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Author Topic: Resample to 44.1 or record at 44.1 (midside Q)  (Read 10067 times)

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Resample to 44.1 or record at 44.1 (midside Q)
« on: December 20, 2008, 12:59:57 AM »
I've been playing around with my recorder when doing MS (raw encoded) stuff, and I've tried doing 88.2 resampled to 44.1 in Audacity and recording straight to 44.1.

From a technical midside decoding standpoint, is there an advantage to doing the resample, or is it a bust and recording at 44.1 is good enough? Does it make a difference in the end product (at 16/44.1)?

I haven't had a chance to do any real comps in a controlled environment, so I don't know. I figure someone out there has looked into this before.
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #1 on: December 20, 2008, 02:05:29 AM »
Sampling rate discussions can go on as long as the pious could argue during the Middle Ages about how many angels could dance on the head of a pin.  People who really know (Lavry) have argued and written at length that above 48kHz is kind of a waste.  I may be wrong about the exact number, but the fellow argues that filters and so on make really high sampling rates ridiculous.  I also wonder if binary math is easier on numbers which are multiples of two in decimal; i.e., 88.2 to 44.1  That said, I record at 24/48.  I know some folks say they can hear the difference between 48 and 96 and 192.  I have yet to see one person do this in a double blind test (ABX).  If it has been done, please show me the test results. 

For MS I cannot imagine why rates would be any different than other mic arrays.  I use MS a lot.  Love it in clubs, etc., where I have to set up close and in a hurry.  It is also inconspicuous which makes the performers and management happy.

As usual, YMMV     8)
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #2 on: December 20, 2008, 02:16:31 AM »
Sampling rate discussions can go on as long as the pious could argue during the Middle Ages about how many angels could dance on the head of a pin.  People who really know (Lavry) have argued and written at length that above 48kHz is kind of a waste.  I may be wrong about the exact number, but the fellow argues that filters and so on make really high sampling rates ridiculous.  I also wonder if binary math is easier on numbers which are multiples of two in decimal; i.e., 88.2 to 44.1  That said, I record at 24/48.  I know some folks say they can hear the difference between 48 and 96 and 192.  I have yet to see one person do this in a double blind test (ABX).  If it has been done, please show me the test results. 

For MS I cannot imagine why rates would be any different than other mic arrays.  I use MS a lot.  Love it in clubs, etc., where I have to set up close and in a hurry.  It is also inconspicuous which makes the performers and management happy.

As usual, YMMV     8)

Yeah, I can't hear much difference between 44.1 and 48, and I can't tell a difference between 48 and 96 at all, however from a math and processing standpoint, I'm curious if it affects the final result of the MS decode, and thus mixing at 88.2 then downsampling would produce a different result then just mixing at 44.1. I have a tendancy to agree, in that I don't think it does, but I'm curious.
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #3 on: December 20, 2008, 01:02:33 PM »
boojum, you wrote:

> I also wonder if binary math is easier on numbers which are multiples of two in decimal; i.e., 88.2 to 44.1

Some SRC routines run more efficiently with simple integer ratios while others don't, so this depends on what software you have. I ran my own little test just now: I took 3 minutes of music recorded at 96 kHz, and converted it both to 48 kHz and to 44.1 kHz. With Adobe Audition 1.5 the conversion to 48 kHz took about half as much time as the conversion to 44.1, but with Sound Forge 7.0 the times required for the same two conversions were within 1% of each other. So, let's beware of generalizations that are overly broad.

Precision and accuracy are what could affect the sound quality of the result, and they're a separate issue in the design of an SRC routine. (They shouldn't be too hard to measure, but I have yet to see any data at all about that in any software reviews.) In the end, I'd say that whether the eventual 44.1 kHz recording might sound better if you start out recording at some higher sampling frequency depends on the equipment that you record with originally, on the particular SRC method that you use, and on whether more accurate audio pleases your particular ears. There can't possibly be one answer that everyone would always agree with.

Technically there's no reason why 44.1 kHz can't sound as good as higher sampling rates, but that is not to say that all existing 44.1 kHz converters sound good to everyone. Whatever equipment you're working with--regardless of what might have been possible in theory if things had been done differently--you surely want to use it in the way that you think sounds best. There's something to be said for the "let the Wookiee win" principle.

--best regards
« Last Edit: December 21, 2008, 09:38:57 AM by DSatz »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #4 on: December 20, 2008, 10:21:49 PM »
I've been playing around with my recorder when doing MS (raw encoded) stuff, and I've tried doing 88.2 resampled to 44.1 in Audacity and recording straight to 44.1.

From a technical midside decoding standpoint, is there an advantage to doing the resample, or is it a bust and recording at 44.1 is good enough? Does it make a difference in the end product (at 16/44.1)?

I haven't had a chance to do any real comps in a controlled environment, so I don't know. I figure someone out there has looked into this before.

I take a pragmatic approach to this question (as I have failed to hear any difference in A/B blind listening).

If I am working on an audio project it is 44.1. If I am working on audio for video then 48 kHz. If I am recording samples with the view to transposing them downward (and so care about ultrasonic frequencies) I try to work at 2X the final sample rate (88.2 or 96 kHz).

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #5 on: December 24, 2008, 11:32:41 PM »
I don't get the 2 times logic, please fill me in.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #6 on: December 25, 2008, 04:40:37 AM »
I don't get the 2 times logic, please fill me in.
nyquist--you have to sample at twice the frequency of your signal in order to avoid aliasing
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #7 on: December 25, 2008, 10:27:28 AM »
An important consideration is target format for the 24-bit product.  A DVD-A will carry 44.1k or 48k  but a DVD-V format "audio DVD" is restricts LPCM streams to 48k or 96k.  Not all DVD players will read DVD-A format; DVD-V format is universally compatible.   

If you plan to circulate 24-bit discs, you should record at 48k so that you have the option to author into both formats. 
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #8 on: December 25, 2008, 10:42:22 AM »
I don't get the 2 times logic, please fill me in.
nyquist--you have to sample at twice the frequency of your signal in order to avoid aliasing

I think he's talking about twice the sampling frequency for the working bit stream, not twice the highest frequency he intends to reproduce.  88.2 is 4X the highest audible frequency one would expect to reproduce from a CD. 

The usual argument for processing at the higher rates and then down sampling is that the digital processing (filters, EQ, etc) produce better results when working with more data.  I don't have any experience with post processing tools so I don't know if any difference in product is audible.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #9 on: December 25, 2008, 11:38:14 AM »
I think when I do 24 bit in safe venues with my laptop, I am going to run 24/48 from now on. I normally go 24/44.

Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #10 on: December 25, 2008, 05:34:50 PM »
When you know you're going to release something in a 16-bit format, it makes complete sense to record 24-bit, and do all your processing with that increased level of precision before you set your final levels and dither the signal down to 16 bits.

However, it makes no such sense to record at a higher sampling rate (sampling frequency) unless you have some evidence that your particular signal processing chain works better that way--a conclusion which would require careful testing. In general it shouldn't be expected; there's certainly no theoretical reason to expect it. Sampling rate and bit depth are entirely different things.

It could be an advantage in a particular case, due to some shortcoming of a particular 44.1 kHz converter, for example. But if that were a widespread situation, then recording at 96 followed by conversion to 44.1 ought to sound distinctly better to the people who prefer 96 kHz recording than recording at 44.1 in the first place sounds. In general that doesn't seem to be true.

Think this through with me, please: When you sample at (say) 96 kHz, the signal components below 22.05 kHz aren't recorded any more accurately than they are recorded by sampling at 44.1 kHz. But the filtering occurs at 48 kHz instead of at 22.05, and even though no one can hear between 22.05 and 48 kHz, the signal components below 22.05 kHz may be handled in a more linear fashion. Such is the rationale for 96 kHz recording in a nutshell.

Unfortunately, when you then convert down to 44.1 for release, you must filter a second time (at 22.05 kHz) to avoid aliasing distortion. Signal components between 22.05 and 48 kHz can't be allowed at that point. So in the end, you put the signal through more filtering--not less--by starting out at a higher sampling rate. The entire approach is self-defeating in that sense.

--best regards
« Last Edit: December 27, 2008, 07:42:06 PM by DSatz »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #11 on: December 25, 2008, 07:21:59 PM »
If you seriously think that now, or someday, you may want to playback on a format where the required bitrates are 48k or 96k, then record at that rate, and resample to 44.1k for CDs.  But if you don't see that happening, or can't be bothered with backing up those original files, then you are better off to record at 44.1k

I think it is a huge misconception that people have that recording at 48k, 96k, or 192k and then resampling to 44.1k in post is superior to an original recording at 44.1k.  That's not true.  It can't be.  Your A/D oversamples the signal by a large factor, and it has lots of information in front of it to make good decisions, and it is MUCH better suited to make these decisions than your computer.  Let it do it's job!  In post your computer has much less information to go buy, so it takes a 48k data file and basically says "use 146 samples out of 147, and then skip that 147th one", at least that's a simple approach that some software might take.  There are more exotic algorithms than that, but basically they are all trying to solve a tough mathematical problem that doesn't need to be a problem in the first place!  http://en.wikipedia.org/wiki/Sample_rate_conversion

Most of the time I record at 24/48 because I really do upload some of those 24/48 flac files to the LMA, and I burn DVD backups of them for "someday".  Rarely, I will even do that at 96k because of peer pressure.  But, the last show I went to, I knew that would not be the case... so I recorded at 44.1k
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #12 on: December 27, 2008, 05:43:50 PM »

I think he's talking about twice the sampling frequency for the working bit stream, not twice the highest frequency he intends to reproduce.  88.2 is 4X the highest audible frequency one would expect to reproduce from a CD. 

The usual argument for processing at the higher rates and then down sampling is that the digital processing (filters, EQ, etc) produce better results when working with more data.  I don't have any experience with post processing tools so I don't know if any difference in product is audible.


I was indeed just making the sample conversion to 44.1 or 48 kHz (theoretically) more accurate by sticking to an exact multiple of the final frequency (working habit not based on any audible difference). I only record above 44.1 kHz if (as I noted) I am going to do some sample manipulation and slow the recording down, so that ultrasonics become audible, such as you may do when recording Bats or smashing glass for example.

It has always intrigued me when I see how much signal manipulation goes on in the studio (and the miles of cables/patchbays/gear/desks etc) the fuss people make over details of digital audio :)

I will remain on the record as rarely hearing any meaningful difference between good preamps (except for noise levels) and gear. Particularly in comparison to the huge difference mic placement & post production (can/does) make to the sound. I used to think I did, but got into the habit of blind listening as the final arbiter.

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« Last Edit: December 27, 2008, 05:47:17 PM by digifish_music »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #13 on: December 27, 2008, 09:32:47 PM »
It seems sampling opinions can be geographic.  On other boards folks will argue, and prove, that hgiher sampling rates do make a difference, and they can hear them!  I still am not sure.    8)
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #14 on: December 28, 2008, 12:06:34 AM »
It seems sampling opinions can be geographic.  On other boards folks will argue, and prove, that hgiher sampling rates do make a difference, and they can hear them!  I still am not sure.    8)

Prove? I have (as yet) never seen any properly conducted experiment that showed people can tell the difference between any sample rate above 44.1 kHz or bit depth above 16 (all else being equal), the data points to the contrary. Got links?

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« Last Edit: December 28, 2008, 12:09:33 AM by digifish_music »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #15 on: December 28, 2008, 02:08:42 AM »
Prove? I have (as yet) never seen any properly conducted experiment that showed people can tell the difference between any sample rate above 44.1 kHz or bit depth above 16 (all else being equal), the data points to the contrary. Got links?

digifish

I find it amusing that the referenced article is offered in two formats "(PDF-615KB)" and "(HI-RES PDF-9.4MB)". 

Unfortunately the article is available to AES members only.

In my playback system I have done AB testing and while sample rate doesn't seem to make a difference, 24-bit vs 16-bit does.  Could be the authoring of the 24-bit material or maybe the DAC does a better job with 24-bit sources.  I don't know why the 24-bit sounds different to me but I've always allowed that maybe I imagine it.  I've read several reviews of up-sampling DACs that stated the up-sampled 16-bit sources sound every bit as good as the native 24-bit material.  In anycase, I use tubes so the noise level and frequency range of my gear is above and below the respective thresholds for 16-bit material anyway. 
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #16 on: December 28, 2008, 01:11:15 PM »
digifish_music, bit depth (e.g. 16- vs. 24-bit) is a clear-cut, practical matter about which there is no great controversy or mystery. It corresponds directly to the dynamic range of a recording.

If you don't know in advance what the maximum sound pressure level at a performance will be, and you're there to record it, 24-bit recording offers a definite advantage since it allows you to set your levels very conservatively. Then once the concert is over and you know what the peak levels actually were, you can renormalize the recording and dither down to 16 bits. If the maximum levels were, say, -8 dBFS then you can have a nearly 8 dB improvement in the dynamic range of the finished product, as compared with recording the concert at 16-bit resolution using the same level settings.

I record classical concerts professionally, and am generally the only audio engineer at most of the events I record. To me this approach seems far more responsible (though less exciting, perhaps) than the old way of trying to guess and occasionally, guessing wrong which I did for 35+ years before I changed over to 24-bit recording for most events.

In my opinion 16 bits is usually more than enough for domestic audio, but occasionally some program material comes along (e.g. percussion ensembles) which can really exploit almost its full range, so I'm glad to have it. And for me, 24-bit recording is the low-stress way to get an optimal 16-bit recording--it's not an end in itself. As far as a concrete proof (or demonstration) of this is concerned, that is actually not difficult at all, and if you'd like to arrange something I'd be glad to help. But it's simply not correct to claim that there has been no proof of it up to now--it is an elementary principle which has been very well known for decades.

The question of sampling rate (sampling frequency) is rather a different sort of topic to discuss critically, and I don't want to try to go there in the same message. I'd rather try to establish some clarity about bit depth, since that really is pretty simple.

The only strong caveat with bit depth is to remember that there is always noise in any audio system, and the actual signal-to-noise ratio of any real-world recording (i.e. a rendering of acoustic events, rather than something algorithmically generated) will always be distinctly less than the maximum which the 24-bit format would allow. 24 bits = almost 143 dB dynamic range, which exceeds by far the dynamic range of the quietest physically possible microphones, mike preamps and A/D converters (let alone the quietest ones which are actually available). Thus by definition any 24-bit recording has less than 24-bit "actual" resolution (i.e. signal-to-noise ratio), and it's important not to confuse the container with its contents.

--best regards
« Last Edit: December 28, 2008, 01:16:54 PM by DSatz »
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Offline boojum

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #17 on: December 28, 2008, 01:40:27 PM »
Nor am I convinced.  I have been down this road many, many times.  Oh, yes.  Green magic markers on the edges of CD and all the rest of the hoo-ha touted by folks "who know" or claim they know or print expensive and revered magazines.  Remember the Shin-Mook wooden dots?  Made everything sound better.  Oh, yeah, right.

There are a couple of other boards, pro boards, which are very strong for recording at 96kHz or higher.  Indeed, I have seen 48 kHz pooh-poohed on this board as "not good enough for serious music recording."  But I, too, have never seen anyone put forward double blind ABX tests to support their positions.  They just "know" they can hear the difference without testing.  Well, good for them.  I am still not sure.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #18 on: December 28, 2008, 05:21:05 PM »
digifish_music, bit depth (e.g. 16- vs. 24-bit) is a clear-cut, practical matter about which there is no great controversy or mystery. It corresponds directly to the dynamic range of a recording.

I agree, but I also wasn't talking about production/post production, I was talking about listening to the final mix. Can you find any scientific paper that shows people can reliably tell a 24 bit from 16 bit derivation of the same 24 bit recording under controlled (normal level) listening conditions?

Evidence against.

Quote from: DSatz
24 bits = almost 143 dB dynamic range, exceeds by far the dynamic range of the quietest physically possible microphones, mike preamps and A/D converters (let alone the quietest ones which are actually available). Thus by definition any 24-bit recording has less than 24-bit "actual" resolution (i.e. signal-to-noise ratio), and it's important not to confuse the container with its contents.

Indeed again, in reality the bit depth of most "24 bit" D/A's is 18-20 bits ~20 is approx the limits imposed by electronic components at room temperature, and as good as it is ever going to get. IMO there is still much more sonic milage to be gained by focusing on making the analog stages better (so that a good D/A or A/D can make use of the bits it has).

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« Last Edit: December 28, 2008, 09:14:03 PM by digifish_music »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #19 on: December 28, 2008, 05:33:25 PM »
They just "know" they can hear the difference without testing.  Well, good for them.  I am still not sure.

Cheers

Nor should you be. Despite years of research there has been very scant evidence that people respond to frequencies above 20 kHz (much less if you are older than 20) and many studies that show there is no effect. The recent 'The Emperor's New Sampling Rate, summarises one of the better (recent) studies.

Also it's worth having a read of this...

Perceptual Discrimination between Musical Sounds with and without Very High Frequency Components, NHK Labs.

and this

David Griesinger's (of Lexicon) Experiments in Ultrasonics

IMO it's time to put this to bed and move on. After all the reason for the push has been equipment manufacturers trying to stop their market stagnating and heading for low-cost where there are small margins. Record labels trying to replicate the 80's boom where people re-purchased all their old LPs on CD, they were hoping for it to happen again with DVD audio or SHCD, fat chance.

I think one of the big problems for the average punter is that they buy some new gear that has 24 bit @ 96 kHz sampling and it does sound better than their old gear, so they assume it's all about the bit depth. In reality it's got more to do with the better analog design and possibly better quality D/A or A/D, but not the # bits or kHz.

digifish
« Last Edit: December 28, 2008, 07:29:24 PM by digifish_music »
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #20 on: December 29, 2008, 12:21:22 AM »
digifish_music asked:

> Can you find any scientific paper that shows people can reliably tell a 24 bit from 16 bit derivation of the same 24 bit recording under controlled (normal level) listening conditions?

If I were to put forth a fancy-sounding claim that a person with normal hearing can perceive a 1 kHz tone at 65 dB SPL, it would be rather silly--not because it's wrong, but because everybody already knows it. If I were to write up my claim as a research paper and describe clinical trials with controlled, double-blind listening experiments, I'd have some difficulty getting it published. It's the old "dog bites man" vs. "man bites dog"--it wouldn't have any news value.

Similarly, if someone were to carry out controlled listening tests on telling a 24-bit system apart from a 16-bit system, I can see very little reason for anyone to publish the result. The outcome is easily predictable from basic knowledge that's been available for decades, and it would just be a waste of journal space.

In a very quiet playback environment (< 20 dB ambient SPL for example), if you set the gain of a playback system so that the peak levels are at 120 dB SPL, then the noise floor of a dithered 16-bit system will be discernably higher than the ambient noise level. (I chose 120 dB SPL because it is close to the highest SPL that I have directly measured with a number of the opera singers I have recorded; let's say I want to play back their singing at precisely realistic levels.) At an identical playback gain, however, the noise floor of a dithered 20-bit system would fall distinctly below the ambient noise level. And so would that of a dithered 24-bit system. This difference in noise floor would be obliterated in the presence of any program material, but would be discernable in the "silence" between tracks.

You want to doubt it? Doubt it--but try it some time, please. The noise floor of a dithered 16-bit system is extremely low when compared to analog tape of all kinds, but it isn't so low that it is never, ever audible. With 20- and 24-bit systems the noise floor is distinctly lower, and there's simply no reason for that difference to be inaudible to a consumer when the playback gain is set high enough.

--best regards
« Last Edit: December 29, 2008, 05:05:25 PM by DSatz »
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Offline digifish_music

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #21 on: December 29, 2008, 01:41:30 AM »

In a very quiet playback environment (< 20 dB ambient SPL for example), if you set the gain of a playback system so that the peak levels are at 120 dB SPL, then the noise floor of a dithered 16-bit system will be discernably higher than the ambient noise level.


I don't doubt it (indeed the paper I linked too says the same, "at very elevated levels"), but then those are not 'normal' listening conditions (so that peaks are 120 dB). It's very hard to find spaces with 20 dB background noise and very few recordings are made that use 120 dB dynamic range...

The paper I pointed to suggests the opposite about the differences between 44.1@16 bit vs others -

http://www.aes.org/e-lib/browse.cfm?elib=14195

24 & 16 bit are indistinguishable over the range of 'normal' & 'loud' listening levels.

For production then 24 bit makes a lot of sense, for playback not at all.

So someone did 'waste space' - Abstract:

"Claims both published and anecdotal are regularly made for audibly superior sound quality for two-channel audio encoded with longer word lengths and/or at higher sampling rates than the 16-bit/44.1-kHz CD standard. The authors report on a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz “bottleneck.” The tests were conducted for over a year using different systems and a variety of subjects. The systems included expensive professional monitors and one high-end system with electrostatic loudspeakers and expensive components and cables. The subjects included professional recording engineers, students in a university recording program, and dedicated audiophiles. The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems. The noise of the CD-quality loop was audible only at very elevated levels."

digifish 
« Last Edit: December 29, 2008, 05:47:17 AM by digifish_music »
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #22 on: December 30, 2008, 09:00:20 AM »
digifish, in that study (written by two guys I know from Boston, and definitely a fun read), recordings made at sampling rates higher than 44.1 kHz were converted down to 16-bit/44.1 kHz. The study didn't show that no sound quality was lost--only that the subjects in that study, under particular listening conditions, didn't reliably perceive any loss in sound quality.

That's a nice data point, but it doesn't paint a complete, permanent picture. Sample populations are always finite, people can learn to hear some sonic defects that they couldn't hear before, and another well-constructed, controlled study could perhaps be published some day in which one or more people do hear a difference to a reasonable degree of statistical confidence. I don't personally expect that--but that's as much a matter of personal belief as it is of science.

In any case some people here (including me) like to go beyond merely "loud" to VERY LOUD at times. The study didn't deal with that, and it has a direct bearing on the issue of bit depth.

If someone feels that a particular 96 kHz recording sounds better to him (whatever "better" may mean to him at the time) than recordings from whatever 44.1 kHz systems he happens to have heard before, I feel that I have to leave that alone. I may have big doubts, but I simply don't know of any way to tell a person that for reason x, y and/or z he can't possibly be right about his own opinion, without insulting him and making a fool of myself. Maybe he hears certain things better than I do--I've definitely met some people who do.

--best regards
« Last Edit: December 30, 2008, 12:45:47 PM by DSatz »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #23 on: January 06, 2009, 09:30:51 PM »
I still find it interesting... we go to a rock show.  Guitarist plays guitar into amp.  Amp is mic'ed with SM57 with 'frequency response' or 40-15,000 hz.  Vocalist sings into SM58 with similar response.  All this goes into mixing board, into stacks, we sit 100' from stacks with "good mics" with response from 20-20,000hz to record the room.  A 44.1k bitrate really will capture all of the above.  Maybe the lady shrieking next to my Earthworks (up to 30,000 hz) will benefit from a higher sample rate recording, but she's not that important to me.  Still, the taper next to me will insist on recording at 96k, even though there is nothing on that wav file higher than 20khz.

Recording a symphony (or instrumental bands stage lip??) without all the above intermediate products might actually get some value  from higher bitrates, but it's pure overkill unless your mics can pick it up, and most of our mics drop off quickly after 20khz.

I agree that recording and archiving at 24bit are well worth it.  Someday we may look back on those raw tapes like the high quality SBD reels from early Grateful Dead and be glad we kept them.
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #24 on: January 06, 2009, 11:43:45 PM »
I still find it interesting... we go to a rock show.  Guitarist plays guitar into amp.  Amp is mic'ed with SM57 with 'frequency response' or 40-15,000 hz.  Vocalist sings into SM58 with similar response.  All this goes into mixing board, into stacks, we sit 100' from stacks with "good mics" with response from 20-20,000hz to record the room.  A 44.1k bitrate really will capture all of the above.  Maybe the lady shrieking next to my Earthworks (up to 30,000 hz) will benefit from a higher sample rate recording, but she's not that important to me.  Still, the taper next to me will insist on recording at 96k, even though there is nothing on that wav file higher than 20khz.

                     <snip>


My take exactly.  Just how good does your gear have to be to record 57's and 58's??  Acoustic recordings are a whole other ball game, but recording 57's and 58's out of a pair, or more, or stack, well, there is a lot of gear which would do the job pretty well.  And not all of it is expensive.     8)
Nov schmoz kapop.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #25 on: January 23, 2009, 05:43:55 PM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #26 on: January 23, 2009, 06:49:12 PM »
If what you are saying is valid it would seem the higher the frequency the fewer sampling points and the obviously degraded signal.  Low notes would be accurate, high notes would not.  A 22.05 kHz note would have two "describing points", a 10 Hz note would have 4.41 thousand.  Does that sound right???
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #27 on: January 23, 2009, 08:02:53 PM »
If what you are saying is valid it would seem the higher the frequency the fewer sampling points and the obviously degraded signal.  Low notes would be accurate, high notes would not.  A 22.05 kHz note would have two "describing points", a 10 Hz note would have 4.41 thousand.  Does that sound right???

Yeah. 

Ok, so next question.  The human ear can hear 20 Hz to 20 kHz, for the sake of this, let's say worthwhile sound tops out at 16 kHz.  If you sample at 44.1 kHz, the frequencies near 16 kHz are only going to have ~3 sampled points to represent the waveform right?  But at the same time, lower frequency sounds are going to have >3 points, so they are probably really well sampled.  So does this mean recording at 48 kHz might improve the representation of the waveform for higher frequency sounds?

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #28 on: January 23, 2009, 08:15:05 PM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

Unless something is wrong with your interpolation scheme, both will recreate the 4 KHz signal perfectly (well, perfectly subject to bit-depth constraints).

Edit:  Imperfect analogy:  Does having 3 points define a line any better than 2 points?
« Last Edit: January 23, 2009, 08:57:12 PM by Will_S »

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #29 on: January 24, 2009, 01:16:36 AM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

Unless something is wrong with your interpolation scheme, both will recreate the 4 KHz signal perfectly (well, perfectly subject to bit-depth constraints).

Edit:  Imperfect analogy:  Does having 3 points define a line any better than 2 points?

The answer to your analogy is yes, 3.

My understanding is that there are a few different interpolation functions in a DAC.

http://en.wikipedia.org/wiki/Digital-to-Analog_Converter#DAC_types

So which one is the best?  Which one will give you the best interpolation of those three points that deffine your wave?

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #30 on: January 24, 2009, 09:13:24 AM »
Edit:  Imperfect analogy:  Does having 3 points define a line any better than 2 points?

The answer to your analogy is yes, 3.

You think two points are inadequate to define a straight line (not a line segment, a line)?

Actually I suppose more points might help, if there was noise in the position of the individual points vs the line they describe. Is that what you meant? 

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #31 on: January 24, 2009, 10:13:10 AM »
prof_peabody, your analogy leaves out something very important. Can I have a moment to explain it in a roundabout way? You are asking excellent questions but the method of visual imagery which you (and about a million other people) are trying to get answers from can't possibly do the trick.

Let me suggest that you consider the analog signals going into the digital recorder, the analog signals that come back out in playback, and a third analog signal: the output subtracted from the input--a/k/a the difference or change or error (a/k/a noise and distortion) caused by the recorder's imperfections.

We're trying to identify the inherent limitations of sampled representations of continuous waveforms, so let's make an "assumption for the sake of argument" that the converters in this recorder are as linear and quiet as they can ever be. We want to know which flaws are inherent to the process, and can't be overcome by better converters, better analog circuitry, better flashing lights or generally, by spending more money.

OK. To start with, of course one full cycle of a low-frequency waveform will be sampled more times than one cycle of a high-frequency waveform will be. But it's a mistake to conclude that the low-frequency waveform is getting a better description as a result. The conventional visual analogy breaks down completely on this point, and gives a wrong answer. What tells the truth is the analog "difference" or "error" signal that I mentioned earlier. In a well-implemented digital recorder, the magnitude of the "difference" or "error" signal will be about the same for the high-frequency waveform as it is for the low-frequency waveform.

That isn't a hypothetical statement. Almost everyone in this forum has the equipment and/or software to create "error" signals of this kind. They can be observed and measured by anyone, and indeed have been listened to and looked at and measured for decades. Doing so can be a big "oh!" moment for understanding digital recording. It's the real-world fact, and any mental models that we construct to help ourselves understand the process must take it into account.

So where does the "three data points vs. fifty data points" model go off the tracks? You might see something very interesting (but probably not hear it) if you could remove the anti-aliasing filters from your digital recorder. Nowadays they tend to be an integral part of the D/A converters, but in the early and mid 1980s when digital recording first entered the recording studios, they were physically separate components (though usually on the same circuit board). Those D/A converters really did deliver the stepped waveforms that many people imagine are at the output of a digital audio recorder; then the filters smoothed those waveforms out (though not just by simple linear interpolation as many people seem to imagine).

Still, you could insert a probe in between the two components and look at the D/A converter's direct output. And if you did that, and if you created an "error" or "difference" signal at that point, you'd see that the "difference" or "error" signal on high frequency inputs would have a different frequency spectrum from the corresponding error signal for low frequency inputs. Probably there would also be some difference in magnitude, just because nothing's perfect. But the main energy of both error signals would fall above the "one-half the sampling rate" frequency--and not incidentally, above the range of human hearing. So the anti-aliasing filter would leave you with smooth sine waves at both frequencies, if that's what you'd put in.

--best regards

P.S.: A closely related issue is low-level vs. high-level signals. A full-scale waveform is "described" with the full range of sample values--for a 16-bit system, all 16 bits are used, etc.--while a low-level waveform (say, at -60 dBFS) exercises far fewer bits. The "connect the dots" model predicts that the low-level signal will be less accurately represented, and will therefore have much greater distortion. Yet with any well implemented digital recorder this doesn't occur--a fact which it's not at all difficult to show nowadays.
« Last Edit: January 24, 2009, 10:58:32 AM by DSatz »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #32 on: January 24, 2009, 12:43:32 PM »
So where does the "three data points vs. fifty data points" model go off the tracks? You might see something very interesting (but probably not hear it) if you could remove the anti-aliasing filters from your digital recorder. Nowadays they tend to be an integral part of the D/A converters, but in the early and mid 1980s when digital recording first entered the recording studios, they were physically separate components (though usually on the same circuit board). Those D/A converters really did deliver the stepped waveforms that many people imagine are at the output of a digital audio recorder; then the filters smoothed those waveforms out (though not just by simple linear interpolation as many people seem to imagine).

Still, you could insert a probe in between the two components and look at the D/A converter's direct output. And if you did that, and if you created an "error" or "difference" signal at that point, you'd see that the "difference" or "error" signal on high frequency inputs would have a different frequency spectrum from the corresponding error signal for low frequency inputs. Probably there would also be some difference in magnitude, just because nothing's perfect. But the main energy of both error signals would fall above the "one-half the sampling rate" frequency--and not incidentally, above the range of human hearing. So the anti-aliasing filter would leave you with smooth sine waves at both frequencies, if that's what you'd put in.


So how good are the filters at smoothing?  There are many implementations of these filters right?

The interpolation of the points is not perfect - ie, it doesn't perfectly represent the input signal.  If you sample at a slightly higher rate, you leave yourself less prone to interpolation error.

Playing with sine waves isn't necessarily helpful as well - 3 points to describe one sine wave will have a unique solution...

From a political standpoint - the jury is still out on the sample rate issue.  There are papers both for and against using higher sample rates, and many experts who back each viewpoint.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #33 on: January 24, 2009, 07:57:32 PM »
> The interpolation of the points is not perfect - ie, it doesn't perfectly represent the input signal.  If you sample at a slightly higher rate, you leave yourself less prone to interpolation error.

You're jumping to a completely false conclusion about sampling at higher rates and what you call "interpolation error." If your beliefs were correct, a 5 kHz tone recorded at some relatively normal level with 44.1 kHz sampling would show distinctly greater distortion in playback than the same tone recorded at the same level with (say) 96 kHz sampling.

Similarly if you doubled the frequency to 10 kHz, according to your beliefs the distortion (in a 44.1 kHz system) would increase considerably as compared to what you got at 5 kHz. The closer you came to the upper limit of the system the greater the distortion would have to be, because fewer samples are taken per cycle.

In the real world, however, none of the above generally happens. Hmmm ...

One definite confusion factor is your use of "one cycle of a wave" as your yardstick rather than a set period of time. Given a constant clock speed, of course a lower-frequency waveform will get more samples during the interval that your mind's eye is allotting to it. So you're not using a "level playing field," and that misleads you somewhat.

Apparently it's difficult for most people to visualize what anti-aliasing filters do without oversimplifying them. So I say skip that and go by the quality of the signals that come out of the filters. For audible purposes, who cares what the signal looks like prior to the output filters, as long as a clean enough signal emerges from them.

It would be a mistake to point to the shape of a signal that no one ever hears, and complain that it doesn't look beautiful in your mind's eye.

--best regards
« Last Edit: December 26, 2009, 08:06:32 PM by DSatz »
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #34 on: February 04, 2009, 01:17:29 PM »

The interpolation of the points is not perfect - ie, it doesn't perfectly represent the input signal.  If you sample at a slightly higher rate, you leave yourself less prone to interpolation error.


This seems like a reasonable statement, and perhaps to a point it is.  This is why your A/D converter probably oversamples at 128times as fast as 44.1k.  Then it has the filters and the data to decide the best way to produce a 44.1k result, and it does a great job.  But making an assumption that you should sample at 48k and then try to convert from 48k to 44.1k on a computer without the filters and the 128x oversampling, this is where the assumptions fall apart IMO.  The best approach is simply to record at the rate you want in the end.  Conversions are the real compromise, because there you don't have extra data and RE-interpolation error is a problem.
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #35 on: February 04, 2009, 03:24:33 PM »
> The interpolation of the points is not perfect - ie, it doesn't perfectly represent the input signal.  If you sample at a slightly higher rate, you leave yourself less prone to interpolation error.

Well, nothing in this world is absolutely perfect. But you're jumping to a completely false conclusion about sampling at higher rates and what you call "interpolation error." What an anti-aliasing filter does is critical to the overall result, and deserves to be understood factually (or conversely, maybe not so well understood but accepted, once its actual behavior has been determined).

Since you mentioned politics: It's not that there can be no benefits from higher sampling rates. I know of no one who actually says that--though it's the position which most advocates of higher rates argue against, as if it were commonly held. That kind of "straw man argument" muddies the discussion a lot sometimes.

But please consider: If your assumptions were correct, a 5 kHz tone recorded at some relatively normal level with 44.1 kHz sampling would show distinctly greater distortion in playback than the same tone recorded at the same level with (say) 96 kHz sampling. Similarly, if you doubled the frequency to 10 kHz, then according to your theory, the distortion in a 44.1 kHz system would increase considerably, as compared to what you got at 5 kHz. The closer you come to the upper limit of the system, the greater the distortion would be.

So check it out. In the real world, none of the above stuff usually happens. Hmmm ...

One definite confusion factor is your use of "one cycle of a wave" as your yardstick rather than a set period of time. It's a truism that the lower the frequency, the longer the cycle of its waveform. Given a constant clock speed, of course a lower-frequency waveform will get more samples during the longer interval that your mind's eye is allotting to it. So you're not using a "level playing field," and that misleads you somewhat.

Also it may be impossible for most people to visualize what anti-aliasing filters do without oversimplifying them. Parts of the relevant math are beyond me, so I go by the quality of the signals that come out of the filters. All filters are imperfect, but some imperfections are audibly acceptable while others are not. We can decide by listening, by measurement, or (preferably) both.

Please consider that for audible purposes, it would be OK if a signal looked dreadful prior to the output filters. It could be loaded with crap above the cutoff frequency, as long as a clean enough signal emerges from the outputs. It would be a mistake to point to the shape of a signal that no one ever hears, and complain that it doesn't look beautiful in your mind's eye.

--best regards

A couple points:

- Why are you invoking an anti-alias filter?  It has very little to do with the questions I posed.  You can low pass to 10 kHz (for example) and still go through the logic I presented in my earlier posts.  I consider this an effort to change the topic to avoid answering the question.  Aliasing does not need to be discussed to answer the questions.

- even if one changes the "yardstick" to a set period of time you still encounter the same predicament I presented in the initial post (in fact time and frequency domains are directly related - you can use a Fourier transform to see this for yourself).  As the input frequency approaches the half the recording frequency (or if you want the frequency limit imposed by the anti-alias filter before the ADC), the number of samples that define the wave at a given frequency decreases.  As the number of sample decreases, you rely more heavily on interpolation algorithms to recreate the waveform in a DAC. 

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #36 on: February 04, 2009, 03:29:57 PM »

The interpolation of the points is not perfect - ie, it doesn't perfectly represent the input signal.  If you sample at a slightly higher rate, you leave yourself less prone to interpolation error.


This seems like a reasonable statement, and perhaps to a point it is.  This is why your A/D converter probably oversamples at 128times as fast as 44.1k.  Then it has the filters and the data to decide the best way to produce a 44.1k result, and it does a great job.  But making an assumption that you should sample at 48k and then try to convert from 48k to 44.1k on a computer without the filters and the 128x oversampling, this is where the assumptions fall apart IMO.  The best approach is simply to record at the rate you want in the end.  Conversions are the real compromise, because there you don't have extra data and RE-interpolation error is a problem.

Do you have any information that shows the ADC chip initially samples at ~5600 kHz?  I'm not sure this is the case - that's why I'm asking.

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #37 on: February 04, 2009, 05:12:45 PM »

Do you have any information that shows the ADC chip initially samples at ~5600 kHz?  I'm not sure this is the case - that's why I'm asking.

128x oversampling was considered "state of the art" in 2000, and they played it up big time back then. 5.6 Mhz sample clock = 128x 44.1khz, and 6.1Mhz sample clock will give 128x oversampling at 48k.  I suspect a modern (now obsolete?) Mini-Me or V3 will do this, but I don't know.  Now that a lot of devices will do 96k and 192k, I suspect the sample clock rates are the same and you get 64x and 32x oversampling at those high rates.  Rather than explain that, they just don't brag about it.

Here is a  device which claims 128x oversampling.  Not exactly "state of the art".
http://www.midi-store.com/M-Audio-Flying-Cow-24-bit-D-A-and-A-D-Converter-p-16880.html

Chipset from April 2000 showing 5.6Mhz and 6.1Mhz sample clock.
http://www.asahi-kasei.co.jp/akm/en/product/ak5393/ak5393_f04e.pdf

Others:
http://recforums.prosoundweb.com/index.php/t/14013/0/ (2004 geek discussion)

Here is a story where they are debating in 2002 what we are still debating here today  ;D
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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #38 on: February 04, 2009, 06:26:11 PM »

Do you have any information that shows the ADC chip initially samples at ~5600 kHz?  I'm not sure this is the case - that's why I'm asking.

128x oversampling was considered "state of the art" in 2000, and they played it up big time back then. 5.6 Mhz sample clock = 128x 44.1khz, and 6.1Mhz sample clock will give 128x oversampling at 48k.  I suspect a modern (now obsolete?) Mini-Me or V3 will do this, but I don't know.  Now that a lot of devices will do 96k and 192k, I suspect the sample clock rates are the same and you get 64x and 32x oversampling at those high rates.  Rather than explain that, they just don't brag about it.

Here is a  device which claims 128x oversampling.  Not exactly "state of the art".
http://www.midi-store.com/M-Audio-Flying-Cow-24-bit-D-A-and-A-D-Converter-p-16880.html

So if you read the specs the 128 X oversampling is a sigma delta filter, which means it's probably 128X at a 1 bit resolution.  Nice try...


Chipset from April 2000 showing 5.6Mhz and 6.1Mhz sample clock.
http://www.asahi-kasei.co.jp/akm/en/product/ak5393/ak5393_f04e.pdf


Again - 128X oversampling using a sigma delta filter at 1 bit resolution.


Others:
http://recforums.prosoundweb.com/index.php/t/14013/0/ (2004 geek discussion)

Here is a story where they are debating in 2002 what we are still debating here today  ;D
http://emusician.com/daw/emusic_bridging_gap/

So in all the links you provided the over sampling is achieved through a sigma-delta, 1-bit filter. 

Thanks for playing.

edit: so in essence the oversampling involves doing the ADC to essentially a DSD format and then you convert that signal to 16 bit / 44.1 khz.  You'd have to do a lot of math the work out how much oversampling is really being done...  It's not as much as you think.

edit2: I forgot to mention that using a delta-sigma technique in an ADC usually does great noise shaping and often means you don't need an anti-alias filter
« Last Edit: February 04, 2009, 06:33:35 PM by prof_peabody »

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #39 on: February 04, 2009, 07:58:06 PM »
I'm not qualified to debate what A/D methodology is better than another.  I have a little understanding, but I get lost in the math.  I never meant to imply that they are doing a full 24bit value calculation at 5.6Mhz.  What I did mean is that here is a great tool, designed to do what it does, and I think it's best to let it do it's thing.  I think the original question 3 pages back was "should I record at 44.1, or record at 48K and resample on the computer".  I say, record at 44.1k.

~~~~~~~~ next day ~~~~~~~~
I was curious, so looked this up.  According to the manual, a Korg MR1000 recording 1bit DSD at 5.6mhz will fill 1 GB file in 11 minutes.  A 24/192 file is 13 minutes....  so 1bit at 5.6mhz is something like (HUGE generalization here) something like 24/226khz if such a beast existed.  So at 192khz, there isn't much oversampling.... agreed.

Again, my point is that the A/D has the information to prune that down to 44.1k better than a computer does, starting with 48k or 96k.
« Last Edit: February 05, 2009, 05:26:03 PM by SmokinJoe »
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