LinFIR Documentation

Welcome to the official LinFIR documentation. LinFIR is a professional tool for designing FIR filters for loudspeaker crossovers and frequency response corrections.

Quick Navigation

• New to LinFIR? Start with the Overview
• Setting up audio? See Audio Setup & Measurements
• Designing filters? Check Driver Processing
• Need help? Visit Troubleshooting

About This Documentation

This documentation covers all aspects of LinFIR, from basic setup to advanced directivity analysis. Use the sidebar to navigate through different topics, or use the search function (top-left) to find specific information.

Accessing Documentation

You can access this documentation in several ways:

• From within LinFIR: Select Help > Documentation from the menu bar or press H from any window
• Online: Visit https://demaudio.com/doc/linfir/ for the latest version
• Offline: LinFIR automatically serves embedded documentation locally when no internet connection is available

When you access documentation from within LinFIR, the application will:

1. First attempt to open the online version
2. If the online version is unreachable (no internet or timeout after 3 seconds), automatically fall back to the embedded local documentation server
3. The local server runs on port 3030+ and serves the documentation directly from the application

This ensures you always have access to documentation, whether online or offline.

Safari Configuration (Mac OS)

⚠️ Important for Safari users on Mac OS: The local documentation server uses HTTP (not HTTPS) for localhost connections. To access the embedded local documentation in Safari, you need to:

1. Open Safari > Settings (or Safari > Preferences)
2. Go to the Security tab
3. Disable the “Non-secure site connections” option

This allows Safari to connect to the local HTTP server running on http://localhost:3030+.

Alternative: If you prefer to keep Safari’s security settings unchanged, the online documentation at https://demaudio.com/doc/linfir/ is always available and uses HTTPS.

LinFIR - Precision FIR Filter Design for Speaker Crossovers and Corrections
Visit https://linfir.demaudio.com for updates and support

Overview

LinFIR is a professional audio software tool designed for precise FIR (Finite Impulse Response) and IIR (Infinite Impulse Reponse) filter design, tailored specifically for loudspeaker crossovers and frequency response correction. The application provides comprehensive control over both magnitude and phase response characteristics, enabling engineers and enthusiasts to craft high-quality audio filter solutions.

Design Philosophy

LinFIR is designed to be both a filter creation tool and a DSP-faithful system simulator. It shows what the filters will actually do when applied to your measurements, not an idealized version.

What You See Is What You Get: LinFIR operates with real DSP constraints—impulse timing, causality, delay, phase rotation, and resampling effects all matter. The application does not silently recenter impulses or adjust phase for prettier graphs. What is displayed is what a real DSP would produce.

This approach requires some understanding of signal processing fundamentals, but LinFIR automates most technical decisions under the hood to keep the workflow practical. The trade-off is that results may sometimes look less “clean” than in tools that abstract away these constraints, but they are physically accurate and predictable.

Different by Design: LinFIR does not replicate the workflows or interface conventions of other tools. The internal DSP-like architecture drives both the workflow and presentation. This is intentional, the goal is to bring a time-domain perspective into DIY loudspeaker and room-correction work, not to replace existing approaches.

Understanding what LinFIR shows and why helps avoid confusion and makes the tool more effective for its intended purpose.

Dual Operating Modes

LinFIR offers two distinct modes of operation, each optimized for different workflows:

Loudspeaker Design Mode

This mode is dedicated to designing and optimizing multi-way loudspeaker systems. Key applications include:

• Crossover design with multiple filter types
• Individual driver frequency response correction
• Phase and time alignment between drivers
• Directivity pattern analysis and optimization
• Complete system integration and tuning

Room Calibration Mode

Focused on in-room acoustic measurements and correction:

• Multiple measurement position capture and averaging
• Automatic temporal alignment using GCC-PHAT algorithm
• Room response correction filter generation
• Spatially-averaged frequency response optimization
• Integration with existing room correction workflows

The mode is selected when creating a new project and remains fixed throughout the project’s lifetime. To switch modes, simply create a new project with the desired configuration.

Core Features

FIR Filter Design

LinFIR implements sophisticated FIR filtering with:

• Configurable filter length: From 32 to 65536 taps for precise frequency resolution
• Causality control: Continuous adjustment from pure linear-phase (0.0) to minimum-phase (1.0)
• Multiple filter types: Brickwall (Sinc), Linkwitz-Riley, Butterworth, and Bessel characteristics
• Kaiser window shaping: Fine-tune transition band characteristics

Frequency Response Correction

Advanced magnitude and phase correction capabilities:

• Target curve options: Flat response, Harman in-room curve, or custom user-defined targets
• Independent correction modes: Separate control for magnitude and phase correction
• Frequency range limiting: Apply corrections only where needed
• Maximum attenuation control: Prevent over-correction of deep nulls

IIR Filtering

Complement FIR filters with cascaded IIR sections:

• Parametric EQ: Precise peak/notch filters for resonance control
• Shelving filters: Low-shelf and high-shelf for tonal balance
• Crossover filters: Butterworth, Linkwitz-Riley, and Bessel topologies
• All-pass filters: Dedicated phase shaping without magnitude changes
• Auto-EQ: Automatic optimization to target curves with configurable parameters

Measurement System

Built-in exponential sine sweep (ESS, Farina method) for high-quality impulse response capture:

• Harmonic distortion separation: Automatic extraction of H2-H4 through deterministic time positioning
• Driver protection: 4th-order Butterworth high-pass filter with configurable cutoff (-3 dB at fc, -24 dB/octave)
• Configurable sweep parameters: Duration, amplitude, frequency range (starting from 1 Hz)
• Quality validation: Automatic rejection of poor captures (clipping, low SNR)
• Microphone calibration: Import manufacturer calibration files for accurate measurements

Workflow Integration

LinFIR seamlessly integrates into professional audio workflows:

• Import/export WAV, txt, and specialized formats
• Direct export to Powersoft Armonia processors
• Export FIR coefficients in binary, text, WAV or CSV format
• Hypex FA Series amplifier configuration (HFD format)
• CamillaDSP filter export
• Comprehensive project management with auto-save

Platform Support

• Mac OS: Full feature support with low-latency audio
• Windows: Complete functionality with platform-specific audio optimizations
• Cross-platform projects: Projects are fully compatible across platforms

Project Modes

LinFIR offers three specialized operating modes, each designed for specific audio engineering workflows. The mode is selected during project creation and determines the available features and interface elements throughout the project’s lifetime.

Note on Hypex FusionAmp mode: The Hypex FusionAmp mode is a derivative of Loudspeaker Design mode. Everything documented for Loudspeaker Design — driver processing, crossover design, FIR/IIR filtering, measurement import, directivity analysis — applies equally to Hypex FusionAmp projects. The mode adds hardware-specific UI constraints (locked sample rate, fixed channel count, FIR tap limits, IIR biquad limit) that prevent invalid configurations and eliminate the need to manually track hardware limits during the design process.

Loudspeaker Design Mode

Purpose

This mode is optimized for designing and analyzing multi-way loudspeaker systems. It provides comprehensive tools for crossover design, driver integration, and directivity analysis.

Key Features

• Multi-driver support: Design systems with multiple drivers (subwoofers, woofers, midranges, tweeters)
• Crossover: FIR and IIR crossover filters with various types
• Driver correction: Individual magnitude and phase correction for each driver
• Directivity tools: Analyze off-axis response and polar patterns (license required)
• Complete export: Export individual driver filters, global filters, and HFD configurations

Typical Workflow

1. Create a new Loudspeaker Design project
2. Import or measure impulse responses for each driver
3. Design crossover filters (low-pass, high-pass)
4. Apply frequency response corrections
5. Analyze summed system response
6. Export filters for DSP implementation

When to Use

• Designing passive loudspeaker conversions to active DSP
• Optimizing existing multi-way systems
• Analyzing driver interactions and phase relationships
• Creating custom crossover
• Performing anechoic or quasi-anechoic measurements

Room Calibration Mode

Purpose

Dedicated to in-room acoustic measurements and correction filter generation. This mode focuses on capturing multiple measurement positions, aligning them temporally, and creating averaged correction filters.

Key Features

• Multiple measurement positions: Capture IRs at different listening locations
• Automatic temporal alignment: GCC-PHAT algorithm aligns measurements
• Spatial averaging: Creates averaged response across measurement positions
• Global correction only: Simplified interface focused on room correction
• Streamlined export: Export only global correction filters

Typical Workflow

1. Create a new Room Calibration project
2. Configure sweep output channel
3. Capture measurements at 3-5 different listening positions
4. Measurements are automatically aligned using GCC-PHAT
5. Design global correction filters (FIR and/or IIR)
6. Export correction filters for room EQ implementation

UI Adaptations

When in Room Calibration mode, the interface adapts to focus on relevant features:

• Disabled: Directivity analysis tools (not applicable to room measurements)
• Simplified: Filter graphs show only global filters
• Restricted: Export options limited to global correction filters
• Hidden: Individual driver processing controls

Export Restrictions

Room Calibration projects export only:

• Global FIR correction filter (if enabled)
• Global IIR filters (Manual or Auto-EQ)

The following exports are disabled:

• HFD config export
• Detailed reports (TXT/PDF)
• Individual driver filters

This ensures clean, focused output for room correction workflows.

When to Use

• Correcting in-room frequency response
• Creating stereo-linked or mono room correction
• Working with existing loudspeaker systems
• Integrating with convolution engines

Hypex FusionAmp Mode

⚠️ License required: Creating Hypex FusionAmp projects requires a valid LinFIR license. See License for activation details.

Purpose

Hypex FusionAmp mode is a derivative of Loudspeaker Design mode tailored specifically for Hypex FusionAmp series amplifiers (FA122, FA123, FA251, FA252, FA253, FA501, FA502, FA503). All Loudspeaker Design features are available — driver processing, crossover design, FIR/IIR filtering, measurement import, directivity analysis — but several parameters are locked to match the DSP capabilities of the target hardware.

The goal is to eliminate manual bookkeeping: instead of counting biquads, tracking tap budgets, or checking compatibility after the fact, the UI enforces hardware limits in real time so the resulting configuration is always valid and ready to export.

Available Models

LinFIR supports all FusionAmp models:

• FA122: 2-channel amplifier (2 × 125W @ 4Ω)
• FA123: 3-channel amplifier (2 × 125W + 100W @ 4Ω)
• FA251: 1-channel amplifier (1 × 250W @ 4Ω)
• FA252: 2-channel amplifier (2 × 250W @ 4Ω)
• FA253: 2-channel amplifier (2 × 250W + 100W @ 4Ω)
• FA501: 1-channel amplifier (1 × 500W @ 4Ω)
• FA502: 2-channel amplifier (2 × 500W @ 4Ω)
• FA503: 3-channel amplifier (2 × 500W + 100W @ 4Ω)

Hardware Constraints

When in Hypex FusionAmp mode, several parameters are locked to match hardware specifications:

• Sample Rate: Fixed at 93.75 kHz (cannot be changed)
• Channel Count: Fixed by model (1, 2 or 3 channels, cannot add/remove drivers)
• IIR Filters: Maximum 15 biquads per channel
• FIR Filters: Fixed total tap count depends on FIR position (see below)

FIR Processing Modes

FusionAmp mode offers two mutually exclusive FIR processing configurations:

FIR IN (Input FIR)

Global FIR correction applied at the DSP input stage before channel processing.

• Location: Before IIR filters and channel routing
• Total taps: Fixed at 4500 (filter taps + padding)
• Use case: Global room correction, global speaker compensation
• Constraint: FIR Taps + Export padding = 4500 (always)
• UI behavior: Per-driver FIR controls are hidden; adjusting taps automatically adjusts padding to maintain 4500 total

FIR OUT (Output FIR)

FIR correction applied at the output stage, after IIR processing.

• Location: After IIR filters, per-channel or global
• Total taps: Fixed at 1500 per channel (filter taps + padding)
• Use case: Individual driver correction, per-channel equalization
• Constraint: Filter length (taps) + Export padding = 1500 (always, per driver or global)
• UI behavior: Adjusting taps automatically adjusts padding to maintain 1500 total

IIR Filter Constraints

FusionAmp DSP limits each channel to 15 biquads maximum. LinFIR enforces this through:

• Add Filter button: Automatically disabled when at 15 biquads
• Filter type restrictions: Types exceeding the limit are grayed out with tooltips
• Order limitations: Filter order sliders dynamically limited based on remaining capacity
• Auto EQ constraints: “Max Filters” parameter accounts for locked filters’ biquad usage

Note: For details on how biquad counts are calculated for different filter types, see the IIR Filtering section.

Key Features

• Hardware-matched UI: Interface adapts to show only applicable controls
• Automatic validation: Prevents configurations exceeding hardware limits
• Export compatibility: Direct HFD export for FusionAmp amplifiers
• Constraint warnings: Visual indicators when approaching limits

Typical Workflow

1. Create new project: File → New Project → Hypex FusionAmp
2. Select model (FA122, FA123, FA251, FA252, FA253, FA501, FA502, FA503)
3. Choose FIR position (Input or Output)
4. Import/measure impulse responses for each channel
5. Design IIR filters (monitor biquad counter to stay within 15 biquad limit)
6. Configure FIR correction (taps + padding always equals 4500 for IN or 1500 for OUT)
7. Export HFD configuration file for amplifier

UI Adaptations

The interface automatically adjusts based on FIR position:

FIR IN mode:

• Global FIR section visible and active
• Per-driver FIR controls hidden
• Taps and padding controls linked to maintain 4500 total
• Adjusting taps automatically recalculates padding, and vice versa

FIR OUT mode with per-driver FIR:

• Per-driver FIR controls visible
• Each driver has linked taps/padding controls maintaining 1500 total
• Global FIR section hidden or disabled

FIR OUT mode without per-driver FIR:

• Taps and padding controls linked to maintain 1500 total
• Per-driver FIR controls hidden

Export Configuration

Hypex FusionAmp projects support:

• HFD export: Native configuration format for FusionAmp amplifiers
• FIR filter export: Individual channel FIR filters
• IIR filter export: Biquad coefficients per channel

When to Use

• Configuring Hypex FusionAmp series amplifiers
• Ensuring DSP configuration fits hardware constraints
• Exporting ready-to-use HFD configuration files
• Working within strict real-time processing limits

Limitations

⚠️ Hardware constraints cannot be bypassed:

• Sample rate is locked at 93.75 kHz
• Channel count is fixed by model (1, 2, or 3 channels)
• FIR total tap count is fixed (4500 for IN, 1500 for OUT) - taps and padding sum must always equal this value
• IIR biquad limit (15 per channel) is strictly enforced
• Cannot use both FIR IN and FIR OUT simultaneously

Mode Selection

Creating a New Project

Project mode is selected via File → New Project:

1. Click “New Project”
2. Choose between “Loudspeaker Design”, “Room Calibration”, or “Hypex FusionAmp”
3. For Hypex FusionAmp: Select model (FA122/FA123/FA251/FA252/FA253/FA501/FA502/FA503) and FIR position (IN/OUT)
4. For other modes: Configure initial project settings (sample rate, filter length, etc.)
5. Begin working in the selected mode

Mode Permanence

⚠️ Important: Once a project is created, its mode cannot be changed. The mode is permanently associated with the project file.

To work in a different mode:

1. Save your current project (if needed)
2. Create a new project with the desired mode
3. Import measurements or data as required

Choosing the Right Mode

Use Loudspeaker Design Mode when:

• Designing crossovers for multi-driver systems
• Analyzing individual driver characteristics
• Performing directivity analysis
• Working with anechoic or quasi-anechoic data
• Need flexible configuration options

Use Room Calibration Mode when:

• Correcting in-room frequency response
• Creating averaged room correction filters
• Working with existing complete loudspeaker systems
• Focusing on global system correction only

Use Hypex FusionAmp Mode when: (license required)

• Configuring Hypex FusionAmp series amplifiers
• Exporting HFD configuration files
• Need to ensure configurations match hardware limits (93.75 kHz, fixed taps, 15 biquads)

Best Practices

Room Calibration Mode

• Measurement count: Capture 3-5 measurements at different positions
• Position spacing: Keep positions within 30-50 cm of main listening area
• Height consistency: Use consistent microphone height across measurements
• Reference position: First measurement should be at primary listening position
• Correction philosophy: Apply gentle correction, avoid over-equalization
• Deep nulls: Don’t attempt to fill room mode nulls below 300 Hz
• Phase type: Consider minimum-phase FIR for reduced latency
• Acoustic treatment: Combine with room treatment for best results

Loudspeaker Design Mode

• Measurement quality: Use anechoic or quasi-anechoic measurements when possible
• Windowing: Gate reflections using IR time windowing
• Alignment: Align drivers using time delay controls, not FIR compensation delay
• Crossover design: Start with appropriate crossover frequencies and filter slopes
• Phase analysis: Monitor phase relationships between drivers
• Directivity: Capture multiple angles for comprehensive analysis (license required)

Mode Comparison Table

Hypex FusionAmp is a superset of Loudspeaker Design: all ✅ features from Loudspeaker Design are available, with the hardware constraints in the last rows added on top.

Graph Interaction & Display

This section covers all interactive features, display controls, and analysis tools available when working with LinFIR’s graphs.

Graph Toolbar

The graph toolbar provides comprehensive controls for selecting which data to display and how to visualize it. The toolbar adapts based on the project mode (Loudspeaker Design vs Room Calibration).

Loudspeaker Design Mode:

Room Calibration Mode:

Display Mode Selection

Purpose: Choose which stage of the signal processing chain to visualize.

Available modes:

• FIR - FIR filter responses only (crossovers, magnitude/phase correction)
• IIR - IIR filter responses only (parametric EQ, shelves, crossovers)
• FIR+IIR - Combined FIR and IIR responses
• Drivers (Loudspeaker Design) / Measurements (Room Calibration) - Full signal chain applied to driver or room measurements (theoretical system response)

Keyboard shortcuts:

• Press F to cycle through filter modes (FIR → IIR → FIR+IIR)
• Press D to switch to Drivers/Measurements mode

Mode behavior:

• Loudspeaker Design mode: Shows individual driver filters plus global filters
• Room Calibration mode: Shows only global filters in FIR/IIR/FIR+IIR modes

See Display Modes section below for detailed explanations of each mode.

Graph Visibility Toggles — “Graphs” Dropdown

Purpose: Show or hide specific graph types.

Graph types are grouped in the Graphs dropdown in the toolbar. Opening the dropdown reveals toggles for each graph type:

• Magnitude (M) - Frequency response in dB
• Impulse Response (I) - Impulse response in time domain
• Step Response (T) - Step response in time domain
• Group Delay (G) - Group delay in milliseconds
• Phase (P) - Phase response in degrees
• Harmonic Distortion (K) - (Loudspeaker Design / Hypex mode only)

Letters in parentheses indicate keyboard shortcuts. The dropdown stays open when toggling items, allowing multiple graphs to be shown or hidden without reopening it.

Note: Harmonic Distortion is only available in Loudspeaker Design / Hypex mode with Drivers display mode active, as harmonic distortion analysis is not applicable to in-room measurements.

License Feature: The HD graph displays the total harmonic distortion for all users. Individual harmonic curves (H2, H3, H4) are only available with a valid license.

Angle Selector (Loudspeaker Design / Hypex mode only)

Purpose: Select which off-axis measurement angle to display when drivers have directivity data.

• Horizontal angles: Typically ±15°, ±30°, ±45°, ±60°, ±75°, ±90°
• Vertical angles: Same as horizontal
• Toggle H/V: Switch between horizontal and vertical axis selection

Only available when at least one driver has off-axis measurements. The selector is hidden in Room Calibration mode as directivity analysis is specific to anechoic/quasi-anechoic loudspeaker measurements.

See Directivity Analysis for complete details on polar measurements and analysis.

Time Domain Options — “Time Domain” Dropdown

The Time Domain dropdown groups options that affect impulse and step response display.

Normalize

Purpose: Scale impulse and step responses to ±1 range for easier comparison.

When enabled:

• Impulse responses normalized to peak amplitude = 1
• Step responses normalized to maximum value = 1
• Useful for comparing relative timing and shape without amplitude differences

Focus on Summed Response

Purpose: Use the time boundaries of the summed impulse response for all IR and step response plots.

• Loudspeaker Design: All time-domain plots are framed around the summed driver output
• Individual curves remain visible within those boundaries
• Useful for evaluating overall system timing alignment

Phase Options — “Phase Options” Dropdown

The Phase Options dropdown is enabled only when the Phase graph is visible. It contains two options:

Unwrap Phase

Purpose: Display continuous phase response without ±180° wrapping discontinuities.

• Disabled: Phase wraps at ±180° (sawtooth pattern)
• Enabled: Phase continues beyond ±180° showing true accumulated phase shift
• Essential for analyzing linear phase filters and group delay
• Makes phase response easier to interpret across wide frequency ranges

Remove time of flight rotations

What is a time of flight? A pure propagation delay — the time it takes for sound to travel from the driver to the measurement microphone — appears in the phase response as a constant linear slope: the further the impulse response peak is from t = 0, the steeper the slope, and the more the phase accumulates rotations across the frequency range. This linear ramp carries no useful information about the filter or driver behaviour; it just reflects the physical distance between source and mic.

This option removes that linear component, leaving only the non-linear phase rotations introduced by the filters and the acoustics of each driver. The result is a much flatter, easier-to-read phase curve, and — when multiple drivers are shown — it becomes straightforward to compare their phase alignment without the bulk delay dominating the picture.

• Disabled: Phase includes the full linear delay component (steep slope proportional to time of flight)
• Enabled: Linear component subtracted, phase flattens around 0° — non-linear deviations stand out clearly

HD Display Mode (Loudspeaker Design mode only)

Purpose: Select how harmonic distortion values are displayed on the HD plot.

Available display modes:

• Percent (%) - Traditional percentage representation
• dB (relative) - Relative level compared to fundamental
• dB (absolute) - Absolute RMS level of harmonics

Percent (%) Mode:

Displays harmonic distortion as a percentage relative to the total RMS (including fundamental and harmonics).

dB (relative) Mode:

Expresses harmonic distortion relative to the total RMS in decibels. Negative values indicate harmonics are quieter than the total signal. For example:

• -40 dB ≈ 1% distortion
• -60 dB ≈ 0.1% distortion
• -80 dB ≈ 0.01% distortion

Useful for evaluating signal-to-distortion ratio.

dB (absolute) Mode:

Shows the absolute level of the harmonics alone, without normalization. This mode is useful for comparing distortion levels across different SPL measurements, as it shows the actual acoustic power in harmonic components regardless of the fundamental level.

Individual Harmonics:

All three modes also display individual harmonic curves (H2, H3, H4) with dashed/dotted line styles. The selected mode applies to both the total HD curve and individual harmonics.

License Feature: Individual harmonic curves (H2-H4) are only available with a valid license.

Smoothing

Purpose: Apply smoothing to magnitude, HD, directivity, group delay and phase plots.

Smoothing options:

• None - No smoothing (raw response)
• 1/48 octave - Very fine smoothing
• 1/24 octave - Fine smoothing
• 1/12 octave - Moderate smoothing
• 1/6 octave - Coarse smoothing
• 1/3 octave - Very coarse smoothing
• ERB - Perceptually-motivated variable smoothing (see below)

Applies to:

• Magnitude (frequency response)
• Group delay
• Phase (preserves driver alignment via trend removal method)
• HD (Harmonic Distortion)
• Directivity sonograms

Does not affect: Time-domain plots (impulse, step response)

ERB Smoothing:

ERB (Equivalent Rectangular Bandwidth) smoothing uses a variable bandwidth that corresponds to the frequency resolution of the human auditory system. The smoothing bandwidth follows the formula: (107.77f + 24.673) Hz, where f is frequency in kHz.

This results in:

• Heavy smoothing at low frequencies (~1 octave at 50Hz, 1/2 octave at 100Hz, 1/3 octave at 200Hz)
• Moderate smoothing at mid frequencies (approximately 1/6 octave above 1kHz)
• Natural perceptual weighting that reflects how the ear integrates frequency information

ERB smoothing is particularly useful for:

• Understanding what a listener actually perceives
• Comparing measurements to subjective listening impressions
• Evaluating whether fine response details are audible

Note: Smoothing is cosmetic and does not affect filter calculations or exports. See Fractional Octave Smoothing section below for detailed behavior.

Mouse & Trackpad Controls

Zoom Controls

Zoom In - Multiple methods:

• Box zoom: Right-click and drag to select a region
• Trackpad pinch: Pinch gesture to zoom in/out
• Mouse wheel: Scroll to move the view within a graph

Zoom Out:

• Double-click anywhere on a graph to reset to auto-calculated bounds
• Double-click re-enables auto-bounds mode (graphs automatically adjust to data range)

Pan

Left-click and drag to move the view within a zoomed graph.

Legend Interaction

Click on color dots in the legend to show/hide individual curves:

• Hidden curves are grayed out in the legend
• Click again to show the curve
• Useful for isolating specific drivers or filters

Detach Graphs

Right-click on any graph to open it in a separate window:

• All graph types supported (Frequency, Phase, Group Delay, Impulse, Step, HD)
• Detached windows render in real-time following the active display mode
• Multiple graphs can be opened simultaneously for side-by-side comparison
• Ideal for multi-monitor setups and detailed analysis workflows
• Close detached windows: Cmd+W (macOS) / Ctrl+W (Windows)

Display Modes

Project Mode Impact

LinFIR supports two project modes that affect display terminology:

• Loudspeaker Design - Speaker design with drivers
• Room Calibration - Multi-position measurements for room correction

In Room Calibration mode:

• “Drivers” mode is renamed to “Measurements”
• Individual measurement curves are shown in Measurements mode
• Filter modes (FIR/IIR/FIR+IIR) display only global filters

In Loudspeaker Design mode:

• “Drivers” mode displays individual drivers
• Filter modes show both per-driver filters and global filters

Filters vs. Drivers/Measurements Mode

Filters Mode (Keyboard: F):

• Shows individual filter responses (FIR, IIR, or FIR+IIR combined)
• Press F multiple times to cycle through filter submodes:
  • FIR Filters - Shows only FIR filter responses
  • IIR Filters - Shows only IIR filter responses
  • FIR+IIR Filters - Shows combined FIR and IIR responses
• Loudspeaker Design: Per-driver crossover and correction filters displayed separately
• Room Calibration: Only global filters are shown
• Global filters shown as additional curves
• Useful for analyzing filter design and frequency response shaping

Drivers/Measurements Mode (Keyboard: D):

• Loudspeaker Design: Shows combined driver + filter responses (acoustic output)
  • Each curve represents the complete signal chain for that driver
  • Sum curve shows the total system response at listening position
• Room Calibration: Shows individual measurement positions and averaged response
  • Each curve represents a raw measurement location (no filters applied)
  • Sum curve shows the averaged response with global filters applied
• Useful for analyzing final acoustic performance

Graph Visibility Toggles

Show/hide individual graph types using keyboard shortcuts:

• M - Toggle Magnitude (Frequency Response) plot
• P - Toggle Phase Response plot
• G - Toggle Group Delay plot
• I - Toggle Impulse Response plot
• T - Toggle Step Response plot
• K - Toggle HD (Harmonic Distortion) plot

X-Axis Synchronization

Toggle: X key or toolbar button

When enabled, Sync X-axis synchronizes the horizontal axis across related plots:

Frequency Plots (Magnitude, Phase, Group Delay, Directivity Sonograms):

• All plots share the same frequency range
• Zooming any plot updates all frequency plots simultaneously
• Double-click resets all frequency plots to full range

Time Plots (Impulse, Step Response):

• All time-domain plots share the same time range
• Zooming any plot updates all time plots simultaneously
• Double-click resets all time plots to full range

Independent Y-Axes:

• Each plot maintains its own vertical scale
• Y-axis auto-bounds independently for each graph type

Use Cases:

• Compare magnitude and phase behavior at the same frequency
• Analyze impulse and step response in the same time window
• Maintain consistent zoom across multiple graph types

Impulse & Step Response Normalization

Toggle: N key or toolbar button

Normalizes impulse and step responses to peak amplitude = 1.0:

Purpose:

• Allows visual comparison between responses with different amplitudes
• Focuses on filter shape and time-domain characteristics
• Eliminates gain differences for easier visual analysis

Effect:

• All IR and SR curves are scaled to the same peak height
• Does not affect magnitude or phase plots
• Purely visual - does not modify underlying data

Use Cases:

• Compare filter shapes between drivers with different sensitivity
• Analyze time-domain behavior without gain differences obscuring details
• Identify pre-ringing or timing issues across multiple drivers

Focus on Summed Response

Toggle: S key or Time Domain dropdown (Drivers/Measurements mode only)

Synchronizes the time window of all temporal plots (Impulse & Step Response) to the Sum/Average curve boundaries:

How it Works:

1. Analyzes the Sum/Average impulse response to find its energy boundaries
2. Detects first and last samples above -60 dB threshold (1/1000th of peak)
3. Adds configurable margins before and after the detected energy region
4. Applies the same time window to all temporal plots (both Impulse and Step Response)
5. Enables auto-bounds for all time plots

Behavior:

• Requires at least 2 active curves to activate (otherwise ignored)
• Focuses all temporal plots on the main acoustic energy of the Sum/Average response
• Individual driver/measurement curves are zoomed to the same time window as the sum
• Easier to compare arrival times and transient behavior across all curves
• Works with adaptive display mode margins (configurable in Settings)

Use Cases:

• Loudspeaker Design: Quickly identify timing alignment issues across drivers
  • See if all drivers arrive within the sum’s main energy window
  • Detect delays or misalignment relative to system response
• Room Calibration: Focus on the averaged room response energy region
  • Compare individual measurement positions to the average response timing
  • Identify room reflections within the main energy window
• Eliminate clutter from pre-ringing or tail noise outside the main energy region

Only available in Drivers/Measurements Mode (not applicable to individual filters).

Note: Works in both Loudspeaker Design and Room Calibration project modes when display mode is set to Drivers/Measurements.

Phase Display Options

Unwrap Phase

Toggle: U key or toolbar button

Controls phase display format:

Wrapped Phase (default):

• Phase values constrained to ±180°
• Shows discontinuities (phase wraps) at ±180° boundaries
• Easier to read for simple filters
• Standard display format for most audio applications

Unwrapped Phase:

• Continuous phase values extending beyond ±180°
• Shows total accumulated phase shift
• Useful for analyzing group delay trends and overall phase shift

Note: Group delay is always computed from unwrapped phase internally. The Unwrap toggle only affects phase display, not group delay accuracy.

Remove time of flight rotations

Toggle: C key or Phase Options dropdown (next to Unwrap Phase)

A pure propagation delay translates into a linear slope in the phase response — the further the impulse response peak is from t = 0, the steeper the slope, and the more rotations accumulate across the frequency range. This linear ramp is entirely determined by the physical distance between the driver and the microphone and does not reveal anything about filter behaviour or driver characteristics.

This option removes that linear component via a linear regression on the IR peak position, leaving only the non-linear phase introduced by the filters and acoustics. Phase curves become much flatter, easier to read, and directly comparable across drivers regardless of their physical placement.

Removes the linear phase component (constant group delay) from phase responses:

Algorithm:

• Computes linear phase slope from IR peak position for all curves (drivers and filters)
• Subtracts the constant group delay (linear phase shift)
• Flattens phase around 0° while preserving relative phase differences
• Works with both wrapped and unwrapped phase display modes

In Loudspeaker Design Mode (Drivers):

When multiple speakers are active:

• Removes the linear delay of the Sum curve from all responses
• Centers all driver phases around 0° relative to the system response
• Highlights phase alignment issues between drivers
• Makes phase differences visible without bulk delay obscuring them

When single driver is active:

• Removes the linear component of that driver’s phase
• Flattens phase around 0° to show only non-linear phase shifts

In Room Calibration Mode (Measurements):

• Independent correction: Removes the linear phase component of each individual measurement independently
• No shared reference - each measurement position is treated separately
• Flattens each measurement’s phase around 0°
• Useful for comparing phase behavior across different room positions without timing offsets
• The averaged curve also has its own linear component removed

In Filters Mode (FIR, IIR, FIR+IIR):

• Per-filter correction: Removes the linear phase component of each individual filter
• Flattens each filter’s phase around 0°
• Reveals only non-linear phase rotations introduced by the filter
• Useful for analyzing filter phase behavior independent of bulk delay

Use Cases:

• Identify phase alignment issues between drivers without delay obscuring differences
• Visualize phase rotations introduced by filters (independent of linear delay)
• Compare filter phase characteristics across different designs
• Verify minimum-phase vs. linear-phase behavior

Fractional Octave Smoothing

Control: Dropdown menu in toolbar (None, 1/48 oct to 1/3 oct, or ERB)

Apply smoothing to reduce measurement noise and reveal trends:

Fixed Fractional-Octave Smoothing:

• No smoothing - Show raw response (no averaging)
• 1/48 octave - Very fine smoothing (highest detail)
• 1/24 octave - Fine smoothing
• 1/12 octave - Moderate smoothing (default)
• 1/6 octave - Coarse smoothing
• 1/3 octave - Very coarse smoothing (lowest detail)

ERB (Equivalent Rectangular Bandwidth) Smoothing:

ERB smoothing uses a frequency-dependent bandwidth that models the human auditory system’s frequency resolution. Unlike fixed fractional-octave smoothing, ERB adapts its bandwidth at each frequency according to psychoacoustic research.

Bandwidth formula: (107.77f + 24.673) Hz, where f is in kHz

Frequency-dependent behavior:

• 50 Hz → ~1 octave bandwidth (heavy smoothing)
• 100 Hz → ~1/2 octave bandwidth
• 200 Hz → ~1/3 octave bandwidth
• 1 kHz and above → ~1/6 octave bandwidth (leveling out)

When to use ERB smoothing:

• To see what the ear actually perceives (perceptually accurate)
• When evaluating audibility of response features
• To match listening test results with measured data
• For publication-quality graphs that reflect human hearing

When to use fixed fractional-octave smoothing:

• For equalization and correction work (1/6 or 1/12 octave recommended)
• When analyzing crossover behavior in detail
• To maintain consistent resolution across all frequencies
• For technical comparisons between drivers

Effect:

• Wider smoothing = smoother curves, less detail
• Narrower smoothing = more detail, more noise visible
• ERB = perceptually-weighted smoothing (variable with frequency)

Applies to:

• Frequency response (magnitude), HD, Directivity Sonograms, group delay and phase plots
• Does not affect time-domain plots (impulse responses)

Note: Smoothing is applied after all processing (filters, crossovers, correction). It affects display only, not exported data. It can also alter crossover slopes on the graphs.

Impulse Response Analysis

Automatic Pre-Ringing Detection

When hovering over impulse response curves, LinFIR displays additional analysis information:

Pre-ring Level (in dB relative to peak):

• Weighted RMS measurement of signal energy before the main peak
• Analysis stops at the last zero-crossing before peak (excludes rising edge)
• Uses Hilbert envelope to capture perceptible energy regardless of phase
• Perceptually weighted: energy closer to the peak is less penalized (temporal masking)
• Displayed as: Pre-ring: -50.0 dB (Likely masked)

Algorithm Details:

1. Find the impulse peak position and amplitude
2. Find the last zero-crossing before the peak (to exclude the rising edge)
3. Extract the pre-ringing region (from start to zero-crossing)
4. Calculate the Hilbert envelope of the pre-ringing region
5. Apply sigmoid weighting (closer to peak = less perceptual penalty)
   • Masking time at 50% penalty: 0.6 ms (default)
   • Weight floor: 0.2 (minimum weight for distant pre-ring)
   • Weight ceiling: 1.0 (maximum weight for near-peak pre-ring)
6. Calculate weighted RMS of the envelope
7. Convert to dB relative to peak amplitude

Weighting Rationale:

• Pre-ringing close to the peak is masked by the main transient (temporal masking)
• Pre-ringing far from the peak is more audible (less masking)
• Sigmoid function models gradual transition from masked to unmasked

Risk Assessment Labels

Pre-ringing levels are classified based on typical perceptual thresholds (indicative):

Important Notes:

• These are indicative estimates, not absolute rules
• Actual audibility depends on:
  • Program material (transient-rich vs. sustained tones)
  • Listening level
  • Individual hearing sensitivity
  • Room acoustics and background noise
• Use as a guideline for filter design, not strict limits
• Shorter FIR filters or increased causality reduce pre-ringing at the cost of magnitude and phase accuracy

Trade-offs:

• Longer FIR filters → More pre-ringing, better magnitude/phase accuracy
• Shorter FIR filters → Less pre-ringing, reduced correction capability
• Higher causality (e.g., minimum-phase) → Less pre-ringing, potential phase distortion
• Linear-phase (causality = 0) → Maximum pre-ringing, linear phase

Auto-Bounds Behavior

LinFIR automatically adjusts graph bounds to fit data:

Auto-Bounds Enabled (default):

• Graphs automatically scale to show all data
• Y-axis adjusts when switching between drivers or display modes
• X-axis adjusts based on data range (frequency or time)

Auto-Bounds Disabled (after user interaction):

• Manual zoom/pan disables auto-bounds for that axis
• Graph retains user-specified bounds
• Switching modes preserves manual bounds
• Double-click re-enables auto-bounds and resets to calculated range

Per-Axis Independence:

• X-axis and Y-axis auto-bounds are independent
• Dragging vertically disables Y auto-bounds, preserves X auto-bounds
• Dragging horizontally disables X auto-bounds, preserves Y auto-bounds
• Box zoom (right-click drag) disables both X and Y auto-bounds

Per-Graph Independence:

• Each graph type has independent auto-bounds state
• Zooming Frequency plot does not affect Impulse plot
• Switching between display modes preserves per-graph bounds

Keyboard Shortcuts Summary

Display Modes

• F - Switch to Filters display mode
• D - Switch to Drivers/Measurements display mode

Display Controls

• X - Toggle X-axis synchronization
• N - Toggle normalize impulse and step responses
• U - Toggle unwrap/wrap phase response
• C - Toggle time of flight rotation removal
• S - Toggle focus on summed impulse response (Drivers/Measurements mode only)
• R - Toggle Directivity Sonogram window (Loudspeaker Design mode only, requires license)

Graph Visibility

• M - Toggle Magnitude Response plot
• P - Toggle Phase Response plot
• G - Toggle Group Delay plot
• I - Toggle Impulse Response plot
• T - Toggle Step Response plot
• K - Toggle HD (Harmonic Distortion) plot

Related Documentation

• Settings - Graph resolution, titles, colors
• Keyboard Shortcuts - Complete shortcut reference
• Driver Processing - Filters affecting graph display
• System Processing - Global filters and sum calculation

Keyboard Shortcuts

This is a comprehensive reference of all keyboard shortcuts available in LinFIR.

Note: On macOS, use Cmd (⌘). On Windows/Linux, use Ctrl.

File Operations

Context:

• Works in main window and all detached windows
• Disabled when text input fields have focus (to avoid conflicts)
• Auto-save runs every 60 seconds in the background (if enabled)

Window Operations

Context:

• Cmd+W works in detached graph windows, driver IR windows, HFD export window, etc.
• Cmd+, is a system-wide shortcut to access application settings
• H toggles documentation visibility globally

Display Modes

Behavior:

• D: Shows acoustic output (driver impulse + all filters applied)
• F: Shows individual filter responses
  • First press: FIR filters only
  • Second press: IIR filters only
  • Third press: FIR+IIR combined
  • Fourth press: Back to FIR filters

Context: Only active when no text input has focus.

Display Controls

Details:

• X (Sync X-Axis):

  • Frequency plots (Magnitude, Phase, Group Delay) share X-axis
  • Time plots (Impulse, Step) share X-axis
  • Y-axes remain independent

• N (Normalize):

  • Visual only - does not affect exported data
  • Useful for comparing filter shapes with different gains

• U (Unwrap Phase):

  • Wrapped: phase constrained to ±180° (discontinuities)
  • Unwrapped: continuous phase beyond ±180° (smooth)

• C (Remove time of flight rotations):

  • Removes linear phase (constant group delay) via linear regression
  • Flattens phase around 0° to show only non-linear behavior
  • In Drivers mode: removes Sum curve delay from all responses
  • In Filters mode: per-filter linear phase removal

• S (Focus on Summed Response):

  • Only available in Drivers mode
  • Calculates time boundaries from summed impulse
  • Auto-enables auto-bounds for all time plots

Context: Only active when no text input has focus.

Graph Visibility Toggles

Behavior:

• Each toggle is independent
• State persists across sessions (saved in project)

Context: Only active when no text input has focus.

Advanced Features

Requirements:

• R only works in Loudspeaker Design mode (not Room Calibration)
• Requires valid license
• Opens directivity analysis window with frequency-angle sonograms

Context: Only active when no text input has focus.

Undo/Redo

Behavior:

• Full project state snapshots
• Limited undo history to 10 snapshots (constrained by memory)
• Works in main window and all viewport windows (Driver IR, HFD export, etc.)

Context: Disabled when text input fields have focus.

Dialog-Specific Shortcuts

Shortcut Contexts

LinFIR disables shortcuts based on context to avoid conflicts:

When Text Input Has Focus

The following shortcuts are disabled when typing in text fields (IR names, filter frequencies, etc.):

• All single-letter shortcuts (D, F, M, K, G, I, T, P, U, C, X, S, N, R, H)
• Undo/Redo (Cmd+Z, Cmd+Shift+Z)
• Save/Export (Cmd+S, Cmd+E)
• Settings (Cmd+,)

Still active:

• Close window (Cmd+W)

When Any Input Window is Open

Certain shortcuts are disabled when modal dialogs or input windows are open:

• Save (Cmd+S)
• Export (Cmd+E)

This prevents accidental saves while configuring settings or importing files.

Platform-Specific Notes

macOS

• Use Cmd (⌘) for all modifier shortcuts
• Cmd+W closes windows but not the main application window
• Cmd+Q quits the application (system shortcut)
• Cmd+, is the standard macOS shortcut for Preferences/Settings

Windows/Linux

• Use Ctrl for all modifier shortcuts
• Ctrl+W closes windows
• Alt+F4 quits the application (system shortcut)

Quick Reference Card

File

• Cmd+S - Save
• Cmd+E - Export Project

Window

• Cmd+W - Close Window
• Cmd+, - Settings
• H - Documentation

Display

• D - Drivers Mode
• F - Cycle Filters Mode
• X - Sync X-Axis
• N - Normalize IR/SR
• U - Unwrap Phase
• C - Remove time of flight rotations
• S - Focus on Summed Response

Graphs

• M - Magnitude
• P - Phase
• G - Group Delay
• I - Impulse
• T - Step
• K - Harmonic Distortion

Advanced

• R - Directivity Sonogram (license required)

Edit

• Cmd+Z - Undo
• Cmd+Shift+Z - Redo

Related Documentation

• Graph Interaction - Mouse controls, zoom, pan, detach graphs
• Settings - Configure default display modes and preferences
• Driver Processing - Filter controls affected by shortcuts
• System Processing - Global filters and display modes
• Directivity Analysis - Directivity sonogram window (R key)

General Settings

Appearance

Customize the visual appearance of the LinFIR interface.

UI Color Scheme

Choose between two color themes for LinFIR’s interface elements:

• Red (default): Custom red accent colors for headers, titles, and active buttons
• Blue: Standard egui theme with blue accents

What changes:

• Section headers (Parameters, Drivers, Global Filters)
• Graph titles (Frequency Responses, Impulse Responses, etc.)
• Active button colors and selection highlights
• Accent colors throughout the interface

Application: Changes apply immediately without requiring restart.

Theme (Light/Dark Mode)

Switch between light mode and dark mode using the system theme preference buttons.

• Light mode: Bright background, dark text
• Dark mode: Dark background, light text
• Follow system: Automatically matches your operating system theme preference

Filter Processing

Settings that affect how filters are computed and validated.

Clipping Detection

Default: Enabled
Threshold: 0 dB (configurable from −5 to +2 dB)

When enabled, LinFIR analyzes filter chains to detect potential clipping and suggests gain adjustments.

How it works:

• Generates a deterministic periodic pink-noise test signal covering 20 Hz to 20 kHz
  • Spectral slope: −3 dB/octave (represents typical programme material)
  • Band-shaped: 12 dB/oct high-pass at 20 Hz, 24 dB/oct low-pass at 20 kHz
  • RMS ≈ −24 dBFS, peak = −6 dBFS (crest factor ≈ 12 dB)
• Applies all active filters to the test signal
• Warns when filter output exceeds the configured threshold
• Suggests gain reduction to bring peak below 0.95 × threshold (leaving a small safety margin)

Threshold setting: The threshold controls the detection sensitivity relative to digital full scale (0 dBFS):

• 0 dB (default): warn when the filter output reaches or exceeds full scale — the standard safety level
• Negative values (−1 to −5 dB): conservative — trigger the warning before reaching full scale, leaving extra headroom.
• Positive values (+1, +2 dB): lenient — only warn when the output is significantly above full scale. Use with caution: values above 0 dB allow actual clipping to go unreported

⚠️ Setting the threshold above 0 dB means clipping can occur without any warning.

Important clarifications:

⚠️ Measurement levels are not tested - Clipping warnings are purely about filter processing, not your imported measurements. The test signal is synthetic and independent of measurement amplitude.

⚠️ Phase processing can cause clipping even with flat magnitude responses - Filters with visually reasonable frequency responses can produce temporal peaks exceeding 0 dBFS. This is especially common with:

• Phase correction filters combined with other filters (without crossovers)
• All-pass filters that only modify phase
• Multiple FIR filters with different causality settings

Why this happens: Phase distortion (or correction) shifts frequency components in time, causing them to align constructively at certain moments. Classic example: a square wave with magnitude 1.0 can exceed 1.0 after phase processing, even though the Fourier magnitude spectrum remains unchanged.

Why disable:

• May improve performance on slower systems
• Removes safety warnings if you’re confident in your gain staging

⚠️ Warning: Disabling removes safety checks. Always test with your actual audio content when clipping detection is disabled.

Warn when phase FIR guard fails

Default: Enabled

When enabled, LinFIR shows a warning toast if the phase correction auto-correction guard exhausted all its attempts without producing a FIR with flat magnitude response.

The auto-correction guard is always active. It automatically raises the minimum/maximum correction frequency (f_min/f_max) in 1/3-octave steps — up to 100 iterations — until the FIR’s magnitude deviation falls within the per-filter tolerance threshold. If every attempt still exceeds the threshold, the guard gives up and (if this warning is enabled) shows a toast.

To resolve the underlying issue, try:

• Reducing the phase correction amount
• Reducing the Kaiser β parameter (high β clips filter oscillations and directly causes deviations)
• Increasing the tap count
• Increasing the per-filter tolerance threshold (“Guard Tolerance” in the phase correction filter settings)

The per-filter Guard Tolerance parameter (0.1–15 dB, default 0.5 dB) is set directly in each driver’s Phase Correction settings and in the Global FIR Phase Correction settings.

Disabling this setting

• Silences guard-failure toasts only — the guard still runs silently.
• May be useful in advanced workflows where you consciously accept the trade-off.

⚠️ Warning: Disabling hides failures silently. Only disable if you understand the implications.

Auto Causal Alignment

Default: Enabled

Automatically optimizes FIR impulse positioning when using multiple FIR filters with causality, based on logarithmic energy distribution analysis.

How it works:

1. Computes logarithmic energy for each sample
2. Shifts log values to positive range while preserving relative distribution
3. Calculates the log-energy ratio before and after the impulse peak
4. Positions the peak at ratio × n_taps within the FIR window
   • Example: 30% log-energy before peak → peak positioned at 30% of FIR taps
   • Example: 50% log-energy before peak → peak centered at 50% of FIR taps

Why logarithmic energy:

• Gives more weight to weak signal components (pre-ringing, post-ringing) relative to the dominant peak
• Compresses dynamic range to better detect energy asymmetry in the impulse response
• More robust than simple amplitude threshold methods with complex filter combinations

Activation conditions:

• At least 2 FIR filters are active (correction, low-pass, high-pass, or phase filters)
• At least one filter has non-zero causality (> 0.0)

Why it’s useful:

• Prevents truncation of both pre-ringing and post-ringing artifacts
• Adapts positioning to actual signal energy distribution
• Handles asymmetric impulse responses from causal filter combinations
• Maximizes useful signal capture in the final cropped FIR window

When to disable:

• Manual control over impulse positioning is preferred using custom FIR Compensation Delay adjustments
• ⚠️ Only disable if you understand signal processing and impulse alignment implications - incorrect manual alignment can cause truncation artifacts or loss of filter effectiveness

Show Manual FIR Delay Compensation

Default: Disabled (hidden)
Type: ⚙️ Advanced Setting

Controls visibility of manual FIR delay compensation fields in both driver FIR filters and global FIR correction window.

What it shows:

• FIR Offset Delay field in each driver’s FIR filter section
• FIR compensation delay field in the global FIR correction window

Purpose: These fields allow fine-tuning of FIR filter impulse alignment by manually adjusting delay compensation. This is primarily useful when:

• Auto Causal Alignment is disabled and manual control is needed
• Fine-tuning filter timing when FIR ringing characteristics permit adjustments

⚠️ Warning - Advanced Users Only:

This is an advanced feature requiring solid understanding of:

• Digital signal processing fundamentals
• FIR filter behavior and causality
• Impulse response alignment and windowing
• Phase relationships in multi-way crossover systems

Incorrect use can result in:

• Filter truncation and loss of effectiveness
• Audible artifacts (ripples in frequency response)
• Degraded time-domain response

Recommendation: Keep this option disabled unless you have sufficient confidence in your signal processing knowledge. In most cases, use:

• Auto Causal Alignment for automatic optimal positioning
• Time Delay adjustments for primary timing alignment between drivers
• Delay Compensation (in IR Management) for microphone positioning corrections

Default Values for New Projects

Configure default settings that apply when creating new projects. These do not affect existing projects.

Sample Rate

Options: 44.1 kHz, 48 kHz, 88.2 kHz, 93.75 kHz, 96 kHz, 176.4 kHz, 192 kHz
Default: 48 kHz

Default sampling frequency for new projects.

• 44.1 kHz: Compact DSP platforms with limited processing power
• 48 kHz: Professional audio standard, most DSP platforms
• 88.2 kHz: High-resolution audio (2× CD rate, relatively uncommon DSP sampling frequency)
• 93.75 kHz: Hypex DSP platforms (FA123, FA253, FA502, etc.)
• 96 kHz: High-resolution audio
• 176.4/192 kHz: Ultra-high resolution - only use when DSP platform specifically requires it (higher CPU load, larger filter sizes)

FIR length (taps)

Range: 32 to 65536 taps
Default: 512 taps

Default number of taps for individual driver FIR filters (low-pass, high-pass, and correction filters).

• 512-2048 taps: Sufficient for most crossover and correction applications
• 4096 taps: Typical maximum needed for standard loudspeaker design
• 8192-16384 taps: Specialized use for subwoofer phase and magnitude correction in very low frequencies
• >16384 taps: Rarely needed - primarily for extreme low-frequency correction or research purposes

Trade-offs:

• More taps = better frequency resolution and steeper slopes
• More taps = higher latency (taps/2 samples for linear-phase filters)
• More taps = larger file sizes and more DSP processing required

Global FIR Taps

Range: 32 to 65536 taps
Default: 512 taps

Default number of taps for the global FIR correction filter (applied to the summed system response).

Same considerations as FIR length (taps) apply.

Curve Smoothing

Options: None, 1/48 octave, 1/24 octave, 1/12 octave, 1/6 octave, 1/3 octave
Default: None

Default smoothing applied to frequency response curves in plots.

• No smoothing: Shows raw response with all detail
• 1/48 to 1/24 octave: Subtle smoothing, reveals trends while keeping detail
• 1/12 octave: Good balance between detail and readability
• 1/6 to 1/3 octave: Heavy smoothing, shows overall trends only

Adaptive Crossover Smoothing

Options: Enabled, Disabled
Default: Enabled

Controls whether smoothing is adaptively reduced near crossover frequencies to preserve slope visibility.

When enabled:

• Smoothing progressively reduces within ±0.5 octave of detected crossover frequencies (LP/HP filters)
• Reaches minimum smoothing (1/96 octave) at the crossover frequency itself
• Helps visualize the actual filter slopes while maintaining smooth curves elsewhere
• Only affects magnitude display curves - does not modify phase, group delay, or filter generation

When disabled:

• Uniform smoothing is applied across the entire frequency range
• Provides consistency with external measurement tools that use constant smoothing

Note: External measurements with uniform smoothing will show gentler crossover slopes compared to LinFIR’s adaptive mode. This is expected behavior - adaptive smoothing reveals the true filter response near crossovers.

Graph Display Mode

Options: FIR, IIR, FIR+IIR, Drivers/Measurements
Default: FIR

Default graph display mode for new projects.

• FIR: Shows FIR filter responses only
• IIR: Shows IIR filter responses only
• FIR+IIR: Shows combined FIR and IIR responses
• Drivers/Measurements: Shows complete processed driver/measurement responses

Keyboard shortcuts: Press F (Filters mode) or D (Drivers mode) to switch quickly.

Phase Display

Options: Unwrapped, Wrapped
Default: Wrapped

Default phase display mode for new projects.

• Wrapped: Phase values constrained to ±180°, with discontinuities at boundaries
• Unwrapped: Continuous phase values extending beyond ±180°

Note: Group delay is always computed from unwrapped phase; this setting only affects the phase graph display.

Keyboard shortcut: Press U to toggle unwrap/wrap mode.

Remove time of flight rotations

Options: Enabled, Disabled
Default: Disabled

Removes the linear phase component (constant group delay) from phase responses to flatten phase around 0°.

How it works:

• Loudspeaker Design: Removes linear delay of Sum curve from all driver responses (or driver’s own delay for single driver)
• Room Calibration: Removes linear phase component of each individual measurement independently (no shared reference)
• Filters Mode: Removes linear phase component of each individual filter

Use cases:

• Loudspeaker Design: Identify phase alignment issues between drivers relative to system response
• Room Calibration: Compare phase behavior across room positions without timing offsets

Keyboard shortcut: Press C to toggle time of flight rotation removal.

Reset Application Settings

⚠️ Warning: This action cannot be undone!

Completely reset all LinFIR settings to factory defaults.

What gets reset:

• All application settings and preferences
• Application cache and UI state
• Default values for new projects
• Audio device selections
• Graph display preferences

What is NOT reset:

• Existing project files (your work is safe)

Process:

1. Save your current project before resetting
2. Click “🔄 Reset Application Settings” button
3. All settings restored to factory defaults

When to use:

• Troubleshooting configuration issues
• Starting fresh with default settings
• Clearing corrupted preferences
• Removing experimental settings after testing

Graphs Settings

Configure display options and graph behavior.

Display

Plot Resolution

Options: 16k points, 32k points, 64k points
Default: 32k points (balanced)

FFT resolution for frequency response computation.

• 16k points (16384): Faster computation, lower frequency resolution
• 32k points (32768): Balanced performance and resolution
• 64k points (65536): Maximum frequency resolution, slower computation

Impact:

• Higher resolution provides finer low frequency detail in magnitude, phase, and group delay plots
• Affects computation time for filter responses and driver measurements
• Does not affect the actual filter quality, only the analysis resolution

Group Delay & Phase Truncation

Optionally truncate group delay and phase curves to the active frequency region using magnitude thresholds.

Enable Truncation

Default: Enabled

When checked: Group delay and phase curves are shown only where the corresponding magnitude exceeds a threshold below the maximum.

When unchecked: Full-range phase and group delay displayed regardless of magnitude.

Purpose:

• Avoid displaying phase/group delay in stop-band regions
• Focus on pass-band characteristics
• Reduce visual clutter from irrelevant frequency ranges

Truncation Thresholds (dB below maximum)

Independent thresholds for different contexts:

How thresholds work:

• Values are expressed as dB below the maximum magnitude in the reference curve
• Example: -20 dB means “show phase/GD only where magnitude > (max - 20 dB)”
• Adjustable range: -120 dB to -20 dB

Context matching:

• The reference magnitude matches the context
• Example: FIR curves use FIR magnitude as reference
• Drivers use driver magnitude as reference

THD Truncation:

• Only available for Drivers context
• Shows Total Harmonic Distortion only in active magnitude regions

Magnitude Plot Bounds

Configure default Y-axis bounds for magnitude plots.

Drivers Range (dB)

Range: 20 to 200 dB
Default: 60 dB

Y-axis range for speaker magnitude plots.

• Calculation: From (max_response - range) to (max_response + 5 dB)
• Smaller values (20-40 dB): Zoomed view, emphasizes small variations
• Moderate values (60-80 dB): Good balance for most designs
• Larger values (100-200 dB): Wide view, shows full range

Filters Lower Bound (dB)

Range: -200 to 0 dB
Default: -100 dB

Lower bound of Y-axis for filter magnitude plots (IIR, FIR, FIR+IIR).

• Higher values (-40 dB): Focuses on pass-band details
• Lower values (-100 to -200 dB): Shows deep stop-band attenuation

Impulse/Step Response Display

Adaptive display automatically detects and shows only the relevant portion of impulse and step responses.

Enable Adaptive Display

Default: Enabled

When checked: Automatically detects signal boundaries based on amplitude threshold.

• Detection threshold: 1/1000th of peak amplitude
• Automatic margins: Adds configurable time before and after detected signal

When unchecked: Always shows the complete impulse response (full tap length).

Use case: Focuses view on active signal portion, especially useful for long impulse responses with significant zero-padding.

Margin Before (ms)

Range: 0 to 50 ms
Default: 5 ms

Time added before detected signal start.

Purpose: Ensures pre-ringing or early arrivals are visible in the plot.

Margin After (ms)

Range: 0 to 100 ms
Default: 10 ms

Time added after detected signal end.

Purpose: Captures decay tail and late reflections.

Focus on Summed Response

Default: Disabled

When enabled in multi-driver plots:

• All curves (individual drivers and sum) use the summed IR time boundaries
• Focuses on the main energy region of the system response

When disabled (default):

• Uses the union of all individual curve boundaries
• Shows the full extent of all driver responses

Use case: Quickly identify timing alignment issues across speakers by focusing on the system sum.

Settings Summary Table

Performance Considerations

Real-Time Filter Processing Architecture

LinFIR differs fundamentally from traditional FIR filter design software in how it processes and displays results.

Real-time (“online”) filter generation:

• Filters are generated and applied to driver impulse responses in real time based on UI interactions
• The complete signal processing chain is computed immediately as you adjust parameters
• You see the true response of the entire filter chain instantly, including all windowing artifacts and actual behavior

Why this matters:

• Immediate feedback: No “generate” or “apply” button required - changes are visible instantly
• Transparency: Shows exactly what the filters can and cannot do, with all artifacts visible
• Accuracy: No approximation or preview mode - what you see is what you get

Contrast with traditional software:

• Most FIR design tools require you to validate parameters before showing the true response
• They often show theoretical or idealized responses that don’t reflect actual implementation artifacts
• LinFIR prioritizes showing real-world behavior over computational convenience

Filter Application Methods

FIR filters:

• Applied via fast convolution (multiplication in Fourier domain)
• Highly efficient for typical filter lengths
• Computational cost scales with filter length and number of measurements

IIR filters:

• Applied via recursive difference equations (time-domain processing)
• Extremely efficient regardless of filter complexity
• Minimal performance impact even with many cascaded sections

Performance Impact Factors

Filter length:

• Filters up to 4096 taps: Negligible performance impact on modern CPUs
• Filters 4096-8192 taps: May cause slight delays on slower systems
• Filters >8192 taps: Can noticeably slow down the application, especially during interactive adjustments

Off-axis measurements:

• Each off-axis measurement requires the full filter chain to be recomputed
• Projects with many off-axis angles and long filters may experience longer update times
• LinFIR prioritizes UI responsiveness using background processing (see below)

Background Processing for Responsiveness

To maintain smooth UI interaction even with complex projects, LinFIR uses processing prioritization:

Priority processing (synchronous, immediate):

• On-axis measurement (0°, 0°): Always computed first in the main thread
• Currently displayed angle: Computed immediately for instant visual feedback
• Filter parameters: Updated in real-time as you adjust controls

Background processing (asynchronous, non-blocking):

• Remaining off-axis measurements: Computed in a background thread after priority angles
• Directivity index: Calculated asynchronously when all measurements are ready
• System sum calculations: Processed in parallel when multiple drivers are active

User experience:

• Main graph updates are always immediate - you see changes to the selected angle instantly
• UI remains fully responsive even during heavy background computations
• You can continue adjusting parameters while background calculations complete

Optimization Recommendations

For slower systems or very long filters (>8192 taps):

1. Reduce plot resolution: Settings → Graphs → Plot Resolution

   • Default: 64k points (maximum quality)
   • Recommended for performance: 16k or 32k points
   • Performance improvement with minimal visual quality loss

2. Disable clipping detection: Settings → General → Filter Processing

   • Eliminates test signal processing on every update
   • Only disable if you’re confident in your gain staging

3. Limit off-axis measurements:

   • Use reasonable angle increments (10 or 15°)

Performance is always optimal for:

• Projects with standard filter lengths (≤4096 taps)
• IIR-only filtering (always fast regardless of complexity)
• Single on-axis measurements
• Modern multi-core processors (2020 or newer)

Best Practices

Performance Optimization

• Slower systems: Reduce plot resolution to 16k or 32k points
• Complex projects: Consider disabling clipping detection if filters are well-understood

Filter Design

• Auto Causal Alignment: Keep enabled unless you need precise manual control over IR positioning
• Show Manual FIR Delay Compensation: Keep disabled unless you are experienced in signal processing and need manual FIR alignment control
• Clipping Detection: Keep enabled during design, test actual content before deploying

Graph Display

• Truncation: Enable for cleaner phase/GD plots focused on active frequency regions
• Focus on Summed Response: Enable when analyzing multi-driver timing alignment

Measurement Setup

• Sweep Duration: Use 5-7 seconds for full-range, 2-3 seconds for limited bandwidth
• Sweep Level: Adjust based on amplifier sensitivity (-12 to -6 dBFS typical)

Audio Configuration

Before performing measurements or analyzing audio, LinFIR needs to be configured with the correct audio devices and parameters. This section covers the audio settings available in the application.

Accessing Audio Settings

Audio settings are accessed via:

• Menu: View → Settings → Audio Settings
• Keyboard: Cmd+, (macOS) or Ctrl+, (Windows), then select Audio Settings tab

Device Selection

Input Device

The input device captures audio from microphones or measurement equipment:

• Click the dropdown to see available input devices
• Select your measurement microphone or audio interface input
• Verify the device shows the correct channel count
• Ensure the device isn’t in use by another application

Output Device

The output device plays back sweep signals or test tones:

• Select the audio interface connected to your loudspeaker system
• Verify the device supports the desired sample rate
• Check that the device has sufficient channels for your application

Refreshing Devices

If your audio device doesn’t appear:

1. Click “Refresh Devices” to rescan available hardware
2. Ensure the device drivers are properly installed
3. Check that the device is powered on and connected
4. Try restarting the application if the device remains unavailable

Sample Rate Configuration

Choose a sample rate matching your needs:

• 44.1 kHz: CD quality, most common for consumer audio
• 48 kHz: Professional audio standard, video production
• 88.2/96 kHz: High-resolution audio, improved time-domain precision
• 176.4/192 kHz: Ultra-high resolution (use only when necessary for THD analysis)

⚠️ Note: The sample rate must be supported by both your input and output devices. If a rate is unavailable, try a different value.

Sample Format Detection

LinFIR automatically detects and selects the best compatible audio sample format between your input and output devices:

• 16-bit integer (I16): Standard for most consumer audio devices
• 24-bit integer (I24): Professional audio interfaces (e.g., Dante Virtual Soundcard, RME, MOTU)
• 32-bit float (F32): High-end interfaces, ASIO and Core Audio devices
• 32-bit integer (I32): Some professional hardware

The format is automatically negotiated during measurement setup - no manual configuration required. If LinFIR cannot find a compatible format, try:

• Using ASIO drivers (Windows) for better professional interface support
• Updating your audio interface drivers
• Checking device sample rate compatibility

Buffer Size

Controls latency vs. stability trade-off:

• Smaller buffers (128-256 samples): Lower latency but may cause dropouts
• Recommended (512-1024 samples): Good balance for measurements
• Larger buffers (2048+ samples): More stable but higher latency

For sweep measurements, 256 or 512 samples is typically optimal.

Input Gain Compensation

Software-based gain adjustment applied to the input signal:

• Range: -32 dB to +32 dB
• Purpose: Compensate for fixed or unavailable hardware gain controls
• Use case: Particularly useful for USB microphones like UMIK-1 on Mac OS

Why Use Input Gain Compensation?

Some measurement microphones have fixed gain or no hardware gain control, particularly on certain platforms:

• Mac OS USB microphones: Mac OS enforces very low fixed gain for USB microphones
• UMIK-1 example: No hardware control available in Mac OS system settings
• Solution: Use positive gain compensation (+18 to +24 dB) to achieve optimal recording levels

Important Notes

• Input gain compensation does not improve signal-to-noise ratio
• It’s a convenience feature for level matching, not a replacement for proper gain staging
• Optimal SNR still requires appropriate acoustic levels and microphone positioning
• Use hardware gain controls when available for best results

Default Sweep Parameters

Configure default settings for the built-in sweep measurement system. These apply to new measurement captures and can be accessed in the Audio Settings tab under Measurement Configuration.

Sweep Duration

Range: 1.0 to 10 seconds
Default: 5 seconds

Default length of exponential sine sweep (ESS) for impulse response measurements.

• Shorter (1.0-2s): Faster measurements, lower frequency resolution
• Longer (5-10s): Better low-frequency resolution, improved signal-to-noise ratio, better harmonic distortion measurement

Benefits of longer sweeps:

• Improved signal-to-noise ratio for harmonic distortion analysis
• Better frequency resolution at low frequencies
• More reliable detection and separation of weak harmonics

⚠️ Important: Longer sweeps deliver more energy to the driver at low frequencies (exponential sweeps spend more time in the bass region). For drivers with limited excursion capability, this may require raising the start frequency or reducing the sweep level to avoid over-excursion.

Recommendation: 5-7 seconds for full-range measurements, 2-3 seconds for quick driver checks.

Sweep Level

Range: -40 to 0 dBFS
Default: -6 dBFS

Default output level for sweep signals.

Adjust this to match your amplifier’s input sensitivity and desired acoustic level. Lower values provide headroom for systems with high gain.

Start Frequency

Range: 1 to 5000 Hz
Default: 1 Hz

Default cutoff frequency for the 4th-order Butterworth high-pass filter applied to the sweep signal for driver protection.

Filter Characteristics:

• Attenuation at Start Frequency: -3 dB (Butterworth characteristic)
• Roll-off below cutoff: -24 dB/octave (deterministic, 4th order)

Frequency Guidelines:

• 1 Hz: Lowest configurable frequency, captures near-DC response
• 20 Hz: Typical for full-range loudspeaker measurements
• 60-70 Hz: Example for 80 Hz passband (accounts for -3 dB at cutoff)
• >1000 Hz: Tweeter protection (above resonance frequency)

⚠️ Critical: Position Start Frequency slightly below your desired passband start to compensate for the -3 dB attenuation at cutoff, but always above driver safety limits. Setting too low can cause mechanical or thermal damage.

See Start Frequency: Driver Protection for detailed guidance on driver-specific settings.

Note on high frequency: The high frequency is automatically calculated to optimize harmonic distortion measurement quality (1/48 octave before Nyquist), typically ~23.5 kHz at 48 kHz sample rate, ~47 kHz at 96 kHz.

Using Default Values

These default values are applied when:

• Opening a new measurement window for the first time
• Clicking the Reset Start Freq button in the measurement window
• Creating a new driver without saved sweep parameters

Changing these defaults affects only future measurements - existing measurements retain their configured parameters.

Platform-Specific Considerations

Mac OS

• Timing stability: Excellent (< 0.01ms typical jitter)
• Timing reference: Optional when the same device is used as input and output but recommended for best accuracy
• Multi-interface: Highly reliable even with different input/output devices
• USB microphones: May require input gain compensation due to OS-enforced low gain with usb devices

Windows

• Timing stability: Variable (1-150ms jitter due to scheduler instability)
• Timing reference: Strongly Recommended (Electric or Acoustic mode) for absolute timing accuracy
• Timing inconsistency: IR arrival time may vary without timing reference
• ASIO drivers: Provide improved stability compared to WASAPI
• Measurement averaging: Works with or without timing reference (continuous capture method)

A timing reference on Windows improves absolute timing consistency when comparing measurements across drivers.

Configuration Checklist

Before starting measurements, verify:

• Input device selected and recognized
• Output device selected and functional
• Sample rate matches project settings
• Buffer size set to 256 or 512 samples
• Input gain compensation configured (if needed)
• Microphone calibration loaded (if available)
• Timing reference configured (see Reference Timing below)
• Devices not in use by other applications

Reference Timing for Sweep Measurements

Accurate timing is critical for consistent impulse response measurements, especially when capturing multiple measurements for averaging or multi-position analysis. LinFIR provides three timing reference modes to accommodate different hardware configurations and platform requirements.

Why Timing Reference Matters

Audio playback and recording systems introduce latency from:

• Digital-to-analog conversion (output)
• Analog-to-digital conversion (input)
• Buffer processing delays
• Operating system scheduler variability

Without a stable timing reference, each measurement may have slightly different latency, making:

• Driver alignment unreliable
• Directivity analysis inaccurate (time-of-flight variations)

Timing Reference Modes

None

No timing reference - relies solely on system scheduler timestamps.

How it works:

• Uses operating system audio clock
• Timestamps based on buffer callback timing
• No additional hardware or connections required

When to use:

• Mac OS systems (low scheduler jitter, typically < 0.01 ms)
• Single-shot measurements without averaging
• Simple setups where high precision isn’t critical

Limitations:

• Windows: Not recommended due to significant scheduler jitter (1-150 ms)
• IR arrival time may vary between captures

Mac OS: Reliable and accurate for most applications when using the same device as input and output. Windows: Avoid for professional work - use Electric or Acoustic mode instead.

Electric (Loopback)

Uses a physical cable connection between output and input to provide a stable timing reference.

How it works:

• Connect line output to line input with a cable
• Loopback signal captures exact OS and electrical latency
• Combined electrical and conversion latency measured
• Provides sample-accurate timing reference

Setup:

1. Connect output channel to spare input channel with audio cable
2. Select “Electric” timing reference mode in Audio Settings
3. Ensure loopback channel is not the measurement input channel
4. Run sweep - loopback captures timing automatically

Advantages:

• Most accurate timing reference
• Platform-independent reliability
• Compensates for interface-specific latency
• Eliminates scheduler jitter completely

When to use:

• Windows systems (strongly recommended)
• Designing crossovers
• Professional applications requiring highest accuracy
• Different input/output devices (aggregate devices)

Limitations:

• Requires spare input channel for loopback
• Requires physical cable connection
• May not be available on simple USB audio interfaces

Acoustic

Uses another driver (reference driver) as a timing microphone.

How it works:

• Reference driver captures sweep output acoustically
• Provides stable timing reference based on acoustic arrival
• Measures relative timing between reference and measured driver
• Does not capture true absolute acoustic + electrical travel time

Setup:

1. Select a reference driver (must reproduce high frequencies > 5 kHz)
2. Position reference driver microphone to capture sweep clearly
3. Select “Acoustic” timing reference mode in Audio Settings
4. Reference driver captures timing on each sweep

Advantages:

• No loopback cable required
• Works when electrical loopback unavailable
• Good alternative for Windows users without spare input channels
• Suitable for multi-driver directivity measurements

When to use:

• Loopback hardware not available
• Windows systems (recommended alternative to None)

Limitations:

• Only measures relative timing (not absolute acoustic delay)
• Reference driver must reproduce high frequencies cleanly
• Acoustic path must be clear and consistent
• Less accurate than Electric mode

Important: Reference driver should be full-range or at least capable of reproducing 5-10 kHz cleanly for reliable timing detection.

Timing Offset (Preferences)

Adjustable time offset to shift impulse position.

• Range: Typically ±100 ms
• Primary use: Acoustic mode when reference driver is farther than measured driver
• Purpose: Compensate for acoustic propagation delay differences

Example:

• Reference driver is 50 cm from its microphone
• Measured driver is 100 cm from its microphone
• Acoustic delay difference ≈ 1.5 ms (50 cm / 343 m/s)
• Apply negative offset of ~1.5 ms to align impulse peaks

Fine-tuning:

1. Run test measurement
2. Check impulse response time alignment
3. Adjust offset in small increments (1-2 ms)
4. Re-measure to verify alignment

Platform Requirements

Windows

• Averaging: Works with or without timing reference (continuous capture eliminates jitter issues)
• Scheduler jitter: Significant (1-150 ms variable latency) for initial timing only
• Recommendation: Timing reference still recommended for absolute timing accuracy across sessions
• Single measurements: Work with “None” mode but may have timing variation

Without timing reference on Windows:

• Each measurement has unpredictable latency
• Directivity analysis compromised
• Time and phase alignment between drivers will lead to wrong results

Mac OS

• Scheduler jitter: Minimal (< 0.01 ms typical)
• Recommendation: Timing reference optional but still beneficial for best accuracy
• Multi-interface: More stable than Windows even without reference

Choosing the Right Mode

Configuration Workflow

1. Assess requirements:

   • Single measurement or averaging?
   • Mac OS or Windows?
   • Spare input channel available?

2. Select mode:

   • Windows averaging → Electric or Acoustic
   • Mac OS → None acceptable, Electric for best results
   • No loopback → Acoustic (if reference driver available)

3. Setup hardware (if Electric):

   • Connect output to spare input channel
   • Verify loopback signal path

4. Setup reference driver (if Acoustic):

   • Select driver with good high-frequency response
   • Position reference microphone to capture sweep clearly

5. Configure in Audio Settings:

   • Select timing reference mode
   • Adjust timing offset if needed (Acoustic mode)

6. Test:

   • Run test measurement
   • Verify timing consistency
   • Adjust offset if required

Timing Reference Troubleshooting

Inconsistent Measurements

Symptom: Each measurement has different arrival time
Cause: No timing reference active
Solution: Enable Electric or Acoustic mode

Reference Signal Not Detected

Symptom: Error message about missing reference
Cause: Loopback not connected or reference driver not capturing sweep
Solution:

• Electric: Verify cable connection and input channel selection
• Acoustic: Ensure reference driver can reproduce sweep frequencies

Timing Offset Not Working

Symptom: Impulse still misaligned after offset adjustment
Cause: Incorrect offset direction or magnitude
Solution:

• Try opposite sign (positive ↔ negative)
• Increase offset magnitude
• Verify reference driver is capturing sweep correctly

Best Practices

• Windows users: Always use Electric or Acoustic mode for professional work
• Loopback preferred: Most accurate, eliminates all timing variables
• Acoustic alternative: Good fallback when loopback unavailable
• Test before session: Verify timing consistency with test measurements
• Document settings: Record timing mode and offset in project notes
• Consistent hardware: Use same interface for all measurements in a session

Troubleshooting

Device Not Listed

• Click “Refresh Devices” to rescan
• Ensure device drivers are installed
• Check that device isn’t in use by another application
• Try different sample rates or buffer sizes
• Restart LinFIR

Configuration Invalid

• Selected devices may have been disconnected
• Sample rate may not be supported by device
• Channel count may exceed device capabilities
• Try selecting default devices and refreshing

Audio Dropouts

• Increase buffer size
• Close other applications using audio
• Check for system resource constraints
• Update audio interface drivers

No Sound During Sweep

• Verify output device is selected and powered on
• Check system audio settings
• Ensure exclusive mode isn’t blocking access (Windows)
• Try different buffer sizes
• Verify output channel selection in IR Management window

Microphone Calibration

Manage calibration files for measurement microphones and define default calibrations for new drivers.

Overview

Microphone calibration compensates for the frequency response and sensitivity of your measurement microphone. LinFIR supports:

• Multiple calibration files stored in application settings
• Per-driver calibration selection in the IR Management window (see IR Management)
• Default calibration for newly created drivers
• Automatic application to new projects when a default is set
• Minimum-phase correction computed in real-time from calibration data

Calibration scope:

• Calibration data is embedded in project files (.lnf) for portability
• Calibrations from projects are also stored in application settings for reuse
• Each driver can use a different calibration file or none at all

Calibration File List

The main section displays all imported calibration files in a scrollable grid:

Automatic driver updates:

• When you delete a calibration file that’s currently assigned to one or more drivers, those drivers are automatically reset to “None”
• A warning toast will notify you which drivers were affected
• This prevents broken references and ensures project stability

Default Calibration

Select a calibration file to automatically apply to newly created drivers.

Dropdown selector:

• Shows “None” when no default is set
• Lists all imported calibration files by name
• Changes apply immediately to new drivers and projects

When the default is applied:

• Manual driver creation: When you add a new driver via the Drivers toolbar
• New Loudspeaker Design project: All initial drivers receive the default calibration
• New Room Calibration project: The initial measurement receives the default calibration

Note: Changing the default does not affect existing drivers in the current project or previously saved projects. Only new drivers created after the change receive the new default.

Import Calibration File

Click “Import Calibration File” to load a microphone calibration file.

Supported file format:

• Plain text files with two columns: Frequency (Hz) and Magnitude (dB)
• Whitespace or comma-separated
• Lines starting with non-numeric characters are treated as comments/headers
• Optional header line with sensitivity factor and serial number

Example file format:

# Sens Factor = -2.5dB, SERNO: 41486651
# Frequency (Hz)  Magnitude (dB)
20.0              -0.5
50.0              -0.3
100.0             -0.1
200.0              0.0
500.0              0.1
1000.0             0.0
2000.0            -0.2
5000.0            -0.5
10000.0           -1.0
20000.0           -2.0

Sensitivity factor and serial number:

• Sensitivity factor (in dB) is automatically extracted from the file header if present (line containing “Sens Factor”)
• Serial number is automatically extracted if present (SERNO: field in header)
• If no sensitivity factor is found in the file, it defaults to 0 dB
• Serial number is used to differentiate calibrations with identical filenames when sharing projects

Storage: Calibration data (frequency response, sensitivity, and serial number) is embedded in project files (.lnf) to ensure portability:

• Projects can be shared without requiring recipients to have the calibration files
• When loading a project, calibrations are automatically added to application settings if not already present
• If a calibration with the same name exists:
  • With serial number: differentiated using serial number (e.g., filename (SN:7016078))
  • Without serial number: differentiated using numeric suffix (e.g., filename (1), filename (2))
• You can also manually import calibrations to your settings library for reuse across projects

How Calibration is Applied

Minimum-phase correction: LinFIR converts the imported frequency response correction into a minimum-phase FIR filter applied during IR processing.

Processing steps:

1. Frequency filtering: Bins below 5 Hz are removed to prevent phase distortion at very low frequencies
2. Interpolation: Calibration data is interpolated to match your IR’s sample rate and FFT size
3. Minimum-phase conversion: Magnitude correction is converted to minimum-phase using Hilbert transform
4. Sensitivity compensation: The sensitivity factor (dB) is applied as a gain adjustment
5. IR application: The resulting correction filter is convolved with the driver’s impulse response

Phase behavior: Because the correction uses minimum-phase reconstruction, the calibration affects both magnitude and phase of the measurement. This is physically accurate for measurement microphones, which exhibit mainly minimum-phase behavior in their frequency response deviations.

Per-Driver Calibration Selection

Each driver can use a different calibration file (or none). To assign a calibration to a specific driver, use the microphone calibration dropdown in the IR Management window.

See: IR Management → Microphone Calibration

Best Practices

File organization:

• Use descriptive filenames that include microphone model and serial number
• Example: UMIK-1_SN123456.txt or Earthworks_M30_SN789.txt

Calibration maintenance:

• Import calibration files for all your measurement microphones
• Set your most frequently used microphone as the default

Project workflow:

• Assign calibrations in the IR Management window after importing measurements
• Use “None” for measurements taken with already-flat reference microphones
• Different drivers in the same project can use different calibrations (useful when combining measurements from multiple sessions)

Frequency range:

• Ensure calibration files cover the full measurement bandwidth
• LinFIR interpolates between provided points, but extrapolation beyond the calibration range may produce inaccurate results
• Most measurement microphones provide calibration data from 20 Hz to 20 kHz

Sensitivity factor:

• Use the value from your microphone’s calibration certificate
• This is typically a negative dB value (e.g., -0.5 dB, -2.4 dB)
• If unsure, use 0 dB and rely solely on the frequency response correction

License

The License tab allows you to activate and manage your LinFIR license.

License Features

A valid LinFIR license unlocks advanced analysis features:

• Off-axis measurements: Import and analyze measurements at angles other than 0° (horizontal and vertical)
• Directivity Index (DI): Calculate and visualize directivity patterns across frequency
• Directivity Sonogram: Access the Directivity Sonogram window for detailed visualization
• Multi-angle visualization: View multiple measurement angles simultaneously in graphs
• Harmonic distortion details: View individual harmonic levels (THD breakdown by harmonic)
• Hypex FusionAmp mode: Create projects for Hypex FA series amplifiers with hardware-locked parameters and HFD export

Without a license, LinFIR operates with full functionality for on-axis measurements only (0°, 0°). The following features are unavailable without a license: off-axis analysis, directivity tools (DI and sonogram), harmonic distortion breakdown by harmonic, and Hypex FusionAmp project creation.

Activating Your License

If you don’t have an active license, the License tab will display an activation form.

Activation Process

1. Enter your license key

   • Format: xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx
   • Provided in your purchase confirmation email

2. Enter your email address

   • Must match the email used for purchase
   • Used for license validation

3. Click “Activate License”

   • Creates a new activation tied to this specific machine
   • Each activation is identified by a unique instance name and ID

Important:

• Each license allows one activation at a time
• Each activation is machine-specific
• You must deactivate one machine before activating another
• You can manage activations from the License tab

License Status

When a license is active, the License tab displays:

Current License Information

• Customer name: License holder’s name
• Email: Email address associated with the license
• Product: LinFIR license type
• Status: Current license status (active, expired, etc.)
• Activations: Number of active machines (e.g., “1 / 1”)
• Last checked: Date and time of last validation

This Machine

• Instance Name: Automatically generated name for this activation
• Instance ID: Unique identifier for this machine’s activation

Managing License Activations

Validate License

Click “Validate License” to check the current status of your license with the licensing server.

Use cases:

• Verify license is still active
• Update activation count
• Check for license changes

Deactivate This Machine

Click “Deactivate This Machine” to free up this activation slot.

When to use:

• Switching to a different computer
• Freeing up an activation for another machine
• System reinstall or hardware change

Effect:

• Removes this machine’s activation
• Frees one activation slot
• License features will no longer be available on this machine
• You can reactivate on this or another machine later

Managing Other Activations

The “All License Activations” section displays all machines where the license is currently active.

For each activation:

• Name: Instance name
• Instance ID: Unique identifier
• Created: Activation date

Actions:

• This Machine: Marked with ✓ (cannot be deactivated from this view)
• Other machines: Click “Deactivate” to remove that activation remotely

Use case: If you activated LinFIR on a machine you no longer have access to, you can deactivate it from here to free up that slot.

Troubleshooting

“No Active License” but I activated

• Click “Validate License” to refresh status
• Check internet connection
• Verify license key and email match your purchase

Maximum activations reached

• View “All License Activations” section
• Deactivate machines you no longer use
• Contact support if you need additional activations

License not validating

• Ensure internet connection is active
• Check firewall settings (licensing server access required)
• Verify license hasn’t expired

Privacy & Data

What is stored:

• License key (encrypted)
• Email address
• Instance name and ID (machine identifier)
• Activation timestamps

What is sent to licensing server:

• License key
• Email address
• Instance information (for activation/validation)

No project data, measurements, or filter designs are transmitted during license operations.

IR Management

The IR (Impulse Response) Management window provides comprehensive tools for obtaining, processing, and managing impulse responses for each driver or measurement position in your system.

Accessing IR Management

Loudspeaker Design Mode:

• To open: Driver → Manage IR
• Purpose: Import, capture, and process driver impulse responses at multiple angles

Room Calibration Mode:

• To open: Measurements → Manage IR
• Purpose: Capture IR at a given in room position

Driver/Measurement Name

Manually enter a descriptive name for the impulse response.

• Label: “Driver Name” (Loudspeaker Design) or “Measurement Name” (Room Calibration)
• Purpose: Identify IRs in project and exports
• Persistence: Names are saved with the project
• Optional: Empty names default to “Driver X” or “Measurement X”
• Best practice: Use descriptive names like “Woofer_Left”, “Tweeter_RefXYZ”, “Position_Center”

Capturing IR

Button: 🎤 Measure

Triggers a measurement sweep with the configured parameters to capture the impulse response.

Features:

• Exponential sine sweep generation with configurable parameters
• Automatic harmonic distortion analysis (THD)
• Timing reference support for multi-driver measurements

For complete documentation on sweep parameters, driver protection, measurement workflow, and best practices, see Sweep Measurements.

Off-Axis Measurements 🔒

Available in: Loudspeaker Design mode only
License required: Valid LinFIR license needed for off-axis measurements

Overview

Off-axis measurements enable directivity analysis by capturing impulse responses at different horizontal and vertical angles. These measurements are essential for understanding speaker radiation patterns and optimizing crossover design for consistent off-axis response.

Measurement Organization

Measurements are displayed in two columns:

• Horizontal column: Measurements with vertical angle = 0°, varying horizontal angle

  • Example: -90°, -60°, -30°, 0°, +30°, +60°, +90°

• Vertical column: Measurements with horizontal angle = 0°, varying vertical angle

  • Example: -60°, -40°, -20°, 0°, +20°, +40°, +60°

On-axis reference: The (0°, 0°) measurement appears in both columns.

Axis Selection

Before importing or capturing an off-axis measurement, select the measurement axis:

• Horizontal button: Measurements along horizontal axis (left/right)
• Vertical button: Measurements along vertical axis (up/down)

The selected axis determines whether the angle applies to horizontal or vertical position.

Measurement Angle

Field: “Measurement angle”
Range: -180° to +180°
Indicates: Angle for next import or capture operation

How it works:

• If Horizontal axis selected: angle applies to horizontal position (vertical = 0°)
• If Vertical axis selected: angle applies to vertical position (horizontal = 0°)
• The field shows which axis is active: “(horizontal)” or “(vertical)”

Restrictions:

• Disabled until on-axis (0°, 0°) measurement exists
• Requires valid LinFIR license
• Must import on-axis measurement first before adding off-axis angles

Per-Measurement Actions

Each measurement in the table has two action buttons:

Export Button (⬇)

Click to export this specific measurement:

1. Opens export format dialog
2. Choose WAV or TXT format
3. Option to include distortion metadata
4. File is saved with angle suffix (e.g., Driver1_v0h30.wav, Driver1_v-15h0.wav)

Delete Button (❌)

Remove a measurement from the driver:

• Off-axis measurements: Can be deleted freely
• On-axis (0°, 0°) measurement: Can only be deleted if all off-axis measurements are removed first
  • This restriction exists because the on-axis measurement serves as the reference for all filter calculations

Tooltip guidance:

• Enabled: “Delete measurement”
• Disabled: “Delete all off-axis measurements first”

Export All Button

Button: ⬇ Export All
Location: Above “Measurement angle” field
Purpose: Export all measurements in a single operation

How it works:

1. Click “⬇ Export All”
2. Select output directory in file dialog
3. All measurements are exported as WAV files
4. Files use consistent naming: DriverName_v{vertical}h{horizontal}.wav

File naming examples:

• On-axis: Driver 1_v0h0.wav
• Horizontal +30°: Driver 1_v0h30.wav
• Vertical -15°: Driver 1_v-15h0.wav
• Horizontal -45°: Driver 1_v0h-45.wav

Format:

• All files exported as WAV (32-bit float, mono)
• Sample rate matches original measurement or importation sample rate
• Distortion metadata option applies to all files
• No individual format selection (always WAV)

Benefits:

• Fast export of complete measurement sets
• Consistent naming for re-import or external processing
• Files can be re-imported via batch import with automatic angle detection

Use cases:

• Archiving complete measurement sets
• Sharing measurements with collaborators
• Processing measurements in external tools (FIR Designer, Matlab, etc.)

Symmetry Tool

Button: ⚖ Symmetry
Location: Below each column
Purpose: Automatically duplicate measurements to opposite angles

How it works:

• Horizontal Symmetry: If you have +30°, creates -30° by mirroring the measurement
• Vertical Symmetry: If you have +20°, creates -20° by mirroring the measurement

Use case: Save measurement time by capturing one side and mirroring to the other side, assuming the speaker has symmetric radiation.

Workflow for Off-Axis Measurements

1. Import or capture on-axis measurement (0°, 0°) first
2. Select axis (Horizontal or Vertical)
3. Set measurement angle (e.g., +30°)
4. Import file or capture measurement
5. Repeat for additional angles
6. Use Symmetry tool to fill opposite angles (optional)
7. Analyze directivity patterns in main application graphs

Import from File

Load impulse responses from external files.

LinFIR works internally with impulse responses (IRs). All imported files are ultimately converted to an IR before being used for filter design. Two import paths exist:

• Direct IR formats (WAV, TXT, TMD): Files already contain time-domain impulse response samples. They are resampled to the project sample rate and loaded directly.
• Frequency-domain format (FRD): Files contain frequency response data (magnitude + phase). LinFIR reconstructs an impulse response via the FRD to IR Converter window before proceeding.

Supported Formats

WAV Files

• Channels: Mono or stereo (left channel used for stereo files)
• Bit depth: 16-bit, 24-bit, or 32-bit PCM
• Automatic resampling: The IR is resampled to match the project sample rate for processing. The original samples are preserved internally at their native rate, so changing the project sample rate later does not cause quality loss from successive resamplings. This is handled transparently in the background.

With distortion metadata:

• If WAV contains custom ‘dist’ chunk (exported from LinFIR with “Include dist.” option)
• Enables THD (Total Harmonic Distortion) computation

Without distortion metadata:

• Standard WAV import
• Only practical/cropped IR available
• THD computation not available

TXT Files

Plain text files containing impulse response samples, one value per line or space/comma-separated. LinFIR auto-detects this format.

Note: .txt files containing FRD-formatted data (3 columns: frequency, magnitude, phase) are automatically detected and processed as FRD files (see FRD Files section below).

TMD Files

MATAA (Mat’s Audio Analyzer) time-domain data format. Each line encodes a sample with its timestamp, allowing the sample rate to be derived from the data itself.

• Format: Two-column text file (time in seconds, amplitude)
• Comments: Lines starting with * are metadata/comments
• Sample rate: Inferred from the time intervals between samples. If the result is within 1% of a standard rate (44.1, 48, 88.2, 96, 176.4, 192, 384 kHz, etc.), LinFIR snaps to that exact rate to correct for the systematic sample rate error introduced by limited timestamp precision.
• Timestamp precision: Timestamps must have enough decimal places for accurate sample rate inference (e.g. 9 digits for 96 kHz). Insufficient precision can produce a slightly off sample rate that will still be corrected if it falls within the 1% snapping tolerance.
• Precision: Full floating-point precision (no bit-depth limitations)

Example TMD file:

* TMD data written by MATAA
* Measurement description
0.000000000	0.000000
0.000020833	0.125000
0.000041667	0.850000
...

CLIO Files

Time-domain impulse response exports from CLIO (Audiomatica) measurement software. These are plain .txt files with three columns.

• Format: Text file with a header row followed by data rows
  • Column 1 — Time[s]: sample timestamp in seconds
  • Column 2 — Real[Pa]: signal amplitude (used as the IR)
  • Column 3 — Imag[Pa]: imaginary part (ignored by LinFIR)
• Header line: the first non-empty line must start with Time (case-insensitive)
• Sample rate: inferred from the time intervals between samples, then snapped to the nearest standard rate within 1% tolerance (same algorithm as TMD)
• Precision: full floating-point precision

Example CLIO file:

Time[s]         Real[Pa]        Imag[Pa]
0.000000E+00    -4.790139E-03   -4.583853E-10
2.083333E-05     4.526122E-03   -8.076313E-10
4.166667E-05     3.454284E-04    1.007265E-10
...

FRD Files

Frequency response data files (commonly exported by REW, VituixCAD, or provided by manufacturers). These files do not contain an impulse response directly—LinFIR reconstructs one via the FRD to IR Converter window (see dedicated section below).

Format:

• 3 columns per line: Frequency (Hz), Magnitude (dB), Phase (degrees)
• Text-based, space or tab delimited
• Can use either .frd extension or .txt extension (auto-detected)

Processing:

• Automatic interpolation to required frequency points using PCHIP (Piecewise Cubic Hermite Interpolating Polynomial)
• Configurable magnitude extrapolation for DC and high-frequency regions
• Phase handling with group delay compensation

Extrapolation Modes:

When importing FRD files, LinFIR opens the FRD to IR Converter window with the following extrapolation controls:

• DC Extrapolation: Controls how magnitude is extended below the lowest measured frequency

  • Constant: Flat extrapolation (holds the lowest measured value)
  • Roll-off: Applies a 12 dB/octave high-pass roll-off for realistic loudspeaker behavior
  • Default: Roll-off

• High Frequency Extrapolation: Controls how magnitude is extended above the highest measured frequency toward Nyquist

  • Constant: Flat extrapolation (holds the highest measured value)
  • Roll-off: Applies a 12 dB/octave low-pass roll-off, automatically compensates for upward slopes
  • Default: Constant

Benefits of Roll-off mode:

• More realistic impulse responses from incomplete FRD data
• DC roll-off mimics natural loudspeaker low-frequency limitations
• HF roll-off prevents artifacts from measurements that don’t extend to Nyquist
• Adaptive: compensates for incorrect slope trends in measured data

Important notes:

• FRD files are sparse (often logarithmically spaced frequency points)
• Requires complex interpolation, potentially introducing artifacts
• Generally do not contain the time of flight required to properly time align drivers
• Recommendation: Prefer importing impulse responses (WAV) whenever possible

Automatic Noise Truncation

Once an impulse response has been imported or reconstructed (from any format), LinFIR automatically truncates it to optimize file length and processing performance. This is an internal post-processing step applied uniformly to all IR data before it is stored in the project.

Truncation Algorithm:

1. Peak detection: Locates the main impulse peak (maximum absolute value in the IR)
2. 5-second limit: Truncates the IR to 5 seconds maximum after the peak
   • Corresponds to approximately 1.7 km acoustic distance
   • Sufficient for capturing full decay in most practical scenarios
3. Noise threshold: Removes trailing samples below 0.1% of peak amplitude
   • Adaptive threshold based on the actual signal level
   • Eliminates noise floor without affecting useful signal
4. Minimum duration: Always preserves at least 1.0 seconds of the IR
   • Ensures sufficient data for low-frequency analysis

Benefits:

• Reduces processing load for FFT operations and convolutions
• Decreases file sizes for project saving and export
• Removes unnecessary noise floor that doesn’t contribute to filter design

What gets removed:

• Excessive silence beyond 5 seconds after the main impulse
• Noise floor below the 0.1% threshold (≈ -60 dB relative to peak)
• Applies to both anechoic and in-room measurements

What is preserved:

• The complete main impulse and initial decay
• At least 1.0 seconds of IR data regardless of noise level
• All frequency content relevant for crossover design
• Sufficient time-domain information for phase analysis

Impulse Response Timing Requirements

Peak delay limit: LinFIR rejects impulse responses where the main peak is located more than 200 ms after the start of the file.

Why this limit exists:

LinFIR performs extensive signal processing operations—FFT transformations, convolutions, resampling, and real-time filter preview—that must remain responsive. These operations scale with impulse response length, and excessive zero padding before the actual signal significantly impacts performance without adding useful information.

Physical context:

A 200 ms delay corresponds to approximately 68 meters of acoustic travel distance. In practical loudspeaker measurement and room calibration scenarios, such excessive delays indicate that no timing reference method was used during capture (loopback trigger, electrical reference, or pre-alignment in the measurement software).

Impact on workflow:

Without proper timing reference, relative time alignment between drivers becomes unreliable, as the arbitrary delays vary unpredictably between measurements. This compromises crossover phase coherence and summation accuracy—core functions of LinFIR.

Best practice:

When capturing impulse responses, use measurement software that provides:

• Loopback triggering (audio interface output → input reference)
• Electrical reference triggering (synchronized start)
• Pre-alignment features that position the IR peak near the beginning of the recording

These methods ensure minimal file length while preserving the precise time-of-flight information needed for accurate driver alignment and filter design.

If your measurement contains a valid signal buried in initial silence, pre-process the file to remove leading zeros before importing into LinFIR.

Import Workflow

1. Click “📁 Import” button
2. Select WAV, TXT or FRD file
3. File is automatically detected and processed:
   • WAV: Resampled to project sample rate, metadata extracted
   • TXT (impulse samples): Either auto-imported (with metadata) or user prompted for sample rate
   • TXT/FRD (frequency data): Opens FRD to IR Converter window
4. IR appears in impulse response graph
5. Manually enter IR name in text field
6. Apply windowing if necessary (see Time Windowing section)

Special case for FRD files: When importing .frd or FRD-formatted .txt files, the FRD to IR Converter window opens automatically. See the dedicated section below for details.

⚠️ Display Mode Check: If the imported impulse response does not appear in the graphs, verify that Drivers/Measurements display mode is selected in the graph toolbar (not FIR, IIR, or FIR+IIR modes which only show filter responses). The graph title and legend clearly indicate which mode is active.

Batch Import of Multiple Measurements

Available in: Loudspeaker Design mode (off-axis measurements)
License required: Valid LinFIR license needed for off-axis measurements

Overview

Batch import enables importing multiple measurement files simultaneously with automatic angle detection from filenames. This significantly speeds up the workflow when importing multi-angle measurement sets from tools like VituixCAD, ARTA, or REW.

Filename Angle Parsing

LinFIR automatically detects measurement angles from filenames using conventions from other measurement software.

Important: Angle patterns must be at the end of the filename (before extension). If the axis marker (h/v) has a prefix format (like h30), it must not be preceded by a letter. This prevents false detection in cases like “Driverh30” while allowing “SIG270-4h30” or “Driver 2_h30”.

Format with single axis specification (H = Horizontal, V = Vertical):

• H30, h30, H-20 → Horizontal axis, angles: +30°, +30°, -20°
• V15, v15, V-45 → Vertical axis, angles: +15°, +15°, -45°
• 30H, 30h, -20H → Horizontal axis, angles: +30°, +30°, -20°
• 15V, 15v, -45V → Vertical axis, angles: +15°, +15°, -45°

Format without axis (@ marker):

• @30, @-20, Tweeter@15 → Angle detected, axis must be specified by user
• When files use @ notation, a dialog prompts you to select Horizontal or Vertical axis

MF 2D format (both H and V in filename):

• v0h30, V0H30, v000h030 → Horizontal +30° (horizontal plane: V=0)
• v30h0, V30H0, v030h000 → Vertical +30° (vertical plane: H=0)
• h30v0, H30V0, h030v000 → Horizontal +30° (order-independent)
• h0v30, H0V30, h00v030 → Vertical +30° (order-independent)
• v0h0, V0H0, v000h000 → On-axis (0°, 0°)
• v30h50 → Rejected (out-of-plane measurement)

Note: Leading zeros are supported (e.g., 030 = 30)

Note on MF 2D format: Only measurements in a plane are accepted (one angle must be 0). Files with both H and V non-zero are ignored as they represent oblique measurements that can’t be represented in LinFIR’s horizontal/vertical coordinate system. The order of h/v doesn’t matter.

Examples of valid filenames:

• Tweeter_h0.wav → On-axis (0°, 0°)
• Tweeter_h30.wav → Horizontal +30°
• Tweeter_v-20.frd → Vertical -20°
• Woofer@15.txt → Angle +15° (user selects axis)
• measurement_30h.wav → Horizontal +30°
• Left_-45v.frd → Vertical -45°
• Driver_v0h45.frd → Horizontal +45° (MF 2D format)
• Tweeter_v20h0.txt → Vertical +20° (MF 2D format)
• SIG270-4h30.frd → Horizontal +30° (number before axis letter is OK)
• SIG270-4v0h30.frd → Horizontal +30° (MF 2D with number prefix)

Examples of rejected filenames:

• Driverh30.wav → Rejected (letter “r” directly before “h”)
• Testh30v0.frd → Rejected (letter “t” directly before “h”)

Files without angle markers:

• Imported as on-axis (0°, 0°) measurement
• You can still manually set the angle before importing if needed

Batch Import Workflow

1. Multi-File Selection

1. Click “📁 Import” button
2. Select multiple files in the file dialog (Ctrl+Click or Shift+Click)
3. Click Open

2. Automatic Processing

LinFIR automatically analyzes all selected files:

If all files have explicit axis (H/V in filename):

• Files are processed immediately
• Each file imported at the angle specified in its filename
• No user interaction needed

If files use @ notation (ambiguous axis):

• Axis Selection Dialog appears
• Select either Horizontal or Vertical axis
• Click Continue
• Selected axis applies to all @ files in the batch

3. FRD Converter Integration

If the batch includes FRD files:

1. LinFIR opens the FRD Converter for the on-axis FRD file (if present)
2. Configure conversion parameters (sample rate, delay, extrapolation modes)
3. Click “Use This Impulse Response”
4. All remaining FRD files automatically use the same parameters
5. No need to configure each file individually

How it works:

• Parameters are configured using the on-axis measurement (0°, 0°)
• Same delay and extrapolation settings applied to all off-axis FRD files
• Ensures consistent conversion across the measurement set

⚠️ Important Limitation - Missing Relative Time Delays:

FRD files contain frequency response data only (magnitude and phase) without encoding the relative time delays between different measurement angles. When importing multiple FRD files at different angles, all files use the same delay parameter configured on the on-axis measurement.

Why this matters for off-axis measurements:

In real loudspeakers, off-axis measurements have different acoustic path lengths and interference patterns compared to on-axis:

• Sound from tweeter and woofer arrives at different relative times depending on measurement angle
• These time differences create frequency-dependent interference (comb filtering)
• Proper time alignment between drivers varies with angle

Consequence:

• Batch-imported FRD files will have identical relative timing at all angles
• Phase relationships between drivers won’t reflect actual acoustic behavior
• Directivity analysis and crossover optimization will be inaccurate
• Simulated off-axis response won’t match real-world interference patterns

Recommendation:

• Prefer WAV impulse response files for multi-angle measurements whenever possible
• WAV files preserve the actual time-of-flight at each angle
• Only use FRD batch import for preliminary analysis or when WAV files aren’t available
• If using FRD files, be aware that off-axis crossover behavior won’t be accurately modeled

Valid use cases for batch FRD import:

• Quick magnitude response comparison across angles
• Preliminary directivity visualization
• Single-driver systems where relative timing isn’t critical
• When no impulse response measurements are available

Practical Example

Scenario: You have a set of horizontal measurements exported from REW:

Tweeter_h0.frd
Tweeter_h15.frd
Tweeter_h30.frd
Tweeter_h45.frd
Tweeter_h60.frd

Workflow:

1. Select all 5 files in file dialog
2. FRD Converter opens for Tweeter_H0.frd (on-axis)
3. Set sample rate to 96000 Hz, delay to 5 ms
4. Choose DC extrapolation: Roll-off
5. Click “Use This Impulse Response”
6. Files H15, H30, H45, H60 automatically converted with same settings
7. All 5 measurements appear in the horizontal column

Result:

• All angles have same delay (5 ms) configured on on-axis measurement
• ⚠️ Relative time-of-flight differences between angles are lost
• For accurate crossover modeling, prefer WAV impulse response files

Limitations

• Batch import only works with files having consistent format (all WAV, all FRD, or mixed)
• Files must follow supported angle naming conventions
• On-axis measurement (0°, 0°) should be included when importing FRD files for optimal parameter configuration
• If mixing WAV and FRD formats, FRD files must include on-axis or parameters will use defaults
• FRD files don’t preserve relative time delays between angles—prefer WAV files for multi-angle measurements

Tips

Consistent naming convention:

• Use descriptive base names: Tweeter_h30.wav instead of just h30.wav
• Include driver name in filename for easier identification
• Use lowercase h/v for consistency with LinFIR exports
• MF 2D format is order-independent: v0h30 = h30v0

Large measurement sets:

• Import horizontal and vertical measurements separately if you have both
• Use Symmetry tool after batch import to fill missing opposite angles
• Verify measurement table after import to ensure correct angle assignment

FRD files:

• ⚠️ Not recommended for off-axis measurements (missing relative time delays)
• Always include on-axis (H0 or V0) measurement in batch for parameter reference
• Configure extrapolation modes carefully for first file—settings propagate to all
• Verify impulse response quality after batch conversion
• Consider this a preliminary workflow—prefer WAV files for accurate directivity analysis

Recommended format priority:

1. WAV files: Best option—preserves all timing information
2. TXT with metadata: Acceptable if exported from LinFIR
3. FRD files: Only for magnitude analysis—timing not preserved

FRD to IR Converter

When importing FRD files, LinFIR opens a dedicated conversion window with advanced tools for interpolation, visualization, and quality assessment.

Overview

The FRD to IR Converter transforms sparse frequency-domain data into time-domain impulse responses using sophisticated interpolation and extrapolation techniques. It provides real-time visualization and quality metrics to help you assess the conversion quality.

Conversion Parameters

Sample Rate

Options: 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz
Default: 96 kHz

Target sampling frequency for the resulting impulse response.

Considerations:

• Higher sample rates provide better frequency resolution
• Must match or exceed your project’s sample rate
• FRD data is interpolated to fill all frequency bins from DC up to Nyquist

Delay Compensation

Range: 0.0 to 100.0 ms
Default: 5.0 ms

Time offset applied to the impulse response to allow for pre-ringing visualization and processing.

Purpose:

• Provides time buffer before the main impulse
• Essential when FRD data contains non-minimum phase characteristics
• Allows visualization of pre-ringing artifacts

Warning indicator: If pre-ringing duration exceeds the delay setting, a warning appears suggesting either:

• Increase delay to accommodate pre-ringing
• Enable Minimum Phase transformation

Minimum Phase Transformation

Default: Disabled

Applies Hilbert transform to convert the impulse response to minimum phase, eliminating all pre-ringing.

When to use:

• FRD data shows significant pre-ringing (> 5% energy before peak)
• Phase data is corrupted, smoothed, or contains wrapping errors
• You need a causal, physically realizable impulse response
• Time-domain alignment is critical

Effects:

• Removes all energy before the main peak
• Preserves magnitude response
• Changes phase response to minimum phase equivalent
• Results in a causal system (no pre-ringing)

Note: Pre-ringing in FRD-derived impulses is often an artifact of measurement or file creation, not a physical characteristic of the device.

DC Extrapolation

Modes: Constant, Roll-off
Default: Roll-off

Controls how magnitude is extended below the lowest measured frequency toward DC (0 Hz).

Constant mode:

• Flat extrapolation using the lowest measured value
• Use when FRD data already reaches very low frequencies

Roll-off mode (recommended):

• Applies 12 dB/octave high-pass characteristic
• Mimics natural loudspeaker low-frequency roll-off
• More physically realistic for incomplete low-frequency data

High Frequency Extrapolation

Modes: Constant, Roll-off
Default: Constant

Controls how magnitude is extended above the highest measured frequency toward Nyquist.

Constant mode (recommended):

• Flat extrapolation using the highest measured value
• Maintains adequate signal level to Nyquist
• Minimizes noise amplification

Roll-off mode:

• Applies 12 dB/octave low-pass characteristic
• Automatically compensates for upward slope trends
• Use when FRD data shows unrealistic high-frequency behavior

Driver Offset Compensation

The FRD to IR Converter provides Driver Offset controls to compensate for the physical displacement of the driver relative to the reference point (typically the front baffle center or another driver).

Parameters:

• X (Horizontal): Driver offset in the horizontal direction, in millimeters
• Y (Vertical): Driver offset in the vertical direction, in millimeters
• Located below the Extrapolation controls in the converter window

Coordinate System and Geometry:

Reference Point (Origin):

The reference point (X=0, Y=0) represents the desired acoustic axis of the loudspeaker system. In the diagram example:

• The tweeter is positioned at the origin (0, 0) — it defines the acoustic axis
• The woofer is offset by -200 mm vertically (0, -200)

Note for designers: The choice of origin is not necessarily the center of a driver — it’s an engineering decision. You could choose:

• The acoustic center of a specific driver (as shown in the example)
• A point between drivers (e.g., midway between tweeter and woofer)
• Any other reference point that makes sense for your design goals

What matters is consistency: all driver offsets are measured relative to this chosen origin.

Important: For off-axis measurements, only a driver with offset (woofer in the example) will have geometric delay applied. A driver at the origin (tweeter in the example) has no delay added or subtracted for off-axis angles — it remains the reference. This delay difference is then used to calculate interference patterns between drivers at off-axis angles.

How it works:

The driver offset introduces an additional geometric delay for off-axis measurements, calculated using the far-field approximation with Euclidean distance:

$$\Delta t = \frac{\sqrt{(x \cdot \sin\theta)^2 + (y \cdot \sin\varphi)^2}}{c}$$

Where:

• \(x\) = horizontal driver offset (mm)
• \(y\) = vertical driver offset (mm)
• \(\theta\) = horizontal measurement angle (degrees)
• \(\varphi\) = vertical measurement angle (degrees)
• \(c\) = speed of sound ≈ 343,000 mm/s

In practice (measurements in a single plane):

Since LinFIR only supports measurements in the horizontal plane (\(\theta \neq 0\), \(\varphi = 0\)) or vertical plane (\(\theta = 0\), \(\varphi \neq 0\)), the formula simplifies:

• Horizontal plane: \(\Delta t = \frac{x \cdot \sin\theta}{c}\) (only x offset matters)
• Vertical plane: \(\Delta t = \frac{y \cdot \sin\varphi}{c}\) (only y offset matters)

The Euclidean formula is used in the code to be mathematically correct and future-proof should oblique measurements be supported.

This geometric delay is added to the user-configured delay before converting the FRD data to an impulse response.

Example:

• Tweeter at origin (0, 0): no geometric delay applied at any angle
• Woofer offset: X = 0 mm, Y = -200 mm
• Measurement at vertical -30° (30° below axis)
• Geometric delay for woofer: \(200 \times \sin(30°)/343000 \approx 0.291\) ms
• This delay difference creates the correct interference pattern between tweeter and woofer at -30°

Purpose and Far-Field Approximation:

⚠️ Purpose: Driver offsets are used to predict far-field interference patterns between drivers when the listener is in the far field (listening distance >> driver spacing).

Key insight: Since typical FRD data has its phase recalculated (constant delay removed), the original measurement conditions don’t matter. What matters is modeling the acoustic geometry at the listening position.

When to use driver offsets:

• Listening distance ≥ 10× driver spacing (far-field condition for the listener)
• Example: For drivers spaced 200 mm apart, listener at ≥ 2 meters
• Typical home listening (2-4m) and studio monitoring (2-3m) scenarios
• Predicting directivity and off-axis response in real-world use

Geometric assumption:

The formula assumes acoustic rays are parallel at the listening position:

• Valid when listener distance >> driver spacing
• Path difference depends only on driver offset and angle

Near-field listening considerations:

For close listening distances (< 1m), the parallel ray approximation becomes less accurate for the direct sound, and the predicted response will deviate more from reality as you get closer to the drivers.

However, the approximation remains valid for the reflected sound field since walls, floor, and ceiling are typically far enough that the acoustic path length is much larger than driver spacing. This is important because off-axis directivity affects how sound interacts with room boundaries.

Best practice:

• Use driver offsets in most scenarios, including near-field listening
• The approximation is excellent for far-field listening (≥ 2m)
• For near-field use (< 1m), understand that direct sound predictions will be approximate, but reflected field modeling remains valid
• Leaving offsets at 0 mm corresponds to a coaxial driver configuration (all drivers at the same point)

Display Parameters

These parameters control visualization only and do not affect the conversion.

IR Start / Stop Time

Range: 0.0 ms to impulse duration
Purpose: Zoom into specific time region of the impulse response graph

Useful for examining:

• Pre-ringing characteristics
• Main peak detail

Frequency Min / Max

Range: 1 Hz to 48000 Hz
Purpose: Set frequency range for magnitude and phase graphs

Allows focusing on:

• Specific frequency bands of interest
• Problem areas in the response
• Comparison between original and interpolated data

Wrap Phase

Default: Disabled

Toggles between wrapped (-180° to +180°) and unwrapped phase display.

Wrapped:

• Phase limited to -180° to +180° range
• Shows discontinuities at ±180° boundaries
• Familiar format for some users

Unwrapped:

• Continuous phase (can exceed ±180°)
• Shows true phase accumulation
• Better for assessing group delay trends

Remove Delay Phase

Default: Disabled

Removes the linear phase component corresponding to the delay setting.

When enabled:

• Subtracts linear phase = -360° × f × delay
• Reveals underlying phase structure
• Useful for comparing final phase response with the original one

Data Quality Assessment

The converter displays comprehensive quality metrics to help you evaluate the FRD data.

Coverage Metric

Format: “X / Y frequency points (Z% coverage)”

• X: Number of frequency points in FRD file (within 0 Hz to Nyquist)
• Y: Expected number of points for complete FFT coverage
• Z: Coverage percentage = (X / Y) × 100

Quality indicators:

• ✅ Good (≥ 50%): Green - Sufficient data density
• ⚠️ Moderate (20-50%): Yellow - Acceptable but sparse
• ❌ Low (< 20%): Red - Very sparse, heavy interpolation required

FFT bin calculation: Based on sample rate and FFT size (4× next power of 2)

Info Button - Detailed Diagnostics

Click the ℹ Info button to open the Data Quality Information window with detailed analysis.

Missing DC (0 Hz):

• Impact: Low-frequency extrapolation required
• Consequence: Uncertainty in bass response below lowest measured frequency
• Solution: Use Roll-off extrapolation for DC

Missing Nyquist Data:

• Impact: High-frequency extrapolation required
• Consequence: Uncertainty in high-frequency response
• Recommended: FRD should extend to at least Nyquist frequency

Sparse Frequency Data:

• Threshold: < 50% coverage
• Impact: Heavy interpolation between measured points
• Risk: Artifacts from PCHIP interpolation of widely-spaced points
• Note: Shows exact percentage of missing data points

Pre-ringing Detection:

Analyzed automatically after conversion. Three severity levels:

• Minor (0.1% - 0.5% energy before peak):

  • Blue indicator
  • May indicate slight phase smoothing or measurement artifacts
  • Usually acceptable for most applications

• Moderate (0.5% - 1.0% energy before peak):

  • Yellow/Orange warning
  • Phase data likely smoothed or contains wrapping errors
  • Consider minimum phase transformation

• Significant (≥ 1.0% energy before peak):

  • Red warning
  • Severe phase corruption or non-physical characteristics
  • Strongly recommended: Enable Minimum Phase transformation
  • Will affect transient response and phase correction accuracy

Pre-ring duration: Time span from first significant energy to main peak

Physical interpretation: Real loudspeakers cannot produce energy before the stimulus arrives. Pre-ringing in FRD files indicates:

• Phase data smoothing in measurement software
• Phase wrapping/unwrapping errors
• File export artifacts
• Non-minimum phase processing applied to data

Impact Warnings

The Data Quality Information window provides context on how issues affect the conversion:

Missing Data Effects:

• Artifacts in reconstructed impulse response
• Estimated information through interpolation/extrapolation
• Final impulse may not fully represent original system

Best Practices for FRD Files:

• Include DC (0 Hz) measurement point
• Extend data to at least Nyquist frequency
• Aim for > 50% frequency coverage
• Prefer linear frequency spacing over logarithmic
• Avoid phase smoothing in export settings

Time-Domain Preference:

The converter emphasizes: When possible, import impulse responses (WAV files) instead of FRD.

Advantages of time-domain data:

• No interpolation artifacts
• Preserves time-of-flight information (with proper timing reference)
• Essential for proper driver alignment in crossover design
• Natural representation of system behavior

FRD limitations for crossover design:

• Often lacks time-of-flight data
• Driver motors may not be coplanar
• Diaphragm geometry affects acoustic center
• Horn loading adds group delay
• Relative delays between drivers are essential

Visualization Graphs

The converter displays three synchronized graphs:

1. FRD Data (Original + Interpolated):

• Left: Magnitude response
• Right: Phase response
• Red curve: Original FRD measurements
• Dashed blue curve: PCHIP interpolated curve

2. Impulse Response:

• Reconstructed time-domain impulse
• Zoom using IR Start/Stop controls
• Shows pre-ringing and overall IR structure

3. Reconstructed Frequency Response:

• Left: Magnitude computed from impulse via FFT
• Right: Phase computed from impulse via FFT
• Validation: Should closely match interpolated FRD (with Remove Delay Phase button enabled)
• Differences indicate reconstruction issues

Workflow

1. Import FRD file - Converter window opens automatically
2. Review quality metrics - Check coverage and warnings
3. Adjust extrapolation - Set DC and HF modes as needed
4. Set sample rate - Match or exceed project sample rate
5. Configure delay - Ensure adequate pre-ringing buffer
6. Enable minimum phase - If pre-ringing > 1% or phase corrupted
7. Inspect graphs - Verify interpolation and reconstruction quality
8. Click “Use This Impulse Response” - Apply to selected measurement angle

The converted impulse is automatically stored in the driver’s measurement at the specified horizontal/vertical angle.

Microphone Calibration

Apply frequency response and sensitivity calibration corrections for your measurement microphone.

Overview

Each driver or room measurement can be assigned a microphone calibration that compensates for:

• Frequency response deviation from ideal flat response
• Sensitivity factor (dB conversion from microphone output to SPL)

Calibration management:

• Calibration files are managed in Settings → Mic Calibration (see Settings → Mic Calibration)
• Default calibration can be set for new drivers
• Each driver/measurement can use a different calibration or none at all

Calibration Dropdown

Location: Below the Driver/Measurement Name field

Options:

• None: No calibration applied (use for reference microphones or pre-calibrated measurements)
• [Calibration file names]: All imported calibration files from Settings → Mic Calibration

Behavior:

• Selection is saved with the project file
• Calibration data is stored in application settings, not in project files
• Changing the selection immediately recomputes the driver’s windowed IR and all filter responses

How Calibration is Applied

Processing pipeline:

1. Measurement is imported and windowed (see Time Windowing)
2. Microphone calibration correction applied (if selected)
   • Minimum-phase FIR filter created from frequency response data
   • Sensitivity factor (dB) applied as gain adjustment
   • Frequencies below 5 Hz filtered out before processing
3. Corrected IR used for all subsequent filtering (crossovers, correction filters, etc.)

Phase behavior:

• Correction uses minimum-phase reconstruction, affecting both magnitude and phase
• This is physically accurate for measurement microphones, which exhibit mainly minimum-phase behavior

Real-time updates:

• Calibration is recomputed whenever you change the windowing parameters
• Filter responses update automatically when you change calibration selection

When to Use Calibrations

Apply calibration when:

• Using consumer or measurement microphones with non-flat response
• You have calibration data from the manufacturer or third-party calibration service
• Measurements were taken with the same microphone used for the calibration file

Use “None” when:

• Measurements were taken with an already-flat reference microphone
• Importing measurements already compensated externally

Per-Driver vs. Default Calibration

Per-driver assignment (this window):

• Overrides any default set in Settings
• Useful when combining measurements from different microphones in one project
• Saved with the project file

Default calibration (Settings → Mic Calibration):

• Automatically applied to newly created drivers/measurements
• Does not affect existing drivers
• See Settings → Mic Calibration for configuration

Time Windowing

Isolate the direct sound from an impulse response and exclude reflections or noise.

Note: Time windowing is not available in Room Calibration mode. This is intentional: Room Calibration mode is designed to capture the full speaker-plus-room response, including reflections and room decay. Applying a time window would produce an anechoic-like impulse response — corrections derived from it would be wrong for the actual listening environment — what you hear is the direct sound from the speakers combined with all the reflections that follow, and the correction filters must account for that full response.

Parameters

Start Time

Beginning of the time window in milliseconds.

• Purpose: Remove pre-ringing or early noise
• Typical: 0.0 to 5.0 ms before main peak
• Effect: Sets the beginning of the extracted portion

Stop Time

End of the time window in milliseconds.

• Purpose: Exclude late reflections or room decay
• Typical: 5.0 to 100.0 ms after main peak
• Effect: Sets the end of the extracted portion

Duration Display

Wndow duration (Stop Time - Start Time).

Auto Set Window

Click “Auto Set Window” to automatically detect and configure optimal window boundaries.

How It Works:

The algorithm analyzes the on-axis measurement (0°, 0°) and automatically sets the start and stop times:

1. Detect main peak: Locates the largest absolute amplitude in the impulse response
2. Compute Hilbert envelope: Calculates the amplitude envelope for robust reflection detection
3. Adaptive threshold: Adjusts detection sensitivity based on signal-to-noise ratio (3-8% of peak)
4. Set start time: Places window start 3 ms before the main peak (clamped to 0.0 minimum)
   • This margin accounts for off-axis measurements where the sound path may be shorter (gives a margin of ~1m)
   • Prevents truncation of the direct sound in off-axis measurements
5. Search for reflections: Scans the envelope for the first significant reflection/echo after the main peak:
   • Must occur at least 3 ms after the main peak
   • Must exceed the adaptive threshold
   • Must be a local maximum in the envelope
6. Set stop time: Places window stop 0.5 ms before the first detected reflection

If no significant reflection is detected, the stop time is set to the end of the search window or IR length.

Detection Method:

The algorithm uses Hilbert envelope analysis to detect reflections. This provides more robust detection than simple amplitude analysis by following the overall energy contour of the signal, making it less sensitive to oscillations within the waveform.

Window Margins:

• Start margin (3 ms): Ensures off-axis measurements are not truncated when rotating large loudspeakers. At the speed of sound (~343 m/s), 3 ms corresponds to approximately 1 meter of path length difference. During rotation, drivers may move closer to or further from the microphone—this margin provides sufficient safety for rotations of large cabinet.
• Stop margin (0.5 ms): Small margin before the detected reflection to avoid including the beginning of the echo.

Best Use Cases:

• ✅ Quasi-anechoic measurements: Ideal for measurements with clear, isolated reflections (e.g., outdoor, treated rooms, gated measurements)
• ✅ Controlled environments: Works well when the direct sound is clearly separated from reflections
• ✅ Multi-angle measurements: The 3ms start margin prevents truncation of off-axis measurements
• ✅ Time-saving: Quick initial window setup that can be manually refined if needed

Limitations:

• ❌ Reverberant environments: Not suitable for highly reflective rooms where reflections arrive continuously
• ❌ Dense reflection patterns: May not detect the first reflection correctly if multiple reflections arrive in quick succession
• ❌ Room measurements: For full-room measurements including reflections, manual windowing is more appropriate

Recommendation:

Use “Auto Set Window” as a starting point in quasi-anechoic conditions, then verify and adjust the window boundaries manually while observing the impulse response graph. In reverberant environments, set window parameters manually based on your measurement conditions.

Windowing Purpose

• Anechoic approximation: Simulate anechoic conditions from room measurements
• Reflection gating: Remove floor, ceiling, and wall reflections
• Crossover and correction design: Essential for accurate filter design using measured data
• Clean response: Isolate direct driver response

Window Function

LinFIR uses a Tukey window (tapered cosine/raised cosine) for smooth fade-in and fade-out transitions. This prevents spectral artifacts and high-frequency ringing caused by abrupt signal truncation.

Windowing Convention:

The fade-out starts at the stop parameter:

• Fade-in: [start → start + taper]
• Full signal: [start + taper → stop]
• Fade-out: [stop → stop + taper]

Taper Duration: 0.5 ms for both simple and adaptive windowing

Preview

The impulse response graph shows the windowed result in real-time as you adjust start and stop times. The window boundaries are visually indicated.

Adaptive Window

Advanced frequency-dependent windowing that preserves low-frequency response while gating high-frequency reflections.

How It Works

When enabled, LinFIR uses a wavelet transform to apply different window lengths for different frequencies:

• Start parameter: Applied uniformly to all frequencies
• Stop parameter: Applied with frequency-dependent extensions
  • Lower frequencies get longer decay times
  • Higher frequencies use shorter windows

Examples of automatic extensions:

• 30 Hz: +33 ms beyond stop time
• 480 Hz: +2 ms beyond stop time
• 7.5 kHz: +0.1 ms beyond stop time
• 18 kHz: +0.06 ms beyond stop time

This reflects the physical reality that longer wavelengths make it difficult to temporally separate direct sound from reflections.

Benefits

• Preserves bass response: Low frequencies automatically get the time they need
• Avoids bass roll-off: Short windows don’t unnaturally cut off bass
• Frequency-aware gating: Removes high-frequency reflections while keeping bass intact
• No phase distortion: Maintains natural phase relationships

When to Use

Use Adaptive Window when:

• Gating reflections but need to preserve bass response
• Using short measurement windows where bass would be compromised
• Simple windowing causes unnatural bass attenuation
• You want frequency-dependent decay times matching physical acoustics

Don’t use Adaptive Window when:

• Working with anechoic or quasi-anechoic measurements with very clean direct sound (simple window is cleaner)
• You have sufficient window length for all frequencies uniformly
• You prefer uniform windowing for consistency

Enabling

Check the “Adaptive Window” checkbox in the IR Windowing section.

Delay Compensation

Shift the impulse response earlier in time without truncating data.

How It Works

• Non-destructive: IR data remains intact in memory
• Display offset: Delay applied as metadata for display and calculations
• Purpose: Bring acoustic peak closer to time zero to flatten the phase

Auto Detect Delay

Click “Auto Detect Delay” to automatically detect and apply optimal delay compensation across all drivers:

Behavior:

1. Detects the peak position for each driver (based on on-axis measurements)
2. Finds the minimum detected delay across all drivers
3. Applies this minimum delay to all drivers simultaneously

Advantages:

• Preserves temporal alignment: All drivers maintain their relative timing relationships
• Removes common delay: Eliminates the shared propagation delay from all measurements
• Prevents errors: Ensures consistent delay compensation across all drivers automatically

Detection Method:

• Locates the position of the main peak in the impulse response
• Uses on-axis measurement (0°, 0°) as reference for each driver
• Aligns all driver peaks precisely at time zero

Manual Adjustment

Manually enter delay value in milliseconds to fine-tune compensation for individual drivers.

Sync Toggle (enabled by default):

When the Sync toggle is enabled, any manual delay adjustment automatically applies to all drivers in the project. This ensures consistent compensation across all drivers without needing to adjust each one individually.

• Enabled: Changes to the delay parameter propagate to all drivers automatically
• Disabled: Changes only affect the current driver, allowing independent delay values
• Activating Sync: If Sync is currently disabled and you enable it, an automatic delay detection is triggered to ensure all drivers have coherent delay values

When to use independently (Sync disabled):

• Only when drivers were measured with significantly different microphone positions
• Even then, prefer fixing microphone position and re-measuring rather than relying on delay compensation

Purpose and Limitations

What delay compensation is FOR:

• Removing the time of flight of sound from the measurement (optional, depends on personal preference)
• Correcting for different microphone positions between measurements

What delay compensation is NOT for:

• ❌ Time-aligning drivers during crossover design (this is handled separately by the Time Delay parameter)
• ❌ A tool for precise multi-driver coherence (proper crossover phase design is required instead)

Important: Microphone Position Corrections

Delay compensation can correct for microphone position differences, but this method has significant limitations:

⚠️ High-frequency measurement errors scale with wavelength:

• 10 cm microphone position difference → Errors above ~3400 Hz
• 5 cm microphone position difference → Errors above ~6900 Hz
• Ultra-short wavelengths (high frequencies) amplify measurement uncertainties

Best practice:

• ✅ Prefer re-measuring with fixed microphone placement rather than compensating for position differences
• ✅ Maintain consistent microphone position across all measurements when possible
• ✅ Delay compensation cannot replace careful measurement technique

Off-Axis Measurements:

• The selected delay compensation applies uniformly to all off-axis measurements (cannot vary by angle)
• This affects the relative timing between different measurement angles consistently

Measurement Notes

Section: Collapsible header (default closed)
Purpose: Document measurement conditions and hardware setup

Overview

The Measurement Notes field provides a space to record important information about your measurement setup, hardware configuration, environmental conditions, and any other relevant details.

Features

• Multi-line text editor: Supports up to 1000 characters
• Auto-save: Notes are saved with the project
• Hint text: Helpful placeholder text suggests what information to include
• Character counter: Live display of character count (e.g., “127/1000 characters”)

What to Document

Hardware configuration:

• Microphone model and serial number
• Audio interface model and settings
• Amplifier configuration
• Cable connections and routing

Measurement conditions:

• Date and time of measurement
• Room conditions (temperature, humidity if relevant)
• Microphone position and distance
• Driver orientation and mounting

Reference information:

• Calibration file version used
• Sample rate and sweep settings
• Any unusual conditions or observations
• Version of measurements for tracking

Example Notes

Microphone: UMIK-1 (S/N: 123456) with cal file v2.3
Interface: Focusrite Scarlett 2i2 (48kHz, 512 buffer)
Distance: 1m on-axis, driver mounted in baffle
Room: Semi-anechoic (windows gated at 10ms)
Date: 2025-12-14
Notes: First measurement after driver break-in period

Best Practices

• Be specific: Include model numbers and serial numbers when relevant
• Date your measurements: Track when measurements were taken
• Note calibration: Record which calibration file was used
• Document changes: If re-measuring, note what changed
• Reference conditions: Record anything that might affect comparisons

Exporting Impulse Responses

Impulse responses can be exported individually or in batch using two methods:

Individual Export (Per-Measurement)

For each measurement (horizontal and vertical angles):

1. Click the ⬇ (Export) button next to the measurement angle
2. Export dialog appears with format options
3. Choose export format (WAV or TXT) and options
4. File is saved with automatic angle suffix

File naming format:

• Format: DriverName_v{vertical}h{horizontal}.extension
• Examples:
  • On-axis (0°, 0°): Driver 1_v0h0.wav
  • Horizontal +30°: Driver 1_v0h30.wav
  • Vertical -15°: Driver 1_v-15h0.wav
  • Horizontal -45°: Driver 1_v0h-45.wav

Room Calibration mode (no angles):

• Format: MeasurementName.extension
• Example: Measurement 1.wav

Batch Export (All Measurements)

Button: ⬇ Export All (above “Measurement angle” field)

Process:

1. Click “⬇ Export All” button
2. Select output directory in folder picker
3. All measurements exported simultaneously as WAV files
4. Files use consistent v{vertical}h{horizontal} naming

Format (batch export):

• All files exported as WAV (32-bit float, mono)
• Sample rate matches project sample rate
• “Include distortion” option applies to all files
• No format selection (always WAV for batch)

Benefits:

• Fast export of complete measurement sets
• Consistent naming for re-import compatibility
• Files can be re-imported via batch import with automatic angle detection
• Compatible with external tools

Use cases:

• Archiving complete measurement sets
• Sharing measurements with collaborators
• External processing (FIR Designer, Matlab, REW)
• Backing up measurements before making changes

Export Format Options

When you click the export button, a dialog presents two format options:

WAV Format

File format:

• Channels: Mono
• Bit depth: 32-bit PCM float
• Sample rate: Project sample rate

Include Distortion Metadata option:

When checked:

• Exports full deconvolved IR plus custom ‘dist’ metadata chunk
• File name automatically gets _dist.wav suffix
• Enables THD computation when re-imported into LinFIR
• Contains sweep parameters: start frequency, end frequency, duration

When unchecked:

• Exports only the active (cropped/windowed) working IR
• Standard WAV file without metadata
• Cannot be used for THD computation upon re-import

TXT Format

File format:

• One sample per line in scientific notation (e.g., 1.234567e-03)
• Metadata stored as comment lines starting with #:
  • # fs=<value> - Sample rate in Hz
  • # IR data: - Marks the start of sample data
• Compatible with external tools (e.g., Matlab, Python)
• Can be re-imported into LinFIR

Include Distortion option:

TXT export does NOT support distortion export:

• The “Export as TXT” button is disabled when “Include distortion” is checked
• TXT format can only export the cropped IR (main impulse only)
• For distortion data export, use WAV format which supports embedded metadata

THD Availability

Total Harmonic Distortion (THD) graphs are only available when:

• IR was captured directly via LinFIR’s built-in sweep, or
• IR was imported from WAV with ‘dist’ metadata chunk (exported with “Include dist.” option in LinFIR)

Plain WAV files without ‘dist’ chunk or TXT files cannot produce THD curves.

Best Practices

For Captures

• Use proper gain staging to avoid clipping
• Monitor levels during sweep - aim for -6 dB to -12 dB peak
• Save captures with “Include dist.” for future THD analysis

For Windowing

• Use anechoic or quasi-anechoic measurements when possible
• Apply windowing before designing correction filters
• Check impulse response graphs to verify proper window placement
• Use Adaptive Window cautiously - verify bass response is realistic

For File Management

• Use descriptive IR names for organization
• Export with distortion metadata if you need THD analysis later
• Keep original unprocessed captures as backups
• Document measurement conditions (position, distance, etc.)

Quick Reference

Sweep Measurements

LinFIR features a built-in exponential sine sweep generator for impulse response capture. This system provides high-quality measurements with automatic quality validation.

Overview

The sweep measurement system generates an exponential sine sweep that excites the system across a specified frequency range. After playback and recording, deconvolution with the inverse sweep extracts the impulse response.

Accessing Sweep Controls

Sweep controls are located in the IR Management window:

• To open the window: Driver → Manage IR
• Section: Measurement Settings
• Start button: 🎤 Measure

Sweep Parameters

⚠️ Safety Warning

Low frequency content can damage small drivers:

• Sweeps starting below 100 Hz contain high-energy bass content
• Never measure unprotected tweeters or midrange drivers with full-range sweeps
• High excursion at low frequencies can cause mechanical damage or voice coil failure
• For tweeters: Use start frequency >1000 Hz
• For midrange: Use start frequency >200-500 Hz (depending on driver specs)
• Always verify your driver’s rated frequency range before measurement

When measuring individual drivers, adjust the start frequency to match the driver’s safe operating range.

Important note on driver protection methods:

If you use an external high-pass filter (physical or DSP) to protect a driver during measurement instead of adjusting the sweep start frequency, be aware that:

• The filter will affect the phase response and group delay of the measurement
• This phase shift will be captured in the impulse response
• If you later remove or change the protection filter, the driver alignment will be incorrect
• Recommendation: Protect drivers by setting the appropriate sweep start frequency rather than using external filters
• This ensures accurate phase and timing measurements without filter-induced artifacts

Duration

Length of the sweep signal in seconds.

• Range: 1 to 15 seconds
• Shorter (1-3s): Faster measurements, lower frequency resolution, reduced harmonic distortion accuracy
• Longer (5-15s): Better low-frequency resolution, improved SNR, better harmonic distortion measurement
• Recommended: 5-7 seconds for full-range measurements

Benefits of longer sweeps:

• Improved signal-to-noise ratio for harmonic distortion analysis
• More reliable detection and measurement of weak harmonics
• Better frequency resolution in the bass region

⚠️ Important: Longer sweeps deliver more energy to the driver at low frequencies because exponential sweeps spend more time in the bass region. For drivers with limited excursion capability, longer sweeps may require:

• Raising the start frequency to avoid the lowest frequencies
• Reducing the sweep level (amplitude) to limit excursion
• Monitoring the driver during measurement to ensure it doesn’t bottom out

Amplitude

Output level of the sweep signal.

• Range: -50 to 0 dB
• Purpose: Control the acoustic level of the measurement
• Typical: -12 to -6 dB for moderate levels
• Maximum (0.0): Full digital scale output

Adjust amplitude based on driver sensitivity and desired measurement level. Too low may result in poor SNR; too high may cause clipping, distortion or damage.

Start Frequency

Cutoff frequency for the 4th-order Butterworth high-pass filter applied to the sweep for driver protection.

• Default: 1 Hz (captures near-DC response)
• Range: 1 Hz to 20 kHz
• Full-range: 20 Hz (typical for full-bandwidth measurements)
• Subwoofers: 10-20 Hz for low-frequency drivers
• Tweeters: >1000 Hz for high-frequency analysis only (protect from low frequency damage)
• Midrange: >200-500 Hz depending on driver specifications

Understanding the Butterworth Filter Protection

LinFIR applies a 4th-order Butterworth high-pass filter to the sweep signal with cutoff frequency fc = Start Frequency. This provides:

• Attenuation at cutoff: -3 dB at the Start Frequency
• Roll-off below cutoff: -24 dB/octave (deterministic, frequency-independent)
• Minimal pre-ringing: Butterworth maximally-flat passband characteristic
• Continuous protection: Applied across the entire sweep duration

⚠️ Critical positioning guideline:

Because of the -3 dB attenuation at the cutoff frequency, set Start Frequency slightly below your desired passband start, but always above driver safety limits:

• Example: For 80 Hz passband → set Start Frequency to 60-70 Hz
• This accounts for the -3 dB attenuation while ensuring full response in the desired band
• For tweeters: position above resonance frequency (fs) to avoid damage
• For woofers: set at a frequency that limits excursion to safe levels

⚠️ Important: Setting the start frequency too low for small drivers can cause mechanical damage. Always verify your driver’s rated frequency range and adjust accordingly.

For full-range measurements of complete speaker systems, 20 Hz is appropriate. For individual driver measurements, adjust to the driver’s safe operating range.

High Frequency (Automatic)

The high frequency limit is automatically calculated to optimize harmonic distortion measurement quality.

• Calculation: Based on sample rate - set to 1/48 octave before Nyquist frequency
• Range: Typically ~23.5 kHz at 48 kHz sample rate, ~47 kHz at 96 kHz
• Display: The actual calculated frequency is shown in the UI (read-only)
• Optimization: Ensures best separation and detectability of harmonic components (H2-H4)

Why automatic? The high frequency is chosen to provide optimal spacing between the fundamental and its harmonics in the time domain, ensuring accurate distortion measurement. Manual adjustment is not necessary and could compromise measurement quality.

Reset Start Freq Button

Quickly restore the start frequency to its default value:

• Action: Sets Start Freq to default frequency (configured in Audio Settings)
• Use: Return to default start frequency after using restricted frequency ranges for driver protection

Driver Protection and Automatic Windowing

LinFIR applies a sophisticated combination of 4th-order Butterworth high-pass filtering and frequency-domain cosine windowing to protect drivers while ensuring clean measurements.

Butterworth High-Pass Filter for Driver Protection

A 4th-order Butterworth high-pass filter with cutoff frequency fc = Start Frequency is continuously applied across the entire sweep for deterministic driver protection:

Filter Characteristics:

• Cutoff frequency (fc): Set by the Start Frequency parameter
• Attenuation at fc: -3 dB (Butterworth characteristic)
• Roll-off: -24 dB/octave below cutoff (4th order)
• Pre-ringing: Minimal (Butterworth provides best balance)

Why Butterworth?

• Provides deterministic, frequency-independent protection slope
• Minimal pre-ringing compared to steeper filters
• Smooth passband without ripple
• Predictable behavior for all driver types

Protection Mechanism:

• Exponentially reduces energy below the cutoff frequency
• Protects tweeters from low-frequency voltage (thermal damage)
• Limits woofer excursion at sub-bass frequencies (mechanical damage)
• Applied in frequency domain for continuous, artifact-free protection

Understanding the Need for Windowing

When a sweep signal starts or ends abruptly at specific frequencies, it creates discontinuities in the time domain. These discontinuities manifest as Gibbs oscillations in the frequency domain near sharp cutoffs.

These artifacts are not physical - they’re mathematical consequences of abrupt spectral transitions and can mask real acoustic behavior.

How Automatic Windowing Works

LinFIR combines the Butterworth filter with smooth cosine windowing at the boundaries:

Low-Frequency: Butterworth + Cosine Fade-In:

• Butterworth filter (fc = Start Frequency) provides continuous -24 dB/octave protection
• First 1/8 octave uses cosine taper combined with Butterworth to force clean 0 dB at sweep start
• Smooth transition eliminates pre-ringing artifacts
• Combined protection: cosine windowing × Butterworth filtering

High-Frequency Fade-Out:

• Begins at the automatically calculated high frequency (1/48 octave before Nyquist)
• Gradually decreases amplitude using a cosine taper
• Physical sweep extends up to 1/96 octave before Nyquist frequency
• Optimized to maximize harmonic distortion SNR

Adaptive Behavior:

• Fade-out is always applied for optimal high-frequency behavior
• This ensures the sweep always covers its full intended bandwidth with continuous protection

Benefits of Combined Protection

Driver Safety:

• Deterministic -24 dB/octave roll-off below Start Frequency
• Continuous protection without sudden energy changes
• Suitable for all driver types with appropriate frequency selection

Clean Impulse Responses:

• Well-defined peaks without oscillations
• Minimal pre-ringing and post-ringing
• Accurate representation of system behavior

Optimized for Distortion Analysis:

• High frequency chosen to maximize harmonic separation
• Better signal-to-noise ratio for harmonic distorsion detection

No Configuration Required:

• Works optimally for all measurement scenarios
• Eliminates need for user windowing decisions
• Consistent, predictable behavior

Best Practices for Frequency Range Selection

Start Frequency: Driver Protection

The Start Frequency is critical for protecting drivers from damage. Loudspeakers can be damaged by two mechanisms:

1. Over-excursion (mechanical damage)

• Occurs when the cone reaches its physical limits and “bottoms out”
• Results in distortion, mechanical stress, and potential permanent damage
• Most critical for medium to large woofers with high motor force

2. Voice coil overheating (thermal damage)

• Low frequencies behave like DC current, causing excessive heat buildup
• Most critical for midrange drivers and tweeters
• Sending bass frequencies to a tweeter is equivalent to feeding it DC current, which can burn the voice coil
• Small voice coils have limited thermal mass and dissipate heat poorly

Recommended approach by driver type:

Woofers and Subwoofers:

• Adjust Start Frequency to avoid over-excursion
• Use driver simulations to predict excursion limits (e.g., VituixCAD, WinISD)
• Start frequency should be below or at the driver’s resonance frequency for accurate measurement
• Example: 12“ woofer with 15mm Xmax → start at 20-30 Hz to avoid mechanical limits

Tweeters:

• Place Start Frequency well above the driver’s resonance frequency
• Avoid exciting the resonance peak which causes extreme excursion and heat
• Must still measure the useful passband including the crossover region
• Example: Tweeter with 800 Hz resonance and 2 kHz crossover → start at 1000-1500 Hz
• This captures the crossover transition while protecting against resonance excitation

Midrange drivers:

• Balance between capturing low-frequency rolloff and avoiding resonance excitation or over-excursion
• Typically start 100-300 Hz below the intended crossover frequency
• Monitor excursion if measuring near resonance frequency
• Example: Midrange crossing at 500 Hz with 200 Hz resonance → start at 300-400 Hz

General rule:

• Always verify your driver’s specifications before measurement
• When in doubt, start conservative (higher frequency) and lower gradually while monitoring

Starting a Measurement

Measure Button (🎤 Measure)

Clicking the 🎤 Measure button initiates the sweep measurement process.

Default behavior (confirmation enabled):

1. Click 🎤 Measure
2. Confirmation dialog appears with warnings:
   • Check microphone and speaker positioning
   • Set appropriate gain levels to avoid clipping
   • The sweep will be audible
   • Windows-specific timing warning (if applicable)
3. Click Start Measurement to proceed or Cancel to abort
4. Sweep plays and records automatically
5. IR is processed and validated

Direct mode (confirmation disabled):

• Measurement starts immediately when clicking 🎤 Measure
• No confirmation dialog shown
• Useful for quick repeated measurements

Disabling Confirmation Dialog

To disable the confirmation dialog:

1. Open Settings (Cmd/Ctrl + ,)
2. Navigate to Audio Settings
3. Enable “Skip confirmation dialog” option
4. Measurements will now start immediately

Warning: Disabling confirmation means sweeps start instantly. Ensure your setup is ready before clicking the button to avoid unexpected loud sweeps.

Stopping a Measurement

Emergency Stop

You can interrupt a sweep at any time during capture:

• Stop Capture button: Click the red button that appears during measurement
• Spacebar: Press the spacebar key for instant interruption

When to use emergency stop:

• Incorrect parameters: Realized the frequency range or level is inappropriate for the driver
• Excessive gain: Input signal is clipping or distorting
• Driver protection: Driver showing signs of stress (unusual sounds, excessive excursion)
• Environmental issues: Unexpected noise or interference occurred

The measurement will stop immediately when interrupted. You can then adjust parameters and start a new measurement.

Automatic Stop on Clipping

LinFIR monitors the input signal in real time during the entire sweep and will automatically cancel the measurement the moment clipping is detected — without waiting for the sweep to finish.

What happens:

• The input callback continuously tracks the peak level of every audio buffer
• As soon as any sample reaches digital full-scale (±1.0), the sweep is immediately faded out and stopped
• A notification toast identifies the cancellation as clipping-triggered

Toast messages:

• “Measurement cancelled: clipping detected! Reduce the output level or input gain and try again.” — clipping occurred on the main measurement channel
• “Measurement cancelled: clipping on the timing reference channel! Reduce the timing reference output level.” — clipping occurred on the timing reference channel (Electric or Acoustic mode); detected in the first ~1.2 s of the sweep, before the main sweep begins

Why this matters:

• Without real-time detection, clipping is only discovered at the end of the sweep after wasting the full measurement duration
• Immediate stop prevents unnecessary acoustic exposure of the driver to a potentially damaging clipped signal
• The automatic stop uses the same graceful fade-out as the manual Stop Capture button — no abrupt clicks

What to do:

• Reduce the audio interface input gain
• Lower the sweep amplitude
• Move the microphone farther from the driver
• For timing reference clipping: lower the timing reference output level in Audio Settings

Automatic Early Stop on Weak Timing Reference Signal

When a timing reference is active (Electric or Acoustic mode), LinFIR checks the level recorded on the timing reference channel shortly after the chirp ends, and cancels automatically if the signal was too weak to be reliable.

What happens:

• During the chirp window, the peak level on the timing reference input channel is continuously tracked
• After the end of the chirp, the accumulated peak is evaluated
• If the peak never exceeded −25 dBFS, the measurement is cancelled immediately (before the main sweep even begins)
• Toast message: “Measurement cancelled: timing reference signal too low (X.X dBFS < −25 dBFS). Check the timing reference channel connection and level.”

Why this matters:

• A weak timing reference signal makes latency detection unreliable or impossible
• Catching the problem immediately — before the full sweep — avoids wasting the measurement time

What to do:

• Check that the timing reference channel is connected correctly
• Increase the timing reference output level in Audio Settings
• For Acoustic mode, verify that the microphone can capture the high-frequency chirp (5–20 kHz)

Automatic Early Stop on Low Signal (Multi-Average)

When a measurement is configured with 2 or more averages, LinFIR checks the signal level at the end of the first sweep — before the second sweep begins — and cancels automatically if the signal was too weak.

What happens:

• During the first sweep, the peak input level is continuously tracked
• ~200 ms into the post-sweep silence (after the sweep signal has ended), the accumulated peak is evaluated
• If the peak never exceeded −25 dBFS, the measurement is cancelled before the second average starts
• Toast message: “Measurement cancelled: signal too low on first sweep (< −25 dBFS). Increase input gain or move the microphone closer.”

Why this matters:

• A low-level signal produces a poor signal-to-noise ratio; averaging more sweeps will not fix it
• Stopping after the first sweep avoids wasting time on subsequent averages that would yield unusable results

What to do:

• Increase the audio interface input gain
• Move the microphone closer to the driver
• Increase the sweep amplitude (Output Level in Sweep Parameters)

Note: The multi-average level check requires ≥ 2 averages. For single-sweep measurements, the level check occurs after the full capture completes and the result is simply rejected if too low. The timing reference level check (above) applies regardless of the number of averages.

Window Behavior During Capture

To ensure reliable interruption, the IR Management window:

• Remains on top of all other windows during measurement
• Maintains keyboard focus throughout the capture
• Returns to normal window behavior when measurement completes or is stopped

This prevents accidental loss of control during the sweep. The spacebar shortcut will always work, even if you accidentally click elsewhere during measurement.

Level Recommendations

Target levels for best measurement quality:

• Optimal: -6 dB to -12 dB
• Acceptable: -12 dB to -20 dB
• Too low: Below -25 dB (poor SNR, noisy measurements)
• Too high: Above -3 dB (risk of clipping)

Quality Validation

LinFIR automatically validates each capture and rejects poor-quality measurements.

Clipping Detection

• Criteria: Any sample reaching digital full-scale (±1.0)
• Result: Capture rejected immediately
• Reason: Clipping introduces harmonic distortion that corrupts the impulse response

Solution: Reduce input gain or lower acoustic level (move microphone farther or reduce driver volume).

Level Check

• Criteria: Peak level must be ≥ -25 dB
• Result: Capture rejected if too low
• Reason: Low levels result in poor signal-to-noise ratio

Solution: Increase input gain, raise driver volume, or move microphone closer.

Best Practices

Microphone Positioning

• On-axis measurements: Position microphone directly in front of driver at typical listening distance
• Listening position: For room measurements, place microphone at primary listening position
• Height: Maintain consistent height, typically at seated ear level (90-150 cm)
• Distance:
  • Farfield recommended: 0.5-1.5 meters with windowing (quasi-anechoic technique)
  • Avoid nearfield: Nearfield measurements (<20cm) capture reactive near-field effects (evanescent waves) that don’t propagate to listening position
  • Consistency critical: Use the same microphone position for all drivers to maintain correct relative phase and amplitude relationships

Why Avoid Nearfield for Multi-Way Systems?

Physical limitations:

• At different distances, the reactive impedance field varies between drivers
• Nearfield captures evanescent waves (reactive air-driver coupling) that decay exponentially and never reach the listener
• These reactive effects create phase rotations and magnitude errors that don’t represent real acoustic radiation

Practical consequences:

• Impossible to maintain consistent distance across drivers of different sizes (tweeter vs woofer require different nearfield distances)
• Phase relationships corrupted: Crossover alignment becomes unreliable when drivers measured at different distances
• Baffle effects missed: Nearfield ignores diffraction and baffle step that dominate the actual system response

Recommended approach:

• Measure at 0.5-1m distance (same position for all drivers)
• Use time windowing to eliminate room reflections (quasi-anechoic)
• Captures correct far-field radiation including baffle interactions
• Maintains phase coherence between drivers for accurate crossover design

Acoustic Environment

• Minimize noise: Turn off HVAC, fans and noisy appliances during measurement
• Reduce reflections: For anechoic-like measurements, gate reflections using time windowing
• Room measurements: Accept reflections, use time windowing to exclude late reflections only
• Outdoor measurements: Recommended to avoid room gain and obtain better results at low frequencies

Measurement Microphone

• Type: Use calibrated measurement microphone (e.g., UMIK-1, Earthworks M23, Beyerdynamic MM-1)
• Calibration: Load manufacturer calibration file before measurements
• Quality: Consumer microphones lack flat frequency response and will give inaccurate results

Gain Staging

1. Start with moderate input gain
2. Run a test sweep
3. Check peak level indicator
4. Adjust gain to achieve -6 dB to -12 dB peaks
5. Re-measure if clipping or too low

Sweep Duration

• Full-range (20 Hz - 20 kHz): 5-7 seconds recommended
• Limited bandwidth: Shorter sweeps acceptable (2-3 seconds)
• Very low frequencies (< 20 Hz): Use longer sweeps (7-10 seconds)

Capture Rejection Messages

“Capture Rejected - Clipping Detected”

Cause: Input signal exceeded digital full-scale.

Solutions:

• Reduce audio interface input gain
• Lower driver volume
• Move microphone farther from driver
• Check gain staging in signal chain

“Capture Rejected - Signal Too Low”

Cause: Peak level below -25 dB threshold.

Solutions:

• Increase audio interface input gain
• Raise driver volume
• Move microphone closer to driver
• Verify microphone connection and phantom power
• Check audio routing and device selection

Advanced Topics

Sweep Output Channel

In the IR Management window, you can select which output channel plays the sweep:

• Purpose: Direct sweep to specific amplifier channel
• Room Calibration: Select channel connected to main loudspeaker (excluding timing reference)
• Multi-way systems: Measure each driver individually by routing to appropriate channel

Multiple Measurements

For averaging or multi-position captures in Room Calibration mode:

1. Configure timing reference (Electric or Acoustic) in Audio Settings
2. Capture first measurement (this becomes the timing reference)
3. Capture subsequent measurements at different positions or conditions
4. In Room Calibration mode, measurements are automatically aligned using GCC-PHAT

Measurement Averaging

Averaging works on all platforms without requiring a timing reference:

• LinFIR uses a continuous capture approach: one recording containing N consecutive sweep segments
• After alignment, segments are extracted and averaged to reduce noise
• Operating system scheduler jitter only affects the initial timing, not the relative timing between segments

Timing reference recommendation:

• Not required for averaging (continuous capture eliminates timing inconsistency between segments)
• Still recommended on Windows for absolute timing accuracy if comparing measurements across drivers
• Mac OS: System clock is already reliable for all timing needs

Technical Details

Exponential Sine Sweep (Farina Method)

LinFIR uses an exponential sine sweep (ESS) for impulse response measurements, based on Angelo Farina’s method:

Basic ESS Properties:

• Frequency distribution: Equal energy per octave
• Low-frequency emphasis: More time spent at low frequencies
• Deconvolution: Analytical inverse filter applied to extract impulse response

Harmonic Distortion Detection:

The exponential sweep provides natural time separation of harmonic distortion products:

• Deterministic harmonic positions: All harmonic distortion products (H2, H3, H4) appear at predictable time positions in the deconvolved impulse response
• Time separation formula: Δt(Hₙ) = (T/ln(f₂/f₁)) × ln(n)
  • H2 (2nd harmonic) appears at: Δt × ln(2) ≈ 0.693 Δt before the main peak
  • H3 (3rd harmonic) appears at: Δt × ln(3) ≈ 1.099 Δt before the main peak
  • H4 (4th harmonic) appears at: Δt × ln(4) ≈ 1.386 Δt before the main peak

Deconvolution

After recording the sweep response LinFIR automatically:

1. Compute the inverse sweep signal
2. Convolve recorded response with inverse sweep
3. Extract impulse response from convolution result
4. Apply microphone calibration (if loaded)
5. Validate quality (clipping and level check)

Quality Criteria

• Clipping: Zero tolerance - any sample at ±1.0 rejects capture
• Minimum level: -25 dB peak threshold ensures adequate SNR
• These are conservative: Ensure high-quality measurements

Workflow Example

1. Configure Audio: Set input/output devices, sample rate, buffer size
2. Load Calibration: Import microphone calibration file (if available)
3. Position Microphone: Place at measurement location
4. Set Sweep Parameters: Duration = 5s, Amplitude = -6 dB, Full range (20 Hz - 22 kHz)
5. Test Sweep: Run a test to check levels
6. Adjust Gain: Aim for -6 dB to -12 dB peak levels
7. Measure: Click 🎤 Measure button to record impulse response
8. Confirm: Click Start Measurement in the confirmation dialog (or skip if disabled in settings)
9. Verify: Check impulse response graph for quality
10. Repeat: Capture additional measurements as needed

Driver Processing

Each driver in LinFIR can be individually processed with a comprehensive signal chain combining FIR crossovers, FIR correction filters, IIR equalization, polarity inversion, and time/gain adjustments.

The processing chain for each driver follows this order:

Driver Management

Adding and Removing Drivers

Add Driver:

• Maximum: 10 drivers per project
• Click ➕ Add Driver button in the Drivers toolbar
• New driver initialized with default settings

Remove Driver:

• Click 🗑 button in the driver’s bottom action bar
• Only visible when more than one driver is present
• Hidden in Hypex mode (driver count is locked by the selected model)

Copy / Paste Driver Parameters:

• Click 📋 to copy all parameters of a driver to the clipboard
  • IRs and driver name are not copied — only filter and DSP settings
  • Copied parameters include: FIR/IIR filters, gain, time delay, polarity, tap lengths, windowing, correction settings, mic calibration, and sweep parameters
• Click 📥 to paste the clipboard parameters onto another driver
  • The target driver’s IRs and name are preserved
  • The 📥 button is grayed out when no parameters have been copied yet
• Useful for duplicating crossover or EQ settings across multiple drivers without redoing each one manually

Driver Organization:

• Drivers are organized in collapsible sections in the left toolbar
• Each section shows the driver name (or “Driver 1”, “Driver 2”, etc. if unnamed)
• Driver names are automatically set when importing impulse responses

Driver Controls

Enable/Disable and Solo

Enable Toggle:

• Button to enable/disable individual drivers
• Disabled drivers are excluded from calculations and the summed response
• Useful for A/B testing or isolating specific drivers

Solo Button:

• Instantly disables all other drivers and enables only the selected driver
• Quick way to review individual drivers in isolation

All Button:

• Re-enables all drivers simultaneously
• Convenient after using Solo mode

Driver Interface Overview

Each driver’s interface provides access to all processing and configuration options, organized into three main sections:

• Manage IR: Import/capture impulse responses and configure windowing
• FIR Filters: Low-pass, high-pass, correction filters, and FIR offset delay (if enabled in global settings)
• IIR Filters: Parametric EQ and filter design
• Driver Adjustments: Polarity inversion, gain, and time delay adjustments

The following sections detail each of these components.

Impulse Response Management

Loading Impulse Responses

Click Manage IR to open the impulse response management window for:

• Importing pre-recorded impulse responses
• Capturing new measurements via audio interface
• Configuring IR windowing parameters
• Managing off-axis measurements for directivity analysis

See IR Management for detailed documentation.

IR Status Indicator:

• Gray “No IR loaded”: Driver has no impulse response
• Green “✅ IR loaded”: Valid impulse response loaded

⚠️ Note: A loaded impulse response is required for filter design and correction. Without an IR, only theoretical filter responses can be displayed.

Target DSP Settings

The Target DSP Settings section configures fundamental parameters that determine filter design constraints and export characteristics. These settings directly impact frequency resolution, latency, and compatibility with your target DSP platform.

⚠️ Note: You can change these settings at any time to match different DSP characteristics, but doing so will affect the precision, effectiveness, delays, and slopes of any FIR filters you have already set.

Sample Rate

Options: 44.1 kHz, 48 kHz, 88.2 kHz, 93.75 kHz, 96 kHz, 176.4 kHz, 192 kHz
Current project: Fixed at project creation

The sample rate is set when creating a new project and cannot be changed afterward. It determines:

• Nyquist frequency: Maximum representable frequency (sample rate / 2)
• Filter frequency resolution: Higher rates provide finer resolution
• Latency: For a given tap count, higher sample rates have lower latency in milliseconds
• DSP compatibility: Must match your target DSP platform’s operating sample rate

Common choices:

• 44.1 kHz: Compact DSP platforms with limited processing power
• 48 kHz: Professional audio standard, most DSP platforms
• 88.2 kHz: High-resolution audio (2× CD rate, relatively uncommon for DSP)
• 93.75 kHz: Hypex DSP platforms (FA123, FA253, FA502, etc.)
• 96 kHz: High-resolution audio, some professional DSP platforms
• 176.4/192 kHz: Ultra-high resolution - only when DSP platform specifically requires it

Enable per-driver tap lengths

Default: Disabled (shared tap length applies to all drivers)

When disabled (default):

• Single Filter length (taps) control in Target DSP Settings applies to all drivers
• Single Export padding control in Target DSP Settings applies to all drivers
• Simpler configuration for uniform systems
• All drivers use the same FIR filter length

When enabled:

• Filter length (taps) and Export padding controls move from Target DSP Settings to individual driver sections
• Each driver gets its own controls in its collapsible section
• Allows optimization: fewer taps for tweeters, more for woofers/subwoofers
• Useful for minimizing DSP load while maintaining quality where needed

Example use case: Use 2048 taps for subwoofer (needs low-frequency resolution), 512 taps for tweeter (high frequencies need less resolution).

⚠️ Note: Linear phase FIR filters and phase-correction FIR filters introduce noticeable delays that depend on the number of taps. Different tap lengths between drivers will likely introduce time misalignments, which you can correct using the Time Delay parameter in the Driver Adjustements section.

Filter Length (Taps)

Range: 32 to 65536 taps
Default: 512 taps
Location:

• Shared mode (per-driver tap lengths disabled): In Target DSP Settings section - applies to all drivers
• Per-driver mode (per-driver tap lengths enabled): In each driver’s collapsible section

Controls the length of all FIR filters for this driver (crossover and correction filters).

Frequency Resolution:

• Resolution (Hz) = Sample Rate / Tap Count
• Example at 48 kHz: 512 taps = 93.75 Hz resolution, 4096 taps = 11.72 Hz resolution
• Lower frequencies need more taps for precise control

Latency (Linear-Phase):

• Latency = (Taps / 2) / Sample Rate
• Example at 48 kHz: 512 taps = 5.33 ms, 4096 taps = 42.67 ms
• Can be reduced using Causality control at the cost of phase linearity

Practical guidelines:

• 256-512 taps: Tweeters, minimal latency applications
• 512-2048 taps: Most crossover and correction applications
• 2048-4096 taps: Typical maximum for full-range loudspeaker designs
• 4096-8192 taps: Subwoofer correction, room correction
• 8192-16384 taps: Specialized low-frequency correction below 20 Hz
• >16384 taps: Rarely needed (extreme cases, research purposes)

Trade-offs:

• ✅ More taps = better frequency resolution, steeper filter slopes, more precise correction
• ❌ More taps = higher latency (for linear-phase), increased DSP load, larger export files

Export Padding

Range: 0 to 65536 taps
Default: 0
Location:

• Shared mode (per-driver tap lengths disabled): In Target DSP Settings section - applies to all drivers
• Per-driver mode (per-driver tap lengths enabled): In each driver’s collapsible section

Adds zero-padding only to exported files, without affecting internal processing or design.

Purpose:

• Some DSP platforms require specific filter lengths (e.g., powers of 2: 512, 1024, 2048, 4096)
• Padding allows meeting DSP requirements without changing internal design
• Does not affect latency, delay, or any graph displays

How it works:

• Internal design: Uses configured tap count
• Export: Appends zeros to reach (Taps + Padding) length
• Frequency response remains identical (zero-padding doesn’t alter frequency content)

Example scenario:

• Design with 3000 taps, DSP requires power-of-2 → Set padding to 1096 → Exports 4096 taps

When not needed: If your DSP accepts arbitrary filter lengths, leave padding at 0.

FIR Crossover Filters

FIR Low-Pass Filter

Enable: Toggle FIR LP button
Configure: Click Configure button (only available when enabled)

Applies a linear-phase low-pass filter to limit the driver’s high-frequency response.

Parameters:

Filter Type

Choose the crossover filter design:

• Brickwall: Ideal brickwall (Sinc) filter using Kaiser window

  • Sharpest possible slope
  • Controlled transition width via Kaiser β parameter

• Linkwitz-Riley: Linkwitz-Riley aligned response

  • -6 dB at crossover frequency (sums to 0 dB when paired with matching HP)
  • Constant power summing when combined with complementary HP filter
  • Order: 2, 4, 6, 8, or 10 (even orders only)

• Butterworth: Classic analog-style Butterworth response

  • Maximally flat magnitude in passband
  • Order: 1 to 10

• Bessel: Bessel/Thomson response

  • Originally optimized for linear phase response and minimal group delay variation
  • Gentler roll-off than Butterworth
  • Order: 1 to 10

Cutoff Frequency

Range: 1 Hz to Nyquist frequency
Step: 1 Hz

The -6 dB point for brickwall filters, or the characteristic frequency for Butterworth/Linkwitz-Riley filters.

Typical values:

• Woofer-Midrange: 300-800 Hz
• Midrange-Tweeter: 2-4 kHz

Kaiser β

Range: 0.0 to 50.0
Default: 8.0

Controls the Kaiser window applied to the FIR filter impulse response. This parameter adjusts the trade-off between transition width and stopband ripple:

For Brickwall filters:

• Lower values (3-5): Narrower transition BUT higher stopband ripple (higher sidelobe levels)
• Moderate values (6-10): Balanced transition width and stopband attenuation
• Higher values (12-20): Wider transition BUT lower stopband ripple (better attenuation)

For Butterworth, Linkwitz-Riley and Bessel filters:

• Controls FIR windowing to suppress ringing and ripple in the stopband
• Lower values (3-5): Sharper response BUT higher ripple/ringing in stopband
• Moderate values (6-10): Good balance between sharpness and ripple suppression
• Higher values (12-20): Gentler response BUT maximum ripple suppression

Common choices:

• β = 5: Narrow transition with noticeable ripple (not necesarrily audible as the complementary driver will mask it)
• β = 8-9: Standard crossover design, good compromise
• β = 12-15: Wide transition with excellent stopband attenuation

Order (Butterworth/Linkwitz-Riley/Bessel only)

Butterworth/Bessel Range: 1, 2, 3, 4, 5, 6, 7, 8, 9, 10
Linkwitz-Riley Range: 2, 4, 6, 8, 10 (even only)

Higher order = steeper slope (more dB/octave attenuation).

• 2nd order: 12 dB/octave (gentle slope)
• 4th order: 24 dB/octave (most common for LR designs)
• 6th order: 36 dB/octave
• 8th order: 48 dB/octave (steep slope)

Causality

Range: 0.0 (linear-phase) to 1.0 (minimum-phase)
Default: 0.0 (linear-phase)

Interpolates between linear-phase and minimum-phase behavior for the FIR crossover filter.

Purpose: Limits the delay introduced by the FIR filter while maintaining steeper slopes than IIR filters, at the cost of phase distortion.

Linear-Phase (0.0):

• Symmetric pre- and post-ringing
• Constant group delay across all frequencies
• No phase distortion
• Highest latency (taps/2 samples)
• Best for high-fidelity applications

Minimum-Phase (1.0):

• No pre-ringing (all energy is causal)
• Variable group delay
• Phase follows magnitude via Hilbert transform
• Minimal latency
• Best for low-latency applications

Intermediate values:

• Blend between linear and minimum phase
• Reduced pre-ringing compared to linear-phase
• Some phase distortion compared to linear-phase
• Lower latency than full linear-phase

Typical values:

• 0.0: High-fidelity systems, studio monitors, reference designs
• 0.5: Balanced compromise
• 1.0: Low-latency systems, live sound, PA applications

Trade-off: Increasing causality reduces latency but introduces phase distortion. For multi-driver systems, matching causality settings across all drivers maintains phase coherence.

FIR High-Pass Filter

Enable: Toggle FIR HP checkbox
Configure: Click Configure button (only available when enabled)

Applies a linear-phase high-pass filter to limit the driver’s low-frequency response.

Parameters: Identical to FIR Low-Pass (Type, Cutoff, Kaiser β, Order, Causality)

Linkwitz-Riley and Brickwall Crossover Design: When using complementary LR filters (e.g., woofer LP at 500 Hz + midrange HP at 500 Hz, both 4th order), the combined magnitude response is flat at the crossover point.

FIR Correction Filter

Enable: Toggle FIR Corr button
Configure: Click Configure button (only available when enabled)

Applies frequency-domain magnitude and/or phase correction using the loaded impulse response.

The correction filter window has two main sections: Magnitude and Phase.

Magnitude Correction

Enable Magnitude Correction

Toggle to enable/disable magnitude response correction independently from phase.

Calculation Point

Choose where correction is calculated in the processing chain:

• pre-IIR: Correction computed from raw IR before IIR filters

  • Use when IIR filters are for fine-tuning only
  • Simpler, doesn’t account for IIR response

• post-IIR: Correction computed after IIR filter application

  • Accounts for IIR filter effects in the correction calculation
  • Use when IIR filters significantly alter the response
  • More accurate for complex filter chains

Target Curve

Click Configure to open the target curve configuration window.

Purpose: Define the desired magnitude response for this driver.

See Target Curves for detailed configuration.

Frequency Range

Start Frequency:

• Range: 1 Hz to Nyquist
• Default: 20 Hz
• Lower limit for magnitude correction

End Frequency:

• Range: 1 Hz to Nyquist
• Default: Nyquist
• Upper limit for magnitude correction

Purpose: Restrict correction to a specific frequency range, avoiding correction where measurements are unreliable or correction is undesirable.

Important: Magnitude correction applies a smooth taper (transition zone) at the frequency range boundaries to prevent discontinuities:

• Taper width: Quarter-octave (1/4 octave) above and below the correction range
• Taper function: Raised cosine (smooth fade-in/fade-out)
• Below Start Frequency: Correction weight fades from 0% at Start / 2^(1/4) to 100% at Start
• Above End Frequency: Correction weight fades from 100% at End to 0% at End × 2^(1/4)
• Example: For Start = 200 Hz, the taper begins at 168 Hz. For End = 10 kHz, the taper extends to 11.9 kHz
• Purpose: Ensures smooth transitions without abrupt magnitude steps or phase artifacts

Critical Understanding: Correction Source Signal

The magnitude correction filter is computed from the driver’s impulse response without FIR low-pass or high-pass filters applied (those are applied later in the signal chain). The correction sees:

• ✅ IIR filters (if “post-IIR” calculation mode is enabled)
• ❌ FIR LP/HP crossovers (not applied yet)

Practical Implications:

1. With IIR Crossovers (post-IIR mode):

• The correction attempts to flatten the IIR crossover rolloff
• ⚠️ This removes your crossover slope, which is usually undesirable
• Solution: Limit the frequency range to roughly your driver’s final passband
• Example: For a tweeter with IIR HP at 2 kHz, set correction range to 2.5 kHz - 20 kHz

2. With pre-IIR mode or FIR crossovers:

• The correction sees the driver’s raw natural rolloff
• Risk: Amplifying rolloff increases noise and distortion outside the intended band
• Sometimes helpful: Can improve summation in the crossover transition region
• Solution: Set frequency range to where summation matters, don’t extend further
• Example: For a midrange crossing at 500 Hz and 3 kHz, limit to 400 Hz - 4 kHz

Best Practice: Configure the frequency range to match your driver’s intended operating band considering your crossover strategy, not just where the measurement looks good.

Resolution

Options: 1/48 oct, 1/24 oct, 1/12 oct, 1/6 oct, 1/3 oct
Default: 1/12 octave

Fractional-octave smoothing applied to the measured response before computing the correction filter. This determines the resolution of features that will be corrected.

Purpose: Avoid correcting measurement artifacts (reflections, diffraction ripples, noise) that are not inherent to the speaker’s response.

• 1/48 oct to 1/24 oct: Minimal smoothing - risks correcting artifacts and creating excessive filter complexity
• 1/12 oct: Recommended balance - corrects real speaker behavior while ignoring fine ripples
• 1/6 to 1/3 oct: Heavy smoothing - corrects only broad trends

Key principle: The correction filter resolution should match the reliability of your measurement, not its raw resolution. Narrow features (<1/12 oct) are often artifacts, not real acoustic properties worth correcting.

Recommendation: Use 1/12 octave for typical speaker correction, 1/6 octave for room measurements or when measurement quality is uncertain.

Max. Attenuation (dB)

Range: 0 to 30 dB
Default: 10 dB

Maximum attenuation applied by the magnitude correction filter.

How it works: The correction filter works by “cutting from the top” - it attenuates peaks in the measured response to bring them down toward the target curve. This parameter limits how much attenuation is applied.

Purpose:

• Prevents excessive notching of narrow peaks (which may be measurement artifacts)
• Keeps the correction filter smooth and stable
• Avoids creating deep notches that could be audible as resonances

Typical values:

• 5-8 dB: Conservative correction for well-behaved drivers
• 10 dB: Standard value for most applications
• 12-15 dB: More aggressive correction (check impulse response for artifacts)
• >15 dB: Rarely needed, may indicate measurement issues

Important: The correction filter primarily attenuates rather than boosts. Use Correction Gain (below) to compensate for the overall level reduction.

Correction Gain (dB)

Range: -20 to +20 dB
Default: 4 dB

Gain applied within the correction frequency range to compensate for the attenuation introduced by the magnitude correction filter.

Purpose:

• Restores overall level after attenuation-based correction
• Allows fine-tuning the driver’s output level
• Maintains consistent loudness across the corrected frequency band

Typical values:

• 0-4 dB: Minimal compensation (gentle correction)
• 4-6 dB: Standard compensation for moderate correction
• 6-10 dB: High compensation (use with caution)

⚠️ Warning: Excessive correction gain can cause clipping in the internal DSP signal chain or damage your drivers. Keep gain reasonable (< 6 dB recommended) to maintain headroom. If LinFIR’s clipping detection is enabled (see Settings), you’ll receive warnings about potential clipping.

Kaiser β (Magnitude)

Range: 0.0 to 50.0
Default: 8.0

Kaiser windowing parameter for the magnitude correction FIR filter.

Effect:

• Lower values (3-5): Sharper correction BUT may amplify ripple from crossover filters (LP/HP) if active
• Moderate values (6-10): Good balance between precision and ripple suppression
• Higher values (12-20): Smoother correction, reduced ripple amplification, but less precise

When to adjust:

• If you see excessive ripple in the stopband after applying correction with active crossover filters, increase Kaiser β
• If correction seems too gentle or imprecise, decrease Kaiser β

Recommendation: Start with default 8.0, increase to 12-15 if ripple is problematic with crossover filters active.

Causality (Magnitude)

Range: 0.0 (linear-phase) to 1.0 (minimum-phase)
Default: 1.0 (minimum-phase)

Controls the phase characteristic of the magnitude correction filter.

• 0.0 (Linear-phase): Symmetric pre- and post-ringing, constant group delay
• 1.0 (Minimum-phase): No pre-ringing, all energy is causal, variable group delay

Typical choice: 1.0 (minimum-phase) to avoid pre-ringing artifacts.

Why minimum-phase is preferred: Most acoustic and electronic systems are minimum-phase (magnitude and phase are coupled via Kramers-Kronig relations). Correcting magnitude with a minimum-phase filter automatically corrects the associated phase deviation, providing physically accurate correction. Use 0.0 (linear-phase) only if you specifically need constant group delay with a non-flat target curve and can tolerate pre-ringing.

Phase Correction

Enable Phase Correction

Toggle to enable/disable phase response correction independently from magnitude.

🚫 Non-anechoic measurements: phase correction is strongly discouraged

Phase correction requires clean, accurate phase data. In-room measurements contain reflections, room modes, and position-dependent artifacts that pollute the phase response. Applying phase correction to such data produces unpredictable — and often worse — results.

Only use phase correction with anechoic or near-anechoic measurements (e.g., measurements gated to remove reflections, or made in an anechoic chamber). For in-room measurements, restrict correction to magnitude only.

⚠️ Critical Limitation: Phase Correction vs. Low Latency

You cannot have both near-zero latency FIR filters AND active phase correction. This is a fundamental limitation, not a software bug.

Why this incompatibility exists:

Even if you configure your FIR crossover filters (HP/LP) and FIR magnitude correction with causality = 1.0 (minimum-phase, low latency), which positions the FIR impulse peak near t=0 (left-aligned), enabling phase correction will shift the combined FIR impulse peak to the center (around ntaps/2), introducing significant latency.

The reason is fundamental:

• Phase correction = temporal correction: Phase distortion is fundamentally a time-domain problem (different frequencies arriving at different times)
• To correct temporal defects, you need time: The filter needs temporal “space” before and after the main impulse to reposition frequency components correctly
• This requires centering the impulse: Phase correction filters must have headroom both before (pre-ringing) and after (post-ringing) the main signal to perform time-domain manipulations
• Result: The combined FIR impulse peak moves from t≈0 (low latency) to t≈ntaps/2 (high latency)

Practical impact:

• Without phase correction: Causality=1.0 filters → impulse peak at t≈0 → latency ≈ 0 samples
• With phase correction enabled: Impulse peak moves towards t≈ntaps/2 → latency ≈ ntaps/2 samples (e.g., 2048 samples at 48kHz = 42.7ms)

Recommended approach:

• For low-latency applications (live sound, PA, monitoring): Use causality = 1.0 on crossover/magnitude filters, disable phase correction, accept phase as-is
• For high-fidelity applications (studio monitors, reference systems): Use causality = 0.0 on crossover, 1.0 on magnitude filters, enable phase correction if needed, accept the latency
• For hybrid approach: Use moderate causality (0.3-0.7) for reduced latency with some phase correction capability (but still expect significant latency when phase correction is active)

Calculation Point

Same as magnitude: pre-IIR or post-IIR calculation mode.

pre-IIR: Phase correction computed from raw IR before IIR filters
post-IIR: Phase correction computed after IIR filter application

Resolution (Phase)

Options: 1/48 oct, 1/24 oct, 1/12 oct, 1/6 oct, 1/3 oct
Default: 1/12 octave

Fractional-octave smoothing applied to the measured phase response before computing the correction filter.

Recommendation: Use heavier smoothing (1/6 to 1/3 octave) for phase correction to avoid chasing measurement noise and room reflections.

Intensity

Range: 0% to 100%
Default: 100%

Controls the amplitude/strength of the phase correction.

Purpose:

• Allows partial phase correction for smoother transitions
• Reduces phase correction aggressiveness if full correction creates artifacts
• Useful for blending between corrected and uncorrected phase

Typical values:

• 100%: Full phase correction (default)
• 50-80%: Partial correction for gentler results
• 0%: No phase correction (disabled)

Phase Offset

Range: -180° to +180°
Default: 0°

Phase offset applied to the filter’s phase response.

Purpose: Controls the symmetry of the impulse response by adjusting the phase offset. This allows fine-tuning the balance between pre-ringing and post-ringing in the time-domain response.

How it works:

• 0°: Default even symmetric behavior
• Positive/Negative offset: Shifts impulse response characteristics towards odd symmetry

Use case: Advanced parameter for optimizing impulse response symmetry in multi-driver systems.

Recommendation: Leave at 0° unless you have specific symmetry requirements and understand impulse response trade-offs.

Kaiser β (Phase)

Range: 0.0 to 50.0
Default: 1.0

Kaiser windowing parameter for the phase correction FIR filter.

Effect:

• Lower values (1-5): Less aggressive windowing — the correction filter’s oscillations (pre- and post-ringing) are preserved more fully, allowing accurate phase correction at the cost of slower roll-off at the filter edges
• Moderate values (6-10): Balanced windowing — trade-off between accuracy and artifact suppression
• Higher values (12-20): Aggressive windowing — strongly attenuates the filter’s pre- and post-ringing; if correction energy is clipped by the window, this introduces magnitude deviations

When to adjust:

• If the guard warning appears (the auto-guard has already narrowed the Start/End Frequencies as far as it can): decrease Kaiser β — see Phase Correction Magnitude Artifacts below
• If phase correction seems imprecise or creates excessive ringing, increase Kaiser β

Recommendation: Start with default 1.0. If the guard warning still appears after the guard’s automatic frequency adjustments, try reducing β toward 0.

Guard Tolerance

Range: 0.1 dB to 15 dB
Default: 0.5 dB

Maximum magnitude deviation the auto-correction guard will tolerate before narrowing the correction band. Above this threshold, the guard automatically raises the Start Frequency in 1/3-octave steps (and eventually lowers the End Frequency) until the phase FIR stays within tolerance — see Phase Correction Magnitude Artifacts.

• Tighter (0.1–0.5 dB): Stricter — the guard narrows the band more aggressively. Recommended for high-fidelity correction.
• Looser (1–5 dB): More permissive — allows a wider correction band at the cost of some magnitude coloration.

Frequency Range (Phase)

Start Frequency:

• Range: 0 Hz to Nyquist
• Default: 0 Hz

End Frequency:

• Range: 0 Hz to Nyquist
• Default: Nyquist

Limits the frequency range over which phase correction is applied.

Use cases:

• Focus correction on the driver’s primary operating band (e.g., 200 Hz – 10 kHz for a midrange)
• Exclude deep bass where phase rotations would require very long filters
• Anchor the correction band manually when the auto-guard’s automatic adjustments are not desirable

Note: The auto-correction guard already raises the Start Frequency automatically when artifacts are detected. Only set it manually to intentionally anchor the correction start frequency.

Phase Correction Magnitude Artifacts

A phase correction FIR filter is designed to be magnitude-neutral — it should modify only the phase of the signal, leaving energy at each frequency unchanged. However, in practice, magnitude deviations can appear after applying phase correction. LinFIR includes an auto-correction guard that detects and corrects these automatically.

Why artifacts occur

Phase correction works in the time domain: correcting a phase rotation at a given frequency means shifting energy from one moment in time to another. This produces an impulse response with oscillations extending both before and after the central peak (pre- and post-ringing). The lower the frequency being corrected, the longer these oscillations extend, because a given phase rotation translates to a longer time shift at lower frequencies.

Two mechanisms can cause the filter to deviate from magnitude-neutral behavior:

1. Insufficient tap count

If the number of taps allocated to the phase correction filter is too small, the filter’s oscillations extend beyond the available window — they are truncated at the boundaries. Truncating a non-zero portion of the filter’s impulse response distorts its frequency response, introducing magnitude ripples proportional to the energy that was cut off.

This is analogous to abruptly cutting off a signal in the time domain and observing Gibbs-like ringing in the frequency domain.

2. Overly aggressive Kaiser windowing (high β)

The Kaiser window tapers the filter’s impulse response toward zero at both ends. A high β value applies a strong taper. This strongly attenuates the pre- and post-ringing of the phase correction filter — even if those oscillations were within the allocated tap window.

Attenuating oscillations that carry real correction energy distorts the filter’s frequency response, producing magnitude deviations — typically attenuations that grow stronger as the window becomes more aggressive.

Auto-correction guard

LinFIR includes an auto-correction guard that runs automatically every time the phase correction FIR is computed — it is always active. The guard:

1. Computes the phase correction FIR with the current Start and End Frequencies
2. Measures the maximum magnitude deviation within the active passband
3. If the deviation exceeds Guard Tolerance, raises the Start Frequency by 1/3-octave steps and retries
4. Once the Start Frequency reaches its ceiling (~15 kHz), lowers the End Frequency instead (down to ~18 kHz)
5. Repeats up to 100 iterations until the deviation stays within the Guard Tolerance threshold

The guard silently narrows the effective correction band until the phase FIR becomes magnitude-neutral. Under normal circumstances no warning appears — the guard resolves artifacts transparently.

When the guard warning appears

If the guard exhausts all 100 iterations without reducing the deviation below Guard Tolerance, it gives up and — if Warn when phase FIR guard fails is enabled in Settings — shows a warning. At this point the guard has already narrowed the frequency band as far as it can, yet the FIR still deviates. Manual intervention may help:

1. Reduce Kaiser β (most effective)
   High β aggressively clips filter oscillations, directly causing magnitude deviations. Since the guard only adjusts frequency boundaries and not Kaiser β, reducing β is the first thing to try.
   → Reduce from the current value toward 1.0 or lower.

2. Increase the tap count
   More taps give the filter more temporal “space” to express its oscillations without truncation. This is especially important for corrections at low frequencies.
   → Double the tap count and check if the warning disappears.

3. Raise Guard Tolerance
   If the residual deviation is small enough to be acceptable, relax the threshold.
   → Set Guard Tolerance to 1–2 dB and assess whether the audible impact is acceptable.

4. Reduce the phase correction intensity
   Lower intensity reduces filter complexity and correction energy.
   → Try setting Intensity to 70–80%.

⚠️ Before adjusting filter parameters, inspect the phase and group delay plots first.

A guard warning that resists all of the above is often a symptom of a measurement quality problem, not a filter configuration problem. Accumulated phase rotations caused by reflections or resonances in the impulse response force the phase correction filter to work on chaotic, non-monotonic data — producing an unstable, artifact-prone FIR regardless of the parameters chosen.

In Loudspeaker Design mode: Measurements must be taken with the least possible room influence. Use time windowing (Start/Stop Time in IR Management → Time Windowing) to gate out reflections before applying phase correction. Use the Auto Set Window function as a starting point, then review the phase and group delay plots — they should be smooth and monotonic in the driver’s passband. Irregular phase jumps or group delay peaks indicate reflections or resonances that windowing should remove.

In Room Calibration mode: Impulse windowing is intentionally not available — this mode is designed to capture the full speaker-plus-room response. Applying a temporal window would yield an anechoic-like response, which would be misleading: corrections derived from it would be wrong for the actual listening environment, because your ears do not apply a time window. To reduce spatial variability and limit the influence of isolated reflections on the measured phase, capture at least 5 or 6 measurements at different listening positions and rely on spatial averaging — see Room Calibration Mode.

Even so, phase correction is strongly discouraged in room calibration mode. The room dominates the phase response — modal resonances, flutter echo, and comb filtering from reflections produce chaotic phase behavior that varies dramatically across the listening area. Even with spatial averaging, a phase correction filter derived from in-room data is likely to be spatially unstable: it may improve the phase at the measurement positions while worsening it elsewhere, and may introduce audible coloration or artifacts. No DSP technique can substitute for acoustic treatment when it comes to correcting room-induced phase problems.

FIR Correction Best Practices

1. Start conservative:

   • Begin with 1/6 or 1/3 octave smoothing
   • Limit maximum gain to 6-12 dB

2. Narrow frequency range:

   • Only correct the driver’s intended operating band
   • Don’t correct below woofer resonance or above tweeter range
   • Avoid correcting crossover stopband region

3. Magnitude before phase:

   • Get magnitude response flat first
   • Then add phase correction if needed
   • Phase correction is most effective with flat magnitude

4. Check impulse response:

   • Excessive pre-ringing indicates too aggressive correction
   • Increase smoothing or reduce max gain if pre-ringing is severe

5. Combine with IIR:

   • Use FIR correction for broad trends
   • Use IIR filters for narrow notches and fine adjustments

IIR Filtering

Enable: Toggle IIR Filters checkbox
Configure: Click Configure button (only available when enabled)

Apply cascaded biquad IIR filters for parametric equalization and precise frequency shaping.

IIR Filter Tabs

The IIR window has two tabs:

Manual Tab

Design custom IIR filter chains with up to 50 filters.

Add Filter: Click ➕ Add Filter button
Remove Filter: Click ❌ button next to filter

Available Filter Types:

• Peak/Notch: Parametric EQ (boost/cut at specific frequency)
• Low Shelf: Boost/cut below shelf frequency
• High Shelf: Boost/cut above shelf frequency
• Low-Pass: Frequency-selective attenuation (high frequencies)
• High-Pass: Frequency-selective attenuation (low frequencies)
• All-Pass: Phase adjustment without magnitude change

Filter Parameters:

Frequency:

• Center frequency for Peak/Notch
• Cutoff frequency for LP/HP/Shelves
• Corner frequency for All-Pass

Gain (dB):

• Boost (+) or cut (-) amount
• Only for Peak/Notch and Shelving filters
• Range: -40 to +40 dB

Q Factor:

• Filter bandwidth/sharpness
• Higher Q = narrower bandwidth
• Typical range: 0.5 to 20
• Low Q (0.5-2): Wide, gentle curves
• Medium Q (2-5): Standard parametric EQ
• High Q (10-20): Narrow notches for resonance control

Order (LP/HP only):

• Filter slope steepness
• Range: 1 to 10 (depending on filter type)
• Each order adds 6 dB/octave slope

Pass Type:

• Low-Pass, High-Pass: Frequency-selective filtering
• Peak/Notch: Standard parametric EQ

Locked Filters (🔒):

Individual filters can be locked to preserve them during operations:

• Lock toggle: Click the 🔓/🔒 icon next to any filter to lock/unlock it
• Protected from Clear: Locked filters are not deleted when clicking “Clear Filters”

Use Cases:

• Lock protective filters (high-pass, low-pass) to preserve them when clearing other filters
• Preserve manually tuned filters when experimenting with different EQ approaches
• Protect critical filters (driver protection, subsonic filters) from accidental deletion

Auto Tab

Automatically generated equalization with manual control.

Adding Filters:

• Click ➕ Add Filter to manually add filters in Auto EQ tab
• Add protective filters (high-pass, low-pass, notches) before running Auto EQ
• Lock them with 🔒 icon to preserve during Auto EQ generation

Manual Editing:

• All filters in Auto EQ tab are editable (frequency, gain, Q)
• Adjust auto-generated filters directly without cloning
• Lock important filters to prevent removal during regeneration

Generate Auto EQ:

• Click Generate to automatically create corrective filters
• Auto EQ respects locked filters (integrates HP/LP into target curve)
• Unlocked filters are replaced with new auto-generated corrections

Copy to Manual:

• Click Clone from Auto EQ to copy the entire filter set to Manual tab
• Useful for preserving a working Auto EQ configuration while experimenting in Manual

Auto EQ Parameters

When you click Generate in the Auto EQ tab, the optimizer uses the following parameters to create corrective filters:

Gain Offset:

• Range: -20 to +20 dB
• Default: 0 dB
• Purpose: Adjusts the overall target level before EQ optimization
• Use case: Compensate for overall system level shifts without changing filter gains

Gain Min / Gain Max:

• Range: -20 to +20 dB
• Default: -6 to +6 dB
• Purpose: Limits the boost/cut range for each individual filter
• Use case: Prevent excessive corrections that could cause clipping or sound unnatural

EQ Boost Cap:

• Range: 0 to +20 dB
• Default: 6 dB
• Purpose: Soft-clips the correction target to prevent filter stacking at the same frequency
• Use case: Set lower values (e.g., 4 dB) for more conservative corrections

Q Min / Q Max:

• Range: 0.1 to 20.0
• Default: 0.3 to 8.0
• Purpose: Controls the bandwidth/sharpness of generated filters
• Use case: Lower Q = wider corrections, higher Q = narrow notches for resonances

Max Filters:

• Range: 1 to 30
• Default: 12
• Purpose: Maximum number of filters to generate (including shelves if enabled)
• Use case: More filters = more precise correction, but more phase impact

Freq Min / Freq Max:

• Range: 10 to 22,000 Hz
• Default: 20 to 20,000 Hz
• Purpose: Defines the frequency range for filter placement and correction
• Important: Auto EQ applies a quarter-octave taper (1/4 octave) above and below these frequency limits to ensure smooth transitions without discontinuities in the correction
• How the taper works:
  • Below Freq Min: Correction weight gradually reduces from 100% at Freq Min to 0% at Freq Min / 2^(1/4) (quarter-octave below)
  • Above Freq Max: Correction weight gradually reduces from 100% at Freq Max to 0% at Freq Max * 2^(1/4) (quarter-octave above)
  • Outside taper zones: No correction applied (weight = 0%)
• Use case: Focus correction on the driver’s operating band, exclude problematic regions (e.g., below resonance, room modes, or above useful bandwidth)

Low Shelf / High Shelf:

• Toggle: Enable/disable shelf filters
• Purpose: Allow AutoEQ to include shelving filters for broad tonal adjustments
• Use case: Enable for better overall tonal balance, disable for peaking-only corrections

Resolution:

• Options: 1/48, 1/24, 1/12, 1/6, 1/3 octave
• Default: 1/12 octave
• Purpose: Smoothing applied to the current response before optimization
• Use case: Higher smoothing (1/3 oct) = broader corrections, lower smoothing (1/48 oct) = more detailed corrections

IIR Filter Applications

Parametric EQ (Peak/Notch):

• Boost/cut specific frequencies
• Correct narrow resonances or notches
• Fine-tune response after FIR correction

Shelving Filters:

• Broad treble/bass adjustments
• Tilt EQ for overall tonal balance
• Baffle step compensation
• Psychoacoustic tuning

Notch Filters:

• Suppress driver resonances

Low-Pass/High-Pass:

• Additional crossover filtering (complements FIR)
• Subsonic/ultrasonic filtering

All-Pass:

• Phase correction without affecting magnitude
• Time alignment adjustments
• Advanced multi-driver phase matching

IIR Best Practices

1. Use sparingly:

   • Fewer filters = cleaner phase response
   • Each biquad adds phase rotation
   • Prefer FIR correction for broad magnitude shaping

2. Narrow adjustments:

   • IIR excels at narrow peaks/notches
   • Use Q > 5 for resonances
   • Use Q < 2 for broad tonal adjustments

3. Check phase impact:

   • View phase plot when adding IIR filters
   • High-Q filters create significant phase rotation
   • Consider if phase impact is acceptable

4. Combine with FIR:

   • FIR correction for broad trends (1/3 to 1/6 oct smoothing)
   • IIR filters for narrow resonances and fine details
   • Best of both worlds

Driver Adjustments

Polarity Inversion

Toggle: Polarity button

Inverts the driver’s signal (180° phase shift across all frequencies).

Use cases:

• Correct out-of-phase drivers
• Optimize multi-driver summation (especially for dipole/bipole designs)
• Fix inverted wiring
• Align asymmetric crossovers

How to check:

• Observe step response: positive step should have initial positive rise
• Check summed magnitude: in-phase drivers sum coherently (+6 dB), out-of-phase drivers cancel

Gain Adjustment

Range: -40 to +40 dB
Step: 0.1 dB
Default: 0 dB

Adjusts the driver’s output level.

Use cases:

• Compensate for driver sensitivity differences
• Set target SPL for each driver
• Balance multi-driver systems
• Fine-tune summed response

Time Delay

Range: 0.0 to 100.0 ms
Step: equivalent to 1 sample (depends on target sampling frequency) Default: 0.0 ms

Adds pure time delay to the driver’s output.

Use cases:

• Align drivers physically offset from each other
• Compensate for acoustic center differences
• Time-align multi-way systems for coherent summation
• Correct for DSP processing delays

Calculation:

• Distance (m) to delay (ms): delay = distance / 0.343 m/ms
• Example: 10 cm offset = 0.291 ms delay

Delay Step Size:

• Automatically adjusted based on sample rate
• Higher sample rates allow finer delay precision
• Use fine adjustments for critical time alignment

FIR Compensation Delay

Range: 0.0 to 100.0 ms
Step: equivalent to 1 sample (depends on target sampling frequency) Default: 0.0 ms

Additional delay applied only to the FIR filter chain, not the entire signal.

Purpose:

• Manual control over FIR filter impulse positioning
• Fine-tune FIR filter alignment when Auto Causal Alignment is disabled
• Shift FIR maximum to optimal crop window position

Use case: When combining multiple causal FIR filters, the impulse maximum may shift. FIR Compensation Delay allows manually repositioning the combined FIR response to avoid truncation. It is not the primary timing alignment tool—use the Time Delay feature for that.

Recommendation: Use Auto Causal Alignment unless you need precise manual control.

Processing Chain Summary

The complete driver processing chain:

1. Raw Impulse Response

   • Loaded via IR Management window
   • Windowed with configurable start/stop times

2. FIR Crossover Filters

   • High-pass filter (if enabled)
   • Low-pass filter (if enabled)
   • Applied in time domain

3. FIR Correction Filter

   • Magnitude correction (if enabled)
   • Phase correction (if enabled)
   • Applied in time domain

4. Combined FIR Filters

   • All FIR filters convolved into single impulse response
   • Cropped to configured tap length
   • Causality interpolation applied

5. IIR Filtering

   • Cascaded biquad sections
   • Applied in time domain (as SOS transfer function)

6. Driver Adjustments

   • Polarity inversion (if enabled)
   • Gain adjustment
   • Time delay

7. Final Output

   • Processed driver response
   • Summed with other drivers for system response

Workflow Recommendations

Starting a New Driver

1. Load impulse response via Manage IR
2. Configure windowing to isolate direct sound
3. Set filter length (512-4096 taps for most designs)
4. Add FIR crossover (LP/HP) if needed
5. Apply FIR correction (start conservative: 1/6 oct smoothing, 12 dB max gain)
6. Check summed response, adjust gain/delay for coherent summation
7. Add IIR filters for fine-tuning (narrow resonances, shelves)
8. Optimize causality if latency is critical
9. Export filters for DSP deployment

Multi-Way Design

1. Load all driver IRs (woofer, midrange, tweeter)
2. Set crossover frequencies based on driver capabilities
3. Apply complementary crossovers (e.g., LR4 at matching frequencies)
4. Check phase alignment and IR (step) summation in phase and IR (step) plots
5. Adjust time delays to align acoustic centers
6. Check summed magnitude for flat on-axis response
7. Fine-tune with IIR for final voicing
8. Verify step response for correct polarity and alignment

Keyboard Shortcuts

• Cmd+W / Ctrl+W: Close configuration windows (LP, HP, Correction, IIR)
• F: Switch to Filters display mode
• D: Switch to Drivers display mode
• U: Toggle wrapped/unwrapped phase
• C: Toggle time of flight rotation removal

See Keyboard Shortcuts for complete reference.

Related Documentation

• IR Management: Detailed impulse response loading and windowing
• System Processing: Global FIR correction and system-level processing
• Export: Exporting filters for DSP platforms
• Settings: Application-wide settings and defaults

System Processing

LinFIR provides global (system-wide) processing that applies corrections to the summed system response after all individual driver filters. This is available in both Loudspeaker Design and Room Calibration modes.

Global processing consists of two main components:

• Global FIR Correction Filter: Magnitude and phase correction of the summed system response
• Global IIR Filters: System-wide parametric EQ for room correction and final tuning

This page documents the global FIR correction filter. For global IIR filtering, see the corresponding section below.

Loudspeaker Design

Driver responses are summed to form the combined system response before correction.

Room Calibration

Driver responses are averaged across measurement positions before correction.

Global FIR Correction Filter

The Global FIR Correction Filter corrects the summed system response using FIR filters. It applies after all individual driver filters are combined, providing final system-level magnitude and phase correction.

Access the global FIR correction window from the Drivers/Measurements column (Global FIR button in the Global Filters section).

When to Use Global Correction

Loudspeaker Design Mode:

• Correct summed driver response/entire loudspeaker
• Align final system response to target curves (Flat, Harman, Custom)
• Compensate for baffle diffraction effects on the combined response
• Final system voicing and tuning

Room Calibration Mode:

• Room correction applied to the averaged measurements
• Correct combined room + speaker response
• Export global correction for integration into DSP processors

💡 Tip: Use global correction for overall tonal balance, and system-wide frequency response issues. Use per-driver correction (in Loudspeaker Design mode) for individual driver resonances, breakup modes, or response irregularities.

Global FIR Filter Length

FIR Taps

• Range: 32 to 65,536 taps
• Controls: the length (and thus frequency resolution) of the global FIR correction filter

How to choose:

• Higher tap counts → better low-frequency resolution, longer processing latency
• Lower tap counts → lower latency, reduced low-frequency precision

⚠️ Latency Note: The global FIR filter’s latency depends on tap count, magnitude causality, and phase correction:

• Causality = 1.0 (minimum-phase) + no phase correction: ~0 latency (impulse at t ≈ 0)
• Causality = 0.0 (linear-phase): ~ntaps/2 samples latency (impulse centered)
• Phase correction enabled: adds latency regardless of causality (shifts impulse to center)

For low-latency applications, use Causality = 1.0 and disable phase correction.

Export Padding

• Range: 0 to 65,536 taps
• Purpose: Add zero-padding to the exported FIR filter files only

Export padding does not affect internal processing—it only extends the exported FIR coefficients with zeros. This is useful for:

• Meeting minimum FIR length requirements of external DSP platforms
• Aligning export formats across multiple filters

Example: If the global FIR filter is 4096 taps and you set export padding to 512, the exported file will contain 4608 coefficients (4096 actual + 512 zeros).

FIR Compensation Delay

• Range: -100 to +100 ms
• Purpose: Fine-tune the alignment of the global FIR filter in the processing chain

This parameter shifts the global FIR filter in time relative to the taps window.

Use Cases:

• Compensate for small timing misalignments introduced by FIR processing
• Adjust global FIR placement when mixing minimum-phase and linear-phase filters

💡 Tip: In most cases, leave this at 0 ms. Only adjust if you observe timing issues and ringing truncation in the impulse response.

Compensate FIR Delay on Graphs

• Toggle: Shift the impulse response left in plots to remove visual delay

When enabled:

• The impulse response plot shifts left by approximately ntaps/2 samples
• The phase and group delay plots are also affected
• Exported filters are not modified

Purpose:

• Provides a clearer view of the aligned impulse response across drivers
• Useful for comparing driver responses without the visual offset from FIR latency

When to enable:

• When visually analyzing impulse alignment between drivers
• When comparing group delay and phase between the individual drivers and the sum

When to disable:

• When you need to see the true latency of the system

Magnitude Correction

The Magnitude Correction section corrects the frequency response (magnitude) of the summed system.

Enable Magnitude Correction

Toggle to enable/disable magnitude response correction independently from phase.

When disabled, magnitude correction settings are grayed out.

Calculation Point (Magnitude)

• Options: pre-IIR or post-IIR
• Purpose: Choose where to measure the system response for magnitude correction

pre-IIR:

• Measures the system before global IIR filters
• Corrects the summed driver response (FIR crossovers + driver FIR corrections + driver IIR)
• Use when you want to correct the raw driver summation before applying IIR EQ

post-IIR:

• Measures the system after global IIR filters
• Corrects the final system response (drivers + global IIR)
• Use for final system tuning or room correction that accounts for global IIR EQ

💡 Tip: For room calibration, use post-IIR to correct the final room + speaker response. For loudspeaker design, use pre-IIR to correct driver interactions before applying global voicing EQ.

Target Curve

• Button: Configure
• Purpose: Select and configure the target curve for magnitude correction

Clicking Configure opens the Target Curve Window where you can:

• Select Flat or Harman as base curves
• Add inflection points and shape the curve

The magnitude correction FIR filter will match the response to this target curve.

See the Target Curves section in the documentation for details on curve types and customization.

Resolution (Magnitude)

• Options: 1/48 octave, 1/24 octave, 1/12 octave, 1/6 octave, 1/3 octave
• Purpose: Control the smoothing applied to the magnitude correction

Lower fractions (e.g., 1/48 oct) → finer detail, corrects small ripples
Higher fractions (e.g., 1/3 oct) → broader smoothing, corrects only major trends

How to choose:

• Room correction: Use 1/6 octave or 1/3 octave to avoid over-correcting room modes (which vary with position)
• Loudspeaker design: Use 1/12 octave or 1/24 octave for detailed driver correction

⚠️ Warning: Very fine smoothing (1/48 oct) can lead to excessive correction of measurement noise. Use with caution.

Max Attenuation

• Range: 0 to 30 dB
• Default: 10 dB
• Purpose: Limit the maximum attenuation (cut) applied by magnitude correction

This parameter “cuts from the top”—it limits how much the correction filter can reduce peaks in the frequency response.

Why limit attenuation?

• Prevents excessive cuts that waste headroom
• Avoids over-correcting narrow resonances (which may not be audible)
• Maintains system efficiency

Example:

• If the system has a +15 dB peak at 200 Hz and Max Attenuation is set to 10 dB, the correction will cut only 10 dB (leaving a +5 dB peak)

Recommended values:

• Room correction: 10–15 dB (room modes can be large)
• Loudspeaker design: 6–10 dB (driver peaks are typically smaller)

Correction Gain

• Range: -20 to +20 dB
• Default: 4 dB
• Purpose: Add gain to the correction filter to compensate for attenuation and maintain system level

Magnitude correction typically attenuates peaks, which reduces overall loudness. Correction Gain adds a fixed boost to restore the level.

How to choose:

• Set Correction Gain to approximately match the average attenuation applied
• For example, if the correction cuts ~4 dB on average, set Correction Gain to +4 dB

💡 Tip: Adjust Correction Gain by observing the final system level after correction. Too much gain can cause clipping or driver damage; too little leaves the system quieter than before.

Kaiser β (Magnitude)

Range: 0.0 to 50.0
Default: 8.0

Kaiser windowing parameter for the magnitude correction FIR filter.

Effect:

• Lower values (3-5): Sharper correction BUT may increase ringing
• Moderate values (6-10): Good balance between precision and ringing suppression
• Higher values (12-20): Smoother correction, reduced ringing, but less precise

Recommendation: Start with default 8.0, decrease to 1-2 if ringing is not problematic.

Frequency Range (Magnitude)

• f_min (minimum frequency): Starting point for magnitude correction
• f_max (maximum frequency): Ending point for magnitude correction
• Fade-in/Fade-out: Smooth transitions at the boundaries

Purpose: Limit magnitude correction to a specific frequency range.

Example:

• Set f_min = 40 Hz and f_max = 12,000 Hz to correct only mid-bass through mid-treble, leaving deep bass and extreme treble unaffected

💡 Tip: Avoid correcting frequencies outside the measurement’s reliable range. Measurement noise dominates at the extremes.

Causality (Magnitude)

Range: 0.0 (linear-phase) to 1.0 (minimum-phase)
Default: 1.0 (minimum-phase)

Controls the phase characteristic of the magnitude correction filter.

• 0.0 (Linear-phase): Symmetric pre- and post-ringing, constant group delay
• 1.0 (Minimum-phase): No pre-ringing, all energy is causal, variable group delay

Typical choice: 1.0 (minimum-phase) to avoid pre-ringing artifacts.

Why minimum-phase is preferred: Most acoustic and electronic systems are minimum-phase (magnitude and phase are coupled via Kramers-Kronig relations). Correcting magnitude with a minimum-phase filter automatically corrects the associated phase deviation, providing physically accurate correction. Use 0.0 (linear-phase) only if you specifically need constant group delay with a non-flat target curve and can tolerate pre-ringing.