Look I don't really want to get this into another long winded sample rate debate where people get their panties in a bunch. I was merely being sarcastic and my last post should be taken in jest, because I felt you were a little rude Desmond.

Please feel free to lock this baby down or whatever needs to be done so this doesn't spin out of control way away from the OP's original question.

Quote Jocoserious:

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because I felt you were a little rude Desmond.

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If you took it that way, I apologise, but it really wasn't meant to insult you.

I felt that, from how you were describing things, that you really didn't have a grasp on the mechanics at work here. Perhaps you weren't expressing yourself very well, or perhaps I interpreted wrongly, but that's how it came across to me - if you took that comment personally, once again, it wasn't intended to be insulting at all.

After all, not understanding something is not equivalent to being an idiot, and I certainly didn't use that term or anything close (nor would I.)

Quote Jocoserious:

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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...

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Yeah. How much actual energy is in those frequencies though is very small - I don't know if tests have been done to determine how much energy is up in those frequencies, and certainly the jury is out on whether we can perceive them (although we're fairly sure we can't "hear" them, at least as we understand conventional hearing.)

Quote Jocoserious:

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Not sure who he is or how knowledgeable he really is, I am sure you are more qualified then he is.

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I can assure you I am by far less qualified than Bob Katz, don't worry...

Quote Interweaved:

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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.

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lol

This must be a joke.

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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No. It means that provided your signal was properly band width limited you will be able to perfectly reconstruct any sine waves below the nyquist frequency. That's all it means. You don't need any more points to reconstruct the sine wave, just like you don't need any more than two points to draw a straight line.

Quote:

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No. It means that provided your signal was properly band width limited you will be able to perfectly reconstruct any sine waves below the nyquist frequency. That's all it means. You don't need any more points to reconstruct the sine wave, just like you don't need any more than two points to draw a straight line.

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Yeah taking it out of context sure...The only thing I was trying to get at, a point which not many people seem to bring up, a point which most people seem to ignore and look past is the point of using a higher sample rate to push most of the distortion into an inaudible region.

I'm not trying to get into the more points argument, the more accurate argument, the whatever else argument. The above point about degradation and distortion was the only one I was trying to communicate and open for discussion.

If you would like I am sure you can go back to my posts and pull a lot of other one liners out that on their own are totally useless. Put them within the whole story and still you get a hard to follow muddled kind of explanation but it was all meant to try and explain ONE point, nothing else.

I am not saying that what you say is incorrect or anything of the sort, I am only saying that what you are correcting me on was not what I was talking about.

If someone has something to say regarding using higher sample rates to push distortion into inaudible ranges then I would love to hear it and will take any kind of tongue lashing and name calling and corrections you want to throw at me cause I am most definitely no expert, but please don't try correcting me on things I was not even talking about. That only shows you are either not reading the thread and my posts, skimming until you find something you can interject with your own wit and wisdom to make yourself look like the scholar, or else I make absolutely zero sense to anyone trying to follow which brings about the simpleton explanations of rudimentary nature.

Sorry for the long wind of hard to follow banter. I shall revert to the sidelines, I promise to never, ever get involved in another sample rate discussion.

haha although they do always spur on a lot of discussion lol

Quote Interweaved:

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Let me repeat my argument again...

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With the greatest respect, there is no point in repeating a totally flawed argument that doesn't adhere to the fundamental principle of the underlying maths as laid down in Nyquist's and/or Shannon's sampling theorum.

What you are claiming, while accurate, is irrelevant because sampling doesn't work if you use a sample rate which is exactly double the frequency of the signal of interest. The sample rate HAS to be MORE than twice the bandwidth of the source material. If it isn't, then it won't work, as you have shown. That's what the theorum explains, and that's why everything samples at a rate of more than twice the bandwidth being sampled. End of story.

Quote:

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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.

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Oh dear... It appears that you don't understand the Nyqusit theorem. Can I respectfully suggest you read this article which will go some way to explaining the process. Or the Lavry article cited above.

Quote:

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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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There would be no point as this example contravenes the Nyquist theorum, and it won't work. So no system will try to operate this way. It's about as useful as saying, see how this petrol engine won't run on diesel fuel, and therefore the whole internal combustion theory is crap!

Quote:

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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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It wouldn't just be 'closer', it would be totally accurate -- and the reason is that by sampling at a rate of more than twice the bandwidth of the source signal, you are now adhereing to the Nyquist rules and, surprise, surprise, it all works perfectly! Magic! But it is very wasteful and inefficient if you aren't particularly interested in the upper portion of the audio bandwidth.


Hugh

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Technical Editor, Sound On Sound

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.

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No. sampling at 44.1kHz allows the system to know preciselty what is going on up to about 20kHz (or whereveer the anti-alias filter rolls off). That 20kHz audio is 'reflected' above and below the 44.1kHz sample rate as 'images.' Again, check out that digital myths article in February's SOS to see what I mean.

The processing is only applied to the region defined by the sampling, between 20Hz and 20kHz (or whatever the filters allow).

There are some advantages to processing at higher rates, for sure. HF EQ can be made to work in a way which replicates analogue EQs much better. Dynamic processing is more accurate, as is metering...

But in plug-ins and systems that use oversampling internally while ostensibly working at 44.1kHz, the original audio bandwidth remains unchanged -- it is still
20Hz to 20kHz.

In systems that operate from the outset at 96kHz (or whatever), the original audio bandwidth will be 20Hz to 40kHz (or thereabouts) -- although whether there is actually any useful content above 20kHz depends on the quality and design of the microphones and other sources.

Hugh

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Technical Editor, Sound On Sound

Quote Jocoserious:

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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

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Calm down, please. He hasn't called you an idiot, but it does appear that you don't really understand it -- and there is no shame in that because this is a complicated topic that very, very few really understand completely. And it's not helped by the enormous about of misinformation that so many people pass around as solid facts!

Quote:

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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.

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See... you're not really sure are you. You are right, though, that a higher sampling rate allows a greater audio bandwidth to be described. So while a 44.1kHz sample rate caters for an audio bandwidth from, say, 20Hz to 20kHz, a 96kHz sample rate would cater for an audio bandwidth from 20Hz to 40kHz -- in broad terms.

Quote:

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Since we can't hear above 20khz anything we apply over a 96khz platform will move more into the inaudible region...

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Sampling at a 96kHz rate doesn't move anything. It just allows whatever is in the region between 20kHz and 40kHz to be captured.

Quote:

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"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".

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The first part of the sentence is correct. The second is confusing and misleading, and really needs more context to justify the claim. Bob Katz knows his stuff, but I fear this quote isn't doing him justice.

Quote:

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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.

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Don't apologise -- it is a hard topic to understand and discuss -- and of those millions of threads and articles, 90% are wrong or grossly misleading, sadly.

Quote:

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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.

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Correct.

Quote:

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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.

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No, this is not correct. These are not the same things, and the differences are important to a correct and reliable understanding of the subject. We are talking about a very clearly defined, but immensely complex process. Using confused terminology and inappropriate analogies doesn't help anyone.

Hugh

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Technical Editor, Sound On Sound

Quote desmond:

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Yeah. How much actual energy is in those frequencies though is very small - I don't know if tests have been done to determine how much energy is up in those frequencies, and certainly the jury is out on whether we can perceive them (although we're fairly sure we can't "hear" them, at least as we understand conventional hearing.)

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Yes, tests have been done. The amount of energy various, as you would probably expect, with the source. Things like trumpets and cymbals generate a huge amount of ultrasonic audio energy. So do things like switched mode (universal) power supplies and the inverters that drive the backlights in LCD screens. And if the microphones have a wide enough bandwidth (and many do, these days) all of this ultrasonic garbage that is inaudible to us naturally is being captured in pristine quality if you operate at 96kHz!

Worse... if there is some non-linearity in some of your digital processing the resulting intermodulation products may well render these ultrasonic signals to a lower frequency range where they will become audible

Hugh

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Technical Editor, Sound On Sound

Quote Jocoserious:

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Yeah taking it out of context sure...The only thing I was trying to get at, a point which not many people seem to bring up, a point which most people seem to ignore and look past is the point of using a higher sample rate to push most of the distortion into an inaudible region.

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What distortion? There is no distortion if the system is sampling properly according to the Nyquist theorum (and ignoring any quantising artefacts for the moment). Sampling is a perfect process, provided the filtering complies with the requirements -- which they do in all well designed systems. It doesn't add distortion, and sampling at 96kHz doesn't change that. It doesn't 'push' disortion products anywhere, either.

Sampling at 96kHz allows more accurate signal processing in some ways than working at 44.1kHz, and if you are using badly designed anti-alias and reconstruction filters, the audibility of their functioning will be reduced since the turnover is an octave higher. But that's it.

You are right though, threads on any aspect of digital audio are always busy and tempers tend to fray as those who do understand the topic get frustrated with those who don't, and those who don't get upset when their misunderstandings are exposed.

But as I've said, it is a very complex subject that is inherently hard to understand, and very few explanations in colleges, on the web, and even in some books get it right. To really get to grips with sampling theory, you also need a solid undertsanding of modulation theory and information theory -- both highly technical subjects in themselves.

Hugh

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Technical Editor, Sound On Sound

Hi Hugh,

I think the question has got way too complex. Surely the simple answer for 99.9% of people recording at home in a semi-pro - or even pro setup would be to suggest that 44.1kHz is far more appropriate than 96kHz.

Simple as that.

I would only suggest there is any point in recording at significantly higher frequencies if very high end gear was being used throughout for the purposes of audio research or very very high end (classical) recordings. The simply truth is that most other systems have other factors that affect the recording quality to a much larger degree than the sampling frequency ie. Jitter, D-A inaccuracies and distortion in both analogue and digital domains. Not to mention 'distortion' in the mixing process! And given that, again, 99.9% of people listen to music on CD or mp3 players surely recording at 96kHz becomes a bit of a nonsense.

BTW - Huge fan of SOS - bought the very first issue!

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

Quote Jonnypopisical:

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I think the question has got way too complex.

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I don't think questions can get too complex, only the answers! But this is a complex topic and there is still a huge amount of misunderstanding about it, as has been shown here. Hopefully, threads like this will help, in some small way, to reduce this kind of misunderstanding over time. That's why I perservere with it, anyway

Quote:

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Surely the simple answer for 99.9% of people recording at home in a semi-pro - or even pro setup would be to suggest that 44.1kHz is far more appropriate than 96kHz.

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I'd say more practical, rather than more appropriate.

Quote:

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The simply truth is that most other systems have other factors that affect the recording quality to a much larger degree than the sampling frequency

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You're certainly right there. Moving a microphone a few inches one way or the other would make a far more audible difference than switching between 44.1 and 96kHz sample rates.

Quote:

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And given that, again, 99.9% of people listen to music on CD or mp3 players surely recording at 96kHz becomes a bit of a nonsense.

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Not a nonsense. The source recordings should always be of the highest possible quality because anything that is then done to the material can only reduce quality. That's why high speed and wide format tapes were used in the studios back in the day when the consumer had manky 7-inch singles and ceramaic cartridges, or cassette players, or AM radio.

With modern digital equipment, the processing losses are far less than they were with analogue gear, but the principle remains the same. More worryingly, it is now quite common for domestic users to have replay equipment which is beginning to rival that of professional studios. DVD-A can accommodate 24/192 material, DTS discs can accommodate 24/96 material, and SACD is roughly equivalent to 24/96. MP3 is a convenience at the moment, like cassette was twenty years ago, but it may well be superceded by something far better in the not too distant future.

Quote:

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BTW - Huge fan of SOS - bought the very first issue!

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Thanks muchly

Hugh

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Technical Editor, Sound On Sound

I've been interested in this arguement for a while. Over on another thread, we have been discussing the article...

mixonline.com/recording/mixing/audio_emperors_new_sampling/

...where Paul Lehrman (one of Mr Robjohns' peers in the audio journalism world I believe) talks about a study that points to higher sampling rates not really making a noticable difference. Of course, the study refers to stereo recordings and it may still be better to record multitrack at a higher rate before downsampling but I wonder if it does make a difference. Before the days of oversampling plug-ins which seem to improve distortion and eq algorithms (as I've found out through using Guitar Rig, PSP products with and without the 'Fat' selected and my UAD plugs) the argument for 96K may have been stronger but I'd like to see a study testing the difference with using higher rates throughout the production process.
When I say I wonder, I'm not just using a rhetorical device as I do indeed wonder. I imagine people like Mr Robjohns have much more opportunity to test these things than I do but after reading the article above I'm going to stick to 44.1 and use oversampling plugs like the UAD 1073 when I can and stay happy.
As has been said previously in this thread - if there are differences that humans can hear, they are pretty darn small if we are still arguing about them and other factors should have much more effect. I'll stick with the extra tracks and CPU thanks.

Oh, yes. While we are handing out compliments, I have Hugh Robjohns (and Paul White, despite the 'buss' travesty) from SOS on my favourite authors list on Myspace. I've been reading you guys for coming up to 20 years now. Keep it up, Mr Robjohns, I get enormous pleasure in reading your articles every month. Seeing HR, PW or Martin Walker on the byline makes me feel like I'm listening to old friends.

Quote JohnnyT:

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...where Paul Lehrman (one of Mr Robjohns' peers in the audio journalism world I believe) talks about a study that points to higher sampling rates not really making a noticable difference.

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In theory, it shouldn't make any difference. If you accept that we humans really can't hear anything above about 20kHz (and the majority of evidence supports that premise), then any sampling system that captures and reproduces audio up to 20kHz should be perfectly adequate, and sampling at a higher rate to capture a wider bandwidth shouldn't sound any different.

However, the real world is a tad more complex as there are a lot of variables to consider. It was certainly the case a decade or two ago that the anti-aliasing and reconstruction filters were imperfect and had an audible effect. Sampling at a higher rate removed the audibility of the filters, and hence many claimed higher rate sampling was better. It was -- but because of filter imperfections, not higher sample rates per se.

There is also the issue of JF intermodulation to consider. There is a lot of ultrasonic noise in our modern world that we humans are not aware of because we can't hear it -- yet if the electronic equipment (analogue or digital) suffers non-linearities intermodulation can result and new audible signals can be created. Again, a decade or two ago, higher sample rate systems tended to suffer less from this than lower sample rate systems, but technology has moved on and it is much less of an issue now. So again, there is negligable sonic difference between standard and higher rate sampling systems.

Another aspect was the clocking precision and jitter. To work at all, higher sample rate systems have to have a very precise clock and a while back these tended to be better than systems that only operated at standard rates. But as technology and design experience has matured, clocking is no longer the challenge it once was, and so the differnece between standard and high sample rate systems has diminished.

But myths and ideas persist for decades in the audio industry -- people are very slow to change their ideas and update their beliefs, which is why these old chestnuts keep coming around and around.

Quote:

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I imagine people like Mr Robjohns have much more opportunity to test these things than I do but after reading the article above I'm going to stick to 44.1 and use oversampling plugs like the UAD 1073 when I can and stay happy.

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I think that is a very sensible and pragmatic approach to take -- assuming you have reasonably modern and well designed converters. Personally, I will continue to make the majority of my source recordings at 96kHz simply because most are stereo and very few involve more than eight tracks, the equipment I have makes it easy to do without penalty, and it makes sense from a maximal quality archive point of view. But I don't break down at the thought of recording projects at 44.1 or 48 -- and I do both frequently, and if a project is going to involve a lot of source tracks, then 44.1 or 48 is way more mangeable and uber-quality is generally less of an issue because it would normally mean I was into close miking and DIs and all that.

Hugh

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Technical Editor, Sound On Sound

Quote Hugh Robjohns:

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However, the real world is a tad more complex as there are a lot of variables to consider. It was certainly the case a decade or two ago that the anti-aliasing and reconstruction filters were imperfect and had an audible effect. Sampling at a higher rate removed the audibility of the filters, and hence many claimed higher rate sampling was better.

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This is still sort of the case with the vast majority of oversampled ADCs as the decimation filters are more often then not half band designs which are only -3db at 0.5FS. They fall off quickly, but that 0.05FS window is sufficient to make ultrasonic rich sources sound unnatural. This is aggravated by the outputs of many electronic instruments containing a lot of ultrasonic hash from the noise shaping used in the DACs (Building competent post filters costs money).

The fix is easy, run the ADC output at 96Khz, add an analogue filter going over at 25 or 30Khz, and resample the output of the ADC with a correctly designed decimating filter in the software....

Of course doing this is a trade off as it adds noise (the extra electronics), adds latency (the additional FIR resampler) and adds CPU and IO load. You could of course put a small cpu (or modest gate array) in the ADC box itself to do this final downsample correctly there at which point the apparent output will be at 44.1/48 without the problem, but nobody that I know in the prosumer card market does this.

Regards, Dan.

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Audiophiles use phono leads because they are unbalanced people!

I really like the last two posts. I'm not very convinced by subjective statements but the mention of JT inteference, the explanations of ultra-sonic complications within electrical systems and the non-human hearing based ideas make sense to me. Of course, all things need to be equal and it makes sense that equipment designed to work well at the higher rates should logically work better when not pushed as hard. The fact that many plug-ins do not oversample internally may also mean a better result when working with them at higher rates. If I seem a little anti high rates, then I'm sorry. I do think we need to find out why there are many reports from trusted sources saying that high rates are better, and if it's not the recorded sound itself then what it is that results in the better quality. I'm actually very interested in using 96K soon (I'm about to do some location sound for video and I'm going to use my M-Audio recorder at 96 as an experiment) but I've been burned with new concepts before - I have yet to be asked to mix or record anything in 5.1, despite spending a couple of grand on a monitoring setup and making sure everything I have is 5.1 capable. I realise this is a different type of issue but it has made me sceptical! As I said, I will continue to use 44.1 for multitrack until such time as I have excess CPU cycles and hard-disk space and the rest of my audio path is good enough. Or until someone can show me, either in a good demonstration or via a good study that it is worth it for me.

I would tend to concur with your scepticism, as I suspect that the cases where 96K sounds better are really papering over the cracks in poorly implemented hardware or software (And there is an awful lot of it out there).

It is not that hard to do right, but it will always cost more (extra electronics for hardware, more CPU load for plugins), and the headline SNR specs for hardware will be slightly worse then they would have been with it done wrong.

Unfortunately the benefits are not always obvious, especially on a spec sheet as the frequency response and SNR are worse in the box done right then they are in the one without the extra filters, and most marketing runs on specs, not real world performance.

Regards, Dan.

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Audiophiles use phono leads because they are unbalanced people!

The topic is so complicated and really the only thing which matters is that you either use 44.1/48 or 88.2/96 and you hopefully try and record whatever you are trying to record well. Guitars Amps, Bass Cabs, Singers, Drums.... if not recorded well... who cares if it was recorded at 192 because it ends up being 44.1 anyway and is still crap

I think if you record at the source well, and capture what you want then regardless if it was recorded at 44.1 or 88.1 it is going to sound good.

I just feel sorry for Hugh's fingers... his keyboard must be dented from all that typing

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There is nothing Grim about this Reaper
We Fell From The Sky

This thread should be a sticky, along with a thread about multi track recording connections and equipment.

I record at 96khz lately, mostly for the lower latency I can get with my M Audio AP192 used as monitor with VST effects path digitally connected to my EMU 1212M through SPDIF connections.

Quote Henry-S:

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The topic is so complicated and really the only thing which matters is that you either use 44.1/48 or 88.2/96 and you hopefully try and record whatever you are trying to record well.

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Well, sort of....

But it has always seemed to me that if you want to be able to consider yourself an engineer as opposed to a technician, then you really need to understand how your tools work and the hairy edge cases where your tools stop behaving ideally and collide with the real world (And how to fix it when it happens)!
This is not really all that complex, compared to say grade 8 music theory!

There is a place for both skill sets of course, and I am well aware that I am far better at system design and implementation then I am at using that system to a creative end, my stuff does not hum, and you can use GSM phones around it with impunity, but my mixing skills (for studio based music) are still a work in progress (Live is different, mostly because good enough now is better then perfect in 3 weeks).

Regards, Dan.

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Audiophiles use phono leads because they are unbalanced people!

Quote Hugh Robjohns:

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Oh dear... It appears that you don't understand the Nyqusit theorem. Can I respectfully suggest you read <a href="/sos/feb08/articles/digitalaudio.htm" target="_blank">this article</a> which will go some way to explaining the process. Or the Lavry article cited above.

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Eh, I know well what the theorem states... I'm stating that these guys are over-stating the results of the theorem. Believe me, I sat there at the same university Nyquist got his PhD at arguing with some well established physics professor, who was a great hobbyist with accoustics and these things, the same point I'm arguing with you.

See, if F is the frequency you're interested in, 2F is necessary in order to avoid aliasing. 2F won't give you a 100% accurate reproduction of the original wave though.

Say you're interested in recording from 0 to 22 kHz...
A) If you record at 44kHz, you won't get aliasing. Aliasing makes extra noises where they don't belong.
B) If you record at 44kHz, you won't get a 100% perfect reproduction of all signals in the 0 to 22 kHz range.

People who think that A) implies a "perfect reproduction" (of the 0-22kHz range) aren't thinking this through and don't seem to understand that sometimes results get overstated.

I could actually prove what I'm saying with my equipment, but I'm sure you'd still say I was wrong because you believed the overstatement of the theorem.

Quote Interweaved:

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See, if F is the frequency you're interested in, 2F is necessary in order to avoid aliasing.

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And there you just proved that you completely clueless about the whole matter.

It's sad that people that pick up a few key words from some forum come here and self proclaim themselves as some kind of authority, even referencing supposed academic credentials and then just go on to regurgitate a load of crap completely messing themselves up in the procvess and possibly confusing a lot of people.

I hope most people reading this forum understand that you don't know what you are talking about.

Interweaved - are you trying to argue that Nyquist and Shannon were wrong? Or are you saying that current equipment falls short of perfection?

And don't forget that it is > rather than =. And while we're at it, don't ignore the filters.

Cheers

James.

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JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Quote James Perrett:

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Interweaved - are you trying to argue that Nyquist and Shannon were wrong? Or are you saying that current equipment falls short of perfection?

And don't forget that it is > rather than =. And while we're at it, don't ignore the filters.

Cheers

James.

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My argument is that people mis-interpret what the theorem does provide and what it doesn't.

Quote Interweaved:

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Eh, I know well what the theorem states... I'm stating that these guys are over-stating the results of the theorem.

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Eh? The theorum defines the correct mathematical operation needed to achieve perfect sampling, the correct application of which allows the source signal to be recreated perfectly. How can that be overstated?

Quote:

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See, if F is the frequency you're interested in, 2F is necessary in order to avoid aliasing.

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No, that is not the case. For a start, the sample rate is related to the bandwidth of the source signal, not just it's frequency... but ignoring that nicety for the moment, if F is the frequency you are interested in, the sampling rate must be 2F+n, where n is sufficiently large to enable the real-world filters to achieve the necessary attenuation of the unwanted image(s).

Quote:

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2F won't give you a 100% accurate reproduction of the original wave though.

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Absolutely. Which is why no practical audio sampling system samples at 2F. So why are you harping on about something that isn't done, isn't a problem, and isn't relevant?

Quote:

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Say you're interested in recording from 0 to 22 kHz...
A) If you record at 44kHz, you won't get aliasing. Aliasing makes extra noises where they don't belong.

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That's debateable. The bottom of the lower sideband will fall precisely on the top of the wanted audio, and thus any wanted signal components at 22kHz won't be perfectly retrievable because of its superimposed alias.

Quote:

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B) If you record at 44kHz, you won't get a 100% perfect reproduction of all signals in the 0 to 22 kHz range.

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Obviously, because the sample rate isn't high enough and doesn't conform to the Nyquist requirements.

Quote:

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I could actually prove what I'm saying with my equipment, but I'm sure you'd still say I was wrong because you believed the overstatement of the theorem.

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Not at all. I could prove what you are saying with my equipment too. But the point is that what you are arguing is NOT what the theorum states. Nyquist and Shannon both state that the sample rate must be MORE THAN twice the bandwidth of the wanted signals. Not equal to twice.

Nothing is being overstated, but there does still seem to be some misunderstanding or confusion... Hopefully you can now see where that error lies.

Hugh

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Technical Editor, Sound On Sound

Quote Interweaved:

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My argument is that people mis-interpret what the theorem does provide and what it doesn't.

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They do indeed... and it is categorically your good self that has misinterpreted it. Sad but true.

Hugh

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Technical Editor, Sound On Sound

Quote Hugh Robjohns:

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Nyquist and Shannon both state that the sample rate must be MORE THAN twice the bandwidth of the wanted signals. Not equal to twice.

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May of us on this thread have stated this multiple times, and yet still Interweaved goes "yeah, but when it's not more than twice the bandwidth, it doesn't work".

We know. This is what the theory says. I don't understand what you are debating...

Quote desmond:

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Quote Hugh Robjohns:

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Nyquist and Shannon both state that the sample rate must be MORE THAN twice the bandwidth of the wanted signals. Not equal to twice.

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May of us on this thread have stated this multiple times, and yet still Interweaved goes "yeah, but when it's not more than twice the bandwidth, it doesn't work".

We know. This is what the theory says. I don't understand what you are debating...

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I think this is what is usually called a troll?

Quote Hugh Robjohns:

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Quote Interweaved:

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My argument is that people mis-interpret what the theorem does provide and what it doesn't.

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They do indeed... and it is categorically your good self that has misinterpreted it. Sad but true.

Hugh

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Ouch !

To answer the original question the as is 44

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

Quote Hugh Robjohns:

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Not at all. I could prove what you are saying with my equipment too. But the point is that what you are arguing is NOT what the theorum states. Nyquist and Shannon both state that the sample rate must be MORE THAN twice the bandwidth of the wanted signals. Not equal to twice.

Nothing is being overstated, but there does still seem to be some misunderstanding or confusion... Hopefully you can now see where that error lies.

Hugh

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That doesn't matter so much. MORE THAN yo mamma.

But, eh, you can record at your frequency, I'll record at mine. (I actually am using 44, though, because there really isn't much content in the upper frequencies in most music I'm doing).

There's a bloke down the pub..

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Stuff what I done

my homemade test

16 bits 44 khz
http://motthieu.free.fr/freq/with-EQ-1644.aif

24 bits 96 khz converted to 16 bits 44 khz
http://motthieu.free.fr/freq/with-EQ-2496%20converted%20to%201644.aif< br />
with my soundcard, Echo Audiofire 8, you'd better record at 96 khz. do the test with your soundcard

I like your answers...You are free to put me to shame any day Hugh, I always learn something when you do.

I should really learn to stay out of these ones as I can never compete with a lot of you guys and just end up looking stupid....

That being said I still think I understand...somewhat....I just am absolutely terrible at trying to take what is in my brain and put it into words that all you really smart people can understand....Not only that like you say it is a very complicated subject and I apologize if I appeared to be ranting or angry. I try and keep my place as much as possible in here, which is at the bottom of the food chain.I really was only trying to make a point and unfortunately sometimes you need to push a few buttons to do so...

p.s. For the record I hate to quote anybody directly out of context, and try and refrain from doing so. I used that small quote from Katz because really, I would have had to use 5 pages to make it make any sense

Ok, here's a wave file of 18, 19, 20, and 21 kHz sine waves at 24 bit 192 kHz... you should be able to play this back at different rates yourself and hear the difference.

If you do not turn down your volume before listening to this file, I will laugh when you come back here crying about it.

http://www.soundupload.com/audio/davrr8p5utguaj7z

Quote Jocoserious:

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I like your answers...You are free to put me to shame any day Hugh, I always learn something when you do.

I should really learn to stay out of these ones as I can never compete with a lot of you guys and just end up looking stupid....

That being said I still think I understand...somewhat....I just am absolutely terrible at trying to take what is in my brain and put it into words that all you really smart people can understand....Not only that like you say it is a very complicated subject and I apologize if I appeared to be ranting or angry. I try and keep my place as much as possible in here, which is at the bottom of the food chain.I really was only trying to make a point and unfortunately sometimes you need to push a few buttons to do so...

p.s. For the record I hate to quote anybody directly out of context, and try and refrain from doing so. I used that small quote from Katz because really, I would have had to use 5 pages to make it make any sense

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Hey that sounds familiar to someone I know..

People who aren't afraid to make mistakes learn the quickest.

Hugh's only chink in his armour is that he doesn't know how us stupid people think.

Aren't we getting bogged down into talking about a part of a whole thing?

If Hugh says that 96k can help emulate analogue EQs in a way that deals with the high frequencies, is this then a mathematical side to the processing argument as opposed to the "sampling" process of capturing an audio signal, which is mechanical?

Quote Interweaved:

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Ok, here's a wave file of 18, 19, 20, and 21 kHz sine waves at 24 bit 192 kHz... you should be able to play this back at different rates yourself and hear the difference.

If you do not turn down your volume before listening to this file, I will laugh when you come back here crying about it.

http://www.soundupload.com/audio/davrr8p5utguaj7z

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First impressions......

Sounded very clear and crisp (over laptop) speakers.

Would be good if you processed some cymbals with EQ to give as an idea of differences.