The Echo effect can now have separate left and right channel delay times.
Optimisations giving smaller code size and better efficiency.
Small corrections and updates in the HTML User Guide.
Various bugfixes.
Full details are in /doc/Yoshimi_2.3.2_features.txt
Yoshimi source code is available from either:
https://sourceforge.net/projects/yoshimi
Or:
https://github.com/Yoshimi/yoshimi
--
Will J Godfrey {apparently now an 'elderly'}
https://willgodfrey.bandcamp.com/http://yoshimi.github.io
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Hello, all,
I'm talking an Introduction to Audio Processing course at our local
community college. In our third class, the instructor presented
visually waveforms, using Apple ProTools, and we listened as they
combined.
The examples showed equal frequency waves with a phase delay, showing
cancellation and amplification. We also listened to the beats of waves
at frequencies like 440 and 445 or 450Hz. Finally we listened to pure
sine wave, then the wave with 2nd, 3rd, 4th, 5th, etc. harmonics.
However, we never saw a visual representation of the combined
waveforms. As a former teacher, I thought it would enhance the lessons
to also visualize the waveform.
I think this could be done in Audacity, by mixing the waveforms to a
final track, and viewing it. However, I think this would be somewhat
clumsy. I also think that I could do this in GNU Radio, with mixing
nodes and oscilloscope displays.
Can anyone suggest a tool that would allow visualization of waveforms
based on combined sine waves? My ideal setup would allow sliders to
change the parameters, say, the phase delay, or frequency difference,
in real time, and continuously display the combined waveform.
Any thoughts or guidance is appreciated. Thanks for your advice.
-Kevin
Hi
Today I've pushed support for the Neural Amp Modeler Core module to github.
https://github.com/sdatkinson/NeuralAmpModelerCore
<https://github.com/sdatkinson/NeuralAmpModelerCore>
That means it is now possible to load nam profiles directly in guitarix.
I've tried to implement it as seamless as possible, so I'm now
interested in users feedback before I release and push it to the vst
version.
The module could be find now under the "Distortion" tab.
I like to know if building from source works out of the box or if a
extra call to
|git submodule update --init --recursive|
is needed.
I like to know if model loading works seamless.
I like to know if the resampling works seamless
I like to know if someone is able to crash it.
I like to know if the input/output controls cover the needed ranges.
So, if you are brave and have some time to test it out, go and checkout
the new sources and give it a try.
https://github.com/brummer10/guitarix
<https://github.com/brummer10/guitarix>
Hi,
Running jackd2 on an Edirol UA-25 USB interface as
jackd -t2000 -dalsa -r44100 -p512 -n2 -Xseq -D -Chw:UA25 -Phw:UA25 -i2 -o2
I can not get clients to connect after having resumed the computer from
"systemctl suspend".
Upon resuming, jackd reports
status error: No such device
ALSA: channel flush for playback failed (No such device)
JackAudioDriver::ProcessAsync: read error, stopping...
and hangs.
Trying to use mpv or puredata with this locked server then throw the same messages:
Cannot read socket fd = 13 err = Success
CheckRes error
JackSocketClientChannel read fail
Cannot open mpv client
JackShmReadWritePtr1::~JackShmReadWritePtr1 - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
Nevertheless, qjackctl indicates a running server but throws
Cannot create new client
JackPosixProcessSync::LockedTimedWait error usec = 5000000 err =
Connection timed out
Driver is not running
Cannot create new client
journalctl/dmesg show that the USB sound card is discovered as a new
device after resuming from the suspend state.
Stopping the server takes some time then, throwing:
ALSA: channel flush for playback failed (No such device)
Cannot stop driver
port deleted: UA-25:midi/playback_1
port deleted: UA-25:midi/capture_1
Released audio card Audio2
audio_reservation_finish
11:05:51.251 JACK was stopped
This problem does not exist when using jackd1.
The computers internal soundcard (intel-hda) survives resuming from
suspend with jackd2.
What can I try next?
Thanks!
Peter
This is the first release of Guitarix.vst
Guitarix.vst is the full blown guitarix stack as VST3 plugin for Linux,
using Juce to wrap the guitarix engine into a VST3 plugin.
It allow to load/save your presets, download presets from online and
load external LV2 plugs and IR Files, like the guitarix stand-alone version.
But all that as a VST3 plugin in your DAW. All parameters been exposed
to the DAW, so accessible for automation.
Other than the stand-alone, the VST3 version allows to switch the input
to a real stereo input, so it may match better your channel strip in the
DAW.
For Hdpi users, the GUI is full scalable.
The binary package
[Guitarix.vst3.zip](https://github.com/brummer10/guitarix.vst/releases/downl…
is a x86-64 Linux binary.
To build from source please use the Guitarix.vst3_0.1.tar.gz package, as
only that one contain the needed submodules.
enjoy.
Release Page is
[here](https://github.com/brummer10/guitarix.vst/releases/tag/v0.1)
Project Page is [here](https://github.com/brummer10/guitarix.vst)
If you like to support the guitarix.vst development consider a donation:
[Donate](https://paypal.me/brummer1010)
ImpulseLoader.lv2 is a simple, robust, mono IR-File loader/convolver
plug allowing to browse the File system for IR-Files to load them. As
well supported is drag and drop, when the host support that.
It provide a pop up menu for quick access to all IR-Files in the current
loaded path.
IR-Files will be resampled on the fly to match the session SampleRate.
If the IR-File have more then 1 channel, only the first one will be loaded.
Controls for input gain and Dry/Wet been available. The input gain
didn't affect the dry part of the signal so a mix could be easily created.
Project page is here:
https://github.com/brummer10/ImpulseLoader.lv2
Release page is here:
https://github.com/brummer10/ImpulseLoader.lv2/releases/tag/v0.2
The Release page provide the source package and as well ready to use
binary bundle packages for Linux x86_64 and Windows 64.
ImpulseLoaderStereo.lv2 is a stereo version of this, providing the same
feature set, except that it provide stereo channels. If a IR-File have
only one channel, it will use this on both channels, if it have more
then 2 channels, only the first 2 channels will be used.
Project page is here:
https://github.com/brummer10/ImpulseLoaderStereo.lv2
Release page is here:
https://github.com/brummer10/ImpulseLoaderStereo.lv2/releases/tag/v0.2
The GUI's been made with libxputty:
https://github.com/brummer10/libxputty
the convolution engine is based on zita-convolver and the resampler is
based on zita-resampler:
https://kokkinizita.linuxaudio.org/linuxaudio/
Audio file handling is based on libsndfile:
http://www.mega-nerd.com/libsndfile/
The source code of the GUI been released under the 0BSD license while
the complete plugin itself is under the GPL v2+ license.
regards
hermann
Greetings All,
I just came into possession of a Behringer Arp 2600 clone.
I would like some eye candy while I fiddle my way through learning synthesis.
Can anybody suggest some oscilloscope software?
I've tried a few things already but I'm interested in other's favourite solutions.
Cheers
John
Sent with Proton Mail secure email.
ImpulseLoader is a simple, mono, IR-File loader/convolution LV2 plug.
IR-Files could be loaded via the integrated File Browser, or, when
supported by the host, via drag and drop.
If the IR-File have more then 1 channel, only the first channel will be
used.
IR-Files will be resampled on the fly to match the session Sample Rate.
Binaries been available for Linux and Windows.
Get the latest release here:
https://github.com/brummer10/ImpulseLoader.lv2/releases/tag/v0.1
Project page is here:
https://github.com/brummer10/ImpulseLoader.lv2
regards
hermann
Hello,
Got a new notebook with the following audio device amongst others:
64:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD]
ACP/ACP3X/ACP6x Audio Coprocessor (rev 63)
...
card 1: acp63 [acp63], device 0: DMIC capture dmic-hifi-0 []
It is using, I think, the snd-soc-ps-mach kernel module but it seems
this module does not respect the index option. I'd like to give this
device a higher index so it doesn't get index 0. If I use something like
the following configuration in /etc/modprobe.d/ it doesn't get picked up:
options snd-soc-ps-mach index=12
Anybody any idea? And yeah, I know index is not a valid option for this
module but the ALSA documentation states that all top level snd modules
should respect this option. Or is this Pink Sardine stuff a whole
different animal?
And for the snd-hda-intel kernel module, is it possible to discern
between devices when assigning indexes? As in, the snd-usb-audio module
for instance has vid and pid options to discern between different USB
audio and MIDI devices based on their vendor and product ID's. This is
really handy as it allows you to assign fixed index numbers to these
devices. The model and id options are not suited for this, they only
allow you to actually set the model and/or id. Could very well this is
not possible at all but I just want to be sure.
Talking about USB, going lower than 64 frames/period floods dmesg with
the following messages:
[ 1223.745771] retire_capture_urb: 2362 callbacks suppressed
[ 1223.759483] xhci_hcd 0000:66:00.4: WARN Event TRB for slot 1 ep 5
with no TDs queued?
Is there anything I could do about that or is this the very limit this
system can handle? No problem anyway since I barely run lower than 128
frames/period normally and even at 64 this system is pretty stable with
bigger projects. But I just like to know how low this notebook can go.
It's an AMD Ryzen 7 system so USB controllers are from AMD (USB ID's are
1d6b:0002/0003)
Thanks in advance!
Jeremy
Hello, all,
I've signed up for an Introduction to Audio Production course at my
local community college. I mainly want to learn how to record audio in
the field, and edit it into something like radio news segments or
podcasts (for those too young to have heard of 'radio').
When I was playing around with this 5-10 years ago, the go-to software
was Audacity, which I used and liked a lot. However, I understand that
Audacity has gone through some upheavals.
I'm on a Ubuntu 22.04 desktop, and I experimented at one time with
PipeWire, but I can't remember what audio tools are on my system
currently.
Do I need a Digital Autio Workstation (DAW)? Which one, in the FOSS
world, would you recommend? What audio tools (jack? pipewire?) do I
also need?
I'm sure that the college will have professional tools, in a lab, for
us to use, but I'd like to see if I can duplicate all the assignments
using FOSS tools.
Thanks for any advice or guidance for me.
-Kevin
