A Fedora COPR repository for Audio creation
Practical audio routing and monitoring
The goal is to help new users get sound working first, then gradually explore the deeper features of the system.
From input (mic or instrument) to output (speakers/headphones), understand how audio flows in a basic setup using Carla.
You’ve explored the essentials with Audinux. It is time to start building your audio workflow.
Audinux is a Fedora COPR repository designed to provide a complete audio production environment on Fedora. Audinux maintains the latest upstream versions of audio packages to ensure users have access to the most up-to-date features and fixes. Below is a categorized summary of the main software available in Audinux:
This guide demonstrates how to set up a flexible audio routing and monitoring environment on Fedora using tools provided by Audinux. Whether you’re a podcaster, musician, or voice-over artist, understanding signal flow and real-time monitoring is crucial.
Proper audio routing allows you to:
Carla is a powerful modular plugin host. It’s especially useful for setting up your audio environment graphically with JACK.
Steps:
Example:
[Mic Input] → [LSP Compressor (LV2)] → [LSP Parametric EQ (LV2)] → [Ardour Track Input]
The Patchbay in Carla allows you to visually connect audio inputs and outputs between devices, plugins, analyzers, and monitoring endpoints. This graphical interface is ideal for setting up complex signal chains with real-time control.
Objective: Route microphone input through effects, analyze the signal, and monitor the output via headphones.
Steps:
LSP Spectrum Analyzer
(LV2) — visualizes frequency responsex42 Stereo Meter
— shows stereo balance and peak levelsEach node in Carla’s Patchbay represents a JACK client or port. Simply click and drag connections between them to build your signal chain. This setup provides precise control and immediate visual feedback, ideal for recording or live use.
Goal: Apply compression and EQ to voice while monitoring with real-time spectrum analysis.
Steps:
Input:
USB microphone
LSP Compressor (LV2)
– dynamic control of voice levelsLSP Parametric EQ (LV2)
– clean up and enhance voice toneAnalysis:
LSP Spectrum Analyzer (LV2)
– monitor vocal spectrum in real time
Ardour
for multi-track recordingsystem:playback
(via JACK or PulseAudio bridge) for live monitoringThis workflow is built in Carla with the JACK backend. All plugins are available via the Audinux COPR repository.
Goal:
Set up a guitar sound using AI-based amp modeling, allowing for real-time playing and recording.
Steps:
Input:
Connect your guitar to a USB audio interface.
NAM Loader
plugin (Neural Amp Modeler) in Guitarix to load an AI-based amp model.Tube Screamer
or Rat
.system:playback
.Note:
This setup is typically built using Carla with the JACK backend. All plugins are available via the Audinux COPR repository.
For a more detailed guide and background, see Yann Collette’s article:
Using Artificial Intelligence to Set a Guitar Sound
LSP Plugins (LV2) include a set of high-quality analysis tools for real-time visual monitoring during mixing or recording.
These tools provide visual feedback on:
Use Cases:
Audinux provides a robust and modular toolkit for managing audio routing and monitoring workflows. By combining Carla, JACK, and powerful analyzer plugins, you can build a flexible and efficient audio environment tailored to your creative and technical needs.
The voice chain is the signal path from the microphone to the final output, which often includes various stages such as EQ, compression, and monitoring. Understanding each component and how they work together is crucial for achieving optimal sound quality in your recordings. In this guide, we’ll explore the essential stages of a voice chain and how you can use them to enhance your vocal recordings.
This section focuses on the physical setup (microphone selection and audio interface), followed by the routing of the voice through Carla. The key idea is to create a good starting point with solid hardware and ensure the signal is correctly routed within Carla(
In audio production, a voice chain refers to the series of audio processing stages your vocal signal passes through — from the microphone input, through various effects (plugins), until it reaches the final output. Each stage shapes or controls the sound for clarity, consistency, and musicality.
Unlike instrument chains or MIDI-based setups (which are very popular topics in the Linux audio community, especially around synths, guitars, or modular setups), a voice chain is unique because:
Voice is highly dynamic and varies greatly between speakers.
Human hearing is especially sensitive to vocal nuances (dynamic swings, breathing noise, plosives).
Real-time monitoring is often more critical (you hear yourself through the chain during recording).
Low latency and clean dynamics are much more important than modulation or creative effects.
In short:
That’s why focusing on the voice chain offers a super practical guide for:
Podcasting
Voice-overs
Singing
Online meetings or live streaming
To begin building a solid voice chain, the first step is setting up the hardware and ensuring the appropriate software routing in Carla. Here’s how you can do this effectively.
The foundation of your voice chain starts with your microphone and audio interface. The quality of your hardware will significantly affect the final result. Common setups include:
Once your hardware is set up, you can use Carla to route audio to and from your recording environment. Carla is a flexible, modular audio plugin host designed for real-time audio routing and plugin management. Carla gives you direct visual control over how your hardware, software instruments, and effects are connected.
Engine processing mode: This setting (visible under Engine settings) defines how Carla communicates with JACK. Carla’s default Multi-client (JACK) mode lets each plugin act as an independent JACK client, making it easy to route audio freely between plugins, hardware, and other software.
Rack/Patchbay mode controls how Carla manages plugins internally.
The following image demonstrates typical Graphical User Interface (GUI) of the Carla Rack:
Best for: Complex signal chains, live performance setups, combining software and external gear. The Patchbay canvas in Carla gives you a visual way to connect plugins, hardware, and system inputs/outputs like building blocks.
The following image demonstrates typical Graphical User Interface (GUI) of the Carla Patchbay Canvas:
In/Out Ports (Audio Ports) represent audio signals entering or leaving a particular device or plugin.
This section introduces LSP (Linux Studio Plugins) as essential tools for shaping the sound of your voice. The emphasis is on using EQ to refine the sound and the LSP Spectrum Analyzer for real-time feedback, enabling users to monitor and adjust the voice chain during the recording or live monitoring process.
EQ (Equalization) is used to shape the frequency response of your voice by boosting or cutting specific frequencies. Proper EQ can make your voice sound clearer, fuller, or more present, depending on your needs.
The following image demonstrates typical Graphical User Interface (GUI) of the LSP Parametric EQ:
The LSP Parametric Equalizer x8 Mono gives you 8 frequency bands to work with, which offers plenty of flexibility to shape the sound of a mono source. This makes it ideal for solo recordings or a single microphone setup (for instance, a voice-over recording or an acoustic guitar mic’d up in a simple setup).
Purpose: Determines the central frequency for each band. You can adjust where the EQ will target specific frequencies on the spectrum.
Load the EQ plugin in Carla (Example: LSP Parametric Equalizer x8 Mono). Right-click on the plugin block, select Enable, and make sure the plugin’s name (such as ‘LSP Parametric Equalizer x8 Mono’) is visible.
Common Adjustment:
Low-End: Set around 80–100 Hz to cut off unwanted rumble (High-Pass).
Midrange: Set around 250–1,000 Hz for controlling muddiness or boosting clarity.
High-End: Set around 4–12 kHz for boosting presence or adding air to the voice.
How to Adjust: Click and drag the Frequency knob left or right. You’ll see the frequency change in Hz. Alternatively, you can also adjust the EQ by clicking and dragging a frequency point upward or downward directly on the graph. Dragging upward boosts the selected frequency, while dragging downward cuts it.
In audio processing, a dry signal is the original, unprocessed sound, while a wet signal is the sound after it has been processed with effects like EQ, compression, or reverb. Imagine recording your voice with a microphone. The raw recording, without any effects, is the dry signal. If you add compression to control the dynamics and a bit of reverb to make it sound fuller, the result is the wet signal.
The following image demonstrates typical Graphical User Interface (GUI) of the bypass option on LSP plugins:
Right-click on the rack window of a plugin and tick the checkbox of ‘bypass’.
Always A/B test your changes. Bypass the EQ and listen to how your voice sounds without it, then engage the EQ and listen again. Make sure you’re not over-processing your voice.
Compression controls the dynamic range of your voice, ensuring that quieter parts are brought up in volume while loud parts are reduced. This results in a more consistent sound level throughout your recording.
The following image demonstrates typical Graphical User Interface (GUI) of the LSP Compressor plugins:
To save a configuration in Carla, you can use a .carxp file, which stores the setup of your patchbay, including all audio and MIDI routing, plugin settings, and connections. This file allows you to quickly restore your exact setup for future sessions without needing to manually reconfigure everything. To save a configuration, simply go to the File menu in Carla, choose Save Configuration As, and select the destination where you want to store your .carxp file. This file will contain all the details about your current patchbay, including the arrangement of plugins, the routing between them, and any device or system parameters. When you want to load the configuration later, you can open the .carxp file, and Carla will automatically restore your entire environment, making it a convenient and efficient way to manage your workflow.
x1 is simple, lightweight, perfect for voice work. Mono matches your voice chain since microphone is usually mono.
The following image demonstrates typical Graphical User Interface (GUI) of the LSP Spectrum Analyzer:
Singing helps you detect how plugins (especially EQ and compression) react to dynamic shifts and emotional expression. You can also catch subtle issues like breath noise or high-frequency peaks that don’t always show up during normal speaking.
Understanding your voice chain is essential for achieving the best possible sound quality in your recordings. By using EQ, compression, and monitoring tools correctly, you can shape and control your voice to suit your needs, whether for podcasts, voice-overs, or music recordings.
With a solid understanding of these tools and techniques, you can improve the quality of your voice recordings and achieve professional-sounding results.
You have explored the essentials with Audinux. Your next steps can align with your creative goals, whether recording a podcast, building live setups, experimenting with plugin chains, or diving into full multi-track production. Audinux gives you a modular foundation, but your journey continues with tools like Ardour, Pipewire-jack, and advanced plugin workflows. Your path depends on how deep you want to go - performance, production, or precision mixing. You’re not just using tools now. You’re shaping your audio environment.
Audinux offers opportunities with open-source tools: you’ve already explored plugin hosting, routing, and monitoring. The next step is personal—build a repeatable workflow, trust your ears, and keep it simple at first. As with any creative tool, familiarity grows with use. Audinux doesn’t aim to mimic commercial DAWs—it gives you the freedom to shape your sound, in your way.
Ardour Tutorial - Learn more about recording and editing audio on Linux. It covers setup, basic workflows, and essential tools in a concise format, making it ideal for newcomers.
Latency Tuning - Low-latency audio is key to real-time performance and smooth monitoring. Achieving it on Linux involves choosing the right kernel (often low-latency or real-time), tuning buffer sizes, and ensuring your system isn’t introducing unnecessary delays. Tools like JACK or PipeWire allow fine-grained control, and using interfaces with solid driver support helps. For a practical approach, including how AI can even help dial in guitar tones with low latency, check out the Fedora Magazine article on latency tuning.
Introduction