Manifold ~ Audio Batch Processing Tool


Manifold is a desktop audio batch-processing tool built for inspecting, transforming, and exporting large collections of audio files with a visual workflow editor. Useful for checking and pre-processing audio for game-audio production, podcasts and broadcast, it is designed to make repetitive tasks more efficient and reproducible.

It combines a polished ImGui-based interface with a custom node-based audio pipeline, allowing you to build repeatable processing chains for tasks such as loudness normalization, peak limiting, DC offset cleanup, resampling, and format conversion.Why Manifold?

Manifold was designed to make audio batch work feel more structured and reproducible. Instead of manually running ad-hoc commands or scripting every step, you can:

  • import whole folders of audio files,
  • inspect their metadata and analysis results,
  • create reusable processing chains,
  • export processed files with consistent naming and format settings.

This makes it useful for workflows such as podcast preparation, mastering prep, broadcast delivery, and game-audio asset processing.

Key Features:

  • Batch import of files and folders
  • Drag-and-drop support from the file explorer
  • Built-in audio playback for quick previews
  • File table with metadata and analysis metrics
  • Configurable processing workflows using nodes
  • Output settings for folder, naming, format, and subtype
  • Background processing with task history and logs
  • Parallel analysis for loudness and DC offset metrics

Supported Processing Nodes:

Manifold includes a growing set of audio processing and analysis nodes, including:

  • Loudness analysis
  • Loudness normalization
  • True peak limiting
  • DC offset removal
  • Peak normalization
  • Channel mixing
  • Silence trimming
  • Fade in/out
  • Loudness compliance checks
  • Clipping detection
  • Dynamic range analysis

These nodes can be chained together to create custom pipelines for different production scenarios.

github.com/AnClark/Manifold

NxTakt ~ Native Linux DAW


The workflow is Ableton Live’s: a grid of clips you launch against a global tempo grid, quantised to musical boundaries. The architecture is not Live’s, and that is the point.

Your set is text

A project is one line-oriented plain-text file with a byte-identical round-trip and stable per-entity uids. A melody is eight note <beat> <len> <pitch> <vel> lines. Diff a mix, branch an arrangement, review a track in a pull request, generate a set from a script.

The audio thread shares nothing

No allocation, no locks, no pointers into GUI memory: the interface reaches the engine through a lock-free command ring and a block of atomics. Launches are split inside the buffer, so a clip starts on the exact frame of its quantum rather than at the top of the next block. nxtaktd already runs that engine as its own process, playing clips out of a shared-memory sample pool.

CI plays what it builds

Every push builds the whole application, runs the headless suite — 2,500+ assertions across the engine, the IPC layer, the daemon and the devices — and then renders five passes of the demo set to FLAC, printing the peak and RMS of each into the run summary. The pipeline does not just compile this DAW. It hands you the audio.

nxtakt user interface

github.com/nerdrx/nxtakt

acidcat ~ Audio File Forensics


A pure-Python inspector, editor, and forensic tool for audio files and synth/DAW presets: read the metadata, decode the format structure byte by byte, flag anomalies, repair broken containers, and identify how a file was made.

Reads BPM, key, duration, tags, and format info from WAV, AIFF, MP3, FLAC, OGG, Opus, M4A, MIDI, and Serum presets. Also structurally decodes Bitwig (.bwpreset/.bwclip), Native Instruments (Massive/Absynth/Kontakt/NKS/KORE), Vital, NCW, SoundFont (SF2/SF3), tracker modules (MOD/XM/IT), MP4, VST FXP, ReCycle RX2, and RMID containers via inspect. Beyond reading: repair and validate fix and check container structure (stale sizes, offset tables, counts, pad bytes) without touching a byte of audio, and audit gives a forensic verdict (structure, integrity, hidden data, and the writing tool). One pure-Python dependency (mutagen); the native inspect walkers need nothing. Optional librosa analysis for BPM/key detection and ML feature extraction.

Also ships per-library SQLite indexes (acidcat index) tracked in a small global registry, plus an MCP server (acidcat-mcp) so an LLM can query your whole collection across libraries by bpm, key, tags, or full-text.

github.com/hed0rah/acidcat
github.com/hed0rah/acidcat-playground
www.databurn.org

Fender Studio ~ Record, Jam & Create


Record, jam, and capture your creativity whenever and wherever it strikes. Packed with authentic Fender tones, Fender Studio is fast, fun, and free for guitar players and music creators of all types.

fender.com/pages/fender-studio-app

HARP ~ Hardware Audio Runtime Protocol


Make hardware instruments behave like plugins. A Raspberry Pi 4B plays as an instrument track in Ableton Live today — Git-style total recall through the project’s own save/reopen, knobs as sample-accurate automation lanes, and offline bounce through the physical box.

Hardware synths, drum machines, and effects have connected to computers for forty years, and the connection is still fragile in one specific way: recall. Audio mostly works, MIDI mostly works — but reopen last month’s project and the hardware is not in the state you saved, and nothing will tell you. The deep integrations that fix this (multichannel USB audio, plugin-shell control, total recall) have existed only inside closed single-vendor ecosystems, because every vendor has to rebuild drivers, state sync, and DAW compatibility from scratch.

HARP is a complete, working implementation of that integration — with an open spec underneath, if you want to build a device on it. The reference device is a Raspberry Pi running a 16-part multitimbral, 8-voice-polyphonic synth (13 params per part); anything that speaks the protocol gets the same treatment from any conforming host:

  • Total recall, Git-style — device state is content-addressed and hash-verified; a saved project reopens with the hardware provably in the saved state, every overwrite preceded by a free snapshot, every mismatch resolved through explicit safe actions — never silently. The archive doubles as patch time-travel: every state the box has ever been pushed over is one click away.
  • Audio as a plugin — a dedicated stream into the plugin shell that bypasses the OS audio stack (no aggregate-device hacks), including a host-paced mode where the hardware renders deterministically, byte- identical, faster than real time: offline bounce through a physical box at ~25× real time, with 16 ms of reported latency at DAW buffers ≤ 256 — in the neighborhood of a good audio interface.
  • Multitimbral, addressed like plugins — one physical device is one session, and several shell instances can share it: drop the plugin on a handful of DAW tracks and each instance drives its own part — its own channel, params, recall state, and stereo output (16 parts, with a summed main mix alongside the per-part outputs). A recall-safe Part knob persists each track’s part in the project, and that per-part state moves intact between the VST3 and AU formats (the CLAP shell writes the same recall bundle).
  • Polyphonic, with per-voice modulation — each part is an 8-voice pool with deterministic voice allocation, so overlapping notes ring out on their own voices instead of stealing one. Modulation is non-destructive and per voice: a VST3 Note Expression or a CLAP per-note parameter modulation bends a single sounding note’s filter cutoff without touching the stored patch — and the two formats render byte-identically (the device applies the §9.5 mod the same regardless of which shell sent it).
  • Sample-accurate everything — DAW automation becomes device-interpolated ramps applied within ±1 sample; notes travel as UMP; an event fence makes “applied late” structurally impossible rather than statistically rare. Turn a knob on the hardware and the DAW records the automation (echo).
  • Musical time — devices follow the DAW’s transport: tempo, position, loops. The reference device’s arpeggiator locks to Live’s grid sample-exactly, survives loop wraps, and renders a byte-identical groove — determinism extended to musical time.
  • Identity, timing, diagnostics — engine versioning and parameter-map hashing protect old songs from new firmware, latency is measured (never guessed), and error counters at every layer end “it glitched” support threads with evidence.
  • Roadie-proof sessions — unplug the cable mid-song and plug it back: the shell reconnects, re-asserts the project’s state, and audio resumes. Hostile or corrupt wire input ends in a clean session reset, never a crash (every parser is fuzzed; a live abuse test is part of CI).

github.com/kschzt/harp

jamailmar ~ Multi-threaded Ogg Vorbis Decoder


Linux code to decode Ogg Vorbis files with multi-thread support

sourceforge.net/projects/jamailmar

Producer Player ~ Mastering Workspaces


A desktop app for producers who bounce a lot. Drag in a folder of exports and Producer Player auto-groups versions, organizes your album, and gives you a full mastering workspace — all in one place.

ethansk.github.io/producer-player
github.com/EthanSK/producer-player

FlavourMTC ~ Mixbus Tone Control


FlavourMTC follows classic “passive” equalizer designs where the EQ circuits itself are not able to amplify signals but a dedicated amplifier stage takes care of it. Those EQ designs are well known for allowing very transparent frequency changes while their amplifier designs do add some icing on the cake quite often.

FlavourMTC implements this by utilizing 1st order shelving filter designs avoiding unwanted resonances and takes advantage of “zero delay” implementations for most accurate higher order filtering and w/o introducing curve warping near Nyquist frequency. The output amplifier stage of the plugin can be calibrated according specific mixing levels, provides a distinct “box tone” and glues everything together. Parts of the plugin are oversampled internally for maximum transparency and sound quality.

varietyofsound.wordpress.com/downloads

shntool ~ WAVE Processing & Reporting Utility


shntool is a multi-purpose WAVE data processing and reporting utility. File formats are abstracted from its core, so it can process any file that contains WAVE data, compressed or not – provided there exists a format module to handle that particular file type.

shntool has native support for .wav files. If you want it to work with other lossless audio formats, you must have the appropriate helper program installed. The “Helper programs” section below contains links to helper programs for each format that shntool supports.

shnutils.freeshell.org/shntool
github.com/bayun/shntool

CLAMPS ~ Common Lisp Aided Music Production System


Clamps, short for “Common Lisp Aided Music Production System”, is a software system for realtime and non-realtime music production written in Common Lisp. It enables a seamless workflow between high-level structures to define musical processes all the way down to low level DSP definitions for sound creation including browser based interfaces for interactive work and control useable for live performances. In that respect it combines features of systems like OpenMusic, SuperCollider or the Pure Data/Max family of software.

codeberg.org/ormf/clamps
codeberg.org/ormf/clamps-install
github.com/arclanguage/Clamp

Cakewalk ~ Free Digital Audio Workstation


The most complete music production package

The creative experience. Only Cakewalk by BandLab offers: advanced technology, effortless workflow, and an interface that amplifies inspiration.

Compose ~ Easily compose complete songs and performances with creative songwriting tools and instruments

Record ~ Capture inspired performances with pristine quality and unlimited audio and MIDI tracks

Edit ~ Fix mistakes, arrange parts, and manipulate pitch, time, and any other aspect of your recordings

Mix ~ Achieve pristine pro-studio sounds and create big, clear, dynamic mixes with advanced mixing tools

Master ~ Polish your finished mixes to perfection with built-in mastering tools

Share ~ Export and publish your songs directly to YouTube, SoundCloud, Facebook and more

www.bandlab.com/products/cakewalk
https://bandlab.github.io/cakewalk/docs/FAQ/images/screenshot-cbb.png

Correlometer ~ Multi-band Correlation Meter


Correlometer is an analog-style stereo multi-band correlation meter.

Multi-band correlation meter is an advanced way to check for presence of out-of-phase elements in the mix. Broadband correlation metering reports overall phase issues and may misrepresent problems present in select spectral bands, while multi-band correlation meter easily highlights problems present in mid to high frequencies that are not easily heard by ear, but may still reduce clarity of the mix. Another application of multi-band correlation metering is phase- and time-aligning of channels and tracks, especially bass and bass-drum pairs, guitar mic and D.I. source pairs, two-microphone stereo recordings, etc.

Correlometer can display 4 to 64 individual spectral bands, with adjustable band quality factor that controls the degree of band’s selectivity. Averaging time of correlation estimator can be adjusted. Correlometer supports side-chain inputs for easy correlation estimation between separate audio tracks.

  • 4 to 64 band correlation meter.
  • Adjustable band quality factor.
  • Adjustable averaging time.
  • Side-chain input.
  • Resizable user interface.
  • All sample rates support.
  • Zero processing latency.
  • User interface color schemes.
  • Retina and High DPI support.
plugins4free.com/plugin/2957/