SpectMorph 0.6.1 has been released.
The main changes are:
- Instrument editor improvements
- Support for multiple banks for WavSources
- New standard instruments
- The code is now hard RT capable
- UI fixes for macOS
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: http://www.spectmorph.org
Download: http://www.spectmorph.org/downloads
There are many audio demos on the website, which demonstrate morphing
between instruments.
List of Changes in SpectMorph 0.6.1:
------------------------------------
#### Instrument Editor
* Support click & drag sample to scroll & zoom (#22).
* Support stereo to mono conversion when loading stereo samples (#14).
* Add manual volume editing / normalization.
* Implement automatic selection triggered by midi.
#### New instruments
* Bass Flute
* Soprano Saxophone
* Clarinet, Bass Clarinet
* Tenor Trombone
* Viola, Double Bass
* Make samples and meta information for standard instruments available
on github.
#### Improvements
* Support multiple banks for WavSources / instrument editor.
* Avoid allocations in DSP thread to be hard RT capable.
* Allow overriding analysis parameter for frame stepping to get higher
time resolution.
#### Fixes
* Make UI work properly in Ableton Live (and possibly other hosts) on macOS.
* Fix UI scaling problem on M1 macOS builds.
* Fix crash if instrument editor is closed without any samples.
* Fix cases of undefined behaviour.
* Fix timing problems for long notes, reproduce long WavSource notes
with exact tempo.
* Fix use-after-free for outdated control events.
* Fix freetype related memory leak.
--
Stefan Westerfeld, http://space.twc.de/~stefan
Hello all,
Release 0.4.1 of jnoisemeter is available at the usual place:
<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>
Jnoisemeter is a small Jack app for accurate measurement of
audio signals, in particular noise signals. See README for
the details.
New in this release: IEC class 0 octave band filters.
Ciao,
--
FA
Hi there,
I have released SoundTracker v1.0.4. Comparing to the last
pre-release, 1.0.3-pre2, it contains several important bugfixes, so I
recommend everyone to upgrade. Its main revolutionary feature is a
facility to process samples by any external program that can
communicate through STDIO. The interaction with such a program is
described used XML specs (not documented yet, but the comprehensive
example: much of SOX functions, is included), so one need not
recompile ST, only restart after adding new specs.
Other main features of the 1.0.4 release are:
* General:
- Full-screen mode
- Volume of all samples of an instrument / all samples in the module can be
set to a value, not only gained. Also all samples panning can be adjusted.
These functions can also modify the explicitly given values of volume /
panning of notes in patterns
- Improved compatibility with FastTracker, also MOD files are played more
correctly
- Unused patterns are listed in the Pattern Sequence Editor
- Sampling rate can be specified while saving a sample as WAV
- New module optimizer with many control parameters
* Track Editor:
- Moving notes up / down is implemented
* Sample Editor:
- The whole sample is drawn after recording
- Exponential / reverse exponential volume fade transients
* Instrument Editor:
- Envelope inversion and shifting is implemented
You can find more new features in NEWS file.
SoundTracker download page:
https://sourceforge.net/projects/soundtracker/files/
Regards,
Yury.
Anklang version 0.2.0 is released.
Anklang is a digital audio synthesis application for live
creation and composition of music. It is released as Free
Software (MPL-2.0) and runs under Linux.
The real-time sound engine is implemented in C++, the UI runs
in Electronjs, Firefox or Chrome. Assistance with development,
porting or creative efforts is very welcome.
Anklang provides a MIDI sequencer, Undo/Redo capabilities for note
editing, real-time synthesis, and support for CLAP plugins.
The source code and binary packages are available here:
https://github.com/tim-janik/anklang/releases/tag/v0.2.0
The project website with further resources is at:
https://anklang.testbit.eu/
=======================================================================
This release has significant improvements to audio synthesis
capabilities and user interface. The documentation has been improved in
several places, and automated generation was integrated into the CI.
Audio synthesis enhancements include support for new CLAP (draft)
extensions, such as transport information and file references. We now
support non-linear mappings for BlepSynth ADSR times, and a new audio
plugin Freeverb by Jezar at Dreampoint was added, with fixes to the
damping mode in the original version. Additionally, a Jack PCM driver
based on Stefan Westerfelds’ code was integrated.
Performance improvements were achieved by adding a new optimizing memory
allocator and supporting low-latency scheduling via sched_* or RtKit.
Saving projects will now automatically create backups of recent versions.
The user interface has also seen significant improvements in this
release. These include context help via F1 key in various UI components,
fixes to mouse wheel sensitivity for modern browsers, improved tooltips
and note editing in the piano roll. The UI can now be zoomed via new
menu entries, the color palette was updated, and integration of
TypeScript annotations allowed improving the UI JavaScript code quality.
In terms of packaging, AppImage builds saw major compatibility
improvements, and we increased the frequency of Nightly releases,
building from nearly all significant trunk merge commits. We aim to
generate more regular releases in the future instead of having lots of
Nightly builds between regular releases.
--
Anklang Free Software DAW:
https://anklang.testbit.eu/
Hello all,
Release 0.9.2 of the tetraproc and tetrafile processors for first
order Ambisonic microphones is now available at:
<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>
Main changes: tetrafile now defaults to the Ambix format for
both channel gains and order. It is still possible to create
old style 'Fuma' files, see README.
Ciao,
--
FA
Neural Capture is a VST3/LV2 plug to make the process of cloning
external soft/hardware a bit more comfortable.

Neural Capture is build using the DISTRHO Plugin Framework
[DPF](https://github.com/DISTRHO/DPF)
It features a round trip measurement routine.
This allows to load the plug, connect the output to the system output,
loop over external gear (soft or hardware) and back to the Profiler input.
Simply press "Capture" to play the "input.wav" file to the output and
record the returning input
delayed by the measured round trip latency. The peek meter will show you
if your signal is in the expected range.
The round-trip latency will be measured on each "Capture" start.
Resulting recorded "target.wav" file will be perfectly in sync with the
used "input.wav" file.
Currently, both files would be saved under "$(HOME)/profiles/".
The "target.wav" file will be overwritten on each Capture run, so there
will be always only one target file.
You need to save it from that directory in order to use it with the
AIDA-X or NAM trainer.
The "input.wav" file comes as resource with the plug (hence the big size
of the binary packages) and get copied over to that folder, when no
input.wav file was found there.
This allows advanced users to use their own input.wav file by simply
replace the one in that folder.
The target.wav file get checked during record and run to a normalization
function when needed.
(Only when the max peek in target is above the max peek in input).
The record will be saved in the PCM24 wav format (same as the input.wav
file).
The UI provide a progress bar, a peek meter, and well the profile button.
As requested by the AIDA-X and the NAM trainer, only 48kHz been
supported to run this plug.
(You'll get a warning otherwise)
Project side: https://github.com/brummer10/neuralcapture
The release provide binary packages for multiple platforms
release: https://github.com/brummer10/neuralcapture/releases/tag/v0.1.1
regards
hermann