yorolpal


Wail, I, myself, bein of semi-sound mind and knowin full well that I ain't got a shovel big enough to git rid of what's bein piled up here would like ta try three or four faingers of whatever it is my ol pal Freddie is drainkin.  That there's some powerful juice.

Okay..I am going to throw my .02 cents in this war...but for me the extra headroom and bit depth was necessary to preserve the quality of the audio when CONVERTING BACK DOWN TO 44.1KHZ.....esp on the bits since all conversions do have to take away some of the bits and if you freaking stay in 44.1khz and then convert to say mp3 again, conversion, bit lost(why mp3's sound like junk) you loose that audio quality, so for me is that extra headroom....

bitflipper

Okay..I am going to throw my .02 cents in this war...but for me the extra headroom and bit depth was necessary to preserve the quality of the audio when CONVERTING BACK DOWN TO 44.1KHZ.....esp on the bits since all conversions do have to take away some of the bits and if you freaking stay in 44.1khz and then convert to say mp3 again, conversion, bit lost(why mp3's sound like junk) you loose that audio quality, so for me is that extra headroom....

Wow...where to begin...so much fuzzy logic swirling around here...

OK, let's start with a few basic points - and there will be a quiz later, so repeat after me: "headroom" and "bit depth" have nothing to do with sample rates. And you don't "lose bits" (or dynamic range) when you downsample, you only lower your upper frequency limit. 48KHz is (somewhat) justified if your final target is a native 48KHz medium such as DVD, for the simple reason that you can avoid a sample rate conversion.

And let's not be too hard on Freddie. One of two things is going on here: either a) 96KHz really does sound better with his interface, or b) he only thinks it does. The former is a real possibility, but if true it more likely reflects a limitation in his equipment, rather than some general truism about sample rates. The latter is also a real possibility, but only those who have never tweaked the wrong fader and "heard" a difference are allowed to cast stones.

I say if it makes you feel better to record at 48, 88.2, 96 or even 192, or if it makes it easier to swap files with others for collaboration, then knock yourself out! Just refrain from stepping outside the reality of math and physics when singing the sample rate hallelujah, because noobs are listening.

Jeff Evans


Danny I still think it has something to do with the fact that the Layla cards might still sound better at 96Khz and maybe dont perform quite as well at 48 Khz. Sorry to say this but I would not put the Layla stuff right up in the high end of sound cards or audio interfaces. For 48 Khz to sound excellent it requires a great audio interface. I bet an Apogee system at 48 Khz would match your Layla or even better it at 96Khz.

Switching sample rates and comparing them on the same interface is useless. 96 K is going to win almost every time! The only way to find the real answer to this is to make the 48 Khz converter the best on the planet and then compare everything else to that at any sample rate.

Wiz


unless you double blind test yourself....you are releaving yourself into the wind.

info@tomflair.com


well..so all you 96 guys surely have rooms treated for 40k $ professional studiomonitors (price range 7k upwards) professional recording chains and high quality da convertesr to HEAR all this praised "leap in quality" ?

i have been in audiobusiness for mor than 15 years now - and i worked on low, middle and high end equippment - and i can also assure you that i am a hardware, and specs freak - actually if i could afford it i would buy the most expensive stuff available (and yes there are classes over apogee and similar stuff ;-)  but just because i am a freak not because there is a real reason to to have "the best" soundg better

yet - please stick to technical facts: all the "better" whatever you hear is just in your head - actually if a soundcard sounds different between 88 and 96 than its just badly built ...maybe you even like the worse sound with more jitter better - who knows...maybe also digital distortion can be pleasant to the children of mp3 files ;-)
its just a mettar of FACT that humans cant hear anything over 20 (and  pushing it to the extremest possible some specialits claim that there is "sound" percieved up to  a max of 26k but that sound is not percieved acustically but via the skin or just via stereo allocation depth...yet this is already pretty specuzlative)

on the other hand folks do as you are pleasd: if you have disk space to waste (i have but i dont waste it in spite of) if you reduce the benefits of better and faster computers by doubling the load...well - its good for the industry - but SURELY not for your tracks...

i wished people would at least react to facts here - the cited double blind randomized tests have all be done - and some exponents of "higher is better" just didnt look good in those tests - cos NO ONE statisically EVER guessed the "better sources"...this is just the truth - and here we are talking about tests with ultimate highend gear in a sort of perfect test environment ...
so pleas stop spreding panic - there are so many professionals still working at 44...

btw.. i agree with most people that 48 is better than 44 - and i also experimented with 88 (cos why 96 if a cd will be the target) and i also sometimes THOUGHT that it sounded better - but its just the psychology of a wish :-)
12 xears ago i dindt care about any specs and rules and in logic i alwayxs applied dither in ec´very renderinbg step (BIG SIN ;-)  - well those old tracks sound as fat as they could - so even 100 steps of dither during production did not any harm --- hey its a bout "MUSIC" not numbers !!!!

cheers

Jeff Evans


Freddie what sound card are you using. I keep quoting Bob Katz's book on mastering and I will again. They did tests and compared 48Khz to 96Khz and above. But they did build the very best converters to do it. They also got some of the best ears in the business to listen to the AB tests. Conclusion, no one could really tell the difference. Conclusion, you dont really need sample rates above 48 Khz. End of story. But the converters have to be good and that is where the problem may lie.

What you may be hearing, is you sound card may be performing better at the higher sampling rate but it is not so good at lower rates. Also you may also be hearing the better detail in the 24 bit depth.

If you are switching yourself you may be experiencing what you think is better. Get someone else to do the A/B test and also make sure the levels between both rates are exactly the same. And by the way when you do an A/B test like this you should use a real analog source eg a high quality turntable or reel to reel machine as the sound source. Do this test with a turntable (and someone else switching and you need to be blindfolded as well or not look!) instead and let us know the result.

You are wasting disc space and making your system work harder for no reason.

A1MixMan


Hey Jeff,
 
I just finished reading Bob's book. What a great book highly recommended for all here. Anyway, here is a quote from page 301 under the heading What Sample Rate?
 
"Until around 2000, I recommended that mix engineers try to work at 44.1 kHz if possible for CD, considering the then poor state of typical sample rate converters. This is no longer necessary nor desirable; high quality sample rate converters can convert between 96 kHz and 44.1 kHz with high integrity, as described in Chapter 20. The best recommendations are for the mix engineers to work at the highest practical sample rate and the longest available wordlength."
 
If it's good enough for mix engineers, it's good enough for me. 96/24 it is.
And with today's fast computers and Terabyte hard drives, the argument for too much disk space and your computer working too hard doesn't hold water anymore. Not for me anyway. You wouldn't buy an HDTV to watch analog signals would you? 

rhythminmind


Ohhh freddie freddie I'm just going to copy/paste at this point.  



"Recording @ 48k is currently the best compromise of clean A/D D/A audible bandwidth (20hz-24khz) & DSP functionality. "


" it's not an optimum rate for that EMU interface, nevermind unnecessary. "


Dan Larvy of , arguably the best current A/D D/A designer posted a great reply.

"
Regarding of the impact of sample rate on accuracy:

We all know that sampling way too slow (say at 1KHZ) is not a good idea. Why? It will only allow frequencies of 500Hz or lower (at best). Sampling that low for audio is simply ridicules.

But we also know that sampling audio at say 1GHz is ridicules. Why? Because at 1GHz we can only get very few bits of accuracy, so the noise and distortions will be really bad. The data size will be huge, and for no good reason at all (we are talking about audio, not 1GHz oscilloscope or telecome gear...).

Therefore, there is a concept we need to adopt – the concept of an OPTIMAL sample rate. Clearly it is above 500Hz, and bellow 1GH. So what is the optimum rate?

The optimum rate for conversion is driven by the application. For weighing scales, the process is slow (a second or more) but we need a lot of accuracy. For video, we need some few MHz bandwidth, but lucky for us, the eye is a lot less accurate then the ear, so having fewer bits (then audio) is OK. For medical applications, it depends on the specific cases (CAT scan, MRI and so on, all have their requirements)…

Generally, if you look at converters, you find out that the faster rates yield less accuracy, the slower rates yield higher accuracy. Why is it so? Well, this is not the only case where doing things faster is a tradeoff for accuracy, and taking your time often enables more precision.

Say you want to charge a cap. If you “take your time”, the final charge will be closer to the intended charging voltage. If you do not wait long enough, it will not charge fully. So you try and reduce the cap size, and now you have other problems show up (beyong the scope of this post)…

Say you want to have an OPamp track an input signal accurately. A circuit designer knows that the best accuracy is at low frequencies, down to DC, and at some point, going to higher frequencies, we lose accuracy. The devices (transistors) inside the amplifier “loose steam” at higher frequencies. Yes you can find real fast transistors, but then you trade off accuracy in different ways.

The charging cap and OPamp are just the first 2 “element” in the AD circuit. There are a lot of caps charging and a lot of OPamps inside an AD… It would take too long to get into the details why speed accuracy is always a tradeoff. Such tradeoffs do exist in electronics and other areas. In electronics one can trade off speed vs. power, size vs. temerature (heat) and so on. It is true that technology is moving forwards, so the tradeoffs today are different then those of say 10 years ago. Please take my comments about tradeoffs as correct for a given time in the history of technology. Lets not compare the speed of today’s gear with that of 40 years ago… Lets talk about the tradeoffs that exist today, or 10 years ago, or at any specific point in time.

For a while, I was interested in writing a whole paper about the technical reasons for speed accuracy tradeoff. But then, I needed to buy a new Audio Precision test system for the production final testing. Now, these guys there at Audio Precision are makers of the finest audio measurement gear, and we are not talking about inexpensive gear. They have a converter based system called ATS2, and you can buy it to accommodate unto a little over 100KHz (for 44-96KHz sampling) at very nice accuracy, or you can buy it with an option to extend to around 200KHz (for 192KHz gear). The point is: if you want to do 192KHz the measurement system is limited to 16 bits! When the best test gear maker has to cut down on precision significantly, for going from 96 to 192KHz, you know that there is speed accuracy tradeoff. Again, the gear I am talking about far from inexpensive. At that level, they do not trade off converter quality for lower price. So when I saw that “proof” that speed costs accuracy, I sort of lost interest in writing a more detailed description as to why speed and accuracy pull in different directions. I can now say
"I rest my case"...

We are talking about audio, so we need to cover the hearing range. What is it? It is a little higher then 20KHz (for some people), so the theory suggests that we can sample at a little higher then 40KHz (twice the bandwidth). But theory assumes that we can “do theoretical things”, in this case, we would need brick wall decimation filters, going from passing audio fully intact at say 20Khz, to blocking it completely at say 20.000001KHz. We can not do such things. We end up needing some margins to “bridge the gap” between theory and good practice.

In my view, that margin should be up to the design engineers. The ear should tells us what we can hear and where the limits are, and the designer gets to find how much margin is sufficient to accomodate the ear. Too often, mastering and recording people, or even gear salesmen, step over thier boundries into the design area, which is a sad fact responsible to the false notion that 192KHz sampling will be better, which is false.

Nowdays there are a number of good designers and ear people that find 60-70KHz sample rate to be the optimal rate for the ear. It is fast enough to include what we can hear, yet slow enough to do it pretty accurately. Faster rate means less accuracy, with some unwanted side effects – increased data size, need for more powerful DSP compute engine, and there is no up side to going faster. Going slower gets the designer “squeezed” at the 20KHz range, which we need to include for high quality audio.

"

Interesting! That means I'm 100 % right then and proof that I'm right.

bitflipper

Okay..I am going to throw my .02 cents in this war...but for me the extra headroom and bit depth was necessary to preserve the quality of the audio when CONVERTING BACK DOWN TO 44.1KHZ.....esp on the bits since all conversions do have to take away some of the bits and if you freaking stay in 44.1khz and then convert to say mp3 again, conversion, bit lost(why mp3's sound like junk) you loose that audio quality, so for me is that extra headroom....

Wow...where to begin...so much fuzzy logic swirling around here...

OK, let's start with a few basic points - and there will be a quiz later, so repeat after me: "headroom" and "bit depth" have nothing to do with sample rates. And you don't "lose bits" (or dynamic range) when you downsample, you only lower your upper frequency limit. 48KHz is (somewhat) justified if your final target is a native 48KHz medium such as DVD, for the simple reason that you can avoid a sample rate conversion.

And let's not be too hard on Freddie. One of two things is going on here: either a) 96KHz really does sound better with his interface, or b) he only thinks it does. The former is a real possibility, but if true it more likely reflects a limitation in his equipment, rather than some general truism about sample rates. The latter is also a real possibility, but only those who have never tweaked the wrong fader and "heard" a difference are allowed to cast stones.

I say if it makes you feel better to record at 48, 88.2, 96 or even 192, or if it makes it easier to swap files with others for collaboration, then knock yourself out! Just refrain from stepping outside the reality of math and physics when singing the sample rate hallelujah, because noobs are listening.