This actually leads to the one reason that higher sampling rates actually has some benefit.

You can design your cutoff filter to roll off much more gently up in the inaudiable range of the spectrum, which leads to less artifacts in the audiable range.

Of course this comes at the cost of 1/2 or 1/4 the plugin performance though. :-(

In most cases decent clocking at 44.1KHz will have a far bigger impact than 88.2KHz sampling rate.

That whole nyquist thing is also why a lot of digital EQs sound ropey and brittle up at the top end of the spectrum. Their EQ Bells are asymetric as they hit nyquist.

The Sony EQ does some pretty hairy maths to counter this and actually can simulate HF bells with the center frequency up to 26KHz which is way above nyquist at 44.1KHz sampling.

They are lurrrverly..

If you use really expensive AD/DA converters you can get around it too. The prism sound AD-1 and DA-1 work internally at about 350 khz. If you can afford 2 of the same converters in the drive bay you can get 16 channels worth of digital audio where it doesnt alias past 175 khz. Course, it will set you back about 12 grand and if you are recording to CD/DVD audio, you will have to dither it down to at least 96 khz anyway for DVD-audio playback. Not a bad way to spend your money though if you have thousands of disposible $$$ bills.

Ultimately most home studio users will work up between 44.1 and 96 khz simply because its much less strain on computer resources. More channels and more instruments and more effects on the go at once and it makes monitoring a mix easier on consumer computers. I think up to a point its more import you get that right before worrying about aliasing...

Im a big advocate of 24/96 though and bouncing as necessary. The extra dynamic range alone at 24 bits is worth it and I'm starting to get used to adding 1 instance of SIR which takes up 33% CPU load.

quote:

Originally posted by -mk- Yes, but D/A converters have filters that cutoff at the nyquist frequency.. for obvious reasons.

Unless they have anti aliasing filters. But then we are getting well into the realms of diminishing returns with regards to how much money you have to spend in order to minimise the whole issue of aliasing.

Anyone do analogue recording to tape?

quote:

Originally posted by funkysouls sure, a Sample Reason file will do it for me. thanks very much

Here's a combinator patch we use.

http://www.vpmrecords.com/reason/
Right click save as FreqAnalyzer.cmb

Very easy -
1. Connect output of whatever it is you wonna analyze to Combi Input
2. Connect Combi Output to whatever input your original output went to (ex. Channel 2 on mix)

P.S. Use decay on BV512 to control fall off rate

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Thank dude!

quote:

Originally posted by DJ Teknique Here's a combinator patch we use. http://www.vpmrecords.com/reason/ Right click save as FreqAnalyzer.cmb Very easy - 1. Connect output of whatever it is you wonna analyze to Combi Input 2. Connect Combi Output to whatever input your original output went to (ex. Channel 2 on mix) P.S. Use decay on BV512 to control fall off rate

that's just teh same as i showed him but I didn't use the combi

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hey booster thanks for tip on the vocoder, im gonna check that out later. I have a question for you. I want to check the frequencies of my different synths, melodies bass etc in my track. Of course you mix your track down by ear as you cannot you an analyser in reason. Would your tip be the best way to check frequencies of individual parts of a track?

I'm not sure what you mean, but if you mean checking freq of parts of your track you can connect absolutely anything to the vocoder to analyze it...

hope that helps...?

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How do you use the vocoder as an actual vocoder?

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==> Hard Mix - 01/10/04 <==

connect it like you connect any other FX

then put something to the extra modulator in (some string/pads could do good) you'll see quite easily :-)

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quote:

Originally posted by dj_palm well not all overtones beacuse a baseline at 100Hz can have an harmonic overtone on 200Hz, 400Hz and 800Hz and all other freqs, this is what makes the actual sound.

Allow me to add 300, 500, 600 & 700, so nobody gets the wrong idea (unless it's some special waveform, like the square that was mentioned, which has no first harmonic because of... well symmetry essentially, which causes that first harmonic fourier coefficient to vanish).
Thanks for this thread, people, took 25 mins of my life, but it's nice to see such no-bullshit threads. Thumbs up!