Can anyone tell me what the difference is between sound cards and converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything sounds
thin. I guy at sam ash told me that the converters in the delta were crappie
but could not suggest a fix for my problem. so my question is do I need to
purchase a better converter or a better sound card or are they the same
thing differnt name.

Can anyone tell me what the difference is between sound cards and
converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything sounds
thin. I guy at sam ash told me that the converters in the delta were
crappie
but could not suggest a fix for my problem. so my question is do I need to
purchase a better converter or a better sound card or are they the same
thing differnt name.

It's much more likely a matter of your mic(s) and mic preamp(s).

"Sugarite" wrote in message ...
Can anyone tell me what the difference is between sound cards and
converters

A sound card is simply a thing that gets audio to and from your computer.
Converters do just what their name implies -- convert analog audio into
digital data.

, im currently running a delta 1010

Pretty good sound card.

thin. I guy at sam ash told me that the converters in the delta were
crappie

Most Sam Ash employees aren't even musicians or involved with pro audio.
They wouldn't know what a good converter was. He was trying to sell you
something else with "good" converters.

purchase a better converter or a better sound card or are they the same
thing differnt name.

The converters are in the sound cards. To upgrade to better converters,
yes, you could buy a new sound card. Or you could use external A/D
converters. They aren't cheap and your 1010 is not the problem (unless of
course it's broken which I doubt).

It's much more likely a matter of your mic(s) and mic preamp(s).

Agreed. Tell us what other gear you're using.

Digital Straits wrote:

Can anyone tell me what the difference is between sound cards and converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything sounds
thin.

That could be any of a variety of problems... from bad sounding conversion
processes to a mediocre to poor signal chain... garbage microphones... lack of
skills to achieve the desired results [buying a Stratocaster® doesn't make you
Eric Clapton overnight]... etc.

I guy at sam ash told me that the converters in the delta were crappie
but could not suggest a fix for my problem. so my question is do I need to
purchase a better converter or a better sound card or are they the same
thing differnt name.

They're similar things... you need a 'soundcard' to interface with your
computer... as long as the 'soundcard' accepts digital inputs and has digital
outputs you can employ superior sounding "outboard converters" which should net
you superior results.

It's mostly a question of "need" [like if you're doing 'product' vs. doing
'writing demos'... 'product' kinda demands better quality tools than 'writing
demos' to assist in translating the emotion of the musical statement(s) where as
'writing demos' are just about getting ideas recorded quickly and easily so you
have your musical ideas sketched out prior to creating the 'product level
presentation of your music].

Best of luck with it.
--
Fletcher
Mercenary Audio
TEL: 508-543-0069
FAX: 508-543-9670
http://www.mercenary.com
"this is not a problem"

Another improvement can be made with cards that have word clock sync- which
is the addition of a high quality clock generator. I use a Lucid Genex96
with my RME Multiface.

Lot's of good info at www.digido.com - and his book is worth owning as well.

C

"Thomas Bishop" wrote in message
. ..
"Sugarite" wrote in message ...
Can anyone tell me what the difference is between sound cards and
converters

A sound card is simply a thing that gets audio to and from your computer.
Converters do just what their name implies -- convert analog audio into
digital data.

, im currently running a delta 1010

Pretty good sound card.

thin. I guy at sam ash told me that the converters in the delta were
crappie

Most Sam Ash employees aren't even musicians or involved with pro audio.
They wouldn't know what a good converter was. He was trying to sell you
something else with "good" converters.

purchase a better converter or a better sound card or are they the
same
thing differnt name.

The converters are in the sound cards. To upgrade to better converters,
yes, you could buy a new sound card. Or you could use external A/D
converters. They aren't cheap and your 1010 is not the problem (unless of
course it's broken which I doubt).

It's much more likely a matter of your mic(s) and mic preamp(s).

Agreed. Tell us what other gear you're using.

Om_Audio wrote:

Another improvement can be made with cards that have word clock sync- which
is the addition of a high quality clock generator. I use a Lucid Genex96
with my RME Multiface.

At the risk of rekindling an old debate that I've been fortunate enough
to ignore, I have to ask what makes people think the designers of a
digital circuit would fail to give due consideration to the internal
clock generation, but somehow excel at providing the means to cleanly
and accurately handle an incoming clock signal? Sorry, but it seems to
me that the clock itself is the easy part. Getting an accurate timing
signal out of one box and into another without messing it up has to be
a lot more difficult. And common sense dictates that a product
designer devote more attention to the former than the latter.

Am I wrong, Dude?

ulysses

On Sat, 30 Aug 2003 20:00:28 -0500, "Digital Straits"
wrote:

Can anyone tell me what the difference is between sound cards and
converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything
sounds
thin. I guy at sam ash told me that the converters in the delta were
crappie
but could not suggest a fix for my problem. so my question is do I
need to
purchase a better converter or a better sound card or are they the
same
thing differnt name.



check out http://www.lynxstudio.com/index.htm
look at the lynx 2 or L22, all depending on your i/o needs.
digital master

"Digital Master" wrote
Can anyone tell me what the difference is between sound cards and
converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything
sounds thin.

Soundcards have convertors built in.
Dedicated convertors may be better quality just as dedicated preamps may be
better than the ones in your mixing console.
The Delta 1010 has a very good reputation so I would look first at the
microphone, pre-amp, acoustics etc. etc.
You may also think that your sound is thin because it hasn't been compressed
or doubled etc.
These will all have a much, much bigger effect than the quality of your
convertors.

I guy at sam ash told me that the converters in the delta were
crappie

Don't ever trust a salesman. They tend to think that everything is crap
except for the equipment which they are trying to sell you. Most salesmen
have never even tried the stuff they sell, no matter how much they tell you
that this is what they have at home.

--
Anthony Gosnell

to reply remove nospam.

On Sat, 11 Oct 2003 12:39:38 +0200, "anthony.gosnell"
wrote:

"Digital Master" wrote
Can anyone tell me what the difference is between sound cards and
converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything
sounds thin.

Soundcards have convertors built in.
Dedicated convertors may be better quality just as dedicated preamps
may be
better than the ones in your mixing console.
The Delta 1010 has a very good reputation so I would look first at the
microphone, pre-amp, acoustics etc. etc.
You may also think that your sound is thin because it hasn't been
compressed
or doubled etc.
These will all have a much, much bigger effect than the quality of
your
convertors.

I guy at sam ash told me that the converters in the delta were
crappie

Don't ever trust a salesman. They tend to think that everything is
crap
except for the equipment which they are trying to sell you. Most
salesmen
have never even tried the stuff they sell, no matter how much they
tell you
that this is what they have at home.







how the hell did you screw this msg up?
please refer to orginal.

Converters....a subject very much argued and debated about.

The A/D stage of your recording is the most crucial part and don't let
anyone tell you differently.

Remember that the converter itself is only one component of the entire
scheme. It's the circuitry around the
converter and the power supply as well that makes it even better.

Anyone can slap a nice AKM or Crystal converter in a box and say that it's
good. What about the Op Amp or
the series of caps and resistors around it? If those are low grade with lots
of DC offset/low THD readings than
you're getting lame quality conversion with no stereo imaging. This is what
leads to thin sounding recordings.

Granted as long as you know what you are doing in terms of recording
techniques then you can make anything
sound good.

Also a factor into good recording is the clock circuitry within the design.
Jitter and a very rampant fast spiking clock
design can make your sound files full of unwanted noise and artifacts. Also
make sure to always record at 24 bits to get the full
bandwidth of the signal. Better resolution makes a better picture.

If you want cheap, buy cheap. Just remember you get what you paid for.

Opus


"Digital Master" wrote in message
...
On Sat, 11 Oct 2003 12:39:38 +0200, "anthony.gosnell"
wrote:

"Digital Master" wrote
Can anyone tell me what the difference is between sound cards and
converters
, im currently running a delta 1010
for sound but im not getting the sound im looking for everything
sounds thin.

Soundcards have convertors built in.
Dedicated convertors may be better quality just as dedicated preamps
may be
better than the ones in your mixing console.
The Delta 1010 has a very good reputation so I would look first at the
microphone, pre-amp, acoustics etc. etc.
You may also think that your sound is thin because it hasn't been
compressed
or doubled etc.
These will all have a much, much bigger effect than the quality of
your
convertors.

I guy at sam ash told me that the converters in the delta were
crappie

Don't ever trust a salesman. They tend to think that everything is
crap
except for the equipment which they are trying to sell you. Most
salesmen
have never even tried the stuff they sell, no matter how much they
tell you
that this is what they have at home.







how the hell did you screw this msg up?
please refer to orginal.

"opus" wrote in message
. net
Converters....a subject very much argued and debated about.

The A/D stage of your recording is the most crucial part and don't let
anyone tell you differently.

I'll baldly tell you that the mic, room, your recording technique and the
musician are all more important than the difference between the converters
of a Delta 1010 and an ideal, stone-perfect converter.

Do you really want to claim something different?

Boldly, yes. As an Apogee Electronics support/technician I will tell you
that the A/D stage is the most crucial
part of the "digital' recording stage.

If you read my reply again you will notice I did say this:
"Granted as long as you know what you are doing in terms of recording
techniques then you can make anything
sound good."

Opus
"Arny Krueger" wrote in message
...
"opus" wrote in message
. net
Converters....a subject very much argued and debated about.

The A/D stage of your recording is the most crucial part and don't let
anyone tell you differently.

I'll baldly tell you that the mic, room, your recording technique and the
musician are all more important than the difference between the converters
of a Delta 1010 and an ideal, stone-perfect converter.

Do you really want to claim something different?

From: "opus"
Also make sure to always record at 24 bits to get the full
bandwidth of the signal.

The recorded word size has absolutely nothing to do with bandwidth. Sample rate
does.

Granted as long as you know what you are doing in terms of recording
techniques then you can make anything
sound good.

Really? I guess that depends upon your concept of "good."

Stuart Welwood
http://members.aol.com/StuWelwood

Stuart

You must look at the "whole" picture here. Yes, sample rate has a lot to do
with bandwidth "BUT" with only
16 bit operation you are not getting a smooth a snapshot as with 24 bit.
Believe me, working for Apogee
we study these aspects through scopes and AP systems.

The sample rate is only a "part" of the entire scheme of things. Also with
24 bit recordings you don't need to get
as much signal in as you did with 16 bit machines to get the total digital
bandwidth as needed.

I have some material at work that can truly enlighten us to the reason for
this. I will forward it to my home and post it here
for everyone to see.

Until then, I speak what the studies have found.

Opus
"StuWelwood" wrote in message
...
From: "opus"
Also make sure to always record at 24 bits to get the full
bandwidth of the signal.

The recorded word size has absolutely nothing to do with bandwidth. Sample
rate
does.

Granted as long as you know what you are doing in terms of recording
techniques then you can make anything
sound good.

Really? I guess that depends upon your concept of "good."

Stuart Welwood
http://members.aol.com/StuWelwood

On Sun, 12 Oct 2003 17:02:45 GMT, "opus" wrote:

You must look at the "whole" picture here. Yes, sample rate has a lot to do
with bandwidth "BUT" with only 16 bit operation you are not getting a smooth
a snapshot as with 24 bit.

Are you refering to stairways effect? I think I've heard something
about that, but couldn't understand. Is 24bit sample rate to help our
drummer, who plays fills and rolls that sound just like boxes
stumbling dow the stairs, to get more warmth in his groove.

... working for Apogee
we study these aspects through scopes and AP systems.

What's AP system.

The sample rate is only a "part" of the entire scheme of things.

What else is there?

Also with 24 bit recordings you don't need to get
as much signal in as you did with 16 bit machines to get the total digital
bandwidth as needed.

You mean I can play only a part of my song and the rest will be
guessed by 24bit sample rate?

I have some material at work that can truly enlighten us to the reason for
this. I will forward it to my home and post it here
for everyone to see.

Please, by all means.

Vladan
www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

You must look at the "whole" picture here. Yes, sample rate has a lot to
do
with bandwidth "BUT" with only 16 bit operation you are not getting a
smooth
a snapshot as with 24 bit.

Are you refering to stairways effect? I think I've heard something
about that, but couldn't understand. Is 24bit sample rate to help our
drummer, who plays fills and rolls that sound just like boxes
stumbling dow the stairs, to get more warmth in his groove.

Yes and No. The stairway effect is more on the sample resolution on how the
samples are taken. The lower the sample rate the more block shaped it is.
The higher the sampling
rate the more smoother it is. It has nothing to do with performance of the
musician and the groove.


... working for Apogee
we study these aspects through scopes and AP systems.

What's AP system.

Audio Precision machine. Machine for DPS analyzing and THD recording.

The sample rate is only a "part" of the entire scheme of things.

What else is there?

Bit depth and sampling rates are the two major factors to the digital realm

Also with 24 bit recordings you don't need to get
as much signal in as you did with 16 bit machines to get the total
digital
bandwidth as needed.

You mean I can play only a part of my song and the rest will be
guessed by 24bit sample rate?

No. With 16 bit converters you had to get the hottest signal possible to
truly take
advantage of the digital bandwidth. With 24 bit converters it's not
necessary. Obviously you still want a hot
signal but you don't have to with 24 bit converters. Hope that makes sense.

I have some material at work that can truly enlighten us to the reason
for
this. I will forward it to my home and post it here
for everyone to see.

Please, by all means.

Once I post that it will help clear up any confusion and enlighten us all

Opus

www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

On Sun, 12 Oct 2003 19:57:39 GMT, "opus" wrote:


Yes and No. The stairway effect is more on the sample resolution on how the
samples are taken. The lower the sample rate the more block shaped it is.
The higher the sampling
rate the more smoother it is.

Does it mean drivers in my speakers won't abruptly jump forward and
backward any more, but rather swing smoothlly?

What's AP system.

Audio Precision machine. Machine for DPS analyzing and THD recording.

Where can I get one? I want my recordings THD.

No. With 16 bit converters you had to get the hottest signal possible to
truly take
advantage of the digital bandwidth. With 24 bit converters it's not
necessary. Obviously you still want a hot
signal but you don't have to with 24 bit converters. Hope that makes sense.

Sorry, but I don't know what is hot and what is not.

Once I post that it will help clear up any confusion and enlighten us all

Please. ASAP.
Vladan
www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

opus wrote:

I have some material at work that can truly enlighten us to the reason for
this. I will forward it to my home and post it here
for everyone to see.

Until then, I speak what the studies have found.

Looking forward. Signal processing theory tells us that the
effects of sample rate and word width are orthogonal, the
former affecting only bandwidth and the latter only
quantization error (noise.) If they overlap due to deeper
considerations I'd love to gain an understanding.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob Cain wrote:

opus wrote:

I have some material at work that can truly enlighten us to the reason for
this. I will forward it to my home and post it here
for everyone to see.

Until then, I speak what the studies have found.

Looking forward. Signal processing theory tells us that the
effects of sample rate and word width are orthogonal, the
former affecting only bandwidth and the latter only
quantization error (noise.) If they overlap due to deeper
considerations I'd love to gain an understanding.

Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.

And you do know very well better than to ask me any questions about
that... g

--
ha

I hope everybody following this thread will take a close look at the
extremely insightful comments made by Opus. I have edited together a
few of the most illustrative statements below for your convenience. He
has told me everything I needed to know about Apogee products.

opus wrote:

Boldly, yes. As an Apogee Electronics support/technician I will tell
you that the A/D stage is the most crucial part of the "digital'
recording stage.

The A/D stage of your recording is the most crucial part and don't let
anyone tell you differently.

What about the Op Amp or the series of caps and resistors around it?
If those are low grade with lots of DC offset/low THD readings than
you're getting lame quality conversion with no stereo imaging. This
is what leads to thin sounding recordings.

Also a factor into good recording is the clock circuitry within the
design. Jitter and a very rampant fast spiking clock design can make
your sound files full of unwanted noise and artifacts. Also make sure
to always record at 24 bits to get the full bandwidth of the signal.
Better resolution makes a better picture.

You must look at the "whole" picture here. Yes, sample rate has a lot
to do with bandwidth "BUT" with only 16 bit operation you are not
getting a smooth a snapshot as with 24 bit. Believe me, working for
Apogee we study these aspects through scopes and AP systems.

Yes and No. The stairway effect is more on the sample resolution on
how the samples are taken. The lower the sample rate the more block
shaped it is. The higher the sampling rate the more smoother it is.
It has nothing to do with performance of the musician and the groove.

Wow. I am literarily amazed. Seriously. Is every Apogee Electronics
support/technician as knowledgeable and informed as you are? Did they
ask you personally to come to this forum and help us to understand
digital audio? I'm really impressed by what you have said above. I
don't even know what to say. I have no previous direct experience with
Apogee products, but your input to this forum has absolutely
illustrated Apogee's level of dedication to audio and has made my
purchasing decision for me. I truly do hope the company is aware of
the impact you're having on their image and future sales. Seriously.


ulysses

"opus" wrote in message
news:gBeib.744074$Ho3.182873@sccrnsc03

"opus" wrote in message
t.net

The A/D stage of your recording is the most crucial part and don't let
anyone tell you differently.

As an Apogee Electronics support/technician I will tell
you that the A/D stage is the most crucial part of the "digital'
recording stage.

Still not right. Wanna try again?

LeBaron & Alrich wrote:

Looking forward. Signal processing theory tells us that the
effects of sample rate and word width are orthogonal, the
former affecting only bandwidth and the latter only
quantization error (noise.) If they overlap due to deeper
considerations I'd love to gain an understanding.

Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.


That sounds seductively intuitive so to be sure I looked up
the derivation of quantization noise in "Digital Signal
Processing" by Proakis and Manolakis and the stastical
analyisis that leads to the usual 6 dB SNR per conversion
bit is independant of sample rate. (It's good to check our
assumptions now and again.)

And you do know very well better than to ask me any questions about
that... g

:-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob Cain wrote:


Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.


That sounds seductively intuitive so to be sure I looked up
the derivation of quantization noise in "Digital Signal
Processing" by Proakis and Manolakis and the stastical
analyisis that leads to the usual 6 dB SNR per conversion
bit is independant of sample rate. (It's good to check our
assumptions now and again.)


Du'oh. I fired that off without enough thought. What you
said sounds seductively intuitive because it's right. If
you double the sample rate to twice the audio band then the
_total_ RMS noise only be a function of the converter width
but the noise will be evenly distributed over twice as wide
a spectrum, half of which you can't hear, so that the
audible noise will be reduced by half.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob Cain wrote:

Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.

That sounds seductively intuitive so to be sure I looked up
the derivation of quantization noise in "Digital Signal
Processing" by Proakis and Manolakis and the stastical
analyisis that leads to the usual 6 dB SNR per conversion
bit is independant of sample rate. (It's good to check our
assumptions now and again.)

So tell me what you just said, please. It's a wrong thing or a right
thing? (Sorry to be so dense, you I doubt you're surprised!)

--
ha

On Tue, 14 Oct 2003 04:16:13 GMT, (LeBaron &
Alrich) wrote:

Bob Cain wrote:

Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.

I think that's related to shaping the dither noise; broader the
spectrum, further from the hearing range you can place it, or boost
it, so it's at lowest level within hearing range.

Vladan
www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

"Bob Cain" wrote in message

Bob Cain wrote:


Geez, you'll know I'm where I don't belong when I input this, but I
read some time ago, and I think it came from someone at Waves, that
extending the sample rate resulted in something like incresing bit
depth because conversion involved a fixed amount of quantization
noise that wound up fit into the resulting bandwidth. Extending
that bandwidth with a greater sample rate was said to spread the
same amount of quantization noise over a broader spectrum,
resulting in a lower noise floor, as if the bit depth had been
increased.


That sounds seductively intuitive so to be sure I looked up
the derivation of quantization noise in "Digital Signal
Processing" by Proakis and Manolakis and the stastical
analysis that leads to the usual 6 dB SNR per conversion
bit is independent of sample rate. (It's good to check our
assumptions now and again.)


Du'oh. I fired that off without enough thought. What you
said sounds seductively intuitive because it's right. If
you double the sample rate to twice the audio band then the
_total_ RMS noise only be a function of the converter width
but the noise will be evenly distributed over twice as wide
a spectrum, half of which you can't hear, so that the
audible noise will be reduced by half.

More specifically, all other things being equal, if you double the converter
bandwidth, noise goes up by the inverse of the square root of two.

If the noise spectrum is flat (like white noise) then the noise in the lower
half is related to total noise by the square root of two, so you break even.

When the market transition from 48 KHz to 96 KHz converters took place all
things weren't equal. Happily, there were significant technical advances on
several fronts. This piled on top of converter performance in the top cards
from the previous generation (e.g. Echo Layla20) that was already adequate
for most audio production purposes.

Lower-end audio production sound card converters (e.g. Echo Mia) roughly
follow the pattern I just described and show substantial noise performance
gains when run at 48 KHz and below, as compared to 96 KHz.

Mid-end audio production sound card converters (e.g. Card Deluxe) don't
follow pattern I just described but happily show substantial noise
performance gains over previous-generation cards, even when run at 96 KHz.

High-end audio production sound card converters (e.g. LynxTWO) are simply
stunning performers and are IMO practically incomparable to
previous-generation cards. We might see even better performance in the
future, but its practical benefits will be even harder to explain.

Happily, any of these cards have so much dynamic range that they provide
more than enough headroom and noise floor margins to force other parts of
the production chain to be the weakest links.

Addressing the noise-shaping issue raised by Vladan - as a rule even
lower-end production-grade sound card converter chips DON'T show the noise
floor variations one would expect if in-band noise shaping were an issue.
There IS spectral shaping of the quantization error in *all* modern
converters, but the quantization step is oversampled and the errors fall
outside the conversion band, even if the sample rate is *just* 44.1 KHz.

Even with 44.1 KHz and in-band noise shaping, the ear's sensitivity above 16
KHz is so poor that about 25% of the conversion band is available to be a
figurative garbage dump for quantization errors. It has been exploited this
way for years if not a decade or more.

Bob Cain wrote:

Bob Cain wrote:


Geez, you'll know I'm where I don't belong when I input this, but I read
some time ago, and I think it came from someone at Waves, that extending
the sample rate resulted in something like incresing bit depth because
conversion involved a fixed amount of quantization noise that wound up
fit into the resulting bandwidth. Extending that bandwidth with a
greater sample rate was said to spread the same amount of quantization
noise over a broader spectrum, resulting in a lower noise floor, as if
the bit depth had been increased.


That sounds seductively intuitive so to be sure I looked up
the derivation of quantization noise in "Digital Signal
Processing" by Proakis and Manolakis and the stastical
analyisis that leads to the usual 6 dB SNR per conversion
bit is independant of sample rate. (It's good to check our
assumptions now and again.)


Du'oh. I fired that off without enough thought. What you
said sounds seductively intuitive because it's right. If
you double the sample rate to twice the audio band then the
_total_ RMS noise only be a function of the converter width
but the noise will be evenly distributed over twice as wide
a spectrum, half of which you can't hear, so that the
audible noise will be reduced by half.


That sounds suspicious. The entropy in the passband should be the same,
by conservation.

Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein


--
Les Cargill

"opus" wrote in message news:Tmiib.743141$YN5.679914@sccrnsc01...
You must look at the "whole" picture here. Yes, sample rate has a lot to
do
with bandwidth "BUT" with only 16 bit operation you are not getting a
smooth
a snapshot as with 24 bit.

Nonsense!!! Ever hear of dither? Ever hear of random noise?


Are you refering to stairways effect? I think I've heard something
about that, but couldn't understand. Is 24bit sample rate to help our
drummer, who plays fills and rolls that sound just like boxes
stumbling dow the stairs, to get more warmth in his groove.

Yes and No. The stairway effect is more on the sample resolution on how the
samples are taken. The lower the sample rate the more block shaped it is.
The higher the sampling
rate the more smoother it is. It has nothing to do with performance of the
musician and the groove.

Where is Dick Pierce when we need him!! If this represents Apogee's
technology, $5 sound cards are a much better investment. Ever hear of
Nyquist?????

Also with 24 bit recordings you don't need to get
as much signal in as you did with 16 bit machines to get the total
digital
bandwidth as needed.


Digital bandwidth refers to the data rate in bits/second. Of course
the digital bandwidth will be higher for higher resolution. This does
not necessarily imply that the encoded audio signal will be better.

You mean I can play only a part of my song and the rest will be
guessed by 24bit sample rate?

No. With 16 bit converters you had to get the hottest signal possible to
truly take
advantage of the digital bandwidth. With 24 bit converters it's not
necessary. Obviously you still want a hot
signal but you don't have to with 24 bit converters. Hope that makes sense.

Not particularly, but it makes more sense than many of your previous
assertions.


I have some material at work that can truly enlighten us to the reason
for
this. I will forward it to my home and post it here
for everyone to see.

Please, by all means.

Yes, please, please submit this material to the IEEE. I am sure that
they will want to learn about the errors of their ways in their DSP
publications of the last half-century.

On Tue, 14 Oct 2003 08:47:42 -0400, "Arny Krueger"
wrote:

Addressing the noise-shaping issue raised by Vladan - as a rule even
lower-end production-grade sound card converter chips DON'T show the noise
floor variations one would expect if in-band noise shaping were an issue.
There IS spectral shaping of the quantization error in *all* modern
converters, but the quantization step is oversampled and the errors fall
outside the conversion band, even if the sample rate is *just* 44.1 KHz.

Ok, I beleive you, although, must say, I never mentioned convertor
chips, but was more thinking about algos, ie mangling data by an plug
in, within application (provided Waves being mentioned and I think of
waves as of software company). Also since Waves being mentioned, I
think I remember reading something like that in their help files.
Neverthrless, let it be.

Vladan
www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

Arny Krueger wrote:

More specifically, all other things being equal, if you double the converter
bandwidth, noise goes up by the inverse of the square root of two.

Got a reference for that, Arny, prefereably something online
if you know one? Mine (Proakis and Manolakis) shows a
statistical derivation of total noise that is not dependant
on converter bandwidth.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Just to ad in context of previous messages and plain words, why 96k
sampling rate does not help with noise, or substitutes for resolution
....

.... because not only noise is spread over broader range, but the whole
signal. Therefore, lots of energy is placed in inaudiable range and
although the signal may produce the same SPL as signal sampled at
lower rate, it will be perceived as softer(less loud) To acheive same
perceived loudness (SPL in audiable range) the level has to be raised
certain ammount. that ammount is such, that the noise, in audiable
range, comes to the same level as if it was sampled at lower sampling
rate.

Example in software processing: I have 96K file and 48k file of the
same signal, both peaking at 0dBFs. On listen 96k file will sound
softer, because ther's more of it in inaudiable range - therefore less
in audiable.

Should I resample 96K file to 48K, the whole file will get lower in
volume by cca 2,4dB. After boosting downsampled signal to be of the
same level as original 48K, they will all sound about the same.
Vladan
www.geocities.com/vla_dan_l
www.mp3.com/lesly , www.mp3.com/shook , www.mp3.com/lesly2
www.kunsttick.com/artists/vuskovic/indexdat.htm

Opus, I hope you'll be soon posting the material from your work to here..

Tommi wrote:

Opus, I hope you'll be soon posting the material from your work to here..


I'm guessing somebody at Apogee caught wind of Opus sharing his wide
breadth of knowledge in their name and put a stop to it. He's back to
work delivering box lunches to the engineers now.

ulysses

"Bob Cain" wrote in message

Arny Krueger wrote:

More specifically, all other things being equal, if you double the
converter bandwidth, noise goes up by the inverse of the square root
of two.

Got a reference for that, Arny, prefereably something online
if you know one?

Not readily available.

Mine (Proakis and Manolakis) shows a
statistical derivation of total noise that is not dependant
on converter bandwidth.

They must be leaving something out or taking a really narrow view of what a
"converter" is.

Arny Krueger wrote:


Mine (Proakis and Manolakis) shows a
statistical derivation of total noise that is not dependant
on converter bandwidth.

They must be leaving something out or taking a really narrow view of what a
"converter" is.

I rather doubt that but if so I'd sure like to know what.
If you run across a deeper analysis please let me know.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Bob Cain" wrote in message

Arny Krueger wrote:


Mine (Proakis and Manolakis) shows a
statistical derivation of total noise that is not dependant
on converter bandwidth.

They must be leaving something out or taking a really narrow view of
what a "converter" is.

I rather doubt that but if so I'd sure like to know what.

Regrattably I lack access to the cited text. I think the overall argument
would be that total noise is irrelevant to converter bandwidth is based on
the fact that quantization noise is the dominant kind of noise. Quantization
noise is related to bit depth, not sample rate (bandwidth).

That seems like it would be true if a converter were simply a quantizer.
However, real-world converter chips are at their simplest also amplifiers
(regardless of gain), and have incidental noise from other sources than
quanitization. Modern converters have such good resolution that ordinary
interfacing components that may or may not be on the converter chip can be
significant sources of noise. Internally, modern converters have a number of
separate elements in their block diagrams that seem like they can contribute
noise.

If you run across a deeper analysis please let me know.

How does your text's block diagram of a converter compare with a real world
part, like this one?

http://www.cirrus.com/en/pubs/whiteP...s43122wp-1.pdf page 4.

Arny Krueger wrote:


How does your text's block diagram of a converter compare with a real world
part, like this one?

You are right, Arny. The analysis only concerns itself with
quantization noise and doesn't address other noise that may
be in the signal prior to the quantization or enter it from
the process in some way. I had thought that the topic was
the effect of sample rate on total quantization noise rather
than all possible noise sources. In that limited context,
there is no effect.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Bob Cain" wrote in message

Arny Krueger wrote:


How does your text's block diagram of a converter compare with a
real world part, like this one?

You are right, Arny. The analysis only concerns itself with
quantization noise and doesn't address other noise that may
be in the signal prior to the quantization or enter it from
the process in some way.

OK, then I agree with it as it stands. This is a good point for a text to
make.

I had thought that the topic was
the effect of sample rate on total quantization noise rather
than all possible noise sources. In that limited context,
there is no effect.

Agreed. This subthread started with this statement:

"Signal processing theory tells us that the
effects of sample rate and word width are orthogonal, the
former affecting only bandwidth and the latter only
quantization error (noise.)"

I know of no exceptions to this theoretical statement.

"If they overlap due to deeper considerations I'd love to gain an
understanding."

The deeper considerations seem to relate only to real-world
implementations...

"Bob Cain" wrote in message

Arny Krueger wrote:


How does your text's block diagram of a converter compare with a
real world part, like this one?

You are right, Arny. The analysis only concerns itself with
quantization noise and doesn't address other noise that may
be in the signal prior to the quantization or enter it from
the process in some way.

OK, then I agree with it as it stands. This is a good point for a text to
make.

I had thought that the topic was
the effect of sample rate on total quantization noise rather
than all possible noise sources. In that limited context,
there is no effect.

Agreed. This subthread started with this statement:

"Signal processing theory tells us that the
effects of sample rate and word width are orthogonal, the
former affecting only bandwidth and the latter only
quantization error (noise.)"

I know of no exceptions to this theoretical statement.

"If they overlap due to deeper considerations I'd love to gain an
understanding."

The deeper considerations seem to relate only to real-world
implementations...

Arny Krueger wrote:


"If they overlap due to deeper considerations I'd love to gain an
understanding."

The deeper considerations seem to relate only to real-world
implementations...

Fair enough. I'd still enjoy reading a more thorough
analysis of this should you ever run across it.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein