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MisterE
 
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Default question on sample rate (and conversion etc)

ok, from what i've gathered, usually, converting sample rate seems to be
a no-no. something you want to avoid at all costs. actually it seems
like any time i've asked about it i usually got that response without
any real explanation or what would be the best way to actually do it if
you had to. although, ive read a little and seen mentioned somewhere
that if you are going down to a sampling rate that is exactly half of
the one you started in, that is 'ok'. because the math is more clean, as
there is no estimation required etc. something to that extent anyway.

now, since i learned years ago that it is best to keep my bit depth as
high as possible until the final mixdown, then to dither and finally
convert the bit depth to 16bit if i want to do a cd, ive been working
that way. i understand how the math is better when you are using
plugins/effects etc which are doing all kinds of calculations. its more
accurate to have a bigger word length through all of that. so, ive been
working in 32bit, but still only at 44.1khz.

now, i HAVE seen plenty of talk on the bit depth issue, but not too much
regarding sampling rate, as far as if it would be of benefit to work at
a higher sampling rate, and then convert down at the end, after multiple
software effects processing. i DID see one or two people suggest that it
WAS better, but not enough to sell me on it. most people seem to say its
a bad idea.

so i want to know once and for all, supposing i had softsynths, vst
effects, etc, by the ton. as long as i check each one and see that they
support higher sampling rates, and as long as i make that original
export from my host program at a sampling rate that is exactly twice as
high as the one i want to end up with, would i gain anything from that?
would the fact that the plugins are processing at that higher rate, and
THEN at the end of all that processing, the sample rate is cut in half,
would there be any gain from that?

if so,
a) would i want to convert the sample rate down on each individual track
in the mix after i have applied all of the effects to them, before i mix
them together, or would i mix them the tracks into the final mixdown,
then convert it down? (that's maybe a stupid question)

b) would it really matter what program i used to do the conversion if im
going down exactly by half, and if so, which programs are best for that?

c) what about the conversion up to a rate which is double. say im using
some samples that are 44.1 and have lots of effects on them. if i
exported that at 88.2 id have to assume that first the sample is
upsampled then ran through the vsts at that rate then the result
exported. would that be ok (or worth doing)? or would i only maybe gain
anything from say, a softsynth which actually creates the original sound
at the higher rate?

i guess thats a big chunk of questions, any words of wisdom would be
appreciated.
  #2   Report Post  
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Pettinhouse.com
 
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Default question on sample rate (and conversion etc)

Hi, I would like to help you and I can do it but I think that you are
too confuse. Help us to answer you that we will help to you to find the
answer.
I always say: play, record, enjoy your music.

Pettinhouse sounds for your music
www.pettinhouse.com

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Joseph Ashwood
 
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Default question on sample rate (and conversion etc)

"MisterE" wrote in message
...

Unless your effects are doing some really interesting things (which is
possible, but not likely) you won't see any real benefit to using a higher
sample rate. Try it and see, mix up the same diddle session in 32/176.4 and
32/44.1 and see if there's any quality difference, you might actually
acquire artifacts in the 176.4. I suppose I should give a few words on why I
said 176.4 instead of 192, simple when you convert to 44.1; 176.4 drops 3
out of every 4 samples (or should, but your program might apply smoothing
anyway which could create artifacts), with 192 smoothing will be necessary
and will introduce artifacts. In my opinion, the best way is to know your
equipment, know the artifacts that each operation adds, and decide which
artifacts are desirable in your art.
Joe


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Mike Rivers
 
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Default question on sample rate (and conversion etc)


MisterE wrote:
ok, from what i've gathered, usually, converting sample rate seems to be
a no-no. something you want to avoid at all costs.


Don't people realize that this is a computer thing, and that computer
things change every few minutes? Sample rate conversion used to be a
kind of iffy thing, but now there are many systems that work very well.
If you have to use it, it's not something to worry about.

Anyone who still tells you that he can always hear when something has
been sample rate converted either has a vivid imagination or needs
something better to do.

now, i HAVE seen plenty of talk on the bit depth issue, but not too much
regarding sampling rate, as far as if it would be of benefit to work at
a higher sampling rate, and then convert down at the end, after multiple
software effects processing.


Thaat's because there isn't a lot of strong support for high sampling
rates other than from the manufacturers who have to sell you new
equipment in order to stay in business. The only thing that raising the
sample rate does is gives you wider bandwidth. It's only been very
recently that we've had transducers that have useful response above 20
kHz, and it's still questionable whether there's any value to
reproducing audio above 20 kHz as long as normal humans are listening
to it.

The early justification for a higher sample rate used to be related to
the need to assure that nothing higher in frequency than 1/2 the sample
rate got into the A/D converter, and that everything higher than 1/2
the sample rate got filtered out of the D/A converter output. By using
a sample rate that was 4x the highest frequency that you wan to record
rather than 2x, you could use a filter with a more gradual cutoff
rather than a sharp "brick wall" response curve. This reduced problems
introduced by group delay within the filter and things sounded better.
But we build converters and filters differently today (oversampling
effectively multiplies the sample rate where it matters), and it's no
longer to design a "brick wall" at one half the basic sample rate.

One place where increasing the sample rate is advantageous is where you
have material that isn't generated musically, for instance if you're
restoring a noisy and crackly record. By opening up the bandwidth, you
can more accurately record the clicks that have frequency components
outside the audio bandwidth, and therefore you can more accurately
decide how to remove or reduce them. So, yes, de-noiseing software
still works better at higher sample rates.

This is not to say that no normal person can hear the difference
between 44.1 and 96 kHz audio. Many can. But of those many, most won't
really care, particularly if it means buying new equipment.

so i want to know once and for all, supposing i had softsynths, vst
effects, etc, by the ton. as long as i check each one and see that they
support higher sampling rates, and as long as i make that original
export from my host program at a sampling rate that is exactly twice as
high as the one i want to end up with, would i gain anything from that?


Absolutely not. There is no reason for any of those things to actually
have usable frequency content above 20 kHz. If you were talking about
recording cymbals for the sake of recording cymbals, or if you were
recording a pipe organ with each pipe on a separate track (someone has
actually done this and offers a system for sale at about half the
installed price of a real pipe organ) then there might be an advantage,
but if that's all you're working with, save the disk space. And if you
have to put it on DVD and sample rate convert up, just do it.

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Arny Krueger
 
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Default question on sample rate (and conversion etc)

"MisterE" wrote in message


ok, from what i've gathered, usually, converting sample
rate seems to be a no-no. something you want to avoid at
all costs.


It all depends.

I've long recommended viewing sample rate conversions (SRC) as being pretty
benign. This thinking relies on my long happy experience with Adobe
Audition, which has an excellent reputation for clean SRC. Frankly, I
figured that if Adobe can sell a complete multitracking DAW program with
tons of EFX and noise reduction features for about $300, all of the more
expensive feature-rich products (like Nuendo) would be as good. Recently
somebody posted the results of some diagnostic tests of a number of DAW
programs including Audition. It turns out that it was not possible for me to
give them all the same clean bill of health. The most likely audible problem
related to spurious responses due to incomplete filtering.

Here's a rerun of that discussion:

-------------------- begin long quote ---------------

Pawel Kusmierek" wrote in message

oups.com


Carey Carlan wrote:


Audition does good SRC. It also has good dither. In fact, it has 5
different modes, 11 different noise shapes, and your choice of dither
bits. Have at and try until you find something you like.



Some SRC comparisons between various programs are available he
http://src.infinitewave.ca/




Very interesting.

Only Goldwave was bad enough in the dynamic range test to raise serious
concerns in my mind.


However there were a goodly number of well-known products with response
outside (above) the ideal range, which could IMO lead to audible problems
with aliasing:


They include:


Abletron
Anytime
Barbabatch
Bias
Cubase
Digital Performer
Goldwave
Izotope
Nuendo
Protools
Sequoia
Soundforge
Spark
Weiss (Both)


Shame shame. But, as predicted, performance of the digital filtering was
the most critical factor.


-------------------- end long quote ---------------


actually it seems like any time i've asked
about it i usually got that response without any real
explanation or what would be the best way to actually do
it if you had to. although, ive read a little and seen
mentioned somewhere that if you are going down to a
sampling rate that is exactly half of the one you started
in, that is 'ok'. because the math is more clean, as
there is no estimation required etc. something to that
extent anyway.


This is a myth. There's nothing magical about SRC involving integer
frequency ratios. AFAIK none of the SRC software out there treats them as
special cases.

There's no purpose for any SRC hardware to have special handling for integer
ratios, because integer ratios of clock frequencies pretty much don't exist
in the real world of hardware unless one clock is derived from the other.

You may think that one hardware clock is 44,100.0000000 KHz and another is
88,200.0000000 KHz but in reality they probably vary randomly by up to
0.01%. Anything but precisely exact ratios breaks any magic that might
exist. Therefore, for all practical purposes there is no magic.

now, since i learned years ago that it is best to keep my
bit depth as high as possible until the final mixdown,
then to dither and finally convert the bit depth to 16bit
if i want to do a cd, ive been working that way.


The problem here is that its pretty rare to find situations where the rule
of the weakest link does not dictate final performance. Once you downsample
to 44,100 Hz all the time and money you put into processing at higher clock
rates goes completely out the window. It's kinda like using the finest
coffee beans to make instant coffee and then peeing into the cup just to
make sure it tastes bad. Except, there's no indication that there ever was
anything wrong with 44,100 Hz sampling in the first place.

All things considered the worst thing about working at higher sample rates
is the possibility that your DAW software has a dicy SRC (see list of
suspect SRCs above) and you actually end up with something worse than what
you would have had if you started working at your final SRC and just carried
it through to the consumer.

if so,


a) would i want to convert the sample rate down on each
individual track in the mix after i have applied all of
the effects to them, before i mix them together, or would
i mix them the tracks into the final mixdown, then
convert it down? (that's maybe a stupid question)


In principle a well-written EFX would produce audibly identical results as
long as the sample rate was high enough to cover the audible range. In fact
I've seen exceptions, but they were rare.

b) would it really matter what program i used to do the
conversion if im going down exactly by half, and if so,
which programs are best for that?


Please see the long quote above.

c) what about the conversion up to a rate which is
double. say im using some samples that are 44.1 and have
lots of effects on them. if i exported that at 88.2 id
have to assume that first the sample is upsampled then
ran through the vsts at that rate then the result
exported. would that be ok (or worth doing)? or would i
only maybe gain anything from say, a softsynth which
actually creates the original sound at the higher rate?


"If its not broke, don't fix it". The purpose of an EFX is to create a
synthetic sound. Even if an EFX sounds a little different at different
sample rates, what difference does it make as long as you get a sound you
like?

i guess thats a big chunk of questions, any words of
wisdom would be appreciated.


I'd say that among the poorly-informed the so-called benefits of high sample
rates are overstated. There's no evidence that regular music like just about
all of us record is necessarily signfiicantly altered by recording at higher
sample rates than those used by the *lowly* audio CD. Many of us do
advocate 16 bits when lots of processing is involved, so that you deliver
the best possible dynamic range to the end-user who is listening to 16 bits.

Note that SACD and DVD-A, both of which bet the farm on higher sample rates
and higher dynamic range than CD, are now generally conceeded to have failed
miserably in the marketplace. Some of the people who bet their jobs on them
no longer have those jobs.




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MisterE
 
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Default question on sample rate (and conversion etc)

thank you to all who shared their knowledge about the subject. i now
feel good about staying at 44.1khz. if its good enough for you guys its
good enough for me. just wanted to get it straight.

tim
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Julian
 
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Default question on sample rate (and conversion etc)

On Tue, 28 Mar 2006 08:47:35 -0500, "Arny Krueger"
wrote:

Carey Carlan wrote:

Only Goldwave was bad enough in the dynamic range test to raise serious
concerns in my mind.


However there were a goodly number of well-known products with response
outside (above) the ideal range, which could IMO lead to audible problems
with aliasing:


They include:


Abletron
Anytime
Barbabatch
Bias
Cubase
Digital Performer
Goldwave
Izotope
Nuendo
Protools
Sequoia
Soundforge
Spark
Weiss (Both)


Pro Tools? Sound Forge?

Please explain.

Julian


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Arny Krueger
 
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Default question on sample rate (and conversion etc)

"Julian" wrote in message

On Tue, 28 Mar 2006 08:47:35 -0500, "Arny Krueger"
wrote:




http://src.infinitewave.ca/

Only Goldwave was bad enough in the dynamic range test
to raise serious concerns in my mind.


However there were a goodly number of well-known
products with response outside (above) the ideal range,
which could IMO lead to audible problems with aliasing:


They include:


Abletron
Anytime
Barbabatch
Bias
Cubase
Digital Performer
Goldwave
Izotope
Nuendo
Protools
Sequoia
Soundforge
Spark
Weiss (Both)


Pro Tools? Sound Forge?

Please explain.


Go to

http://src.infinitewave.ca/


Select "transition band"

Select the respective products.

If the green line goes to the right of the white line, that means that
signals 22.05 KHz can be aliased down to below 22.05 KHz.


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Posted to rec.audio.pro
 
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Default question on sample rate (and conversion etc)

Some years ago, I put a track through 19 serial passes of SRC, using
CoolEditPro (now Audition). Comparing to the original, I couldn't hear any
difference.

Anyone can try this.
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Joseph Ashwood
 
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Default question on sample rate (and conversion etc)

"philicorda" wrote in message
news
I wonder if 'lossless' sample rate conversion is possible, in the case
where a file is converted to a higher rate, and then back down to a lower
one?

In general, no. For most purposes though, you can get it close enough to not
matter. For reasonable purposes you can even get it perfect. If you convert
between 44.1 and 88.2 you can do it perfect every time, but if you convert
to 88.201 your samples will very slowly drift no matter how you do it. You
can of course get very close approximations using b and cubic splines for
interpolation, and even to a lesser degree the linear approximation that is
commonly used. The difference will not likely be significantly audible for
the first few dozen conversions (possibly the first few hundred, maybe
thousand), but if you look at the sample values themself you'll see them
start to drift.
Joe


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Bob Cain
 
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Default question on sample rate (and conversion etc)



Joseph Ashwood wrote:

In general, no. For most purposes though, you can get it close enough to not
matter. For reasonable purposes you can even get it perfect. If you convert
between 44.1 and 88.2 you can do it perfect every time, but if you convert
to 88.201 your samples will very slowly drift no matter how you do it.


Drift? To where?

You
can of course get very close approximations using b and cubic splines for
interpolation,


And much much better using a polyphase or a real sinc reconstruction
filter. The accuracy of sinc reconstruction is limited only by the
length of its windowed approximation (a sinc is infinite in both
directions and monotonically decreasing in amplitude away from its center.)

and even to a lesser degree the linear approximation that is
commonly used.


Neither linear approximation nor spline interpolation are used in any
DAW software or plugins. They are the quickest and dirtiest methods
that can possibly be applied. Only if you had a microprocessor with a
very limited calculation rate would such a method be considered.


Bob
--

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

A. Einstein
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