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kvm/internal/audio/config_constants.go

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package audio
import "time"
// AudioConfigConstants centralizes all hardcoded values used across audio components.
// This configuration system allows runtime tuning of audio performance, quality, and resource usage.
// Each constant is documented with its purpose, usage location, and impact on system behavior.
type AudioConfigConstants struct {
// Audio Quality Presets
// MaxAudioFrameSize defines the maximum size of an audio frame in bytes.
// Used in: buffer_pool.go, adaptive_buffer.go
// Impact: Higher values allow larger audio chunks but increase memory usage and latency.
// Typical range: 1024-8192 bytes. Default 4096 provides good balance.
MaxAudioFrameSize int
// Opus Encoding Parameters - Core codec settings for audio compression
// OpusBitrate sets the target bitrate for Opus encoding in bits per second.
// Used in: cgo_audio.go for encoder initialization
// Impact: Higher bitrates improve audio quality but increase bandwidth usage.
// Range: 6000-510000 bps. 128000 (128kbps) provides high quality for most use cases.
OpusBitrate int
// OpusComplexity controls the computational complexity of Opus encoding (0-10).
// Used in: cgo_audio.go for encoder configuration
// Impact: Higher values improve quality but increase CPU usage and encoding latency.
// Range: 0-10. Value 10 provides best quality, 0 fastest encoding.
OpusComplexity int
// OpusVBR enables Variable Bit Rate encoding (0=CBR, 1=VBR).
// Used in: cgo_audio.go for encoder mode selection
// Impact: VBR (1) adapts bitrate to content complexity, improving efficiency.
// CBR (0) maintains constant bitrate for predictable bandwidth usage.
OpusVBR int
// OpusVBRConstraint enables constrained VBR mode (0=unconstrained, 1=constrained).
// Used in: cgo_audio.go when VBR is enabled
// Impact: Constrained VBR (1) limits bitrate variation for more predictable bandwidth.
// Unconstrained (0) allows full bitrate adaptation for optimal quality.
OpusVBRConstraint int
// OpusDTX enables Discontinuous Transmission (0=disabled, 1=enabled).
// Used in: cgo_audio.go for encoder optimization
// Impact: DTX (1) reduces bandwidth during silence but may cause audio artifacts.
// Disabled (0) maintains constant transmission for consistent quality.
OpusDTX int
// Audio Parameters - Fundamental audio stream characteristics
// SampleRate defines the number of audio samples per second in Hz.
// Used in: All audio processing components
// Impact: Higher rates improve frequency response but increase processing load.
// Common values: 16000 (voice), 44100 (CD quality), 48000 (professional).
SampleRate int
// Channels specifies the number of audio channels (1=mono, 2=stereo).
// Used in: All audio processing and encoding/decoding operations
// Impact: Stereo (2) provides spatial audio but doubles bandwidth and processing.
// Mono (1) reduces resource usage but loses spatial information.
Channels int
// FrameSize defines the number of samples per audio frame.
// Used in: Opus encoding/decoding, buffer management
// Impact: Larger frames reduce overhead but increase latency.
// Must match Opus frame sizes: 120, 240, 480, 960, 1920, 2880 samples.
FrameSize int
// MaxPacketSize sets the maximum size of encoded audio packets in bytes.
// Used in: Network transmission, buffer allocation
// Impact: Larger packets reduce network overhead but increase burst bandwidth.
// Should accommodate worst-case Opus output plus protocol headers.
MaxPacketSize int
// Audio Quality Bitrates - Predefined quality presets for different use cases
// These bitrates are used in audio.go for quality level selection
// Impact: Higher bitrates improve audio fidelity but increase bandwidth usage
// AudioQualityLowOutputBitrate defines bitrate for low-quality audio output (kbps).
// Used in: audio.go for bandwidth-constrained scenarios
// Impact: Minimal bandwidth usage but reduced audio quality. Suitable for voice-only.
// Default 32kbps provides acceptable voice quality with very low bandwidth.
AudioQualityLowOutputBitrate int
// AudioQualityLowInputBitrate defines bitrate for low-quality audio input (kbps).
// Used in: audio.go for microphone input in low-bandwidth scenarios
// Impact: Reduces upload bandwidth but may affect voice clarity.
// Default 16kbps suitable for basic voice communication.
AudioQualityLowInputBitrate int
// AudioQualityMediumOutputBitrate defines bitrate for medium-quality audio output (kbps).
// Used in: audio.go for balanced quality/bandwidth scenarios
// Impact: Good balance between quality and bandwidth usage.
// Default 64kbps provides clear voice and acceptable music quality.
AudioQualityMediumOutputBitrate int
// AudioQualityMediumInputBitrate defines bitrate for medium-quality audio input (kbps).
// Used in: audio.go for microphone input with balanced quality
// Impact: Better voice quality than low setting with moderate bandwidth usage.
// Default 32kbps suitable for clear voice communication.
AudioQualityMediumInputBitrate int
// AudioQualityHighOutputBitrate defines bitrate for high-quality audio output (kbps).
// Used in: audio.go for high-fidelity audio scenarios
// Impact: Excellent audio quality but higher bandwidth requirements.
// Default 128kbps provides near-CD quality for music and crystal-clear voice.
AudioQualityHighOutputBitrate int
// AudioQualityHighInputBitrate defines bitrate for high-quality audio input (kbps).
// Used in: audio.go for high-quality microphone capture
// Impact: Superior voice quality but increased upload bandwidth usage.
// Default 64kbps suitable for professional voice communication.
AudioQualityHighInputBitrate int
// AudioQualityUltraOutputBitrate defines bitrate for ultra-high-quality audio output (kbps).
// Used in: audio.go for maximum quality scenarios
// Impact: Maximum audio fidelity but highest bandwidth consumption.
// Default 192kbps provides studio-quality audio for critical applications.
AudioQualityUltraOutputBitrate int
// AudioQualityUltraInputBitrate defines bitrate for ultra-high-quality audio input (kbps).
// Used in: audio.go for maximum quality microphone capture
// Impact: Best possible voice quality but maximum upload bandwidth usage.
// Default 96kbps suitable for broadcast-quality voice communication.
AudioQualityUltraInputBitrate int
// Audio Quality Sample Rates - Frequency sampling rates for different quality levels
// Used in: audio.go for configuring audio capture and playback sample rates
// Impact: Higher sample rates capture more frequency detail but increase processing load
// AudioQualityLowSampleRate defines sample rate for low-quality audio (Hz).
// Used in: audio.go for bandwidth-constrained scenarios
// Impact: Reduces frequency response but minimizes processing and bandwidth.
// Default 22050Hz captures frequencies up to 11kHz, adequate for voice.
AudioQualityLowSampleRate int
// AudioQualityMediumSampleRate defines sample rate for medium-quality audio (Hz).
// Used in: audio.go for balanced quality scenarios
// Impact: Good frequency response with moderate processing requirements.
// Default 44100Hz (CD quality) captures frequencies up to 22kHz.
AudioQualityMediumSampleRate int
// AudioQualityMicLowSampleRate defines sample rate for low-quality microphone input (Hz).
// Used in: audio.go for microphone capture in constrained scenarios
// Impact: Optimized for voice communication with minimal processing overhead.
// Default 16000Hz captures voice frequencies (300-3400Hz) efficiently.
AudioQualityMicLowSampleRate int
// Audio Quality Frame Sizes - Duration of audio frames for different quality levels
// Used in: audio.go for configuring Opus frame duration
// Impact: Larger frames reduce overhead but increase latency and memory usage
// AudioQualityLowFrameSize defines frame duration for low-quality audio.
// Used in: audio.go for low-latency scenarios with minimal processing
// Impact: Longer frames reduce CPU overhead but increase audio latency.
// Default 40ms provides good efficiency for voice communication.
AudioQualityLowFrameSize time.Duration
// AudioQualityMediumFrameSize defines frame duration for medium-quality audio.
// Used in: audio.go for balanced latency and efficiency
// Impact: Moderate frame size balances latency and processing efficiency.
// Default 20ms provides good balance for most applications.
AudioQualityMediumFrameSize time.Duration
// AudioQualityHighFrameSize defines frame duration for high-quality audio.
// Used in: audio.go for high-quality scenarios
// Impact: Optimized frame size for high-quality encoding efficiency.
// Default 20ms maintains low latency while supporting high bitrates.
AudioQualityHighFrameSize time.Duration
// AudioQualityUltraFrameSize defines frame duration for ultra-quality audio.
// Used in: audio.go for maximum quality scenarios
// Impact: Smaller frames reduce latency but increase processing overhead.
// Default 10ms provides minimal latency for real-time applications.
AudioQualityUltraFrameSize time.Duration
// Audio Quality Channels - Channel configuration for different quality levels
// Used in: audio.go for configuring mono/stereo audio
// Impact: Stereo doubles bandwidth and processing but provides spatial audio
// AudioQualityLowChannels defines channel count for low-quality audio.
// Used in: audio.go for bandwidth-constrained scenarios
// Impact: Mono (1) minimizes bandwidth and processing for voice communication.
// Default 1 (mono) suitable for voice-only applications.
AudioQualityLowChannels int
// AudioQualityMediumChannels defines channel count for medium-quality audio.
// Used in: audio.go for balanced quality scenarios
// Impact: Stereo (2) provides spatial audio with moderate bandwidth increase.
// Default 2 (stereo) suitable for general audio applications.
AudioQualityMediumChannels int
// AudioQualityHighChannels defines channel count for high-quality audio.
// Used in: audio.go for high-fidelity scenarios
// Impact: Stereo (2) essential for high-quality music and spatial audio.
// Default 2 (stereo) required for full audio experience.
AudioQualityHighChannels int
// AudioQualityUltraChannels defines channel count for ultra-quality audio.
// Used in: audio.go for maximum quality scenarios
// Impact: Stereo (2) mandatory for studio-quality audio reproduction.
// Default 2 (stereo) provides full spatial audio fidelity.
AudioQualityUltraChannels int
// CGO Audio Constants - Low-level C library configuration for audio processing
// These constants are passed to C code in cgo_audio.go for native audio operations
// Impact: Direct control over native audio library behavior and performance
// CGOOpusBitrate sets the bitrate for native Opus encoder (bits per second).
// Used in: cgo_audio.go update_audio_constants() function
// Impact: Controls quality vs bandwidth tradeoff in native encoding.
// Default 96000 (96kbps) provides good quality for real-time applications.
CGOOpusBitrate int
// CGOOpusComplexity sets computational complexity for native Opus encoder (0-10).
// Used in: cgo_audio.go for native encoder configuration
// Impact: Higher values improve quality but increase CPU usage in C code.
// Default 3 balances quality and performance for embedded systems.
CGOOpusComplexity int
// CGOOpusVBR enables Variable Bit Rate in native Opus encoder (0=CBR, 1=VBR).
// Used in: cgo_audio.go for native encoder mode selection
// Impact: VBR (1) adapts bitrate dynamically for better efficiency.
// Default 1 (VBR) provides optimal bandwidth utilization.
CGOOpusVBR int
// CGOOpusVBRConstraint enables constrained VBR in native encoder (0/1).
// Used in: cgo_audio.go when VBR is enabled
// Impact: Constrains bitrate variation for more predictable bandwidth.
// Default 1 (constrained) provides controlled bandwidth usage.
CGOOpusVBRConstraint int
// CGOOpusSignalType specifies signal type hint for native Opus encoder.
// Used in: cgo_audio.go for encoder optimization
// Impact: Optimizes encoder for specific content type (voice vs music).
// Values: 3=OPUS_SIGNAL_MUSIC for general audio, 2=OPUS_SIGNAL_VOICE for speech.
CGOOpusSignalType int
// CGOOpusBandwidth sets frequency bandwidth for native Opus encoder.
// Used in: cgo_audio.go for encoder frequency range configuration
// Impact: Controls frequency range vs bitrate efficiency.
// Default 1105 (OPUS_BANDWIDTH_FULLBAND) uses full 20kHz bandwidth.
CGOOpusBandwidth int
// CGOOpusDTX enables Discontinuous Transmission in native encoder (0/1).
// Used in: cgo_audio.go for bandwidth optimization during silence
// Impact: Reduces bandwidth during silence but may cause audio artifacts.
// Default 0 (disabled) maintains consistent audio quality.
CGOOpusDTX int
// CGOSampleRate defines sample rate for native audio processing (Hz).
// Used in: cgo_audio.go for ALSA and Opus configuration
// Impact: Must match system audio capabilities and Opus requirements.
// Default 48000Hz provides professional audio quality.
CGOSampleRate int
// CGOChannels defines channel count for native audio processing.
// Used in: cgo_audio.go for ALSA device and Opus encoder configuration
// Impact: Must match audio hardware capabilities and application needs.
// Default 2 (stereo) provides full spatial audio support.
CGOChannels int
// CGOFrameSize defines frame size for native Opus processing (samples).
// Used in: cgo_audio.go for Opus encoder/decoder frame configuration
// Impact: Must be valid Opus frame size, affects latency and efficiency.
// Default 960 samples (20ms at 48kHz) balances latency and efficiency.
CGOFrameSize int
// CGOMaxPacketSize defines maximum packet size for native encoding (bytes).
// Used in: cgo_audio.go for buffer allocation in C code
// Impact: Must accommodate worst-case Opus output to prevent buffer overruns.
// Default 1500 bytes handles typical Opus output with safety margin.
CGOMaxPacketSize int
// Input IPC Constants - Configuration for audio input inter-process communication
// Used in: input_ipc.go for microphone audio capture and processing
// Impact: Controls audio input quality and processing efficiency
// InputIPCSampleRate defines sample rate for input IPC audio processing (Hz).
// Used in: input_ipc.go for microphone capture configuration
// Impact: Must match microphone capabilities and encoding requirements.
// Default 48000Hz provides professional quality microphone input.
InputIPCSampleRate int
// InputIPCChannels defines channel count for input IPC audio processing.
// Used in: input_ipc.go for microphone channel configuration
// Impact: Stereo (2) captures spatial audio, mono (1) reduces processing.
// Default 2 (stereo) supports full microphone array capabilities.
InputIPCChannels int
// InputIPCFrameSize defines frame size for input IPC processing (samples).
// Used in: input_ipc.go for microphone frame processing
// Impact: Larger frames reduce overhead but increase input latency.
// Default 960 samples (20ms at 48kHz) balances latency and efficiency.
InputIPCFrameSize int
// Output IPC Constants - Configuration for audio output inter-process communication
// Used in: output_streaming.go for audio playback and streaming
// Impact: Controls audio output quality, latency, and reliability
// OutputMaxFrameSize defines maximum frame size for output processing (bytes).
// Used in: output_streaming.go for buffer allocation and frame processing
// Impact: Larger frames allow bigger audio chunks but increase memory usage.
// Default 4096 bytes accommodates typical audio frames with safety margin.
OutputMaxFrameSize int
// OutputWriteTimeout defines timeout for output write operations.
// Used in: output_streaming.go for preventing blocking on slow audio devices
// Impact: Shorter timeouts improve responsiveness but may cause audio drops.
// Default 10ms prevents blocking while allowing device response time.
OutputWriteTimeout time.Duration
// OutputMaxDroppedFrames defines maximum consecutive dropped frames before error.
// Used in: output_streaming.go for audio quality monitoring
// Impact: Higher values tolerate more audio glitches but may degrade experience.
// Default 50 frames allows brief interruptions while detecting serious issues.
OutputMaxDroppedFrames int
// OutputHeaderSize defines size of output message headers (bytes).
// Used in: output_streaming.go for message parsing and buffer allocation
// Impact: Must match actual header size to prevent parsing errors.
// Default 17 bytes matches current output message header format.
OutputHeaderSize int
// OutputMessagePoolSize defines size of output message object pool.
// Used in: output_streaming.go for memory management and performance
// Impact: Larger pools reduce allocation overhead but increase memory usage.
// Default 128 messages provides good balance for typical workloads.
OutputMessagePoolSize int
// Socket Buffer Constants - Network socket buffer configuration for audio streaming
// Used in: socket_buffer.go for optimizing network performance
// Impact: Controls network throughput, latency, and memory usage
// SocketOptimalBuffer defines optimal socket buffer size (bytes).
// Used in: socket_buffer.go for default socket buffer configuration
// Impact: Balances throughput and memory usage for typical audio streams.
// Default 131072 (128KB) provides good performance for most scenarios.
SocketOptimalBuffer int
// SocketMaxBuffer defines maximum socket buffer size (bytes).
// Used in: socket_buffer.go for high-throughput scenarios
// Impact: Larger buffers improve throughput but increase memory usage and latency.
// Default 262144 (256KB) handles high-bitrate audio without excessive memory.
SocketMaxBuffer int
// SocketMinBuffer defines minimum socket buffer size (bytes).
// Used in: socket_buffer.go for low-memory scenarios
// Impact: Smaller buffers reduce memory usage but may limit throughput.
// Default 32768 (32KB) provides minimum viable buffer for audio streaming.
SocketMinBuffer int
// Scheduling Policy Constants - Linux process scheduling policies for audio threads
// Used in: process_monitor.go for configuring thread scheduling behavior
// Impact: Controls how audio threads are scheduled by the Linux kernel
// SchedNormal defines normal (CFS) scheduling policy.
// 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
// Impact: Controls system resilience and recovery from audio process failures
// MaxRestartAttempts defines maximum number of restart attempts for failed processes.
// Used in: supervisor.go for limiting restart attempts to prevent infinite loops
// Impact: Higher values increase resilience but may mask persistent problems.
// Default 5 attempts allows recovery from transient issues while detecting persistent failures.
MaxRestartAttempts int
// RestartWindow defines time window for counting restart attempts.
// Used in: supervisor.go for restart attempt rate limiting
// Impact: Longer windows allow more restart attempts but slower failure detection.
// Default 5 minutes provides reasonable window for transient issue recovery.
RestartWindow time.Duration
// RestartDelay defines initial delay before restarting failed processes.
// Used in: supervisor.go for implementing restart backoff strategy
// Impact: Longer delays reduce restart frequency but increase recovery time.
// Default 2 seconds allows brief recovery time without excessive delay.
RestartDelay time.Duration
// MaxRestartDelay defines maximum delay between restart attempts.
// Used in: supervisor.go for capping exponential backoff delays
// Impact: Prevents excessively long delays while maintaining backoff benefits.
// Default 30 seconds caps restart delays at reasonable maximum.
MaxRestartDelay time.Duration
// Buffer Management - Memory buffer configuration for audio processing
// Used across multiple components for memory allocation and performance optimization
// Impact: Controls memory usage, allocation efficiency, and processing performance
// PreallocSize defines size of preallocated memory pools (bytes).
// Used in: buffer_pool.go for initial memory pool allocation
// Impact: Larger pools reduce allocation overhead but increase memory usage.
// Default 1MB (1024*1024) provides good balance for typical audio workloads.
PreallocSize int
// MaxPoolSize defines maximum number of objects in memory pools.
// Used in: buffer_pool.go for limiting pool growth
// Impact: Larger pools reduce allocation frequency but increase memory usage.
// Default 100 objects provides good balance between performance and memory.
MaxPoolSize int
// MessagePoolSize defines size of message object pools.
// Used in: Various IPC components for message allocation
// Impact: Larger pools reduce allocation overhead for message passing.
// Default 256 messages handles typical message throughput efficiently.
MessagePoolSize int
// OptimalSocketBuffer defines optimal socket buffer size (bytes).
// Used in: socket_buffer.go for default socket configuration
// Impact: Balances network throughput and memory usage.
// Default 262144 (256KB) provides good performance for audio streaming.
OptimalSocketBuffer int
// MaxSocketBuffer defines maximum socket buffer size (bytes).
// Used in: socket_buffer.go for high-throughput scenarios
// Impact: Larger buffers improve throughput but increase memory and latency.
// Default 1048576 (1MB) handles high-bitrate streams without excessive memory.
MaxSocketBuffer int
// MinSocketBuffer defines minimum socket buffer size (bytes).
// Used in: socket_buffer.go for memory-constrained scenarios
// Impact: Smaller buffers reduce memory but may limit throughput.
// Default 8192 (8KB) provides minimum viable buffer for audio streaming.
MinSocketBuffer int
// ChannelBufferSize defines buffer size for Go channels in audio processing.
// Used in: Various components for inter-goroutine communication
// Impact: Larger buffers reduce blocking but increase memory usage and latency.
// Default 500 messages provides good balance for audio processing pipelines.
ChannelBufferSize int
// AudioFramePoolSize defines size of audio frame object pools.
// Used in: buffer_pool.go for audio frame allocation
// Impact: Larger pools reduce allocation overhead for frame processing.
// Default 1500 frames handles typical audio frame throughput efficiently.
AudioFramePoolSize int
// PageSize defines memory page size for alignment and allocation (bytes).
// Used in: buffer_pool.go for memory-aligned allocations
// Impact: Must match system page size for optimal memory performance.
// Default 4096 bytes matches typical Linux page size.
PageSize int
// InitialBufferFrames defines initial buffer size in audio frames.
// Used in: adaptive_buffer.go for initial buffer allocation
// Impact: Larger initial buffers reduce early reallocations but increase startup memory.
// Default 500 frames provides good starting point for most audio scenarios.
InitialBufferFrames int
// BytesToMBDivisor defines divisor for converting bytes to megabytes.
// Used in: memory_metrics.go for memory usage reporting
// Impact: Must be 1024*1024 for accurate binary megabyte conversion.
// Default 1048576 (1024*1024) provides standard binary MB conversion.
BytesToMBDivisor int
// MinReadEncodeBuffer defines minimum buffer size for CGO audio read/encode (bytes).
// Used in: cgo_audio.go for native audio processing buffer allocation
// Impact: Must accommodate minimum audio frame size to prevent buffer underruns.
// Default 1276 bytes handles minimum Opus frame with safety margin.
MinReadEncodeBuffer int
// MaxDecodeWriteBuffer defines maximum buffer size for CGO audio decode/write (bytes).
// Used in: cgo_audio.go for native audio processing buffer allocation
// Impact: Must accommodate maximum audio frame size to prevent buffer overruns.
// Default 4096 bytes handles maximum audio frame size with safety margin.
MaxDecodeWriteBuffer int
// IPC Configuration - Inter-Process Communication settings for audio components
// Used in: ipc.go for configuring audio process communication
// Impact: Controls IPC reliability, performance, and protocol compliance
// MagicNumber defines magic number for IPC message validation.
// Used in: ipc.go for message header validation and protocol compliance
// Impact: Must match expected value to prevent protocol errors.
// Default 0xDEADBEEF provides distinctive pattern for message validation.
MagicNumber uint32
// MaxFrameSize defines maximum frame size for IPC messages (bytes).
// Used in: ipc.go for message size validation and buffer allocation
// Impact: Must accommodate largest expected audio frame to prevent truncation.
// Default 4096 bytes handles typical audio frames with safety margin.
MaxFrameSize int
// WriteTimeout defines timeout for IPC write operations.
// Used in: ipc.go for preventing blocking on slow IPC operations
// Impact: Shorter timeouts improve responsiveness but may cause message drops.
// Default 5 seconds allows for system load while preventing indefinite blocking.
WriteTimeout time.Duration
// MaxDroppedFrames defines maximum consecutive dropped frames before error.
// Used in: ipc.go for IPC quality monitoring
// Impact: Higher values tolerate more IPC issues but may mask problems.
// Default 10 frames allows brief interruptions while detecting serious issues.
MaxDroppedFrames int
// HeaderSize defines size of IPC message headers (bytes).
// Used in: ipc.go for message parsing and buffer allocation
// Impact: Must match actual header size to prevent parsing errors.
// Default 8 bytes matches current IPC message header format.
HeaderSize int
// Monitoring and Metrics - Configuration for audio performance monitoring
// Used in: metrics.go, latency_monitor.go for performance tracking
// Impact: Controls monitoring accuracy, overhead, and data retention
// MetricsUpdateInterval defines frequency of metrics collection and reporting.
// Used in: metrics.go for periodic metrics updates
// Impact: Shorter intervals provide more accurate monitoring but increase overhead.
// Default 1000ms (1 second) provides good balance between accuracy and performance.
MetricsUpdateInterval time.Duration
// EMAAlpha defines smoothing factor for Exponential Moving Average calculations.
// Used in: metrics.go for smoothing performance metrics
// Impact: Higher values respond faster to changes but are more sensitive to noise.
// Default 0.1 provides good smoothing while maintaining responsiveness.
EMAAlpha float64
// WarmupSamples defines number of samples to collect before reporting metrics.
// Used in: metrics.go for avoiding inaccurate initial measurements
// Impact: More samples improve initial accuracy but delay metric availability.
// Default 10 samples provides good initial accuracy without excessive delay.
WarmupSamples int
// LogThrottleInterval defines minimum interval between similar log messages.
// Used in: Various components for preventing log spam
// Impact: Longer intervals reduce log volume but may miss important events.
// Default 5 seconds prevents log flooding while maintaining visibility.
LogThrottleInterval time.Duration
// MetricsChannelBuffer defines buffer size for metrics data channels.
// Used in: metrics.go for metrics data collection pipelines
// Impact: Larger buffers reduce blocking but increase memory usage and latency.
// Default 100 metrics provides good balance for metrics collection.
MetricsChannelBuffer int
// LatencyHistorySize defines number of latency measurements to retain.
// Used in: latency_monitor.go for latency trend analysis
// Impact: More history improves trend analysis but increases memory usage.
// Default 100 measurements provides good history for analysis.
LatencyHistorySize int
// Process Monitoring Constants - System resource monitoring configuration
// Used in: process_monitor.go for monitoring CPU, memory, and system resources
// Impact: Controls resource monitoring accuracy and system compatibility
// MaxCPUPercent defines maximum valid CPU percentage value.
// Used in: process_monitor.go for CPU usage validation
// Impact: Values above this are considered invalid and filtered out.
// Default 100.0 represents 100% CPU usage as maximum valid value.
MaxCPUPercent float64
// MinCPUPercent defines minimum valid CPU percentage value.
// Used in: process_monitor.go for CPU usage validation
// Impact: Values below this are considered noise and filtered out.
// Default 0.01 (0.01%) filters out measurement noise while preserving low usage.
MinCPUPercent float64
// DefaultClockTicks defines default system clock ticks per second.
// Used in: process_monitor.go for CPU time calculations on embedded systems
// Impact: Must match system configuration for accurate CPU measurements.
// Default 250.0 matches typical embedded ARM system configuration.
DefaultClockTicks float64
// DefaultMemoryGB defines default system memory size in gigabytes.
// Used in: process_monitor.go for memory percentage calculations
// Impact: Should match actual system memory for accurate percentage calculations.
// Default 8 GB represents typical JetKVM system memory configuration.
DefaultMemoryGB int
// MaxWarmupSamples defines maximum number of warmup samples for monitoring.
// Used in: process_monitor.go for initial measurement stabilization
// Impact: More samples improve initial accuracy but delay monitoring start.
// Default 3 samples provides quick stabilization without excessive delay.
MaxWarmupSamples int
// WarmupCPUSamples defines number of CPU samples for warmup period.
// Used in: process_monitor.go for CPU measurement stabilization
// Impact: More samples improve CPU measurement accuracy during startup.
// Default 2 samples provides basic CPU measurement stabilization.
WarmupCPUSamples int
// LogThrottleIntervalSec defines log throttle interval in seconds.
// Used in: process_monitor.go for controlling monitoring log frequency
// Impact: Longer intervals reduce log volume but may miss monitoring events.
// Default 10 seconds provides reasonable monitoring log frequency.
LogThrottleIntervalSec int
// MinValidClockTicks defines minimum valid system clock ticks value.
// Used in: process_monitor.go for system clock validation
// Impact: Values below this indicate system configuration issues.
// Default 50 ticks represents minimum reasonable system clock configuration.
MinValidClockTicks int
// MaxValidClockTicks defines maximum valid system clock ticks value.
// Used in: process_monitor.go for system clock validation
// Impact: Values above this indicate system configuration issues.
// Default 1000 ticks represents maximum reasonable system clock configuration.
MaxValidClockTicks int
// Performance Tuning - Thresholds for adaptive audio quality and resource management
// Used in: adaptive_optimizer.go, quality_manager.go for performance optimization
// Impact: Controls when audio quality adjustments are triggered based on system load
// CPUThresholdLow defines CPU usage threshold for low system load.
// Used in: adaptive_optimizer.go for triggering quality improvements
// Impact: Below this threshold, audio quality can be increased safely.
// Default 20% allows quality improvements when system has spare capacity.
CPUThresholdLow float64
// CPUThresholdMedium defines CPU usage threshold for medium system load.
// Used in: adaptive_optimizer.go for maintaining current quality
// Impact: Between low and medium thresholds, quality remains stable.
// Default 60% represents balanced system load where quality should be maintained.
CPUThresholdMedium float64
// CPUThresholdHigh defines CPU usage threshold for high system load.
// Used in: adaptive_optimizer.go for triggering quality reductions
// Impact: Above this threshold, audio quality is reduced to preserve performance.
// Default 75% prevents system overload by reducing audio processing demands.
CPUThresholdHigh float64
// MemoryThresholdLow defines memory usage threshold for low memory pressure.
// Used in: adaptive_optimizer.go for memory-based quality decisions
// Impact: Below this threshold, memory-intensive audio features can be enabled.
// Default 30% allows enhanced features when memory is abundant.
MemoryThresholdLow float64
// MemoryThresholdMed defines memory usage threshold for medium memory pressure.
// Used in: adaptive_optimizer.go for balanced memory management
// Impact: Between low and medium thresholds, memory usage is monitored closely.
// Default 60% represents moderate memory pressure requiring careful management.
MemoryThresholdMed float64
// MemoryThresholdHigh defines memory usage threshold for high memory pressure.
// Used in: adaptive_optimizer.go for aggressive memory conservation
// Impact: Above this threshold, memory usage is minimized by reducing quality.
// Default 80% triggers aggressive memory conservation to prevent system issues.
MemoryThresholdHigh float64
// LatencyThresholdLow defines acceptable latency for high-quality audio.
// Used in: adaptive_optimizer.go for latency-based quality decisions
// Impact: Below this threshold, audio quality can be maximized.
// Default 20ms represents excellent latency allowing maximum quality.
LatencyThresholdLow time.Duration
// LatencyThresholdHigh defines maximum acceptable latency before quality reduction.
// Used in: adaptive_optimizer.go for preventing excessive audio delay
// Impact: Above this threshold, quality is reduced to improve latency.
// Default 50ms represents maximum acceptable latency for real-time audio.
LatencyThresholdHigh time.Duration
CPUFactor float64
MemoryFactor float64
LatencyFactor float64
InputSizeThreshold int
OutputSizeThreshold int
TargetLevel float64
// Priority Scheduling
AudioHighPriority int
AudioMediumPriority int
AudioLowPriority int
NormalPriority int
NiceValue int
// Error Handling - Configuration for error recovery and retry mechanisms
// Used in: error_handler.go, retry_manager.go for robust error handling
// Impact: Controls system resilience and recovery behavior
// MaxRetries defines maximum number of retry attempts for failed operations.
// Used in: retry_manager.go for limiting retry attempts
// Impact: More retries improve success rate but may delay error reporting.
// Default 3 retries provides good balance between persistence and responsiveness.
MaxRetries int
// RetryDelay defines initial delay between retry attempts.
// Used in: retry_manager.go for spacing retry attempts
// Impact: Longer delays reduce system load but slow recovery.
// Default 100ms provides quick retries while avoiding excessive load.
RetryDelay time.Duration
// MaxRetryDelay defines maximum delay between retry attempts.
// Used in: retry_manager.go for capping exponential backoff
// Impact: Prevents excessively long delays while maintaining backoff benefits.
// Default 5 seconds caps retry delays at reasonable maximum.
MaxRetryDelay time.Duration
// BackoffMultiplier defines multiplier for exponential backoff retry delays.
// Used in: retry_manager.go for calculating progressive retry delays
// Impact: Higher values increase delays more aggressively between retries.
// Default 2.0 doubles delay each retry, providing standard exponential backoff.
BackoffMultiplier float64
// ErrorChannelSize defines buffer size for error reporting channels.
// Used in: error_handler.go for error message queuing
// Impact: Larger buffers prevent error loss but increase memory usage.
// Default 50 errors provides adequate buffering for error bursts.
ErrorChannelSize int
MaxConsecutiveErrors int
MaxRetryAttempts int
// Timing Constants
DefaultSleepDuration time.Duration // 100ms
ShortSleepDuration time.Duration // 10ms
LongSleepDuration time.Duration // 200ms
DefaultTickerInterval time.Duration // 100ms
BufferUpdateInterval time.Duration // 500ms
StatsUpdateInterval time.Duration // 5s
SupervisorTimeout time.Duration // 10s
InputSupervisorTimeout time.Duration // 5s
ShortTimeout time.Duration // 5ms
MediumTimeout time.Duration // 50ms
BatchProcessingDelay time.Duration // 10ms
AdaptiveOptimizerStability time.Duration // 10s
MaxLatencyTarget time.Duration // 50ms
LatencyMonitorTarget time.Duration // 50ms
// Adaptive Buffer Configuration
LowCPUThreshold float64 // 20% CPU threshold
HighCPUThreshold float64 // 60% CPU threshold
LowMemoryThreshold float64 // 50% memory threshold
HighMemoryThreshold float64 // 75% memory threshold
TargetLatency time.Duration // 20ms target latency
// Adaptive Optimizer Configuration
CooldownPeriod time.Duration // 30s cooldown period
RollbackThreshold time.Duration // 300ms rollback threshold
LatencyTarget time.Duration // 50ms latency target
// Latency Monitor Configuration
MaxLatencyThreshold time.Duration // 200ms max latency
JitterThreshold time.Duration // 20ms jitter threshold
// Microphone Contention Configuration
MicContentionTimeout time.Duration // 200ms contention timeout
// Priority Scheduler Configuration
MinNiceValue int // -20 minimum nice value
MaxNiceValue int // 19 maximum nice value
// Buffer Pool Configuration
PreallocPercentage int // 20% preallocation percentage
InputPreallocPercentage int // 30% input preallocation percentage
// Exponential Moving Average Configuration
HistoricalWeight float64 // 70% historical weight
CurrentWeight float64 // 30% current weight
// Sleep and Backoff Configuration
CGOSleepMicroseconds int // 50000 microseconds (50ms)
BackoffStart time.Duration // 50ms initial backoff
// Protocol Magic Numbers
InputMagicNumber uint32 // 0x4A4B4D49 "JKMI" (JetKVM Microphone Input)
OutputMagicNumber uint32 // 0x4A4B4F55 "JKOU" (JetKVM Output)
// Calculation Constants
PercentageMultiplier float64 // 100.0 for percentage calculations
AveragingWeight float64 // 0.7 for weighted averaging calculations
ScalingFactor float64 // 1.5 for scaling calculations
SmoothingFactor float64 // 0.3 for adaptive buffer smoothing
CPUMemoryWeight float64 // 0.5 for CPU factor in combined calculations
MemoryWeight float64 // 0.3 for memory factor in combined calculations
LatencyWeight float64 // 0.2 for latency factor in combined calculations
PoolGrowthMultiplier int // 2x growth multiplier for pool sizes
LatencyScalingFactor float64 // 2.0 for latency ratio scaling
OptimizerAggressiveness float64 // 0.7 for optimizer aggressiveness
}
// DefaultAudioConfig returns the default configuration constants
// These values are carefully chosen based on JetKVM's embedded ARM environment,
// real-time audio requirements, and extensive testing for optimal performance.
func DefaultAudioConfig() *AudioConfigConstants {
return &AudioConfigConstants{
// Audio Quality Presets
// Rationale: 4096 bytes accommodates largest expected audio frames with safety margin
// for high-quality audio while preventing buffer overruns on embedded systems
MaxAudioFrameSize: 4096,
// Opus Encoding Parameters
// Rationale: 128kbps provides excellent quality for KVM audio while maintaining
// reasonable bandwidth usage over network connections
OpusBitrate: 128000,
// Rationale: Maximum complexity (10) ensures best quality encoding, acceptable
// on modern ARM processors with sufficient CPU headroom
OpusComplexity: 10,
// Rationale: VBR enabled (1) optimizes bitrate based on audio content complexity,
// reducing bandwidth for simple audio while maintaining quality for complex audio
OpusVBR: 1,
// Rationale: Unconstrained VBR (0) allows maximum bitrate flexibility for
// optimal quality-bandwidth balance in varying network conditions
OpusVBRConstraint: 0,
// Rationale: DTX disabled (0) ensures consistent audio stream for KVM applications
// where silence detection might interfere with system audio monitoring
OpusDTX: 0,
// Audio Parameters
// Rationale: 48kHz sample rate is professional audio standard, provides full
// frequency range (up to 24kHz) for accurate system audio reproduction
SampleRate: 48000,
// Rationale: Stereo (2 channels) captures full system audio including spatial
// information and stereo effects common in modern operating systems
Channels: 2,
// Rationale: 960 samples = 20ms frames at 48kHz, optimal balance between
// latency (low enough for real-time) and efficiency (reduces packet overhead)
FrameSize: 960,
// Rationale: 4000 bytes accommodates compressed Opus frames with overhead,
// prevents packet fragmentation while allowing for quality variations
MaxPacketSize: 4000,
// Audio Quality Bitrates (kbps)
// Rationale: Low quality optimized for bandwidth-constrained connections,
// output higher than input as system audio typically more complex than microphone
AudioQualityLowOutputBitrate: 32,
AudioQualityLowInputBitrate: 16,
// Rationale: Medium quality balances bandwidth and quality for typical usage,
// suitable for most KVM scenarios with reasonable network connections
AudioQualityMediumOutputBitrate: 64,
AudioQualityMediumInputBitrate: 32,
// Rationale: High quality for excellent audio fidelity, matches default Opus
// bitrate for professional applications requiring clear audio reproduction
AudioQualityHighOutputBitrate: 128,
AudioQualityHighInputBitrate: 64,
// Rationale: Ultra quality for audiophile-grade reproduction, suitable for
// high-bandwidth connections where audio quality is paramount
AudioQualityUltraOutputBitrate: 192,
AudioQualityUltraInputBitrate: 96,
// Audio Quality Sample Rates (Hz)
// Rationale: 22.05kHz captures frequencies up to 11kHz, sufficient for speech
// and basic audio while minimizing processing load on constrained connections
AudioQualityLowSampleRate: 22050,
// Rationale: 44.1kHz CD-quality standard, captures full audible range up to
// 22kHz, excellent balance of quality and processing requirements
AudioQualityMediumSampleRate: 44100,
// Rationale: 16kHz optimized for voice/microphone input, captures speech
// frequencies (300-8000Hz) efficiently while reducing bandwidth
AudioQualityMicLowSampleRate: 16000,
// Audio Quality Frame Sizes
// Rationale: 40ms frames reduce processing overhead for low-quality mode,
// acceptable latency increase for bandwidth-constrained scenarios
AudioQualityLowFrameSize: 40 * time.Millisecond,
// Rationale: 20ms frames provide good balance of latency and efficiency,
// standard for real-time audio applications
AudioQualityMediumFrameSize: 20 * time.Millisecond,
AudioQualityHighFrameSize: 20 * time.Millisecond,
// Rationale: 10ms frames minimize latency for ultra-responsive audio,
// suitable for applications requiring immediate audio feedback
AudioQualityUltraFrameSize: 10 * time.Millisecond,
// Audio Quality Channels
// Rationale: Mono (1 channel) reduces bandwidth by 50% for low-quality mode,
// acceptable for basic audio monitoring where stereo separation not critical
AudioQualityLowChannels: 1,
// Rationale: Stereo (2 channels) preserves spatial audio information for
// medium/high/ultra quality modes, essential for modern system audio
AudioQualityMediumChannels: 2,
AudioQualityHighChannels: 2,
AudioQualityUltraChannels: 2,
// CGO Audio Constants
// Rationale: 96kbps provides good quality for CGO layer while being more
// conservative than main Opus bitrate, suitable for embedded processing
CGOOpusBitrate: 96000,
// Rationale: Lower complexity (3) reduces CPU load in CGO layer while
// maintaining acceptable quality for real-time processing
CGOOpusComplexity: 3,
// Rationale: VBR enabled (1) allows bitrate adaptation based on content
// complexity, optimizing bandwidth usage in CGO processing
CGOOpusVBR: 1,
// Rationale: Constrained VBR (1) limits bitrate variations for more
// predictable processing load in embedded environment
CGOOpusVBRConstraint: 1,
// Rationale: OPUS_SIGNAL_MUSIC (3) optimizes encoding for general audio
// content including system sounds, music, and mixed audio
CGOOpusSignalType: 3, // OPUS_SIGNAL_MUSIC
// Rationale: OPUS_BANDWIDTH_FULLBAND (1105) enables full 20kHz bandwidth
// for complete audio spectrum reproduction
CGOOpusBandwidth: 1105, // OPUS_BANDWIDTH_FULLBAND
// Rationale: DTX disabled (0) ensures consistent audio stream without
// silence detection that could interfere with system audio monitoring
CGOOpusDTX: 0,
// Rationale: 48kHz sample rate matches main audio parameters for
// consistency and professional audio quality
CGOSampleRate: 48000,
// Rationale: Stereo (2 channels) maintains spatial audio information
// throughout the CGO processing pipeline
CGOChannels: 2,
// Rationale: 960 samples (20ms at 48kHz) matches main frame size for
// consistent timing and efficient processing
CGOFrameSize: 960,
// Rationale: 1500 bytes accommodates Ethernet MTU constraints while
// providing sufficient space for compressed audio packets
CGOMaxPacketSize: 1500,
// Input IPC Constants
// Rationale: 48kHz sample rate ensures high-quality microphone input
// capture matching system audio output for consistent quality
InputIPCSampleRate: 48000,
// Rationale: Stereo (2 channels) captures spatial microphone information
// and maintains compatibility with stereo input devices
InputIPCChannels: 2,
// Rationale: 960 samples (20ms) provides optimal balance between latency
// and processing efficiency for real-time microphone input
InputIPCFrameSize: 960,
// Output IPC Constants
// Rationale: 4096 bytes accommodates largest audio frames with safety
// margin, preventing buffer overruns in IPC communication
OutputMaxFrameSize: 4096,
// Rationale: 10ms timeout provides quick response for real-time audio
// while preventing blocking in high-performance scenarios
OutputWriteTimeout: 10 * time.Millisecond,
// Rationale: Allow up to 50 dropped frames before error handling,
// providing resilience against temporary network/processing issues
OutputMaxDroppedFrames: 50,
// Rationale: 17-byte header provides sufficient space for frame metadata
// including timestamps, sequence numbers, and format information
OutputHeaderSize: 17,
// Rationale: 128 message pool size balances memory usage with throughput
// for efficient audio streaming without excessive buffering
OutputMessagePoolSize: 128,
// Socket Buffer Constants
// Rationale: 128KB optimal buffer provides good balance between memory
// usage and network throughput for typical audio streaming scenarios
SocketOptimalBuffer: 131072, // 128KB
// Rationale: 256KB maximum buffer accommodates burst traffic and high
// bitrate audio while preventing excessive memory consumption
SocketMaxBuffer: 262144, // 256KB
// Rationale: 32KB minimum buffer ensures adequate buffering for basic
// audio streaming while minimizing memory footprint
SocketMinBuffer: 32768, // 32KB
// Scheduling Policy Constants
// Rationale: SCHED_NORMAL (0) provides standard time-sharing scheduling
// suitable for non-critical audio processing tasks
SchedNormal: 0,
// Rationale: SCHED_FIFO (1) provides real-time first-in-first-out
// scheduling for critical audio processing requiring deterministic timing
SchedFIFO: 1,
// Rationale: SCHED_RR (2) provides real-time round-robin scheduling
// for balanced real-time processing with time slicing
SchedRR: 2,
// Real-time Priority Levels
// Rationale: Priority 80 ensures audio processing gets highest CPU
// priority for latency-critical operations without starving system
RTAudioHighPriority: 80,
// Rationale: Priority 60 provides elevated priority for important
// audio tasks while allowing higher priority system operations
RTAudioMediumPriority: 60,
// Rationale: Priority 40 provides moderate real-time priority for
// audio tasks that need responsiveness but aren't latency-critical
RTAudioLowPriority: 40,
// Rationale: Priority 0 represents normal scheduling priority for
// non-real-time audio processing tasks
RTNormalPriority: 0,
// Process Management
// Rationale: 5 restart attempts provides resilience against transient
// failures while preventing infinite restart loops
MaxRestartAttempts: 5,
// Rationale: 5-minute restart window allows recovery from temporary
// issues while resetting attempt counter for long-term stability
RestartWindow: 5 * time.Minute,
// Rationale: 1-second initial restart delay prevents rapid restart
// cycles while allowing quick recovery from brief failures
RestartDelay: 1 * time.Second,
// Rationale: 30-second maximum delay prevents excessive wait times
// while implementing exponential backoff for persistent failures
MaxRestartDelay: 30 * time.Second,
// Buffer Management
// Rationale: 1MB preallocation provides substantial buffer space for
// high-throughput audio processing while remaining reasonable for embedded systems
PreallocSize: 1024 * 1024, // 1MB
// Rationale: 100 pool size limits memory usage while providing adequate
// object pooling for efficient memory management
MaxPoolSize: 100,
// Rationale: 256 message pool size balances memory usage with message
// throughput for efficient IPC communication
MessagePoolSize: 256,
// Rationale: 256KB optimal socket buffer provides good network performance
// for audio streaming without excessive memory consumption
OptimalSocketBuffer: 262144, // 256KB
// Rationale: 1MB maximum socket buffer accommodates burst traffic and
// high-bitrate audio while preventing excessive memory usage
MaxSocketBuffer: 1048576, // 1MB
// Rationale: 8KB minimum socket buffer ensures basic network buffering
// while minimizing memory footprint for low-bandwidth scenarios
MinSocketBuffer: 8192, // 8KB
// Rationale: 500 channel buffer size provides adequate buffering for
// inter-goroutine communication without blocking
ChannelBufferSize: 500, // Channel buffer size for processing
// Rationale: 1500 audio frame pool size accommodates frame reuse for
// efficient memory management in high-throughput scenarios
AudioFramePoolSize: 1500, // Audio frame pool size
// Rationale: 4096-byte page size aligns with system memory pages for
// optimal memory allocation and cache performance
PageSize: 4096, // Memory page size
// Rationale: 500 initial buffer frames provides adequate startup buffering
// without excessive memory allocation during initialization
InitialBufferFrames: 500, // Initial buffer size in frames
// Rationale: 1024*1024 divisor provides standard MB conversion for
// memory usage calculations and reporting
BytesToMBDivisor: 1024 * 1024, // Divisor for converting bytes to MB
// Rationale: 1276 bytes minimum buffer accommodates smallest CGO audio
// read/encode operations while ensuring adequate processing space
MinReadEncodeBuffer: 1276, // Minimum buffer size for CGO audio read/encode
// Rationale: 4096 bytes maximum buffer provides sufficient space for
// largest CGO audio decode/write operations without excessive allocation
MaxDecodeWriteBuffer: 4096, // Maximum buffer size for CGO audio decode/write
// IPC Configuration
// Rationale: 0xDEADBEEF magic number provides distinctive header for
// IPC message validation and debugging purposes
MagicNumber: 0xDEADBEEF,
// Rationale: 4096 bytes maximum frame size accommodates largest audio
// frames while preventing excessive memory allocation
MaxFrameSize: 4096,
// Rationale: 5-second write timeout provides reasonable wait time for
// IPC operations while preventing indefinite blocking
WriteTimeout: 5 * time.Second,
// Rationale: Allow up to 10 dropped frames before error handling,
// balancing audio continuity with quality maintenance
MaxDroppedFrames: 10,
// Rationale: 8-byte header provides sufficient space for basic IPC
// metadata including message type and size information
HeaderSize: 8,
// Monitoring and Metrics
// Rationale: 1-second metrics update interval provides timely monitoring
// without excessive overhead for performance tracking
MetricsUpdateInterval: 1000 * time.Millisecond,
// Rationale: 0.1 EMA alpha provides smooth exponential moving average
// with 90% weight on historical data for stable metrics
EMAAlpha: 0.1,
// Rationale: 10 warmup samples allow metrics to stabilize before
// making optimization decisions based on performance data
WarmupSamples: 10,
// Rationale: 5-second log throttle interval prevents log spam while
// ensuring important events are captured for debugging
LogThrottleInterval: 5 * time.Second,
// Rationale: 100 metrics channel buffer provides adequate buffering
// for metrics collection without blocking performance monitoring
MetricsChannelBuffer: 100,
// Rationale: 100 latency measurements provide sufficient history for
// statistical analysis and trend detection
LatencyHistorySize: 100, // Number of latency measurements to keep
// Process Monitoring Constants
MaxCPUPercent: 100.0, // Maximum CPU percentage
MinCPUPercent: 0.01, // Minimum CPU percentage
DefaultClockTicks: 250.0, // Default clock ticks for embedded ARM systems
DefaultMemoryGB: 8, // Default memory in GB
MaxWarmupSamples: 3, // Maximum warmup samples
WarmupCPUSamples: 2, // CPU warmup samples
LogThrottleIntervalSec: 10, // Log throttle interval in seconds
MinValidClockTicks: 50, // Minimum valid clock ticks
MaxValidClockTicks: 1000, // Maximum valid clock ticks
// Performance Tuning
CPUThresholdLow: 0.20,
CPUThresholdMedium: 0.60,
CPUThresholdHigh: 0.75,
MemoryThresholdLow: 0.30,
MemoryThresholdMed: 0.60,
MemoryThresholdHigh: 0.80,
LatencyThresholdLow: 20 * time.Millisecond,
LatencyThresholdHigh: 50 * time.Millisecond,
CPUFactor: 0.7,
MemoryFactor: 0.8,
LatencyFactor: 0.9,
InputSizeThreshold: 1024,
OutputSizeThreshold: 2048,
TargetLevel: 0.5,
// Priority Scheduling
AudioHighPriority: -10,
AudioMediumPriority: -5,
AudioLowPriority: 0,
NormalPriority: 0,
NiceValue: -10,
// Error Handling
MaxRetries: 3,
RetryDelay: 100 * time.Millisecond,
MaxRetryDelay: 5 * time.Second,
BackoffMultiplier: 2.0,
ErrorChannelSize: 50,
MaxConsecutiveErrors: 5,
MaxRetryAttempts: 3,
// Timing Constants
DefaultSleepDuration: 100 * time.Millisecond,
ShortSleepDuration: 10 * time.Millisecond,
LongSleepDuration: 200 * time.Millisecond,
DefaultTickerInterval: 100 * time.Millisecond,
BufferUpdateInterval: 500 * time.Millisecond,
StatsUpdateInterval: 5 * time.Second,
SupervisorTimeout: 10 * time.Second,
InputSupervisorTimeout: 5 * time.Second,
ShortTimeout: 5 * time.Millisecond,
MediumTimeout: 50 * time.Millisecond,
BatchProcessingDelay: 10 * time.Millisecond,
AdaptiveOptimizerStability: 10 * time.Second,
MaxLatencyTarget: 50 * time.Millisecond,
LatencyMonitorTarget: 50 * time.Millisecond,
// Adaptive Buffer Configuration
LowCPUThreshold: 0.20,
HighCPUThreshold: 0.60,
LowMemoryThreshold: 0.50,
HighMemoryThreshold: 0.75,
TargetLatency: 20 * time.Millisecond,
// Adaptive Optimizer Configuration
CooldownPeriod: 30 * time.Second,
RollbackThreshold: 300 * time.Millisecond,
LatencyTarget: 50 * time.Millisecond,
// Latency Monitor Configuration
MaxLatencyThreshold: 200 * time.Millisecond,
JitterThreshold: 20 * time.Millisecond,
// Microphone Contention Configuration
MicContentionTimeout: 200 * time.Millisecond,
// Priority Scheduler Configuration
MinNiceValue: -20,
MaxNiceValue: 19,
// Buffer Pool Configuration
PreallocPercentage: 20,
InputPreallocPercentage: 30,
// Exponential Moving Average Configuration
HistoricalWeight: 0.70,
CurrentWeight: 0.30,
// Sleep and Backoff Configuration
CGOSleepMicroseconds: 50000,
BackoffStart: 50 * time.Millisecond,
// Protocol Magic Numbers
InputMagicNumber: 0x4A4B4D49, // "JKMI" (JetKVM Microphone Input)
OutputMagicNumber: 0x4A4B4F55, // "JKOU" (JetKVM Output)
// Calculation Constants - Mathematical constants used throughout audio processing
// Used in: Various components for calculations and conversions
// Impact: Controls calculation accuracy and algorithm behavior
// PercentageMultiplier defines multiplier for percentage calculations.
// Used in: Throughout codebase for converting ratios to percentages
// Impact: Must be 100 for standard percentage calculations.
// Default 100 provides standard percentage conversion (0.5 * 100 = 50%).
PercentageMultiplier: 100.0, // For percentage calculations
// AveragingWeight defines weight for weighted averaging calculations.
// Used in: metrics.go, adaptive_optimizer.go for smoothing values
// Impact: Higher values give more weight to recent measurements.
// Default 0.7 (70%) emphasizes recent values while maintaining stability.
AveragingWeight: 0.7, // For weighted averaging calculations
// ScalingFactor defines general scaling factor for various calculations.
// Used in: adaptive_optimizer.go, quality_manager.go for scaling adjustments
// Impact: Controls magnitude of adaptive adjustments and scaling operations.
// Default 1.5 provides moderate scaling for quality and performance adjustments.
ScalingFactor: 1.5, // For scaling calculations
SmoothingFactor: 0.3, // For adaptive buffer smoothing
CPUMemoryWeight: 0.5, // CPU factor weight in combined calculations
MemoryWeight: 0.3, // Memory factor weight in combined calculations
LatencyWeight: 0.2, // Latency factor weight in combined calculations
PoolGrowthMultiplier: 2, // Pool growth multiplier
LatencyScalingFactor: 2.0, // Latency ratio scaling factor
OptimizerAggressiveness: 0.7, // Optimizer aggressiveness factor
}
}
// Global configuration instance
var audioConfigInstance = DefaultAudioConfig()
// UpdateConfig allows runtime configuration updates
func UpdateConfig(newConfig *AudioConfigConstants) {
audioConfigInstance = newConfig
}
// GetConfig returns the current configuration
func GetConfig() *AudioConfigConstants {
return audioConfigInstance
}
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