Hello all,
The purpose of this post is to gather some opinions on the
use of 'tape distortion' plugins.
The way an analog magnetic tape recorder distorts the signal
is quite complicated. The non-linear response is mainly due
to the hysteresis of the magnetisation process, and how this
interacts with the HF bias added to the recorded signal.
Simplifying things a bit, as the tape moves past the recording
head and away from it, it goes through several cycles of an
hysteresis loop with decreasing amplitude, and it finally
converges to some value more or less proportional to the signal.
The bias signal itself does not remain on the tape, or at most
at a very low level.
The same process but with just a very high level HF bias is
used to erase the tape.
Simulating this accurately is quite complicated, and very CPU
intensive. The sample rate must be at least twice the bias
frequency (around 100 kHz for 'pro' equipment) and for each
sample you need to evaluate the hysteresis loop up to a hundred
times (depending on tape speed) to simulate the convergence.
Over the past few months I've written the code to do this,
with the aim to present a useful 'tape emulation' plugin
at the next LAC. It uses a somewhat simplified form of the
Preisach algorithm for the hysteresis.
<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/preisach-model.pdf>
But the first results are somewhat sobering.
Have a look at
<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/tapesaturation.png>
This shows the tranfer function (recorded magnetic signal as
a function of the input) for three values of the HF bias.
The blue line is what you get without bias. It result in a
very smooth saturation for high level signals, but a low
amplitude signal will be severly distorted due to the flat
region around zero.
As bias level is increased, this effect is reduced and the
low level gain increases up to some maximum. This is shown
by the orange curve.
The recommended way to set the correct bias level is to
further increase it until the small signal gain is reduced
by a few dB (typically 1 or 2 dB, depending on tape type).
This is shown by the green curve.
The central part of this is very linear (less than 1% THD).
But now the smooth saturation is almost transformed into
hard clipping. The sharp bend in the curve occurs when the
signal amplitude is higher than the bias amplitude.
The only effect that remains in a complete simulation is
the result of the EQ applied to the signal to be recorded.
Higher frequencies have a 'self erasure' effect (which is
nicely reproduced by the simulation) and so need to be
amplified. The net result is that they will saturate at
a lower input level.
How much EQ is required depends mainly on tape speed, higher
speeds need less. At the most popular 'pro' speed (381 mm/s)
this would be something around 10 dB at 10 kHz. At the higher
(762 mm/s) speed typically used for 'master tapes' it's just
a few dB.
So what does effectively remain of the 'smooth saturation
and compression' that is claimed to give tape recording its
magical 'warm' character ? Is it just a myth ?
I also simulated the green curve directly without going
through the complicated full simulation, and honestly,
to me that sounds just the same. And unless you really
use very high levels (much more than would actually be
used) the net effect is marginal. Maybe the hard clipping
can be useful when applied to individual tracks (e.g.
drums or bass), but then there are much simpler ways
to do this than 'tape emulation'.
Comments invited !!
Hi.
What do I need to know going forward trying to switch to Pipewire on a newer
machine? I skipped PulseAudio for several reasons.
I am basically using plain jackd with the tools like jack-lsp and
jack-connect.
from the command-line. That was relatively simple, since
I would specify the desired device on the jackd command-line,
and that was basically it. How do I use Pipewire without a graphical
control panel, practically speaking?
In particular:
* How do I select the desired device to use?
(Typical laptop + docking station setup easily has 2 sound cards these
days...)
* What is the equivalent of jack-connect and jack-lsp, i.e. how do I
connect clients to eachother and to a hardware output?
* Do I have to care about volume controls, which I didn't have to
previously with jackd since I would adjust volume on the clients?
--
CYa,
⡍⠁⠗⠊⠕
This is a relatively minor bugfix release. However there are some other general
improvements included. We've been steadily updating the code, to make it meet
current standard, but (hopefully) doing so in a way that improves performance
but makes no detectable difference to the actual sound.
--
Will J Godfrey {apparently now an 'elderly'}
After the second move this year, I've finally started to set up my home studio properly again. (Ubuntu 24.04.1 LTS, Mixbus, RME ADI-2 Pro for monitoring, Focusrite Scarlett 18i20 2nd generation for integrating analog hardware)
Since I've also updated Ubuntu, I've switched my workstation to Pipewire, as I've had quite good experiences with it on my laptop and running several audio interfaces in parallel is quite problematic.
My first attempt with a smaller project (17 audio tracks) ran without any problems.
My second attempt with a medium-sized project (~30 audio tracks), however, is driving me to despair. I have a DSP load of 50-60%, no buffer underruns but constant stuttering and occasional beeping.
The log says the following
Nov 12 22:37:13 muse pipewire[2048]: spa.alsa: hw:4c: snd_pcm_mmap_commit error 1022 514 1022: Data transfer interrupted (broken pipe)
Nov 12 22:37:24 muse pipewire[2048]: spa.alsa: hw:4c: follower delay:873 target:1536 thr:1024 resample:1, resync (2 suppressed)
It is interesting that when I start the safe mode in Mixbus, i.e. all effects are switched off, I have a DSP load of 10 percent and the stuttering and beeping is just as present. I don't have any analog hardware connected at this stage, so I read all tracks directly from disk and send them stereo to the RME interface.
I didn't have these problems under jack2, but then the RME was also connected to the Scarlett via TOSLink, so there was only one interface visible to the system.
I therefore suspect that it is a problem with the hard disk usage, as I can't think of any other reason.
Unfortunately I have not found a way to optimize hard disk access and I don't even know if this is done directly by Pipewire or if one of the usual libraries is used for this.
Have you seen this phenomenon before or do you have any ideas what else I can try?
Thanks for any suggestions!
Holger
--
Holger Dehnhardt
holger(a)dehnhardt.org
https://www.dehnhardt.org
Hi list,
wanting to output audio from two applications at the same time using
ALSA and not ~/.asoundrc in place, I am getting the following error for
the second application (mpv) wanting to access the soundcard:
ALSA lib pcm_dmix.c:999:(snd_pcm_dmix_open) unable to open slave
[ao/alsa] Playback open error: Device or resource busy
[ao] Failed to initialize audio driver 'alsa'
Could not open/initialize audio device -> no sound.
Querying my system for devices using $ arecord -L I get (amongst other
interfaces and cards):
default
hw:CARD=PCH,DEV=0
HDA Intel PCH, ALC269VC Analog
Direct hardware device without any conversions
plughw:CARD=PCH,DEV=0
HDA Intel PCH, ALC269VC Analog
Hardware device with all software conversions
I remember that the hw: devices may not allow simultaneous access, but
the plughw: might. So will it help to declare the plughw:CARD=PCH,DEV=0
as the "default", and how would I go about to do so?
https://www.alsa-project.org/wiki/Asoundrc indicates I should setup dmix
but I somehow wonder if/why the above doesn't work out of the box
without an .asoundrc?
Thanks for all ideas!
Peter
Forwarding to LAU in case there are interested users who are not on the
Rosegarden mailing list ;-)
-------- Forwarded Message --------
Subject: [Rosegarden-user] Rosegarden 24.12 Released
Date: Wed, 4 Dec 2024 14:17:27 -0500
From: Ted Felix <ted(a)tedfelix.com>
To: rosegarden-user <rosegarden-user(a)lists.sourceforge.net>
====== ROSEGARDEN 24.12 RELEASED ======
The Rosegarden team is proud to announce the release of version 24.12
of Rosegarden, a MIDI sequencer that features a rich understanding of
music notation along with basic support for digital audio.
http://www.rosegardenmusic.com/
Included in this release...
===== Bug Fixes =====
* LV2: Fix link problem with no gtk2. [2e168f0d]
* LV2: Add lv2 check for GUI library ok. [9cb306b3]
* Manage MIDI Banks and Programs dialog now allows the same bank
MSB/LSB to be used for percussion and non-percussion banks on
the same device. Bug #1692.
* Fix wrong ports being assigned on file load. Bug #1687.
[663099cc]
* Fix autoscroll and highlighting behavior with multiple segments
on a track. Bug #1672.
* Fix missing scroll bar on Instrument Parameters Bank dropdown.
Bug #1695. [295a2f7a]
* Fix "turn repeats into copies" misses last segment. Bug #1696.
[afcf2f0e]
* Fix shortcut order ignored. Bug #1702. [b1272a6d]
* Fix unexpected cursor position when moving from note to note
on a bar with a clef or time signature. Bug #1704. [215b9d27]
* Fix unexpected translation of bank and program names. Bug #1705.
[6dbd6bcd]
* Fix unexpected Save As... directory. Feature #522. [15db9d10]
* Fix broken tempo line in tempo ruler after adding a time
signature change. Bug #1706. [6a7ef89e]
* Fix "Modify MIDI Filters" dialog does not mark document as changed.
Bug #1707. [4e505977]
* Fix LV2 sfizz plugin support. [8fce0614]
* Fix group box formatting issue in native/light theme. Bug #1678.
[d2df6376]
* LV2: Fix crashes and locking related to atom buffers. [95b09722]
[be37b31e]
===== New Features =====
* Make MIDI File Division value configurable for Export.
GH #9. [991e4839]
* WAV export of audio and soft synth tracks.
File > Export > Export WAV File... [5b4d109e]
* Apply interpretations to more than one segment.
Segment > Interpret... Feature #517. [b8a243a5]
* LV2 support has been promoted to beta. [a602850c]
* Advanced Looping has been promoted out of beta. [25bdbf69]
===== General Improvements =====
* Manage MIDI Banks and Programs dialog is now full-featured and can
be used to edit every aspect of an .rgd device file.
Studio > Manage MIDI Devices > Banks...
* Tempo and Time Signature Editor has been rewritten and improved.
===== Additional Contributions =====
* Korg-M3-Factory-ALL.rgd (Chuck Elliot)
* Korg-M1.rgd (Chuck Elliot)
* Updated Polish translation (Grzegorz Pruchniakowski)
* Updated Chinese translation (Icenowy Zheng)
* Yamaha-MM6-MM8.rgd (Icenowy Zheng)
* Yamaha-PSR-SX600.rgd (Huanyu Liu)
_______________________________________________
Rosegarden-user mailing list
Rosegarden-user(a)lists.sourceforge.net - use the link below to unsubscribe
https://lists.sourceforge.net/lists/listinfo/rosegarden-user
Hi LAU,
I have now switched to pipewire on my main laptop, with an overall
positive experience especially JACK-aware applications typically working
'out of the box' and (more or less) co-existing with other 'desktop'
applications (I might talk about some caveats in a different thread).
Something which I still can't totally work out though is jack
frames/period (aka buffer size aka 'quantum') and its management.
For 'everyday' use I keep this at 1024 (which I think is the default)
but in some cases I want to push it down even to 128, and I must say
that with recent versions of pipewire and with a reasonable 'cpu
settings' (I have a tuxedo machine and it has its own cpu thingy) this
actually works quite well.
What I'm still not understanding completely is how to best manage the
frames/period setting.
Pre-pipewire in this use case I'd have dedicated qjackctl presets and
just select the desired frames/period one for the sound device of
interest, restart jack and be good to go.
Now with Pipewire it seemed that this would be handled via pw-metadata
and setting the clock.force-quantum value (I even made myself a script
and then a little python-tk 'gui' for this and setting the sample rate).
Up to recently indeed all of the previous preset-related settings in
qjackctl (most notably sound device, sample rate, frames/period and
periods/buffer) where disabled, but then now _only_ the frames/period
setting dropdown seems to be available again and selectable. And indeed
jack software (e.g. Ardour) seems to report that buffer size set in
qjackcrl. But then if I do:
pw-metadata -n settings 0
I see 1024 for clock.quantum and 0 for clock.force-quantum
Explicitly setting clock.force-quantum also works. So I'm a bit
confused. It seems the idea is to be able to change buffer size 'on the
fly' or at least without restarting applications (e.g. in Ardour I _did_
have to disconnect and reconnect to JACK-pipewire, but without
closing/reopening).
Then I remember lots of discussions and advice about setting, for
instance, the periods/buffer to 3 for USB sound devices etc. etc. With
Pipewire we don't care about this any more?
I also had a couple of instances where Ardour complained about the
session being at 48000 but jack running at 44100 when actually it was
running (reportedly) at 48000 - but then this is another matter, maybe.
Hopefully getting insight on this could be helpful to other LAU
'transitioning' to Pipewire[-jack] :-)
Lorenzo
