sample rates

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dobro

dobro

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I understand the importance of bitdepth in audio processing, but what about sample rate?

I recently wrote Gadgetlabs, asking them if my 24 bit, 48 kHz soundcard is upgradable to 96 kHz. It isn't. Here's what they said:

"Since true 96kHz
support requires different converter chips, the Wave/824 can't be upgraded
96kHz. When we settled on the design, we decided that including 96kHz would
have increased the cost too much. Also, there are very few PCs available
that have the performance necessary for 8 in x 8 out @ 96kHz, and the hard
disk space doubles even though the audible improvement is marginal."

Okay, you can't upgrade a soundcard's sample rate. But I'm taking that 'audible improvement is marginal' with a grain of salt, because Gadgetlabs themselves released the Wave 4/24 recently, which is 24-bit, 96 kHz. Are they doing this because 96 kHz sounds better than 48 kHz, or because people *think* 96 kHz sounds better?

It seems logical to me that a higher sample rate means higher resolution recording, a smoother sound, but can anybody here actually *hear* the difference between 48 and 96?
 
I don't think this Gadget Labs thing is any more sinister than them holding out to get a good price on 96KHz converters they could design (read: budget) into the 4/24.
Probably scouring the planet playing chipmakers against each other for the biz.
 
I'm convinced there is a benefit to 96k recording. I have no basis for my belief other than general theory, though. Those with better ears than I maintain that 96k makes a difference, but a marginal difference. I have interpreted that to mean that I can probably improve things a lot more with better mics, better monitors, some flavorful preamps, some more good plug-ins, etc., given my current rig.
 
I've always been under the impression that if you're going to put your music onto a CD it's best to just record at 44.1khz because that's the speed at which Cd's play. I'm not 100% certain, but I think even if you record at 96K or 100Ghz (not really possible...), if you're going to resample down to 44.1khz for a CD, it's only going to sound as good as 44.1, and not what you recorded at. Of course if you're only goal is to bring people to your computer for listening, then 96 (if there is any noticle difference) will be better. What we release to the public is only as good as the meduim it's on. Can someone argue with me here please? I'd LOVE to know if I'm wrong.
 
of course recording in 96khz will be better quality, however there is no real audible difference between 96khz and 88.2khz, so i think you should do all your recordings in 24/88.2. Brian Grey......why should we record in a higher sample rate when we will later convert down for cd? because you want to record in the highest sample rate and bit rate possible for a better quality recording which you can then convert to "cd quality standard". To sum it up we don't need 96khz yet, it wastes extra storage space, there is no real audible difference between 96 and 88.2. we will only need tpo record in 24/96 when DVD becomes the standard.
 
wheres ed on this one? do a search for all threads with "96 khz" in it and use "sonusman" as the member name.. at LEAST 10 different threads will pop up arguing 96 khz vs 44.1 khz and the "sampling down" to cd issue...

- eddie -
 
I read somewhere that Rupert Neve said we needed to record at 24bit/192khz to equal analogue - any comments on that remark??
 
He might be right. I imagine Rupert Neve has a fantastic mic collection. We know he's got some okay preamps and a few okay consoles :D. I bet his monitors don't suck. And his ears are probably not too bad.

As much as I believe there are improvements at higher sampling rates, I just can't stress this enough: 24/44.1 is probably NOT the quality bottleneck in my system.
 
Okay, everybody thinks the higher sample rate is better, but nobody's saying it makes a big, obvious difference. So my first conclusion is that 96 kHz is not as big a deal as other things in getting a good sound (mic, mic pre, mic placement, compression and EQ and reverb, just for a start).

After reading virtualray's comment and Ed's thumbs up on that comment, my question is now about converters. How do you judge converters? Shall I start a new thread? :)
 
How to judge converters? You probably know what my answer is going to be :D.
 
Well, there are two good ways to judge: the best is listen to 'em and decide for yourself; a less good but often useful way is to ask other people if they have any experience with different converters. :D :D
 
<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:</font><HR>Originally posted by dobro:
Okay, everybody thinks the higher sample rate is better, but nobody's saying it makes a big, obvious difference. So my first conclusion is that 96 kHz is not as big a deal as other things in getting a good sound (mic, mic pre, mic placement, compression and EQ and reverb, just for a start).

After reading virtualray's comment and Ed's thumbs up on that comment, my question is now about converters. How do you judge converters? Shall I start a new thread? :)
<HR></BLOCKQUOTE>

So am I to assume now that it PROBABLY isn't too big of a deal that I'm recording at 44.1 instead of 96?
If so, the topic of converters... I'd have to go for Sound Forge. Of course it's all I've ever really used. I'd like to know what Pglewis would suggest, I havn't been around to know his style. And also, I'd like to know if I should upgrage my sound card on the basis that it can't record at 96khz, or should I just stick with 44.1
 
Think of it like this...

DSP works with samples. It deals with samples only. It doesn't really care much what the sampling rate is because it is just looking at the bits that are available.

In 24 bit, you have 24 bits of info per sample.

So, when you go to apply DSP, which is based upon algorythms that work with induvitual bits of information, the algorythm is set, it is not going to do any more then what was coded. Right? Right.

So, since you have X amount of bits per sample, if you have more samples, you have more bits of infomation for the algorythm to play around with. What this usually means is that the processing will be smoother.

When you down sample, or resample, another algorythm is used. With higher sampling rates, of course this algorythm is going to work better because it is some sort of DSP. Not only will a higher sampling rate mean better DSP "percieved accuracy" when it is applied to the higher sampling rate file, but DSP "percieved accuracy" when you use the DSP to down sample. So, you start with a good file, then apply DSP for whatever, and it is good DSP because more processes took place, then the process of "dummying it down" is also better for two reasons: 1- because there is a better audio source for it to work with 2- more processes are going to take place during the process of "dummying it down".

Now you know.

Rubert Neve is an electrical engineer, not a recording engineer. I am sure he has picked a few things up over the years in the studio, but I have never seen his name on a CD credit.

About the 192KHz sampling rate deal, this is not really true or false. It has merits in some circles, and none is others. Hell, there are people who still believe that 20Hz-20KHz is adequate for "real sounding" audio.... :) Since digital can effectively record half it's sampling rate with a great degree of accuracy, then it would stand to figure that being able to record sound that is around 98KHz would provide significant sonic accuracy. In theory, this is a good idea. Many brass instruments produce tones that push 100KHz. But, one small little problem with this....

It would not be that hard to create 24/192 converters with a full dynamic range and signal to noise ratio available. But, a sound has to be delivered to it. That is the problem. EVERY microphone has a self noise. Even Rupert Neve's preamps generate noise, and only have around (at best) 100db of sound to noise ratio. So, while sound that we hear live if effected by frequencies that are well beyond what we can hear, you also need to be able to catch that sound on tape to have "true" accuracy. Not a single microphone I know of, even test microphone that are used for research, are capable of "accurately" catching tones that are pushing 90KHz. The mic itself will change those tones through it's mechanical conversion. Throw in the preamps limitations, and the wires that are used to connect this stuff into to mix, and well, you just can't record stuff that high with true accuracy. Period. Presently, it cannot be done, and if it can (who knows) it cannot be done affordably (even to the big time studios). So, 192KHz sampling rate is bogus presently with even the best mics and preamps and wire available.

Analog recorders only can record up to about around 40KHz, and above I believe around 30KHz, it starts to really suffer from some weird stuff that will effect the audio.

I don't think that 192KHz sampling rate is what is needed to produce "analog sounding" results in the digital realm. I think that digital will always have a more "cleaner" sound to it, and that converter quality is a big issue. 96KHz sampling rate is relatively new, and I believe that as the manufactures look to "reinvent" the stuff, that 96KHz will prove to be quite adequate to produce "analog type" quality. But, analog, even at it's best, is not very sonically accurate, so even if higher sampling rates become the rage, it still will not sound like analog, because it is not analog, and it will never be analog. I think the only way you will get digital to really "sound" like analog is through emulation. Of course, now we are back to having higher sampling rates available so that the DSP can replicate the "modeled" tape deck much closer.

You wanna know what will be next is digital? Higher internal bit processing. Most stuff is at 24 bit (the stuff most of us use) and most high end pro gear starts at 48 bit, with a few high quality digital boxes providing up to 72 bit.

Once more actual processes can be applied to the audio that is in the digital form, the better "percieved" accuracy DSP will have. Until internal bit resolution is up'ed significantly, digital is going to suffer from it's present day lack of "warmth" and "subtlety" that high end analog enjoys.

Hell, I could go on and on, but the point is that more bits are available for DSP at higher sampling rates. Plus, aliasing and what not are significantly minimized with 96KHz. While you ears may not be able to tell much difference between 96KHz and 48KHz, you DSP will!!! :)

Ed
 
AS Ed said, with a higher sampling rate,
it's not so much the Freq' range ( which is also a factor) but the fact that with trying to avoid aliasing we dont add as much distortion to the audio.

I have read about Another reason higher rates should be used but have not totaly understood the issue. It's about localization. If I understand it correctly - there is a delay between when we hear the sound in each ear when something is in the center of a stereo image. About 6 microsecs.

As we increase a delay to control the position of a transient in the stereo image, the higher the sample rate the more improved the image.
People claim that in 44.1, the image may be smeared.

I would be glad to hear more about this.

[This message has been edited by Shailat (edited 07-12-2000).]
 
Dobro: Toche :D. When do I get to hear some of your stuff, though?

Brain: You can get pretty good recordings with 44.1. I've honestly been recording everything at 16/44.1 so far... I don't want anyone thinking I'm talking from personal experience when I chime-in about 24/96. I think someone with Sonus' experience can milk a lot more out of higher sampling rates than someone like me who's still learning the ropes a lot. I think I can get a lot better mixes out of my current equipment with more practice and experience. YMMV, of course. On "converters", we're talking about the analog to digital (A/D) converters that take your "real-world" sound and convert 'em to bits. Converters are NOT all created equal by a long-shot. It's usually another one of those "you get what you pay for" deals.



[This message has been edited by pglewis (edited 07-12-2000).]
 
The Nyquist theorem says that you need to have a sample rate at twice that of the highest frequency you want to record. So 44.1kHz will allow us to record a highest frequency of 22.05kHz
At 96kHz, we can reproduce HF up to 48kHz.
So the question is, how good are we at hearing sound between 22.1 and 48kHz ?
Information above 22kHz will certainly add a "sparkle" to the music, but we can't hear much above 28kHz - and even this decreases with age.
 
pglewis - coupla months, maybe. But it's just voice and guitar, so no big deal recording-wise.
 
The question isn't so much about how well we hear sound between 20.05 and 48kHz but rather about how it affects the sound in what is usually referred to as the audible range (20-20kHz). With a test system, which consisted of s friends $8000 HiFi system and a tone generator, I could hear a sinus wave from around 10Hz to 17kHz (and I'm only 25 years old). However, when a supersonic sinus wave at 30kHz was added to an "audible" sinus wave at 7kHz, the difference was definetly audible.

I'm convinced that the supersonic frequencies in music affects the audible sound enough to affect the overall sound in an audible way.

However, I think that the largest benefit of 24/96 for most of us is better signal processing, as already pointed out, as most people's playback systems aren't really up to the task of doing the supersonic frequencies justice anyway. Maybe you could reap the full benefit of 24/96 with a really good DVD player and *really* good amps and speakers. That is, of course, if you have *really* good mics, preamps and A/D converters to record with.

/Ola
 
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