Hi,
I have just bought a Presonus FP10 as my first home studio recording equipment (up until now I have just been using a laptop sound card).

Should I record at 96khz now that I have the capability? What benifit does this bring to me?

My recordings sound ok at 44khz so im assuming the actual quality of the sound wont change?

Thanks,
AM.

Debatable. If you can, do so, but you'll find that it chews up resources far faster than when working with 44.1. Bitrate (24bit vs 16bit) would be my preference in a pinch, as it gives more flexibility as well as (Debatably) better sound quality. Actually there's loads of articles that'll be far more useful than people's opinions. This: http://www.soundonsound.com/sos/sep07/articles/digitalmyths.htm is a good starting point, carry on from there

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Stuff what I done

Stick to 44.1kHz - you'll be wasting your time & hard disc space running 97kHz

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Mac Pro, Logic Pro, lots of software and 17 hard drives!

I dont know about that

I can hear a huge difference between 96 and 97khz

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Matt
www.krcollective.org

i have worked with "big name producers" that record at 44.1

though this is all pretty much rock guitar bandy stuff.

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Live Sound and Studio Engineer

44, unless you have a really specific reason for 96, and your room acoustics and recording chain are top rate, and you have the available resources to "waste" (double the CPU load, double the disk space, etc).

what about 48khz?

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www.myspace.com/shambolic-charm

You always want to maintain the highest sample rate as long as possible. However just because you have the capability to record at 96 does not mean that you should. There are a lot of factors to consider and I won't go into detail as I am no expert but space is one of them, power is another, internal components of your AD is another. Even if you have the capability that doesn't mean that your gear operates best at 96. It really is quite confusing and there are a million previous threads debating such questions, one or two of them mine.

Bottom line after all the reading and research I feel I know less than when I started, or else I know more which only makes it more confusing thus it feels like I know less

I can hear a difference between 96 and 48, however what that difference is, is debatable and once I listen to 48 for a while it sounds just fine.

At the end of all of that I would say unless you are recording Classical music I would stick to 48/24, or 44.1/24 and forget about trying to understand any of it.

48k is for film and broadcast - I usually export "up" to 48k from 44.1 when handing in broadcast stuff.

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Stuff what I done

Some say certain plugins sound better at higher sample rates. And I have seen convincing claims that lower end converters perform better at higher sample rates as well, this seems counterintuitive I know. But space and computing power will most likely be the deciding factor for most home studios. Dedicated DSP cards like those in the UAD line and HD Pro Tools systems up sample to as high as 192khz for their internal processing I believe, which negates the need to record at higher sample rates.

I could be wrong, this is just what I have gathered from my untrained research!

Quote Jocoserious:

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You always want to maintain the highest sample rate as long as possible.

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Hmm... bit of a blanket statement that.

Why?

Quote passerrby3141:

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Some say certain plugins sound better at higher sample rates.

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Yes. There are technical reasons for this, but it can be true that, especially for aliasing reasons, some virtual instruments benefit from going up that high. But this is a completely different issue to audio recording.

Quote passerrby3141:

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Dedicated DSP cards like those in the UAD line and HD Pro Tools systems up sample to as high as 192khz for their internal processing I believe, which negates the need to record at higher sample rates.

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It's nothing to do with DSP cards per se, it's the DSP algorithms used. Some algorithms and processes benefit from upsampling, and some native plugins do this as well where it makes sense. However, upsampling does tend to make the plugin much heavier on the CPU, so plugin designers try to avoid high-cost algorithms unless it makes sense, or is necessary to use them.

Basically.....

Human hearing goes up to about 20 kHz.... but you need to double that to eliminate something called aliasing, which is something like a "rounding error" that causes mucho distortion and strange frequencies.

However, I regularly argue that 80 kHz or higher is necessary to get a good representation of the frequencies around 20 kHz.

If you imagine a sine wave at 20 kHz... and then you sample that twice per period (which is a 40 kHz sampling rate), you don't happen to get a great representation of the wave... For example, if you're unlucky and you happen to sample the sine wave at the two zeros each period, you'll get nothing.

in fact, you're guaranteed to get a "lower volume" reproduction of the wave. Thus, I suggest at least 4 samples per period of the highest frequency you're interested in reproducing accurately.

192 kHz means you're getting 8 samples per period of a 24 kHz wave... that ought to reproduce those high frequencies much better.

Anyhow - that'd be the difference! Much music is "low end heavy" anyway.

Like others have said, I'd stick with 44.1 unless you're doing something very delicate in a very nice room. I used to do everything at 96. It was a pain to mix large sessions because my computer really couldn't handle all of that data. Now my classical recordings are done at 96/24 while pretty much everything else is at 44.1/24. A friend of mine uses 48 to get away from a few of the side-effects of 44.1. He even uses 48 for classical work and there's no problem in the sound. Unless you can do 192k or... SACD rates (2822k etc...) I'd stick with lower rates. 192 and sacd are definitly far superior, but is it worth the strain on your computer?

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Winding Road Studios
Speaking Audio Blog

Ok, i made a rough approximation... because I'm hitting the same problems at 48 versus 96...

Basically, I roughly computed that if you wanted 48 to sound more like 96, if you put an EQ that raised 20 kHz by 1 dB, 10kHz by 0 db, and made a smooth line between the two (eh, I have a hand drawn eq)... it'd be pretty close.

So, then I tested it with some songs in my collection... basically, if you've already boosted the high end on cymbals and similar, it doesn't really matter too much.. just puts a small punch on the cymbals.

Whelp, so. I find this stuff fun to think about.

Well I think 96k sounds better...the difference gets smaller as converters get better but I think it's clearly a sonic advantage IF you're actually recording stuff and not just using loads of samples.

However it is really inconvenient because it chews resources, makes backups take ages, and introduces a boring extra conversion at the end of the process.

My advice would be: Anything that's really important, try it at 96 because it's certainly a bonus. Do your demos etc at 44.1 because it's convenient and sounds very good.

J

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www.jackruston.com

This has been covered (extensively) before, but just to add my 2 penneth...

recording at 96k doubles your PC overhead - for an improvement that is hardly perceivable in real-life use. (don't quote your maths at me)

It ain't worth it.

Just remember to record 24 bit, 'cause that makes one hell of a difference.

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www.thediplomatz.com

Check out Paul Lehrmans artice in Mix Magazine (available on the web) - The Emperors New Sampling Rate.

The title says it all really. Even Ethan Winer, of Real Traps who I respect greatly when it comes to sound (his ears are much more golden than mine) suggests that differences heard (and I'm not saying that people who hear a difference are lying) can be due to minute changes in head position between changing the material, affecting the phase of the sound. Mr Lehrmans article refers to double blind tests where there was no statistical difference and apparently 'educated' ears were used. Of course, he has been flamed within an inch of his life in some places but many people, in particular manufacturers and those who have been saying for years that they can hear a difference have a vested interest in 88.2 and up.

Now, I'm not one to go against the word of many much more experienced and sucessful engineers on this but in my (admittedly very limited and unscientifc) tests I can't hear a difference in my little studio. And as I can run a few more plug ins (not as many as you would think, I run Guitar Rig in HQ mode which does internal upsampling and use UAD plugs which also upsample where required) and many more tracks I keep using 44.1 or 48.

Here's the link. Interesting article, thanks for posting that:

http://mixonline.com/recording/mixing/audio_emperors_new_sampling/

Quote JohnnyT:

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Mr Lehrmans article refers to double blind tests where there was no statistical difference and apparently 'educated' ears were used.

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Double blind tests, in a big enough sample of people, are about the only thing that actually proves anything. All the rest can be easily explained by the placebo effect.

I believe any hesitation in shouting about the results is due to the size of the sample, if I recall. I didn't think to post the link - my, how 20th Century!

I correct that, I just re-read the article. The sample size was 'hundreds' of people. Perhaps not definitive but pretty persuasive, certainly enough to stop and think about it and hope for more studies. In the meantime, maybe stick to 44.1 :-)?

Once again, sorry. Hundreds of trials.

Quote desmond:

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Here's the link. Interesting article, thanks for posting that:

http://mixonline.com/recording/mixing/audio_emperors_new_sampling/

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Interesting article. I enjoyed reading that

Quote JohnnyT:

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Once again, sorry. Hundreds of trials.

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I can not find anything wrong with the study at all. I suppose the only thing you could do to improve it would be having some kind of frame to lock the listener head in the exact same position every time. Something like they had in clockwork orange maybe? lol

I know I can not wear any differences in 96KHz audio as opposed to 44.1KHz, but then again my hearing is not great after a few years playing gigs so Id on't really think this proves much.

It reminds of a test I saw in some audiophile forum once where they did a blind test between themselves in order to prove they could hear a difference between 16/44.1 Khz and 320 Kbps MP3. They only guessed correctly about 40% of the time, and conclude that deaf people could probably do as good a job as they did or better, since anyone has 50% probability of guessing right by picking a random answer.

Anyway, I blame all this on gearslutz.com. lol

Quote Interweaved:

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If you imagine a sine wave at 20 kHz... and then you sample that twice per period (which is a 40 kHz sampling rate), you don't happen to get a great representation of the wave... For example, if you're unlucky and you happen to sample the sine wave at the two zeros each period, you'll get nothing.

in fact, you're guaranteed to get a "lower volume" reproduction of the wave. Thus, I suggest at least 4 samples per period of the highest frequency you're interested in reproducing accurately.

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This is why you need a sampling rate greater than instead of equal to twice the highest frequency you are sampling.

It's only when the frequency of the sampled signal is exactly half the sampling frequency that this problem occurs, as there are an infinite number of sine waves of frequency fs/2 with different phases and amplitudes that could all give rise to the same sampled data.

If however the sampling rate is even sightly greater than twice the frequency of the sampled sine wave then there is only a single sine wave with a frequency less than fs/2 that will fit through all the sample points, and thats your original signal.

Mark has already tackled this, but it is a common fallacy and deserves the emphasis:

Quote Interweaved:

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Basically.....

Human hearing goes up to about 20 kHz.... but you need to double that to eliminate something called aliasing, which is something like a "rounding error" that causes mucho distortion and strange frequencies.

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The Shannon/Nyquist sampling theorem states that you need to sample at a rate which is more than twice the bandwidth of the required signal. Sampling is a modulation process, and the 'more than twice' element is to ensure that the source and it's modulated sidebands remain separate and separable.

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However, I regularly argue that 80 kHz or higher is necessary to get a good representation of the frequencies around 20 kHz.

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It is one way to do it, but it is not 'necessary.' Given a wanted bandwidth of 20kHz, how much higher than 40kHz the sample rate needs to be is determined by the efficacy of the anti-alias and reconstruction filtering.

To be honest, 44.1 was too low when the standard was set -- 60kHz would have been a far better choice for a host of reasons. Equally, 96kHz is arguably wastefully high. However, technology hasn't stood still for the last 25 years, and most decent converters can now provide very satisfactory performance sampling at 44.1kHz.

Twenty plus years ago, you needed the very best of converters to achieve adequate performance sampling at 44.1. Thankfully, that is (arguably) no longer the case, and there are some extremely competent mid-price converters out there now.

Interweaved Quote:

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If you imagine a sine wave at 20 kHz... and then you sample that twice per period (which is a 40 kHz sampling rate), you don't happen to get a great representation of the wave... For example, if you're unlucky and you happen to sample the sine wave at the two zeros each period, you'll get nothing.

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This is true, but hardly relevant since the theorem requires the sample rate to be more than twice the bandwidth of interest. In the case you cite, the lower modulation sideband of the source signal will lie precisely on top of it, and what you hear will be the summation of the two, which -- if they are perfectly aligned because the sample rate is exactly twice the source frequency -- could be nothing at all, as you say!

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Thus, I suggest at least 4 samples per period of the highest frequency you're interested in reproducing accurately.

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Yes, this will obviously work, but it is a bull in a china shop approach, which is wasteful of the information capacity provided by sampling at that high a rate.

The theorem is 100% accurate in its claims. The only issue is in the practicalities of implementing it -- specifically in designing the filters to do what they are required to do without damaging the wanted signal in the process.

There was a time when working at 96kHz (or higher) provided a clear sonic advantage to working at 44.1kHz, simply because the filter artefacts where removed above the human hearing range. However, as filter design and clever techniques like delta-sigma converters have evolved, the difference has become far less pronounced -- in some cases to the extent that the difference is barely audible at all.

Countering that, computer processing power and data storage capacity has increased (and continues to increase) enormously, and while working at 96kHz involved serious overheads for most systems a few years ago, that is less the case now.

Personally, I generally work at 24/96 unless specifically requested not to, because all of my equipment can handle that rate doing the kind of work I do without a significant performance reduction. But on the odd occasions that I am required to work at 44.1 or 48kHz, I don't hang my head in shame at the poor quality -- the difference is often undetectable.

But as a working principle, it makes sense to record source material at the highest possible resolution and quality, because the quality can only suffer after that.

Quote Interweaved:

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Basically, I roughly computed that if you wanted 48 to sound more like 96, if you put an EQ that raised 20 kHz by 1 dB, 10kHz by 0 db, and made a smooth line between the two (eh, I have a hand drawn eq)... it'd be pretty close.

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I think what you are suggesting here is compensation for the transition area of the 'brick wall filter(s)' since some (not all) exhibit a mild roll off as they approach the turnover frequency. But of course, amplitude is not the only aspect of a filter to consider -- phase is as important (if not more so) and by introduceing your HF lift, you will also introduce phase shifts that will change the character of the sound in some (indeterminate) way.

Hugh

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Technical Editor, Sound On Sound

Edited by Hugh Robjohns (05/07/08 01:37 PM)

Not much to add to the above, save to say that if you have (or aspire to) a Focusrite Liquid Mix, you can run 32 instances at 44/48k as opposed to 8 instances at 88/96k.

That's how much extra DSP power the higher sampling rate requires.

192k is definitely for the birds (or anyone else with genuinely ultrasonic hearing).

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Dynamite with a laser beam...


Edited by Steve Hill (05/07/08 01:36 PM)

Quote Interweaved:

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Basically.....

Human hearing goes up to about 20 kHz.... but you need to double that to eliminate something called aliasing, which is something like a "rounding error" that causes mucho distortion and strange frequencies.

However, I regularly argue that 80 kHz or higher is necessary to get a good representation of the frequencies around 20 kHz.

If you imagine a sine wave at 20 kHz... and then you sample that twice per period (which is a 40 kHz sampling rate), you don't happen to get a great representation of the wave... For example, if you're unlucky and you happen to sample the sine wave at the two zeros each period, you'll get nothing.

in fact, you're guaranteed to get a "lower volume" reproduction of the wave. Thus, I suggest at least 4 samples per period of the highest frequency you're interested in reproducing accurately.

192 kHz means you're getting 8 samples per period of a 24 kHz wave... that ought to reproduce those high frequencies much better.

Anyhow - that'd be the difference! Much music is "low end heavy" anyway.

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That's a nice thought provoking explanation.

I keep thinking of the maths and mechanics of how the highs are dealt with (and lows) because this is where the audio begins to suffer in standard home recordings.

I believe software and components hold the key to how the sound is dealt with.
I mean it's easy for a soundcard to state audio clarity,but add a dodgy plugin(which you don't know about)and it will mess your audio signal up.
It's a minefield but I love digital.

Quote Music Manic:

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but add a dodgy plugin(which you don't know about)and it will mess your audio signal up.

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This is just as relevant to hardware and analog processing too. At the end of the day, you are putting your signal into a "black box", which is closed and does some undefined process on that signal, the design process and choices of which you weren't part of, and then you get the output.

Without knowledge of what's happening inside the box, you have no idea what's going on, apart from what you can deduce from the output signal.

It doesn't matter whether the box is a software plugin designed by a 12-year old with a poor understanding of audio signal processing, a plugin written by some DSP maths genius, a hardware box an the entry level with cheap poorly designed electronics or a boutique piece of expensive analog gear.

At the end of the day, you have to understand and be familiar with your gear, educate yourself and make informed choices about how to process, why, and what to use, and this will give you better results than just inserting some random magical black box on a signal and hoping it's going to make your song sound better, somehow...

I found:
www.lavryengineering.com/documents/Sampling_Theory.pdf
Very interesting.

I have been doing 88.2 projects for a couple of years now, and I have to say that I think the quality of the sound I am getting is amazing. It doubles the system resources required however, and that can put a kink in things when I am working with large projects (Core2Duo + 3 UAD-1 cards).

It has been so long since I have done a 44.1 project that it is hard to remember the difference, HOWEVER, I have been recently taking a trip down memory lane in my car going through a bunch of old sessions that I have recorded in my studio and not listened to in a long time. Bands that have long since broken up and moved on, etc.

Just from listening to the CDs, I could not tell you which ones were recorded at 44.1 and which ones were recorded at 88.2, and trust me, I tried as hard as I could to see if I could tell the difference!

Maybe it is my old, overused ears...

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James
"Never interrupt your enemy when he is making a mistake" ~Napoleon Bonaparte~

Quote Hugh Robjohns:

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The Shannon/Nyquist sampling theorem states that you need to sample at a rate which is more than twice the bandwidth of the required signal. Sampling is a modulation process, and the 'more than twice' element is to ensure that the source and it's modulated sidebands remain separate and separable.

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Let me repeat my argument again... Assume the Shannon/Nyquist theorem only eliminate aliasing, so your sidebands are eliminated. But it does not guarantee anything about your sample being a great reproduction.

Now, suppose we're sampling at 40 kHz, and there's a 20 kHz sine wave, like thus:



Now... say your sampler samples at the blue points. Your final sample will just be zeros. Say i samples at the yellow points. Your final sample will be something like a 20 kHz sine wave, but reduced in volume for the original. So, on average, 20 kHz waves will be somewhat lower in volume when sampled in 40 kHz than your initial sample.

However, if you sampled at 80 or 160 kHz... you'd get a much closer representation of the original wave.

Quote Interweaved:

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Now, suppose we're sampling at 40 kHz, and there's a 20 kHz sine wave, like thus:

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But we are not, we are sampling at 44.1KHz, the minimum defined "CD quality" standard for the full range of the human hearing.

This is "more than" 2x the sample rate, rather than "exactly" 2x the sample rate.

Quote Interweaved:

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However, if you sampled at 80 or 160 kHz... you'd get a much closer representation of the original wave.

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Have you actually been *reading* this thread..?

Sampling at that frequency is redundant and not necessary. 44 is fine. Remember, sampling at higher sample rates does not give a "more accurate" reconstruction of the signal than 44. More dots does not equal more accuracy.

Okay now I may be way off base here and probably will not explain it well enough to be understood but here goes.

Even if someone cannot hear the difference between say 44.1 and 96 isn't there still some benefit to be had from using a higher sample rate??

There a million variables in all of this and this may not be of significance for anyone to actually hear but what about the degradation caused by plug-ins, calculations etc.?? If you are using 96khz shouldn't any degradation caused by such calculations be spread over a larger frequency range?? Thus when you convert down it should actually eliminate more of that degradation?? This makes sense to me I just don't know if I am explaining it correctly.

Not really, it's the bit depth that carries the accuracy of the maths involved, not the sample rate.

The sample rate determines which samples are used to reconstruct the wave - whether the wave is made from 44100 samples per second, or 96000 samples per second, the resultant reconstructed waveform is exactly the same, as long as your wanted audio is under the Nyquist limits for your sample rate...

For me, there is really only one tangible, fairly obvious benefit, in that some virtual instruments have a better sound at 96KHz, due to the aliasing calculations being done (some synths have an internal upsampling mode as well).

Your milage may vary, etc...

Sorry that doesn't work for me....Or rather that does not do anything to change my opinion on my previous post.

Yes I understand bit rate and all that, that really wasn't what I was getting at. Maybe my explanation is too convoluted but it still makes sense to me.

If you are using 44.1 does that not mean that the range is from 0-44.1? So when you apply something like a plug in it is contained within 0-44.1. When you convert it down it is now gone from 0-44.1 to say 0-20 thus losing 24.1 of range. Would a plug in not cover the whole spectrum? Thus any degradation it causes would be over that whole spectrum. Therefore at 44.1 you lose 24.1 yet if you are using 96 and you convert down you would be losing 76khz of range?? Wouldn't this allow the plug in degradation to be spread over a larger spectrum thus eliminating more of it when converting down???

This seems really simple to me yet oh so confusing.

Now what about calculations? I understand that the bit rate is the determining factor in the calulation but the degradation it causes would still be included in the range right? So wouldn't a higher sample rate allow more degradation to be spread out and thus when converted down you lose more of it??

Sorry if my terminology is un professor like, hopefully someone can shed some light on this not often talked about issue....

Quote Jocoserious:

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If you are using 44.1 does that not mean that the range is from 0-44.1? So when you apply something like a plug in it is contained within 0-44.1. When you convert it down it is now gone from 0-44.1 to say 0-20 thus losing 24.1 of range. Would a plug in not cover the whole spectrum? Thus any degradation it causes would be over that whole spectrum. Therefore at 44.1 you lose 24.1 yet if you are using 96 and you convert down you would be losing 76khz of range?? Wouldn't this allow the plug in degradation to be spread over a larger spectrum thus eliminating more of it when converting down???

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Depends on the effect. Imagine a delay/echo effect... it really doesn't care about your sampling frequency at all. Now, imagine a pitch shifter... ever notice how they sound better pitch shifting up instead of down? Now... flanger's do "pitch shifting" to some degree, so it likely has an effect there. Chorus's do some... but not a whole lot... so, maybe not so much there.

Quote Jocoserious:

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If you are using 44.1 does that not mean that the range is from 0-44.1?

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What "range"? 44.1 refers to the fact that 44100 sample are taken per second on a single mono stream. The actual amplitude values depend on the bit depth.

There is no 0-44.1 "range" of which you speak.

Quote Jocoserious:

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So when you apply something like a plug in it is contained within 0-44.1. When you convert it down it is now gone from 0-44.1 to say 0-20 thus losing 24.1 of range. Would a plug in not cover the whole spectrum? Thus any degradation it causes would be over that whole spectrum. Therefore at 44.1 you lose 24.1 yet if you are using 96 and you convert down you would be losing 76khz of range?? Wouldn't this allow the plug in degradation to be spread over a larger spectrum thus eliminating more of it when converting down???

This seems really simple to me yet oh so confusing.

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Unfortunately it seems that you really don't seem to have a grasp of how sampling and digital systems work, so any intuitive feelings you have, or conclusions you reach, are based on a flawed understanding are are therefore not going to be valid.

A bit like asking me to advise on, say, choreography or flower arranging.

I don't really like you, or rather your posts, not very helpful.

Like I said before I knew I was being convoluted. No need to call me an idiot, I can do that myself.

Yes I do understand pal, I was trying to explain something in more of a Lehmens term however I think I may have been too confusing. Essentially we are saying the same thing, either way a higher sample rate is going to provide a wider frequency range because having more samples per second allows you to capture higher frequencies because those travel faster, or something like that. I am not referring to dynamic range I am referring to frequency range. On a spectral frequency it shows from 0 up to half of the sample rate, at least that is what shows in the program I am using. Therefor 44.1 will not contain as much info as 96. Since we can't hear above 20khz anything we apply over a 96khz platform will move more into the inaudible region...

How to explain this properly is better left to someone much more qualified. Someone like Katz perhaps. "Thus there is considerable advantage of doing all our processing at higher sample rates, which moves the distortion products into the inaudible spectrum above 20khz".

Not sure who he is or how knowledgeable he really is, I am sure you are more qualified then he is. So sorry for not being able to simply communicate this very easy to understand topic. I mean there are only a million threads, articles and books on sample rate simply because it is so easy to understand and everyone enjoys debating known facts for the simple reason of debating.

All I know is that 96khz will contain more info than 44.1khz will regardless of whether we can hear it or not and regardless of exactly how and why it does. To me saying 0-44.1 or 0-96 is the same as saying samples per second sine wav this and that yada yada yada. To argue over semantics is a waste of time.

Cheers!