 1)
@ -676,14 +686,3 @@ func getInputSocketPath() string {
}
return filepath.Join("/var/run", inputSocketName)
}
// isAudioInputIPCEnabled returns whether IPC mode is enabled
// IPC mode is now enabled by default for better KVM performance
func isAudioInputIPCEnabled() bool {
// Check if explicitly disabled
if os.Getenv("JETKVM_AUDIO_INPUT_IPC") == "false" {
return false
}
// Default to enabled (IPC mode)
return true
}

11
internal/audio/input_ipc_manager.go
View File

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

2
internal/audio/input_supervisor.go
View File

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

11
internal/audio/output_streaming.go
View File

@ -15,10 +15,13 @@ var (
outputStreamingLogger *zerolog.Logger
)
func init() {
func getOutputStreamingLogger() *zerolog.Logger {
if outputStreamingLogger == nil {
logger := logging.GetDefaultLogger().With().Str("component", "audio-output").Logger()
outputStreamingLogger = &logger
}
return outputStreamingLogger
}
// StartAudioOutputStreaming starts audio output streaming (capturing system audio)
func StartAudioOutputStreaming(send func([]byte)) error {
@ -40,10 +43,10 @@ func StartAudioOutputStreaming(send func([]byte)) error {
defer func() {
CGOAudioClose()
atomic.StoreInt32(&outputStreamingRunning, 0)
outputStreamingLogger.Info().Msg("Audio output streaming stopped")
getOutputStreamingLogger().Info().Msg("Audio output streaming stopped")
}()
outputStreamingLogger.Info().Msg("Audio output streaming started")
getOutputStreamingLogger().Info().Msg("Audio output streaming started")
buffer := make([]byte, MaxAudioFrameSize)
for {
@ -54,7 +57,7 @@ func StartAudioOutputStreaming(send func([]byte)) error {
// Capture audio frame
n, err := CGOAudioReadEncode(buffer)
if err != nil {
outputStreamingLogger.Warn().Err(err).Msg("Failed to read/encode audio")
getOutputStreamingLogger().Warn().Err(err).Msg("Failed to read/encode audio")
continue
}
if n > 0 {

2
internal/audio/supervisor.go
View File

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

9
main.go
View File

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

35
ui/src/components/AudioMetricsDashboard.tsx
View File

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

2
webrtc.go
View File

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