audiophile sound on FreeBSD ?

[AikiZen]

Hello,

I would like know if FreeBSD can used for audiophile listening and
composing ? I am a music player and i would like listen with the best
quality are possible and record me with my guitar.

I test FreeBSD as a long time but my objective at this moment are
different.

I would like know if FreeBSD give a better audio experience than
GNU/Linux.

Best Regard,
Aiki Zen.

PS: Why does not exist anymore French mailing list ?

[@lbutlr]

Post by AikiZen
I would like know if FreeBSD can used for audiophile listening and
composing ? I am a music player and i would like listen with the best
quality are possible and record me with my guitar.

Depends, how many orders of magnitude beyond the range of human hearing do you think you need?

[Polytropon]

Post by AikiZen
I would like know if FreeBSD can used for audiophile listening and
composing ? I am a music player and i would like listen with the best
quality are possible and record me with my guitar.

Even though sophisticated Linux distributions like Ubuntu Studio
or KXStudio offer great software solutions for audio as well as
good hardware support, FreeBSD can be utilized as well, because
it is a multi-purpose OS. You still have to choose your hardware
wisely to make sure it will fully support the OS. Many applications
not available natively for FreeBSD can be used with its Linux ABI.

I've been using FreeBSD for my (more or less simple) music
applications, without any problems. I've been using a CMI-based
sound card (PCI) and my own homemade guitar pickup amplifiers, PA,
and connectors to tape recorder equipment. However, I have not
tried to get MIDI working - the last time I saw working MIDI was
at a time where Atari ST was considered a professional platform
for musicians... ;-)

Post by AikiZen
I would like know if FreeBSD give a better audio experience than
GNU/Linux.

It's probably not such a matter of the OS as it is of the hardware
and the peripherials. Oh, and compression also matters a lot:
MP3 with maximum PCM-like compression just sounds terrible on
_any_ OS. :-)

[Chris Hill]

Post by Polytropon

Post by AikiZen
I would like know if FreeBSD can used for audiophile listening and
composing ? I am a music player and i would like listen with the best
quality are possible and record me with my guitar.

Even though sophisticated Linux distributions like Ubuntu Studio
or KXStudio offer great software solutions for audio as well as
good hardware support, FreeBSD can be utilized as well, because
it is a multi-purpose OS. You still have to choose your hardware
wisely to make sure it will fully support the OS. Many applications
not available natively for FreeBSD can be used with its Linux ABI.
I've been using FreeBSD for my (more or less simple) music
applications, without any problems. I've been using a CMI-based
sound card (PCI) and my own homemade guitar pickup amplifiers, PA,
and connectors to tape recorder equipment. However, I have not
tried to get MIDI working - the last time I saw working MIDI was
at a time where Atari ST was considered a professional platform
for musicians... ;-)

Post by AikiZen
I would like know if FreeBSD give a better audio experience than
GNU/Linux.

It's probably not such a matter of the OS as it is of the hardware
MP3 with maximum PCM-like compression just sounds terrible on
_any_ OS. :-)

I'll second Polytropon's advice: it's primarily about the hardware and
the support thereof, not so much the OS. My needs have been pretty
simple: In the course of digitizing hundreds of LPs, I'm editing album
sides down to individual tracks and removing gratches.

I got a pair of "good" powered speakers [1] and plugged them into my
FreeBSD machine's audio out jack from the onboard sound card. They sound
very good to me, but I'm getting a lot of noise in the bargain. This can
be mitigated by paying attention to the gain structure. I have attempted
to record into the audio in of the onboard soundcard, but that did not
end well. I'm using a perfectly fine workaround instead. For quality,
low-noise recording and playback, I'm afraid you would need to add
hardware (card or external). What hardware? Good question; I'd answer it
if I could.

And I have no idea what's become of the French mailing list. Je suis
desole (ASCII character set here, sorry).

[1] "Cakewalk by Roland" if anyone cares.

[Erich Dollansky]

Hi,

On Mon, 23 Apr 2018 22:42:42 +0200

Post by AikiZen
Hello,
I would like know if FreeBSD can used for audiophile listening and
composing ? I am a music player and i would like listen with the best
quality are possible and record me with my guitar.
I test FreeBSD as a long time but my objective at this moment are
different.
I would like know if FreeBSD give a better audio experience than
GNU/Linux.

I do not expect so as it uses the same programs.

Erich

[Ralf Mardorf via freebsd-questions]

"Audiophile" is for the sound quality? If so it stands and falls with
the chosen audio device. A long time ago I tested FreeBSD with a
RME HDSPe AIO PCIe card. It's a professional audio quality grade device.

I can't comment on FreeBSD real-time abilities, latency, MIDI jitter
etc., but ass long as you operate with that much latency, that you'll
get no xruns and no other hardware, e.g. the mobo shouldn't cause
such an issue, the sound quality has got nothing to do with the chosen
OS and just a little bit to do with the chosen software.

In short, instead of using a prosumer audio device, you should buy a
professional audio device.

There isn't much choice, since vendors tend not to support drivers for
Linux and FreeBSD.

It's possible to use the RME HDSPe AIO PCIe with FreeBSD and Linux, but
FreeBSD comes without hdspmixer and under Linux you get hdspmixer, but
not all ADAT IOs could be used. Perhaps a class compliant device such as
the RME Babyface works with FreeBSD, too.

When using Linux, then USB is no issue at all, I own a
Focusrite Scarlett 18i20, the usable latency with all IOs used is
better than that of my RME PCIe card with just using 4 IOs.

However, regarding audio quality you can't compare a pro-sumer Focusrite
Scarlett series with any professional RME device, excepted of the ADAT
IOs that require an additional device. IOW you could use a pro-sumer
Focusrite Scarlett series with a professional ADAT device to get
professional audio quality.

[AikiZen]

Hi,

I would like install FreeBSD on my Dell Précision M4700. It's a laptop,
no possible to change sound card...

If i understand well no very difference between Linux and FreeBSD, may
be i would like use FreeBSD because there are a new challenge.

I just want listening my encoded audio CD and record, may be, my music.

Maybe FreeBSD are a best documentation for use, like the handbook and
else...

Thank for your answer,
and take care,
Aiki.

[Polytropon]

Post by AikiZen
I would like install FreeBSD on my Dell Précision M4700. It's a laptop,
no possible to change sound card...

In this case, you can still use an "external sound card",
i. e., a USB-connected audio device. Of course you should
check driver support before buying. If the built-in audio
is sufficient for your needs, it will be supported probably
by the HDA driver (I have no idea what audio chip is inside
a Dell Precision M4700, but the spec mentions "Intel Panther
Point PCH - High Definition Audio Controller" which suggests
that it'll be the HDA driver).

Post by AikiZen
If i understand well no very difference between Linux and FreeBSD, may
be i would like use FreeBSD because there are a new challenge.

That is correct, at least within the margins of your
question.

Post by AikiZen
I just want listening my encoded audio CD and record, may be, my music.

In this case, FreeBSD is able to provide what you'll need.
You probably encoded to MP3 or OGG/Theora? Players are
available. But you can also play 1:1 copies ("raw") of
audio CDs. I've been doing this in my "home studio" in
the past. Many programs for dealing with audio are already
available as ported applications, and in many cases, you
can use Linux programs on FreeBSD, too. It's probably
possible to even use "Windows" programs via wine (but
I've never tried that for audio programs).

Post by AikiZen
Maybe FreeBSD are a best documentation for use, like the handbook and
else...

FreeBSD's documentation is really good on all levels,
compared to specific Linux distributions... ;-)

[Robroy Gregg]

Good day Aiki,

I would like install FreeBSD on my Dell Pr?cision M4700. It's a laptop,
no possible to change sound card...

...

I just want listening my encoded audio CD and record, may be, my music.

Aiki, FreeBSD forums member Phishfry recommended an M-Audio "MobilePre
USB" device to me for this purpose. He found one on eBay for $35 and sent
me the link.

I've used it with audacity on three (very different) computers running
11.1-RELEASE, and it worked really well on all of them. I now have two
microphones attached, and it mixes them together in to one mono channel
for audacity to record.

A sticker on the bottom of this device reads:

PID/Ver: 200F
Firmware: V1.03
Date: 11-2008
PN: AU02-071G1
SN: 071G158B03130

On one of my computers it was necessary for me to manually select the
recording source, just because it wasn't the default, in audacity. And I
did toy around with mixer(1) some. But other than that, it just worked.

I don't have enough experience with listening for sound quality to really
comment on that aspect of it, but it sounds great to me. I use it almost
every single day, and the idea of upgrading to anything fancier hasn't
really crossed my mind.

It also tends to want gain maxed out with my SM58 microphones, or at 75%
or so; the device doesn't seem to support enough amplification to exceed
what's required by much, if any. 'not sure why, but this seems to be less
noticable with two microphones attached than with only one.

It shows up like this when I plug it in:

ugen0.7: <M Audio MobilePre> at usbus0
uaudio0 on uhub4
uaudio0: <MobilePre> on usbus0
uaudio0: Play: 48000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 44100 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 32000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 24000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 22050 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 16000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 11025 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Play: 8000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 48000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 44100 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 32000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 24000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 22050 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 16000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 11025 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: Record: 8000 Hz, 2 ch, 16-bit S-LE PCM format, 2x8ms buffer.
uaudio0: No MIDI sequencer.
pcm2: <USB audio> on uaudio0
uaudio0: No HID volume keys found.

And it shows up like this also (it's the "USB audio" entry):

% cat /dev/sndstat
Installed devices:
pcm0: <NVIDIA (0x001c) (HDMI/DP 8ch)> (play)
pcm1: <NVIDIA (0x001c) (HDMI/DP 8ch)> (play)
pcm2: <USB audio> (play/rec) default
No devices installed from userspace.

Happy days to you Aiki,
Robroy

[Ralf Mardorf via freebsd-questions]

It's a laptop, no possible to change sound card...

Most likely any elCheapo prosumer USB class compliant audio device
provides far better audio quality, than any onboard audio device.

This is a prosumer class compliant audio device:

https://www.thomann.de/de/focusrite_scarlett_2i2_2nd_gen.htm

It much likely sounds thousand times better, than an onbaord device.

This is a professional class compliant USB audio device:

https://www.thomann.de/de/rme_babyface_pro.htm

It for sure does sound thousand times better than a prosumer audio
device.

I don't know your definition of "audiophile", but if you want
professional sound you need to use a professional device, that apart
from the converters also provides a far better analog domain.

Btw. an "audio CD" already provides not that good audio quality. What
you need for high end audio quality are 48 KHz, btw. anything higher
than 48 KHz gains you nothing, to the contrary, not matching pass
filters could make the sound less good and at very high sample rates,
the energy does increase, this means the noise floor does increase.

[RW via freebsd-questions]

On Tue, 24 Apr 2018 20:09:24 +0200

Post by Ralf Mardorf via freebsd-questions
Btw. an "audio CD" already provides not that good audio quality. What
you need for high end audio quality are 48 KHz,

I'd be astonished if you could tell the difference under double blind
conditions - even if you are a child.

[Ralf Mardorf via freebsd-questions]

Post by RW via freebsd-questions

Post by Ralf Mardorf via freebsd-questions
Btw. an "audio CD" already provides not that good audio quality. What
you need for high end audio quality are 48 KHz,

I'd be astonished if you could tell the difference under double blind
conditions - even if you are a child.

I'd be surprised, if most adults with trained hearing would be unable
to perceive a difference.

To keep it short, I'll quote a short and raw explanation:

"> frankly, 48k may be a good enough for distribution, but it is

Post by RW via freebsd-questions
sub-optimal not for production ... and it is horrible for digital
synthesis.

Only if you use 'primitive' algorithms. Unfortunately there's
a lot of those around.

In summary, 96 or 192 kHz will allow you to use simpler algorithms.
That may be a good reason for higher sampler rates, but it doesn't
mean you can't have the same performance at 48 kHz.

Another good reason for higher sampling rates is that the
antialising filters in the converters can have a much wider
transition band (assuming you don't actually use the higher
bandwidth), leading to much reduced latency. It's the reason
why 'digital snakes' used in PA system usually work at 96 kHz.
By starting the transition band at 24 kHz or so they can use
very short filters, a fraction of a millisecond for some.

The same matter makes all the difference between 44.1 and 48 kHz." -
http://linux-audio.4202.n7.nabble.com/jack-oversampling-td89817.html

Loosely speaking:

It all depends on the tuning. You could assume that most gear and
software works best at 48 KHz, this is professional audio studio
standard and apart from audio CD, consumer standard, too.

However, depending on your amp, your speakers or any other part of the
audio chain or depending on the recorded signal, such as e.g. an
electric guitar played via a Celestion speaker, you might be even
unable to hear any loss when using DAT longplay (32 KHz, 12 bit
non-linear).

Since the subject is "audiophile sound on FreeBSD" I recommend to go
with a RME audio interface or another professional audio interface. The
OP should ensure that an audio interface that claims to be
"professional" isn't just a "prosumer" audio interface. Without doubts,
48 KHz is the best choice.

When using a professional audio interface, with an elCheapo Hifi amp
and supermarket speakers, you wont be able to hear the good analog
audio quality of your professional audio interface. If you can't hear
the good analog audio quality, you can't notice any issues of different
sample rates.

[Erich Dollansky]

Hi,

On Wed, 25 Apr 2018 00:21:31 +0100

Post by RW via freebsd-questions
On Tue, 24 Apr 2018 20:09:24 +0200

Post by Ralf Mardorf via freebsd-questions
Btw. an "audio CD" already provides not that good audio quality.
What you need for high end audio quality are 48 KHz,

I'd be astonished if you could tell the difference under double blind
conditions - even if you are a child.

despite knowing the theories behind 'digital' sound, I can tell you out
of own experience that there is a huge difference between 'analog'
sound and 'digital' when money is not an issue. I am not able to tell
you why 'digital' sound is so bad. A friend of mine has real good
'analog' sound equipment. After noticing that he has an original LP of
a CD of mine, we plugged normal digital equipment in. Every child could
hear the difference.

Erich

[Steve O'Hara-Smith]

On Wed, 25 Apr 2018 18:53:30 +0800

Post by Erich Dollansky
After noticing that he has an original LP of
a CD of mine, we plugged normal digital equipment in. Every child could
hear the difference.

There are substantial differences between what is recorded on an
LP and what is recorded on a CD even if they start from the same master
tape.

The audio for the LP will have been processed with RIAA (to be
reversed by the pre-amp on playback) while the audio for the CD will have
been filtered sharply to prevent aliasing.

Then the audio for the LP will be mastered onto the LP with some
care (possibly compression being applied) to ensure that the groove spacing
is adequate to the amplitude being recorded while the audio for the CD will
be digitised.

The playback pitch accuracy will depend on the rotation speed of the
turntable for the LP and for the readout speed of the CD mechanism for the
CD (which may or may not be buffered to prevent mechanical jitter*).

IOW they're bound to sound different - as to which is more accurate
you'd need to do a double blind test against the monitors the mixing
engineer used (at least I *think* that's the most valid reference).

*: I recall reading many years ago that some recording
engineers could tell which machine a digital recording was mastered on
because of the characteristic timing jitter imposed by the machine onto the
recording.

[Erich Dollansky]

Hi,

On Wed, 25 Apr 2018 12:12:49 +0100

Post by Steve O'Hara-Smith
On Wed, 25 Apr 2018 18:53:30 +0800

Post by Erich Dollansky
After noticing that he has an original LP of
a CD of mine, we plugged normal digital equipment in. Every child
could hear the difference.

There are substantial differences between what is recorded on
an LP and what is recorded on a CD even if they start from the same
master tape.

the LP was even purchased before CDs were even invented. So, real old
stuff with all the other noises a CD does not have.

Post by Steve O'Hara-Smith
The audio for the LP will have been processed with RIAA (to be
reversed by the pre-amp on playback) while the audio for the CD will
have been filtered sharply to prevent aliasing.
Then the audio for the LP will be mastered onto the LP with
some care (possibly compression being applied) to ensure that the
groove spacing is adequate to the amplitude being recorded while the
audio for the CD will be digitised.
The playback pitch accuracy will depend on the rotation speed
of the turntable for the LP and for the readout speed of the CD
mechanism for the CD (which may or may not be buffered to prevent
mechanical jitter*).
IOW they're bound to sound different - as to which is more
accurate you'd need to do a double blind test against the monitors
the mixing engineer used (at least I *think* that's the most valid
reference).
*: I recall reading many years ago that some recording
engineers could tell which machine a digital recording was mastered on
because of the characteristic timing jitter imposed by the machine
onto the recording.

There is not doubt for me.

Erich

[Ralf Mardorf via freebsd-questions]

However, if plain analog equipment does sound better than digital
equipment or not is off-topic. Still important is, that even digital
gear at some point converts from/to analog and apart from the quality
of the converters, the quality of the digital gear's analog domain does
matter a lot. If you want to experience a high class headphone amp,
listen to the headphone output of a RME card and compare it with
another good headphone amp, fed by a digital output of a the same RME
card or by a balanced or unbalanced analog output of the same card.
Usually the result is that clear, that there's no need for blind or
double blind tests.

[Ralf Mardorf via freebsd-questions]

Comparing the same album from different sources is tricky for several
reasons, already if you compare two LPs played by the same record
player or two CDs by played by the same CD player. Two LPs or CDs from
different editions done in the same year, without any remastering,
could sound different, caused by fabrication issues. Two LPs or two CDs
from different years could differ by e.g. reduced dynamic, caused by a
remastering with an insane amount of compression, to increase the
impression of loudness, it's known as the loudness war. The peaks might
have the same level, while passages with lower levels might get higher
levels, so there is less difference between a silent passage and a loud
passage of a recording. There are many other known issues, especially
if something was released as a record first and years later was
released as a CD. Theoretically a CD could sound better than a record,
but indeed, often records sound much better. Let alone that a very old
record might suffer from disgusting scratches, but still could be
played, while a very old CD might suffer from fatal data loss, so
playing the CD might be impossible and even recovering of the still
available data and burning the recovered data to a new CD might lead to
nothing.

[Ralf Mardorf via freebsd-questions]

OTOH storage of very good sounding professional analog tapes has got
pitfalls. Assumed professional analog recordings should sound better
than professional digital recordings, perhaps they don't, it's just an
arbitrary assumption, then we shouldn't underrate the issue of
maintaining the archiving. An analog copy does reduce the quality of
the original recording, there is nothing such as a perfect analog copy.
A digital copy doesn't reduce the quality, at least not a perfect
digital copy and perfect digital copies are possible. When coping from
one consumer DAT to another consumer DAT via S/PDIF I experienced
artefacts, but something like this shouldn't happen, if we copy a file
from one HDD to another ;).

[Waitman Gobble]

On Wed, Apr 25, 2018 at 5:31 PM, Ralf Mardorf via freebsd-questions

Post by Ralf Mardorf via freebsd-questions
Comparing the same album from different sources is tricky for several
reasons, already if you compare two LPs played by the same record
player or two CDs by played by the same CD player. Two LPs or CDs from
different editions done in the same year, without any remastering,
could sound different, caused by fabrication issues. Two LPs or two CDs
from different years could differ by e.g. reduced dynamic, caused by a
remastering with an insane amount of compression, to increase the
impression of loudness, it's known as the loudness war. The peaks might
have the same level, while passages with lower levels might get higher
levels, so there is less difference between a silent passage and a loud
passage of a recording. There are many other known issues, especially
if something was released as a record first and years later was
released as a CD. Theoretically a CD could sound better than a record,
but indeed, often records sound much better. Let alone that a very old
record might suffer from disgusting scratches, but still could be
played, while a very old CD might suffer from fatal data loss, so
playing the CD might be impossible and even recovering of the still
available data and burning the recovered data to a new CD might lead to
nothing.
_______________________________________________
https://lists.freebsd.org/mailman/listinfo/freebsd-questions

Hardly anything recorded in a studio the past 20 years (30?) is over
48KHz. The actual master digital recordings are not hi definition. You
can find modern classical music recordings with higher resolution. But
there are also great recordings that were originally analog (like up
until the 1990's) and digitized at 96/192 and you (well at least I)
can tell the difference.

(But there are also people who claim there is no difference in
shooting JPEG format photos compared to shooting RAW, they claim they
cannot see any difference when it's quite obvious there's more detail
in RAW)

If you want 96/192 on a pci card you'd need something several years
old, verify the chipset and use OSS on FreeBSD.

There are a couple good USB audio devices that do 96/192. I do not
think you can get 96/192 out of any "Creative" device on FreeBSD.

[Ralf Mardorf via freebsd-questions]

Post by Waitman Gobble
Hardly anything recorded in a studio the past 20 years (30?) is over
48KHz. The actual master digital recordings are not hi definition. You
can find modern classical music recordings with higher resolution. But
there are also great recordings that were originally analog (like up
until the 1990's) and digitized at 96/192 and you (well at least I)
can tell the difference.
(But there are also people who claim there is no difference in
shooting JPEG format photos compared to shooting RAW, they claim they
cannot see any difference when it's quite obvious there's more detail
in RAW)
If you want 96/192 on a pci card you'd need something several years
old, verify the chipset and use OSS on FreeBSD.
There are a couple good USB audio devices that do 96/192. I do not
think you can get 96/192 out of any "Creative" device on FreeBSD.

Any comparison with a lossy compression image file format is utter
nonsense. Apart from working around latency issues for PA systems 96 and
192 KHz gains you nothing. For listening 48 KHz 16 bit already would do
the job and for production 48 KHz 32 bit floating point should be used.
32 bit floating point doesn't provide a better audio quality, it just
makes production easier. For production usually uncompressed audio
formats are used, but usually not raw header-less PCM, I suspect most
common is WAV. However, regarding all the double-blind test
discussions, lossless compressed audio format are lossless and lossy
compressed audio format are lossy. It doesn't matter how many people
are able to hear a difference or if they should hear a difference to
claim that the culprit is a bad algorithm. To avoid any issue for any
person, using whatever algorithm is provided by the software, lossy
audio formats shouldn't be used.

[AikiZen]

Hi,
and thank all for your reply, i read with care.

I have this usb audio
card : https://www.thomann.de/fr/presonus_audiobox_usb_96.htm?ref=search_prv_3

It's the good card ?
In my home i have an kenwood audio amplifier with good speaker.

I will encode my CD under FLAC audio codec, it's the good way ?
Where good configuration for FLAC encode ?

Take care,
aiki.


Le Tue, 24 Apr 2018 20:09:24 +0200,

Post by Ralf Mardorf via freebsd-questions

It's a laptop, no possible to change sound card...

Most likely any elCheapo prosumer USB class compliant audio device
provides far better audio quality, than any onboard audio device.
https://www.thomann.de/de/focusrite_scarlett_2i2_2nd_gen.htm
It much likely sounds thousand times better, than an onbaord device.
https://www.thomann.de/de/rme_babyface_pro.htm
It for sure does sound thousand times better than a prosumer audio
device.
I don't know your definition of "audiophile", but if you want
professional sound you need to use a professional device, that apart
from the converters also provides a far better analog domain.
Btw. an "audio CD" already provides not that good audio quality. What
you need for high end audio quality are 48 KHz, btw. anything higher
than 48 KHz gains you nothing, to the contrary, not matching pass
filters could make the sound less good and at very high sample rates,
the energy does increase, this means the noise floor does increase.
_______________________________________________
https://lists.freebsd.org/mailman/listinfo/freebsd-questions
To unsubscribe, send any mail to

[Ralf Mardorf via freebsd-questions]

Post by AikiZen
I have this usb audio
card : https://www.thomann.de/fr/presonus_audiobox_usb_96.htm?ref=search_prv_3
It's the good card ?

I don't know this very cheap prosumer audio device. I once tested the
Presonus AudioBox 1818VSL. The 1818VSL seems to be a quite good usable
audio interface for home recording, but it's also a prosumer audio
device. It's quite possible to do professional recordings with a
prosumer audio interface, but with the exception, that you are unable
to estimate fine nuances. I compared the 1818VSL with the professional
RME card's IOs and with a prosumer ADA8000 IOs. I suspect an audiophile
person expects professional audio quality + esoteric humbug. Those
prosumer devices are good enough for most users, but they definitively
don't provide the audio quality of a RME card and this isn't esoteric
humbug.

[Erich Dollansky]

Hi,

On Wed, 25 Apr 2018 21:51:40 +0200

Post by AikiZen
Hi,
and thank all for your reply, i read with care.
I have this usb audio
card : https://www.thomann.de/fr/presonus_audiobox_usb_96.htm?ref=search_prv_3
It's the good card ?
In my home i have an kenwood audio amplifier with good speaker.
I will encode my CD under FLAC audio codec, it's the good way ?
Where good configuration for FLAC encode ?

as disk space is not a real concern, I leave the original file format
intact. FLAC is lossless but it is still an extra layer. When space is
a concern, yes, use it.

Anyway, you can do a simple test at home. Try it with some extreme
pieces you like. It will be your ears listening not ours.

Erich