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I always use double sampling rates if possible, for two important reasons. First reason: to get rid of the characteristics of the anti-imaging filter when working with analog sound sources. What is an anti-imaging filter? Let's say I am recording on 44100 Hz. If I would record a sine wave of less then 10 KHz, you could clearly see the sinewave when you ...

I understand why sampling at 44.1 KHz means frequencies approaching 22.05 KHz begin to suffer from sampling artifacts due to the Nyquist theorem. I can't tell the difference, to be honest. Once my signal leaves the digital domain, my analogue gear probably masks a lot of the details in this region and my hearing is definitely not ...

I tend to prefer something higher than 44.1KHz because (like you said) it's pretty close to the minimum useful frequency (according to Nyquist), and I try not to assume that I'm going to be using the audio in the same way all the time. The best example I can think of is timestretching algorithms. I'm a big fan of Ableton Live's warping, and at some point ...

To have headroom for effects is a theoretically (and practically) valid reason to have a higher sampling rate than twice the human hearing limit. The reason for this is easily visualised by comparing with image editing – if you only have say 800x600 px image with an overall shot of a high contrast brick wall, fishnet, striped textiles, or other finely ...

Pelle ten Cate's answer is great. Another possible concern is that the ultrasonic frequencies might become useful later in processing, even though you can't hear them directly in the recording. If you decide to slow something down for whatever reason, those ultrasonic frequencies will become audible frequencies. If you had filtered them out, the slowed ...

Another good reason to use a higher sampling rate is to work around deficiencies of plugin implementations. Many plugin writers do not properly take into account the bandwidth-expanding effects of nonlinear signal operations, and as a result you can get aliasing effects before you leave the box. For example, a compressor is basically a voltage-controlled ...

For what it's worth, the mathematical rationale, at least to the needs of the audio world, is generally described by the Nyquist-Shannon sampling theorem, sometimes just referred to as the Nyquist theorem, which in basic language just states that to fully reproduce a waveform with a max frequency n Hz, you need 2n samples per second.

independently of hearing or not, there's the nyquist frequency, but I don't think that that should be to much of a factor to worry about. If you do hear the difference then do something about it, if you don't just let it be - it'll be less drain on your hard drive, RAM and processor, not to mention have to re-apply algorithms when bouncing etc. And why? ...

When recording with multiple tracks, I believe the bit depth is more important than the sampling rate. So for example, 24bit would be better that 16bit. This has to do with the way that your tracks are mixed down together and something called "rounding errors" when there are not sufficient bits. Most hardware and software can now easily support 96k & ...

I have asked this question to myself a few times in the past, too. If I can be a bit objective after I gained more experience: The difference between 44.1 or 48.2 and 99.6 will never, ever, amount to 0.2% of the difference between an inspired or non-inspired performance that's captured, or whether it's played in a good or a bad instrument. Bit rate is yes ...

You get far more mileage out of going to 24 bit than you do by increasing the sampling frequency, especially when multitracking. The only reason I would go above 44.1 kHz is if I was especially interested in preserving the phase characteristics of very high frequency sounds (cymbals, for example), since the anti-aliasing filters do not have to be so steep ...

If you can't tell the difference, I wouldn't sweat it. You may, someday, be able to tell the difference, but if you can't now, and your clients can't, I think this is a premature optimization of your setup. There may be a technical reason to do so, such as interchange with other systems/people.

I generate audio in the best quality possible, this means for my equipment 24bit/96 kHz and I am "only" doing a podcast. Reason for this is to have all possibilities open for processing after recording. If this is too much data for you to process, the first step could be to copy the original data to a file with less quality. Please check page 15 of this ...