11 Sep
2005
11 Sep
'05
8:54 p.m.
Hello all,
I am looking for some engineering know-how and advice. Should I track at
44.1 kHz, 48 kHz or 96 kHz?
I record vocals and acoustic guitar for the most part. I have been
routinely recording at 24 bits, 96 kHz simply because I can and I assume
that this gets me the highest fidelity from my recordings (tracking
through AKG Solidtube or a pair of Oktava condensers and a Pendulum
Audio SPS-1 preamp to Delta 1010LT into Jack/Ardour). After a discussion
with a friend, I´m not so sure anymore. He is recording his album at
44.1 kHz, arguing that this will avoid downsampling when finally
pressing a CD. So, are there any opinions about this?
There is a very practical issue here. I am contemplating the addition of
an outboard digital effects processor to take some load off my CPU. Most
units (less than $1000) with s/pdif are 44.1 and 48 kHz, so they would
not work (I don´t think) with digital audio recorded at 96 kHz.
Thanks for all the help you've given me in the past. I hope I am not
coming to the well too many times!!
-Joe Dell'Orfano
--
Joseph Dell'Orfano <fullgo(a)dellorfano.net>
11 Sep
11 Sep
11:32 p.m.
On 9/11/05, Joseph Dell'Orfano <fullgo(a)dellorfano.net> wrote:
Hello all,
I am looking for some engineering know-how and advice. Should I track at
44.1 kHz, 48 kHz or 96 kHz?
I record vocals and acoustic guitar for the most part. I have been
routinely recording at 24 bits, 96 kHz simply because I can and I assume
that this gets me the highest fidelity from my recordings (tracking
through AKG Solidtube or a pair of Oktava condensers and a Pendulum
Audio SPS-1 preamp to Delta 1010LT into Jack/Ardour). After a discussion
with a friend, I´m not so sure anymore. He is recording his album at
44.1 kHz, arguing that this will avoid downsampling when finally
pressing a CD. So, are there any opinions about this?
There is a very practical issue here. I am contemplating the addition of
an outboard digital effects processor to take some load off my CPU. Most
units (less than $1000) with s/pdif are 44.1 and 48 kHz, so they would
not work (I don´t think) with digital audio recorded at 96 kHz.
Thanks for all the help you've given me in the past. I hope I am not
coming to the well too many times!!
-Joe Dell'Orfano
--
Joseph Dell'Orfano <fullgo(a)dellorfano.net>
If you do blind listening tests and can tell that 96k consistantly
sounds better, go with 96k.
Quite honestly I've found that having a super accurate master word
clock (Big Ben for example) will benifit the sound quality more that
boosting the sampling rate. Especially if you're just going to down
sample to 44.1k anyway.
-Reuben
12 Sep
12 Sep
10:56 a.m.
On Sun, 2005-09-11 at 20:32 -0500, Reuben Martin wrote:
> I am looking for some engineering know-how and
advice. Should I track at
> 44.1 kHz, 48 kHz or 96 kHz?
I had just another thought...
In my last post, I suggested that you capture and process your work at
88.2, but it occurred to me that you're original question was more about
comparing lower and higher sample rates. Your friend who suggested that
you work directly at 44.1 may have a point, but it really depends on the
quality of gear you'll have access to.
Working at a higher sample rate is usually a good idea for several
reasons. First, cheaper converters (< $1,500) generally sound better at
higher sample rates due to extra filtering that goes on to reduce
artifacts. (Note: this is not the case for top quality "audiophile"
gear.)
Secondly, if you're planning to send your mixes to a mastering house,
then work at the highest sampling rate you possibly can and let them do
the conversion. Believe it or not, many of the ultra-high end mastering
studios bounce their final product to 1/2 inch two-track analog tape so
they can recapture it a final time at 44.1. If the digital gear is top
of the line and the tape machines are in great shape, this is probably
the highest fidelity down-sampling conversion process on the planet.
So the real question is this: what are you planning to do with your
final product?
Greg
www.steeprockmedia.com
12:41 p.m.
On Monday 12 September 2005 08:55 am, Greg Wilder wrote:
Believe it or not, many of the ultra-high end
mastering
studios bounce their final product to 1/2 inch two-track analog tape so
they can recapture it a final time at 44.1.
Interesting--just the approach I was thinking of as being among the easiest /
potentially best quality. (I mean, I can see all kinds of potential problems
doing the conversion "mathmatically" (sp?) within the computer.)
A similar approach might be to do a D to A and then an A to D within the
computer (e.g., soundcard output to soundcard input).
Randy Kramer
12:56 p.m.
Randy Kramer wrote:
On Monday 12 September 2005 08:55 am, Greg Wilder
wrote:
yes, i've thought about that, but if we're using jack it is not possible
to set different sampling rates in capture and playback simultaneously,
right?
.pltk.
Believe it or not, many of the ultra-high end
mastering
studios bounce their final product to 1/2 inch two-track analog tape so
they can recapture it a final time at 44.1.
Interesting--just the approach I was thinking of as being among the easiest /
potentially best quality. (I mean, I can see all kinds of potential problems
doing the conversion "mathmatically" (sp?) within the computer.)
A similar approach might be to do a D to A and then an A to D within the
computer (e.g., soundcard output to soundcard input).
3:38 p.m.
On Monday 12 September 2005 11:00 am, Peter Lutek wrote:
yes, i've thought about that, but if we're
using jack it is not possible
to set different sampling rates in capture and playback simultaneously,
right?
Sorry, I don't know. Hopefully somebody else will chime in.
Randy Kramer
3:45 p.m.
On Mon, 2005-09-12 at 12:49 -0400, Randy Kramer wrote:
On Monday 12 September 2005 11:00 am, Peter Lutek
wrote:
Correct, it's not possible.
Lee
yes, i've thought about that, but if
we're using jack it is not possible
to set different sampling rates in capture and playback simultaneously,
right?
Sorry, I don't know. Hopefully somebody else will chime in.
1:13 p.m.
On Mon, 2005-09-12 at 10:23 -0400, Randy Kramer wrote:
On Monday 12 September 2005 08:55 am, Greg Wilder
wrote:
Yes, but you'd really have to have amazing cards to hear a real benefit.
Bouncing a final mix to 1/2 inch tape before the final print allows you
to do the 44.1 conversion from a high quality analog source. But, it
assumes that the source recordings and processing was done at an
extremely high level - i.e. DSD. And unfortunately, we don't yet have
any open-source tools for manipulating DSD. I know of a few mastering
houses that would *kill* to be using GNU/Linux if the tools were
available.
Greg
www.steeprockmedia.com
Believe it or not, many of the ultra-high end
mastering
studios bounce their final product to 1/2 inch two-track analog tape so
they can recapture it a final time at 44.1.
A similar approach might be to do a D to A and then an A to D within the
computer (e.g., soundcard output to soundcard input). 
1:31 p.m.
Bouncing a final mix to 1/2 inch tape before the final
print allows you
to do the 44.1 conversion from a high quality analog source. But, it
assumes that the source recordings and processing was done at an
extremely high level - i.e. DSD. And unfortunately, we don't yet have
any open-source tools for manipulating DSD. I know of a few mastering
houses that would *kill* to be using GNU/Linux if the tools were
available.
there is a still quite a lot of controversy over whether or not Sony's 1
bit/MHz clock design actually *is* better than more "traditional"
approaches. i know there is a lot of good marketing to explain why its
so cool, but that doesn't seem to be backed up by any substantive double
blind testing. even the math analysis is heavy on marketing bullshit.
2:17 p.m.
New subject: [linux-audio-user] [OT] DSD +/- [was:44.1, 48 or 96?]
On Mon, 2005-09-12 at 11:32 -0400, Paul Davis wrote:
Hmm - I didn't know that. I'm no DSD expert, but I understand that
there are some clear drawbacks. For example, the DSD process introduces
high levels of ultrasonic noise which can cause problems with certain
playback equipment, and it is very troublesome to attempt DSP with pure
DSD signals. But if you're trying to accurately capture a "real-world"
performance - it certainly is a strong contender.
Cutting edge technology formats often attract controversy for many
reasons. That said, many of the (admittedly few) facilities that
specialize in audiophile authoring (both SACD and DVD-A) rely on DSD
basic capture and editing tools. Speaking as a format agnostic, I think
it would be tough to argue that it doesn't sound damn good.
Greg
www.steeprockmedia.com
Bouncing a
final mix to 1/2 inch tape before the final print allows you
to do the 44.1 conversion from a high quality analog source. But, it
assumes that the source recordings and processing was done at an
extremely high level - i.e. DSD. And unfortunately, we don't yet have
any open-source tools for manipulating DSD. I know of a few mastering
houses that would *kill* to be using GNU/Linux if the tools were
available.
there is a still quite a lot of controversy over whether or not Sony's 1
bit/MHz clock design actually *is* better than more "traditional"
approaches. i know there is a lot of good marketing to explain why its
so cool, but that doesn't seem to be backed up by any substantive double
blind testing. even the math analysis is heavy on marketing bullshit. 
3:34 p.m.
New subject: [linux-audio-user] [OT] DSD +/- [was:44.1, 48 or 96?]
On Mon, Sep 12, 2005 at 12:16:46 -0400, Greg Wilder wrote:
Hum. I'm certainly no expert on this, but my impression is that the DSD
format is just an encoding of the intermediate signal produced by decent
ADCs before conversion into PCM and used in "digital amplifiers". The
advantage being that you save a conversion step in both directions - if
your source and final output are both analogue.
If you have to convert to PCM in either direction (say, conventional
digital processing in ardour or for output over SPDIF) then the advantge
would be wiped out in at least that direction.
It's not really new for what its worth, underneath its the same modulation
system that was used to encode laserdiscs.
- Steve
there is a
still quite a lot of controversy over whether or not Sony's 1
bit/MHz clock design actually *is* better than more "traditional"
approaches. i know there is a lot of good marketing to explain why its
so cool, but that doesn't seem to be backed up by any substantive double
blind testing. even the math analysis is heavy on marketing bullshit.
Hmm - I didn't know that. I'm no DSD expert, but I understand that
there are some clear drawbacks. For example, the DSD process introduces
high levels of ultrasonic noise which can cause problems with certain
playback equipment, and it is very troublesome to attempt DSP with pure
DSD signals. But if you're trying to accurately capture a "real-world"
performance - it certainly is a strong contender.
15 Sep
15 Sep
2:21 a.m.
Thanks to all who answered this thread. I am privileged to get to learn
from everybody's experience!
-Joe D
On Mon, 2005-09-12 at 11:32 -0400, Paul Davis wrote:
--
Joseph Dell'Orfano <fullgo(a)dellorfano.net>
Bouncing a
final mix to 1/2 inch tape before the final print allows you
to do the 44.1 conversion from a high quality analog source. But, it
assumes that the source recordings and processing was done at an
extremely high level - i.e. DSD. And unfortunately, we don't yet have
any open-source tools for manipulating DSD. I know of a few mastering
houses that would *kill* to be using GNU/Linux if the tools were
available.
there is a still quite a lot of controversy over whether or not Sony's 1
bit/MHz clock design actually *is* better than more "traditional"
approaches. i know there is a lot of good marketing to explain why its
so cool, but that doesn't seem to be backed up by any substantive double
blind testing. even the math analysis is heavy on marketing bullshit.
12 Sep
12 Sep
5:31 p.m.
On Mon, Sep 12, 2005 at 11:12:25AM -0400, Greg Wilder wrote:
Bouncing a final mix to 1/2 inch tape before the final
print allows you
to do the 44.1 conversion from a high quality analog source. But, it
assumes that the source recordings and processing was done at an
extremely high level - i.e. DSD. And unfortunately, we don't yet have
any open-source tools for manipulating DSD. I know of a few mastering
houses that would *kill* to be using GNU/Linux if the tools were
available.
Is there even any readily available DSD hardware? I'm aware of some
Sony prototype gear, an editing system called Pyramix, and Sonic Studio,
and I don't imagine that any of those would sell just the hardware or
open up the specs of the hardware for writing drivers.
Additionally, is there even a standard yet that would be the DSD
equivelent of AES or S/PDIF?
--
Joshua D. Boyd
jdboyd(a)jdboyd.net
http://www.jdboyd.net/
http://www.joshuaboyd.org/
13 Sep
13 Sep
10:55 a.m.
On Mon, 2005-09-12 at 15:16 -0400, Joshua Boyd wrote:
Is there even any readily available DSD hardware?
There are many DSD converters out there - the Prism Sound Dream ADA-8XR
has a FireWire interface option and a few other systems are available
with PCI cards.
I don't imagine that any of those would sell just
the hardware or
open up the specs of the hardware for writing drivers.
I've no idea.
Additionally, is there even a standard yet that would
be the DSD
equivelent of AES or S/PDIF?
Sony has SDIF-3, and I think other manufacturers are using mlan and
something called MAC-DSD.
12 Sep
12 Sep
10 a.m.
On Sun, 2005-09-11 at 18:54 -0400, Joseph Dell'Orfano wrote:
After a discussion
with a friend, I´m not so sure anymore. He is recording his album at
44.1 kHz, arguing that this will avoid downsampling when finally
pressing a CD. So, are there any opinions about this?
The quality of the dithering algorithms can make very a noticeable
difference in your final product if you have to down-sample - but that
doesn't necessarily mean that you're relegated to life at 44.1.
IMO, the best solution is to work at 88.2khz whenever possible. This
way, you can down-sample to CD quality without quantization errors. I'm
no DSP guru - but as I understand it, sampling at a multiple of 44.1
(88.2, 176.4, etc.) is the cleanest way to go outside of DSD.
FWIW - I haven't heard a software dither (with quantization) that I
thought was decent - FOSS or otherwise. The best down-sampling I ever
heard came out of a $3,500 Apogee box.
Hope this helps,
Greg
www.steeprockmedia.com
10:26 a.m.
On Mon, 12 Sep, 2005 at 07:59AM -0400, Greg Wilder spake thus:
On Sun, 2005-09-11 at 18:54 -0400, Joseph
Dell'Orfano wrote:
Wow. How hard can this be? Surely it's just like image scaling, but
without that complicated other dimension.
If anyone want's to put me right...
After a discussion
with a friend, I´m not so sure anymore. He is recording his album at
44.1 kHz, arguing that this will avoid downsampling when finally
pressing a CD. So, are there any opinions about this?
The quality of the dithering algorithms can make very a noticeable
difference in your final product if you have to down-sample - but that
doesn't necessarily mean that you're relegated to life at 44.1.
IMO, the best solution is to work at 88.2khz whenever possible. This
way, you can down-sample to CD quality without quantization errors. I'm
no DSP guru - but as I understand it, sampling at a multiple of 44.1
(88.2, 176.4, etc.) is the cleanest way to go outside of DSD.
FWIW - I haven't heard a software dither (with quantization) that I
thought was decent - FOSS or otherwise. The best down-sampling I ever
heard came out of a $3,500 Apogee box. Hope this helps,
Greg
www.steeprockmedia.com
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated
Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)
10:59 a.m.
On Mon, 2005-09-12 at 13:30 +0100, james(a)dis-dot-dat.net wrote:
On Mon, 12 Sep, 2005 at 07:59AM -0400, Greg Wilder
spake thus:
> On Sun, 2005-09-11 at 18:54 -0400, Joseph Dell'Orfano wrote:
> > After a discussion
> > with a friend, I´m not so sure anymore. He is recording his album at
> > 44.1 kHz, arguing that this will avoid downsampling when finally
> > pressing a CD. So, are there any opinions about this?
>
> The quality of the dithering algorithms can make very a noticeable
> difference in your final product if you have to down-sample - but that
> doesn't necessarily mean that you're relegated to life at 44.1.
>
> IMO, the best solution is to work at 88.2khz whenever possible. This
> way, you can down-sample to CD quality without quantization errors. I'm
> no DSP guru - but as I understand it, sampling at a multiple of 44.1
> (88.2, 176.4, etc.) is the cleanest way to go outside of DSD.
in my understanding, dithering has nothing to do with downsampling.
dithering is done to reduce artifacts from sample bit width reduction,
not from sample rate reduction. sample rate changes can be handled by a
variety of interpolation techniques, which exist in a plane defined by
two axes: quality and CPU cycles per sample. you can do cheap linear
interpolation and get a pretty unpleasant effect, or use more cycles for
a better result. dithering comes in its own varieties, but they are not
arranged on a simple 2D axis the way that interpolation is. AFAIK, so-
called "pow-f" dithering is spoken of the most highly at present, but
relies on a proprietary (patented, too, i think) algorithm.
consequently, yes there are good reasons to record at 88.2 if the final
SR will be CD quality at 44.1 because of the ability to almost entirely
avoid interpolation when downsampling. but this has nothing to do with
dithering, which handles the sample bit width change from (presumably)
24 down to 16. if done naively (e.g.
float twenty_four_bit_sample;
short sixteen_bit_sample = (short) twenty_four_bit_sample;
various noisy artifacts end up in the signal. dithering is a way to
reduce their audible effect.
--p
12:04 p.m.
On Mon, 2005-09-12 at 09:00 -0400, Paul Davis wrote:
in my understanding, dithering has nothing to do with
downsampling.
Yup - my bad. I was confusing terms. Thanks for the correction.
sample rate changes can be handled by a
variety of interpolation techniques, which exist in a plane defined by
two axes: quality and CPU cycles per sample. you can do cheap linear
interpolation and get a pretty unpleasant effect, or use more cycles for
a better result.
Thanks for stepping in and clearing this up. Interpolation is the word
I was originally looking for. Too many uncaffeinated brain cells too
early in the morn'
dithering comes in its own varieties, but they are
not
arranged on a simple 2D axis the way that interpolation is. AFAIK, so-
called "pow-f" dithering is spoken of the most highly at present, but
relies on a proprietary (patented, too, i think) algorithm.
Very interesting stuff. Thanks for the help.
Greg
11:18 a.m.
Greg Wilder <greg(a)gregwilder.com> writes:
CD quality
It's a conspiracy;) (44.1kHz)
The best down-sampling I ever heard came out of a
$3,500 Apogee box.
I didn't know they shipped resampling units.
OT: Apogee uses analog devices chips for AD/DA which are great. I
intend to sell my Multiface II and buy one, but I'd never use a
proprietary resampling unit.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
12:04 p.m.
On Mon, 2005-09-12 at 17:17 +0200, Esben Stien wrote:
You're right, they don't. I meant that the best resampling result I've
ever heard came from an Apogee converter that recaptured a higher
resolution signal at 44.1. Sorry for being unclear.
Best,
Greg
www.steeprockmedia.com
The best
down-sampling I ever heard came out of a $3,500 Apogee box.
I didn't know they shipped resampling units.
19 Sep
19 Sep
10:40 p.m.
Esben Stien <b0ef(a)esben-stien.name> writes:
I didn't know they shipped resampling units.
Hmm, the rosetta 200 comes with a SRC. I would never use it though,
over the secret rabbit code;).
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
12 Sep
12 Sep
11:03 a.m.
"Joseph Dell'Orfano" <fullgo(a)dellorfano.net> writes:
I am looking for some engineering know-how and advice.
Then you should check out:
irc://irc.homelien.no/%23musicdsp (efnet) and it's mailinglist music-dsp.
Also check out the rec.audio.pro newsgroup.
Many enthusiasts and recording engineers there
Should I track at 44.1 kHz, 48 kHz or 96 kHz?
The higher the better. Even with 192kHz you're not creating a correct
digital representation of the analog signal. Nyquist says 2 times the
highest frequency, but this still has some problems.
recording his album at 44.1 kHz, arguing that this
will avoid
downsampling when finally pressing a CD.
Sure, but this shouldn't stop you creating the best possible fidelity
in your recordings.
I am contemplating the addition of an outboard digital
effects
processor to take some load off my CPU.
Not advisable; rather invest in processing power where you can use
free software algorithms.
Most units (less than $1000) with s/pdif are 44.1 and
48 kHz, so
they would not work (I don´t think) with digital audio recorded at
96 kHz.
Another reason to stay away.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
7007
days inactive
7015
days old
linux-audio-user@lists.linuxaudio.org
21 comments
11 participants
participants (11)
-
Esben Stien -
Greg Wilder -
james＠dis-dot-dat.net -
Joseph Dell'Orfano -
Joshua Boyd -
Lee Revell -
Paul Davis -
Peter Lutek -
Randy Kramer -
Reuben Martin -
Steve Harris