PDA

View Full Version : What is an anti aliasing filter? (Nero Wave Editor 3)


terryfunku
2008-06-28, 04:24 PM
It says I can choose a fast, medium, or accurate anti aliasing filter. I am converting a 24-bit/44.1 khz recording to a 16-bit/44.1 khz recording.

I have no idea what I'm doing, but it must be done, as the show I taped sounds great. My levels were good, my bass roll-off unit did it's job, and there is no clipping or distortion. Nothing really needs to be tweaked.

I don't want to convert it wrong and ruin a good thing. Please help me!

GRC
2008-06-28, 04:57 PM
Google can be your friend; as can Wikipedia;

"An anti-aliasing filter is a filter used before a signal sampler, to restrict the bandwidth of a signal to approximately satisfy the sampling theorem. Since the theorem states that unambiguous interpretation of the signal from its samples is possible only when the power of frequencies outside the Nyquist bandwidth is zero, the anti-aliasing filter would have to have perfect stop-band rejection to completely satisfy the theorem. Every realizable anti-aliasing filter will permit some aliasing to occur; the amount of aliasing that does occur depends on how good the filter is.

Anti-aliasing filters are commonly used at the input of digital signal processing systems, for example in sound digitization systems; similar filters are used as reconstruction filters at the output of such systems, for example in music players. In the latter case, the filter is to prevent aliasing in the conversion of samples back to a continuous signal, where again perfect stop-band rejection would be required to guarantee zero aliasing.

The theoretical impossibility of realizing perfect filters is not much an impediment in practice, though practical considerations do lead to system design choices such as oversampling to make it easier to realize "good enough" anti-aliasing filters."


Aliasing; also from Wikipedia

"In statistics, signal processing, computer graphics and related disciplines, aliasing refers to an effect that causes different continuous signals to become indistinguishable (or aliases of one another) when sampled. It also refers to the distortion or artifact that results when a signal is sampled and reconstructed as an alias of the original signal.

Temporal aliasing is a major concern in the sampling of video and audio signals. Music, for instance, may contain high-frequency components that are inaudible to us. If we sample it with a frequency that is too low and reconstruct the music with a digital to analog converter, we may hear the low-frequency aliases of the undersampled high frequencies. Therefore, it is common practice to remove the high frequencies with a filter before the sampling is done.

Situations also exist where the low frequencies are removed (if necessary), and the high frequency components are intentionally undersampled and reconstructed as lower ones. Some digital channelizers [1] exploit aliasing in this way for computational efficiency; see IR/RF sampling. Signals that contain no low frequencies are often referred to as bandpass or non-baseband."

So, my take is that the filter will suppress the "distortion or artifact that results when a signal is sampled and reconstructed as an alias of the original signal" - but I can't figure out yet if you want fast, med or accurate.....

paddington
2008-06-28, 05:01 PM
ALWAYS choose the slowest, highest quality, most accurate, etc anti-aliasing filter.

When you convert, little "stair-steps" are left in the digital audio. These are unnatural / non-musical and can cause the glassy swishy effect you hear in MP3s.

Anti-aliasing attempts to smooth those out.. the more time you give it, the better, more natural-sounding job it can do.

terryfunku
2008-06-28, 05:30 PM
Thanks James and GRC! I will choose the accurate option then. That seems to be the lowest one.

I did google this for like an hour. And all I got was the same stuff you found. Which may as well have been in Japanese. Because I didn't understand a lick of it. The stuff made no sense to me.

roomful
2008-06-28, 05:41 PM
I think an anti-alias filter is only used when changing the sample rate. You need to convert the bit depth (24/44.1 > 16/44.1), so you need to dither, and leave the sample rate alone. What audio program are you using?

terryfunku
2008-06-28, 05:43 PM
Nero Wave Editor 3. I'm lost. I think I need a drink. This is getting soo confusing. Plus, this program is a bit clumsy. It won't let me delete a track break if I insert one by mistake.

terryfunku
2008-06-28, 06:00 PM
I just want to convert 24-bit to 16-bit. I think I'm just over-complicating things.

roomful
2008-06-28, 06:52 PM
Sorry, I overlooked the thread title. Does Nero Wave Editor have dithering? Couldn't find it. I'd get Audacity for dithering (I think shaped dithering is preferred) and I'd get CD-Wave to split your show into individual tracks. It would be a good idea to archive a FLAC copy of the show at 24/44.1 as well. Both of these programs are used all the time by tapers and traders.

CDWave (free)
http://www.milosoftware.com/cdwave/download.html

Audacity (free)
http://audacity.sourceforge.net/

If you don't like Audacity you could use Soundforge (you can find it for as low as about $50) or something similar for dithering. Wavelab with Apogee UV22HR is the best for dithering.

terryfunku
2008-06-28, 07:07 PM
Thanks. I'll give it a whirl.

dorrcoq
2008-07-02, 03:54 PM
Yeah, do yourself a favour and get a better editing program than Nero.