correct. but find me a way to do this for f.e. analog like resonant filters with a tan-h (or whatever approximation for the drive) in the feedback path... imo the only way in this case is oversampling, if you would want the satureation/harmonics to be applied to the full frequency response of the filter... no? please correct me if i'm wrong....bootsie wrote:word!DaveGamble wrote: - best solution: find ways to avoid generation of aliasing without oversampling
Is realtime oversampling a dirty/destructive process?
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- KVRAF
- 6168 posts since 26 Sep, 2003 from right here, as you can see ...
regards,
brok landers
BIGTONEsounddesign
gear is as good as the innovation behind it-the man
brok landers
BIGTONEsounddesign
gear is as good as the innovation behind it-the man
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tony tony chopper tony tony chopper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=3103
- KVRAF
- 3561 posts since 20 Jun, 2002
I'd like to hear just 1 bit of sound (anything) that sounds different with its content above 20khz fully filtered.in practice there's not too many signals, especially transient rich signals, that sound good when the transient is reaching up to 21khz
I'd even suggest 16-17khz, but I'm 37 & there may be kids who can really hear the difference. But with 20khz it's pretty safe.
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- KVRian
- 1169 posts since 24 Feb, 2012
That may be true, but I don't have access to such filtering technology. Fact is, even the steepest filters ALWAYS have a smooth transition. Keeping this transition away from the audible band is the reason why you want to tune your nyquist filters a little higher (or better: as high as possible). Just try it out. Try to build a filter that fully filters content above 20kHz.tony tony chopper wrote:I'd like to hear just 1 bit of sound (anything) that sounds different with its content above 20khz fully filtered.in practice there's not too many signals, especially transient rich signals, that sound good when the transient is reaching up to 21khz
Fabien from Tokyo Dawn Records
Check out my audio processors over at the Tokyo Dawn Labs!
Check out my audio processors over at the Tokyo Dawn Labs!
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tony tony chopper tony tony chopper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=3103
- KVRAF
- 3561 posts since 20 Jun, 2002
Why not? You can make a FIR lowpass as giant as needed.Try to build a filter that fully filters content above 20kHz.
I was only saying that whatever is above 16-17khz hardly matters.
DOLPH WILL PWNZ0R J00r LAWZ!!!!
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- KVRAF
- 6168 posts since 26 Sep, 2003 from right here, as you can see ...
well, i wouldn't go that far and say it doesn't matter, but i for one would doubt to hear a change of -3db above 20khz, let alone the phase shift above 20khz. and even if i would hear it, i assume i'd actually prefer the shifted, minus-3db-at-21khz signal....tony tony chopper wrote:Why not? You can make a FIR lowpass as giant as needed.Try to build a filter that fully filters content above 20kHz.
I was only saying that whatever is above 16-17khz hardly matters.
regards,
brok landers
BIGTONEsounddesign
gear is as good as the innovation behind it-the man
brok landers
BIGTONEsounddesign
gear is as good as the innovation behind it-the man
- KVRian
- 1169 posts since 24 Feb, 2012
Again, that may be fine for some signals. But in practice, the process of filtering "whatever is above" will ALWAYS affect "whatever is below" as well. That's my point. And it becomes clear as soon you leave theory and try it yourself.tony tony chopper wrote:I was only saying that whatever is above 16-17khz hardly matters.
Fabien from Tokyo Dawn Records
Check out my audio processors over at the Tokyo Dawn Labs!
Check out my audio processors over at the Tokyo Dawn Labs!
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tony tony chopper tony tony chopper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=3103
- KVRAF
- 3561 posts since 20 Jun, 2002
how?But in practice, the process of filtering "whatever is above" will ALWAYS affect "whatever is below" as well.
but I've done it many timesThat's my point. And it becomes clear as soon you leave theory and try it yourself.
can you post an example?
DOLPH WILL PWNZ0R J00r LAWZ!!!!
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- KVRAF
- 16977 posts since 23 Jun, 2010 from north of London ON
mmmm...I wonder if this would affect bat range hearing....
Barry
If a billion people believe a stupid thing it is still a stupid thing
If a billion people believe a stupid thing it is still a stupid thing
- KVRian
- 1169 posts since 24 Feb, 2012
Beside obvious things such as filter passband ripple, the smooth transition region, and maybe phase shift, don't forget the Gibbs phenomenon. The latter isn't audible, but its time-domain implications can turn audible (e.g. via overloads).tony tony chopper wrote:how?But in practice, the process of filtering "whatever is above" will ALWAYS affect "whatever is below" as well.
It makes no sense to over generalize these things. Reducing the bandwidth introduce side-effects as a matter of fact. No matter if you like them or not.
Last edited by FabienTDR on Mon Mar 04, 2013 3:06 pm, edited 1 time in total.
Fabien from Tokyo Dawn Records
Check out my audio processors over at the Tokyo Dawn Labs!
Check out my audio processors over at the Tokyo Dawn Labs!
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tony tony chopper tony tony chopper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=3103
- KVRAF
- 3561 posts since 20 Jun, 2002
but with a massive FIR that would be less than neglectable.Beside obvious things such as filter passband ripple, the smooth transition region
Again I'm not arguing about filtering, but about this sentence
which has nothing to do with the question whether you can perfectly filter above 21khz (and the answer is yes, you can, you just can't do without a massive latency but you can)in practice there's not too many signals, especially transient rich signals, that sound good when the transient is reaching up to 21khz
obviously yes what's at 20khz matters if you process it further, but that wasn't really the question.The latter isn't audible, but its time-domain implications can turn audible
DOLPH WILL PWNZ0R J00r LAWZ!!!!
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- KVRAF
- 1724 posts since 10 Feb, 2008 from Berlin, Germany
Have you ever verified that impression in a blindtest?bmanic wrote:I can't quite put my finger on it but it's a similar constant annoyance like when a client delivers some song for mastering and he/she hasn't used any dithering when it's required but instead has truncated everything. It's a nasty harshness, usually manifesting itself from about 2kHz onward. Especially noticeable once you start hitting the final mastering limiter/ADDA loop hard.
You really had to mix at low levels and render out in 16bit to ever have an issue with truncation errors.
And nobody would ever send a 16bit file to mastering.
You forget two things here:bmanic wrote:Lets say you have quite a few tracks in a production.. a typical number would be perhaps 50 tracks playing simultaneously (this is quite common for most pop/rock things with a lot of multitracked content). Now you have some compression on most tracks at some point in time. They all have annoying aliasing/nasty stuff below -100dBFS.. at some point in the song you will have some of these inharmonic peaks cancel each other out and at some points they will reinforce each other, basically doubling their volume.
1. Loudness of unrelated signals won't add up indefinetly.
2. The loudness of the actual audio signals (100dB louder than your "nasty stuff") will add up as well. So the relative loudness of your errors stays about the same.
Huh?tony tony chopper wrote:But if your compression envelope is too fast.. it gets in the audible range.. and I don't think that you want a compressor produce "sound" by itself.
I thought, that's what compressors are for.
I totally love the snappy sound with shorter attack-times of some compressors.
+1tony tony chopper wrote:I'd even suggest 16-17khz, but I'm 37 & there may be kids who can really hear the difference. But with 20khz it's pretty safe.
I'm 27 now, and stuff above 20kHz is inaudible to me.
(And I use earplugs to protect my ears in nightclubs.)
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- KVRian
- 1074 posts since 12 Jul, 2009 from Brighton
Ah, to clarify: My original statement wasn't something that one can have an opinion on - multiplying together two full bandwidth signals only generates intermodulation signals that go up to twice nyquist, so x2 in the audio path is all that's necessary. This is assuming, of course, that your VCA is just that - a variable gain stage.bmanic wrote:Not so sure about this. I've done a lot of testing of this and feel that it very much depends on many things:DaveGamble wrote: Dynamics processing doesn't need oversampling by more than x2 in the audio path.
[snip]
Finally, I agree that the most critical component is to oversample the actual sidechain/VCA block thing.. not sure if there is much benefit in oversampling the actual audio path too much.
But, as you say, oversampling the process that determines what gain the VCA should apply does indeed appear to offer benefits.
Dave.
[ DMGAudio ] | [ DMGAudio Blog ] | dave AT dmgaudio DOT com
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- KVRian
- 1074 posts since 12 Jul, 2009 from Brighton
Well, this is precisely the thing. It would be pessimistic to imagine that we'll never have good solutions to this problem.brok landers wrote:correct. but find me a way to do this for f.e. analog like resonant filters with a tan-h (or whatever approximation for the drive) in the feedback path... imo the only way in this case is oversampling, if you would want the satureation/harmonics to be applied to the full frequency response of the filter... no? please correct me if i'm wrong....bootsie wrote:word!DaveGamble wrote: - best solution: find ways to avoid generation of aliasing without oversampling
You already might use an RK-based strategy?
But finding you a way to do this, even though there isn't an obvious way now, is likely going to be one of the big research topics for us all over the next few years.
Either that, or we'll all sink into oversampling complacency until some bright spark comes along with something clever.
Dave.
[ DMGAudio ] | [ DMGAudio Blog ] | dave AT dmgaudio DOT com
- KVRAF
- 10361 posts since 3 Feb, 2003 from Finland, Espoo
Many many times. I do ABX tests all the time. A shootout between 3x tracks of The Glue set to 64x oversampling versus the same 3 tracks set to no oversampling is very easy to pick out.Nokenoku wrote:Have you ever verified that impression in a blindtest?bmanic wrote:I can't quite put my finger on it but it's a similar constant annoyance like when a client delivers some song for mastering and he/she hasn't used any dithering when it's required but instead has truncated everything. It's a nasty harshness, usually manifesting itself from about 2kHz onward. Especially noticeable once you start hitting the final mastering limiter/ADDA loop hard.
However, it might be also due to the oversampling itself.
I've also done plenty of blind "taste" AB tests where I do not concentrate on the differences but simply select the one I feel is better sounding. These usually end up being the heavily oversampled ones.. but not always. There is a lot of material that does suffer from the actual oversampling process. Sometimes things can sound a bit strained or less punchy. Not sure why this happens either as the filtering that needs to be done when oversampling is at such high frequencies.
Cheers!
bManic
"Wisdom is wisdom, regardless of the idiot who said it." -an idiot
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tony tony chopper tony tony chopper https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=3103
- KVRAF
- 3561 posts since 20 Jun, 2002
to me a compressor is an "auto gain", but if the gain envelope is going at such a high rate that it enters the audible, it's not just a gain anymore, it colors the sound. Which may not be a problem, but I wouldn't go as far as saying that it's what compressors are for.I thought, that's what compressors are for.
DOLPH WILL PWNZ0R J00r LAWZ!!!!