[LAD] Re: Jack error

[John Rigg]

On Fri, Aug 04, 2023 at 11:13:46PM +0200, Fons Adriaensen wrote:
> Givven the stutus of BDB, I'd say the long term solution
> would be for jackd to use e.g. LMDB instead of BDB. 

Makes sense.

> And also to make the metadata support a run-time option.

A run-time option would be useful.

John
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[LAD] Re: Calculating logarithmic curve for controller automation

[John Rigg]

On Thu, Feb 23, 2023 at 07:09:51PM +0100, Jeanette C. wrote:
> Hey hey,
> how is a logarithmic curve usually programmed in a DAW or sequencer? Do you
> scale the values of (log1) to log(2) to the desired range and stretch it
> over time? Do you ajudst steepness by either using more less of the log
> function or changing both values like log(20) to log(21)?

If this relates to audio amplitude or frequency you might need
log10(), not log() in C or C++.

John
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[LAD] Re: Status of Pipewire - Ryzen 5

[John Rigg]

On Fri, Feb 10, 2023 at 06:32:36PM +, Will Godfrey wrote:
> On Thu, 9 Feb 2023 20:18:59 +
> John Rigg  wrote:
> >Have you turned off hyperthreading on the Ryzen system (usually called SMT in
> >BIOS settings on AMD)? I keep SMT turned off on my Ryzen systems to avoid
> >possible scheduling problems. So far no problems with 6.x kernels with 
> >dynamic
> >preempt enabled.
> >
> OK. That works thanks. Strange it wasn't needed for kernel 5.10.

The scheduler was modified to allow dynamic preemption (preemption
can be turned on and off with kernel command line), starting with
5.12 kernel.

John
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[LAD] Re: Status of Pipewire - Ryzen 5

[John Rigg]

On Thu, Feb 09, 2023 at 02:33:18PM +, Will Godfrey wrote:
> 
> Something that may (or may not be related)
> There seems to be something odd with Linux image 6.1 preempt
> 
> On a Ryzen 5, Rosegarden keeps randomly losing the transport timer
> sometimes freezing for nearly a second (then blasts poor yoshimi with bunches
> of notes on all 16 channels). This doesn't happen with the 'normal' 6.1 
> kernel,
> nor does it happen with 5.10 preempt.
> 
> However the exact same setup on an older Intel Pentium has no problems at all.

Have you turned off hyperthreading on the Ryzen system (usually called SMT in
BIOS settings on AMD)? I keep SMT turned off on my Ryzen systems to avoid
possible scheduling problems. So far no problems with 6.x kernels with dynamic
preempt enabled.

John
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[LAD] Re: Status of Pipewire

[John Rigg]

On Wed, Feb 08, 2023 at 05:06:45PM +, Rui Nuno Capela wrote:
> for instance, and for crying out loud, pipewire is simply a disaster under a
> PREEMPT_RT kernel, while jack excels with flying colors :)

This is a concern.

I've noticed some pro audio package maintainers are starting to
replace jack dependencies with pipewire-jack (eg. in lsp-plugins
package in Alpine edge). This is quite worrying, considerimg
pipewire doesn't appear to be suitable (yet) for pro audio work.

John
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[LAD] Re: Status of Pipewire

[John Rigg]

On Wed, Feb 08, 2023 at 05:06:45PM +, Rui Nuno Capela wrote:
> for instance, and for crying out loud, pipewire is simply a disaster under a
> PREEMPT_RT kernel, while jack excels with flying colors :)

This is a concern.

I've noticed some pro audio package maintainers are starting to
replace jack dependencies with pipewire-jack (eg. in lsp-plugins
package in Alpine edge). This is quite worrying, considerimg
pipewire doesn't appear to be suitable (yet) for pro audio work.

John
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Re: [LAD] A GUI request

[John Rigg]

On Sun, Jun 05, 2022 at 09:49:13PM +0100, Will Godfrey wrote:
> Therefore please consider either defaulting to a lighter layout, or
> alternatively, at first time start give a choice using a system alert/choice
> window. If you don't provide a lighter option, maybe consider doing so.

Options are good. I'm not too far behind you in age so I empathise.
I actually have the opposite problem and find dark GUIs easier
(especially in a darkened studio or live venue), but they need to have
a good level of contrast. The current trend for ultra-low contrast in
many office applications and web page designs causes me real problems.

Some way of making text larger is a good thing too. I find myself
increasingly using command line programs because it's easier to set
the size of text in the console or xterm than it is in many GUIs.

John
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[LAD] status of RME HDSPe AES (32 channels)

[John Rigg]

Good to know there's a new hdspm driver being worked on.

> The settings of the card can nevertheless be graphically
> set, using alsamixer in the terminal.

I've only had limited success using alsamixer to change settings
on HDSPe MADI cards. I've found amixer and alsactl (both CLI)
more useful here. I made a web page describing their use with the
HDSPe MADI:

http://www.jrigg.co.uk/linuxaudio/hdspe-madi.html

Much of it should also be applicable to the HDSPe AES32 (allowing
for different parameter names and channel numbers).

John
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[LAD] NSM fork

[John Rigg]

The recent fork of NSM (Non Session Manager) and the discussion
around it has highlighted some issues in the Linux Audio
community which I believe need to be addressed.

I don't think the fork was handled very well, with antagonism
on both sides which could easily have been avoided with a
little forethought.

The choice of the name New Session Manager and the re-use of
the NSM initials is an obvious problem IMO. The original
wording on the GitHub README.md, which implied that Non
Session Manager was non-free and contained ads and spyware,
didn't help. (It was changed after I raised an issue about
it).

Now Jonathan Liles has been banned for life from LAD after
an angry email to the list. The email contained ad-hominem
comments, so a block on the post was not unreasonable, but
a life ban for one angry post is certainly unreasonable.

The fact that the list moderator is also one of the team
which forked NSM does not look good. It is a clear conflict
of interest.

At the very least, Jonathan Liles' list membership should
be reinstated. 

I've been using Linux for pro audio work for over 15 years
now, and much of the community spirit that existed when I
started seems to have been replaced with an intolerant
mindset that expects everyone to conform to its view of how
things should be. This doesn't bode well for the future of
Linux audio.

John 
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Re: [LAD] READ THIS IF YOU CARE ABOUT FREEDOM! Fwd: [LAA] Non DAW release including Non Session Manager (i.e. the real NSM)

[John Rigg]

On Fri, Jan 29, 2021 at 05:46:11PM +0100, Thomas Brand wrote:
> So that's why I ask you David Runge to undo the ban. The OP is not the core
> of the problem. By removing OP, you remove symptoms only.

I agree. Please undo the ban.

John
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[LAD] NSM non vs new

[John Rigg]

On Sun, Sep 06, 2020 at 05:46:55AM +0200, Hermann Meyer wrote:
> Including NSM 

As a long term user of non-session-manager I think the use of
the same abbreviation (NSM) for the new fork is confusing.
Would it not be better to refer to it by something different,
eg. NSM2?

John
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Re: [LAD] Looking for some jackd debugging help

[John Rigg]

On Sat, Mar 28, 2020 at 12:38:46PM -0700, Ethan Funk wrote:
> I would like to run jackd in gdb so I can see where it is failing, but
> I am unsure how to build it without breaking my system's installed jack
> package, which is of course built without debugging enabled.

You could try installing the compiled version over the existing
package files. On Debian I add the following to the ./waf configure
command when I want to do this:

--prefix=/usr --libdir=/usr/lib/x86_64-linux-gnu

Substitute the libdir path relevant for your distro
if not Debian based.

John
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Re: [LAD] 'A' note Tuning Range

[John Rigg]

On Thu, Apr 11, 2019 at 12:17:58PM +0200, li...@justmail.de wrote:
> On Thu, 2019-04-11 at 08:37 +0100, Will J Godfrey wrote:
> > On Thu, 11 Apr 2019 08:16:29 +
> > John Rigg  wrote:
> > > A Korg GA-1 tuner can go down to 5 semitones flat. It's quite common
> > > in the heavier styles of rock music to downtune a few semitones.
> > Interesting. Thanks for that.
> 
> Assuming the guitar tuner is a chromatic tuner, dropped and lowered
> guitar tunings don't require anything else than A = 440 Hz and if you
> dislike 440Hz a range from + half of a semitone (+50 cent) to - half of
> a semitone (-50 Cent).
> 
> https://en.wikipedia.org/wiki/List_of_guitar_tunings#Dropped
> https://en.wikipedia.org/wiki/List_of_guitar_tunings#Lowered

That's all very well, but tuning quickly on stage in a live gig
is lot easier if your tuner goes down to the right pitch with
minimal fuss. (Speaking from long experience as gigging guitarist
and bassist).

The GA-1 tuner I mentioned isn't a true chromatic tuner, but
its ability to shift the standard guitar tunings down several
semitones is very useful. In modern metal genres C or B tunings
are probably more common than the standard EADGBE, so this isn't
just an edge case.

John
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Re: [LAD] 'A' note Tuning Range

[John Rigg]

On Tue, Apr 09, 2019 at 09:21:09PM +0100, Will Godfrey wrote:
> Currently in 'Scales' Yoshimi can set this anywhere between 1Hz and 2kHz, 
> which
> is frankly ridiculous.
> 
> This doesn't appear at all in the Scala documentation, so that's no guide.
> 
> I've had suggestions ranging from +- 1/2 semitone to +- half octave as being
> more than enough, considering that there is also semitone master key shift
> covering +- 3 octaves (used to be 5!) along with a fine detune of +63 -64
> cents.

A Korg GA-1 tuner can go down to 5 semitones flat. It's quite common
in the heavier styles of rock music to downtune a few semitones.

John
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Re: [LAD] New(ish) OSS synth plugin

[John Rigg]

On Wed, Sep 26, 2018 at 02:02:08PM -0400, bill-auger wrote:
> derp, the first time i noticed this post i overlooked the egregious
> error in the title
> 
> one should be mindful to avoid terminology (especially acronyms) that
> have multiple interpretations, especially when those alternate
> interpretations can relate to the same subject area - that really
> should not happen but it does when people give names to new things
> without knowing that term is already widely used for something else
> 
> in this case i am clearly referring to 'OSS' - in the context of
> pro-audio, OSS is the "Open Sound System"; and it has been widely known
> as such for about 20 years, and still in use today - that is how i read
> this post at first - now i realize it was a short-hand for
> "open-source"

I too thought it meant Open Sound System at first.

John
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[LAD] Do professionals use Pulse Audio? ; xfce4-mixer

[John Rigg]

On Wed Apr 25 18:27:25 CEST 2018 Will J Godfrey wrote:
> I use one machine specifically for music, permanently connected to Keyboards
> etc. and PA was removed with extreme prejudice. On the audio side it's working
> entirely Jack - MIDI is mostly ALSA.
> 
> On my other 'office' machine it's there and I don't pay any attention to it.

I also use dedicated machines for my DAW systems, with no PA installed. All
audio is working with jack.

My DAWs seldom connect to the internet so I don't have a heavyweight web
browser (or many other desktop-type programs for that matter) installed.

I actually dislike modern desktop Linux so much I switched my 'office'
system to OpenBSD some time ago. That uses sndio (much simpler than PA)
for browser audio.

John
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Re: [LAD] Half-OT: Fader mapping - was - Ardour MIDI tracer

[John Rigg]

Ralf, please stop dictating how others should work. I've been recording in
studios for 35 years and I like to think I know what I'm doing. Not everyone
who has different working methods from yours is an idiot, but that seems to
be what you are implying.

John
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Re: [LAD] Half-OT: Fader mapping - was - Ardour MIDI tracer

[John Rigg]

On Fri, Aug 22, 2014 at 01:10:58AM +0200, Ralf Mardorf wrote:
 On Thu, 2014-08-21 at 20:06 +0100, John Rigg wrote:
  The P+G faders (generally regarded as the best)
 
 Actually you get fader units for Studer with PG, but also with Alps
 faders. Such a module usually costs more than a complete home recording
 mixer.

True, but how much more does it cost to emulate a P+G fader in software
compared with the cheaper ones?

John
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Re: [LAD] Half-OT: Fader mapping - was - Ardour MIDI tracer

[John Rigg]

On Thu, Aug 21, 2014 at 12:39:05PM +0200, Ralf Mardorf wrote:
 IMO each channel by default
 should provide your (Fons') parametric EQ and post fader aux sends.

Those who usually try to record the right sound in the first place might find
this a nuisance. Unfortunately one size doesn't fit all.

 If
 people start mixer automation for the fist time, a notification should
 pop up and mention, that:
 
  Fades are rare in music mixing

Some might find that patronising. I personally mix a lot of recordings with
fade outs, even if it's just to fade the end of the last note or a long reverb
tail.

Regarding fader mappings, faders on analogue mixers vary a lot in their
characteristics. Control panel markings are rarely accurate and don't
necessarily provide a reliable basis for software emulation (unlike actual
measurements).

The '70s Neve mentioned earlier would likely have used Penny  Giles faders
with a roughly logarithmic taper (no VCAs). Since this type of fader is made
by varying the characteristics of the resistive track along its length it's
very tricky to achieve consistency. A logarithmic fader is unlikely to match
one from a different manufacturer, which makes replacement difficult if the
original type is obsolete - often all the channel faders need to be replaced
at once if this is the case.

One thing most analogue faders do have is better resolution than a 128 step
midi controller, so slow fades without audible steps are easier to achieve.

John
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Re: [LAD] Half-OT: Fader mapping - was - Ardour MIDI tracer

[John Rigg]

On Thu, Aug 21, 2014 at 09:01:03AM -0700, Len Ovens wrote:
 On Thu, 21 Aug 2014, John Rigg wrote:
 The '70s Neve mentioned earlier would likely have used Penny  Giles faders
 with a roughly logarithmic taper (no VCAs). Since this type of fader is made
 
 Ok, I have to ask this somewhere, it may as well be here :) Which
 log?

The P+G faders (generally regarded as the best) are close to log10
characteristic, ie. -20dB at about half travel. There's a PDF data sheet
here:

http://www.pennyandgiles.com/Products/Audio-Faders-Video-Controllers/Linear-Manual-Fader-PGF8000.aspx

The Panel Graduations/Slots diagrams on page 3 show the taper characteristic.

John
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Re: [LAD] Beta testers required...

[John Rigg]

On Thu, Jul 24, 2014 at 08:42:48AM +, Fons Adriaensen wrote:
 You need resampling even if the sample rates are equal, unless
 the interconnected system have a common word clock.

Can the resampling be switched off in cases where a common word
clock is available?

John
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Re: [LAD] LV2 plugin for all applications

[John Rigg]

On Fri, Jun 06, 2014 at 09:31:26AM +, Fons Adriaensen wrote:
 The whole user space part of ALSA has become more or less redundant.
 For 'serious' work, people will use Jack which provides the ultimate
 flexibility.

Except for those of still in the multiple PCI-card stone age who need
pcm_multi ;-)

John
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Re: [LAD] LV2 plugin for all applications

[John Rigg]

On Fri, Jun 06, 2014 at 10:42:09AM +, John Rigg wrote:
 On Fri, Jun 06, 2014 at 09:31:26AM +, Fons Adriaensen wrote:
  The whole user space part of ALSA has become more or less redundant.
  For 'serious' work, people will use Jack which provides the ultimate
  flexibility.
 
 Except for those of still in the multiple PCI-card stone age who need
 pcm_multi ;-)

Sorry, typo. Should be:

Except for those of us still in the multiple PCI-card stone age who need
pcm_multi ;-)

John
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Re: [LAD] Releasing source code is not enough, I think...

[John Rigg]

On Tue, Jan 21, 2014 at 12:40:23PM +, Fons Adriaensen wrote:
 On Tue, Jan 21, 2014 at 05:55:04AM +, Filipe Coelho wrote:
 
  I think we should stop assuming releasing source code is enough.
 
 Enough for what ? Users who don't want to install from source
 want packages made for the package manager of their distro,
 which will take care of dependencies etc. You can't expcect a
 developer to provide such packages for each and every distro.
 I don't even provide them for the distro I use myself.

I'm inclined to agree with Fons here. There seems to be a growing culture
of expecting Windows-style hand-holding for free software. In the Windows
(and Mac) world you pay money for this. I think it's unreasonable to
expect the same level of support from unpaid developers. (If they have the
time to do it that's great, but it shouldn't be taken for granted).

John
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Re: [LAD] Releasing source code is not enough, I think...

[John Rigg]

On Tue, Jan 21, 2014 at 03:34:05PM +, Filipe Coelho wrote:
 I think it's not up to the users to understand how software
 compilation works.

That depends on the developer's intention when releasing source
code. Some very good Linux audio software was written for the
developers' own use, and the source code only released as a
favour to those who might also have a use for it. Some of the
LADSPA and LV2 plugins I use most fall into this category.

If a piece of free software isn't available as a distro package,
I think it's very much up to the user to find out how to compile
it if the developer doesn't have time to offer unpaid support.

John
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Re: [LAD] Audio Levitation

[John Rigg]

Would it not be better to continue this discussion off-list?

John
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Re: [LAD] who wants contribute to Advanced Gtk+ Sequencer

[John Rigg]

On Tue, Sep 24, 2013 at 06:40:31PM +0300, Vytautas Jancauskas wrote:
 sourceforge? what is this 1999?

Top-posting? What is this, a non-developer list? ;-)

John
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Re: [LAD] who wants contribute to Advanced Gtk+ Sequencer

[John Rigg]

On Tue, Sep 24, 2013 at 05:32:57PM +0200, Ralf Mardorf wrote:
 On Tue, 2013-09-24 at 17:01 +0200, Joël Krähemann wrote:
  http://sourceforge.net/projects/ags/
 
 ags_0.3.15-0 doesn't build.
 
 $ make
 gcc -g -c main.c -o main.o -I../ `pkg-config --cflags alsa glib-2.0 
 gobject-2.0 gdk-2.0 gdk-pixbuf-xlib-2.0 gmodule-2.0 gtk+-2.0 libxml-2.0 
 sndfile libinstpatch-1.0`
 main.c:21:35: fatal error: ags/audio/ags_channel.h: No such file or directory
  #include ags/audio/ags_channel.h
^
 compilation terminated.
 make: *** [main.o] Error 1
 $ ls
 audio  Documentation  file  lib  license  main.c  main.h  Makefile  object  
 README  widget  X

Same result here. The file's there in the audio directory but it's looking in 
ags/audio. Makefile adjustment required?

John
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Re: [LAD] forking (was Re: Aeolus)

[John Rigg]

On Sat, Sep 21, 2013 at 07:52:02AM +0200, Ralf Mardorf wrote:
 It's completely impossible to be on Fons site. When Fons has such a
 super-mind, why did he chose the GPL? Those simple-minded guy who forked
 Aeolus might have made a little mistake, but doesn't offend the licence.

Ralf, please stop this straw man argument about the GPL. Fons is understandably
irritated by someone forking his project without prior discussion. It's a
question of etiquette, not licence terms, as has been pointed out repeatedly
in this thread. 

It isn't difficult to find out that Aeolus is currently maintained; all it
takes is a look at the README in the sources, which contains release dates.
The 2007 copyright date which I think was mentioned earlier (I'm not going to
re-read this whole sorry thread to confirm that) is the copyright date for
GPL 3, not Aeolus. Again, this stuff is not difficult to check.

Another point: copyright and licence are separate things. It is possible to
violate a copyright while still complying with the GPL. Adding one's own
copyright notice to someone else's original work without making substantive
changes to the work may be such a violation (IANAL, so that's speculation).

John
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Re: [LAD] forking (was Re: Aeolus)

[John Rigg]

On Sat, Sep 21, 2013 at 11:52:05AM +0200, Ralf Mardorf wrote:
 He made a mistake? Ok! And now? Making it a drama while it might be easy
 to solve, by just talking to him? He might think he didn't made a
 mistake, so he perhaps won't contact somebody, but he perhaps will reply
 if Fons or you send a request.

There is one person turning this into a major drama, and it isn't Fons.

The situation could have been avoided simply by reading the README and acting
accordingly.

John
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Re: [LAD] Community Interaction and Working Together

[John Rigg]

On Fri, Sep 20, 2013 at 02:26:28PM +0200, Ralf Mardorf wrote:
 When reading musician forums:
 Seemingly many people are missing fashion software, such as auto-tune

Last time I needed to auto-tune a vocal track I used zita-at1.
It worked very well.

John
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Re: [LAD] zita1 RIP

[John Rigg]

On Wed, Sep 04, 2013 at 07:09:56PM +, Fons Adriaensen wrote:
 The new zita1 will probably be a Fujitsu P510, one of the 
 reasons being that this is one of the few still having at
 least one PCI slot, so I can still use my sound cards. Any
 suggestions for alternatives are welcome !

If you wanted to go the AMD route, ASUS are still making some
motherboards with 2 or more PCI slots. I'm still using Athlon64x2
CPUs on Asus M3A78-PRO boards (with ECC RAM) in my DAWs, but last
time I checked there were still current boards with 3 PCI slots.
Boards with 2xPCI are still fairly common.

In general I prefer ECC memory, high efficiency CPUs and low-power
graphics hardware for audio work. Keeping the power consumption down
allows fan noise to be minimised. My location DAW can run fanless.
I also prefer to run efficient software to keep CPU load (and heat)
down. 

Thanks for writing such efficient software BTW :-)

John
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Re: [LAD] Behringer ADA8000 phase

[John Rigg]

On Mon, Jul 01, 2013 at 09:29:53PM +, Fons Adriaensen wrote:
 The only thing that has happened to the installation over
 the last years is that some of the Behringers failed (power
 supply blown up, one per year on average) and were replaced.
 So I checked those separately. And yes, some of them had their
 output phase inverted w.r.t. the others. Apparently the thing
 exists in two versions, but apart from measuring there's no
 way to tell which is which. So I'll have to recheck things
 each time any of them are replaced again. Thank $GOD we didn't
 use those for the WFS system.

This is appalling, but I'm aware it happens sometimes. I've
observed a similar thing with Chinese condenser microphones,
including some relatively expensive ones. Now I always check
polarity against a known good mic whenever I buy a new one.
Looks like I'll also have to start checking other gear.
Thanks for the warning.

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Re: [LAD] A3 clicks

[John Rigg]

On Tue, Jun 18, 2013 at 06:28:39PM +0200, Nick Copeland wrote:
 On Tue, Jun 18, 2013 at 08:24:29AM -0400, Paul Davis wrote:
 
 
 
  inevitable, since you are changing the order of processors in the channel
 
  strip. in some setups, you will notice the click as this happens, in others
 
  you will not.
 
 
 
 This makes A3 unusable for live work. For the simple reason that nobody 
 expects
 
 such a thing to happen, not any more than e.g. using a PFL should cause 
 clicks.
 
 I will refrain from mentioning the not-so-kind adjectives that would be used
 
 to describe a HW mixer doing such a thing.
 
  add it to the list of several hundred other items that all make A3 unusable 
  for this or that. 
  Fons - could you outline a few of the needs to relocate such taps during a 
 live performance?I don't have much knowledge either way but you always have 
 context behind your remarks andthat might help to put the issue into 
 perspective.
 Having those taps are useful for reviewing signals levels, looking for 
 sources of noise/hum, butI would probably prefer to select those taps during 
 soundcheck rather then when actually live, bythat time it might already be 
 too late, but hey - you have a _lot_ more experience here.
 Regards, nick.  

Not answering for Fons, but I've done a lot of live sound mixing over the years.
In live mixing you often need to meter at different points in the signal chain 
to
check levels, as they are constantly changing, often unpredictably. Nothing ever
stays as it was in a sound check. (I personally wouldn't use a DAW running
on commodity computer hardware for live mixing, but that's another matter ;-)

John
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Re: [LAD] (Modular) Synth and Clipping

[John Rigg]

On Sat, Jun 15, 2013 at 11:33:45PM +0200, Dominique Michel wrote:
  An output transformer will saturate if the frequency is low enough,
  but the signal level required to saturate it is directly proportional
  to frequency. In a properly designed guitar or bass amp there will be
  some transformer distortion at the lowest frequencies but not much
  above that. If you lowered the frequency enough to fully saturate the
  transformer it wouldn't sound very good, as you say. (I design guitar
  amps among other things).
 
 Me too, and I repair them too. I was talking here about cheap power
 transformers used in some brands of commercial guitar amplifiers, not
 about their output transformers. The main frequency is low enough to
 easily saturate them when they are not properly dimensioned, and this
 saturation will go through everything to the speaker.

Power transformer saturation only occurs if the voltage applied to the
primary is too high. It is not affected directly by the load on the
transformer.

 A typical example are the old Peavey Mace, good transistor preamp and
 driver stage, 6x6L6 for the output, but a too small power transformer to
 drive such a power (160 w RMS), and a bias circuit for the power stage
 that kill the dynamic when it is in saturation. The power transformer is
 definitely too small to drive the tubes at full saturated volume. I
 measured such an amp, the maximum power is the same with a clean sound
 and at full saturation. The sound is very good when the power stage is
 not saturated, but very bad when the power stage is saturated, that
 not only because of the lack of dynamic, but also because of the
 saturation of the power transformer.

The effect you are describing is due to the internal resistance of the
transformer windings and other power supply components, not transformer
saturation. When more current is drawn the supply voltage drops due to
resistive losses. If there's a tube rectifier the effect will be more
pronounced. Some people like that effect but not me. I agree that power
transformers in many commercial designs are undersized.

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Re: [LAD] (Modular) Synth and Clipping

[John Rigg]

On Sat, Jun 15, 2013 at 02:56:13PM +0100, Aurélien Leblond wrote:
 - in some cases (or let say modules of a synth), clipping is
 implemented more to copie what an analogue system would do than a
 mandatory part of the algorithm... Let's take an example: 2 sin waves
 mixed together of amplitude -1/1 will just have an amplitude of -2/2
 (as long as they are in phase)... A digital mixer without clipping
 would be able to cope with that, but an analogue one wouldn't... and
 that's why the analogue system would clip the signal..right?

An analogue system with enough headroom wouldn't clip. A DAC will
clip in those circumstances.

 - 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?

You'd also need to avoid generating harmonics above fs/2 to prevent
aliasing, unless that's part of the personality you're going for.

 - Hard clipping is something of the digital world - it doesn't exist
 in the analogue world... right?

Wrong. An analogue amplifier with lots of negative feedback, eg. a
typical op amp or power amp, will usually hard clip.

 - Soft clipping will deform any waves of amplitude -1/1 even if it
 doesn't exceed the accepted threshold, because just before reaching
 the threshold  the algorithm will take over and softly make the signal
 reach the maximum amplitude and keep it there until the original
 signal goes back under a set threshold.right?

Yes, although clipping threshold can be any level you like (within
reason) in floating point.

 - Is there a preferred stage for clipping? In the case of a filter,
 should we clip before filtering, after or both? Or are all these
 options valid and that's what will give an additional personality to
 the filter?

That depends on whether you want to modify the frequency response
before or after the clipping harmonics are added. Both can be valid
options.

John
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Re: [LAD] (Modular) Synth and Clipping

[John Rigg]

On Sat, Jun 15, 2013 at 08:50:39PM +0200, Dominique Michel wrote:
 As example, when you push guitar amps in clipping at full volume, half
 of the clipping you can ear is, with some brands, not the clipping of
 the electronic, but the clipping of the power transformer. That sounds
 very bad -:(, and that imply you may have to change often this
 transformer when such amps are used to play blues or rock like styles
 of music. 

An output transformer will saturate if the frequency is low enough, but
the signal level required to saturate it is directly proportional to
frequency. In a properly designed guitar or bass amp there will be some
transformer distortion at the lowest frequencies but not much above that.
If you lowered the frequency enough to fully saturate the transformer it
wouldn't sound very good, as you say. (I design guitar amps among other
things).

John
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Re: [LAD] Xiph.org - Video:Digital Show and Tell - No difference between analog and digitally processed sound.

[John Rigg]

On Thu, May 23, 2013 at 12:18:25PM +, Fons Adriaensen wrote:
 Take a filter for a 48 kHz DAC. It could be -0.5 dB at 23 kHz,
 -12 dB at 24 kHz, and -100 dB at 25 kHz. Any aliasing will be
 either above 23 kHz or below -100 dB, probably harmless.
 Given the passband and stopband constraints at 23 and 25 kHz,
 the actual value at 24 kHz is more or less irrelevant.
 
 The potential intermodulation effects referred to in the paper
 by Julian Dunn are real, but not realistic. If such signals 
 (high energy well above 20 kHz) are present you'll have some
 serious problems even in a completely analog system.

True, although I'd have more confidence in a filter that eliminated
the possibility entirely.

I couldn't resist responding to the statement in the thread title:
No difference between analog and digitally processed sound.
As in so many things, that depends.

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Re: [LAD] Xiph.org - Video:Digital Show and Tell - No difference between analog and digitally processed sound.

[John Rigg]

On Wed, May 22, 2013 at 10:38:37AM -0400, Bill Gribble wrote:
 There are real effects due to clock jitter on
 both the A/D and D/A end that can cause small but measurable
 distortions.

Not to mention audible if it's severe enough. Decimation filters
that only give 6 or 12dB attenuation at fs/2 (typical in many pro
audio ADC chips) can allow audible aliasing too. I wouldn't expect
an oscilloscope to have enough resolution to detect these effects,
but a good spectrum analyser and/or a good pair of ears often can.

John
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Re: [LAD] RME suggestions

[John Rigg]

On Sat, May 18, 2013 at 08:27:14AM +1000, Geoff Beasley wrote:
 I will be getting a new audio rig in July. I would appreciate
 comments from anyone who uses a RME system. I'm looking into a
 raydat and some converters... and what D?A converters do you use?

I used a Solid State Logic Alpha-Link with my RME card when I had
one. I was using an Alpha-Link SX (24 analogue I/O, AES3 plus MADI)
with an HDSPe MADI card, but the Alpha-Link AX has 24 channels of
ADAT I/O plus MADI so it might be worth a look.

The SSL unit sounded good and was reliable. I also liked the fact
that it had so many channels in a 2U box and had no cooling fan.
(Despite the lack of fan it ran a lot cooler than something like
an Avid/Digidesign 192 I/O).

John
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Re: [LAD] Mixing audio: Implementing pan and balance

[John Rigg]

On Sat, Mar 16, 2013 at 03:47:12PM -0400, Ricardus Vincente wrote:
 On 03/16/2013 09:25 AM, John Rigg wrote:
 
  A lot of mixing consoles don't provide a mono switch, but it's usually
  possible to work around it with sub groups. I still have to work out a
  convenient method for mono checking in Ardour 3.
 
  Doesn't A3 have a mono button on the new monitor section?

Yes it does. I had (stupidly) switched off the monitor section and
forgotten about it. Thanks for the reminder.

It appears to be impossible to add the monitor section to sessions that
were created without it, so it needs to be turned on in the config
before session creation IIUC.

John
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Re: [LAD] Mixing audio: Implementing pan and balance

[John Rigg]

On Sun, Mar 17, 2013 at 06:54:18AM -0400, Paul Davis wrote:
 On Sun, Mar 17, 2013 at 5:17 AM, John Rigg lad...@jrigg.co.uk wrote:
 
  On Sat, Mar 16, 2013 at 03:47:12PM -0400, Ricardus Vincente wrote:
Doesn't A3 have a mono button on the new monitor section?
 
  Yes it does. I had (stupidly) switched off the monitor section and
  forgotten about it. Thanks for the reminder.
 
  It appears to be impossible to add the monitor section to sessions that
  were created without it, so it needs to be turned on in the config
  before session creation IIUC.
 
 
 not so. Session - Properties - Monitoring.
 
 you can add it and remove it to any given session.

Great, thanks. Another new A3 menu item discovered :-)

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Re: [LAD] Mixing audio: Implementing pan and balance

[John Rigg]

On Sat, Mar 16, 2013 at 06:23:09PM +, Fons Adriaensen wrote:
 Zita-mu1 is a simple Jack client used to organise stereo monitoring 
 during recording and mixing. More here: 
 http://kokkinizita.linuxaudio.org/linuxaudio/zita-mu1-doc/quickguide.html
 and download from
 http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-mu1-0.2.0.tar.bz2

Very useful, thanks.

John
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Re: [LAD] Mixing audio: Implementing pan and balance

[John Rigg]

Nobody has mentioned it yet but there is a good reason why it's
useful to have pan controls on each channel of a stereo bus: it
makes it easy to check for phase errors by panning both sides
to the middle to check in mono. I do this in Ardour 2 (and on hardware
mixing consoles) all the time. The end and middle detents on A2's
panners make this quick and easy.

John
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Re: [LAD] Mixing audio: Implementing pan and balance

[John Rigg]

On Sat, Mar 16, 2013 at 11:16:46AM +0100, Ralf Mardorf wrote:
 On Sat, 2013-03-16 at 09:50 +, John Rigg wrote:
  ... check for phase errors ... on hardware mixing consoles ...
 
 I'm doing it too. Mono compatibility seems to be less important for the
 Linux community.

There's possibly a lack of awareness. Anyone who mixes sound as a job
(and wants to keep it) knows it's important to check in mono. Even if
it will only be played back in stereo a phase reversal on one side will
cancel low frequency centre-panned signals when listening between the
speakers.

Judging by the comments I hear from mastering engineers about dimwits
putting a phase reversed bass or kick drum in the mix and not noticing,
it isn't just a problem in the Linux audio community.

 My hardware mixing console doesn't provide a mono switch and since I
 won't mis-adjust the pan pots I rout it mono by subgroups.

A lot of mixing consoles don't provide a mono switch, but it's usually
possible to work around it with sub groups. I still have to work out a
convenient method for mono checking in Ardour 3.

John
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Re: [LAD] Muse - Was: OT: Bitwig beta for Linux reviewed

[John Rigg]

On Fri, Mar 08, 2013 at 11:47:52PM +0100, Ralf Mardorf wrote:
 The CPU always was and still is an AMD Athlon 64-bit dual-core BE-2350
 2.1 GHz. 
... 
 I'm usually using self-build kernel-rt in the past 2.6.x and today 3.x.

FWIW I've never been able to get an -rt patched kernel to work with
a dual core CPU without xruns and lockups. I've had better results
with standard kernels with pre-emption enabled. I don't use MIDI and I
haven't tried recent -rt patches though, so treat that observation
accordingly.

I think I'd be more likely to blame things like nVidia graphics drivers
and non-standard kernels before making a sweeping criticism of Linux
sequencing software, but maybe that's just me.

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Sat, Feb 09, 2013 at 12:03:41PM +0300, Louigi Verona wrote:
 On Sat, Feb 9, 2013 at 2:29 AM, John Rigg lad...@jrigg.co.uk wrote:
  To be fair I wasn't really slagging off Windows and Mac users. Most pro
  audio
  engineers are using those after all. I'm just bemused by the attitude that
  audio processing tools should be anything more than that. Pretty pictures
  and dumbed down control ranges don't help me make better mixes, they just
  get in my way.
 
 But why instantly dumbed down? Or are the generic LADSPA controls so
 intellectual?
 I think beautifully down interface adds to the inspiration as opposed to
 stuff
 that all looks like coding examples.

By dumbed down I mean restricted in a way which may result in
inexperienced users making fewer mistakes, but will also inconvenience
more advanced users. An example of this would be a high mid EQ that won't
sweep above 8kHz. What if I need to EQ 12kHz? There's some excuse for this
kind of thing on analogue hardware, as component cost has to be kept down,
but in a plugin it's totally unnecessary. 

Another pet peeve is lack of a text entry field on controls, as it makes
it difficult to set a parameter to an exact amount. Even worse are detented
controls. What if I need an intermediate setting? The reason for detents on
analogue hardware is for repeatability of settings, but it's totally
redundant in software, unless the developer has neglected to provide
text entry!

I will stress that I'm talking about audio engineering tools, not music
creation software here. I do appreciate that users of the latter have very
different requirements.

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Sat, Feb 09, 2013 at 02:20:58PM +0300, Louigi Verona wrote:
 Right, I hear you John. But I did look at pro mastering software on
 Windows, I don't
 remember any unnecessary restrictions.
 My message would that I oppose this sort of a sweeping judgment of a whole
 audio platform.
 There might be some concrete examples, sure, but if you mean that in
 general all or most
 software on Windows is restricted in such a manner, I personally would want
 some proof.

There's a misunderstanding here. I'm criticising plugin design choices which
make my job difficult, regardless of platform. That includes Linux.

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Fri, Feb 08, 2013 at 01:02:13AM +, Fons Adriaensen wrote:
 But do you really think that when doing a mix, the quality of the
 final result will depend on which of the 15 or so general purpose
 equalisers you use on any particular track ? 

No, it depends on which esoteric piece of hardware the pretty picture
on the GUI looks like, of course.

 The result will depend only on your skill in using any one of
 those EQs.

Heresy!

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Fri, Feb 08, 2013 at 09:55:14PM +0300, Louigi Verona wrote:
 By the way, to once again put up a little defense of people on Windows and
 Mac, I advice everyone to watch a masterclass with Ritchie Hawtin, a
 popular minimal house producer and dj. It is available on YouTube and he
 speaks about how he is using a modular software and hardware setup by using
 MIDI and OSC to create a complicated audio and video setup. It is amazing
 and it shows that many people on Mac and Win are experimenting and even
 writing their own software for themselves, like Richie Hawtin and his team.

To be fair I wasn't really slagging off Windows and Mac users. Most pro audio
engineers are using those after all. I'm just bemused by the attitude that
audio processing tools should be anything more than that. Pretty pictures
and dumbed down control ranges don't help me make better mixes, they just
get in my way.

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Fri, Feb 08, 2013 at 10:14:58PM +, Fons Adriaensen wrote:
 Digital emulations of well-known analog equalisers have become a genre
 of their own... Usually the 'good imperfections' (noise, distortion,...)
 are emulated as well, as if the creators of those EQs actually added
 them on purpose. I can't imagine any of the designers at e.g. Neve or
 SSL ever doing that - they went for the best technical specs they could
 have.

Agreed. Some types of distortion can sound nice on the right material,
but I prefer to add that separately if I think it's required. I don't
want an EQ to make that decision for me.

 Not that all equalisers are equal, far from it. Some of those classic
 designs had some unusual features such as higher order shelf filters
 which are actually quite nice to have.
 
 I wrote an equaliser having those some years ago (not yet published,
 maybe I will some day), and it has become my 'workhorse'. You can
 see some of the frequency responses here:
 http://kokkinizita.linuxaudio.org/linuxaudio/shelf2filt.html

Please do consider releasing that if you get time. I can think of several
situations where it would have been very useful to have that :-)

John
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Re: [LAD] So what do you think sucks about Linux audio ?

[John Rigg]

On Wed, Feb 06, 2013 at 12:09:17PM -0500, Fred Gleason wrote:
 I suspect that many of the points in your list about confusion and
 fragmentation come from users who are expecting this One Perfect System
 and are then disappointed by the reality of having to make choices and
 exercise knowledge.  (And, even those Other OSes that purport to deliver
 this universal platform are more sizzle than steak here, as users who have
 attempted to configure multiple applications with varying requirements for
 MME vs. DirectSound vs. ASIO can attest).  Life is complicated.  Linux
 exposes this and empowers the user to make choices about what fits *his*
 application best, rather than trying to paper them over into an illusion
 of homogeneity.  That's a strength, not a weakness!

I agree with everything in your post, and especially the above point.
Many here appear to be music makers rather than audio engineers. That's
two totally different sets of requirements.

I use Linux for audio engineering (my job) and find it already meets my
needs better than Windows or OS X. There are only two things missing AFAIC,
and neither is the fault of Linux.

The first is the lack of an effective DAW session exchange program, which I
blame on the deliberately obscured session data formats used by Avid etc. The
second is lack of support for many pro level interfaces, thanks to the lack
of programming information from the manufacturers (RME and a couple of others
excepted).

John
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Re: [LAD] [LAU] So what do you think sucks about Linux audio ?

[John Rigg]

On Thu, Feb 07, 2013 at 08:47:55AM -0800, Michael Bechard wrote:
 I think others (most?) in this community want to see Linux audio 
 flourish and not become relegated to an audio environment for 
 programmers. Nobody's slagging audio geeks, really, we just want to see 
 some effort put into making the platform more accessible to a wider 
 audience, and in the process (hopefully) maybe free up that right side 
 of the brain to do its own thing in our own music-making processes. I 
 personally do not like to have to think about very technical issues when
 making music; I just want to create.

I should point out that your requirements as a music maker and those of audio
engineers like myself are mutually exclusive in many respects.

I need flexibility and configurability, plus access to the source code,
in order to do my job effectively. Of course it should be possible to cater
to both sets of users, but I don't see how it can be done easily. Don't
Windows and OS X already offer what you describe?

John
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Re: [LAD] [LAU] So what do you think sucks about Linux audio ?

[John Rigg]

Maybe someone should start new mailing lists for Linux music software
developers and users. I'm serious - there's clearly a demand for it.

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Re: [LAD] C - Change in memory causing seg.fault (but why?)

[John Rigg]

On Fri, Nov 09, 2012 at 03:24:56PM +0100, Muffinman wrote:
 Yeah, that explains why I didn't get any compiling error on a 32bit
 computer and did get on a 64bit one (where I did correct the error).
 However, the latter was satisfied with %li. Apparently in c++ there is
 no difference between long and long int (as far as I can find through
 Google) but for c I could not find much info on a potential difference.

long is the same as long int in both. The problem is that int and long int
are both 32 bits on i386, but on x86-64 int is 32 and long int is 64 bits.

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Re: [LAD] [LAU] Linux Audio 2012: Is Linux Audio moving forward?

[John Rigg]

On Wed, Oct 10, 2012 at 10:01:23PM -0400, drew Roberts wrote:
 Let's say I want at least 24 ins.
 
 What do I get? Where can I find a HOWTO on my options?

Here's a HOWTO on using multiple Delta 1010s (which can also be adapted
for other cards):

http://www.jrigg.co.uk/linuxaudio/ice1712multi.html

Note that the 1010 is still in production and there are so many of
them out there that used replacements should be available for quite
a while if new production ceases.

Another cheap option is a used RME HDSP9652 (also still being made)
with 3xADAT I/O. The PCIe alternative is the HDSPe RayDAT mentioned
elsewhere in this thread.

Going up the price scale there are RME MADI cards, both PCI and PCIe
versions. I used an RME HDSPe MADI with an SSL Alpha-Link for a couple
of years with excellent results, and I don't expect either of those to
go out of production for a while yet.

Future availability of PCI motherboards might be a concern, but there
are still many new boards being made with PCI slots.

John
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Re: [LAD] [LAU] Linux Audio 2012: Is Linux Audio moving forward?

[John Rigg]

On Thu, Oct 11, 2012 at 05:31:19PM +0400, Louigi Verona wrote:
 Speaking of hardware drivers, long time ago I wrote this article on
 E-MU 0404 USB:
 http://www.louigiverona.ru/?page=projectss=writingst=linuxa=linux_emu0404usb
 
 For a long time it was my mostly read article. Some people theorized
 that it is possible to make the soundcard working, but my tests have
 concluded
 that it is surely impossible without voodoo spells.
 
 Is there any system solution to these kind of things, when the specs are
 available,
 but nobody cares?

If it's a popular device shouldn't it be possible to organise the programming
equivalent of a group buy and get interested users to pay someone who knows
the necessary voodoo to get it working?

It might not be a case of nobody cares, but that nobody can afford to drop
their paid work for long enough to look at the problem.

John

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Re: [LAD] Kontakt sampler format (and others like EXS24)

[John Rigg]

On Sat, Sep 01, 2012 at 03:39:27PM +0200, Emanuel Rumpf wrote:
 2012/8/31 John Rigg:
  Thanks for taking the initiative on this.
 
  The lack of high quality samples usable on a Linux system has been quite a 
  problem.
 
 
 What, more closely, is a high quality sample (today) ?
 
 Is it very different, from what it was ten years ago ?
 If yes, why did it fit  formerly, but not today ?
 Have our ears eventually improved  within that time-period ?

What has changed is production budgets. Ten years ago there was more
money available for hiring real musicians along with places to record
them in and technical personnel to do it. If you're recording soundtracks
for TV or film you need samples that are good enough to replace the
real thing unless you're lucky enough to be working on the biggest
productions.

John
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Re: [LAD] Kontakt sampler format (and others like EXS24)

[John Rigg]

Thanks for taking the initiative on this. The lack of high quality
samples usable on a Linux system has been quite a problem.

John
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[LAD] xfmr-plugins-0.0.1

[John Rigg]

xfmr-plugins-0.0.1 has been released. At present it contains one
LADSPA plugin, an anti-aliased transformer distortion emulator
intended to be used on the master bus when mixing in a DAW such
as Ardour.

I made it for my own use, in order to get closer to the sound of
an analogue mix without the extra round trip through the DA-AD
converters. Hopefully others may find it useful. License is GPL v2.

More details and download link here:
http://www.jrigg.co.uk/linuxaudio/xfmr-plugins.html

John
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Re: [LAD] LADSPA plugin undefined symbols - SOLVED

[John Rigg]

On Mon, Jul 09, 2012 at 10:55:12PM +0100, John Rigg wrote:
 It was the -lm option. With that added to the LD line in the makefile
 it could find cos, which was the undefined symbol stopping demolition.
 
 Other undefined symbols like malloc, calloc and free didn't bother it
 (these could be defined with the -l:libc.so option but it wasn't
 necessary).

As a follow up for anyone else confused by these ld options, -lm
makes ld look for libm.so (which is a symlink to libm.so.6, the
system library containing math functions). I could also have used
-lc instead of -l:libc.so.

It's one of those stupidly obvious things I missed when I read the ld
manpage.

John
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[LAD] LADSPA plugin undefined symbols

[John Rigg]

(Hope this is the right place to ask - apologies if not).

I'm trying to test my new LADSPA plugin with demolition but
the latter fails due to undefined symbols. Running objdump -t on
the .so lists several of the standard C functions used in the code
as undefined symbols. I've tried installing libc6-dbg and adding
-g to the CFLAGS in the makefile (which is a modified version of
the one in ladspa_sdk) but it makes no difference.

System is Debian testing amd64 (pre-multi-arch snapshot from
Jan 2012). I thought it might be due to a mismatch in the build
tools, so I tried building the plugin on Debian Lenny (5.0.3).
The same thing happens there.

I've tried running demolition on a couple of the Debian packaged
swh-plugins and it runs without a problem, but any plugin I compile
from source (not just mine) gives the same type of error.

All of the undefined symbols are standard libc functions like malloc,
calloc, free, cos etc. The plugin works perfectly in Ardour2 on both
systems. Anyone have any ideas how to fix this?

John
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Re: [LAD] LADSPA plugin undefined symbols - SOLVED

[John Rigg]

On Mon, Jul 09, 2012 at 09:41:44PM +0200, Robin Gareus wrote:
 On 07/09/2012 09:18 PM, John Rigg wrote:
  I'm trying to test my new LADSPA plugin with demolition but
  the latter fails due to undefined symbols. Running objdump -t on
  the .so lists several of the standard C functions used in the code
  as undefined symbols. I've tried installing libc6-dbg and adding
  -g to the CFLAGS in the makefile (which is a modified version of
  the one in ladspa_sdk) but it makes no difference.
  
  System is Debian testing amd64 (pre-multi-arch snapshot from
  Jan 2012). I thought it might be due to a mismatch in the build
  tools, so I tried building the plugin on Debian Lenny (5.0.3).
  The same thing happens there.
  
  I've tried running demolition on a couple of the Debian packaged
  swh-plugins and it runs without a problem, but any plugin I compile
  from source (not just mine) gives the same type of error.
  
  All of the undefined symbols are standard libc functions like malloc,
  calloc, free, cos etc. The plugin works perfectly in Ardour2 on both
  systems. Anyone have any ideas how to fix this?
 
 since Lenny, Debian switched to multi-arch support. libs will be in
 /usr/lib/triplet/ instead of /usr/lib/  NTL the linker should find
 them -- but check /etc/ld.so.conf.d/* and /etc/ld.so.conf if you've
 upgraded from Lenny or use an early testing system.
 
 To get the standard libs, it is necessary to explicitly use
 LOADLIBES=-lrt -lm when linking objects these days.

It was the -lm option. With that added to the LD line in the makefile
it could find cos, which was the undefined symbol stopping demolition.

Other undefined symbols like malloc, calloc and free didn't bother it
(these could be defined with the -l:libc.so option but it wasn't
necessary).

Thanks,
John
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