The anatomy of a fader cap

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Overall Geometric Topology & Curvature

  • Base Volume: Begin with a standard rectangular cuboid. The lateral sides (left and right) remain strictly planar and parallel to each other.

  • The Primary Saddle (Concave): The top face features a deep, concave cylindrical boolean subtraction running along the transverse axis. This creates the central “scoop” or saddle where the finger rests. The nadir of this curve sits precisely at the center point of the top plane.

  • The Crests and Downward Trail (Convex): The front and back edges of the concave saddle do not end in sharp corners. Instead, they transition into high-radius convex fillets. These rounded crests roll smoothly downward, transforming the horizontal top surface into the vertical front and back faces. This continuous curve creates a soft, sweeping drop-off on both ends of the longitudinal axis.

  • OLYMPUS DIGITAL CAMERA

     

 

Surface Topography: The Grip Grooves

  • Geometry/Displacement: Embedded entirely within the concave saddle are a series of parallel, shallow horizontal striations.

  • Orientation: These grooves run along the transverse axis (perpendicular to the fader’s sliding direction).

  • Shading Execution: In a PBR workflow, these should be handled via a normal map or displacement map using a subtle sine-wave profile. They are designed to catch grazing light, creating alternating horizontal micro-bands of soft specular highlights and self-shadowing (ambient occlusion) within the deeper scoop.

The Indicator Line Wrap

  • UV Projection: The stark indicator line requires a continuous UV projection mapped along the longitudinal axis.

  • Path of the Wrap: The solid band originates on the flat, lower-front vertical face. It sweeps seamlessly up and over the front convex crest, plunges directly through the center of the ribbed concave saddle (bisecting the horizontal grooves at a perfect 90-degree perpendicular angle), climbs up the rear convex crest, and drops down to terminate on the rear vertical face.

  • Material Contrast: The painted line must have an independent material ID or a mask. It requires a high-albedo, purely diffuse surface (Roughness 0.9+) with zero specularity, ensuring the line remains readable and does not reflect light the way the base plastic does.

Material & PBR Shading Instructions

  • Base Material: Dielectric polymer (injection-molded plastic).

    Index of Refraction (IOR): ~1.45

  • Roughness: Set to medium-low (approx. 0.35). The plastic should exhibit a slight sheen. Specular reflections should be visible but slightly blurred, simulating a micro-texture that prevents a perfect, mirror-like gloss.

  • Specular Highlights: The primary lighting interaction will occur on the front and back convex crests. These tight curves should act as sharp specular catchers, grabbing overhead lights and creating bright, curved horizontal glints that define the upper boundaries of the fader cap.

  • Shadowing: Soft shadowing and heavy ambient occlusion should pool in the central concave depression, contrasting with the highly illuminated convex lips.

Drum kit desk

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Merging a professional workspace with an electronic kit like your Alesis Nitro Max is the ultimate space-saving play, Anthony. By treating the desk frame as your drum rack, you can shift from “deep work” to “deep grooves” without ever leaving your seat.

Here is how to re-engineer that Alesis setup into a “Drum-Desk” hybrid.

1. The Core Foundation: Height & Footing

The primary challenge of a hybrid station is vertical clearance.

* The Desk: You almost certainly need an adjustable-height standing desk. A standard fixed desk (typically 29 inches) is too low to accommodate the upward travel of your knees while playing the kick and hi-hat pedals. You’ll want to raise it to “drumming height” to clear your sticks and lower it for “typing height.”

* The Seat: Swap the office chair for a high-quality drum throne. While it lacks a backrest for long-term lumbar support, it provides 360-degree mobility and—crucially—won’t roll away when you bury the beater into the kick pedal.

2. Component Placement: The “Wrap-Around” Strategy

To keep the workspace functional, the drum components must integrate into the desk’s footprint rather than fight it.

The “Under-Desk” Zone (Kick & Snare)

* Kick Pedal: Position the kick pad/trigger directly against a wall or a heavy desk leg to prevent “creep” during play.

* Snare Drum: This is the trickiest piece. Use a dedicated snare stand set to its lowest position. When working, it sits between your knees; when drumming, slide your keyboard back to clear “stick room” for rimshots.

The “Desktop” Zone (Toms & Hi-Hat)

* Hi-Hat: Position the Alesis hat pad to the left of your laptop. If you use a stand-alone hi-hat stand, place it just outside the desk’s left leg.

* Toms: Instead of using the bulky Alesis rack, use multiclamps. You can clamp the four tom pads directly to the desk edges or even to your monitor arms to keep the desk surface clear.

* High Toms: Flank your center monitor.

* Floor Toms: Clamp these to the right side of the desk, near your mouse area.

The “Aerial” Zone (Cymbals)

* Ride & Crashes: Use long cymbal boom arms clamped to the back of the desk or your monitor pole. This keeps the “swing” area of the cymbals above your three monitors so they don’t obstruct your view while you’re working.

3. Tech Integration & Cable Management

Since you’re running three monitors and a laptop, your “brain” (the Nitro Max module) should act as the audio bridge.

* Audio Routing: Connect the Module Out to your computer’s audio interface. This allows you to hear your drums through your studio monitors or workspace headset while simultaneously playing along to Spotify or YouTube lessons on your screens.

* Cable Management: This is critical. Use Velcro ties to run all trigger cables along the underside of the desk frame. A “rat’s nest” of cables near your feet will interfere with your pedal work and ruin the clean aesthetic of your office.

Hybrid Layout Summary

| Component | Placement Strategy |

|—|—|

| Snare | Centered, low-profile stand between the legs. |

| Kick | Floor level, braced against a desk leg. |

| Hi-Hat | Left side, clear of the laptop/outer monitor. |

| Toms | Clamped to desk edges or monitor poles. |

| Cymbals | Boom arms extending from the back of the desk. |

| Module | Mounted under the desk surface for easy “blind” reach. |

 

Drone-pressor

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The Drone-pressor is a specialized audio restoration and spectral processing plugin designed for film post-production, broadcast, and field recording cleanup. Unlike a standard compressor or noise gate, it uses AI-driven motion-tracking algorithms to isolate and suppress the specific non-linear acoustic signatures of Unmanned Aerial Vehicles (UAVs).

Below are the technical specifications for its primary control parameters.

1. Pitch Take-Off Speed

This parameter controls the sensitivity of the plugin’s tracking oscillator. Because drone motors increase in pitch exponentially during takeoff or rapid maneuvers, a standard static filter would fail to track the sound.

* Specification: A temporal-frequency variable that defines the rate (in Hz/ms) at which the suppression filter can “climb” or “dive.”

* Function: Lower settings are used for stable, hovering drones. Higher settings allow the plugin to maintain suppression during aggressive vertical climbs or “punch-outs” where the RPM spikes instantly.

2. Doppler Effect Remover

This is the “spatial de-shifter.” It counteracts the frequency compression and expansion caused by the drone’s movement relative to the microphone.

* Specification: An inverse-motion algorithm that calculates the radial velocity of the sound source.

* Function: It “un-stretches” the audio in real-time. It identifies the pitch drop (the “Nee-Yum” sound) and applies a compensatory pitch-shift in the opposite direction, keeping the fundamental drone frequency centered so the narrow-band notch filters can remain locked on the target.

3. Drone Size (UAV Mass Class)

The acoustic signature of a drone is dictated by the length and material of its propellers. Smaller drones “whine” (high frequency), while larger drones “thrum” (low-mid frequency).

* Specification: A multi-band spectral profile selector ranging from Nano/Micro (250g) to Industrial/Heavy Lift (25kg+).

* Function: Adjusting this shifts the plugin’s “Attention Mask.”

* Small: Focuses on 2kHz to 15kHz.

* Large: Focuses on 150Hz to 1.5kHz.

4. Optical Metadata Integration (The Camera Toggle)

This unique feature allows the plugin to sync with visual data from the film set or the drone’s onboard feed to improve accuracy.

Parameter: Camera Sync (Enabled)

* Specification: Utilizes an Optical Flow Analysis bridge.

* Function: The plugin “looks” at the video file in the DAW. If the drone is visually moving left to right at a certain speed, the plugin automatically calculates the required Doppler correction and suppression strength based on the pixel-distance from the camera. It automates the parameters based on what is seen on screen.

Parameter: Blind Mode (No Camera)

* Specification: Pure Acoustic Inference Engine.

* Function: The plugin relies entirely on the audio signal. It uses a “best-guess” AI model to identify the drone type and distance based solely on harmonic distortion and the sound pressure level (SPL). This is used for audio-only recordings or when the drone is “off-camera.”

Summary Table for DAW Implementation

| Parameter | Unit | Purpose |

| Slew Rate | Hz/ms | Tracks how fast the motor pitch changes. |

| Shift Comp | Percentage | How much of the Doppler shift to flatten. |

| Mass Profile | grams/kg | Adjusts the frequency “sweet spot.” |

| Optic Link | On/Off | Syncs suppression to visual movement. |

To effectively remove the sound of a drone from an audio track, the “Drone-pressor” needs parameters that address the specific physics of multi-rotor flight. Because drones are moving targets with shifting frequencies, a simple static filter won’t work.

Here are the essential parameters for the removal engine:
1. Fundamental Frequency Tracking (f_0)
Since drone motors spin at variable speeds, the “noise” isn’t a single tone—it’s a moving target.
* Auto-Center: A real-time pitch tracker that locks onto the primary whine of the motors.
* Harmonic Multiplier: Drones produce integer multiples of their base frequency (overtones). This parameter allows you to suppress the 2nd, 3rd, and 4th harmonics simultaneously with a single slider.
2. Blade Count & Symmetry
The number of propellers changes the “texture” of the sound.
* Blade Parameter: Select between Tri-blade (smoother, higher frequency) or Bi-blade (choppier, more aggressive).
* Phase Offset: Adjusts for the fact that four motors are never perfectly in sync, creating a “beating” or “pulsing” effect.
3. Spatial Motion Parameters
These deal with the drone’s physical movement through the 3D sound field.
* Radial Velocity (Doppler Fix): A “Look Ahead” feature that predicts the pitch drop as the drone passes the microphone, adjusting the notch filters before the frequency shift occurs.
* Proximity Gain: An inverse-square law filter. As the drone gets closer (louder), the suppression intensity automatically increases to prevent clipping or “bleed.”
4. Turbulence & “Prop Wash” Removal
When a drone descends or turns sharply, it creates chaotic, low-frequency air turbulence that sounds like “buffeting.”
* Wash-Gate: A frequency-dependent transient shaper that targets the “thumping” air sounds without affecting the dialogue or ambient background.
* Jitter Reduction: Smooths out the micro-fluctuations in pitch caused by the flight controller making thousands of tiny motor adjustments per second.
5. Environment & Occlusion
* Reflectivity Scale: Adjusts how the plugin handles “echoes” of the drone bouncing off buildings or trees.
* Optical Occlusion (Camera Link): If the camera “sees” the drone go behind a wall, this parameter automatically softens the high-frequency suppression, as the wall would naturally act as a low-pass filter.
Summary of Parameter Controls
| Parameter Name | Unit | Action |
|—|—|—|
| Harmonic Depth | dB | How aggressively to cut the overtones. |
| Slew Rate | ms | How fast the filter follows a pitch change. |
| Blade Profile | 2 / 3 / 4 | Matches the filter shape to the prop type. |
| Doppler Ratio | M/S | Compensates for the speed of the passing drone. |

VU meter iterations

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Ever wonder why VU meters are always rectangular or circular?

It’s usually a matter of mechanical necessity. In the analog world, the physical sweep of a needle and the housing required to protect it dictated the design. We’ve become so accustomed to these “skeuomorphic” constraints that anything else feels almost alien—mechanically impossible, and therefore, aesthetically foreign.

But when you move from physical hardware to dynamic variables, hooks, and handles, those walls disappear.

The Danger & Joy of “Outside the Box”

Iterating without boundaries is a double-edged sword:
-The Danger: You can lose the user. If a shape is too “unseen,” it loses its familiarity and function.
-The Joy: You unlock unlimited potential. By manipulating the geometry through code, I’ve been riffing on the classic VU meter to see where the math takes me.

I’ve had to invent a new vocabulary just to keep track of these iterations. Say hello to the Squircle, the Squectangle, and the Hex-Dome.

Breaking the Skewmorphic Ceiling:
By leaning into the “mechanically impossible,” we create something that couldn’t exist in a world of gears and glass. It challenges the eye and redefines what an interface can look like.

Personally, the Parking Meter style is my favorite—there’s something inherently authoritative and nostalgic about that heavy arc.

Which of these shapes do you think works best? Or have we pushed “outside the box” too far?

#DesignSystems #UIUX #IterativeDesign #CreativeCoding #VUMeters #ProductDesign

Zero-Copy Sound: How MXL Reinvents Audio Exchange for the Software-Defined Studio

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The broadcast industry is undergoing a fundamental shift from hardware-centric systems to software-defined infrastructure, a move championed by initiatives like the EBU Dynamic Media Facility (DMF). At the heart of this transition lies the Media eXchange Layer (MXL), a high-performance data plane designed to solve the interoperability challenges of virtualized production. While MXL handles video through discrete grains, its approach to audio—via Continuous Flows—represents a sophisticated evolution in how compute resources exchange data using shared memory.

The Move from Sending to Sharing
Traditional IP broadcast workflows rely on a “sender/receiver” model involving packetization and network overhead. MXL replaces this with a shared memory model. In this architecture, media functions (such as audio processors or mixers) do not “send” audio; rather, they write data to memory-mapped files located in a tmpfs (RAM disk) backed volume known as an MXL Domain.
This allows for a “zero-overhead” exchange where readers and writers access the same physical memory, eliminating the CPU cycles usually wasted on copying data or managing network stacks. Continue reading →

Schematic Semantics: Ethernet left or right side

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The debate over whether an Ethernet port functions as a transmitter or a receiver on a schematic is the technical equivalent of the “toilet paper over or under” argument. It is a fundamental disagreement over orientation that often ignores the fact that the utility remains the same regardless of which way the roll is hanging.

Traditionally, schematics follow a rigid left-to-right flow: sources (transmitters) live on the left, and sinks (receivers) live on the right. This worked perfectly for analog audio or serial data where electricity moved in one direction. Ethernet, however, is a bidirectional transceiver technology. It is constantly “pushing” and “pulling” simultaneously, which breaks the traditional rules of drafting.

The Access vs. Consumption Debate

Many designers view the Ethernet switch as the “provider.” In this mental model, the switch is the source of connectivity, sitting on the left side of the page and “feeding” access to the edge devices on the right. The edge device is seen as the consumer of the network.

Conversely, others view the edge device as the “source” of the data itself. If a 4K camera is generating a video stream, that camera is the transmitter, and the switch is merely the consumer of that stream. In this scenario, the camera sits on the left, and the switch sits on the right.

Why It Is Like Toilet Paper

Just like the “over or under” debate, both sides have logical justifications that feel like common sense to the practitioner:

* The “Over” (Switch as Source) Argument

* It prioritizes infrastructure. Without the switch, there is no signal path.

* It follows the logic of power distribution, where the source of “energy” (in this case, data access) starts at the core.

* It treats the network as a utility, similar to a water main providing flow to a faucet.

* The “Under” (Edge as Source) Argument

* It prioritizes the payload. A switch with no devices has nothing to move.

* It maintains the “Signal Flow” tradition. If a microphone generates audio, it must be on the left, regardless of whether it uses an XLR or an RJ45 jack.

* It focuses on the intent of the system (e.g., getting video from a camera to a screen).

The Best Mechanism for Drafting

The shift in modern schematic design is moving away from seeing the switch as a “provider of access.” Instead of trying to force a bidirectional “highway” into a one-way “pipe” layout, the most effective designers are treating the switch as a neutral center point.

By placing the network switch in the center of the drawing, you acknowledge its role as a transceiver. You can then place “Signal Generators” (like cameras or microphones) to the left of the switch and “Signal Consumers” (like displays or speakers) to the right. This acknowledges that while the switch provides the “road,” it is the edge devices that provide the “traffic.”

Ultimately, as long as the drawing is consistent, it doesn’t matter if the “paper” is hanging over or under—as long as the data reaches its destination.

 

Reference of Audio Metering

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The Master Reference of Audio Metering

Audio metering serves two distinct masters: Psychoacoustics (how loud the human ear perceives sound) and Electrical Limits (how much voltage the equipment can handle). No single meter can do both perfectly.

This document compiles the ballistics, scales, visual ergonomics, and technical implementations of the world’s major audio metering standards.

Continue reading →

The 50ms Loop: A Broadcast Engineer’s Confusion with AV Standard Practice

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By: Anthony Kuzub. A Confused Broadcast Engineer

I’ve spent the last twenty years in OB vans and broadcast control rooms. In my world, “latency” is a dirty word. (production offsets is what I like to call them) If a signal is one frame out of sync, we panic. If a host hears their own voice in their IFB (earpiece) with even a 10ms delay, they stop talking and start shouting at us. You can judge an in ear mix by how far they throw the beltpack.

So can you imagine my confusion when I recently stepped into the commercial AV world to help commission a high-end boardroom.

I opened a clients DSP file—a standard configuration for a room with ceiling mics and local voice lift—and I saw something that made me think I was reading the schematic wrong.


There were Acoustic Echo Cancellation (AEC) blocks on everything.

Not just on the lines going to the Zoom call (where they belong), but on the microphones being routed to the local in-room speakers. I turned to the lead AV integrator and asked a genuine question:
“Why are we running the podium mic through an echo canceller just to send it to the ceiling speakers five feet away?”
His answer was, “To stop the echo.”
And that is when my brain broke.

The Chicken, The Egg, and The Latency

In broadcast, we follow a simple rule: Signal flow follows physics.

If I am standing in an empty room and I speak, there is no electronic echo. If I turn on a microphone and send it to a speaker with zero processing, there is still no echo—there might be feedback (squealing) if I push the gain too high, but there is no distinct, slap-back echo.

So, I looked closer at the AEC block in the software.

• Processing Time: ~20ms to 50ms (depending on the buffer and tail length).
Suddenly, the math hit me. By inserting this “safety” tool into the chain, we were effectively delaying the audio by nearly two video frames before it even hit the amplifier.

Here is the loop I saw:

1. The presenter speaks.
2. The DSP holds that audio for 50ms to “process” it.
3. The audio comes out of the ceiling speakers 50ms late.
4. The microphones at the back of the room hear that delayed sound.
5. Because 50ms is well beyond the Haas Effect integration zone, the system (and the
human ear) perceives this as a distinct second arrival. A slap-back. An echo.

Creating the Problem to Fix the Problem

I realized that in this room design, the AEC wasn’t curing the echo; it was the source of it.
Because the system was generating a delayed acoustic signal, the other microphones in the room were picking up that delay. The integrator’s solution? “Oh, just put AEC on those back mics too.”
It felt like watching a doctor break a patient’s leg just so they could bill them for a cast.
In the broadcast world, we use “Mix-Minus” (or N-1). If a signal doesn’t need to go to a destination, you don’t send it. If a signal doesn’t need processing, you bypass it. You strip the signal path down to the copper.

The “Empty Room” Test

I proposed a crazy idea to the team. I asked them to imagine the room completely empty. No Zoom call. No Microsoft Teams. Just a guy standing at a podium speaking to people in chairs.
• Is there a remote caller? No.
• Is there a far-end reference signal? No.
• Is there a need to cancel anything? No

If we simply bypassed the AEC block for the local reinforcement, the latency dropped from 50ms down to about 2ms. At 2ms, the sound from the speakers arrives at the listener’s ear almost simultaneously with the actual voice of the presenter. The “echo” vanishes.

The system became stable not because we added more processing, but because we stopped fighting physics.

A Plea from the Control Room
I’m still learning the ropes of AV, and I know that VTC calls are complex. But I can’t help but feel that we are over-engineering our way into failure.
If you have to use an Echo Canceller to remove an echo that you created by using an Echo Canceller… maybe it’s time to just turn the Echo Canceller off.

Rotary Selector Switch (SelectorSwitch)

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Rotary Selector Switch (SelectorSwitch)

The `SelectorSwitch` is a high-fidelity Tkinter Canvas-based widget designed to model discrete multi-position controls. It mimics the behavior of physical rotary switches found on industrial equipment, laboratory instruments, and high-end audio gear.

Continue reading →

MDP – Multi Dimensional Panner

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MDP – Multi Dimensional Panner

Demo: https://like.audio/MDP/

## Overview

The **Multi-Dimensional Panner (MDP)** is an advanced user interface concept designed for spatial audio mixing, object-based panning (e.g., Dolby Atmos), and complex parameter control. It extends the traditional “Linear Travelling Potentiometer” (LTP) by placing it within a free-floating, rotatable widget on a 2D plane.

Continue reading →

CMDP: Circular Motion Displacement Potentiometer

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CMDP: Circular Motion Displacement Potentiometer

DEMO: http://like.audio/CMDP

# CMDP: Circular Motion Displacement Potentiometer

Overview
The **Circular Motion Displacement Potentiometer (CMDP)** is a novel user interface concept designed for spatial audio mixing, microphone array management, and multidimensional sound control. It combines the precision of linear faders with the intuitive spatial organization of a polar coordinate system, allowing users to visualize and manipulate sound sources in a 360-degree field.

 

Continue reading →

The Great Un-Boxing: Audio’s Transition from Signal to State

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The Great Un-Boxing: Audio’s Transition from Signal to State

For decades, the broadcast world was defined by physics. We built facilities based on the “Box Theory”: distinct, dedicated hardware units connected by copper. The workflow was linear and tangible. If you wanted to process a signal, you pushed it out of one box, down a wire, and into another. The cable was the truth; if the patch was made, the audio flowed.

Today, we are witnessing the dissolution of the box.

The industry is currently navigating a violent shift from Signal Flow to Data Orchestration. In this new paradigm, the “box” is often a skeuomorphic illusion—a user interface designed to comfort us while the real work happens in the abstract.

From Pushing to Sharing

The fundamental difference lies in how information moves. In the hardware world, we “pushed” signals. Source A drove a current to Destination B. It was active and directional.

In the software world of IP and virtualization, we do not push; we share. The modern audio engine is effectively a system of memory management. One process writes audio data to a shared block of memory (a ring buffer), and another process reads it. The “wire” has been replaced by a memory pointer. We are no longer limited by the number of physical ports on a chassis, but by the read/write speed of RAM and the efficiency of the CPU.

The Asynchronous Challenge

This transition forces us to confront the chaos of computing. Hardware audio is isochronous—it flows at a perfectly locked heartbeat (48kHz). Software and cloud infrastructure are inherently asynchronous. Packets arrive in bursts; CPUs pause to handle background tasks; networks jitter.

The modern broadcast engineer’s challenge is no longer just “routing audio.” It is artificially forcing non-deterministic systems (clouds, servers, VMs) to behave with the deterministic precision of a copper wire. We are trading voltage drops for buffer underruns.

The “Point Z” Architecture

Perhaps the most radical shift is in topology. The line from Point A (Microphone) to Point B (Speaker) is no longer straight.

We are moving toward a “Point A → Cloud → Point Z → Point B” architecture. The “interface layer” is now a complex orchestration of logic that hops between cloud providers, containers, and edge devices before ever returning to the listener’s ear. The signal might traverse three different data centers to undergo AI processing or localized insertion, creating a web of dependencies that “Box Thinking” can never fully map.

The era of the soldering iron is giving way to the era of the stack. We are no longer building chains of hardware; we are architecting systems of logic. The broadcast facility of the future isn’t a room full of racks—it is a negotiated agreement between asynchronous services, sharing memory in the dark.

The Open Concept License 

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The Open Concept License

Copyright © 2026 Anthony Kuzub

This license allows for the free and open use of the concepts, designs, and software associated with this project, strictly adhering to the terms set forth below regarding nomenclature and attribution.

1. Grant of License

Permission is hereby granted, free of charge, to any person obtaining a copy of this design, software, or associated documentation (the “Work”), to deal in the Work without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Work, subject to the following conditions.

2. Mandatory Nomenclature

Any implementation, derivative work, or physical hardware constructed using these concepts must formally and publicly utilize the following terminology in all documentation, marketing materials, and technical specifications:

LTP: Linear Travelling Potentiometer

GCA: Ganged Controlled Array

3. Attribution and Credit

In all copies or substantial portions of the Work, and in all derivative works, explicit credit must be given to Anthony Kuzub as the source of inspiration and original concept. This credit must be prominent and clearly visible to the end-user somehow.

4. “As Is” Warranty

THE WORK IS PROVIDED “AS IS”, WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE WORK OR THE USE OR OTHER DEALINGS IN THE WORK.

Rust Headless 96kHz Audio Console

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Architecting a Scalable, Headless Audio Console in Rust

In the world of professional audio—spanning broadcast, cinema, and large-scale live events—the mixing console is the heart of the operation. Traditionally, these have been massive hardware monoliths. Today, however, the industry is shifting toward headless, scalable audio engines that run on standard server hardware, controlled remotely by software endpoints.

This article proposes the architecture for Titan-96k, a scalable, 32-bit floating-point audio mixing engine written in Rust. It is designed to handle everything from a simple podcast setup to complex 7.1.4 immersive audio workflows, controlled entirely via MQTT.

Continue reading →

Crawler visualizer

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Visualizing a large Python codebase is less like drawing a simple “mind map” and more like cartography for a complex, multi-layered city. A standard mind map has one central idea branching out. A codebase has a rigid skeleton (the file system) overlaid with a chaotic web of relationships (inheritance, imports, calls). Continue reading →

(GCA) Ganged Controlled Array

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(GCA) Ganged Controlled Array – Anthony P. Kuzub (Anthony@Kuzub.com)

DEMO:  https://like.audio/GCA/

## Overview

The **Ganged Controlled Array (GCA)**, also known as the **Composite Fader**, is a high-density user interface widget designed to manage multiple related parameters (channels) through a single “Master” fader cap. It solves the problem of controlling groups of values (e.g., a 5.1 surround mix, a drum bus, or an RGB color mix) where maintaining relative offsets is critical, but screen real estate is limited.

# Composite Smart-Fader Design & Style Guide

Continue reading →