Reducing sample rate?

[Kevin Rothi]

I know that when I go down in bit depth I need to apply some kind of dithering depending on the material, but do I need to worry about this if I reduce the sample rate? I'd think not because it's not truncating anything, but rather just dropping samples entirely.

Thanks!

[djsynchro]

Correct, dithering only applies to reducing word length, but in Soundforge (for example) there are quality options for doing the sample rate conversion and the option to switch anti-alias filtering on or off. In Live there are no parameters to set.

[Kevin Rothi]

Correct, dithering only applies to reducing word length, but in Soundforge (for example) there are quality options for doing the sample rate conversion and the option to switch anti-alias filtering on or off. In Live there are no parameters to set.

Um, what?

[zalo]

you are only dropping samples if you divide the sample rate in half, other wise your computer is doing math again

[Kevin Rothi]

you are only dropping samples if you divide the sample rate in half, other wise your computer is doing math again

I really don't understand what you guys are saying here. Is this something I need to worry about or not? Say I have a file that's 24/192, What's the best way to get it to CD quality?

[crumhorn]

you are only dropping samples if you divide the sample rate in half, other wise your computer is doing math again

I really don't understand what you guys are saying here. Is this something I need to worry about or not? Say I have a file that's 24/192, What's the best way to get it to CD quality?

From the manual (page 496)

Rendering audio from Live with a sampling rate other than the one that was used while
working on the project is also a non-neutral operation, and may result in a loss of sound
quality. It is recommended to always render using the original sampling rate, and then
convert the rendered le using a dedicated mastering application that is optimized for
these kinds of CPU-intensive, ofine tasks.
While we recommend that you use a high-quality ofine tool for sample rate conversion,
we recognize that one of Live’s core features is its ability to pitch-shift and warp audio in
real time. For this situation, it is necessary to make a trade-off between CPU performance
and precision. We recommend the use of the Hi-Q button for any clips which undergo
transposition in a given Set. The algorithm behind the Hi-Q switch was rewritten for Live 7,
and now results in considerably lower distortion than in previous versions.

If you are moving from a higher to a lower sample rate and the old sample rate is a whole number multiple of the new one (e.g. going from 96K to 48K) then the process is really simple and any software would create equivalent results.

Going from 48K, 96K or 192K to 44.1K is much harder, this is because the most of the new sample points will fall between the existing sample points, so the software has to interpolate a new sample point between the existing ones.

This interpolation can be done very simply and quickly by assuming that the waveform between the 2 sample points is a straight line - but this would sound awful.

The theoretically correct way is to play it back through a good quality D to A and then resample it with another A to D at the new sample rate. The best software sample rate conversions simulate this process by using a really high intermediate sample rate to simulate the reconstruction filter in a virtual A to D. It's really CPU intensive and still not perfect.

Other methods exist that fall somewhere between these two in quality.

Hope I haven't confused the issue even more. basically use a decent off-line audio app to do your sample rate conversions. Live can't spare the CPU cycles to do it properly.

[Kevin Rothi]

you are only dropping samples if you divide the sample rate in half, other wise your computer is doing math again

I really don't understand what you guys are saying here. Is this something I need to worry about or not? Say I have a file that's 24/192, What's the best way to get it to CD quality?

From the manual (page 496)

Rendering audio from Live with a sampling rate other than the one that was used while
working on the project is also a non-neutral operation, and may result in a loss of sound
quality. It is recommended to always render using the original sampling rate, and then
convert the rendered le using a dedicated mastering application that is optimized for
these kinds of CPU-intensive, ofine tasks.
While we recommend that you use a high-quality ofine tool for sample rate conversion,
we recognize that one of Live’s core features is its ability to pitch-shift and warp audio in
real time. For this situation, it is necessary to make a trade-off between CPU performance
and precision. We recommend the use of the Hi-Q button for any clips which undergo
transposition in a given Set. The algorithm behind the Hi-Q switch was rewritten for Live 7,
and now results in considerably lower distortion than in previous versions.

If you are moving from a higher to a lower sample rate and the old sample rate is a whole number multiple of the new one (e.g. going from 96K to 48K) then the process is really simple and any software would create equivalent results.

Going from 48K, 96K or 192K to 44.1K is much harder, this is because the most of the new sample points will fall between the existing sample points, so the software has to interpolate a new sample point between the existing ones.

This interpolation can be done very simply and quickly by assuming that the waveform between the 2 sample points is a straight line - but this would sound awful.

The theoretically correct way is to play it back through a good quality D to A and then resample it with another A to D at the new sample rate. The best software sample rate conversions simulate this process by using a really high intermediate sample rate to simulate the reconstruction filter in a virtual A to D. It's really CPU intensive and still not perfect.

Other methods exist that fall somewhere between these two in quality.

Hope I haven't confused the issue even more. basically use a decent off-line audio app to do your sample rate conversions. Live can't spare the CPU cycles to do it properly.

Do you have any recommendations on software to use for this purpose?

[crumhorn]

I'm broke so the only audio editor I use is Audacity, which is open source. it uses a method called "High Quality Sinc Interpolation". Not sure what that means

[ethios4]

Be sure to turn on Hi-Q mode in all clips that will have the sample rate changed. This greatly increases the quality of Live's sample-rate conversion.

[djsynchro]

just render at 44.1 16-bit it will be fine, promise!

[Kevin Rothi]

just render at 44.1 16-bit it will be fine, promise!

I can't tell if you're being serious or sarcastic. : (

[ethios4]

He's being serious. There are very fine differences in sample rate conversions, with some softwares being better than others, but it's not anything you'll ever notice. We're talking very very minor differences at this point the game. Be sure to turn on High-Quality mode for each clip (if it's not already) and you'll be fine. If at some point you become such an audiophile that sample rate conversion algorhythms matter to you, you can always invest in some dedicated high-end software.

[Kevin Rothi]

If at some point you become such an audiophile that sample rate conversion algorhythms matter to you, you can always invest in some dedicated high-end software.

What software would that be?

[ethios4]

Here's a site where you can compare different aspects of SRC algorhythms from various softwares.
http://src.infinitewave.ca/
The iZotope 64-bit SRC seems to be the best actually.

[Tone Deft]

just render at 44.1 16-bit it will be fine, promise!

I can't tell if you're being serious or sarcastic. : (

this thread has been up for days and you still don't know what to do?

all this time you should have been trying the various combinations and seeing if you can hear the difference. (you won't be able to hear the difference.)


"Say I have a file that's 24/192, What's the best way to get it to CD quality?"
you mean you want it at a worse quality?