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Sampling Rate past 44 Khz

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Anonymous
February 1, 2005 6:18:23 PM

Archived from groups: rec.audio.pro (More info?)

Recording at a higher sampling rate obviously produces better
recordings. Is it overkill to record higher than 44KHz since CD's have
a sampling rate of 44 KHz? Does it make a difference when the final
product is transferred to CD?

Stan

More about : sampling rate past khz

Anonymous
February 1, 2005 9:36:53 PM

Archived from groups: rec.audio.pro (More info?)

<skingfong@yahoo.com> wrote:
>Recording at a higher sampling rate obviously produces better
>recordings.

Does it? Does wider frequency response necessarily produce better
recordings? If I could reproduce a 200 KHz ultrasonic tone, would it
be a good thing in any way? Faster sampling rate only buys you more
extension on the top end, nothing else. And it comes with some big
disadvantages because it makes clock stability that much more difficult.

>Is it overkill to record higher than 44KHz since CD's have
>a sampling rate of 44 KHz? Does it make a difference when the final
>product is transferred to CD?

No, but do you have any intention of releasing on some other format
some time in the future?
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 1, 2005 11:31:56 PM

Archived from groups: rec.audio.pro (More info?)

Scott Dorsey wrote:
> <skingfong@yahoo.com> wrote:
> >Recording at a higher sampling rate obviously produces better
> >recordings.
>
> Does it? Does wider frequency response necessarily produce better
> recordings? If I could reproduce a 200 KHz ultrasonic tone, would it
> be a good thing in any way? Faster sampling rate only buys you more
> extension on the top end, nothing else. And it comes with some big
> disadvantages because it makes clock stability that much more
difficult.

I'm just starting to find out about the clock stability. It has been
wreaking havoc with my computer. I'm finding that it's much easier to
work with 44Khz sampling. My computer seems to like it more. The higher
sample rates are not worth sacrificing clock stability.


> >Is it overkill to record higher than 44KHz since CD's have
> >a sampling rate of 44 KHz? Does it make a difference when the final
> >product is transferred to CD?
>
> No, but do you have any intention of releasing on some other format
> some time in the future?

No

Stan
Related resources
Anonymous
February 2, 2005 12:12:11 AM

Archived from groups: rec.audio.pro (More info?)

I'm just getting familiar with my new soundcard. (M-Audio Audiophile
192.) To me, higher sampling rate recordings I've recorded sound
slighly warmer and smoother. I've been experimenting with recording
44Khz to 192Khz. The difference was subtle and not really worth going
up to the higher sampling rates at the expense of stabilty and
processing speed.

Stan
Anonymous
February 2, 2005 12:20:43 AM

Archived from groups: rec.audio.pro (More info?)

<skingfong@yahoo.com> wrote in message
news:1107299903.501517.120270@l41g2000cwc.googlegroups.com


> Recording at a higher sampling rate obviously produces better
> recordings.

Reasons being?

> Is it overkill to record higher than 44KHz since CD's have
> a sampling rate of 44 KHz?

yes.

> Ds it make a difference when the final product is transferred to CD?

Not unless there is appreciable nonlinear distortion in the processing
chain, prior to the downsampling. If so, then recording at a higher bitrate
not only makes an audible difference, it is an audible difference that could
be considered to be EFX.
February 2, 2005 2:19:33 AM

Archived from groups: rec.audio.pro (More info?)

When in a top studio in London, I asked the (very experienced)
engineer/producer why he used tape multi-track rather than digital. His
answer was that even at 96k 24-bit digital, with the best converters
money can buy, tape was giving him better FREQUENCY RESPONSE.

Chris
( http:www.chris-melchior.com/strings.htm REAL strings for realistic
prices )
Anonymous
February 2, 2005 3:11:37 AM

Archived from groups: rec.audio.pro (More info?)

On 2/1/05 5:18 PM, in article
1107299903.501517.120270@l41g2000cwc.googlegroups.com, "skingfong@yahoo.com"
<skingfong@yahoo.com> wrote:

> Recording at a higher sampling rate obviously produces better
> recordings. Is it overkill to record higher than 44KHz since CD's have
> a sampling rate of 44 KHz? Does it make a difference when the final
> product is transferred to CD?

In my experience, most projects sound better if I do all my digital
processing at 88.2/96kHz and then step the final down through a high-quality
SRC unit, than if I do it all at standard sample rate (44.1/48kHz). ADC's
and DAC's usually sound better at high sample rates as well, but not always.

The key is to have a very high-quality SRC. If you don't, then the damage of
bad up-sampling/down-sampling will override any advantage of high sample
rate processing.


Allen
--
Allen Corneau
Mastering Engineer
Essential Sound Mastering
Houston, TX
Anonymous
February 2, 2005 8:16:39 AM

Archived from groups: rec.audio.pro (More info?)

<skingfong@yahoo.com> wrote in message...

> Recording at a higher sampling rate obviously produces better
> recordings.


As always, I disagree with this assumption. Higher frequency content
does not inherently make for 'better' recordings. People make better
recordings, for a myriad of reasons.


--
David Morgan (MAMS)
http://www.m-a-m-s DOT com
Morgan Audio Media Service
Dallas, Texas (214) 662-9901
_______________________________________
http://www.artisan-recordingstudio.com
Anonymous
February 2, 2005 10:50:34 AM

Archived from groups: rec.audio.pro (More info?)

<chris@chris-melchior.com> wrote in message
news:1107328773.835762.91320@o13g2000cwo.googlegroups.com...
> When in a top studio in London, I asked the (very experienced)
> engineer/producer why he used tape multi-track rather than digital. His
> answer was that even at 96k 24-bit digital, with the best converters
> money can buy, tape was giving him better FREQUENCY RESPONSE.
>
> Chris
> ( http:www.chris-melchior.com/strings.htm REAL strings for realistic
> prices )
>

Now THAT's a surprise. That means his 30IPS reel to reel must be heading up
towards 60khz.
However, I've never heard a pure tone on analog tape. There's always some
'bumpiness' for lack of a better term. High pitched tones, like 15khz, seem
to have a bunch of harmonics and other modulation noises attached to them.
Oddly, with music, it doesn't seem to be that noticable.


--
Best Regards,

Mark A. Weiss, P.E.
www.mwcomms.com
-
Anonymous
February 2, 2005 11:04:11 AM

Archived from groups: rec.audio.pro (More info?)

<skingfong@yahoo.com> wrote in message
news:1107321131.353864.87680@l41g2000cwc.googlegroups.com

> I'm just getting familiar with my new soundcard. (M-Audio Audiophile
> 192.) To me, higher sampling rate recordings I've recorded sound
> slighly warmer and smoother.

OK, so is that because you're groovin' on a new piece of gear, or because it
actually sounds different? Don't be shy - I'm a gear slut too, but I also
try to maintain some degree of factuality about what I write.

>I've been experimenting with recording
> 44Khz to 192Khz. The difference was subtle and not really worth going
> up to the higher sampling rates at the expense of stabilty and
> processing speed.

Didn't you just contradict what you said in your first paragraph?
Anonymous
February 2, 2005 11:07:16 AM

Archived from groups: rec.audio.pro (More info?)

"Mark & Mary Ann Weiss" <mweissX294@earthlink.net> wrote in message
news:eD%Ld.7049$Ix.5335@newsread3.news.atl.earthlink.net
> <chris@chris-melchior.com> wrote in message
> news:1107328773.835762.91320@o13g2000cwo.googlegroups.com...
>> When in a top studio in London, I asked the (very experienced)
>> engineer/producer why he used tape multi-track rather than digital.
>> His answer was that even at 96k 24-bit digital, with the best
>> converters money can buy, tape was giving him better FREQUENCY
>> RESPONSE.

> Now THAT's a surprise. That means his 30IPS reel to reel must be
> heading up towards 60khz.

Indeed.

> However, I've never heard a pure tone on analog tape.

Nor have I. I've always been very happy that we primarily used analog tape
to record music, not pure tones. Their inability to record pure tones does
give some cause for pause.

>There's always some 'bumpiness' for lack of a better term.

Agreed.

>High pitched tones, like 15khz, seem to have a bunch of harmonics and
>other modulation noises
> attached to them.

I think its the same mechanism as the bumpiness, being reflected down to low
frequencies by the naturally high IM of analog tape.

>Oddly, with music, it doesn't seem to be that noticable.

Agreed.
Anonymous
February 2, 2005 12:01:59 PM

Archived from groups: rec.audio.pro (More info?)

chris@chris-melchior.com wrote:

> When in a top studio in London, I asked the (very experienced)
> engineer/producer why he used tape multi-track rather than digital. His
> answer was that even at 96k 24-bit digital, with the best converters
> money can buy, tape was giving him better FREQUENCY RESPONSE.

What format was the recording ultimately going to be released on
that could take advantage of that frequency response? Did he plan
to sell 15 or 30 inch per second reel-to-reel tape copies at the
retail stores across the nation? Is there a section for that at
Tower Records (or local British equivalent)?

- Logan
Anonymous
February 2, 2005 12:18:58 PM

Archived from groups: rec.audio.pro (More info?)

In article <1107318716.955575.18000@c13g2000cwb.googlegroups.com>,
<skingfong@yahoo.com> wrote:
>
>
>> >Is it overkill to record higher than 44KHz since CD's have
>> >a sampling rate of 44 KHz? Does it make a difference when the final
>> >product is transferred to CD?
>>
>> No, but do you have any intention of releasing on some other format
>> some time in the future?
>
>No

Don't worry about it, then. There is some processing that might benefit
from higher sample rates, and there are some converters that might sound
better at higher sample rates, but there is also other processing that
becomes problematic at higher sample rates because of the additional
CPU horsepower needed, and there are other converters that sound worse
at higher sample rates because of the clocking issues. So it's a wash
overall, although it might be of benefit in some cases.

DSD, on the other hand, really does seem to be a step up. But it is
not catching on that strongly.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 2, 2005 12:22:34 PM

Archived from groups: rec.audio.pro (More info?)

chris@chris-melchior.com <chris@chris-melchior.com> wrote:
>When in a top studio in London, I asked the (very experienced)
>engineer/producer why he used tape multi-track rather than digital. His
>answer was that even at 96k 24-bit digital, with the best converters
>money can buy, tape was giving him better FREQUENCY RESPONSE.

Well, maybe he likes the head bump. Lots of people do, and for rock
stuff the head bump is a real advantage and might even be worth going
to analogue tape alone. You also have some ability to tailor response
with the emphasis adjustments too, which can sometimes be handy.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 2, 2005 1:03:08 PM

Archived from groups: rec.audio.pro (More info?)

"Scott Dorsey" <kludge@panix.com> wrote in message
news:ctqngi$ou5$1@panix2.panix.com

> DSD, on the other hand, really does seem to be a step up. But it is
> not catching on that strongly.


I don't see it. Data rate-wise, dynamic range-wise, DSD is a lesser, subset
format compared to DVD-A 24/192. The basic bit stream process is now known
to have a PCM core for encoding, as Sony implemented it. The other core DSD
technology is just noise shaping which has been around forever and has
already been advantageously retrofitted to CD Audio. The most common way to
decode DSD has turned out to be a modification of a standard PCM DAC, so
there's also zero cost advantage on the decode end.

This discussion does not consider the perceptual consequences of greater
bandpass and dynamic range, to avoid clouding some pretty clear
practical/technical issues.
Anonymous
February 2, 2005 1:32:50 PM

Archived from groups: rec.audio.pro (More info?)

In article <1107328773.835762.91320@o13g2000cwo.googlegroups.com> chris@chris-melchior.com writes:

> When in a top studio in London, I asked the (very experienced)
> engineer/producer why he used tape multi-track rather than digital. His
> answer was that even at 96k 24-bit digital, with the best converters
> money can buy, tape was giving him better FREQUENCY RESPONSE.

That may have been his explanation, but it's not really correct.
Recording engineers and producers rarely verify their perceptions with
laboratory measuements. What he really meant was that he liked the
sound better. Is there something wrong with that?

Another reason is that it's usually easier to record basic tracks on
an analog recorer (or a digital recorder that works like one) rather
than a computer because you don't usually have very much punching and
fixing to do. The work goes quicker during that phase. DAWs are often
used for the "detail" work where you're recording one track at a time
and fixing it one piece at a time.

If you have to do computer-like fixes on your basic tracks, you have a
different problem that's unrelated to the sound of the recorder, and
that you need to solve first.

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
February 2, 2005 1:38:05 PM

Archived from groups: rec.audio.pro (More info?)

Arny Krueger wrote:

> "Scott Dorsey" <kludge@panix.com> wrote in message
> news:ctqngi$ou5$1@panix2.panix.com
>
>
>>DSD, on the other hand, really does seem to be a step up. But it is
>>not catching on that strongly.
>
>
>
> I don't see it. Data rate-wise, dynamic range-wise, DSD is a lesser, subset
> format compared to DVD-A 24/192. The basic bit stream process is now known
> to have a PCM core for encoding, as Sony implemented it. The other core DSD
> technology is just noise shaping which has been around forever and has
> already been advantageously retrofitted to CD Audio. The most common way to
> decode DSD has turned out to be a modification of a standard PCM DAC, so
> there's also zero cost advantage on the decode end.

They do seem to sound really good. You don't think so?
Anonymous
February 2, 2005 2:34:32 PM

Archived from groups: rec.audio.pro (More info?)

"Joe Sensor" <crabcakes@emagic.net> wrote in message
news:36cdkrF50a84fU1@individual.net
> Arny Krueger wrote:

>> Data rate-wise, dynamic range-wise, DSD is a lesser,
>> subset format compared to DVD-A 24/192. The basic bit stream process
>> is now known to have a PCM core for encoding, as Sony implemented
>> it. The other core DSD technology is just noise shaping which has
>> been around forever and has already been advantageously retrofitted
>> to CD Audio. The most common way to decode DSD has turned out to be
>> a modification of a standard PCM DAC, so there's also zero cost
>> advantage on the decode end.

> They do seem to sound really good. You don't think so?

I think that they sound good, and so do a lot of other things.

BTW Joe, you seem to have deleted the following relevant explanatory
paragraph:

"This discussion does not consider the perceptual consequences of greater
bandpass and dynamic range, to avoid clouding some pretty clear
practical/technical issues."

Why did you do that?
Anonymous
February 2, 2005 2:34:33 PM

Archived from groups: rec.audio.pro (More info?)

Arny Krueger wrote:

> BTW Joe, you seem to have deleted the following relevant explanatory
> paragraph:
>
> "This discussion does not consider the perceptual consequences of greater
> bandpass and dynamic range, to avoid clouding some pretty clear
> practical/technical issues."
>
> Why did you do that?

I gotta be honest. I have no idea what it meant. <eek>
Anonymous
February 2, 2005 2:43:34 PM

Archived from groups: rec.audio.pro (More info?)

"Joe Sensor" <crabcakes@emagic.net> wrote in message
news:36ce55F4vl5jfU1@individual.net
> Arny Krueger wrote:
>
>> BTW Joe, you seem to have deleted the following relevant explanatory
>> paragraph:
>>
>> "This discussion does not consider the perceptual consequences of
>> greater bandpass and dynamic range, to avoid clouding some pretty
>> clear practical/technical issues."
>>
>> Why did you do that?
>
> I gotta be honest. I have no idea what it meant. <eek>

IOW it says that I purposfully did not discuss whether they sounded
different, or one sounded bad, or another sounded better.

I have no problems with the sound quality of any of the hi-rez formats,
44/16 included.
Anonymous
February 2, 2005 3:40:48 PM

Archived from groups: rec.audio.pro (More info?)

Joe Sensor <crabcakes@emagic.net> wrote:
>Arny Krueger wrote:
>
>> "Scott Dorsey" <kludge@panix.com> wrote:
>>
>>>DSD, on the other hand, really does seem to be a step up. But it is
>>>not catching on that strongly.
>>
>> I don't see it. Data rate-wise, dynamic range-wise, DSD is a lesser, subset
>> format compared to DVD-A 24/192. The basic bit stream process is now known
>> to have a PCM core for encoding, as Sony implemented it. The other core DSD
>> technology is just noise shaping which has been around forever and has
>> already been advantageously retrofitted to CD Audio. The most common way to
>> decode DSD has turned out to be a modification of a standard PCM DAC, so
>> there's also zero cost advantage on the decode end.
>
>They do seem to sound really good. You don't think so?

I think it sounds really good, but I want to know why. I think Arny is
concerned entirely with that part.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 2, 2005 3:40:49 PM

Archived from groups: rec.audio.pro (More info?)

Scott Dorsey wrote:

>
> I think it sounds really good, but I want to know why. I think Arny is
> concerned entirely with that part.

I think it's because the engineering is better due to the
belief that the format is more revealing, whether or not the
latter is true.


Bob
--

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

A. Einstein
Anonymous
February 2, 2005 5:43:30 PM

Archived from groups: rec.audio.pro (More info?)

Bob Cain <arcane@arcanemethods.com> wrote:
>Scott Dorsey wrote:
>
>>
>> I think it sounds really good, but I want to know why. I think Arny is
>> concerned entirely with that part.
>
>I think it's because the engineering is better due to the
>belief that the format is more revealing, whether or not the
>latter is true.

Well, I'm all in favor of that, then. I think that's the big deal with
the XRCD process, too. If the effect is entirely a psychological one on
engineers, it's just fine by me as long as it results in a better final
product.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 2, 2005 8:22:47 PM

Archived from groups: rec.audio.pro (More info?)

On 2005-02-02, chris@chris-melchior.com <chris@chris-melchior.com> wrote:

> money can buy, tape was giving him better FREQUENCY RESPONSE.

Wider? Or just more accurate within a specific range? Tape with a
400kHz bias? Frequency response measured how?
Anonymous
February 2, 2005 8:22:48 PM

Archived from groups: rec.audio.pro (More info?)

In article <slrnd02343.2vq.fishbowl@radagast.home.conservatory.com>,
james of tucson <fishbowl@conservatory.com> wrote:
>On 2005-02-02, chris@chris-melchior.com <chris@chris-melchior.com> wrote:
>
>> money can buy, tape was giving him better FREQUENCY RESPONSE.
>
>Wider? Or just more accurate within a specific range? Tape with a
>400kHz bias? Frequency response measured how?

Wider and more accurate is not necessarily better. Sometimes better
response is having a head bump.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 2, 2005 10:10:12 PM

Archived from groups: rec.audio.pro (More info?)

Arny Krueger wrote:
> <skingfong@yahoo.com> wrote in message
> news:1107321131.353864.87680@l41g2000cwc.googlegroups.com

>
> OK, so is that because you're groovin' on a new piece of gear, or
because it
> actually sounds different? Don't be shy - I'm a gear slut too, but I
also
> try to maintain some degree of factuality about what I write.

I think it's both. I'm definately a gear slut. I also like to maintain
some degree of factuality. Let me clarify, I was just giving my opinion
not stating a fact.

> >I've been experimenting with recording>
> > 44Khz to 192Khz. The difference was subtle and not really worth
going
> > up to the higher sampling rates at the expense of stabilty and
> > processing speed.
>
> Didn't you just contradict what you said in your first paragraph?
Good call. You got me there.

Stan
Anonymous
February 3, 2005 7:20:19 PM

Archived from groups: rec.audio.pro (More info?)

PROBLEM:
I have a number of live recordings I did during the 60s. I sang and played
guitar at colleges
And coffee houses throughout the US and Canada. . (It was at the Zenith of the
big "Folk music
Scare".) However, I was usually stoned on grass, beer, nutmeg, Ripple, or
anything else non-addictive that would give me a buzz, just for the fun of it,
and so I would appear cool to the hippie chicks I was trying to pick up after the
shows.

The audience didn't seem to mind, as most of them were as stoned as me, or more.
For that
time period, all things were equal and in balance. After all it was during the
Vietnam War, something we’ll never have to go through again, right? Oh, yeah,
right..

BUT, in listening to those performances with a critical ear now, I notice that
the performances
are really SLOW and almost sluggish. A song I would do in 2 and a half minutes
now, took 3 minutes to do back
then. But, the performances were otherwise good. I sang in pitch, and the
audiences reacted
extremely well.

Therefore, I thought I would experiment with some of the old taped performances
by speeding
up the tempo digitally, but without changing the pitch. So this is actually a
question about time
shortening rather than time stretching. I use the reference; "time Stretch"
because that's what
the process is called in WAVELAB and several other programs I have (Cool Edit,
DCLive,
Cakewalk, etc).

VOILA!!!. I have tried different settings, and have found that by taking a 3
minute song, and
time compressing it into 2 and half minutes makes it more exciting and
listenable. But there are little pops, and hiccups, and other artifacts in my
preliminary tests. An engineer pal told me to
convert my 16 bit recordings I did at 44.1 khz and raise it to 32bit at 96 khz
BEFORE I process
them with the "time stretch" tools. In essence, this would give the time
stretching tool more bits
of data to work with, and should show LESS artifacts. I tried this, and it
improved the sound.
Less glitches and pops. HOWEVER, my engineer friend pointed out that there is
LESS of a
problem when you LENGTHEN a song, rather than SHORTENING it, for obvious reasons.

When you shorten a song, you're actually REMOVING chunks of data, and it will be
more
Audible than if you ADD more data.

So, I have come up with an alternate approach I MIGHT try. I suppose I could
speed up the
TAPE before I transfer it over to the computer first, and find the desired TEMPO
I like. Of
course, this will raise the key more than I want it to be. However, then I can
try changing the
pitch back to the original key, digitally. I'm wondering if there will be less
audible artifacts by
altering the key, rather than changing the tempo.

. Of course, I could just indicate in the liner notes that I was stoned, but
that wouldn't be any fun
now, would it? Let's all keep in mind that many of the hit recordings by Fats
Domino and Little
Richard were usually sped up a full key, as were some of Art Garfunkle's hits.
This was
probably done to shorten the songs' length so they would get radio play, or
perhaps to make
them more exciting. (KEEP A KNOCKIN' by Little Richard was EXTREMELY exciting.)

I'm not against using some studio voodoo here, so please, no lectures on the
"aesthetics" of
what I want to accomplish .This is a technical question I'm asking you engineers
to help me
resolve. This is show biz, right? After all, Leonard Bernstein was considered to
be a REALLY
sensitive artist and conductor. However, he ran around his house in a TUTU. Ha
Ha. All art is
therefore as fake as show business. Frank Frazetta is a better comic book artist
than
Michaelangelo was of religious murals on church ceilings. Still, history says
that Michaelangelo
was an artist and that Frank Frazetta is not.

Are there any other suggestions from any of you studio wizards on how to best
speed up the
performances, with minimal artifacts, and without altering the original keys? Of
all the "Time
Stretching" tools that are available, which ones are the best? It would save me
some time
knowing whether to use the tools in Cool Edit Pro, versus, Cuebase, Cakewalk,
Sonar,
DCLIve, etc. I usually work with Cakewalk and Wavelab, and although I have the
other
programs, I haven't worked with them much. In fact, I hadn't tried any time
stretching until this
week, although I've had Wavelab for 6 years.

I suppose I could have just done a joint before I listened to the old
performances, and I wouldn't
have noticed any sluggishess or bothered this board with my problem. But, it's
been a long
time since I did that. Maybe I should ask my kids if they'll sell me some.
Thanks..

Alan Cassaro
Anonymous
February 3, 2005 7:20:20 PM

Archived from groups: rec.audio.pro (More info?)

For heaven's sake, why don't you leave them alone, as an accurate record
of what you did at that time in your life? How satisfying would it be
to you if you could alter the photographs in your high school yearbook
to remove your acne, or change the clothing to remove the bell-bottoms
and peasant shirts, or rub out your girl-friend and make her skinny
instead, and make Grandpa a veteran of the Spanish Civil War, and make
Grandma Miss Poland 1935, and make Uncle Roy a general in the
Confederate army......

If you need a faster version of those songs, get out your guitar...




Alan Cassaro wrote:
> PROBLEM:
> I have a number of live recordings I did during the 60s. I sang and played
> guitar at colleges
> And coffee houses throughout the US and Canada. . (It was at the Zenith of the
> big "Folk music
> Scare".) However, I was usually stoned on grass, beer, nutmeg, Ripple, or
> anything else non-addictive that would give me a buzz, just for the fun of it,
> and so I would appear cool to the hippie chicks I was trying to pick up after the
> shows.
>
> The audience didn't seem to mind, as most of them were as stoned as me, or more.
> For that
> time period, all things were equal and in balance. After all it was during the
> Vietnam War, something we’ll never have to go through again, right? Oh, yeah,
> right..
>
> BUT, in listening to those performances with a critical ear now, I notice that
> the performances
> are really SLOW and almost sluggish. A song I would do in 2 and a half minutes
> now, took 3 minutes to do back
> then. But, the performances were otherwise good. I sang in pitch, and the
> audiences reacted
> extremely well.
>
> Therefore, I thought I would experiment with some of the old taped performances
> by speeding
> up the tempo digitally, but without changing the pitch. So this is actually a
> question about time
> shortening rather than time stretching. I use the reference; "time Stretch"
> because that's what
> the process is called in WAVELAB and several other programs I have (Cool Edit,
> DCLive,
> Cakewalk, etc).
>
> VOILA!!!. I have tried different settings, and have found that by taking a 3
> minute song, and
> time compressing it into 2 and half minutes makes it more exciting and
> listenable. But there are little pops, and hiccups, and other artifacts in my
> preliminary tests. An engineer pal told me to
> convert my 16 bit recordings I did at 44.1 khz and raise it to 32bit at 96 khz
> BEFORE I process
> them with the "time stretch" tools. In essence, this would give the time
> stretching tool more bits
> of data to work with, and should show LESS artifacts. I tried this, and it
> improved the sound.
> Less glitches and pops. HOWEVER, my engineer friend pointed out that there is
> LESS of a
> problem when you LENGTHEN a song, rather than SHORTENING it, for obvious reasons.
>
> When you shorten a song, you're actually REMOVING chunks of data, and it will be
> more
> Audible than if you ADD more data.
>
> So, I have come up with an alternate approach I MIGHT try. I suppose I could
> speed up the
> TAPE before I transfer it over to the computer first, and find the desired TEMPO
> I like. Of
> course, this will raise the key more than I want it to be. However, then I can
> try changing the
> pitch back to the original key, digitally. I'm wondering if there will be less
> audible artifacts by
> altering the key, rather than changing the tempo.
>
> . Of course, I could just indicate in the liner notes that I was stoned, but
> that wouldn't be any fun
> now, would it? Let's all keep in mind that many of the hit recordings by Fats
> Domino and Little
> Richard were usually sped up a full key, as were some of Art Garfunkle's hits.
> This was
> probably done to shorten the songs' length so they would get radio play, or
> perhaps to make
> them more exciting. (KEEP A KNOCKIN' by Little Richard was EXTREMELY exciting.)
>
> I'm not against using some studio voodoo here, so please, no lectures on the
> "aesthetics" of
> what I want to accomplish .This is a technical question I'm asking you engineers
> to help me
> resolve. This is show biz, right? After all, Leonard Bernstein was considered to
> be a REALLY
> sensitive artist and conductor. However, he ran around his house in a TUTU. Ha
> Ha. All art is
> therefore as fake as show business. Frank Frazetta is a better comic book artist
> than
> Michaelangelo was of religious murals on church ceilings. Still, history says
> that Michaelangelo
> was an artist and that Frank Frazetta is not.
>
> Are there any other suggestions from any of you studio wizards on how to best
> speed up the
> performances, with minimal artifacts, and without altering the original keys? Of
> all the "Time
> Stretching" tools that are available, which ones are the best? It would save me
> some time
> knowing whether to use the tools in Cool Edit Pro, versus, Cuebase, Cakewalk,
> Sonar,
> DCLIve, etc. I usually work with Cakewalk and Wavelab, and although I have the
> other
> programs, I haven't worked with them much. In fact, I hadn't tried any time
> stretching until this
> week, although I've had Wavelab for 6 years.
>
> I suppose I could have just done a joint before I listened to the old
> performances, and I wouldn't
> have noticed any sluggishess or bothered this board with my problem. But, it's
> been a long
> time since I did that. Maybe I should ask my kids if they'll sell me some.
> Thanks..
>
> Alan Cassaro
>
Anonymous
February 3, 2005 11:39:07 PM

Archived from groups: rec.audio.pro (More info?)

I have an engineering problem I want to solve. I asked that we make no aesthetic
judgements about it. I've already wrestled with that problem and resolved it in my own
mind. That's why I came to the newsgroup here for a possible technical solution to my
problem.
As far as my high school pictures are concerned, I DID transfer them over to the
computer, and I DID get rid of my ACNE. I also got rid of my girlfriend's crossed eyes.

Finally, I took some old pubicity photographs of a duo I was in for ten years, and
I digitally removed my ex- singing partner from the photos. Technology can do neat
things. Natalie Cole sings with her Dad and gets a Grammy. Hank Williams senior and Jr
appear in a video together singing "TEAR IN MY BEER", with Hank SR mouthing the new
lyrics. . Steve MC QUEEN is currently appearing in a car commercial. Hey, if it
makes you happy, I could just speed all the performances up, key and tempo alike, and
then they'll be like a quarter of the records that were issued in the Fifties, just
to come in under two minutes. The public doesn't give a damn. If they ever gave a damn
about anything bordering on artistic, drum machines would be outlawed. Name something
in music that succeeded that wasn't a little fake.

Thank you for your help by not addressing any of my questions,
Al

Bill Van Dyk wrote:

> For heaven's sake, why don't you leave them alone, as an accurate record
> of what you did at that time in your life? How satisfying would it be
> to you if you could alter the photographs in your high school yearbook
> to remove your acne, or change the clothing to remove the bell-bottoms
> and peasant shirts, or rub out your girl-friend and make her skinny
> instead, and make Grandpa a veteran of the Spanish Civil War, and make
> Grandma Miss Poland 1935, and make Uncle Roy a general in the
> Confederate army......
>
> If you need a faster version of those songs, get out your guitar...
>
> Alan Cassaro wrote:
> > PROBLEM:
> > I have a number of live recordings I did during the 60s. I sang and played
> > guitar at colleges
> > And coffee houses throughout the US and Canada. . (It was at the Zenith of the
> > big "Folk music
> > Scare".) However, I was usually stoned on grass, beer, nutmeg, Ripple, or
> > anything else non-addictive that would give me a buzz, just for the fun of it,
> > and so I would appear cool to the hippie chicks I was trying to pick up after the
> > shows.
> >
> > The audience didn't seem to mind, as most of them were as stoned as me, or more.
> > For that
> > time period, all things were equal and in balance. After all it was during the
> > Vietnam War, something we’ll never have to go through again, right? Oh, yeah,
> > right..
> >
> > BUT, in listening to those performances with a critical ear now, I notice that
> > the performances
> > are really SLOW and almost sluggish. A song I would do in 2 and a half minutes
> > now, took 3 minutes to do back
> > then. But, the performances were otherwise good. I sang in pitch, and the
> > audiences reacted
> > extremely well.
> >
> > Therefore, I thought I would experiment with some of the old taped performances
> > by speeding
> > up the tempo digitally, but without changing the pitch. So this is actually a
> > question about time
> > shortening rather than time stretching. I use the reference; "time Stretch"
> > because that's what
> > the process is called in WAVELAB and several other programs I have (Cool Edit,
> > DCLive,
> > Cakewalk, etc).
> >
> > VOILA!!!. I have tried different settings, and have found that by taking a 3
> > minute song, and
> > time compressing it into 2 and half minutes makes it more exciting and
> > listenable. But there are little pops, and hiccups, and other artifacts in my
> > preliminary tests. An engineer pal told me to
> > convert my 16 bit recordings I did at 44.1 khz and raise it to 32bit at 96 khz
> > BEFORE I process
> > them with the "time stretch" tools. In essence, this would give the time
> > stretching tool more bits
> > of data to work with, and should show LESS artifacts. I tried this, and it
> > improved the sound.
> > Less glitches and pops. HOWEVER, my engineer friend pointed out that there is
> > LESS of a
> > problem when you LENGTHEN a song, rather than SHORTENING it, for obvious reasons.
> >
> > When you shorten a song, you're actually REMOVING chunks of data, and it will be
> > more
> > Audible than if you ADD more data.
> >
> > So, I have come up with an alternate approach I MIGHT try. I suppose I could
> > speed up the
> > TAPE before I transfer it over to the computer first, and find the desired TEMPO
> > I like. Of
> > course, this will raise the key more than I want it to be. However, then I can
> > try changing the
> > pitch back to the original key, digitally. I'm wondering if there will be less
> > audible artifacts by
> > altering the key, rather than changing the tempo.
> >
> > . Of course, I could just indicate in the liner notes that I was stoned, but
> > that wouldn't be any fun
> > now, would it? Let's all keep in mind that many of the hit recordings by Fats
> > Domino and Little
> > Richard were usually sped up a full key, as were some of Art Garfunkle's hits.
> > This was
> > probably done to shorten the songs' length so they would get radio play, or
> > perhaps to make
> > them more exciting. (KEEP A KNOCKIN' by Little Richard was EXTREMELY exciting.)
> >
> > I'm not against using some studio voodoo here, so please, no lectures on the
> > "aesthetics" of
> > what I want to accomplish .This is a technical question I'm asking you engineers
> > to help me
> > resolve. This is show biz, right? After all, Leonard Bernstein was considered to
> > be a REALLY
> > sensitive artist and conductor. However, he ran around his house in a TUTU. Ha
> > Ha. All art is
> > therefore as fake as show business. Frank Frazetta is a better comic book artist
> > than
> > Michaelangelo was of religious murals on church ceilings. Still, history says
> > that Michaelangelo
> > was an artist and that Frank Frazetta is not.
> >
> > Are there any other suggestions from any of you studio wizards on how to best
> > speed up the
> > performances, with minimal artifacts, and without altering the original keys? Of
> > all the "Time
> > Stretching" tools that are available, which ones are the best? It would save me
> > some time
> > knowing whether to use the tools in Cool Edit Pro, versus, Cuebase, Cakewalk,
> > Sonar,
> > DCLIve, etc. I usually work with Cakewalk and Wavelab, and although I have the
> > other
> > programs, I haven't worked with them much. In fact, I hadn't tried any time
> > stretching until this
> > week, although I've had Wavelab for 6 years.
> >
> > I suppose I could have just done a joint before I listened to the old
> > performances, and I wouldn't
> > have noticed any sluggishess or bothered this board with my problem. But, it's
> > been a long
> > time since I did that. Maybe I should ask my kids if they'll sell me some.
> > Thanks..
> >
> > Alan Cassaro
> >
Anonymous
February 4, 2005 4:52:45 PM

Archived from groups: rec.audio.pro (More info?)

Try it. Time-stretch algorithms keep improving. Waves have a very
good one (at a price:-).

Don't expect miracles. A 15% stretch is a lot, on complex material.
The tape noise won't help. Working at a high sample rate is
certainly worth trying. As is doing the shift in several smaller
steps, not all at once.

A live performance often IS slower than a studio recording. You
worry more about communicating the words to your audience. This may
be a good thing. When I make backing tracks for singers I always
advise they should settle for a slightly higher key and a slightly
lower speed than what feels "right" without an audience.

It's also a common performance fault to take familiar material too
fast. Remember it isn't familiar to the audience.

Sorry you didn't want artistic input. You got it anyway. Like,
that's where I'm coming from, man :-)

CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
Anonymous
February 4, 2005 11:04:01 PM

Archived from groups: rec.audio.pro (More info?)

Well, in terms of actual audible content, nothing over about 20KHz is
going to be audible, of course. However, as all DAC's have a built-in
low pass to prevent sounds over the Nyquist frequency from being
reproduced, a higher sampling frequency can produce a better sound.
The gradient of this filter is fairly sharp when you're sampling at 44
KHz, and sharp filter gradients can produce noticeable distortion of
the upper frequencies (since every filter is - in theory - a delay,
cancellation occurs). Sampling at 96 allows for a more gradual filter
gradient, which contributes less distortion. While distortion will
occur when you dither back down to 44 for CD production, if your gear
is good it will be less than that introduced by a 44KHz sampler, and
you can minimize the effects during recording, mixing, and processing
by recording originals at 96.

Hope this answers the question...
Mike Sayre
Anonymous
February 5, 2005 5:38:47 AM

Archived from groups: rec.audio.pro (More info?)

bsuhorndog wrote:
> Well, in terms of actual audible content, nothing over about 20KHz is
> going to be audible, of course. However, as all DAC's have a built-in
> low pass to prevent sounds over the Nyquist frequency from being
> reproduced, a higher sampling frequency can produce a better sound.
> The gradient of this filter is fairly sharp when you're sampling at 44
> KHz, and sharp filter gradients can produce noticeable distortion of
> the upper frequencies (since every filter is - in theory - a delay,
> cancellation occurs). Sampling at 96 allows for a more gradual filter
> gradient, which contributes less distortion. While distortion will
> occur when you dither back down to 44 for CD production, if your gear
> is good it will be less than that introduced by a 44KHz sampler, and
> you can minimize the effects during recording, mixing, and processing
> by recording originals at 96.
>
> Hope this answers the question...

It does, but incorrectly in several regards.


Bob
--

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

A. Einstein
Anonymous
February 5, 2005 11:22:21 AM

Archived from groups: rec.audio.pro (More info?)

bsuhorndog <bsuhorndog@hotmail.com> wrote:
>Well, in terms of actual audible content, nothing over about 20KHz is
>going to be audible, of course. However, as all DAC's have a built-in
>low pass to prevent sounds over the Nyquist frequency from being
>reproduced, a higher sampling frequency can produce a better sound.
>The gradient of this filter is fairly sharp when you're sampling at 44
>KHz, and sharp filter gradients can produce noticeable distortion of
>the upper frequencies (since every filter is - in theory - a delay,
>cancellation occurs). Sampling at 96 allows for a more gradual filter
>gradient, which contributes less distortion.

This problem was solved with oversampling filters, some time around 1990.
Group delay at the end of the passband is no longer an issue with any modern
converters.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 6, 2005 12:33:56 AM

Archived from groups: rec.audio.pro (More info?)

"Scott Dorsey" <kludge@panix.com> wrote in message news:cu2had$kt9$1@panix2.panix.com...
> bsuhorndog <bsuhorndog@hotmail.com> wrote:
> >Well, in terms of actual audible content, nothing over about 20KHz is
> >going to be audible, of course. However, as all DAC's have a built-in
> >low pass to prevent sounds over the Nyquist frequency from being
> >reproduced, a higher sampling frequency can produce a better sound.
> >The gradient of this filter is fairly sharp when you're sampling at 44
> >KHz, and sharp filter gradients can produce noticeable distortion of
> >the upper frequencies (since every filter is - in theory - a delay,
> >cancellation occurs). Sampling at 96 allows for a more gradual filter
> >gradient, which contributes less distortion.
>
> This problem was solved with oversampling filters, some time around 1990.
> Group delay at the end of the passband is no longer an issue with any modern
> converters.
> --scott


I was wondering... I hadn't noticed (of course, I'm getting old) any massive
distortion on my work done at 44.1, but I did give a pause for the cause of
contemplating this one...

--
David Morgan (MAMS)
http://www.m-a-m-s DOT com
Morgan Audio Media Service
Dallas, Texas (214) 662-9901
_______________________________________
http://www.artisan-recordingstudio.com
Anonymous
February 6, 2005 11:11:46 AM

Archived from groups: rec.audio.pro (More info?)

David Morgan \(MAMS\) <mams@NOSPAm-a-m-s.com> wrote:
>
>I was wondering... I hadn't noticed (of course, I'm getting old) any massive
>distortion on my work done at 44.1, but I did give a pause for the cause of
>contemplating this one...

Try an SV-3700 and you'll hear exactly the sort of problem that the original
poster talks about. The converters on the PCM-1630 are even worse in that
regard. I get the feeling that twenty years down the road we won't be seeing
a lot of "vintage digital" gear fetching high prices on Ebay.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 6, 2005 11:19:41 AM

Archived from groups: rec.audio.pro (More info?)

It what regards? Please explain...I'd like to learn.

Thanks
Mike
Anonymous
February 6, 2005 11:21:35 AM

Archived from groups: rec.audio.pro (More info?)

I apologize if I was incorrect. However, don't we also have to
consider the fact that an oversampling filter also involves some sort
of "mystery dither"? I would like more information on this topic, as
it obviously is a hole in my knowledge.

Thanks
Mike
Anonymous
February 6, 2005 6:59:53 PM

Archived from groups: rec.audio.pro (More info?)

In article <1107706895.018666.64170@o13g2000cwo.googlegroups.com> bsuhorndog@hotmail.com writes:

> I apologize if I was incorrect. However, don't we also have to
> consider the fact that an oversampling filter also involves some sort
> of "mystery dither"? I would like more information on this topic, as
> it obviously is a hole in my knowledge.

You don't need a newsgroup, you need a good book. Principles of
Digital Audio by Ken Pohlman will straighten you out.

There's no mystery dither.

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
February 6, 2005 8:31:44 PM

Archived from groups: rec.audio.pro (More info?)

bsuhorndog <bsuhorndog@hotmail.com> wrote:
>I apologize if I was incorrect. However, don't we also have to
>consider the fact that an oversampling filter also involves some sort
>of "mystery dither"? I would like more information on this topic, as
>it obviously is a hole in my knowledge.

No. Dither is a thing that improves linearity when you are converting
data from a wide word to a narrow word.

Oversampling is basically a way of running the converters at a very high
rate of speed then doing the filtering in the digital domain as you are
converting down to a lower sampling rate. This means you only need
filtering to deal with the high speed converter, not the actual storage
rate.

Pretty much all converters made in the past decade and a half have been
oversampling systems of some sort.

All of this stuff is discussed in the FAQ.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
Anonymous
February 7, 2005 1:40:48 AM

Archived from groups: rec.audio.pro (More info?)

Mike Rivers wrote:

> There's no mystery dither.

Though that could be a great brand name. <g>

--
ha
Anonymous
February 7, 2005 1:47:54 AM

Archived from groups: rec.audio.pro (More info?)

Scott Dorsey wrote:

> All of this stuff is discussed in the FAQ.

http://www.recaudiopro.net


with hats off to Harvey Gerst.

--
ha
Anonymous
February 7, 2005 10:49:52 AM

Archived from groups: rec.audio.pro (More info?)

"Scott Dorsey" <kludge@panix.com> wrote in message news:cu552i$i28$1@panix2.panix.com...

> David Morgan \(MAMS\) <mams@NOSPAm-a-m-s.com> wrote:
> >
> >I was wondering... I hadn't noticed (of course, I'm getting old) any massive
> >distortion on my work done at 44.1, but I did give a pause for the cause of
> >contemplating this one...

> Try an SV-3700 and you'll hear exactly the sort of problem that the original
> poster talks about. The converters on the PCM-1630 are even worse in that
> regard. I get the feeling that twenty years down the road we won't be seeing
> a lot of "vintage digital" gear fetching high prices on Ebay.


Probably not... but I still wouldn't sell my 18 year old Mitsubishi, even for the
$500 it might fetch, unless there's really not going to be any more tape for it.
In which case, I suppose it's not worth anything anyway at that point. Then
again, it sampled at 48K so I guess I'm strolling off topic. <g>

DM
Anonymous
February 7, 2005 11:28:22 AM

Archived from groups: rec.audio.pro (More info?)

"Alan Cassaro" <alanleatherwood@worldnet.att.net> wrote in message
>
> Thank you for your help by not addressing any of my questions,


Alan,

Somewhere within your 6 kilobytes of rambling, you did in fact ask
a couple of audio related questions.

Let's review your questions, shall we.... in order of appearance.

1).

> > > After all it was during the Vietnam War, something we’ll never have
> > > to go through again, right?

2).

> > > Of course, I could just indicate in the liner notes that I was stoned, but
> > > that wouldn't be any fun now, would it?

3).

> > > This is show biz, right?

4).

> > > Are there any other suggestions from any of you studio wizards
> > > on how to best speed up the performances, with minimal artifacts,
> > > and without altering the original keys?

5).

> > > Of all the "Time Stretching" tools that are available, which ones
> > > are the best?


I'll take a shot at a response to numbers 4 and 5.

#4... Reduce the amount of time you are attempting to speed up the
program material until the glitches become minimal or non-existant,
and then consider that a software imposed limit which you deal with.
You could probably also deal with moving a half step up and get a
*whole lot* more out of the results.

#5... You described owning a greater variety of audio software than several
typical audio types combined. I don't mean to be prickish here, either,
but with that sort of array to draw from, you should be telling us which
piece of kit actually works the best for the task. Not being able to do so,
probably leaves leaves a bunch of us wondering if you have more money
than brains, or heaps of cracked software that simply isn't working right,
or you're a man too busy to take the time to learn.

Quite honestly, you're apparently in a position that makes most of us incapable
of answering your question. I can't imagine any single individual having all of the
software you purport to have at your disposal, and not being one helluva geek
with all the answers.

On the serious side, dropping the superfluous 5K of commentary into your
post, which was placed inside another thread and might never be seen by
someone who *might* have a recommendation, sounded and looked a bit
like you'd just blown a big fatty and leaped before looking. Maybe you should
narrow the commentary and re-phrase the question as a new post where it
can more easily be seen. (Pssst... and don't tell people you own that much
software but can't be bothered with taking the time to figure out which one
will work the best for you).

--
David Morgan (MAMS)
http://www.m-a-m-s DOT com
Morgan Audio Media Service
Dallas, Texas (214) 662-9901
_______________________________________
http://www.artisan-recordingstudio.com
Anonymous
February 7, 2005 6:38:31 PM

Archived from groups: rec.audio.pro (More info?)

Thanks David for addressing my questions.Apologies for such a long rambling post. No, I
wasn't stoned, I'm just more hypo manic on some occasions than on others. But, I agree,
I was babbling. It's my experience that most engineers don't really want to hear the
story of our lives, and for a moment I forgot where I was posting. (Just one off topic
foortnote: Over the years in my grilling of many lethargic engineers to find out what
kind of music THEY actually like, I was quite surprised the LIZA MANELLI seems to be
high on the list.??????)
In the meantime, I did find another related post in this group (I forget which one),
and I downloaded Audactity, and some other programs, which seem to be interesting from
my initial tests of speeding up the source material. While I do indeed have a number of
audio software programs, I've never had the occasion to try to "doctor" the speed or
keys or tempos on my tracks exept on one occasion. I thought someone here might have a
"favorite" time stretcher, and that would save me some time in trying out ALL the
programs that do the job, some better than others, I would expect. Raising the key a
half step is a good idea, and as I have suggested in my original post, I am also going
to try speeding the entire track up to the tempo I want during the tape transfer to the
PC, which might raise the key a step and a half, and then try to alter the overall Key
back to the original with a pitch control tool, just to see if that gives me less
artifacts than trying to change the tempo. Thanks again,
Alan

"David Morgan (MAMS)" wrote:

> "Alan Cassaro" <alanleatherwood@worldnet.att.net> wrote in message
> >
> > Thank you for your help by not addressing any of my questions,
>
> Alan,
>
> Somewhere within your 6 kilobytes of rambling, you did in fact ask
> a couple of audio related questions.
>
> Let's review your questions, shall we.... in order of appearance.
>
> 1).
>
> > > > After all it was during the Vietnam War, something we’ll never have
> > > > to go through again, right?
>
> 2).
>
> > > > Of course, I could just indicate in the liner notes that I was stoned, but
> > > > that wouldn't be any fun now, would it?
>
> 3).
>
> > > > This is show biz, right?
>
> 4).
>
> > > > Are there any other suggestions from any of you studio wizards
> > > > on how to best speed up the performances, with minimal artifacts,
> > > > and without altering the original keys?
>
> 5).
>
> > > > Of all the "Time Stretching" tools that are available, which ones
> > > > are the best?
>
> I'll take a shot at a response to numbers 4 and 5.
>
> #4... Reduce the amount of time you are attempting to speed up the
> program material until the glitches become minimal or non-existant,
> and then consider that a software imposed limit which you deal with.
> You could probably also deal with moving a half step up and get a
> *whole lot* more out of the results.
>
> #5... You described owning a greater variety of audio software than several
> typical audio types combined. I don't mean to be prickish here, either,
> but with that sort of array to draw from, you should be telling us which
> piece of kit actually works the best for the task. Not being able to do so,
> probably leaves leaves a bunch of us wondering if you have more money
> than brains, or heaps of cracked software that simply isn't working right,
> or you're a man too busy to take the time to learn.
>
> Quite honestly, you're apparently in a position that makes most of us incapable
> of answering your question. I can't imagine any single individual having all of the
> software you purport to have at your disposal, and not being one helluva geek
> with all the answers.
>
> On the serious side, dropping the superfluous 5K of commentary into your
> post, which was placed inside another thread and might never be seen by
> someone who *might* have a recommendation, sounded and looked a bit
> like you'd just blown a big fatty and leaped before looking. Maybe you should
> narrow the commentary and re-phrase the question as a new post where it
> can more easily be seen. (Pssst... and don't tell people you own that much
> software but can't be bothered with taking the time to figure out which one
> will work the best for you).
>
> --
> David Morgan (MAMS)
> http://www.m-a-m-s DOT com
> Morgan Audio Media Service
> Dallas, Texas (214) 662-9901
> _______________________________________
> http://www.artisan-recordingstudio.com
Anonymous
February 15, 2005 2:21:33 AM

Archived from groups: rec.audio.pro (More info?)

"bsuhorndog" <bsuhorndog@hotmail.com> wrote in message
news:1107576241.907158.310500@c13g2000cwb.googlegroups.com

> Well, in terms of actual audible content, nothing over about 20KHz is
> going to be audible, of course. However, as all DAC's have a built-in
> low pass to prevent sounds over the Nyquist frequency from being
> reproduced, a higher sampling frequency can produce a better sound.

That would be a hypothesis.

> The gradient of this filter is fairly sharp when you're sampling at 44
> KHz, and sharp filter gradients can produce noticeable distortion of
> the upper frequencies (since every filter is - in theory - a delay,
> cancellation occurs).

I'll point out that the audibility of this situation is a presumption, not
one supported by reliable listening tests.

Additionally, FWIW I'll agree with Scott that there's no need for a filter
to have in-band delay.

I'll also add a test report of what should be now a pretty well-known
counter-example:

http://www.pcavtech.com/soundcards/LynxTWO/Ph-loop-1644...

Note that there is no appreciable phase shift in the audio band. It's easy
to find power amps with vastly more phase shift below 20 KHz!
!