SAMPLING RATES

sound_designer

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Sampling Rate

Digital sound is produced by sampling a sound (or should I say the electrical version of it) in real time and expressing it in bit words. Once you start sampling or recording digital sound a clock starts and progressive samples of what the sound is are taken. The rate at which the samples are taken is called the sampling rate.

Why 44.1kHz Sampling Rate?

Why not 44, or a nice round number like 50. When the first engineers were inventing digital sound they had worked out the on/off, 0/1, idea and needed a way to represent it. The idea came to use white dots on a TV screen where a white dot was on and a black dot was off. Neat. So you record it like a video picture on a video recorder. That was fine, but the engineers had been caught out before. What about PAL (the European video standard) and NTSC? (the American and Japanese standard.) They weren't going to get caught up in that again, no way, so they configured a number that was compatible between the 528 line NTSC and 625line PAL and the number was 44.1kHz. Just a piece of useless info you might want one day!

What you can see from the above is how the digital recorders were developed. They were Beta Video Recorders with an external processor and digital audio had arrived. The beta video became the DAT and the DAT became the ADat and the D88 and they are all basically video recorder decks. The ADat used a SuperVHS deck while the D88 used a High8 deck. The basic SuperVHS deck was pretty awful and the ADat of today is a completely rebuilt deck. It's a shame the world chose to make the VHS deck the standard because the Beta Decks were far superior. Market forces don't always give the best outcome. Did you know that when an ADat or D88 records on a new track it plays the bit stream off the tape , mixes in the new track, and records it again. Now that's worth thinking about.

In an article by Rupert Neve, I read recently, he said that we should aim for 24bit resolution and 192kHz sampling rate if we want to equal the quality of high quality analogue recording. We will get there. DVD is already up to 24 bit 96kHz sampling so we are on the way. But if your 16bit, 44.1kHz CD sounds bright, consider what makes it bright and you will see that it's a false bright created by the high frequencies sounding like square waves!!
 
The word size is also important. With 24 bit, you'll have a much higher fidelity to the sound. Sound samples converted to a 16bit word has less precision than 24bits. While it may not be immediately apparent, frequent audio manipulation will exacerbate the quantisation errors that occurs when rounding up.
 
sound_designer said:
In an article by Rupert Neve, I read recently, he said that we should aim for 24bit resolution and 192kHz sampling rate if we want to equal the quality of high quality analogue recording. We will get there. DVD is already up to 24 bit 96kHz sampling so we are on the way. But if your 16bit, 44.1kHz CD sounds bright, consider what makes it bright and you will see that it's a false bright created by the high frequencies sounding like square waves!!

Hmm. In real life, it may not be practical. Recording in 24 bit will increase the storage by about 50% as compared to 16 bit. Furthermore, processing in 24 bit will tax the CPU much more than 16 bit. All in all, cost will go up. And really, how much audible difference can one tell between 24 and 16 bit? A little brighter maybe, but most ears can't tell the difference. Somebody at another forum did this experiment by posting recordings in 16 and 24 bits - hardly anybody can tell the difference. Wish I could locate that link. With music nowadays (blaring from loud speakers), the slight difference can go pass totally unnoticed.
 
It's all relative. Diskspace today is roughly $1.2 to a gigabyte. So the extra fidelity is worth it when recording and during the lifetime of the project before mxing down to 16/44.1. Simple recordings are probably not that obviously different, but once the audio is manipulated, there's going to be consequences. From a graphics perspective, it's similar to applying visual effects to constantly resaved lossy jpegs and expecting the quality to be the same as lossless tiffs.

Music nowadays is mastered very loudly. Everything just becomes a blur of sharp loud constant noise. Not that i haven't heard extremely sharp constant noise as music, but i sometimes can't stand how it's done. Rock albums in particular spring to mind. Like Mew's frengers. You'd think Icelandic rock would be full of fire and ice, but there's a lack of dynamics, which would've suited their style of playing, mastered out into a constant loudness.
 
i believe the main difference is when u apply lotsa effects to a file recorded in 16bits vs another in 24bits, loss in fidelity would be less apparant with the 24bit....otherwise if we compare a raw 16 vs 24 unprocessed..the difference is minimal ..which also depends on our equipment and room we listening to

like they always say, " ...its only as good as ur weakest link"
:wink:
 
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