Any use of going beyond 44100?

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bmanic wrote:yeah, distortion is one of those that always seems to benefit (unless you want it to sound gritty) and same goes for 99% of the EQ plugins out there.
- bManic
When talking about the distortions gritty = aliasing noise.

oh, btw, "EIDT" ??? :)

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OK, how about a related question; what would have a more drastic effect on recorded [live] material moving to a higher (let's say 96k) samplerate on a prosumer card [M-Audio and the likes] or buying a high end converter [Apogee, Lucid, Lavry] and staying at 44.1k? I assume it will be the latter correct? Now can we get a bit into how the two relate to each other? I mean will I loose the benefits of a good apogee converter at 44.1k as soon as I start mixing, or will it still be there by the end (I'm guessing yes). Secondly, if all this stuff about 96k representing sine waveforms near nyquist better than 44.1k is true than why will a good 44.1k converter still outperform a mediocre 96k converter?

I'm asking because I'm looking to buy an external converter to save me from increasing samplerate as I'm hearing a common sound [a sort of graininess] in my recordings that I'm not happy with, whereas I know my mics [Rode, Blue, Shure] and pre's [Grace] are quite good. The problem of increasing samplerate is that I'm not sure how great a benefit I'd recieve from an M-Audio Delta 66 converter at 96k than at 44.1k (in theory it could sound worse), and I'd run out of CPU power quickly at mixing at time in a project with 30+ tracks with lots of compressors and EQs as well as the stock synth or two, delay, and reverb.
I'm sorry this post wasn't about techno.

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pj geerlings wrote:It all comes down to this: higher sample rates have much broader transition bands.

The transition band is the frequency band between the highest frequency you care about (typically 20KHz) and the actual Nyquist-Shannon frequency that is always half of the sample rate.

In the case of a 44.1K sample rate the transition band may be specified as 22.5K/20K which equals 1.125 – in log base 2 terms this value works out to 0.17 of an octave. A real-world filter that works over a transition band this small is very difficult to design. (These days, oversampling techniques are used in the DAC hardware to relax the filter requirements for any sampling rate – but for the moment lets not consider this aspect of the issue)

If we bump up the sample rate to 48K things get better. 24K/20K equals 1.2 – or 0.26 of an octave. Designing a filter to work over this transition band is not nearly as difficult. In fact, this seemingly tiny bump in sampling rate reduces the required filtering complexity by an order of approximately 2 to 1.
This is a great explanation! In thinking about this issue, a question came to mind: Given the importance of the transition band size, do you see any advantage to recording at 48KHz instead of 44.1KHz, then using something like Voxengo's R8brain Pro Sample Rate Converter to go to 44.1Khz?

Or, would this additional step degrade the audio and therefore offset the benefit of the higher sample rate? :shock:
Last edited by EnzymeX on Wed Dec 29, 2004 6:53 pm, edited 2 times in total.

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_starcraft_ wrote:i thought it was all nonsense before trying it.....but indeed SOME vstis do sound alot better at 96hz.
wether this is due to poor code or not i have no knowledge to judge......but the sound at 96hz is clearly fuller.
it would solve most problems if all vsti supported oversampling.
for everything else 44.1 hz seems to be just fine.
It's not due to poor coding-it's due to GOOD coding.
Although it's said that the human ear only perceives up to 20k or so,I think our brains perceive frequencies way beyond that.That's what make the old Moog modulars so special sounding-they went out beyond 30k in frequency response.Put a Moog modular against your traditional hardware synth from the 80s-90s and the other synth will sound like there's a blanket in front of the speakers...even though the newre synth is going out to 20k or so...
ew
A spectral heretic...

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stefancrs wrote:
bmanic wrote:yeah, distortion is one of those that always seems to benefit (unless you want it to sound gritty) and same goes for 99% of the EQ plugins out there.
- bManic
When talking about the distortions gritty = aliasing noise.

oh, btw, "EIDT" ??? :)
:hihi:

Hehe, sorry.. I'm sure you knew I wanted to say EDIT: :-o :D

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EnzymeX wrote:Question: Given the importance of the transition band size, do you see any advantage to recording at 48KHz instead of 44.1KHz, then using something like Voxengo's R8brain Pro Sample Rate Converter to go to 44.1Khz? Or would this additional step instead degrade the audio and therefore offset the benefit of the higher sample rate? :shock:
These days I always use 48K - but that's just me ;)

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I see Vanguard was mentioned several times in regards to sounding better at higher rates.... :shock:

Oh, come ooooooooooooonnnnnnn!
The majority of Vanguard users are trying to get 3-oscillator, stacked tarnce leads at 55-voice unison mode. How the f**k can anyone hear any difference with those type of sounds even if it's recorded at 8bit/33KHz ?

:hihi:

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pj geerlings wrote:
EnzymeX wrote:Question: Given the importance of the transition band size, do you see any advantage to recording at 48KHz instead of 44.1KHz, then using something like Voxengo's R8brain Pro Sample Rate Converter to go to 44.1Khz? Or would this additional step instead degrade the audio and therefore offset the benefit of the higher sample rate? :shock:
These days I always use 48K - but that's just me ;)
Thanks for the response, pj. If you don't mind, I have two more quick questions:

-How do you downsample/dither to 44/16?
-Do you notice degredation when you downsample (compared to, say, a straight dither from 44/24 to 44/16)?
Last edited by EnzymeX on Wed Dec 29, 2004 7:27 pm, edited 2 times in total.

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EnzymeX wrote:
pj geerlings wrote:
EnzymeX wrote:Question: Given the importance of the transition band size, do you see any advantage to recording at 48KHz instead of 44.1KHz, then using something like Voxengo's R8brain Pro Sample Rate Converter to go to 44.1Khz? Or would this additional step instead degrade the audio and therefore offset the benefit of the higher sample rate? :shock:
These days I always use 48K - but that's just me ;)
Sound like a good idea for DVD compatibility also. If you don't mind, I have three more quick questions:

-Do you record at 48/32 or 48/24?
-How do you downsample/dither to 44/16?
-Do you notice degredation when you downsample (compared to, say, a straight dither from 44/24 to 44/16)?
DVD is 96kHz/24 bit. DVCam is 48kHz though, and the reason why I'm set to 48kHz.

Devon
Simple music philosophy - Those who can, make music. Those who can't, make excuses.
Read my VST reviews at Traxmusic!

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DevonB wrote: DVD is 96kHz/24 bit. DVCam is 48kHz though, and the reason why I'm set to 48kHz.
Devon
Thanks, DevonB

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kritikon wrote:I see Vanguard was mentioned several times in regards to sounding better at higher rates.... :shock:

Oh, come ooooooooooooonnnnnnn!
The majority of Vanguard users are trying to get 3-oscillator, stacked tarnce leads at 55-voice unison mode. How the f**k can anyone hear any difference with those type of sounds even if it's recorded at 8bit/33KHz ?

:hihi:
Sounds bloody ugly at 44khz, and does not at 96khz. Besides your genre issues, I'm sure you'll hear the difference. And even though it might be popular to bash 'tarnce' the elaborations don't exactly make your own level of expertise shine brightly.

Markus

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pj geerlings wrote:
EnzymeX wrote:Question: Given the importance of the transition band size, do you see any advantage to recording at 48KHz instead of 44.1KHz, then using something like Voxengo's R8brain Pro Sample Rate Converter to go to 44.1Khz? Or would this additional step instead degrade the audio and therefore offset the benefit of the higher sample rate? :shock:
These days I always use 48K - but that's just me ;)
Very wise indeed. It'll sound better. Especially if you run a lot of tracks with lots of EQ etc, then it will sound quite a lot better.

Cheers!
bManic

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YASD, yet another samplerate discussion!

For a starter, I would like to refer here:
http://www.kvraudio.com/forum/viewtopic ... c&start=15

About Connect-the-dots
Transfer of a string of number from the digial domain to a continuous signal is not connect the dots.
Audition/Cooledit has is very good demonstration of this effect. You can zoom into the wave until you can see the sample points, plus you can see the corresponding continuous wave resulting from the sample points (be surprised!).
If you want the ultimate kick, try converting sample rates to 96 kHz or even down to 22 kHz and observe that the continuous wave hasn't changed shape :-o (as long as no signal energy was lost).


Now the difference of cheap vs expensive sound hardware is it's ability to make the continuous wave from the sample points as good as Cooledit can do.


Why not make the same sample rate experiments with an aliasing free synthesizer, like the IBlit, PolyIBlit, Xhip or the like? Then report you can hear a difference between running IBlit at 44.1 and running it at 96?

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ew wrote: It's not due to poor coding-it's due to GOOD coding.
Although it's said that the human ear only perceives up to 20k or so,I think our brains perceive frequencies way beyond that.That's what make the old Moog modulars so special sounding-they went out beyond 30k in frequency response.Put a Moog modular against your traditional hardware synth from the 80s-90s and the other synth will sound like there's a blanket in front of the speakers...even though the newre synth is going out to 20k or so...
ew
i really don't believe in this "other level of listening" theory.....
I just hear what my ear allows me to hear.
I still believe its to do with code being more efficient with higher sampling rates.....but just cos it aids its calculations and helps deal with this alias stuff.we are talkin about generated sound here.i think its more to do with what the computer can hear (numbers) ....than what we can hear as humans.
if i record my voice at 44.1 or 96 it makes no difference whatsoever(no matter how good the mic is).
a dev hopefully will clear this up for us.....

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I'm set to 24 bit 48 Khz with Cubase SX and an Echo Mia sound card. The difference from my old PC (same sound card) running at 16 bit and 44.1 Khz is definitely noticeable.

You simply have to dither your master down to 44.1 and 16 bit when everything is done.

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