Hi list,
I'm researching a project that requires a light-weight computer and I'm
considering using something like an Intel NUC NUC5i7RYH with a low cost
stereo audio interface such as a Behringer UCA222.
Will there be any issues running linux audio on NUCs such as this one?
All the best,
Iain
--
_________
Iain Mott
http://escuta.org
Hello, Ralf--
I hoped that just the headphone amps of the Focusrite
> are weak,
>
i have a Focusrite 2i2. A few years ago, i sat down with a salesman to
audition headphones for it. i called ahead before the session to let him
know what phones interested me (an expensive mixing pair), and he told me
the 2i2 didn't have strong enough drivers to run them. He eventually sold
me a pair of Focal Spirit Professionals, which sounded fantastic--to me, at
least. They were half the price of the pair i first mentioned, too, so
he's an excellent salesman. Anyway, i wouldn't have suspected the same
issue with a larger model. The problem seems to run in the family, i guess.
I suspect the Presonus is the better choice.
>
i'm not the most experienced guy for this opinion, but i have to concur. i
also have to apologize for not chiming in sooner. i own the Presonus of
your original posts and have had success with it--for my limited projects,
anyway. It came packaged with a plugin host i bought, and the company sent
the Presonus with one of these
<https://www.amazon.com/CableWholesales-Speed-Desktop-Powered-Multi/dp/B003V…>.
i've used it to plug the Presonus into my computer when i want to record
and have never had an issue. The MIDI connections even show up in
Patchage. Again, my apologies for not chiming in sooner. i don't own an
iPad and didn't think i could help with that issue.
Keep trying Ralf, though i won't blame you for not trying the Presonus,
again. You'll come across a solution, soon.
Tom
I do not recall seeing a reply on list regarding this.
On Tue, December 27, 2016 6:07 pm, Peter P. wrote:
> starting jackd1 after connecting a motu ultralite AVB sound
> card throws below error. The second startup attempt works then.
> What could be wrong? How could I debug this further? Thanks! Peter
>
> 12:55:29.917 JACK was started with PID=23950.
> loading driver ..
> apparent rate = 44100
> creating alsa driver ...
> hw:AVB|hw:AVB|64|2|44100|24|24|nomon|swmeter|-|32bit
> configuring for 44100Hz, period = 64 frames (1.5 ms), buffer =
> 2 periods ALSA: final selected sample format for capture: 24bit
> little-endian in 3bytes format
> ALSA: use 2 periods for capture
> ALSA: final selected sample format for playback: 24bit
> little-endian in 3bytes format
> ALSA: use 2 periods for playback
> ALSA: cannot set hardware parameters for playback
> ALSA: cannot configure playback channel
> cannot load driver module alsa
Fernando at CCRMA reported a similar problem on the jackdev list, I think
mentioning a similar problem even when changing sample rates. My guess is
that the device has a PLL with a long time constant for locking the
sampling clock, such that when you send a request for a particular sample
rate the device takes many seconds to report that the requested sample
rate is achieved and the device is ready to use.
The solutions I can think of for a device with such behavior is to make
sure the device is running at the desired sample rate before starting
jackd (e.g. using an ALSA utility to setup the device first, playing
something through alsaplay, etc.), just accept that you have to start
jackd twice, or modify the source for jackd to wait longer or have a retry
loop when a device does not report quick success for setting the ALSA
parameters.
--
Chris Caudle
Hi,
first of all, with a non-rt kernel 4.8.13-1-ARCH threadirqs the Presonus
worked with 256 frames/period, in this regard the Focusrite is better,
it's possible to use it with 128 frames/period. I used the same
Qtractor song with fluid-dssi and hexter, as I used for the Presonus. I
didn't test S/PDIF.
I returned the Presonus, so I can't compare the Presonus and Focusrite
directly.
The headphone outputs of the Focusrite are the most worse I ever heard.
I made a test with 44.1 KHz. I ripped "weather with you" from a Crowded
House CD. Ardour played the ripped WAV and a CD player played the
original CD. A comparison with the RME HDSPe AIO's headphone output
makes no sense. I hoped that just the headphone amps of the Focusrite
are weak, so I compared the sound quality of the Focusrite's
analog outputs with a Behringer ADA8000 connected by ADAT with the
Focusrite, using the same cables and mixer channel and then by
listening to the mixing console's headphone amp. After that I used the
same cables to connect the Focusrite and the ADA8000 to a hifi amp,
without a mixer involved. The ADA8000's sound quality always is more or
less equal to a hifi CD player, the sound of the Focusrite was always
vastly worse. Not only nuances are missing in the audio signal, but I
have the impression that the bass is purposely boosted.
Since I need a device that provides ADAT and works with my iPad, I'll
continue testing, but for Linux usage only, without an ADAT device, I
suspect the Presonus is the better choice.
I suspect the Focusrite's headphone and analog outputs are completely
unusable for home recording. At least I would send back the Focusrite,
if I wouldn't need DAT for the iPad. The sound is close to a cheap
no-name Walkman with a bass boost feature. Using studio headphones
there anyway is some 80s Walkman headphone alike sound impression.
It doesn't require blind tests to compare a telephone with a hifi amp.
The Focusrite's sound is better than a telephone, but I want to make
clear, that there's an audible difference compared with other gear, that
doesn't require blind tests.
FWIW the front is made of plastic and a countersunk bolt isn't
completely countersunk, but the cross recess of the screw is damaged.
It's a brand new device, not a b-stock.
Regards,
Ralf
Hi,
Linux tests and more iPad tests coming soon.
Today I got the Focusrite Scarlett 18i20 2nd Gen.
The iPad shows 20 output channels.
Connecting to my iPad works without an active hub.
Channel 7 and 8 are for headphone output 1, channels 9 and 10 are
for headphone output 2. Both work when connected to the iPad.
Until now I didn't check more, but I had time to repair and clean some
of my equipment, so it might be possible, to comment the audio quality,
too.
I can't take a look at the Windows mixer, since for testing purpose
only Windows XP is available, but Windows 7 or a more recent Windows is
required.
To be continued...
Regards,
Ralf
Hi list,
It seems that the latest (packaged see below) ladspa version of the Calf
plugins is not working with the main applications I use which still use
ladspa: Rosegarden, jack-rack and Ardour. The former two provide some
clues about errors happening when trying to load calf.so (which is a
symbolic link in /usr/lib64/ladspa/calf.so pointing to
/usr/lib64/calf/calf.so). Ardour seems to silently "ignore" all of the
ladspa versions of Calf (but I may be missing error messages/clues...).
I am on Fedora 24 and the distribution does provide a
ladspa-calf-plugins package to which I already filed a bug report.
Interestingly if I use the calf.so file compiled with the latest git
version of LMMS [1] (where the provided version of Calf seems to date
back at least 8 month, but even 2 or 3 years for most source files) the
plug ins are visible and usable.
Any info would be interesting, and *maybe* someone is also using ladspa
(for me this especially relevant for existing versions of files I have
using these plugins)
Side note... Plugin "binary-dependency" in the light of longer term
archiving is probably an unsolvable problem whichever the format :)
Lorenzo.
[1] https://github.com/LMMS/lmms/tree/master/plugins/LadspaEffect/calf
On Mon, Jan 2, 2017 at 6:02 AM, Jeanette C. <julien(a)mail.upb.de> wrote:
> Jan 2 2017, Yassin Philip has written:
> ...
>
>> Is there such thing as a default tuning "norm" or (most probably) does
>> every plugin do what it wants to do?
>>
> The default or standard tuning for typical western musical instruments
> is at 440Hz. There are variations, when it comes to concert pitch. I
> believe that the Americans sometimes choose 441 or 442Hz. But every
> hardware synth - that I know of - is tuned to 440Hz with a well-tempered
> scale.
>
Equal tempered. I've never seen a software synth or modern tuner that uses
one of the well tempered systems.
12 equal tones to the octave with the A above middle C tuned to 440Hz is
the standard.
MIDI pitch to Hz can be calculated with the formula: 440 ^ (2 * ( (x - 69)
/ 12))
When I install, say ZASFX, and press the A2 key, is it a 441Hz A, or a
440..?
Is there such thing as a default tuning "norm" or (most probably) does
every plugin do what it wants to do?
How can I know the real tuning of a given instrument plugin, apart from
playing it through a tuner, which is very hard to do on some patches..?
Is there a general rule (like tune your acoustic instruments to 441 and
you'll be fine)?
Happy new year? :)
yPhil
--
Yassin Philip - New album out NOW
http://yassinphilip.bitbucket.org
Best wishes to everyone !
The first track of the year is a long, 12-minutes trance-type travel
that could perhaps be best enjoyed as a background music while doing
something else. Basses should be plentiful on systems that can support
them, while some care was also taken to have them represented in a
higher audio range.
I 'discovered' this 'old' track recently. Its original name was
jam8 while the current jam number is at 85. I guess that qualifies as
'old'. I renamed it 'Jiji' for the playful sounds of the name as well
as for the meaning of the yi jing hexagram 既济 of, say, 'feeling good
having achieved coming to shore while knowing that there will be other
things to face on the land itself'. Or something like that. Which
can also represent in a way having finished a year and starting another
one. Jiji.
It's a strange track in the sense that it can be listened-to easily I
find. While not being muzak.
https://soundcloud.com/nominal6/jiji
As always, comments are welcomed.
Cheers.