This release introduces major improvements to DSP performance and tuning
flexibility.
New Features
*Multithreaded Audio Engine*
Loopino now supports multithreaded audio processing to reduce load on
the main audio thread.
*
Audio processing can be buffered as half-frame or full-frame blocks
*
Buffered DSP blocks are processed in a worker thread
*
Significantly reduces DSP load and xruns in the main audio thread
*
Designed to improve stability under high polyphony and complex
modulation scenarios
*Micro Tuning Support (Scala)*
Loopino now supports microtonal tuning via Scala.
*
Built-in factory tuning scales included
*
Drag & drop support for Scala .scl files
*
Drag & drop support for Scala .kbm key mapping files
*
Flexible keyboard-to-scale mapping for alternative tuning systems
Notes
This update improves real-time performance and expands Loopino’s musical
language beyond standard equal temperament, making it suitable for
high-load sound design and microtonal composition alike.
Project Page:
https://github.com/brummer10/Loopino
Release Page:
https://github.com/brummer10/Loopino/releases/tag/v0.9.5
This release focuses on workflow improvements, clearer signal routing,
and new creative options.
### New Features
- **Drag & Drop Processing Chains**
- Filter and Machine chains can now be reordered via drag and drop
- Machine chain changes trigger a full key cache rebuild
- Filter chain changes apply immediately in real time
- **Reverse Sample Playback**
- Samples can now be played in reverse
- Fully integrated into the existing voice and filter pipeline
- **New Machine: Vintage (TimeMachine)**
- A new offline machine focused on temporal character and coloration
- Operates during key cache generation
- Designed for non-destructive experimentation with timing and feel
---
### Architecture & Workflow
- Clear separation between **offline machines** and **real-time filters**
- Deterministic signal flow from sample → machine → key cache → voices →
filters
- Improved internal consistency and predictability
---
### Documentation
- Added a new [**Loopino
Wiki**](https://github.com/brummer10/Loopino/wiki/User-Documentation)
- User-facing documentation covering:
- Sample loading and destructive trimming
- Machines vs Filters
- Signal flow and processing stages
- Documentation aims to be precise, technical, and transparent
---
### Notes
- Existing projects remain compatible
---
Project Page:
https://github.com/brummer10/Loopino
Release Page:
https://github.com/brummer10/Loopino/releases/tag/v0.9.0
As always, feedback is welcome.
Hi all,
just a quick "addendum" to the earlier announcement here:
The organizing team of this year's Linux Audio Conference (see below) has also
given this conference a "theme" that should spark some ideas for papers or
discussion, and has now added the following blurb to the home page
(see https://lac.linuxaudio.org/2026/) -
###
Conference theme:
Large language models and Free/Libre/open source software.
"I am not comfortable contributing to a project that extensively uses AI."
from a post found in a pull request to a github repository.
This year's LAC theme explores questions relating to the (sometimes uneasy)
relationships that may emerge between LLMs and FLOSS. This of course has many
dimensions, from the purely technical, through to the practical, and finally
to the ethical.
As code repositories such as github roll out support for third-party and their
own LLM agents, this is an area that needs vigorous discussion and assessment.
It is probably not a good idea to ignore it, as it is unlikely to go away.
It may be possible to formulate a position from the LAC community, which we might
carry forward for further consideration in other forums.
Even if such a thing cannot exist, it is still important that we put forward
our ideas in relation to this issue. Therefore, it makes good sense to invite
contributions to this theme and make it a central point of discussion at the LAC.
The LAC2026 organising team.
###
Greetings,
Frank
On Fri, 9 Jan 2026 17:33:01 +0100
Frank Neumann <beachnase(a)web.de> wrote:
> Hi all,
>
> just wanted to share the good news here that in 2026 the LAC (Linux Audio Conference) is
> taking place again, on June 18-20 (Thu-Sat), this time coming back to Maynooth (Ireland)
> where it was already hosted in 2011.
>
> Victor Lazzarini, conference organizer, asked me to help in spreading the word about
> it, so here we go.
>
> All details on music&paper submission process, deadlines, travel and accomodation etc can
> be found at the conference web site: https://lac26.mucs.club/
>
> Greetings, and please feel free to spread the word wherever possible,
> Frank
Hello everybody,
Best wishes for the new year! I've made some small changes to the LAU
mailing list that will hopefully reduce the number of bounces that some
members suffer from:
- The unsubscribe footer has been removed.
- The [LAU] prefix to the subject has been removed.
By making these changes the messages sent to the list should go through
unaltered which would make DMARC/DKIM happier. Especially removing the
[LAU] prefix might be problematic for some, hopefully this doesn't mess
up filtering all too much. So apologies beforehand for any inconvenience
these changes might cause. If any other problems arise from these
changes then let me know!
Best regards,
Jeremy
Hi all,
I'm running a RPi5 with the 8ch HifiBerry DAC (Debian 13), and on top of that JACK (1.9.22 I think). Sometimes when I startup the system, there's a channel "remapping" where channels 1-4 becomes 5-8 and vice versa. This is (reasonably) reliably "fixed" with a restart of the JACK service, albeit non-optimal.
Any ideas on where in the pipeline from JACK to the DAC driver this mess-up could occur? The RPi5 has 4 stereo I2S ports which are coalesced into 8 channels in the DAC driver.
Best regards
/Robert Bielik
The information in this email (including any attachments) may contain confidential and/or proprietary material. Any review, retransmission or use of this information by persons or entities other than the intended, authorized recipient is prohibited. If you received this email in error, please notify the sender and delete the material. For information regarding how Dirac handles personal data, please visit https://www.dirac.com/privacy-policy.
To follow-on to Fernando's response, I originally used
Leland Smith's SCORE program (the old SCORE, not the
manuscript program originally named MS, but renamed
to Score later), with Mus10 to turn the note list
into sound. Then the Samson box came along and
I used Pla for awhile with Sambox (the program that
turned the note list into a synthesizer command stream).
Later I got into the habit of writing a different
program for each piece, or even writing the note
lists by hand (Phrygians and various sound-file
based pieces). CLM and Snd came still later, but
by then I had somehow become a programmer which
was a much easier life than composition. I did
port most the old pieces to that environment. A
major unanticipated problem was hearing loss in old
age which makes it nearly impossible to check that
I'm hearing what others hear.
NeuralRack is a Neural Model and Impulse Response File loader for
Linux/Windows available as Stand alone application, and in the Clap, LV2
and vst2 plugin format.
It supports *.nam files <https://www.tone3000.com/search?tags=103> and,
or *.json or .aidax files <https://www.tone3000.com/search?tags=23562>
by using the NeuralAudio <https://github.com/mikeoliphant/NeuralAudio>
engine.
For Impulse Response File Convolution it use FFTConvolver
<https://github.com/HiFi-LoFi/FFTConvolver>
Resampling is done by Libzita-resampler
<https://kokkinizita.linuxaudio.org/linuxaudio/zita-resampler/resampler.html>
New in this release:
* implement option to move (drag and drop) EQ around
Neuralrack allow to load up to two model files and run them serial.
The input/output could be controlled separate for each model.
It features a Noise Gate, and for tone sharping a 6 band EQ could be
enabled.
Additional it allow to load up a separate Impulse Response file for each
output channel (stereo),
or, mix two IR-files to a two channel mono output.
Neuralrack provide a buffered Mode which introduce a one frame latency
when enabled.
It could move one Neural Model, or the complete processing into a
background thread. That will reduce the CPU load when needed.
The resulting latency will be reported to the host so that it could be
compensated.
ProjectPage:
https://github.com/brummer10/NeuralRack
Release Page:
https://github.com/brummer10/NeuralRack/releases/tag/v0.3.0
SpectMorph 1.0.0-beta3 has been released.
This version contains a new pitch detection algorithm for the instrument
editor and it can read mp3 files. Compared to 1.0.0-beta2 there are
mostly smaller fixes, but since some of them address critical problems
we strongly recommend updating to beta3 if you use a previous beta.
There is a tutorial on YouTube for the new features in the 1.0.0 series:
- https://youtu.be/mwVUsuOTcN0
Feedback for any issues you might experience with the beta version is
appreciated.
What is SpectMorph?
-------------------
SpectMorph is a free software project which allows to analyze samples of
musical instruments, and to combine them (morphing). It can be used to
construct hybrid sounds, for instance a sound between a trumpet and a
flute; or smooth transitions, for instance a sound that starts as a
trumpet and then gradually changes to a flute.
SpectMorph ships with many ready-to-use instruments which can be
combined using morphing.
SpectMorph is implemented in C++ and licensed under the GNU LGPL version
2.1 or later
Integrating SpectMorph into your Work
-------------------------------------
SpectMorph is currently available for Linux, Windows and macOS (Intel
and Apple Silicon), with CLAP/LV2/VST plugins. Under Linux, there is
also JACK Support.
Links:
------
Website: https://www.spectmorph.org
Download: https://www.spectmorph.org/downloads
There are many audio demos on the website, which demonstrate morphing
between instruments.
List of Changes in SpectMorph 1.0.0-beta3:
------------------------------------------
## SpectMorph 1.0.0 beta3
#### New Features
* Implement pitch detection algorithm for instrument editor and smenc (#31).
* Support mp3 format for static plugins and builds with new libsndfile.
#### Instrument Updates
* Trumpet, French Horn: ping pong loop, better tuning
* Bass Trombone: ping pong loop, volume normalization, tuning
* Alto Saxophone: ping pong loop
#### Reduce Memory Usage after Unload
* Avoid global constructors / destructors.
* Use our own TextRenderer instead of cairo to be able to free font cache.
* Ship necessary fonts on macOS for TextRenderer.
* Free various tables and other bits of static data when unloading.
#### Fixes
* Don't crash on invalid utf8 during conversion (use replacement char).
* Fix crash caused by multiple threads modifying control events.
* Fix CLAP's get factory implementation (#30).
* Various ASAN / UBSAN fixes.
* Fix RTSAN issue: make FFT realtime safe.
* Avoid allocating memory in RT thread if events need to be sorted.
* Fix (unlikely) LineEdit crash.
* Validate input for smenc -m and other utils where an integer is
expected (#31).
* Fix smooth tune performance for long input files.
* Build system updates.
* Convert manpages to markdown.
* Documentation updates.
--
Stefan Westerfeld, http://space.twc.de/~stefan
[I wanted to send this to LAD/LAU yesterday, but used the old list addresses by accident]
Hi all,
just wanted to share the good news here that in 2026 the LAC (Linux Audio Conference) is
taking place again, on June 18-20 (Thu-Sat), this time coming back to Maynooth (Ireland)
where it was already hosted in 2011.
Victor Lazzarini, conference organizer, asked me to help in spreading the word about
it, so here we go.
All details on music&paper submission process, deadlines, travel and accomodation etc can
be found at the conference web site: https://lac26.mucs.club/
Greetings, and please feel free to spread the word wherever possible,
Frank
