On 4/27/24 5:42 AM, Fons Adriaensen wrote:
On Fri, Apr 26, 2024 at 12:34:03PM -0400, Tim wrote:
Would you have any insight into how this product achieves this and the
techniques used?
It must be a combination of a lot of different things, carefully
tuned for the best results.
Their patent
.
The latency is practically zero. Stunning.
Would you have any insight into how this product achieves this and the
techniques used?
They say, obviously, there is a lot of magic besides plain FFT going on,
such as onset detection and so on.
Thanks
opt.add_auto_option(
'db',
help='Use Berkeley DB (metadata)')
Sure looks like an option !
See if you can access that option via the WAF configuration script.
Tim.
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your software.
Chris
Aww gee whiz, now you're gonna make me add R3 support in MusE ! ;-)
Congratulations, Chris.
RB is an important library for a DAW like ours that has variable tempo
AND handles waves.
Tim.
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e GObject.bind_property().
https://docs.gtk.org/gobject/method.Object.bind_property.html
This can provide one or two way binding between data and UI objects
as long as they both inherit from GObject.
Regards
--
Tim Orford
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On 11/30/21 5:57 AM, Daniel Swärd wrote:
On Mon, 2021-11-29 at 21:25 -0500, Tim wrote:
If I understand the question and unless something changed recently,
the Linuxsampler plugin GUIs have no controls inside them. They are
blank.
You must actually use something like QSampler to control
g like QSampler to control Linuxsampler.
Tim.
MusE Sequencer project.
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On 3/23/21 5:05 AM, Filipe Coelho wrote:
On 23/03/21 00:50, Tim E. Real wrote:
Hi list. Been curious about this question for some time but have not
had time to rig a test for it.
If Jack Transport is rolling, at a frame near the end of its 32-bit
unsigned limit,
what happens when it
?
Tim.
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are you doing in your plugin host (and why) ?
TIA,
A quick search of MusE code finds no usage of it out of
21 matches on "RTLD_".
Only RTLD_NOW and/or RTLD_DEFAULT.
Tim.
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ver an hour now...
Thanks.
Tim.
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me chip it's OK but here, meh...
So in the end, looking at the code given above, I can see I did much
the exact same thing as Ardour, using a fixed coefficient value
driven by time passage (sample rate).
Cheers.
Tim.
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On 4/14/19 5:42 PM, Tim wrote:
> Hi list.
> When I fist boot each day, this is what I get,
> no hi-res timer:
>
> cat /proc/asound/timers
> G0: system timer : 4000.000us (1000 ticks)
> P0-0-0: PCM playback 0-0-0 : SLAVE
> P0-0-1: PCM capture 0-0-1 : SLAVE
> P2-0-1
On 4/7/19 6:48 AM, Fons Adriaensen wrote:
On Sat, Apr 06, 2019 at 12:32:29PM -0400, Tim wrote:
So the software would be able to deal with sines better,
and measure phase differences rather than a pulse edge.
Sure, the pulse will be 'smeared out' a bit, and its actual
position m
On 4/6/19 9:12 AM, Fons Adriaensen wrote:
On Fri, Apr 05, 2019 at 06:38:59PM -0400, Tim wrote:
[PC] -->--[ex. Par. port]-->--[Triggers]-->--[Sin oscs]
--- |
| |
--<--
On 4/5/19 3:16 PM, Tim wrote:
With a discussion about latency in the Jack lists, I thought
I ask a question I've been wondering:
By connecting a cable from a HW output to a HW input, one
can measure the round-trip latency.
But if the purpose is to try to determine the absolute
late
On 4/5/19 3:16 PM, Tim wrote:
But if the purpose is to try to determine the absolute
output latencies of the output port and the input port,
^^
Sorry, remove that word.
T.
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-<--[Delay measurement]---<---
The remaining unknown - the input port latency would be
given by subtracting this measured value from the total
round-trip latency.
Sound crazy?
Thanks.
Tim.
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On 3/3/19 6:56 AM, Will Godfrey wrote:
Just a heads up for anyone likely to come across this.
It now hides some internal elements that were accessible in V2.x so you might
be advised to review your code - it bit us with Yoshimi :(
Yeah, kinda like Fluidsynth 2.0 :(
__
Hi list.
When I put breakpoints in gui or non-rt audio thread code, no problems.
But when I put breakpoints in rt audio thread code, behaviour is odd.
I can continue the program and even have more breakpoints.
But when the program tries to close, it hangs inside jack_deactivate().
I must kill it
most common general usage in our app.
Tim.
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. Thanks.
Tim.
On 11/20/2018 10:14 AM, Thomas Brand wrote:
On Sun, November 18, 2018 09:22, Will Godfrey wrote:
Linux Audio Music has been dormant for a very long time, but recently I
contacted the the person who hosted and ran it.
The reason he closed it was because of a serious vulnerability was
On 11/20/2018 07:42 PM, drew Roberts wrote:
On Tue, Nov 20, 2018 at 4:45 PM Tim <mailto:termt...@rogers.com>> wrote:
On 11/18/2018 03:22 AM, Will Godfrey wrote:
> Linux Audio Music has been dormant for a very long time, but
recently I
> contacted the
music made with
Linux and yet might only offer closed, finished songs.
Who cares what was used to make a closed, finished song?
Who purposely searches for finished songs made with Linux over
songs made with something else?
Tim.
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On 11/01/2018 01:10 AM, Hermann Meyer wrote:
Am 01.11.18 um 04:00 schrieb Tim:
Hi, as I mentioned in LAD today, I think I found some problems
with Blop and SWH LADSPA RDF files.
I fixed them automatically with a program, and by hand
for the few odd incomplete enumerations.
Hi
We&#x
/corrected.tar.gz?dl=0
https://www.dropbox.com/s/tllg1iudino4ah0/FixLrdfFiles.tar.gz?dl=0
Thanks.
Tim.
The MusE sequencer project.
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On 10/31/2018 04:36 PM, Tim wrote:
> I may have found some serious mistakes in the various .rdf files:
>
And in swh-plugins.rdf they are also all like that:
To illustrate that one, here is swh-plugins.rdf 1416 port #1:
Steve Harr
and_1_gain" />
ladspa:hasLabel="band_2_gain" />
ladspa:hasLabel="band_3_gain" />
[ ... ]
--
The blop and SWH are not correct and do not jive with what
the plugin reports. The result is mismatched enums to
rucial thing, the integrity of existing projects.
So we have Caps Amp I, II, III and so on.
But I replied that hey, no worries, it's not so bad.
They're improvements and it's no big deal to work with them.
Tim.
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On 10/29/2018 10:34 AM, Robin Gareus wrote:
On 10/29/2018 06:52 AM, Tim wrote:
On 10/29/2018 12:40 AM, Hermann Meyer wrote:
Downside of cached information is, that it could clash on plugin load
when the plugin have changed it's ports (updated).
If that happens you have a much bigger
On 10/29/2018 12:40 AM, Hermann Meyer wrote:
Hi Tim
On guitarix we wrap the LADSPA/LV2 plugins to our own internal plugin
format and save instructions in json format.
Downside of cached information is, that it could clash on plugin load
when the plugin have changed it's ports (up
all the work.
Even if the file might lack the ability to fill in those ladspa
enumeration value strings when scanning an unidentifiable plugin.
Thanks.
Tim.
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://quitte.de/dsp/caps.html
http://quitte.de/dsp/caps_0.9.26.tar.bz2
Cheers,
Tim
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On 06/26/2018 04:25 PM, Tim wrote:
On 06/26/2018 03:55 PM, Hans Wilmers wrote:
On 06/26/2018 08:32 PM, Spencer Jackson wrote:
I don't know of anyone really working on polyphonic pitch recognition in
the open source world. I think Bayesian filtering of some kind though
would be compe
sed on human hearing,
to make one pair of these speakers simulate a truer surround.
Tim.
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On 06/26/2018 02:16 PM, Ralf Mardorf wrote:
On Mon, 25 Jun 2018 18:31:12 -0400, Tim wrote:
Then I stumbled across this product, MIDI-Guitar from Jam Origins.
For quite some time now, the free version is installed on my iPad, but
I never tested it and meanwhile I've got two electric gu
On 06/26/2018 02:44 AM, Bengt Gördén wrote:
Den 2018-06-26 kl. 00:31, skrev Tim:
I read they use more than just spectral stuff.
Like AI used in speech recognition and so on.
Amazing what DSP audio and image coding can do these days.
Any thoughts on coding techniques? I've read a l
On 06/25/2018 07:08 PM, Paul Davis wrote:
On Mon, Jun 25, 2018 at 6:31 PM, Tim <mailto:termt...@rogers.com>> wrote:
hen I stumbled across this product,
MIDI-Guitar from Jam Origins.
but can it handle negative harmony?
OK I could make a TON of jokes about tha
ese days.
Any thoughts on coding techniques? I've read a lot of papers!
Some say using FFTs + auto-correlation comparisons.
Some say non-negative matrix.
My head spins, but this team definitely deserves praise.
Can open source come up with something?
Cheers, Tim.
_
ds a start menu item for one-click easy setup of several
real time audio settings like this one. No file editing required.
Also you might want to actually install package ubuntu studio.
It installs several audio related packages for audio production.
It saves you the task of manually installing them
On 11/14/2017 08:39 PM, Christopher Arndt wrote:
Am 15.11.2017 um 01:58 schrieb Tim:
I turned on hexter debugging, and added my own to liblo and hexter:
--
initOSC() url:osc.udp://linux-2lbs:19899/
What does 'linux-2lbs' resolve too?
OMG that'
When I run "hexter -test" the gui *does* appear:
Hexter does a few tricks to run in test mode, so I'm not sure
how relevant the second test is (with "hexter -test").
Sorry that should of course have read "hexter_gtk -te
onfigure called with 'DSSI:PROJECT_DIRECTORY' and './'
hexter.so hexter_get_midi_controller called for port 1
hexter.so hexter_get_midi_controller called for port 2
OscIF::oscShowGui(): v:1 visible:0
OscIF::oscShowGui(): No QProcess or process not running. Starting gui.
OscIF
cessing that can't easily
be supported by current plugin architectures, without building
special embedded support into the application, since it requires
pulling variable lengths of input data.
Thanks.
Tim.
>
> On Sat, Jul 9, 2016 at 6:06 PM, Paul Davis
>
> wrote:
> > VAMP doe
ms so that 'run' can pull and push
whatever lengths it needs.
There would be compatibility information on each
stream so that other streams could accommodate.
I thought I read of an LV2 extension or something...
Or am I imagining some
[Fokke de Jong]
>I’m processing 32 sample-blocks at 48KHz but roughly every 0,6
>seconds I get a large spike in cpu usage. This cannot possibly be
>explained by my algorithm, because the load should be pretty stable.
>
>I am measuring cpu load by getting the time with
>clock_gettime(CLOCK_MONOTONI
d asked for the
inclusion of this flag but would later find it ill-conceived and of no
practical use. Could be wrong though, mists of time and all that. In
any event, ignoring the flag altogether seems to be the most pragmatic
solution.)
Cheers, Tim
__
> How does that work for live streams of data?
As you intuit: you sample up, process, then sample back down, ending
up with one output sample for every input sample.
IIrc, http://quitte.de/dsp/caps.html contains at least two oversampled
plugins and comes with source code.
Ch
mponent which most sane
people agree needs to be filtered out. Too much bass going in makes a
clipper sound farty, and lowpass filters taking some edge off pre- and
post-distortion are highly useful for musical purposes.
And welcome to the list,
Tim
_
On April 27, 2015 07:59:36 PM Ralf Mardorf wrote:
> On Fri, 24 Apr 2015 23:13:12 -0400, Tim E. Real wrote:
> >To reduce latency I even tried putting the guitar through a standard
> >time-domain pitch shifter (up one octave) and then into the detector.
> >Not bad, so so.
&
includes the ability to use the PC keyboard
as a music keyboard, hence the map on the right.
[Fons:] Yeah, believe it or not it works best with *dead* guitar strings
and the guitar's tone knob all the way down.
Don't want too many rich guitar string harmonics in there!
Given that, it wo
On April 24, 2015 10:18:57 PM Tim E. Real wrote:
> 6: Now turn the mouse pointer back on. Done.
Ehm, missed on of the best parts:
6: Now return the mouse pointer to where it was when originally clicked.
7: Now turn the mouse pointer back on. Done.
Although, realizing now that when using t
atency than FFT.
I have read about wavelets for a long time.
They are said to be better for this than FFT.
Hard to find real working examples except in commercial.
In PD, there is a wavelet pitch shifter (I think in PD extended)
but it is *broken*
I emailed the list but got no reply excep
far-fetched but, what about showing a sort of temporary
'proxy' circular-motion knob, somewhat larger and fancier/embellished,
in the centre of the screen when a real control is pressed?
Users focus on that proxy control as they are adju
s now report current gain reduction:
http://quitte.de/dsp/caps.html#Compress
http://quitte.de/dsp/caps_0.9.24.tar.bz2
Cheers,
Tim
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Good morning lads,
among the people still using LADSPA, either directly or wrapped into
more fashionable plugin APIs, is anyone actually dependent on
run_adding()?
(Asking, of course, because I intend to drop support for it.)
Thank you,
Tim
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of the filters.
So in this case, with wave files involved, I inform the user that the
best option here is the last one: 'no clock filtering' at all.
It produces a very dense 'verbose' tempo graph with lots of small
jittery changes - BUT - it is the most accurate for playbac
I've been using pipes for message-passing the way Clemens mentioned in
realtime MIDI and audio threads for years without ever experiencing any
scheduling or blocking problems.
Cheers, Tim
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ng back to non-live high-latency mode, we can safely
turn on those plugins again.
So we absolutely need those higher buffer sizes too sometimes.
I've been dreaming of having such a dual 'mode' switch in MusE.
>From lazy to tight at the push of a button.
Possible?
Thanks.
Tim.
oscillation, and that is of great importance for me personally (YMMV
of course).
Cheers,
Tim
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from the bridge.
This might explain why there doesn't seem to be any electric guitars
shown, only bass and electro-acoustic, because we would need
close-to-the-bridge palm muting, although the ElectroAcoustic model
may be able to get a good sound, after all optical sound is completely
mbedded midi board products.
Only to have a user point out that they didn't do 7-bit RPN
conventionally (they used the LOWER 7 bits of data), but the
company owner said there was nothing he could do until
next designs and he actually argued vigorously that his way
was correct (well, he is rig
[hermann meyer]
> However, this list here is the wrong place to discus that any further,
> I just hope that some people here get interested in such a approve
> (Tim you are more then welcome to join us),
> if, please join us at
> guitarix-develo...@lists.sourceforge.net
Thank you
se.
Faust and Guitarix have some nice sounding tubes.
Guitarix seems to have a solid foundation by using Faust.
The Guitarix Head's sound is awesome Hermann.
Have you ever thought of simulating the Head right down to
adjustable power supply (Van Halen's Vari-AC Brown-Sound),
performed worse when using SCHED_DEADLINE for the helper threads
compared to SCHED_FIFO ...
cheers,
tim
>> since recent kernels provide SCHED_DEADLINE, i'm porting my code to make
>> use of it.
>>
>> * is there any plan to migrate jack1 and/or jack2 to use this schedu
h the current API? afaict, the JackThreadInitCallback is
called before JackClient::AcquireSelfRealTime in jack2. (i cannot use
jack1 due to the old bug that jack corrupts the stack while trying to
pre-fault it)
thnx,
tim
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on par with
the best efforts, I think (not my achievement but Dolson's really).
Tim
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llel processing.
Cheers,
Tim
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uations in closed gate state.
http://quitte.de/dsp/caps.html#Download
Upgrading is recommended.
Enjoy,
Tim
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segmentation fault triggered by the
'model' parameter selection of the CabinetIV plugin.
Thanks guys!
http://quitte.de/dsp/caps.html#Download
Upgrading is recommended.
Enjoy,
Tim
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s itself does not restore some settings (stops?).
But can a connected app restore them by sending the whole lot
as midi controller values? Even better, if it's a plugin, the infrastructure
is already there (in MusE) to automatically restore settings, rather than
requiring some extra
commend a FIFO, which you can build with byte-sized counters.
Hope this helps,
Tim
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The new version of CAPS comes -- among other, minor changes -- with
control input smoothening for Eq4p, establishing gentle manners in the
face of even the most drastic parameter changes:
http://quitte.de/dsp/caps.html#Download
Due to an unrelated change, the plugin will even run slightly quicker
[Fons Adriaensen]
>On Sun, Aug 25, 2013 at 12:21:52AM +0200, Tim Goetze wrote:
>
>> I'm planning to evade this problem by crossfading between two parallel
>> static filters. Some phase mismatch issues can probably be expected
>> when the parameter sweep covers a lar
d
when the parameter sweep covers a larger range, but I'm hopeful it
will turn out not to be much of a problem in actual practice, and that
we will find out soon. It really needs a fix.
Tim
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[Fons Adriaensen]
>On Sat, Aug 24, 2013 at 03:14:14PM +0200, Tim Goetze wrote:
>> Eq4p, a four-way parametric equaliser
>> http://quitte.de/dsp/caps.html#Eq4p
[...]
>The F slider switching between its extremes when using the
>mouse wheel is of course a bug in A3. But it
CAPS Audio Plugin Suite 0.9.12
http://quitte.de/dsp/caps.html
New in this release:
Eq4p, a four-way parametric equaliser
http://quitte.de/dsp/caps.html#Eq4p
(Unlike Fons', this one uses parallel processing but lacks control
smoothening other than that provided by a continuous IIR filter
history.)
t sympathy, but also not without
occasionally speculating what the Linux audio landscape might look
like with less lofty and more UNIX-shaped goals.
Tim
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CAPS 0.9.10
===
http://quitte.de/dsp/caps.html
The CAPS Audio Plugin Suite, a LADSPA library comprising classic sound
effects, various signal generators and guitar tone processing, sees
another update containing some bug fixes, minor sound improvements in
various places and a new plugin,
Hi Aurélien,
>- What method of clipping is used will give a "personality" to the
>module: hard clipping, soft clipping, the method used for soft
>clipping, etc...right?
from my experience, there are two main factors determining the
character of waveshaping distortion: one is the hardness of the
c
On June 9, 2013 04:57:21 PM Tim Goetze wrote:
> [Paul Davis]
>
> >On Sun, Jun 9, 2013 at 5:55 AM, Tim Goetze wrote:
> >> Some MIDI devices do not employ 14-bit CCs or the NRPN mechanism.
> >> Access' Virus for example maps all CCs as single 7-bit values, and a
[Paul Davis]
>On Sun, Jun 9, 2013 at 5:55 AM, Tim Goetze wrote:
>> Some MIDI devices do not employ 14-bit CCs or the NRPN mechanism.
>> Access' Virus for example maps all CCs as single 7-bit values, and a
>> forced 14-bit mode will break communication.
>>
>
>
t CCs or the NRPN mechanism.
Access' Virus for example maps all CCs as single 7-bit values, and a
forced 14-bit mode will break communication.
Tim
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On June 8, 2013 11:59:49 AM you wrote:
> On Thu, 2013-06-06 at 20:58 -0400, Tim E. Real wrote:
> [...]
>
> > That regardless, I must make my own full encoding code anyway for
> > our Jack midi driver, because Jack midi delivers separate events.
> > But that's t
On June 6, 2013 05:36:50 PM you wrote:
> On Thu, Jun 6, 2013 at 5:31 PM, Tim E. Real wrote:
>
> Lacking access to the full midi specs document, I don't know
> if this question is addressed. I've looked at manuals for products
> which support them and searched
nding a *single* HW or GUI control to either 14-bit CC
or 14-bit (N)RPN, would *always* send the value LSB, even if
the LSB did not change but the MSB did, when the control moves?
Do the midi specs address this?
Or do you know of examples of such LSB optimizing-ou
[Fons Adriaensen]
>Exactly the same with the form I proposed, w, a, b need to be computed
>just once, not for every gain change. In fact only w depends on the
>sample rate, a and b are fixed constants.
Ah yes, sorry, I see that now.
If that extra operation comes around to bite hard enough, I'l
[Fons Adriaensen]
>On Tue, Mar 19, 2013 at 03:02:19PM +0100, Tim Goetze wrote:
>> Surely you realise this version executes exactly as many additions and
>> multiplications per sample as a biquad?
>
>Yes. In this case it's possible to remove one multiplication
[Fons Adriaensen]
>On Tue, Mar 19, 2013 at 04:26:21AM +0100, Tim Goetze wrote:
>> A 2nd-order IIR filter is often called a "biquad"; at musicdsp, look
>> for that instead.
>
>Not really. A biquad is one way to implement a 2nd order IIR, and
>in many cases related
[Tim Goetze]
>[Harry van Haaren]
>>How is the "rise time" determined here?
>
>As a function of the filter's damping (zeta = 2*Q) and frequency:
>http://en.wikipedia.org/wiki/Rise_time
Sorry, zeta = 1 / (2*Q).
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[Harry van Haaren]
>On Mon, Mar 18, 2013 at 11:45 PM, Fons Adriaensen wrote:
>> A critically damped
>> second order lowpass with a rise time of 30 ms or so will eliminate all
>> audible artefacts. It's very low on CPU and you only need to run it while
>> the gain is changing.
>
>Although I understa
On March 18, 2013 11:58:48 PM Fons Adriaensen wrote:
> On Mon, Mar 18, 2013 at 07:43:32PM -0400, Tim E. Real wrote:
> > Ah, I may have answered my own question when I said:
> > "(One cannot simply wait for the current data value to be 'zero' because
> >
> &g
On March 18, 2013 07:04:52 PM Tim E. Real wrote:
On March 18, 2013 06:47:16 PM you wrote:
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So I'm
On March 18, 2013 06:47:16 PM you wrote:
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So I'm looking at two techniques:
Zero-crossing / ze
ing to sound like where the volume
changes happen at slightly different times, if it will be noticeable, even
though that is far better than 'zipper' noise.
Also I'm trying to imagine how track cross-fading support would deal
with zero-crossing - if it is better to use ramps in
On March 14, 2013 11:42:41 AM you wrote:
> On 03/12/2013 08:08 PM, Tim E. Real wrote:
> > But having said that, yes I'm wondering about a true 'stereo pan' feature.
> > How would such a feature work?
>
> there is no one true stereo pan.
>
> a pan law fo
On March 12, 2013 09:22:39 PM Fons Adriaensen wrote:
> On Tue, Mar 12, 2013 at 04:24:49PM -0400, Tim E. Real wrote:
> > Interesting about the crossover bit.
> > Wow, I considered adding selectable pan laws but didn't realize
> > crossovers.
>
> The rationale behin
On March 12, 2013 04:28:19 PM you wrote:
On Tue, Mar 12, 2013 at 4:24 PM, Tim E. Real wrote:
I will look at having separate pan controls for each channel on one strip,
as I'm reminded from talking to Paul that Ardour has this :)
not anymore ...
Oh wow, haven't tried A3 yet.
On March 12, 2013 09:56:15 AM Fons Adriaensen wrote:
> On Tue, Mar 12, 2013 at 01:23:02AM -0400, Tim E. Real wrote:
> > I noticed our app uses this pan formula:
> > vol_L = volume * (1.0 - pan);
> > vol_R = volume * (1.0 + pan);
> >
> > where volume i
On March 12, 2013 03:13:44 PM you wrote:
On Tue, Mar 12, 2013 at 3:08 PM, Tim E. Real wrote:
lance, but slightly different levels, but not a true 'stereo pan'.
But having said that, yes I'm wondering about a true 'stereo pan' feature.
first, terminology. just as
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