The Optimal Sample Rate for Quality Audio

[goldenpiggy]

I remember when experts proclaimed that 3.1 megapixels were all you'd ever need from a digital camera. Then it was 6 megapixels. Then 12 was the magic number.

 

Now we have the 36-megapixel Nikon D800 taking the industry by storm. Does anyone need 36 megapixels? Probably only those that print billboards and giant posters. But it's available, (relatively) affordable, and actually works. Why not?

 

As a part time sound engineer, I never considered recording above 24bit/96KHz. I figured everything will be burnt to 44.1KHz redbook CD. These days, we have to keep in mind hi-res distribution which the audiophile market demands. If I was going to distribute 96Khz, I would probably record at 192KHz so sample-accurate editing is not compromised. If I was archiving, then probably go double-rate DSD.

 

Today's computers and audio interfaces easily handle recording and playback of multiple 192KHz channels simulateously. Storage is cheap (getting back to normal after Thailand flood last year).

 

I guess same argument could be made with 32bit DACs...isn't 24 bits plenty? We're talking about S/N ratio that's beyond many measuring equipment, let alone our ears.

 

I would agree that technically 96Khz is plenty. But if the technology is here, affordable, and allows higher res, then go for it. We have bandwidth to burn.

 

Cheers,

JR

[goldenpiggy]

Nope, not because we can. Because there is a market for hi res audio. Not advocating 192Khz, no professional advice or experience offered...JMHO.

 

Ironically, the 36Mp D800 outperforms just about every other 3,6,10,12,16,21,24Mp cameras in low light/high ISO capabiity (except for the low light King Nikon D3s). Such is technology.

[goldenpiggy]

Protocol-wise, sample rates above 192Khz is not a problem with AES/EBU -- it can theoretically support any sample rate as long as clock can be recovered from the data. Bandwidth is usually not an issue as it is common to run AES/EBU over good old 75 Ohm RG6U coax or cat 5e (350MHz) cables. The current spec allows for maximum of 24 bit audio data though.

 

You do have some issues: the more common 110 Ohm characteristic impedance allows a pretty loose +/-20% tolerance. The common XLR connector is not impedance matched. At faster rates, impedance matching between transmitter and receiver is a must. To keep the eye opening bigger and jitter low, faster rise time is needed, so one must also consider EMC/radiated emissions at higher rates given the biphase signaling (not very efficient and quite a noise polluter.) Commercial audio equipment has to still pass FCC class A certification. AES/EBU is almost always transformer-coupled, so at higher rates it will require better and more expensive isolation.

 

In the pro audio world where AES/EBU is prevalent, 96KHz/24bit is common these days, although you'll often see 192KHz in the studio environment. I don't think I've ever seen anything higher though. Probably not a whole lot of equipment that handles LPCM at greater than 192KHz sample rate anyway. AES/EBU was designed to handle long distance transmission in less than ideal environment, rather than sheer transmission speed. With an active equalizer (distribution amp), you can easily go 2-3 football fields over cat 5.

[goldenpiggy]

Hi Jud,

 

Why use SRC? Well for me, since I record at 192k, I need it to create 96k and CD versions.

I use it in mastering too. Even when a mix comes in at 44.1k, one of the first things I'll do is create copies at a higher sample rate. The reason for this is that when applying EQ or other processing, I find the results sound better at higher rates. Further, if done at higher rates and the results later converted to 44.1 with a high quality algorithm (such as iZotope's), *some* of the benefits of the higher rates are preserved. In other words, I've found it creates a better sounding 44.1 version than if the SRC was eliminated and all mastering done at 44.1.

 

I believe SRC means reclocking data, and if you have a very stable master clock, it is in effect a de-jitter. Also, when one goes from 16 bit to 24 bit, this tends to lessen or redistribute quantization error. These should have audible effects.

 

Those of us who were into sampling from the stone age (remember the 8-bit Ensoniq Mirage or original Emu Emax?) can attest to what quantization error does to sound quality. Going from 8 to 12 then to 16 bit sampling tremendously improved sound quality. So you can imagine from 16 to 24 bits.

 

Regards,

JR