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  #21  
Old 07-13-2010, 06:45 AM
daeron80 daeron80 is offline
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

It's both, of course. Coarser is coarser and finer is finer. Finer = more accurate representation, more detail.
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  #22  
Old 07-13-2010, 11:39 AM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by chrisdee View Post
Is this correct ? I can defenetly hear a big difference especially on reverb between 44 and 96KHz at the same bit bandwidth (24bit).
As you know, a higher sampling rate does not "improve", "add detail", or "add resolution" to lower frequencies. A 10k tone will look and sound the same at 44.1, 48, 96 or 192 on reproduction. Since the Nyquist Theorem states that sounds sampled at less than half the sample rate will be reproduced "without error", you can't get any finer or better than "no error". The sound is the same.

Your music between 20 and 20k will sound the same if you use any sampling rate above 40k. Of course we have to take into account the low pass filter to avoid aliasing and that is why the sampling rate is above 40k: to have a more gentle roll off to avoid ringing.

Some may feel that moving the filter higher in frequency, like with 88.2 or 96, will reduce ringing. It can if the filter is properly designed. Also, since there is so little musical energy above 20k, there is little there to cause ringing (straight muted trumpets accepted).
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  #23  
Old 07-13-2010, 12:20 PM
daeron80 daeron80 is offline
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

You're right that less corner ring could influence the perception of clarity, but there's also less phase shift and ripple to consider. Any or all could contribute to verbs sounding better in higher rate sessions.

Quote:
Originally Posted by Park Seward View Post
Since the Nyquist Theorem states that sounds sampled at less than half the sample rate will be reproduced "without error", you can't get any finer or better than "no error". The sound is the same.
To me, that's just another example of why not all mathematicians are great recording engineers. What Nyquist calls "without error" does not necessarily equal what we call "good sound." Something can be mathematically error-free from a given perspective and yet have audible inadequacies - not because there's anything wrong with the math, but because the theory may have failed to take some relevant factor into account.

If Nyquist had explained all we needed to hear, many good plug-ins wouldn't upsample prior to processing. The Theorem may be all you need to know to get sound into and out of the digital domain. But if you do anything to it while it's there, you start to reveal its shortcomings. Or rather, you start to reveal the incredible sensitivity of the human ear.

The following is just thinking out loud. I've never tried programming a reverb, so I don't really know for sure how they work. But when you pass digital audio words to a piece of software, it doesn't necessarily "hear" them the way the human ear hears them after reconstruction. It operates on the bit stream with an algorithm. From the perspective of an algorithm, I imagine that an 11 kHz tone drawn with 8 words per cycle is going to be more accurately represented than one drawn with 4 words per cycle. That sort of thing, intuitively, seems like it should result in a smoother sounding reverb. No?
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  #24  
Old 07-13-2010, 01:30 PM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

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Originally Posted by albee1952 View Post
The math to convert form 88.2K to 44.1K is simple, vs the math to convert from 96K to 44.1K.
Just want to go back and point out this is urban legend. It is simply not true. they are both equally as easy and error free.

What effects the sound of the conversion is not the actual conversion to the new sample rate, it is the reconstruction filter at the Nyquist frequency. But this holds true for any sampling, not just sample rate conversion. The crappier the filter is, the worse it will sound.
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  #25  
Old 07-13-2010, 02:52 PM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by daeron80 View Post
I imagine that an 11 kHz tone drawn with 8 words per cycle is going to be more accurately represented than one drawn with 4 words per cycle. That sort of thing, intuitively, seems like it should result in a smoother sounding reverb. No?
No. The reconstruction filter will recreate the same output as long as the sound is less than half Nyquist. The sound will be the same. A higher sampling rate, by itself, will have no effect.
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  #26  
Old 07-13-2010, 03:51 PM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by O.G. Killa View Post
Just want to go back and point out this is urban legend. It is simply not true. they are both equally as easy and error free.

What effects the sound of the conversion is not the actual conversion to the new sample rate, it is the reconstruction filter at the Nyquist frequency. But this holds true for any sampling, not just sample rate conversion. The crappier the filter is, the worse it will sound.
"The's also why resampling at integer multiples or divisors would be less likely to distort, the math is simpler and less prone to round-off error. If we're downsampling from 96 to 48 all we need do is throw away every other sample. If we're upsampling, each new sample lies exactly halfway between the old and all we have to do is set its value to the average of the two old samples on either side of it. But if that new sample has to be inserted at .345654 of the time between two old samples and the next one is at .897667 of the time difference, and only a few of the new samples fall exactly on the same time mark as the old ones, now we got the problem in interpolation."

http://www.dvinfo.net/forum/all-thin...important.html

Sampling rate conversion is simplified if rates are integer multiples of each other.

Here are some interesting charts comparing SRC with different manufacturers:

http://src.infinitewave.ca/
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  #27  
Old 07-13-2010, 05:31 PM
spicemix spicemix is offline
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by Park Seward View Post
"The's also why resampling at integer multiples or divisors would be less likely to distort, the math is simpler and less prone to round-off error. If we're downsampling from 96 to 48 all we need do is throw away every other sample. If we're upsampling, each new sample lies exactly halfway between the old and all we have to do is set its value to the average of the two old samples on either side of it. But if that new sample has to be inserted at .345654 of the time between two old samples and the next one is at .897667 of the time difference, and only a few of the new samples fall exactly on the same time mark as the old ones, now we got the problem in interpolation."

http://www.dvinfo.net/forum/all-thin...important.html

Sampling rate conversion is simplified if rates are integer multiples of each other.

Here are some interesting charts comparing SRC with different manufacturers:

http://src.infinitewave.ca/
You are quoting some guy who for all I know is a home theatre consumer and has no professional audio expertise whatsoever.

Go ahead and try your sample dropping approach and let us know how it sounds.

Today those factors are irrelevant in the better algos.
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  #28  
Old 07-14-2010, 10:26 AM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by Park Seward View Post
"The's also why resampling at integer multiples or divisors would be less likely to distort, the math is simpler and less prone to round-off error. If we're downsampling from 96 to 48 all we need do is throw away every other sample. If we're upsampling, each new sample lies exactly halfway between the old and all we have to do is set its value to the average of the two old samples on either side of it. But if that new sample has to be inserted at .345654 of the time between two old samples and the next one is at .897667 of the time difference, and only a few of the new samples fall exactly on the same time mark as the old ones, now we got the problem in interpolation."

http://www.dvinfo.net/forum/all-thin...important.html

Sampling rate conversion is simplified if rates are integer multiples of each other.

Here are some interesting charts comparing SRC with different manufacturers:

http://src.infinitewave.ca/
All of the effects seen in the Infinitewave graphs are due to the filters used. they explain that in the FAQ and Help.

ABSOLUTELY NO PROGRAM EVER DIVIDES BY 2 TO DOWNSAMPLE. NONE. NOT GOING TO HAPPEN. IT IS A MYTH. This is exactly the reason why I posted about this. There are so many people that still don't understand how this actually works or have been told incorrectly and then adhere to the wrong information as though it is fact.

Do this little math problem for me please.

44100 x 160 = ?
48000 x 147 = ?

What are the answers to the equations above? Noticing anything similar about the answers? What do you think that means in terms of sample rate conversion?
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  #29  
Old 07-14-2010, 10:33 AM
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

and by the way, that DVinfo forum link is sooooo far off the mark it's scary. They guy is telling people to add DITHER when sample rate converting!?!?!?! Wow... that is scary. These guys in that link are all novice/hobbyist people. Don't put too much stock into what they are saying, because after reading through the thread pretty much all of the info they mentioned was wrong (except for the fact that DVDs use 48KHz and CDs use 44.1KHz).
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  #30  
Old 07-14-2010, 10:46 AM
daeron80 daeron80 is offline
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Default Re: 44.1 kHz vs. 48 kHz - why not use the higher?

Quote:
Originally Posted by Park Seward View Post
No. The reconstruction filter will recreate the same output as long as the sound is less than half Nyquist. The sound will be the same. A higher sampling rate, by itself, will have no effect.
OK, so you send a digital audio stream (from a track, via an aux bus) to a reverb plug-in. The plug-in operates on that stream with an algorithm and outputs the result. Where in that scenario are reconstruction filters used? How is it that the algorithm would produce the same result given different data to process?
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