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Hydrogenaudio Forum => General Audio => Topic started by: RockFan on 2006-08-30 01:47:13

Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 01:47:13
Hi All,

A thread back in March discussed whether normalization of WAV files is 'lossy', and I remember that no-one thought to point out that operative word is actually 'destructive'.

Digital processing of PCM audio such as normalization, compression (or expansion), equalisation, etc' is 'destructive', not 'lossy', although both mean 'irreversable'. It has to be said, this seems as much a matter of semantics or even philosophy as much as exact engineering terminology.

Anyway, this is of some interest to me, I'm using a 24-bit ADC to record from vinyl (and taking it pretty seriously as an archiving project - I get one shot at some discs, so I want to do it right), and as it happens there is  insufficient gain on the line from the phono-amp I'm using to it to get close to 0dB with some discs, many peak at -6, -8, even -12 dB.

However, even -20 dB or more in 24 bits is greater than 16-bits of resolution. A 24-bit recording which peaks at, say, -16dB, can be 'normalized' to 0dB, and will *still* have information below digital silence in 16-bits, and ideally require dither on conversion.

Obviously I could use a preampifier in addition to the phono-stage to get around the level issues, but I'd rather keep to the 'minimal' signal path I'm using now.

What I'd like to ask people here is; in their experience, is normalization completely 'benign', sonically? Are the algoritms used in different applications much the same, or are some better than others?

R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-08-30 02:22:47
Theres nothing worth recording anywhere near 120dB below peak on a record, so you're not losing anything by normalizing.  There would only be a loss if the dynamic range of the album plus the wasted room at the top were greater then your ADC was capable of recording.

Personally I wouldn't use normalization since it doesn't really get you anything replaygain doesn't do.  So I'd just replaygain the albums and then apply it during playback (if desired).
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Hollunder on 2006-08-30 02:46:15
At least peak normalisation should be more or less the same in every application because it's a rather simple process (I guess). But it's always possible that there are differences. And keep in mind that it is irreversible.

You can find a short description if you search for normalization in the scientific R&D forums here and most likely a lot about it on wiki or in the web out there.

But I would also recommend you using replay gain if possible since you don't need to touch the audio data and it works with perceived loudness, not peak amplitude or rms (well it uses rms afaik, but it's just a part of it). http://wiki.hydrogenaudio.org/index.php?title=Replaygain (http://wiki.hydrogenaudio.org/index.php?title=Replaygain)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 02:56:28
The thing is, I don't want to end up with CDR's that peak at -12dB or lower.

By normalizing at 24-bit resolution, some info is effectively 'pulled' up into the usable dynamic range of CD-audio.

To reiterate; the question is, has anyone noticed the effects of the rounding errors in normalization, and are all nornalization algorithms created equal?

A few years back I tried 'remastering' some late 80's CD's that were typically at -6dB or lower (there seemed to be a lot of paranoia in those days about hitting 0dB) , and I thought (NB) they didn't sound as good as the original files.

Rounding errors in 24-bits will be less of concern, presumably
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-30 03:01:17
Yes, it is completely "benign,"  at least within reasonable limits. Working in floating point gives you the best margin. The amplification will not in any way damage the audio, the only serious consideration is the signal to noise ratio. If your recorded signal is low enough, the noise floor of your system will become an audible part of the finished product.

This is not too likely to be a significant factor with a decent 24 bit soundcard, but at some point, if the amplification of your phono stage is low enough, you will gain quality by using a decent mixer or line level preamp to increase the signal in the analogue domain.

I believe this is a fairly simple calculation as digital transforms go, although I have no idea but that there may be some program(s) that makes a (relative) mess of it.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 03:03:33
I use replay gain for my own listening (with Foobar, all my music is on my PC), but for serious lights down and a spliff listening, it's kernel-streaming and no DSP of any kind.

In the context of straight LP to CD transfers, I don't want to use more digital processing than absolutely necessary, and as you might have gathered I'm even a little suspicious of simply adding gain.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-08-30 03:18:24
The thing is, I don't want to end up with CDR's that peak at -12dB or lower.

By normalizing at 24-bit resolution, some info is effectively 'pulled' up into the usable dynamic range of CD-audio.


Well something is pulled up into the dynamic range of the CD, but its not info.  Probably just hiss, since the dynamic range of the CD is so much greater then the record (even including 10 or 20 dB of waste).

Edit:  I am dumb.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 03:19:49
I guess that's the conundrum - add a pre-amp or add gain digitally.

Probably you're right, and I should really get the analogue side optimized. The only problem is that simple, decent and affordable preamps are few and far between (no market for the little suckers, I guess)- I have thought of building a simple opamp based one myself.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-08-30 03:28:05
I guess that's the conundrum - add a pre-amp or add gain digitally.

Probably you're right, and I should really get the analogue side optimized. The only problem is that simple, decent and affordable preamps are few and far between (no market for the little suckers, I guess)- I have thought of building a simple opamp based one myself.


I think you don't understand me.  I'm saying it makes absolutely no difference since the limit is the source not the hardware
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-30 03:32:10
Quote
I have thought of building a simple opamp based one myself.
good option but take care of old opamp like 741 that have round -40dB noise floor and with the power supply to your pre ampl or you'll get more noise than quality. if you don't choose the right components,better is amplify or normalize in 32bit and after that back to 16bit.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 03:35:02
Well something is pulled up into the dynamic range of the CD, but its not info.  Probably just hiss, since the dynamic range of the CD is so much greater then the record (even including 10 or 20 dB of waste).


Well, the whole point of dithering 16-bit PCM is that sounds like tape-hiss, for example, (say at -70 odd dB) can sound rather gritty and 'unnatural' without it.

It's been argued that analogue noise (whether from tape or circuitry), which is essentially 'brownian', shouldn't be any more intrusive than other 'ambience noise'. I'd rather have that tape hiss quantized with as many bits as possible.

It's generally understood very low-level signals (such as ambience cues) aren't reproduced well by CD-audio near the bottom of it's dynamic range - simply not enough bits..
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-08-30 03:41:53

Well something is pulled up into the dynamic range of the CD, but its not info.  Probably just hiss, since the dynamic range of the CD is so much greater then the record (even including 10 or 20 dB of waste).


Well, the whole point of dithering 16-bit PCM is that sounds like tape-hiss, for example, (say at -70 odd dB) can sound rather gritty and 'unnatural' without it.

It's been argued that analogue noise (whether from tape or circuitry), which is essentially 'brownian', shouldn't be any more intrusive than other 'ambience noise'. I'd rather have that tape hiss quantized with as many bits as possible.



I don't think you're understanding this.  The source media you have has maybe 60 or 70dB of dynamic range.  If you're lucky.  The CD you're going to has roughly 100dB, perhaps more with noiseshaping.  Thats 30-40dB of headroom.  Applying dither here isn't going to help you since the bottom 5-7 bit are already randomly distributed and thus contain no actual information. 

It's generally understood very low-level signals (such as ambience cues) aren't reproduced well by CD-audio near the bottom of it's dynamic range - simply not enough bits..


You're not even close to the bottom.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-30 03:59:19
Quote
In the context of straight LP to CD transfers, I don't want to use more digital processing than absolutely necessary, and as you might have gathered I'm even a little suspicious of simply adding gain.

Having done more than 400 LP transfers I submit that reasonable digital processing is no detriment to audio quality. Of course it depends on what you are looking for, but it is easy to produce a product through editing that is better than what came off the LP.

Reading posts about DEA, it seems as though more than a few people are so concerned about the bits they end up with -- far beyond anything that might make an audible difference -- that one might believe they are building their eternal abode in paradise instead of a mobile music collection. Well, whatever you might think about that attitude, it is very silly to try to apply it to recordings you make from an LP. Record the same disk twice and the totally unavoidable differences between the two recordings will be far more significant than any "damage" you could possibly do by normalizing.

Since peaks too near 0dB can produce waveforms as much as 8dB above 0dB (although that extreme is pretty unlikely in music), some DACs can clip on files that have no clipping. I recommend normalizing to about 97% instead of going to maximum.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 04:16:49
I don't think you're understanding this.  The source media you have has maybe 60 or 70dB of dynamic range.  If you're lucky.  The CD you're going to has roughly 100dB, perhaps more with noiseshaping.  Thats 30-40dB of headroom.  Applying dither here isn't going to help you since the bottom 5-7 bit are already randomly distributed and thus contain no actual information.


Analogue sources have no '0dB' reference to allow direct comparison of 'dynamic range' or 'signal to noise' (which two terms are used, wrongly, interchangably).

However, distortion in the analogue 'domain' increases with level, in the digital domain *exactly* the opposite is the case.

Now, LP. You say 60-70 of 'dynamic range', when in fact you meant 'signal to noise', and that would be about right.

However, since information can be stored, retrieved and heard *well* below that noise floor, the dynamic range is actually considerably greater.

Properly mastered, and played back on commensurately good equipment, LP actually has a dynamic range approaching 90dB, and all of this reproduced with distortion of no more than a few %, and this subjectively 'benign' distortion at that.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 04:30:51
Having done more than 400 LP transfers I submit that reasonable digital processing is no detriment to audio quality. Of course it depends on what you are looking for, but it is easy to produce a product through editing that is better than what came off the LP.

Reading posts about DEA, it seems as though more than a few people are so concerned about the bits they end up with -- far beyond anything that might make an audible difference -- that one might believe they are building their eternal abode in paradise instead of a mobile music collection. Well, whatever you might think about that attitude, it is very silly to try to apply it to recordings you make from an LP. Record the same disk twice and the totally unavoidable differences between the two recordings will be far more significant than any "damage" you could possibly do by normalizing.

Since peaks too near 0dB can produce waveforms as much as 8dB above 0dB (although that extreme is pretty unlikely in music), some DACs can clip on files that have no clipping. I recommend normalizing to about 97% instead of going to maximum.


Amongst others, I will be transferring discs my Dad will be sending over from AUstralia (all classical), many of which date back to the 50's and 60's. It's a one-time thing in the case of these LPs, and I would like to do as good job as I can.

Doing a 'good job' always invloves attention to detail, as I'm sure you're aware.

Anyway, on the subject in hand - normailzation - I mentioned that I tried applying this to some rips of overly 'quiet' 80's CD's, and I wasn't impressed with the results - I went back to turning the volume up.

That was of course 16 bit - I'm hoping that the process is more benign at 24 bits.

I agree that you can 'improve' on the LP itself, in that discreet pops ans clicks can be competely masked, and it would be lost opportunity not do do so. I'm less sanguine about removal of other kinds of noise, though.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-08-30 08:25:14
Quote
Analogue sources have no '0dB' reference to allow direct comparison of 'dynamic range' or 'signal to noise' (which two terms are used, wrongly, interchangably).
This is wrong, actually. Vinyl has always had a "reference" level of 0db, referring to a maximum stylus velocity, which, depending on who you ask, is either 5cm/s or 7cm/s peak. From what I understand, cutting levels are calibrated by how they would reproduce a 1khz sine wave, which would have been at an amplitude of 0db on your tape, relative to that level. (EDIT: Just like tape, this 0db reference can be exceeded, with perhaps greater distortion.)

You tend not to ever think about that 0db reference level if all you're recording is 12" LPs. 12" EPs and singles are often cut at much higher levels (+3 or +6db). Test records can quite commonly hit +15db, and a few hit +18db as a torture test. But if you want to actually get an accurate signal-to-noise result with a turntable system, it's probably a good idea to compare against 0db.

I'm pretty much in the same boat as you are with recording LPs at low levels. Most of my recordings trade at around -15 to -20dbFS peak - which is both worse than yours and perfectly reasonable. It would be foolhardy to boost my gain by 15db, because I do have several tracks that legitimately hit -8 to -10db.

To boot, I have a full-discrete RIAA preamp, and it sucks. I think it's worse than the preamp in my receiver. When I have extra time and pennies I'll put together an OPA637-based amplifier. Nowadays it is quite possible to design an opamp preamp with lower noise and distortion than many discrete designs.

Quote
However, since information can be stored, retrieved and heard *well* below that noise floor, the dynamic range is actually considerably greater.
I made the exact same argument a few months ago and I got shot down to hell for it. As I believe somebody (Garf? Woodinville?) told me, to paraphrase, according to modern psychoacoustic theory, at any frequency with a given level of noise, you're only really going to hear a signal that's at most 3-6db underneath that noise. What this means is that even though a signal "seems" to be buried in the noise and is yet still audible - a signal that peaks at -80dbFS being audible behind pink noise at -70dbFS, for instance - those numbers don't mean anything unless they are broken out by frequency. And when you do that you will find that the signal is usually going to be well above the noise level.

Quote
Properly mastered, and played back on commensurately good equipment, LP actually has a dynamic range approaching 90dB, and all of this reproduced with distortion of no more than a few %, and this subjectively 'benign' distortion at that.
Again, the dynamic range figures are highly frequency dependent. At 10hz I would estimate that you would be extremely lucky to get 40db out. If you test only at 1khz, and go from +15db all the way down to whatever is measurable, then I guess you could get 90db. But that number is going to be wildly different at different frequencies. Personally, when I've looked at spectrum plots of my sound card noise versus blank vinyl grooves, the vinyl/preamp noise dwarfs the internal soundcard noise by a very wide margin at all frequencies.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-30 08:54:36
Unless you used a very poor program, the only undesirable thing you could achieve by normalizing is making the existing noise more noticeable. Turning up the volume in the analogue domain will do exactly the same thing. You should proceed in whatever way makes you happy, but you are laboring under a fantasy if you think there is a difference.

With 16 bit files it is possible to either simply truncate the calculation results or dither them. Dithering significantly reduces distortion, and thus is to be preferred, at least in a conceptual sense. With LPs  however (or cassettes), the background noise level is so high relative to the quantization errors from truncating that there isn't any contest. You will never hear the difference (unless you do a sufficiently large number of transforms; normalization is one transform).

It is easy enough to demonstrate this. You can record into a 32 bit floating point format and do all operations there. The quantization errors will all be very far below the theoretical limits of circuit noise floors -- thus there is no way to possibly hear them. Then you resample to 16 bit (with or without dither is almost certain to be audibly undifferentiated). You can then take the initial recording and immediately convert to 16 bit, then duplicate the operations there. You won't be able to differentiate the two in a blind AB test. You can add to this testing by using 24 bit integer files in any manner you want to manipulate them. The results will be audibly identically.

Clicks and pops are not "masked" by any process I'm familiar with. They are removed, either completely or to a large enough extent that they can no longer be distinguished. In the majority of cases there is nothing left to suggest they were ever there.

Removing many other noises is a process with so many variables that many people seem to never get it right. That doesn't mean that it can not be done very well indeed. While it isn't possible to do blind AB tests, since any audio needing noise reduction will sound different after processing, it is possible to do close tolerance AB comparisons. After years of doing so I am quite convinced that my most general results are that there is no significant change in the desired audio, the noise level is just reduced. In cases of very poor condition LP, the best results may indeed make some audible changes to the signal, but the results are thus better than the beginning condition, no contest.


***

In any practical terms, the dynamic range of LP is the difference between the highest amplitude signal and the lowest that can be differentiate from the intrinsic disk surface noise. While some people have speculated that greater signal depths can exist, actual measurable dynamic range is about 55 to 60 dB; in fact in many cases the recording is compressed to that neighborhood in the mastering process. With some fairly heavy duty noise reduction it might be extended to 70 dB in a transfer from LP. Where have you heard/read of anyone measuring anything approaching 90dB?

Every analogue device (any used in audio, anyway) has a linear range. These are routinely graphed in the specification sheets. A properly designed circuit keeps operation in this range. Distortion does not increase with signal level unless the circuit is over driven.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-30 09:12:13
I guess there is a difference in subject matter. I'm writing about dynamic range of any given recording. Axon, and I presume Rockfan, are writing about the possible dynamic range of the medium. NO?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-08-30 10:36:01
Maybe I can put it this way...

If the noise floor of your pre-amp + ADC is well below the noise floor of the most "silent" parts on a given disc for all audible frequencies, then you are transferring the full dynamic range of that recording (as reproduced by your turntable) into the digital domain.

To determine the noise floor of your pre-amp + ADC, try recording the output of your pre-amp when the stylus is not on the record. You might be surprised - I'm betting that there's already noise well above the 16th bit before you even play the LP! (Though maybe not at all frequencies).


In your application, normalisation is perfectly benign. It's a trivial mathematical operation. That doesn't mean all program do it without error, but they should!

When converting from 24 to 16-bits, you should dither.

I agree with other posters that there may be no audible benefit from working at 24-bits, or from dithering. However, given that you already have the hardware and software to do both, they don't cost anything, and can't hurt.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cliveb on 2006-08-30 12:11:00
...While some people have speculated that greater signal depths can exist, actual measurable dynamic range [of an LP] is about 55 to 60 dB; in fact in many cases the recording is compressed to that neighborhood in the mastering process. With some fairly heavy duty noise reduction it might be extended to 70 dB in a transfer from LP. Where have you heard/read of anyone measuring anything approaching 90dB?

There have been numerous examples of frequency spectrum graphs showing apparently very low noise levels from LPs once you get above about 500Hz. The only link I can find to hand is this article on the Audioholics website (http://www.audioholics.com/techtips/specsformats/LPvsCDformats2.php).

However, the real noise floor is not as low as it appears in these graphs. Rather than repeat the details here, if anyone is interested in why I believe that the conclusions drawn are in error, see this section of my LP-to-CD tips page (http://www.delback.co.uk/lp-cdr.htm#record_resolution).
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-08-30 13:44:55
I'm glad you've critiqued that, Clive.

That's the difference between a simple RMS measurement, and one of any number of possible frequency dependent measurements.


Simple RMS is clear enough to define and measure, but can be misleading.

A strictly defined "spectrum level" is also clear (though surprisingly small for most broadband signals). If something (e.g. noise) is spectrally flat, take the RMS level, go from dB to linear, divide by the bandwidth of the noise, and convert back to dB. That's the spectrum level.

When you FFT something (as Cool Edit does), you're not getting the correct spectrum level. The results depend on the window function, the window length, and the reference level.

If you use a shorter window, then you spread information in the frequency domain - this reduces the apparent (peak) level of sine waves. Conversely, if you use a longer window, then you average energy over time while examining smaller frequency bands - this reduces the apparent level of broadband / noise-like signals relative to sine waves.

If you know what you are doing, you can prove almost anything!

And that's even before you compare RMS measurements with a poor estimate of spectrum level - as you point out, two completely different things!!!


The relevant comparisons are those which compare like with like, and use percentually relevant parameters (time resolution, frequency resolution) for whatever is being measured.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-08-30 14:32:06
What David said. Once you move to spectral measurements you have to move away from RMS, peak or A-weighted noise numbers entirely, and work off of your own measurements, for the most part. They're not at all standardized.

Oh, and I'll throw another curveball at the original question: If you only amplify your signal in ~6db increments - ie, if you repeatedly double your signal until going any further will clip - no quantization error will occur. It will be a 100% lossless process. Of course, you can only get away with this when you can specify your signal amplification as an integral ratio instead of as dB of gain, because it's impossible to specify an exact factor of 2 increase in db.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-30 15:06:03
off topic:
this thread is "bookmarked"!

2Bdecided,Axon,cliveb,AndyH-ha,RockFan,Hollunder,Mike Giacomelli,
you are now in my "white book"(hall of fame).
you all posted magnific explanations in few words,is the HA best thread in my taste.

Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 15:11:33
That's very useful info, Clive, thanks for sharing your experience. And BTW, I agree about the 'demise' of Cooledit.

Re. the adequacy of 16-bit for recording LPs; essentially the question more broadly is whether 16 bits is adequate for music, period, and I would agree that it is.

However, if any DSP is going to be carried out, IMO 16 bits is really NOT adequate. This kinds of relates to my question about simple normalization.

An example I vividly remember was applying normalization to an old copy of "School's Out", which was absurdly quiet. At first I was quite happy with the results, but something wasn't right (to be suitably vague), and I ended up binning it. Possibly flaws in the infamously dodgy A/D conversion often used in in the early days of CD had been emphasized.

You mention the need for 'hard measurement', and while no-one would argue with that, one has to recognize the fact that the 'art' (!) of music reproduction is not a fait-accomplis. Music is easily the most complex and cerebral use we make of our auditory system, and some 'parameters' have yet to be nailed down (IMO). Even the best audio engineers/designers in the world will listen to their designs before deciding whether they're any good at playing music.

PS >> my own 'standards' for transferring vinyl are similar to yours - I'm not a "audiophile extremist" but as I've repeatedly said, I want to do it right - I regard it as 'archiving.' I'm using a completely overhauled, DC-driven Townshend Rock MkII (with a cassette-tape belt, believe it or not), heavily-modded/rewired/screened  Rega arm and Garrott-tipped Decca MkIV, a Gram Amp 2SE and the aforementioned stand-alone 24-bit ADC, the latter both battery-powered.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cliveb on 2006-08-30 15:47:33
However if any DSP is going to be carried out, IMO 16 bits is really NOT adequate. This kinds of relates to my question about simple normalization.

In the context of LP transfer, I am not persuaded of the need for working at greater than 16 bits. When you consider that the resolution of an LP is about 10 or 11 bits (12 on a good day with a following wind), then you'd need to do a heck of a lot of DSP for the rounding errors to accumulate sufficiently to bring the 16 bit quantisation noise up above the vinyl noise floor.

And the point here is that, IMHO, LP transfers don't need much DSP at all. The vast majority of the editing I do concerns the removal of impulse noise, which involves edits to small isolated sections of waveform. The changes made to the waveform at those locations vastly swamps any changes to the quantisation noise that may result.

The only "global DSP" I do to LP recordings is normalisation (nearly always), some modest EQ (very rarely), and broadband noise reduction (sometimes, and only in moderation). Apart from normalisation, these other operations again make a much bigger change to the audible nature of the music than the minor change in quantisation noise they cause (which I still maintain will remain beneath the vinyl noise floor).
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 16:16:51
And the point here is that, IMHO, LP transfers don't need much DSP at all. The vast majority of the editing I do concerns the removal of impulse noise, which involves edits to small isolated sections of waveform. The changes made to the waveform at those locations vastly swamps any changes to the quantisation noise that may result.

The only "global DSP" I do to LP recordings is normalisation (nearly always), some modest EQ (very rarely), and broadband noise reduction (sometimes, and only in moderation). Apart from normalisation, these other operations again make a much bigger change to the audible nature of the music than the minor change in quantisation noise they cause (which I still maintain will remain beneath the vinyl noise floor).


I have to disagree - even the DSPs you list should be done at higher than 16-bit.

If one is simply recording to PCM and perhaps remove (!) a few clicks, then straight to 16-bit is fine.

But even normalization involves multiplying each sample by an arbitrary integer without regard to their values relative to each other, and rounding errors will certainly effect the resulting waveform and it's sound - I've heard this definitively. These tiny changes in the relative values of the samples might seem inconsequential in theory, but the ear/brain is a very sensitive 'device'.

Further, it's the very low level (ambience and such like) signals which are brought into prominence, and which in the original waveform are quantized with too few bits to sound realistic.

It's worth bearing in mind that when 16/44.1 PCM was 'invented', there was no such thing as DSP at all (even if it was envisaged), it didn't appear until well into the late 80's. The universal adoption of 24-bit resolution (and much higher sampling rates) was mainly to faciltate them.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-08-30 17:10:08
But even normalization involves multiplying each sample by an arbitrary integer without regard to their values relative to each other, and rounding errors will certainly effect the resulting waveform and it's sound - I've heard this definitively. These tiny changes in the relative values of the samples might seem inconsequential in theory, but the ear/brain is a very sensitive 'device'.


It depends what you mean by "working at more than 16-bits".

If you multiply two numbers together, you will normally generate more bits (more decimal places, if you like). Programs like Cool Edit Pro don't just throw these bits away or round them off - they dither back down to 16-bits by default.

You can convert to floating point, perform the operation, and dither back to 16-bits if you want - but that's pretty much what Cool Edit Pro is doing anyway when "working at 16-bits".

Further, it's the very low level (ambience and such like) signals which are brought into prominence, and which in the original waveform are quantized with too few bits to sound realistic.

It's got nothing to do with "sounding realistic".

With correct dither, it's about noise - pure and simple.

If can be just four bits - it will sound perfectly "realistic", but with bucket loads of noise on top!

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 17:41:28
It depends what you mean by "working at more than 16-bits".

You can convert to floating point, perform the operation, and dither back to 16-bits if you want - but that's pretty much what Cool Edit Pro is doing anyway when "working at 16-bits".

Converting to 24 or 32-float might help, I'm not sure. I would definitely rather work with a 'native' 24 bit recording (which is what I'm doing)
It's got nothing to do with "sounding realistic".

With correct dither, it's about noise - pure and simple.

If can be just four bits - it will sound perfectly "realistic", but with bucket loads of noise on top!


It has everything to do with 'realism'. 16/44  typically introduces several percent quantization distortion below -70dB (typically, played back through a good, linear DAC it reaches 6-10 % by -80dB, and more distortion than signal by -100dB).

Quantization (randomly distributed) distortion is completely unlike 'harmonic' distortion produced in the analogue domain, which can reach 10% or more and still sound like 'colouration' - i.e benign.

Were one to inflict these levels of quantization-distortion (say, 5%) across the entire dynaimic range of a music recording, the result wouldn't just sound bad, it would be unlistenable, quite literally.

R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-08-30 18:07:02
Just to put some hard numbers down on the signal-to-noise ratio of vinyl vs CD.

I took a recording of the Hi Fi News test record, with one of its +15db, 300hz tracks. (It was one of Andy's, actually.) I made a large (5 million point) FFT amplitude spectrum plot, and compared the 300hz peak amplitude with an eyeballed average noise amplitude around the peak. It comes out to be 80-84db. Now, you could probably eke another 3db out of this if your cart was able to track a +18db tone. Very few can. And this SNR number is compromised quite a bit by the speed variation of the table, which spreads the power of the test tone out by quite a bit. You might get 10-20db back in your SNR if you manage to factor that back in. Oh, and there's more background noise at 300hz due to rumble and hiss and what not, and the observed background noise at around 10khz is lower by about 16db, so let's just pretend that we're looking at the 10khz noise floor in this recording, rather than the 300hz noise floor. So for the sake of argument, let's call the measured vinyl dynamic range to be 123db. (84 db measured + 3db from not testing the complete headroom of vinyl + 20db expected from eliminating speed variation issues + 16db for a more ideal noise floor measurement)

Note that most people would call that a very, very generous figure.

In comparison: I made a wav file in Audacity composed of a 300hz tone of amplitude 1.0 I exported it to a 16 bit 44.1 wav with noise-shaped dither.I did another large FFT of the result. The peak-to-average ratio came out to.. drumroll... 151db. No, that is not a typo. A 16-bit WAV file is perfectly capable of 151db of dynamic range.

Conclusion: 16-bit recording has a noise density that is at least 28db lower than vinyl.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cliveb on 2006-08-30 18:16:40
But even normalization involves multiplying each sample by an arbitrary integer without regard to their values relative to each other, and rounding errors will certainly effect the resulting waveform and it's sound - I've heard this definitively. These tiny changes in the relative values of the samples might seem inconsequential in theory, but the ear/brain is a very sensitive 'device'.

So, just to make sure I understand your position on this....

Suppose we have a 24 bit recording of a vinyl LP. Consider two possible normalisation methods:
1. Normalise at 24 bit resolution, then dither down to 16 bit for playback.
2. Dither down to 16 bit, then normalise at 16 bit resolution.
I take it you maintain there will be an audible difference between the two. Have I understood your position correctly?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 18:41:34
So, just to make sure I understand your position on this....

Suppose we have a 24 bit recording of a vinyl LP. Consider two possible normalisation methods:
1. Normalise at 24 bit resolution, then dither down to 16 bit for playback.
2. Dither down to 16 bit, then normalise at 16 bit resolution.
I take it you maintain there will be an audible difference between the two. Have I understood your position correctly?


Yes. To be exact, the normalization would be done at 32-float.

There will definitely be a clear mathematical difference between files created with the two different methods, and yes, an audible one. To what degree depends how much normalization is applied. With my particlular setup this can be +16dB or more.

Converting first needlessly discards useful resolution, and normalizing in 16-bits is far less precise than in 24.

So, carry out the more complex operation first (sample multiplcation) in high-resolution (much bigger numbers for the machine to work with), then the simpler process (truncation + dither).

Whatever, I will not apply any DSPs in 16-bit if it can possibly avoided.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-30 18:46:03
There will definitely be a clear mathematical difference between files created with the two different methods, and yes, an audible one.

OK, ENOUGH! 

You're violating TOS #8 (http://www.hydrogenaudio.org/forums/index.php?showtopic=3974#entry149481).
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-30 18:48:49
Quote
2. Dither down to 16 bit, then normalise at 16 bit resolution

please,don't do that,read the post #208 here: http://www.hydrogenaudio.org/forums/index....134&st=200# (http://www.hydrogenaudio.org/forums/index.php?showtopic=40134&st=200#)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-30 18:53:29
Quote
2. Dither down to 16 bit, then normalise at 16 bit resolution

please,don't do that,read the post #208 here: http://www.hydrogenaudio.org/forums/index....134&st=200# (http://www.hydrogenaudio.org/forums/index.php?showtopic=40134&st=200#)

Ok, before we start listening with our eyes , there is the "Dither Transform Results (increases dynamic range)" setting.  Was/is it checked?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 18:54:02
There will definitely be a clear mathematical difference between files created with the two different methods, and yes, an audible one.

OK, ENOUGH! 

You're violating TOS #8 (http://www.hydrogenaudio.org/forums/index.php?showtopic=3974#entry149481).


oooOOOoo!

edit >> well having read the TOS in question, I don't know what you mean.

Or you don't.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-08-30 18:59:41
Enough is right! This is getting too pedantic and argumentative.

If RockFan is right, then the quantization noise from the operations would be audible - with a blind test. And I'll give the benefit of the doubt here, because it is, perhaps, quite reasonable to listen to classical music that is highly amplified - 60db or more - to catch the end of some ambience or instrument resonance as it fades to the background noise. I've done it before. guru used a similar use case to poke holes in MPC's transparency. It's unlikely, but not completely outside the realm of possiblity, to catch the quantization error in such a situation in an ABX test.

What I'd say is to take a very quiet music selection, amplify it to full scale at both 24-bit and 16-bit resolution, then quantize the 24-bit one to 16-bit. I would also strongly suggest highpassing at 40-50hz, at 24-bit resolution, to knock out the rumble in order to get higher gain during the normalization. (Or just crank your volume up really loud.) Then ABX the 24-bit-processed stuff to the 16-bit-processed stuff for 32 trials.

I would offer to prepare the samples myself from my own classical LPs but I'm going on vacation for a week starting this afternoon.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 19:19:00
Enough is right! This is getting too pedantic and argumentative.

If RockFan is right, then the quantization noise from the operations would be audible - with a blind test. And I'll give the benefit of the doubt here, because it is, perhaps, quite reasonable to listen to classical music that is highly amplified - 60db or more - to catch the end of some ambience or instrument resonance as it fades to the background noise. I've done it before. guru used a similar use case to poke holes in MPC's transparency. It's unlikely, but not completely outside the realm of possiblity, to catch the quantization error in such a situation in an ABX test.

What I'd say is to take a very quiet music selection, amplify it to full scale at both 24-bit and 16-bit resolution, then quantize the 24-bit one to 16-bit. I would also strongly suggest highpassing at 40-50hz, at 24-bit resolution, to knock out the rumble in order to get higher gain during the normalization. (Or just crank your volume up really loud.) Then ABX the 24-bit-processed stuff to the 16-bit-processed stuff for 32 trials.

I would offer to prepare the samples myself from my own classical LPs but I'm going on vacation for a week starting this afternoon.

A perfectly reasonable proposal, if one really feels that these issues need to be 'proved' via ABX.

But I don't really understand anyone taking issue with my stating that;

1) if substantial normalization is needed with an 'under-recorded' 24-bit file, carrying it out at 24-bits will result in better use of the resolution of eventual 16-bit file.

and;

2) that any and all DSPs (including normalisation) are better applied while in 24 bits.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-30 19:19:17
Quote
Ok, before we start listening with our eyes
use your eyes again(to read now) and look this detail from my post:
Quote
...i can host the sample 44.1k-16bit used for test or you can use some other extracted from cda,trust me,you will hear the noise.


Quote
there is the "Dither Transform Results (increases dynamic range)" setting. Was/is it checked?
...yes...whant one screenshot too?
do you get better results if uncheck it?

regards
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-30 19:20:57
I'll leave you audiophiles alone now.

Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-30 19:25:02
I'll leave you audiophiles alone now.


sorry,poor english here.i don't understood 
but i'm sure you're kiddin!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cliveb on 2006-08-30 19:38:26
A perfectly reasonable proposal, if one really feels that these issues need to be 'proved' via ABX.

I was going to propose just such an experiment once RockFan had confirmed that I had correctly understood his position, but Axon beat me to it. Does anyone want to go to the trouble of an ABX test? It's really not that important in the grand scheme of things. And as RockFan says....

But I don't really understand anyone taking issue with my stating that;
1) if substantial normalization is needed with an 'under-recorded' 24-bit file, carrying it out at 24-bits will result in better use of the resolution of eventual 16-bit file.
and;
2) that any and all DSPs (including normalisation) are better applied while in 24 bits.

I certainly don't take issue with that. Obviously doing it at 24 bit cannot be worse than 16 bit, so you've nothing to lose.

All I ever questioned was whether there is an audible difference in the context of doing LP transfers. You believe there is, I don't. We can both be happy with our positions. And when you're talking about LP transfers, all this stuff about quantisation noise is like arguing about angels on the head of a pin compared to the *really* important stuff, like getting the LP properly clean, using a good turntable and phono preamp, etc.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 20:37:40
All I ever questioned was whether there is an audible difference in the context of doing LP transfers. You believe there is, I don't. We can both be happy with our positions. And when you're talking about LP transfers, all this stuff about quantisation noise is like arguing about angels on the head of a pin compared to the *really* important stuff, like getting the LP properly clean, using a good turntable and phono preamp, etc.


I don't think we're really disagreeing fundamentally here, except perhaps on the overall 'wisdom' of using DSP's on 16/44 PCM.

Possibly whats's been lost sight of is that I need to use *substantial* amounts of digital gain on some recordings due to the level-matchng issues I described. There is simply no reason for me to throw out all those bits before I do it.

If I feel the inclination, I might try a few DSP's such as HF boost to a recording sometime, in 24 and 16 bits, and see if the final 16/44 files (or CDRs) can be ABX'd.

Generally speaking, my motto is "if it's worth doing, it's worth overdoing".
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cliveb on 2006-08-30 20:52:00
Possibly whats's been lost sight of is that I need to use *substantial* amounts of digital gain on some recordings due to the level-matchng issues I described. There is simply no reason for me to throw out all those bits before I do it.

No argument with that. Indeed, if you're suffering from very low recording levels, then it's a good thing you *are* recording at 24 bit. For my part, since I am able to achieve decent recording levels, and since my so-called 24 bit soundcard (an M-Audio AP2496) has a noise floor that makes it effectively an 18/19 bit card, I am happy to continue working at 16 bit.

If I feel the inclination, I might try a few DSP's such as HF boost to a recording sometime, in 24 and 16 bits, and see if the final 16/44 files (or CDRs) can be ABX'd.

FWIW, I did just try an experiment. Recorded at a deliberately low peak level (-18dB) at 24 bit resolution (although of course the bottom 5 or 6 bits on the AP2496 are pretty much random). Then prepared two files:
1. Normalised at 24 bit, then dithered to 16.
2. Dithered to 16, then normalised.
Loaded them up into Foobar2000 and ABX'd (or rather, failed to ABX) them.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-30 22:28:18
I have to say, as something of a vinyl die-hard, I've always viewed the 'dynamic range' attributed to CD to be very generous.

Typical distortion through a top-notch DAC is something like this;

-60dB - 0.22%
-70dB - 3%
-80dB - 8%
-90dB - 30+%
-100dB - distortion = signal

The nature of this distortion is little discussed, but it needs to be understood that it is not like the evenly distributed harmonic distortion (typically mostly 2nd, some 3rd a little 4th and so on) that predominates in the analogue domain - it is randomly distributed 'quantization noise' and is extremely obnoxious at levels over a fraction of a percent.

It passes 'unnoticed' because it is only affecting low level stuff like ambience and reverb.

If the entire signal were afflicted with this distortion at approaching 1% it would be rather unpleasent to listen to. At anything over 3% it would be unlistenable.

I am compelled to wonder, then,  how it came to be that signals distorted to the the extent they are below -70dB are included in CD's 'dynamic range' numbers. More properly, they are actually it's 'signal to noise'.

Conversely, vinyl disc is damned for it's 'signal to noise' of <70dB, when in truth it has a 'dynamic range' considerably better than this.

Which is the better music carrier? You decide!

edit >> we could also discuss CD's notional 'bandwidth' or 'time-domain resolution'. Perhaps not.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-31 06:36:26
Since the forum has apparently been offline much of the day, some of what I wrote this morning may be a bit outdated but I think not too much, so here it is. As far as the latest posts go, from where come these amazingly high distortion numbers? They seem exceedingly unlikely and very inconsistent with measured data.

I want to point out a few aspects of the frequency analysis graph vs RMS measurements, just in case they are not clear to everyone. The graphs are misleading only if one doesn't know what they mean. The frequency spectrum is divided into some number of ranges, the frequency windows. The total noise is the sum of noise (or audio signal, depending on what one is measuring) in all those windows. As one changes the FFT for the graph, the graph trace moves up or down accordingly. Increase the FFT, thus the number of divisions being considered, and the dB reading at any given frequency is now lower; decrease the FFT and the reading is higher. If one adds up all those window values he will arrive at the same numbers, giving the same results, as the RMS calculations.

***
I can't know what any particular individual hears but I do know what  has been demonstrated many times under controlled conditions: people often hear what they believe they will hear, in spite of the fact that what they believe they hear isn't even present.

Many people prefer to go on believing their illusions, but illusions are what they often are. I've gone through the experience myself enough times to know this isn't simply nay saying about human abilities. Much the same applies to vision and probably to the other senses.

In this thread
http://www.audiomastersforum.net/amforum/v...opic.php?t=5455 (http://www.audiomastersforum.net/amforum/viewtopic.php?t=5455)
I made arguments that dithering was often irrelevant, even when operating at 16 bits. Dither really only effects very low level signals (in any way that can be perceived). I got a lot of strongly worded counter arguments saying I was wrong, that not dithering transforms produced audibly bad results.

To verify my belief to myself I extracted a reasonably dynamic DDD track from a commercial CD. It contains a fade to a very low level. Its peak value is a little under -4dB. I applied the following three transforms in CoolEdit (at 16 bit), once with dithering transforms engaged, then again with no dithering.

(1) normalize to 100% -- should effect all samples
(2) hard limit to -3dB, Boost Input by 0dB -- should effect only a minority of samples
(3) normalize to my usual 97% -- should effect all samples

The audible results between the two seemed subtle but present. Hardly what I would expect from other people's arguments, but still there -- or so it seemed. Loading the samples I was comparing into WinABX showed me that I could not in fact distinguish between the two treatments.

I expected this for most of the track but I understand the theoretical benefits of dithering at very low levels. I though I might be able to hear dither vs no dither on the fade out, if nowhere else. More ‘objective' comparisons showed otherwise.

I received some half -hearted excuses as to perhaps why not. I issued a challenge for anyone to point out any music samples on which I could repeat the experiment and detect a difference in the results. No one seemed to be able to come up with anything.

So I'll issue the same challenge here.  If you think there is a difference that can stand up to testing, show me. I'm reasonably sure my equipment is capable. Maybe I'm not. Produce results that show you are.

Making level adjustments in the analogue domain, to match a pre-normalized track with a normalized one, is confounding and difficult. Therefore I propose the following alternative. Normalize at 16 bits. You say it makes an audible difference, I say it does not make a difference relative to using an analogue volume control (i.e. no difference that can be heard, only a difference that can be calculated).

Now amplify by the necessary negative amount to bring the level back to the original value. You thus have a track that can be legitimately compared to the original in WinABX, PCABX, or any other ABX testing program. You have done two transforms at 16 bits, thus increasing the possibility of detecting the deterioration you believe you can hear. You can dither the transforms or not, whichever you think increases your probability of success.

I know results have been demonstrated with pure tones near the 16 bit lower limit. Here we are considering real recorded music.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-08-31 13:11:45

It's got nothing to do with "sounding realistic".

With correct dither, it's about noise - pure and simple.

If can be just four bits - it will sound perfectly "realistic", but with bucket loads of noise on top!


It has everything to do with 'realism'. 16/44  typically introduces several percent quantization distortion below -70dB (typically, played back through a good, linear DAC it reaches 6-10 % by -80dB, and more distortion than signal by -100dB).


I have to say, as something of a vinyl die-hard, I've always viewed the 'dynamic range' attributed to CD to be very generous.

Typical distortion through a top-notch DAC is something like this;

-60dB - 0.22%
-70dB - 3%
-80dB - 8%
-90dB - 30+%
-100dB - distortion = signal



It doesn't matter how many times you say this, or how many times some idiot audiofools say it on other boards, it doesn't make it true!

The truth is very simple...

Correct dither prevents quantisation distortion and replaces it with benign "uncorrelated" noise, below which the original signal is infinitely resolvable (as far as the noise can be cancelled / averaged away / "heard through")

From your quotes, I don't think you understand dither. There are some fairly bad explanations on the web, but the principle is simple: add "correct" random noise before the rounding or truncation stage. This makes the truncation a stochastic, rather than deterministic process. This means that, rather than always being rounded to the nearest 16-bit value (or truncated to the one below), a given sample value could be rounded up or down depending on the amplitude of the noise at that instant - and with "correct" noise, the probability of being rounded up vs the probability of being rounded down is directly proportional to the amplitude of the original sample value between the two nearest possible 16-bit sample values.

To put it simply in decimal, an original value of 2.9 has ten times more chance of being rounded to 3.0 that to 2.0, but it can go either way. Without dither, it would always end up as 3.0, and at the limit a sine wave always looks like a square wave. With dither, a sine wave looks like a weird noisy square-ish wave, but sounds like a sine wave plus "uncorrelated" noise. That is because, to have any appreciate of "frequency" (never mind the actual psychoacoustics of how human ears work) you have to look across time. Looking at the dithered output across time, it is a sine wave plus noise. You started with a sine wave, you added noise, this gave you a sine wave plus noise. Then you truncated, but you still have a sine wave plus noise!

(To the really smart people: I know I'm just explaining rectangular dither, while triangular is optimal. To the really really smart people: I know the noise isn't genuinely uncorrelated, but its uncorrelated to its second moment with triangular dither, which Lipshitz and Vanderkooy seem to regard as sufficient - who am I to argue?)

Quote
The nature of this distortion is little discussed, but it needs to be understood that it is not like the evenly distributed harmonic distortion (typically mostly 2nd, some 3rd a little 4th and so on) that predominates in the analogue domain - it is randomly distributed 'quantization noise' and is extremely obnoxious at levels over a fraction of a percent.


Without dither, quantisation noise is typically largely harmonic, with the caveat that, being generated in the sampled domain, it will alias above fs/2.

Here are some pictures from Cool Edit Pro.

[attachment=2493:attachment]

Without dither, the -90dB FS sine wave has 50% harmonic distortion at 16-bits.

With dither, the harmonic distortion is absent, and there is broadband noise instead.

With noise shaped dither, the level of this noise in the most sensitive region of hearing is lowered, thus increasing the perceived signal to noise ratio.


My claim about 4-bit being distortion-free wasn't an idle boast. I've tried it.

I've even put a 6-bit example on line here...

http://mp3decoders.mp3-tech.org/24bit2.html (http://mp3decoders.mp3-tech.org/24bit2.html)

Scroll down to "To dither, or not to dither?" and have a listen.

What do you think?

(the noise doesn't sound quite right because I mp3 encoded the result to post it to the web - you can try the experiment yourself and listen to the pure linear PCM output using Cool Edit Pro/Audition)


Quote
edit >> we could also discuss CD's notional 'bandwidth' or 'time-domain resolution'. Perhaps not.


We have done so many times before. The threads are in the FAQ.

I won't say any more (like scream TOS 8) because I'm trying to be constructive, but I'm surprised we haven't had a moderator in here!

I hope this post is helpful.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-31 15:37:18
Quote
What do you think?

http://www.hydrogenaudio.org/forums/index....ost&id=2493 (http://www.hydrogenaudio.org/forums/index.php?act=Attach&type=post&id=2493)

that dither is one "spray of white noises" over the whole audio and is audible,even giving a general better result in the audio.
this benefit is good for people that can't here this noise because we
Quote
can't know what any particular individual hears

the link posted by AndyH-ha from "some idiot audiofools say it on other boards"(as you wrote)is showing that clearly.

anyone still have doubts? (http://img428.imageshack.us/img428/2869/whererm6.gif)


regards!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-31 16:34:43

Typical distortion through a top-notch DAC is something like this;

-60dB - 0.22%
-70dB - 3%
-80dB - 8%
-90dB - 30+%
-100dB - distortion = signal


It doesn't matter how many times you say this, or how many times some idiot audiofools say it on other boards, it doesn't make it true!


What? Those numbers are typical of measured results from a good DAC.  You do understand that? measurements at the output of a DAC, not digital-domain analysis?

You seem to be one these people that take exception to suggestions that there are any shortcomings in digital audio (and CD in particular) at all, that it's anything but flawless (just like Phillips said it was back in '83), even advocacy of analogue and vinyl disc, to the point of ready and childish name-calling like "audiofool".

If you're happy with CD, I'm not going to take it way from you (although Sony/Phillips might), unlike what befell those who wished to continue buying LPs 15 or so years ago.

The problem is, you see, a not inconsiderable number of people actually would like to be able to choose which format they buy their music on, including vinyl disc, and the obduracy of views like yours don't help very much.

Analogue does have particular virtues (along with it's intractable flaws, of course) in it's reproduction of music and CD is far from "perfect" , however often and loudly militant digiphiles insist otherwise.

And BTW matey, speaking of TOS, you'll probably find the use of insults, even if you think you're being clever and using them obliquely, is also a breech of them.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-31 16:43:35
And BTW matey, you'll probably find the use of insults, even if you think you're being clever and using them obliquely, is in breech of the TOS.

Then I guess you two are even.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-31 16:47:23
Quote
I'll leave you audiophiles alone now.

you break your promisse.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-31 16:49:25
Quote
I'll leave you audiophiles alone now.

you break your promisse.


Couldn't resist.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-08-31 17:18:19


Typical distortion through a top-notch DAC is something like this;

-60dB - 0.22%
-70dB - 3%
-80dB - 8%
-90dB - 30+%
-100dB - distortion = signal


It doesn't matter how many times you say this, or how many times some idiot audiofools say it on other boards, it doesn't make it true!


What? Those numbers are typical of measured results from a good DAC.  You do understand that? measurements at the output of a DAC, not digital-domain analysis?


You didn't use that faulty test CD did you? (IIRC it's the Hi-Fi news one, but I could be wrong about that; it is one CD from someone you would expect to know better) This disc has not dither, so it has hideous amounts of distortion present in the digital data itself! Even some Hi-Fi magazines have used that one.

I know how to generate test signals, I knew (but have now probably forgotten) how to use an audio precision, and I know the difference between correlated distortion and uncorrelated noise. So I also know that there are plenty of DACs with barely measurable distortion at the 16-bit level. Even a couple of half decent sound cards connected together can prove this.

These links are vaguely relevant...

http://www.dcsltd.co.uk/technical_papers/bits.pdf (http://www.dcsltd.co.uk/technical_papers/bits.pdf)
http://www.dcsltd.co.uk/dcs_elgar_plus.html (http://www.dcsltd.co.uk/dcs_elgar_plus.html)


Quote
You seem to be one these people that take exception to suggestions that there are any shortcomings in digital audio (and CD in particular), that it's anything but flawless (just like Phillips siai it was back in '83), or even the advocacy of analogue and vinyl disc, to the point of ready and childish name-calling like "audiofool".

If you're happy with CD, I'm not going to take it way from you (although Sony/Phillips might), unlike what befell those who wished to carry on buying LPs 15 or so years ago.


You sound very bitter. This isn't a vinyl versus CD debate. (I have thousands of both - I have even more 78s). This is actually about simple mathematics. It's not even science (where hypotheses can be disproven after years of being accepted) - it's maths. Once proven, it stays proven. Dither has been around for decades. The best papers wrt audio are from Lipshitz, Vanderkooy and Wannamaker, from ~1984-1992. You'll note that it's all maths.


Quote
The problem is, you see, a not inconsiderable number of people actually would like to be able to choose which format they buy their music on, including vinyl disc, and the obduracy of views like yours don't help very much.


What relevance does this diatribe have to whether 16-bit audio is, in reality, distorted or not due to the 16-bit word length limit?

FWIW I could find many more things to complain about in the record industry than the decline of vinyl records!


Quote
Analogue does have particular virtues (along with it's intractable flaws, of course) in it's reproduction of music and CD is far from "perfect" , however often and loudly militant digiphiles insist otherwise.


I'm on record in this forum as saying I suspect CD isn't good enough.
http://www.hydrogenaudio.org/forums/index....=9311&st=50 (http://www.hydrogenaudio.org/forums/index.php?showtopic=9311&st=50)

That, yet again, is irrelevant to the present discussion. I'm disputing your claim that low level (e.g. -90dB) signals in a 16-bit signal are hideously distorted. They are not. They are very noisy (just as they are on, say, analogue tape), but there is no intrinsic distortion - and there are plenty of DACs which will reproduce them faithfully in this respect.

Quote
And BTW matey, you'll probably find the use of insults, even if you think you're being clever and using them obliquely, is in breech of the TOS.


Obliquely? I haven't insulted you at all.

I don't think calling people in other places, who refuse to understand dither and nyquist "audiofools" is against the TOS on HA. In fact, I think it's an implicit requirement of HA to hold this view, or come around to this view.


What's sad is that my previous post was a genuine attempt to explain dither to you. Not to knock vinyl, or support record companies, or declare my love for Sony and Philips - just to explain why your original question had a specific answer. I didn't run Cool Edit Pro, generate a signal, process it, capture various different spectrograms, stitch them together in PhotoPaint, and upload them to HA to insult you or anyone. I did it because a quick Google suggested that the best explanation of dither was now a "404 page", and the others were either written by people who didn't really understand it, or without the pictures I think explain it the best.


This topic (and nyquist) has been done to death. Please don't feel insulted, but let me warn you of this: the people who spend the most time trying to deny the basics of digital audio are the ones who feel the most silly when they realise they were wrong.


If or why CD isn't perfect is an interesting discussion. 30% harmonic distortion at -90dB is not there is reality, and so is not a possible reason!


Whether you believe me or not, the ABX test that's been suggested may at least put your mind at rest for your own purposes.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: GeSomeone on 2006-08-31 17:34:05
If you only amplify your signal in ~6db increments - ie, if you repeatedly double your signal [..] Of course, you can only get away with this when you can specify your signal amplification as an integral ratio instead of as dB of gain, because it's impossible to specify an exact factor of 2 increase in db.

I was thinking 6 db was equal to a factor of 2. It isn't?

(yes, sorry a bit off topic)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-08-31 17:50:06
I was thinking 6 db was equal to a factor of 2. It isn't?

It isn't.

A factor of 2 is roughly 6.02dB
6dB is roughly a factor of 1.995

dB = 20 * log10(factor)
factor = 10^(dB/20)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-08-31 18:16:16
It has everything to do with 'realism'. 16/44  typically introduces several percent quantization distortion below -70dB (typically, played back through a good, linear DAC it reaches 6-10 % by -80dB, and more distortion than signal by -100dB).

Hold on a sec. Are you saying that PCM itself is flawed as in it can't faithfully represent a signal without adding nonlinear distortions (during quantization) just because you measured the output of some DACs with god knows what kind of possibly flawed digital signal?

If not, we probably use different terms to name things. "Distortion" has a pretty broad meaning. I'm assuming you mean non-linear distortions. Of course, quantization alters a signal and introduces errors. If you don't do it correctly it's possible that there'll be non-linear distortions, too. However, if you DO it correctly (via dithering) the (digital) system is provably "linear" as in "quantized_output = input + noise" where the noise is statistically independant (up to a any moment you wish) from the input (!!!!!) and is possibly shaped instead of being white (to get almost any PSD of the noise floor you desire).

It's one thing to be not satisfied with current DACs or your current test signal.
It's another thing to deny mathematical facts.

Quantization (randomly distributed) distortion is completely unlike 'harmonic' distortion produced in the analogue domain, which can reach 10% or more and still sound like 'colouration' - i.e benign.

Huh?

Were one to inflict these levels of quantization-distortion (say, 5%) across the entire dynaimic range of a music recording, the result wouldn't just sound bad, it would be unlistenable, quite literally.

Excuse my ignorance. But what do you mean by "quantization-distortion" and how do you measure it? What does 5% mean? Is this related to THD (http://en.wikipedia.org/wiki/Total_harmonic_distortion)?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-31 21:50:08
Quote
the link posted by AndyH-ha from "some idiot audiofools say it on other boards"(as you wrote)is showing that clearly.
Huh? I suppose this is pretty petty in the midst of such cerebral discussions, but what did I do?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-31 22:42:46
Quote
but what did I do?
nothing wrong AndyH-ha.( i wrote about the link from your post and not about you.)
you posted the link to audiomasters and 2bedecided in the next post wrote "some idiot audiofools". (just before the too big letters. lol)
i used this "strange qualifications" in my answer using the link that you posted as referencial with the words that 2bedecided had used.

Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-08-31 23:21:08
I see. While everyone has their biases, the unquestioned pet beliefs of most people who uses the AudioMasters site with any frequency is generally much closer to that of ‘audio engineer' than to ‘audiophile,' so the reference was confusing.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-31 23:37:03
You sound very bitter. This isn't a vinyl versus CD debate. (I have thousands of both - I have even more 78s). This is actually about simple mathematics. It's not even science (where hypotheses can be disproven after years of being accepted) - it's maths. Once proven, it stays proven. Dither has been around for decades. The best papers wrt audio are from Lipshitz, Vanderkooy and Wannamaker, from ~1984-1992. You'll note that it's all maths.

I'm on record in this forum as saying I suspect CD isn't good enough.

I don't think calling people in other places, who refuse to understand dither and nyquist "audiofools" is against the TOS on HA. In fact, I think it's an implicit requirement of HA to hold this view, or come around to this view.

What's sad is that my previous post was a genuine attempt to explain dither to you. Not to knock vinyl, or support record companies, or declare my love for Sony and Philips - just to explain why your original question had a specific answer. I didn't run Cool Edit Pro, generate a signal, process it, capture various different spectrograms, stitch them together in PhotoPaint, and upload them to HA to insult you or anyone. I did it because a quick Google suggested that the best explanation of dither was now a "404 page", and the others were either written by people who didn't really understand it, or without the pictures I think explain it the best.

This topic (and nyquist) has been done to death. Please don't feel insulted, but let me warn you of this: the people who spend the most time trying to deny the basics of digital audio are the ones who feel the most silly when they realise they were wrong.

If or why CD isn't perfect is an interesting discussion. 30% harmonic distortion at -90dB is not there is reality, and so is not a possible reason!

Whether you believe me or not, the ABX test that's been suggested may at least put your mind at rest for your own purposes.

Cheers,
David.


I sound bitter? Diatribe? Take a look back over this last of yours. I am angry, that industrial 'fascism' denies people their legitimate choice in the pursuit of one of the simplest and most basic pleasures the modern world affords, in the pursuit of profit, yeh.

It is indeed all maths to you, apparently. If you want to pretend the model is reality itself, knock yourself out.

Nyquist? You presumably believe that bandwidth is a function of sampling rate? What is the sampling rate of an analogue system, and implicitly it's bandwidth? Oh, well, that's diiferent isn't it? Tell you what, capture that analogue waveform digitally, then you'll be able to answer the question  ....er..........

The simple fact is that Nyquist was concerned with data-trasmission, not audio, let alone music reproduction,.

You evidently believe music is sound is information (is data is bits), and therefore you have everything you need at your disposal to 'understand' it. Extraordinary hubris, IMV.

You presume to lecture and bore the crap out of me with an iteration on the subject of 'dither' (which I could  inform myself of in detail with a single google search, bringing up pages of hits written by people a good deal smarter than you, I dare say), and then claim that it's out of altruism and your desire to 'educate' me.

Give me a break.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-08-31 23:53:29
@ AndyH-ha
Quote
most people who uses the AudioMasters site with any frequency is generally much closer to that of ‘audio engineer' than to ‘audiophile,'
yes,is one cool forum too. 
you and 2bedecided are there.... lol...2bdecided,are you an "audiophile"? (just kiddin)
 

@ all
take it easy,the thread is growing amazingly and the "solution" for the doubts will came later.
i will that the mod don't close here and in the end...we all lose.

thanks.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-08-31 23:58:39
Excuse my ignorance. But what do you mean by "quantization-distortion" and how do you measure it? What does 5% mean? Is this related to THD (http://en.wikipedia.org/wiki/Total_harmonic_distortion)?


Yes THD, all harmonic disortion, er, totalled. If it consists of an arbitrary spread across many harmonics, even and odd, it's ugly, atonal, un-musical.

Distributed on a descending curve through the first 2 or 3, as usually exhibited in the analogue domain. it's subjectively benign, often called 'colouration'.

edit >> all distortion is 'harmonic', that's the only way we have of measuring and quantifying it, but it can be, and should be, weighted according to it's spead of harmonic components.

5% (say) pure 2nd harmonic is competely 'benign', of no real consequence, it simply 'warms' timbre somewhat. 5% third is much less so, it's atonal and 'sharpens' timbre. And so on.

Even-order harmonics are by definition exact octaves above a fundamental, and are therefore 'euphonic', unlike odd-order.

All of which you already knew.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-09-01 01:00:17
Nyquist? You presumably believe that bandwidth is a function of sampling rate?


Thats not really something you need to believe in.  Its actually proven (literally), so its true regardless of what one thinks.

What is the sampling rate of an analogue system, and implicitly it's bandwidth? Oh, well, that's diiferent isn't it?


Yes, the Nyquist criteria only applies to sampled data.

Tell you what, capture that analogue waveform digitally, then you'll be able to answer the question  ....er..........


That would be the easiest way to do it if you have an ADC.  Otherwise, use a tuner and do it the old fashioned way in analog.

The simple fact is that Nyquist was concerned with data-trasmission, not audio, let alone music reproduction,.


And the simple fact is that audio is data, and CD and records and computers are media for transmission that obey the rule Nyquist and Shannon discovered

You evidently believe music is sound is information (is data is bits), and therefore you have everything you need at your disposal to 'understand' it.


Why not believe it?  Nyquist proved this generations ago.  Unless you've found a flaw in the proof, I see no issue with it. 

Extraordinary hubris, IMV.


Not really, the test is quite trivial.  Play back PCM through a DAC.  If you hear music, then clearly sound is a form of data.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-01 01:11:43
Quote
And the simple fact is that audio is data...If you hear music, then clearly sound is a form of data.
 
means that when i hear a acoustic guitar, my dog,one airplane,one pretty girl talking with me...i hear data? lol
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-09-01 01:44:09
Digital data isn't music, or any kind audio, but then neither is the arrangement of magnetic domains on an analogue tape or the groove traces in an LP. They are all only means of storing information/instructions about how to reproduce an audio signal which can be converted into sound waves. In one case it is digital data, in another case it is analogue data.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 02:04:46
Nyquist's 'theorem' applys to constant RF pilot tones carrying 'multiplexed' digital data, but I'm skeptical about it's acceptance as a 'law' for defining audio bandwidth or 'time domain resolution'.

Have you ever looked at what 16/44 does to pure tones over 8KHz or so? That ain't 'fidelity' if you ask me.

Mother Of Tone (http://www.mother-of-tone.com/cd.htm) (Altmann Micro Machines)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-09-01 04:55:30
I don't now if everything he says is as nonsensical, but those graphs are made by plotting sample point amplitudes and drawing straight lines between them. That's about as meaningful as measuring temperature every hour and believing that the variations show that infinitely fast temperature shifts have occurred between one measurement and the next.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-01 09:47:08
Yes THD, all harmonic disortion, er, totalled.

Well, then you're simply wrong to assume harmonic distortions are impossible to avoid when working with PCM -- as 2Bdecided and myself already pointed out. In fact, it's mainly the analogue world that plagues us with non-linearity.

I'm OK with you not understanding the sampling theorem or what dithering is about and how it works. Unfortunately you're one of those who want to point out flaws without even trying to understand what's going on exactly. It couldn't just be that you're wrong, eh?

With you ignoring things we're just going round in circles.

I just had a look at the page you linked. Oh boy! Now I see what you mean. You're right. It looks ugly. But that just can't be due to the way the plotting program works (ie no proper reconstruction!). No, so, PCM is to blame .... riiiiight ....
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 10:43:07
Well, then you're simply wrong to assume harmonic distortions are impossible to avoid when working with PCM -- as 2Bdecided and myself already pointed out. In fact, it's mainly the analogue world that plagues us with non-linearity.

Well, then you're simply wrong to assume harmonic distortions are impossible to avoid when working with PCM -- as 2Bdecided and myself already pointed out. In fact, it's mainly the analogue world that plagues us with non-linearity.

I'm OK with you not understanding the sampling theorem or what dithering is about and how it works. Unfortunately you're one of those who want to point out flaws without even trying to understand what's going on exactly. It couldn't just be that you're wrong, eh?

With you ignoring things we're just going round in circles.

I just had a look at the page you linked. Oh boy! Now I see what you mean. You're right. It looks ugly. But that just can't be due to the way the plotting program works (ie no proper reconstruction!). No, so, PCM is to blame .... riiiiight ....


If you'd done a little more than 'look' at the page, and scrolled down , you'd have found that he goes into some detail about Shannon and oversampling, and includes actual screen shots from an analogue 'scope of HF tones from digitally filtered (oversampling) DAC.

edit>>> BTW, he's talking about CD specifically, not PCM generally, late in the article he discusses the desirabilty of higher sampling rates. But who needs higher sampling rates, CD is perfect, right?.

(Altmann of course builds non or zero OS DACS (among other things), which I've heard an HA memeber declare to be simply "broken" and implicitly not worth listening to  ).

We may well seem to be going round in circles, because I get exasperated with people attempting to 'prove' that everything in the garden is rosy with 16/44 PCM  -  I'm not the only one "ignoring things". Just read your first paragraph again. Non-linearity in the analogue world "plagues" us? No distortion of any consequence exists in 16/44 PCM?

Every attempt to state what should be obvious - that our ears and how enjoyable and 'realistic' music playback is (or isn't) should ultimately arbitrate on music reproduction is met with the catch-all "prove it. Show your ABX results", and TOS invoked. Unfortuntately this leaves little room for any meaningful debate.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-01 11:08:45
Quote
but what did I do?
nothing wrong AndyH-ha.( i wrote about the link from your post and not about you.)
you posted the link to audiomasters and 2bedecided in the next post wrote "some idiot audiofools". (just before the too big letters. lol)


The letters apparently weren't large enough.

I certainly wasn't referring to the audiomasters forums (of which I am a member - it's descended from the old Cool Edit forums, of which I was a very active member) when I talked about "audiofools". I had in mind somewhere like rec.audio.opinion or audio asylum.

Quote
i used this "strange qualifications" in my answer using the link that you posted as referencial with the words that 2bedecided had used.


I wasn't quite sure what you meant in your previous post (which is why I didn't reply). I think I'm being a bit slow, or having a language problem this week!

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-01 11:42:49
Nyquist? You presumably believe that bandwidth is a function of sampling rate?


It is RockFan. In a sampled system, it sets the absolute limit.


Quote
What is the sampling rate of an analogue system


It doesn't have a sample rate because it isn't a sampled system. You could measure the bandwidth with an (analogue!) sine generator and a spectrum analyser or (analogue!) scope etc. You can do exactly the same with a digital system, and see how close it comes to the Nyquist limit in practice.


Quote
The simple fact is that Nyquist was concerned with data-trasmission


You may be looking at the wrong theory. Nyquist is sometimes called Shannon, and there's another theory by Shannon concerned with data transmission.

Or maybe you mean the original paper by Nyquist, but that is truly obscure: when people talk about the Nyquist theorem, they mean that a sampled system can fully represent all frequency components up to but not including half the sampling rate. The actual field of the original paper in which this was first mentioned is irrelevant. As a mathematical proof of a mathematical system, it is universally true.

The ways in which it can fall down in practice with real world components (i.e. the assumptions in the proof which are difficult to realise in practice) are interesting, and may explain some of the problems with CDs (especially early ones).


Quote
You presume to lecture and bore the crap out of me with an iteration on the subject of 'dither' (which I could  inform myself of in detail with a single google search, bringing up pages of hits written by people a good deal smarter than you, I dare say), and then claim that it's out of altruism and your desire to 'educate' me.



Searching Google for dither (here in the UK at least - Google results are regionalised, even if you select all of the web)...

The 1st hit is wikipedia. In the section devoted to audio, it seems to me that the graphs are simply wrong. Even the dithered version is shown with significant harmonic distortion.

The 2nd hit is earlevel.com. It's a fair enough explanation, though I don't find the analogy about waving your fingers useful (others might) and there are no pictures. I like pictures.

The 3rd hit is by Nika Aldrich. Nika turned up on one forum years ago (I can't remember if it was r3mix, mp3.com, or somewhere else) proudly announcing his new article on dither. Let me say first that, in his area of expertise, Nika Aldrich is widely respected. However, dither apparently wasn't his area of expertise at the time, and his article was roundly criticised for being simply incorrect. From the advice on that forum (which would have included advice from people who are on HA now) I believe he corrected his article. This article is now probably very useful, because it is written from the point of view of someone learning about dither. However, I can't bring myself to like it simply because I remember how wrong it was in its first draft, and how many of the r3mix/mp3.com/HA crew are the true authors, and don't get any credit.

The 4th hit is a group called Dither.

The 5th hit is about graphics

The 6th hit is about Netscape's colour pallette

The 7th is a blog?!

The 8th hit - finally - is by Bob Katz. Now, this guy is a genius.

http://www.digido.com/portal/pmodule_id=11...der_page_id=27/ (http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=27/)

Read that one. Not the first 7. It's written by someone who knows exactly what they're talking about, it includes some pictures, and he's a very nice guy too (he helped me with ReplayGain).



Have you ever looked at what 16/44 does to pure tones over 8KHz or so? That ain't 'fidelity' if you ask me.

Mother Of Tone (http://www.mother-of-tone.com/cd.htm) (Altmann Micro Machines)


This is complete and utter nonsense. It's been done to death here on Hydrogen Audio. Here are some relevant threads...

Theoretical discussion : 44 KHz (CD) not enough !? (Nyquist etc.), plethora of distortion frequencies? (http://www.hydrogenaudio.org/forums/index.php?act=ST&f=1&t=9311)
Listening test : 96 vs. 48 or 44.1 kHz sampling --> scientific test, perhaps here is the 1. listening test ! (http://www.hydrogenaudio.org/forums/index.php?act=ST&f=1&t=6150)
Another discussion : Sound, the human ear, and the digital world (http://www.hydrogenaudio.org/show.php/showtopic/13298)

These are lifted straight out of the FAQ.


So you're arguing against things which are already in the FAQ of HA.

And rather than go away and try to understand this, you want to have a go at me?

Quote
Give me a break.


Cheers,
David.


Every attempt to state what should be obvious - that our ears and how enjoyable and 'realistic' music playback is (or isn't) should ultimately arbitrate on music reproduction is met with the catch-all "prove it. Show your ABX results", and TOS invoked. Unfortuntately this leaves little room for any meaningful debate.


You can enjoy your music as and when you please.

The purpose of ABX is to demonstrate that an audible difference exists.

This is in the HA TOS.

If you want to claim that X, Y or Z makes an audible difference without ABXing first, you're on the wrong board!

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 11:45:47
I don't now if everything he says is as nonsensical, but those graphs are made by plotting sample point amplitudes and drawing straight lines between them. That's about as meaningful as measuring temperature every hour and believing that the variations show that infinitely fast temperature shifts have occurred between one measurement and the next.


Groan. So you're saying that some or most if what he says is "nonsensical"?

He discusses the use of over-sampling/digital-filtering further down the page.

Comparing an audio waveform to room-temparature variation is bit of stretch, isn't it? Actually, come to think of it, I suppose a bomb going off could generate a flash of a few milliseconds.

There are many instruments (for e.g. almost any muted brass) which generate edges much faster than CD could ever hope to follow accurately.

edit >> actually, before the pedants weigh in, more correctly, they have ultasonic content which is extremely problematic at 16/44 . Muted cornet generates substantial pressure levels at over 40, even 50 KHz (7th harmonic and beyond), and these are not subtle overtones.

That we can't hear them as discreet components, or a recording system 'band limits' or low-passes at the limit of human hearing is irrelevent, they are intrinsic to the shape of the captured waveform inside the audio band, just as they are to a squarewave.

Interestingly, only non-OS DACs play them back 'accurately' (both examples, muted brass and squarewaves) because  they inflict no ringing or phase-shifting as digital filters do.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 12:03:47

Have you ever looked at what 16/44 does to pure tones over 8KHz or so? That ain't 'fidelity' if you ask me.
Mother Of Tone (http://www.mother-of-tone.com/cd.htm) (Altmann Micro Machines)


This is complete and utter nonsense. It's been done to death here on Hydrogen Audio. Here are some relevant threads...

Theoretical discussion : 44 KHz (CD) not enough !? (Nyquist etc.), plethora of distortion frequencies?
Listening test : 96 vs. 48 or 44.1 kHz sampling --> scientific test, perhaps here is the 1. listening test !
Another discussion : Sound, the human ear, and the digital world

These are lifted straight out of the FAQ.

So you're arguing against things which are already in the FAQ of HA.

And rather than go away and try to understand this, you want to have a go at me?



"..... complete and utter nonsense"? Prove it.

ciao,
R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-01 12:06:32
It's proven in those threads!

EDIT: Having read the page carefully, I think he knows exactly what he's saying, and intends to let naive people draw the conclusion that CD isn't good enough, while actually including enough information (not explained!) to prove that it is!

For example, he knows (or should know) perfectly well that an easily realisable reconstruction filter will give no beats on any of his measurements. However, he tests a typical cost-effective commercial filter, and surprise surprise - finds beats!

Also, he knows (or should know) that while infinite filters are needed in theory, using finite length filters is only a detectable problem if those filters are too short. Get beyond several thousand taps, and any errors are lost in the noise, even in a 24-bit system. Not infinite, but more than good enough!

Also, he must realise that his 4x CD sampling rate waveform (which he describes as music to my eyes  ) can be generated by taking a CD and resampling correctly to 176.4kHz - something any one of us can do in Cool Edit! Thus proving that all you need is right there on the 44.1kHz sampled CD!

He claims "sharp filters do not necessarily sound best to our ears". Well, he would say that, wouldn't he?. It's no surprise his filterless designs sound "different" - all that ultrasonic crap getting through is going to cause hideous intermodulation distortion in even the best equipment.

What isn't forthcoming (even from people like Bob Katz, who have tried) is that good quality filters above 20kHz are audible. Bad ones can be, but good ones?

If its that critical, where are the ABX results?

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: AndyH-ha on 2006-09-01 12:17:32
English is a hard language to communicate in. Not that it matters but I did not compare "an audio waveform to room-temparature variation." If the message isn't coming through it isn't worth re-wording.

It is well know that not everything can be perfectly captured at a given sample rate. Square waves are perhaps the most commonly cited example. Does this make any difference to music? While there are all sorts of arguments, most of the data isn't on the side of those who what to believe this a defect in CD audio.

You might consider that there are a large number of people who will present ideas very convincingly to the unwary when they hope to sell something, and that there are quite a few people who devise something to sell for the simple reason that they hope to make a living of it. Remember the traveling medicine shows? Remember that fellow who said 'there's one born every minute'?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-01 12:39:22
There are many instruments (for e.g. almost any muted brass) which generate edges much faster than CD could ever hope to follow accurately.

edit >> actually, before the pedants weigh in, more correctly, they have ultasonic content which is extremely problamatic at 16/44 . Muted cornet generates substantial pressure levels at over 40, even 50 KHz (7th harmonic and beyond), and these are not subtle overtones.

Whether we can hear them as discret components, or a recording system 'band limits' or low-passes at the limit of human hearing is irrelevent, they are intrinsic to the shape of the captured waveform inside the audio band, just as they are to a squarewave.


Hang on a moment - if we can't hear any difference when signal components above 20kHz are removed, why on earth would be care if they're recorded or not?

I'm not recording music to look at the waveform - I want to listen to it. If stuff above 20kHz is irrelevant to human ears, then we don't need to record it.

To prove that it is relevant you have to ABX with and without. If you read those threads I pointed to, you would find such experiments.

Quote
Interestingly, only non-OS DACs play them back 'accurately' (both examples, muted brass and squarewaves) because  they inflict no ringing or phase-shifting as digital filters do.


Ah, now that's pure marketing BS from the no filter DAC guy. The thing is, you clearly can't have a faster rise time in a "correct" 44.1kHz sampled system than the rising edge of a 22.05kHz sine wave. However, in a "correct" 44.1kHz sampled system, that rising edge can be anywhere. It can be between sample points. In fact, any signal can have sub-sample timing accuracy (even the phase of a 1kHz tone) in a "correct" 44.1kHz sampled system. However, without correct filtering, that breaks down. Sure, you can have a near-instantaneous rise time, but only at sample boundaries. So if the real rising edge was a little earlier or later, it's tough! It's been moved! (Imagine how much jitter that is, if you want to think of it in those terms!) More over, you'll alwayshave instantaneous rise times at sample boundaries (unless the signal is stationary), even where they weren't present in the original signal.

The only reason these non-OS DACs are even listenable is precisely because the ear doesn't respond to all the ultrasonic crap that's allowed through.

If you think I'm talking nonsense, try it at a sampling frequency (e.g. 5kHz) where the extra components are audible. It's difficult to do in Cool Edit, because even if you switch the filters off, it still uses some filtering. There may be other software which can do this though. I have done it in MATLAB. It sounds truly awful!



It is well know that not everything can be perfectly captured at a given sample rate. Square waves are perhaps the most commonly cited example. Does this make any difference to music? While there are all sorts of arguments, most of the data isn't on the side of those who what to believe this a defect in CD audio.


You can't sample square waves because they have an infinite number of frequency components going infinitely high. Funny thing is though that human ears don't respond particularly well to sine waves of infinite frequency either. Or even 30kHz. Or 0.001Hz for that matter. That's why we don't bother recording them!

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-01 12:57:32
If you'd done a little more than 'look' at the page, and scrolled down , (...)

I did actually. But it's obvious (maybe not for you 'cause you possibly think the same way) that the auther has a skewed view of things. He's showing misleading "stair step" graphs wich doesn't mean a thing.

edit>>> BTW, he's talking about CD specifically, not PCM generally, late in the article he discusses the desirabilty of higher sampling rates. But who needs higher sampling rates, CD is perfect, right?.

But I'm talking about PCM in general. You always come up with 16/44 and point out "issues" about harmonic distortions and problems with high frequency stuff. Well 16/44 is just one example of PCM. If you say 16/44 has problems you're attacking PCM in general.

We may well seem to be going round in circles, because I get exasperated with people attempting to 'prove' that everything in the garden is rosy with 16/44 PCM

Nowhere did I (as well as 2Bdecided) state that.

I'm not the only one "ignoring things". Just read your first paragraph again. Non-linearity in the analogue world "plagues" us? No distortion of any consequence exists in 16/44 PCM?

You're using the very broad term "distortions" again.
Also, I didn't say that it's impossible to create 16/44 files with harmonic distortions.
We just keep saying "when done properly you can avoid having harmonic distortions".
Regardless whether you use 16/44, 24/96 or whatever.

Every attempt to state what should be obvious - that our ears and how enjoyable and 'realistic' music playback is (or isn't) should ultimately arbitrate on music reproduction is met with the catch-all "prove it. Show your ABX results", and TOS invoked. Unfortuntately this leaves little room for any meaningful debate.

Well, what would you do if every now and then someone comes to you with crackpot theories?
After a lot of you explaining and him not listening you just get tired.

Darn! I already wasted too much time with this thread.
(Yeah "wasted", 'couse you didn't learn something.)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 13:36:05
Hang on a moment - if we can't hear any difference when signal components above 20kHz are removed, why on earth would be care if they're recorded or not?

I'm not recording music to look at the waveform - I want to listen to it. If stuff above 20kHz is irrelevant to human ears, then we don't need to record it.


Read this again, carefully;

Whether we can hear them as discret components, or a recording system 'band limits' or low-passes at the limit of human hearing is irrelevent, they are intrinsic to the shape of the captured waveform inside the audio band, just as they are to a squarewave.

Put another way, what happens to squarewave if you remove it's ultrasonic content (edit >>) or shift it's harmonics? (I don't mean a 'perfect' squarewave, just a reasonable one from one an olde worlde signal generator).

Now you're recording music to listen to, not to look at?! That's my line isn't it?!

ciao,
R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-01 14:02:32
But I'm talking about PCM in general. You always come up with 16/44 and point out "issues" about harmonic distortions and problems with high frequency stuff. Well 16/44 is just one example of PCM. If you say 16/44 has problems you're attacking PCM in general.


Absolutely not. I'm critical of 16/44 specifically, if you want to say I'm "attacking" it, that's up to you.

For the record, I believe that CD was introduced before PCM (and digital optical disc) was a mature technology - had it waited a few more years, and used at least double the sampling rate (greater bit depth is only necessary for DSPs), we probably wouldn't be arguing about it.

R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-01 14:37:14
hi all (again) 

why cd players have LPF in the output? (seems off topic but it's not)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-01 15:34:16

But I'm talking about PCM in general. You always come up with 16/44 and point out "issues" about harmonic distortions and problems with high frequency stuff. Well 16/44 is just one example of PCM. If you say 16/44 has problems you're attacking PCM in general.

Absolutely not. I'm critical of 16/44 specifically, if you want to say I'm "attacking" it, that's up to you.

You are due to the reasoning you give for higher sampling rates. Now, there are people who suggest higher rates and understand the theory behind it. Their reasoning however (about designing practical reconstruction filters and the filters' impact on "ringing") is totally different from yours. You keep mentioning distortions (meaning harmonic distortions) where in fact every (good) textbook on DSP covering dither proves that you can circumvent harmonic distortions. It has been said many times and you never picked it up and responded to that. -- Or did I miss it?

Quote
Put another way, what happens to squarewave if you remove it's ultrasonic content (edit >>) or shift it's harmonics? (I don't mean a 'perfect' squarewave, just a reasonable one from one an olde worlde signal generator).

So? Where are you going with this? It's just a filtered square wave. It won't sound any different to you unless you're a bat. Surprise: We don't care about how a wave looks and you shouldn't either.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-01 16:52:18
Whether we can hear them as discret components, or a recording system 'band limits' or low-passes at the limit of human hearing is irrelevent, they are intrinsic to the shape of the captured waveform inside the audio band, just as they are to a squarewave.


What is "the audio band"?

Most people would define it as the range of frequencies which are audible to the human ear. Typically 20Hz-20kHz for young listeners, though there are more careful definitions. What's your definition?


I'll try an example, and you tell me where you disagree...

A 10kHz square wave [EDIT: sawtooth wave!] (for example) has content at 10kHz, 20kHz, 30kHz ... That's what it is. You don't even need fourier analysis or digital processing to prove it - just take a signal generator, an analogue notch filter and a scope. Sweep the notch filter's frequency to see which parts of the frequency spectrum are relevant to the waveform. You'll find it's 10kHz, 20kHz, 30kHz... oh what a surprise, fourier was right!

Now, the first frequency component (10kHz) is within the audio band. The second one (20kHz) is on the edge (though it's outside for me!), the third one (30kHz) is seriously outside it.


If we drop everything above 22kHz, the shape of the waveform certainly changes (it looks more like a sine wave - see wikipedia sawtooth), but these changes are not "intrinsic to the shape of the captured waveform inside the audio band" because we haven't touched anything within the audio band.

Quote
Put another way, what happens to squarewave if you remove it's ultrasonic content (edit >>) or shift it's harmonics? (I don't mean a 'perfect' squarewave, just a reasonable one from one an olde worlde signal generator).


If all the changes are to the components outside the audio bandwidth, then it looks different but sounds the same.

I am aware of anecdotal evidence which claims to suggest otherwise, but it hasn't been scrutinised, peer reviewed, or repeated.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-09-02 06:20:29
Nyquist's 'theorem' applys to constant RF pilot tones carrying 'multiplexed' digital data, but I'm skeptical about it's acceptance as a 'law' for defining audio bandwidth or 'time domain resolution'.

Have you ever looked at what 16/44 does to pure tones over 8KHz or so? That ain't 'fidelity' if you ask me.

Mother Of Tone (http://www.mother-of-tone.com/cd.htm) (Altmann Micro Machines)


I'll admit I only managed to read 2/3 of that page, but those 2/3 were completely full of crap.

For starters, he lectures about Nyquist and Shannon, and then confuses their respective work.  Thats not a good sign when he can't remember which theorem is which.

Second his plots showing PCM data are blatently wrong.  The points in his graphs have clearly being interpolated to form those stair steps, and his choice of an interpolating function creates the image he wants to show, not the one that actually occurs in a real DAC, nor the one assumed in Nyquist's work.  At best hes a fool, more likely hes simply dishonest.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 13:11:02
Second his plots showing PCM data are blatently wrong.  The points in his graphs have clearly being interpolated to form those stair steps, and his choice of an interpolating function creates the image he wants to show, not the one that actually occurs in a real DAC, nor the one assumed in Nyquist's work.  At best hes a fool, more likely hes simply dishonest.


You should show a llittle more restraint with your name-calling - nevermind borderline slander.

These are constant sine-wave (!) tones created by NCH tone generator of 18, 20, and 21 KHz on my own PC.

(http://i5.tinypic.com/27yoc4j.jpg)

You were saying?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 13:49:17
What is "the audio band"?

Most people would define it as the range of frequencies which are audible to the human ear. Typically 20Hz-20kHz for young listeners, though there are more careful definitions. What's your definition?


No sh*t! You could probably be a little more pedantic if you really tried. (you'll have to forgive the sarcasm, but I take a dim view of being patronized gratuitously )

If all the changes are to the components outside the audio bandwidth, then it looks different but sounds the same.


Say that again?

It this point it is you who appears not to "understand" the issue (either that or you're being deliberately obtuse) - that is quite the absurdest statement I've heard in a long time.

You ever looked at a spectrum analysis of a high note on a muted trumpet, held by a good player? It 's a row of razor-sharp, discreet harmonics which impicilty disappear right off the end of the (20Hz to 20KHz) display. Have the ones outside the the system's band-width (or our hearing) simply 'ceased to exist'? Of course not, any more than they have when the same thing is done with a squarewave. They are indeed,  just as I put it, "intrinsic".

The timbre, the distinct tone of that instrument (and the shape of it's waveform inside the audio band) is composed of these ultrasonic harmonics, just as a squarewave is. Remove them, or move them around (phase shift them) relative to each other, either in capturing or playing back a recording of them, and the timbre of the instrument is changed, the square wave ceases to be square wave.

Band limiting or low-pass filtering does NOT, contrary to what you appear to be saying, "change the shape of the waveform" or the timbre of that recorded trumpet (or violin, or soprano voice) , but digital filtering in a 16/44 DAC certainly does.

ciao,
R.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 14:30:10
This is absurd.

Of course filtering (which changes the shape of the spectrogram) changes the waveform.

But it shouldn't alter the timbre if done properly.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 14:53:33
This is absurd.

Of course filtering (which changes the shape of the spectrogram) changes the waveform.

But it shouldn't alter the timbre if done properly.


"filtering" can mean a lot of things - in the context of a 'bit-stream' DAC it means over-sampling and noise-shaping, which bears no resemblence to simple low-pass filtering, digital or analogue, in it's intended purpose or effect.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 15:39:56

This is absurd.

Of course filtering (which changes the shape of the spectrogram) changes the waveform.

But it shouldn't alter the timbre if done properly.


"filtering" can mean a lot of things - in the context of a 'bit-stream' DAC it means over-sampling and noise-shaping, which bears no resemblence to simple low-pass filtering, digital or analogue, in it's intended purpose or effect.

You're taking my post out-of-context. By filtering I meant the things you mentioned in your previous post, which do indeed alter the spectrogram. (Unless the filtering wasn't needed.)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 16:49:12
And a change in the spectrogram is a change in the waveform. But it shouldn't alter the timbre, if the changes are out of hearing range.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 17:28:43


This is absurd.

Of course filtering (which changes the shape of the spectrogram) changes the waveform.

But it shouldn't alter the timbre if done properly.


"filtering" can mean a lot of things - in the context of a 'bit-stream' DAC it means over-sampling and noise-shaping, which bears no resemblence to simple low-pass filtering, digital or analogue, in it's intended purpose or effect.

You're taking my post out-of-context. By filtering I meant the things you mentioned in your previous post, which do indeed alter the spectrogram. (Unless the filtering wasn't needed.)


I have to confess that as I'm not an engineer, I probably use terms incorrectly at times, such as 'filtering' and 'band limiting'.

16/44 is inherently 'band limited' to 22KHz.

Filtering is (I assume) meant to refer to a real-time process performed on a signal.

If you low-pass filter a signal being captured (for e.g.) in 16/44 but no part of the filter roll-off is within 16/44's band limit, it will not affect the captured waveform or it's sound.

Any roll-off below 16/44's band-limit, it will indeed change the waveform, it's sound, and a spectrogram of it.

Subjectively the change will be obvious, it will be less bright, muffled  -  but this is not changing 'timbre' in the sense it's generally understood.

What does change timbre (and implictly fidelity or faithfulness) is the 'digital filtering' applied to the signal on D/A conversion in a bitstream DAC. It's observable, measurable. Most graphically it can be seen in what happens to a square wave (or saw-tooth).

Quantified as 'THD', the numbers are minute, and it's therefore (according to some) 'inaudible'.

The problem is it is certainly NOT inaudible - the pre/post ringing on edges and tiny phase shifts in harmonic content are easily perceptable as ..... change in timbre.

You want that proved by ABX? Well, I can't claim to have done such a test, but I don't think anybody who has actually listened carefully will question that multibit and bitstream DACs sound very different, and I'd be willing to put substantial money on my being able to tell my 13 year-old, 18-bit Audio Alchemy from any bit-stream one.

NB - I'm not claiming the AA is 'better' than any bitstream DAC (recent ones anyway) - in terms of measured performance, 'linearity' and THD, the opposite would be quite easily proved (although it will certainly be superior in terms of noise). It does hoewever sound better to me.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 17:51:32
Subjectively the change will be obvious, it will be less bright, muffled

No, for people who can't hear anything above 17kHz, a change at 18kHz won't mean a thing.

I don't understand the rest of your post, but please stop looking at spectrograms or waveforms, because a change in shape does not imply a change in perception.

What does change timbre (and implictly fidelity or faithfulness) is the 'digital filtering' applied to the signal on D/A conversion in a bitstream DAC. It's observable, measurable. Most graphically it can be seen in what happens to a square wave (or saw-tooth).

I'm no engineer either, but if you think square waves can't be recorded properly as PCM, I think you've been had.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-02 18:00:08
Quote
a change in shape does not imply a change in perception.
no? 
and why cd player need lpf/dac? (is one question only,nothing against your comments    )

http://www.ee.washington.edu/conselec/CE/k...ebit/primer.htm (http://www.ee.washington.edu/conselec/CE/kuhn/onebit/primer.htm)

http://www.audiocircuit.com/A-HTML/AA-Gene..._____-O-A01.htm (http://www.audiocircuit.com/A-HTML/AA-General-GEN/940-GEN-HTW_________-O-A01.htm)

http://www.ee.washington.edu/conselec/CE/k...audio2/95x7.htm (http://www.ee.washington.edu/conselec/CE/kuhn/cdaudio2/95x7.htm)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 18:08:37
Quote
a change in shape does not imply a change in perception.
no? 

No.

Here's a very simple example which I'm sure you'll understand.

Mix together a 15kHz sine wave and a 30kHz sine wave and make a spectrogram. Lowpass it at 22kHz and make a new spectrogram. They'll probably sound the same to most people.

I don't know much about signal processing so I can't answer your other question; I only know enough to refute the most common sort of FUD when I see it.

PS. Maybe I should have said "a change in shape does not always imply a change in perception"?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 18:26:22
No, for people who can't hear anything above 17kHz, a change at 18kHz won't mean a thing.

Why do think I mentioned "roll-off"? Filters don't just 'brick-wall' over a few Hz or 100's of Hz, even digital ones. If, say, 3 dB of attenuation is seen at 18KHz, the HF roll-off will probably have started a KHz or more lower.

But anyway, that's missing the point. Lets assume a filter that starts rolling off at 14 KHz,  is that better?

I'm no engineer either, but if you think square waves can't be recorded properly as PCM, I think you've been had.


I didn't say they can't be recorded, I was talking about D/A conversion - playback.

Bitstream DACs mangle them practically out of all recognition above 8KHz or so with ringing on signal edges. Multibit DAC's are rather better, and non/zero oversampling ones output them almost perfectly (although they do some pretty bizarre things to the signal in other respects, and according to many, not least the experts on this forum, they are theoretically unusable).
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-02 18:34:21
kjoonlee,
a.audition1.5 is one good editor to do that test?

thanks so much! 

@ all
off topic: (but still in topic)
if you all don't mind,i want to post it here,please read the link.
i was reading this post now from one respectable and advanced audio member in doom9 forum:
"I've recently had the opportunity to hear a couple of high-quality vinyl reissues that were digitized at 24-96 from a really good turntable. I've honestly never heard the albums sound so good. The definition and separation of the instruments, the dynamics, the overall sense of "natural" sound was astounding. However, these were recorded from a high-quality turntable with moving coil cartridge, separate dedicated phono preamp, etc."
start here: http://forum.doom9.org/showthread.php?p=870322#post870322 (http://forum.doom9.org/showthread.php?p=870322#post870322)
and back to the beginning of the thread if needed.

regards.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-02 19:21:52
But anyway, that's missing the point. Lets assume a filter that starts rolling off at 14 KHz,  is that better?

No, because with square waves, the second harmonic is at what, 3 times the fundamental? People can't (and don't need to) hear 42kHz.

I didn't say they can't be recorded, I was talking about D/A conversion - playback.

Bitstream DACs mangle them practically out of all recognition above 8KHz or so with ringing on signal edges. Multibit DAC's are rather better, and non/zero oversampling ones output them almost perfectly (although they do some pretty bizarre things to the signal in other respects, and according to many, not least the experts on this forum, they are theoretically unusable).

Doesn't matter. I'd still say you were had.

kjoonlee,
a.audition1.5 is one good editor to do that test?

thanks so much! 

I don't think my soundcard can play 30kHz, so it would probably sound the same to me.

"I've recently had the opportunity to hear a couple of high-quality vinyl reissues that were digitized at 24-96 from a really good turntable. I've honestly never heard the albums sound so good. The definition and separation of the instruments, the dynamics, the overall sense of "natural" sound was astounding. However, these were recorded from a high-quality turntable with moving coil cartridge, separate dedicated phono preamp, etc."

Just imagine how it would have sounded if the master tapes had been digitized directly!

Lately I've been using Korean idiom increasingly in my English posts. Here goes: even the grandaddy of a respectable and advanced audio member at doom9 has to provide objective results to be taken seriously at HA.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-02 21:14:40
Quote
I don't think my soundcard can play 30kHz
...of course,mine probably can't too.

Quote
Just imagine how it would have sounded if the master tapes had been digitized directly!
poor audio editors only can imagine. 

Quote
...provide objective results...
hummm..i know what you mean.

found one interesting .pdf that can break lots of argumments but need to read the whole article first.
from Dan Lavry:
http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-09-02 21:57:02

Second his plots showing PCM data are blatently wrong.  The points in his graphs have clearly being interpolated to form those stair steps, and his choice of an interpolating function creates the image he wants to show, not the one that actually occurs in a real DAC, nor the one assumed in Nyquist's work.  At best hes a fool, more likely hes simply dishonest.


You should show a llittle more restraint with your name-calling - nevermind borderline slander.



Its only slander if its not true. 

These are constant sine-wave (!) tones created by NCH tone generator of 18, 20, and 21 KHz on my own PC.

http://i5.tinypic.com/27yoc4j.jpg (http://i5.tinypic.com/27yoc4j.jpg)

You were saying?


All this shows is that you don't understand what you're discussing.  You haven't refuted my claim, you've simply made the same mistake as the ridiculous source you quoted.

We're talking about digital audio.  Digital as in discrete time.  You posted a continuous time waveform (because you connected the points).  This is fine for software because it makes it somewhat easier to visualize the data (stem plots tend to look ugly at high sample rates).  Where the issue comes in is when you present these abstractions as actual audio.  Thats when you're either clueless or dishonest because they are NOT actual waveforms, rather they're just abstractions invented by the software to help you visualize purely numeric information.




No, for people who can't hear anything above 17kHz, a change at 18kHz won't mean a thing.

Why do think I mentioned "roll-off"? Filters don't just 'brick-wall' over a few Hz or 100's of Hz, even digital ones. If, say, 3 dB of attenuation is seen at 18KHz, the HF roll-off will probably have started a KHz or more lower.


You really don't know what you're talking about.  Try FFTing the output of a sinc function on a DAC.  Even the cheap DAC will have the 3 dB point above 19.5k.  Maybe even above 20-21k.  I've done the measurements.  A typical sound card can produce essentially 100% amplitude up to around 20k, and good ones go higher then that.  Thats the whole idea of oversampling, it allows you to build DACs with extremely sharp cutoff filters very easily, and its why you can have $5 DACs with transistion bands above 20k.

But anyway, that's missing the point. Lets assume a filter that starts rolling off at 14 KHz,  is that better?


Why stop there?  If we're making things up, why not assume it starts at 10k?  Or 1K?  Hell lets assume there is no pass band 

Look, I'm sorry if this makes me sound like an ass, but get a decient scope and measure these things.  You're making a lot of assumptions that have no basis in reality.  If you take a few minutes to do the experiments and see for yourself what equipment is actually capable of, I think you will realize how silly this issue really is.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-02 22:39:20


Second his plots showing PCM data are blatently wrong.  The points in his graphs have clearly being interpolated to form those stair steps, and his choice of an interpolating function creates the image he wants to show, not the one that actually occurs in a real DAC, nor the one assumed in Nyquist's work.  At best hes a fool, more likely hes simply dishonest.

You should show a llittle more restraint with your name-calling - nevermind borderline slander.

Its only slander if its not true. 

Keep digging.


These are constant sine-wave (!) tones created by NCH tone generator of 18, 20, and 21 KHz on my own PC.
http://i5.tinypic.com/27yoc4j.jpg (http://i5.tinypic.com/27yoc4j.jpg)
You were saying?

All this shows is that you don't understand what you're discussing.  You haven't refuted my claim, you've simply made the same mistake as the ridiculous source you quoted.
We're talking about digital audio.  Digital as in discrete time.  You posted a continuous time waveform (because you connected the points).  This is fine for software because it makes it somewhat easier to visualize the data (stem plots tend to look ugly at high sample rates).  Where the issue comes in is when you present these abstractions as actual audio.  Thats when you're either clueless or dishonest because they are NOT actual waveforms, rather they're just abstractions invented by the software to help you visualize purely numeric information.

Utter rubbish. You're making this up as you go along. Can you read? These are the actual output of a perfectly respectable digital  tone-generator application.

They are are exactly what one one would get if a signal generator's output was captured by a 16/44 ADC. Mathematical interpolation is what allows them to be 'reconstituted' and the aliasing removed by a typical DAC (at the expense of other aspects of fidelity).


No, for people who can't hear anything above 17kHz, a change at 18kHz won't mean a thing.

Why do think I mentioned "roll-off"? Filters don't just 'brick-wall' over a few Hz or 100's of Hz, even digital ones. If, say, 3 dB of attenuation is seen at 18KHz, the HF roll-off will probably have started a KHz or more lower.

You really don't know what you're talking about.  Try FFTing the output of a sinc function on a DAC.  Even the cheap DAC will have the 3 dB point above 19.5k.  Maybe even above 20-21k.  I've done the measurements.  A typical sound card can produce essentially 100% amplitude up to around 20k, and good ones go higher then that.  Thats the whole idea of oversampling, it allows you to build DACs with extremely sharp cutoff filters very easily, and its why you can have $5 DACs with transistion bands above 20k.
But anyway, that's missing the point. Lets assume a filter that starts rolling off at 14 KHz,  is that better?

Why stop there?  If we're making things up, why not assume it starts at 10k?  Or 1K?  Hell lets assume there is no pass band 

Look, I'm sorry if this makes me sound like an ass, but get a decient scope and measure these things.  You're making a lot of assumptions that have no basis in reality.  If you take a few minutes to do the experiments and see for yourself what equipment is actually capable of, I think you will realize how silly this issue really is.

Here's a simple "thought experiment", but anyone who happens to have the hardware could do it for real.

Connect a signal generator up to a good amp and monitors, or headphones, and have a variable slope low-pass filter in line.

Configure the generator to produce, say, an 8KHz square wave and listen to it. Now start rolling HF off with your filter.

When do you start to hear a difference in the sound of the square wave???? No earlier than when the knee in the filter slope starts at 8KHz or lower -  above that it will make NO difference at all.

If you want to confirm it with measurements? Run the signal into a spectrum analyzer, or capture it (digitally, shall we say) and do the same.

The spectrogram will reveal that the the 8KHz square wave has harmonics extending in to the ultrasonic, even if practically everything above 8KHz in the actual signal has been has been rolled off.

How strange! Are we learning yet?

The somewhat misleadingly named 'digital filtering' of an D/A covnvertor is another matter. Compare the sound of your 'live' square wave to the one mangled by the 1 bit, over-sampled, noise-shaped processing of a typical 1-bit DAC. It WILL be different, I can guarantee it.

Exactly the same applies to the much more complex sound of musical instruments, especially those whose timbre is defined by an extended harmonic structure. What instrument's isn't, as a matter of fact?

edit >> a final edit for clarity (!)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Steve999 on 2006-09-03 01:06:00
Exactly.

Lately I've been using Korean idiom increasingly in my English posts. Here goes: even the grandaddy of a respectable and advanced audio member at doom9 has to provide objective results to be taken seriously at HA.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 02:16:17
Compare the sound of your 'live' square wave to the one mangled by the 1 bit, over-sampled, noise-shaped processing of a typical 1-bit DAC. It WILL be different, I can guarantee it.

Where are your ABX results?
Exactly the same applies to the much more complex sound of musical instruments, especially those whose timbre is defined by an extended harmonic structure. What instrument's isn't, as a matter of fact?

Snare drum?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 02:57:55

Compare the sound of your 'live' square wave to the one mangled by the 1 bit, over-sampled, noise-shaped processing of a typical 1-bit DAC. It WILL be different, I can guarantee it.

Where are your ABX results?
Exactly the same applies to the much more complex sound of musical instruments, especially those whose timbre is defined by an extended harmonic structure. What instrument's isn't, as a matter of fact?

Snare drum?


Snare drum? Most definitely.

edit>>  ABX results? Oh dear. Well -  disregard everything I've said.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Radetzky on 2006-09-03 03:16:44
Quote
And the simple fact is that audio is data...If you hear music, then clearly sound is a form of data.
 
means that when i hear a acoustic guitar, my dog,one airplane,one pretty girl talking with me...i hear data? lol


Why don't you leave this thread to the grown ups?

Actually, if you really want to be a smart ass, we could say your ear is actually sampling the data that is being transmitted by air.  Electric signals are what are being sent to your brain.  So we have an DAC -> air <- ADC -> brain chain that is being innefficient.

But, then, I am just being stupid myself.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Radetzky on 2006-09-03 03:27:10
There are many instruments (for e.g. almost any muted brass) which generate edges much faster than CD could ever hope to follow accurately.


... let's assume what you say is correct.  Even if that instrument was played back live in front of you, you couldn't hear the frequencies above about 18kHz (your an adult, right?).  That's below the 22.05kHz a CD can reproduce (well, assuming your equipment can!).

BTW, your vinyl table would not be able to follow that frequency either.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 03:30:44
facetiousness aside - how about any instrument  - Miles Davis's trumpet, Pinkas Zuckerman's violin, Murray Perahia's piano, Suzanne Vega's and Deborah Harry's voices ........

Yeh, I'm greedy, I want to hear them.


There are many instruments (for e.g. almost any muted brass) which generate edges much faster than CD could ever hope to follow accurately.


... let's assume what you say is correct.  Even if that instrument was played back live in front of you, you couldn't hear the frequencies above about 18kHz (your an adult, right?).  That's below the 22.05kHz a CD can reproduce (well, assuming your equipment can!).

BTW, your vinyl table would not be able to follow that frequency either.


Oh yes, it could.

edit >> LP is capable of (slightly wayward) response to 30KHz+. It's a whole different ball-game to CeeDee.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 03:33:11
PCM doesn't have to follow the waveform accurately to sound accurate. Stop looking, and start listening.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: saratoga on 2006-09-03 03:44:04
[quote name='RockFan' post='426789' date='Sep 2 2006, 14:39']
These are the actual output of a perfectly respectable digital  tone-generator application.
[/quote]

Err, whats your point? 

[quote name='RockFan' post='426789' date='Sep 2 2006, 14:39']
They are are exactly what one one would get if a signal generator's output was captured by a 16/44 ADC.
[/quote]

Completely false.  Have you ever used an actual ADC?  I'm not talking about software for recording, but the actual hardware that will let you see the data and not an interpolated or abstracted view of it.  Try it sometime and save the PCM data to a disk.

[quote name='RockFan' post='426789' date='Sep 2 2006, 14:39']
Mathematical interpolation is what allows them to be 'reconstituted' and the aliasing removed by a typical DAC (at the expense of other aspects of fidelity).
[/quote]

This is correct.  However, the its also what your "perfectly respectable digital  tone-generator application" is doing when it displays that neat little output. 

[quote name='RockFan' post='426789' date='Sep 2 2006, 14:39']
[quote name='Mike Giacomelli' post='426774' date='Sep 2 2006, 12:57']
[quote name='RockFan' post='426724' date='Sep 2 2006, 10:26']
[quote name='kjoonlee' post='426710' date='Sep 2 2006, 08:51']
No, for people who can't hear anything above 17kHz, a change at 18kHz won't mean a thing.
[/quote]
Why do think I mentioned "roll-off"? Filters don't just 'brick-wall' over a few Hz or 100's of Hz, even digital ones. If, say, 3 dB of attenuation is seen at 18KHz, the HF roll-off will probably have started a KHz or more lower.
[/quote]
You really don't know what you're talking about.  Try FFTing the output of a sinc function on a DAC.  Even the cheap DAC will have the 3 dB point above 19.5k.  Maybe even above 20-21k.  I've done the measurements.  A typical sound card can produce essentially 100% amplitude up to around 20k, and good ones go higher then that.  Thats the whole idea of oversampling, it allows you to build DACs with extremely sharp cutoff filters very easily, and its why you can have $5 DACs with transistion bands above 20k.
[quote name='RockFan' post='426724' date='Sep 2 2006, 10:26']
But anyway, that's missing the point. Lets assume a filter that starts rolling off at 14 KHz,  is that better?
[/quote]
Why stop there?  If we're making things up, why not assume it starts at 10k?  Or 1K?  Hell lets assume there is no pass band 

Look, I'm sorry if this makes me sound like an ass, but get a decient scope and measure these things.  You're making a lot of assumptions that have no basis in reality.  If you take a few minutes to do the experiments and see for yourself what equipment is actually capable of, I think you will realize how silly this issue really is.
[/quote]
Here's a simple "thought experiment", but anyone who happens to have the hardware could do it for real.

Connect a signal generator up to a good amp and monitors, or headphones, and have a variable slope low-pass filter in line.

Configure the generator to produce, say, an 8KHz square wave and listen to it. Now start rolling HF off with your filter.

When do you start to hear a difference in the sound of the square wave???? No earlier than when the knee in the filter slope starts at 8KHz or lower -  above that it will make NO difference at all.

If you want to confirm it with measurements? Run the signal into a spectrum analyzer, or capture it (digitally, shall we say) and do the same.

The spectrogram will reveal that the the 8KHz square wave has harmonics extending in to the ultrasonic, even if practically everything above 8KHz in the actual signal has been has been rolled off.

How strange! Are we learning yet?



[/quote]

I think you confused me with someone else since that reply doesn't appear to have anything to do with my post.

[quote name='RockFan' post='426856' date='Sep 2 2006, 19:30']
facetiousness aside - how about any instrument  - Miles Davis's trumpet, Pinkas Zuckerman's violin, Murray Perahia's piano, Suzanne Vega's and Deborah Harry's voices ........

Yeh, I'm greedy, I want to hear them.

[quote name='Radetzky' post='426855' date='Sep 2 2006, 18:27']
[quote name='RockFan' post='426316' date='Sep 1 2006, 02:45']
There are many instruments (for e.g. almost any muted brass) which generate edges much faster than CD could ever hope to follow accurately.
[/quote]

... let's assume what you say is correct.  Even if that instrument was played back live in front of you, you couldn't hear the frequencies above about 18kHz (your an adult, right?).  That's below the 22.05kHz a CD can reproduce (well, assuming your equipment can!).

BTW, your vinyl table would not be able to follow that frequency either.
[/quote]

Oh yes, it could.

edit >> LP is capable of (slightly wayward) response to 30KHz+. It's a whole different ball-game to CeeDee.
[/quote]

Can you hear above 20kHz?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 03:48:38
I've only read the most recent posts in this thread, but now it seems to be very off-topic from normalization.

It's probably rehashing stuff from the FAQ, so I humbly propose a thread split or a thread lock.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 03:54:56
You just don't get it

I'm really sorry.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 04:05:55
"The spectrogram will reveal that the the 8KHz square wave has harmonics extending in to the ultrasonic, even if practically everything above 8KHz in the actual signal has been has been rolled off."

You'll get it if you really try.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 04:07:31
No, it's you. You just don't understand what you're criticising, so by definition, you can only resort to straw-man attacks.

I hope you enjoy your digital music as much as I do mine.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Radetzky on 2006-09-03 04:17:25
These are the actual output of a perfectly respectable digital  tone-generator application.

They are are exactly what one one would get if a signal generator's output was captured by a 16/44 ADC. Mathematical interpolation is what allows them to be 'reconstituted' and the aliasing removed by a typical DAC (at the expense of other aspects of fidelity).


<sarcasm>
I really should ask for a refund of my electrical engineering courses.  Those Universities in Canada really teaches us crap.

Would you mind postulating for a position as a professor in signal processing?
</sarcasm>

You really believe this is what is being sent to your speakers?  This is what a DAC, even a cheap one, outputs?  Your cute.

I noticed many smart users stopped participating to this thread.  I think I should do the same.  This is really getting pathetic.

edit >> LP is capable of (slightly wayward) response to 30KHz+. It's a whole different ball-game to CeeDee.


I propose to add a section in the Wiki.  An all of fame for the funnies.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 04:27:04
No, it's you. You just don't understand what you're criticising, so by definition, you can only resort to straw-man attacks.

I hope you enjoy your digital music as much as I do mine.


I have very rewarding digital playback;

CD's ripped via a Plextor drive and Plextools, to Monkey's Audio, output via M-Audio DIO2448 card, Foobar2000 kernel-steaming, a video-broadcast quality coax, to an Audio Alchemy DAC , a nice Rotel RA 820BX4 amp and JPW Gold-monitors + passive sub.

Impressed? You would be if you heard it.

But true analogue is for those who make the effort.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 04:30:55
/me yawns. Still want to get off-topic?

Where are your ABX results comparing individual components in your setup with other components?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: RockFan on 2006-09-03 04:34:11
I've only read the most recent posts in this thread, but now it seems to be very off-topic from normalization.

It's probably rehashing stuff from the FAQ, so I humbly propose a thread split or a thread lock.


Yeah, but it's intersting all the same, No?
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: kjoonlee on 2006-09-03 04:42:08
No. I find it sad.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-04 08:33:44
These are the actual output of a perfectly respectable digital  tone-generator application.

By "output" you mean the digital signal or the visualization of it?

case: "visualization"
Mike already pointed out this is an easy way to visualize the set of sample values that has been generated. If you think it looks ugly then yes, we can all agree on it. However, it's just a simple visualization which obviously serves other purposes than to be close to what a DAC should output.

case: "digital signal"
This is probably the root of all the problems here. You think that the continues function you see plotted (stair steps) is somehow coded in your previously generated WAVE file and that the program you used to visualize the wave "just plots" it the way it's in the WAVE file. Well, you're wrong. Your digital PCM signal is just a bunch of measurements (sample points) for certain equidistant points in time. It does not dictate any visualization program to draw stair steps. The "values in-between" (between the sample points) have to be properly reconstructed (bandlimited interpolation). Even the author of the page you referenced says something about proper reconstruction (does "sin(t)/t" ring a bell?). But he seems to prefer showing stair steps for his visualizations and thus misleads the reader.

Cheers!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SamK on 2006-09-04 11:41:01
For sure, the way samples are visualized in wave editor softwares must have misled quite a few.
It seems Adobe Audition is the only one able to show the waveform as is obtained by bandlimited interpolation rather than joining the dots, so I suggest Mr Rockfan to download some demo of this software and create various signals and see what happens. (the waveforms won't be perfect fit to squares / sawtooths - they don't need to be...- but they'll be a lot closer than what he got previously).
The reconstructed waveform is just like the original one, stripped from its high-frequency content.
A waveform is pretty, but unless you can mentally compute its fourier transform, it doesnt say much about what's actually heard by human ears.

Next step is to understand that PCM44 works by assuming the stripped content is not audible, whence the "roundering" of sharp edges in the reconstruction process is a non-issue given that assumption. It's a bit of shock to learn the ear can't tell the difference between a neat, sharp square-wave and a wavy rounded-one, but after a while one gets used to it and doesn't give the same importance to waveforms views. (just realize the difference between those 2 sounds is a sound whose energy only lies in very high frequencies, and thus unaudible).

Then arguments can only be made on :
1. validity of the assumption, i.e. whether components above 20kHz can have any effect on human ears. (and how much does the ear behave in a *linear* way ?)
2. how good hardware do that reconstruction (whether 22050 Hz sampling gives enough roll-off for cheap DACs to work with, etc..)

Both those points are valid, and can lead to interesting debates.
But for now, Rockfan is defending wrong asumptions, and no-one has been able to bring him to debate on constructive subjects.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-04 12:28:07

What is "the audio band"?

Most people would define it as the range of frequencies which are audible to the human ear. Typically 20Hz-20kHz for young listeners, though there are more careful definitions. What's your definition?


No sh*t! You could probably be a little more pedantic if you really tried. (you'll have to forgive the sarcasm, but I take a dim view of being patronized gratuitously )


RockFan,

Let's be clear here. You're wrong about the topic of this thread. Every engineer and interested amateur who has dropped in to try to help has told you you're wrong.

The thought experiments you have proposed are wrong, and quite bizarre. The things you've stated are self contradictory, but you don't seem to notice.

This suggests to me that there's at least one fundamental belief that you've picked up from somewhere which you believe to be true, but which isn't. I don't mean the whole "CD vs LP" thing - I mean something about human hearing, or sound, or filtering, or something.


I have come close to patronising you when I thought you were a troll on the first page of this thread. But we're way past that now.


So the post of mine that you replied to wasn't patronising at all. I looked at what you'd posted, I tried to figure out how you and I could have such a completely different view of it.

Unfortunately, you've got very angry, and haven't posted anything helpful, so we're not getting anywhere.



Quote
Band limiting or low-pass filtering does NOT, contrary to what you appear to be saying, "change the shape of the waveform" or the timbre of that recorded trumpet (or violin, or soprano voice) , but digital filtering in a 16/44 DAC certainly does.


Here's a simple "thought experiment", but anyone who happens to have the hardware could do it for real.

Connect a signal generator up to a good amp and monitors, or headphones, and have a variable slope low-pass filter in line.

Configure the generator to produce, say, an 8KHz square wave and listen to it. Now start rolling HF off with your filter.

When do you start to hear a difference in the sound of the square wave???? No earlier than when the knee in the filter slope starts at 8KHz or lower -  above that it will make NO difference at all.

If you want to confirm it with measurements? Run the signal into a spectrum analyzer, or capture it (digitally, shall we say) and do the same.

The spectrogram will reveal that the the 8KHz square wave has harmonics extending in to the ultrasonic, even if practically everything above 8KHz in the actual signal has been has been rolled off.


Quote
The somewhat misleadingly named 'digital filtering' of an D/A covnvertor is another matter.



Those three quotes suggest that you don't know what filtering is, how it's done, or any of the implications - in the analogue or digital domain.

There was a fourth quote about how accurate a digital filter can't be in the frequency domain, which was also wrong. A digital filter can do anything you care to define which doesn't violate the time/frequency equivalent of the uncertainty principle. I would put "given enough processing power", but I can make a filter to slice a single 1Hz-wide signal from a 44.1kHz sampled signal here on my PC (not that you'd want to!), so I don't think "given enough processing power" is relevant these days (at least at the level of proving things - you still wouldn't do some of this stuff in cheap commercial real-time devices yet).


Later, in the thread, you quoted yourself ...

Quote
"The spectrogram will reveal that the the 8KHz square wave has harmonics extending in to the ultrasonic, even if practically everything above 8KHz in the actual signal has been has been rolled off."

You'll get it if you really try.


The fact that you quoted this is quite disturbing! I get the feeling you think this was your finest hour, revealing the something about the true nature of digital recording - something which every audio engineer failed to grasp, but RockFan uniquely understood.

I'm almost speechless.

Isn't is possible that the hundreds of thousands of people who understand this subject are right, and you (who by your own admission don't actually understand filtering) are wrong?


What you've written above is actually "The spectrogram will reveal that the 8KHz square wave has harmonics extending in to the ultrasonic, even if there are no harmonics extending in to the ultrasonic."

You're either trying to invent a new kind of spectrogram, or invent a new kind of "filter" that doesn't "filter".

Try visiting reality - it makes a surprising amount of sense. Filers filter, and spectrograms show what is there (not what isn't). It's great!

(OK, that was patronising, but the rest of the post was serious.)

If you want to work through this, and figure out what someone has told you which has skewed your thinking so badly on this, fine. If you want to read a DSP text book and learn something, fine.

If you want to come back, say that we're all wrong, and propose incorrect thought experiments which you won't let anyone challenge or dissect, then I think you'll wear out your welcome very quickly.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-04 17:08:33
RockFan, this is for you!
I can do plots, too. 
The name of this pic is "reconstruct 18 kHz"
Drop me 20 bucks and I'll make you a poster.

(http://homepages.uni-paderborn.de/sgeseman/reconstruct18kHz.png)

Magic? Not really. This approach only corresponds to a reconstruction filter whose impulse response is a good deal closer to the normalized sinc function (see Whittaker–Shannon interpolation formula (http://en.wikipedia.org/wiki/Whittaker%E2%80%93Shannon_interpolation_formula)) than your stair step thingy.

Of course, this isn't the only practical way to get a decent reconstruction -- but one of the better ones and IMHO well-suited for visualisation within a Wave editor.

Cheers!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: singaiya on 2006-09-04 18:01:57
What I'd like to ask people here is; in their experience, is normalization completely 'benign', sonically? Are the algoritms used in different applications much the same, or are some better than others?

R.


Why ask this question, only to argue against everybody's answers received for the next five pages? If you (think you) knew the answer already, why bother asking, unless you are trolling?

IMO, people have been more than patient enough, and going out of their way trying to explain it to you.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-04 18:51:53
Quote
...people have been more than patient enough...

for me is good,i'm learning lots with their patience answers.
 
go ahead boys.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: equalrightsforwerewolves on 2006-09-05 07:11:11
for me is good,i'm learning lots with their patience answers.


i second that! 
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: uart on 2006-09-06 04:59:35
(http://img71.imageshack.us/img71/2686/pcm14kat44k16bitxx5.jpg)

(http://img479.imageshack.us/img479/31/cro14kadp7.jpg)

(http://img71.imageshack.us/img71/8690/cro14kbui0.jpg)


Top Trace : 14khz tone as 44.1k 16bit PCM file. This is the view of the actual file when loaded into "Audacity".

Middle Trace : Actual analog out when above PCM file is played (foobar) without any resampling or DSP. Soundcard is onboard "soundmax" AC97 of cheap motherboard. (Photo of osciloscope image)

Bottom Tave : As above but zoomed in.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: bhoar on 2006-09-06 07:35:51
Top Trace : 14khz tone as 44.1k 16bit PCM file. This is the view of the actual file when loaded into "Audacity".

Middle Trace : Actual analog out when above PCM file is played (foobar) without any resampling or DSP. Soundcard is onboard "soundmax" AC97 of cheap motherboard. (Photo of osciloscope image)

Bottom Tave : As above but zoomed in.


Magic!*

-brendan

* or is it just reality conforming to the theory... 
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pepoluan on 2006-09-06 17:15:02
I get stressed when I see this thread... clearly remembered those Digital Signal Processing classes which I failed the first time I took it...

Why? Because I was thinking like you, RockFan. The flawed "Connect-the-dots" point-of-view.

But when someone gently and patiently guide me... showing me things that are basically what SebastianG showed...

Ah! The light comes out!

I retake the course and scored an 'A' there...

So, please RockFan. Before commenting further, do buy some introductory and intermediate books on Digital Signal Processing instead of relying on your (flawed) point of view.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-06 17:28:35
I think RockFan bowed out of this discussion.

After checking Hydrogenaudio Forums > Misc. > Recycle Bin, I think I know why.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pepoluan on 2006-09-06 18:50:53
Quite rare seeing TOS#2 got invoked
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: cabbagerat on 2006-09-06 19:07:51
Cool, thanks uart! What brand of scope is that? Digital scopes are really great tools, but they just don't have the same feel as the old school green phosphor CRT type 
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2006-09-06 20:45:07
Quote
2bdecided wrote:
Searching Google for dither (here in the UK at least - Google results are regionalised, even if you select all of the web)...


The 3rd hit is by Nika Aldrich. Nika turned up on one forum years ago (I can't remember if it was r3mix, mp3.com, or somewhere else) proudly announcing his new article on dither. Let me say first that, in his area of expertise, Nika Aldrich is widely respected. However, dither apparently wasn't his area of expertise at the time, and his article was roundly criticised for being simply incorrect. From the advice on that forum (which would have included advice from people who are on HA now) I believe he corrected his article.


He did -- it's noted with a date at the end of the article.  I believe his book (Digital AUdio Explained for the Recording Engineer) incorporates the correct information too (and has a thank-yous section)

Quote
This article is now probably very useful, because it is written from the point of view of someone learning about dither.


It was to me.

Quote
However, I can't bring myself to like it simply because I remember how wrong it was in its first draft, and how many of the r3mix/mp3.com/HA crew are the true authors, and don't get any credit.


Having corresponded with and talked to Nika, it's hard for me to imagine he would purposely deny them credit if due.  Have any contacted him to complain?


Quote
The 8th hit - finally - is by Bob Katz. Now, this guy is a genius.

http://www.digido.com/portal/pmodule_id=11...der_page_id=27/ (http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=27/)

Read that one. Not the first 7. It's written by someone who knows exactly what they're talking about, it includes some pictures, and he's a very nice guy too (he helped me with ReplayGain).


Yes, I've had a couple of nice email interactions with him as well.  I like his book very much too (Mastering Audio).  I do wish he would publish some of his blind/ABX test results though, given some of the things he claims are audible.


Quote

Every attempt to state what should be obvious - that our ears and how enjoyable and 'realistic' music playback is (or isn't) should ultimately arbitrate on music reproduction is met with the catch-all "prove it. Show your ABX results", and TOS invoked. Unfortuntately this leaves little room for any meaningful debate.


You can enjoy your music as and when you please.

The purpose of ABX is to demonstrate that an audible difference exists.



Not to mention that blind tests are the best known way to make sure one's *ears* are 'arbitrating', rather than one's eyes or one's prejudices. If someone's not willing to do a blind comparison, maybe they don't really trust their ears.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pepoluan on 2006-09-06 21:48:57
Not to mention that blind tests are the best known way to make sure one's *ears* are 'arbitrating', rather than one's eyes or one's prejudices. If someone's not willing to do a blind comparison, maybe they don't really trust their ears.
If they don't trust their ears, perhaps they need to use some DSPs (http://en.wikipedia.org/wiki/Marijuana)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: dv1989 on 2006-09-06 21:52:16
Let's just hope he's gone for good. uart has proved what everyone else knew. Thanks, uart!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-07 16:40:44
(images snipped)
Top Trace : 14khz tone as 44.1k 16bit PCM file. This is the view of the actual file when loaded into "Audacity".

Middle Trace : Actual analog out when above PCM file is played (foobar) without any resampling or DSP. Soundcard is onboard "soundmax" AC97 of cheap motherboard. (Photo of osciloscope image)

Bottom Tave : As above but zoomed in.


This bit (with images) should be in the FAQ!

The fact that it even works with an on-board AC97 sound card says it all!

(Mind you, it's probably resampled to 48kHz internally, though that's not relevant to the point you're proving).

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: uart on 2006-09-07 16:53:52
Cool, thanks uart! What brand of scope is that? Digital scopes are really great tools, but they just don't have the same feel as the old school green phosphor CRT type 


Hi cabbagerat. It was my brothers osciloscope that I'd borrowed last weekend for another project. I cant recall the model number right now but I think it's only a cheapie. Anyway I was reading this thread and the silly claims made on that page that Rockfan linked to so I thought I should snap a view photos of actual waveforms to help debunk it. I didn't get around to uploading them for a few days but I'm glad I did. I know it just confirms what most of us already knew anyway.

BTW. The small artifacts that you can see on those osciloscope traces are properties of the oscilscope and not the waveform. If rockfan is still reading I can assure him that if I connect an analog signal (sine) generator to the same scope that the waveforms look exactly the same.

Well as far as the "16bit 44.1kHz PCM is not good enough for really high quality audio" debate goes, I'm not totally closed minded to the possiblity that higher bit depths and sample rates might offer some small improvement to some people, but I really hate to see CD digital audio unfairly criticized like in that above link. Personally I am somewhat skeptical of whether the higher bitrate stereo (like 96kHz 24 bit etc) will ever be consistantly ABX'able over a well mastered CD. Sure it's a great idea to master the music at higher bitdepth and samlpe rate. Higher sample rates make the analog anti-aliasing filters much easier and higher inital bitdepths mean that edits can be lossless at the final bitdepth level.  I think they choose the standards pretty well with the good old CD. They certainly the didn't go overboad on the bitrate side, with only 650MB per disc they couldn't afford to, but 44.1,16 still gives 20khz bandwidth and about 100dB dynamic range and damit that's good enough for me.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-07 17:48:31
Quote
Higher sample rates make the analog anti-aliasing filters much easier

Aliasing occurs on the A/D side of things, not the D/A side of things.

I think the misuse of this word was corrected a while back.

EDIT:  A while back in a different thread, sorry.
http://www.hydrogenaudio.org/forums/index....&pid=427756 (http://www.hydrogenaudio.org/forums/index.php?act=findpost&pid=427756)

Of course a higher sample rate will require a more simple prefilter but I'm pretty sure you're talking about playback, not sampling.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: uart on 2006-09-07 18:04:47
Aliasing occurs on the A/D side of things, not the D/A side of things.
I think the misuse of this word was corrected a while back.


Yes that's what I meant. When I said mastering I really meant "recording and mastering", the A/D side was most definitely what I had in mind when I mentioned anti-aliasing filters.

The quote in context was
Quote
Sure it's a great idea to master the music at higher bitdepth and samlpe rate. Higher sample rates make the analog anti-aliasing filters much easier and higher inital bitdepths mean that edits can be lossless at the final bitdepth level.


Maybe I didn't really make it clear what I was trying to say. I was trying to say that I can see definite advantages to recording and mastering at higher bit-rates but at this point in time I'm still somewhat skeptical about whether higher bit-rates (96k 24bit etc) will offer any real advantage for stereo playback. Not for most listeners in any case.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-07 18:13:40
Sure it's a great idea to master the music at higher bitdepth and samlpe rate. Higher sample rates make the analog anti-aliasing filters much easier and higher inital bitdepths mean that edits can be lossless at the final bitdepth level.


Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2006-09-07 20:52:16
Check out this spin on the 'square wave'  'problem'. 
(It's Chris(tine) Tham again ...who used dubious methods to show that LPs have more dynamic range than CDs, in another Audioholics article)

http://www.audioholics.com/techtips/specsf...igitalAudio.php (http://www.audioholics.com/techtips/specsformats/DigitalAudio.php)

scroll to the bottom part about 'non-sine waves'
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-07 21:09:50
I read your post regarding the conclusion of that link prior to editing.

Quote
This is perhaps true, but the counter argument is that if a digital player is not playing back a 0dB 19997kHz sawtooth (or a 0dB 10kHz square wave) with even amplitude accuracy (let alone harmonic accuracy), then that represents a kind of “distortion” that is audible.

It is hillarious how the author can talk about "amplitude accuracy" without bothering to say anything about quantization error.  What is being talked about here is solely "harmonic accuracy."
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-08 08:28:54
The article started out pretty good. But just to make sure that nobody gets the wrong idea: She made a big mistake.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-08 08:38:10
Her square wave was produced within Adobe Audition after choosing 44.1kHz/16-bit for the settings.  I get exactly the same shape as what is shown in the article.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2006-09-08 08:51:23
Are you implying something?

EDIT:
Because a 10 kHz square wave is composed out of the following harmonics: 10 kHz, 30 kHz, 50 kHz, 70 kHz, 90 kHz, .... the only thing you should get at 44.1 kHz sampling rate is a 10 kHz sine wave. Assuming you're right (I don't have Adobe Audition to test it) you may want to check the spectrum view. I'm sure you'll see a lot of aliasing.  ==> Don't use Adobe Audition for generating other-than-sine waves as it samples them without removing harmonics above the Nyquist frequency first.

EDIT2:
You would expect Adobe Audition to create proper waveforms. She probably trusted Adobe Audition. Thou shall not jump to conclusions without knowing exactly what the software does you are working with. This applies to all the RockFan-type people. In his case he was fooled by a software's wave form visualization.

EDIT3: I just tested Audacity. You'll get horrible aliasing as well when you generate sqaure waves. Google for alias-free waveform generation and/or bandlimited step if you're intestested in digital sound synthesis.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: greynol on 2006-09-08 08:58:51
That she knew the square wave that she produced was bandwidth limited.

The label of the plot was "Figure 9: 10kHz 0dB square wave sampled at 44.1kHz 16 bits"

Either this or I fail to grasp your point.  I mean, what would a non-bandlimited 10kHz 0dB square wave look like except for what was in the plot?

EDIT:  Ah, yes (ligthbulb goes on), a prefiltered 10kHz square wave would be a 10kHz sine wave.  I failed to grasp your point!
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2006-09-08 11:57:37
Cool Edit Pro (so presumably audition) is useless for alias-free square waves

10kHz generated at 44.1kHz...

Waveform: [attachment=2511:attachment]

Spectrum: [attachment=2512:attachment]

If you listen to it, it sounds like the fundamental is at 100Hz, with lots of screeching on top!

TBH, you would have to be a bit daft to listen to the result and still think that the software was working properly.

When I first spotted the problem, I "solved" it by generating square waves (actually swept square waves) using Cool Edit Pro at 100x the target sample rate (i.e. 4410kHz sampled) and resampled them back down to 44.1kHz. It gets rid of most of the false harmonics, but there's still a few ~30dB down.

Thinking about it, there are neater solutions, even in Cool Edit. e.g. generate the required (in band) harmonics manually by summing sine waves (settings in the tone generator let you do this).

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: uart on 2006-09-08 14:33:24
Thinking about it, there are neater solutions, even in Cool Edit. e.g. generate the required (in band) harmonics manually by summing sine waves (settings in the tone generator let you do this).


Yep, that's exactly what Sebastian was pointing out. Both of those "problem samples", the 10kHz square wave and the 19.997kHz triangular wave, would each precisely be a perfect sine wave if they were band-limited to 20KHz before sampling as they should have been. The only relevant question that the author could legitimately have posed would have been whether the human ear could distinguish between those waveforms and the corresponding filtered sine wave when played though high end stereo equipment. Certainly the author made no attempt at all to demonstrate that she could hear the difference between a 19.997kHz triangular wave and a 19.997kHz sine-wave.
.

Quote
About Christine Tham
Christine Tham has always been a keen "hi fi" enthusiast, which is an affliction she inherited from her father. She was interested enough in the subject to enroll in an Electrical Engineering degree at Sydney University , but decided that was not for her and graduated instead with a University Medal in Computer Science (Honours) and subsequently a Master of Applied Finance from Macquarie University.


Hey, too bad that she pulled out of that EE course before they covered the stuff pre-sample filtering.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pepoluan on 2006-09-08 18:08:55
Hey, too bad that she pulled out of that EE course before they covered the stuff pre-sample filtering.
And not even all EE programs has Digital Signal Processing... case in point: There are 7 majors in the EE program of my university, and only 2 of them requires the DSP courses; optional for the other majors...

Edit: Added "my university" there.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-09-08 18:23:18
When I went to college - for an EE degree - the DSP class was optional. And I didn't take it. I really wish I did, but as it turns out, one of the books I wound up having had a very good treatment of the topic, so I've studied up.

Once you have a good grasp of Fourier transforms you can figure the rest out. Unfortunately, only EEs (well, and probably MEs and physics majors) ever have a need to delve into that stuff.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pepoluan on 2006-09-08 19:23:08
Yes, all majors in EE requires a knowledge of Fourier transforms.

However I finally recall that Discrete Signal Processing (in my university) is mandatory only for the Telecommunications major. It is optional for all the other majors... and due to its complexity, naturally no one wants to take that satanic course
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2006-09-08 20:08:02
I posted a complaint thread about Tham's stuff on Audioholics.  Any takers here for the reply from the editor?

http://forums.audioholics.com/forums/showt...6095#post206095 (http://forums.audioholics.com/forums/showthread.php?p=206095#post206095)


Quote
Perhaps you can direct the more knowledgable people on those forums to construct a new test sequence that is more real world? We're open to additional input.
__________________
Clint DeBoer
Editor in Chief
Audioholics
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: jlt on 2006-09-08 21:31:50
Quote
We're open to additional input
why he don't came here as here was started?
we're open too(and a long time)

back to the title: 'Normalization' of PCM audio - subjectively benign?

i have read about lossy and lossless,lossy lose few details and lossless don't lose anything.
seems ok!
imagine if one (sample)music that we don't know what was done,if normalized,or equalized or if was changed from 16 to 32bit and back to 16bit again,or was 44.1k to 48k and back to 44.1k again,etc.

@ moderators
can i host the (sample)music for analysis?


if i can host and you all download,as nobody knows if any effect was used,i ask:

1- was normalized(how much?) or was not normalized?
2- if was normalized,is benign or not?
3- is 44.1k now but was 48k?
4- if was 48k and now is 44.1,what is lost?
5- if was 48k and now is 44.1k,how was dithered?
6- was really dithered? how?
5- was equalized or not?
6- if was equalized,where and how much was done?
..of course,we can encrease the numbers of questions.

if anyone can tell what was done with the source that i want to host,i can trust that everybody can tell if normalization is benign or not and if have real audibles differences between lossy and lossless.
if nobody can't hear,what is the difference if normalization is benign or not?

the central point: is possible to have one true answer?

thanks
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2006-09-08 21:56:48
I'll take on the dynamic range article if somebody else comments on the sampling article.

EDIT: I replied to that thread, comments on my analysis welcome.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2006-09-12 19:44:55
I'll take on the dynamic range article if somebody else comments on the sampling article.

EDIT: I replied to that thread, comments on my analysis welcome.



Thanks Axon, perhaps Clint will consider replacing her article with yours?  I suspec though that he'd want some actual data from using your protocol.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Radetzky on 2006-09-21 13:48:07
Yes, all majors in EE requires a knowledge of Fourier transforms.

However I finally recall that Discrete Signal Processing (in my university) is mandatory only for the Telecommunications major. It is optional for all the other majors... and due to its complexity, naturally no one wants to take that satanic course


Where I did my computer engineering course (University of Sherbrooke in Quebec, Canada), all Computer Engineering & Electrical Engineering students had to take a "Communications" course where we would be exposed to Fourrier and more generally to Laplace transforms.  We would study Shanon and Nyquist.  The final project was the simulate, using Matlab, the quantization & sampling process occuring with music @ 16/44.1.  We had to reconstruct the signal and show the difference with the original signal.  How cool is that?  It was very instructive.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2008-07-29 20:21:24
I posted a complaint thread about Tham's stuff on Audioholics.  Any takers here for the reply from the editor?

http://forums.audioholics.com/forums/showt...6095#post206095 (http://forums.audioholics.com/forums/showthread.php?p=206095#post206095)


Quote
Perhaps you can direct the more knowledgable people on those forums to construct a new test sequence that is more real world? We're open to additional input.
__________________
Clint DeBoer
Editor in Chief
Audioholics




Something (a glitch?) made that thread appear invalid for awhile.  But it's still there.

http://forums.audioholics.com/forums/showthread.php?p=206095 (http://forums.audioholics.com/forums/showthread.php?p=206095)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: pdq on 2008-07-29 20:54:19
The link doesn't work for me.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2008-07-29 21:24:46
(http://www.high-techproductions.com/indian_h.jpg)


I can't see the article either.

EDIT: I will save a "J'ACCUSE!" moment for a short while longer. This could get sensitive.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2008-07-30 05:10:05
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2008-07-30 05:16:46
Yeah... it's gone "sensitive".
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2008-08-01 17:09:58
Yeah... it's gone "sensitive".


and now....it's back.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SpasV on 2008-08-03 23:57:56
 It is obvious you know that, unless you normalize at the pick amplitude, the normalization changes the sound. Another question is how audible it is.
If normalization decreases the sample values it introduces spectrum distortions also.
In any case, I think, it does not influence the quantization noise as seems to me the discussion is about.

For the simple case when the quantization noise is considered “white”, and this model is realistic, as it is said in “Discrete Time Signal Processing” by Alan V. Oppenheim and Ronald W. Shafer when “…the signal is sufficiently complex and quantization steps are sufficiently small so that the amplitude of the signal is likely to traverse many steps from sample to sample”,
I think: The quantization noise depends only on the quantizer used and does not depend on the signal.

So, to me, the normalization in the case of increasing the signal does not distort it as long as it stays within the acceptable number range.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2008-09-07 17:04:11
Once more into the breach....

Over on AVSForum, someone named 'DulcetTones" cited the 'Mother of Tone' page as an authoritative reference against the idea that we get 'perfect sine waves' from digital audio,

http://www.avsforum.com/avs-vb/showpost.ph...mp;postcount=74 (http://www.avsforum.com/avs-vb/showpost.php?p=14598158&postcount=74)


Quote
I might as well point to these again as some seem intent on saying the usual vague facts.
http://www.mother-of-tone.com/cd.htm (http://www.mother-of-tone.com/cd.htm)
Article is correct, no-one showed anything wrong last time. Apart from some vague comments as usual.
In other words those who say you end up with a perfect sine wave are wrong.
It seems for some reason there are a few who decide to talk about CD and say its a sinewave and yet do not understand the significance of the rectangular sinusoidal component.
This is shown in the above link, you only end up with the sine wave after the reconstruction filters work their logic on the rectangular sinusoidal components shown in the URL.
Also the URL is good as it shows some other challenges with sine waves and PCM.


I pointed out that that article had been, um, *critiqued* on HA.org, (pointing to this thread) as being rather skewed.
(I also agreed that we don't get 'perfect' reconstruction of the input signal from any audio repro medium, but digital properly done is the closest by far)

Much hooing and hahing ensues, during which DT claims HA folk don't know what they're talking about. Finally this:

http://www.avsforum.com/avs-vb/showpost.ph...p;postcount=120 (http://www.avsforum.com/avs-vb/showpost.php?p=14608422&postcount=120)

Quote
Krab,
Ok to prove which of the two sources we are quoting actually know what they are talking about.
The person you hold in high regard who is basically stating Mother of Tone do not know what they are talking about commented in the forum you hold in high regard that Mother got the Nyquist and Shannon formulas wrong way round.
In fact your guy was quite explicit in saying this was a fundamental mistake. 

Well hate to burst your bubble bud:
1st Part of the equation discussion on Mother.
Quote

Well, what did Nyquist say ?

maximum data rate in a noiseless channel = C = 2*W log base2( L ) bits/sec

* where 2W is 2 times the highest frequency contained in the noiseless channel, and

* where L = number of discrete levels (e.g., binary = two levels, 0 and 1)

As Nyquist seems to have been more interested in data transmission than in high-fidelity, we should not wonder, that his statement just defines a maxiumum data-rate of a communications channel.


This is EXACTLY CORRECT and pertains to the original Nyquist paper I suggested you read.
http://www.nctt.org/pages/resources/simulations/nyquist.php (http://www.nctt.org/pages/resources/simulations/nyquist.php)

Nice simple comparison, exact formula.

Then the next part in Mother expanding on the original works of Shannon:

Quote
Later, Claude Shannon said:

If a function s(x) has a Fourier transform F[s(x)] = S(f) = 0 for |f| > W, then it is completely determined by giving the value of the function at a series of points spaced 1/(2W) apart. The values sn = s(n/(2W)) are called the samples of s(x).

This goes much further than Nyquist's words, in that it states, that a signal which consists of sine waves with a maximum frequency of W is completely described by recording its values twice as fast as W.

The real cool thing is that Shannon also gave a interpolation formula to get back to the original signal:


Unfortunately this formula includes an infinite sum... What does that mean ?


Again this can be seen from a link that just focuses on the data from Nyquist and Shannon.
http://www.fact-index.com/n/ny/nyquist_sha...ng_theorem.html (http://www.fact-index.com/n/ny/nyquist_shannon_sampling_theorem.html)

Save the effort of reading it all here it is:

Quote
If a function s(x) has a Fourier transform F[s(x)] = S(f) = 0 for |f| > W, then it is completely determined by giving the value of the function at a series of points spaced 1/(2W) apart. The values sn = s(n/(2W)) are called the samples of s(x).
Also the Mother of Tone went on to show the Interpolation formula (that does not copy into here) and this matches again EXACTLY the paper by Shannon (can find it in Section II The Sampling Theorem).


So the guy your quoting actually made what I would call FUBAR statement in his argument against Mother of Tone.

The person you quote argues just as persistently as you do, makes mistakes such as stating the "infinite filter" are already here due to filters having 10000s of taps, this is rather obtuse and why I asked you earlier do you know which products are using 256 or 1024 tap lengths.

Notice 1024 is substantially smaller than the 10000s your colleague states and the 1024 type are only found in the significantly more expensive DAC-CD players.

Now why the heck would I want to wade into the bull on another forum when it is giving me a headache arguing with someone even on this forum who does not even take the time to read and follow up on both :
Nyquist Certain Topics in Telegraph Transmission Theory
Shannon Communication in the Presence of Noise



(NB: DT appears to have conflated what MikeG wrote about 'Mother of Tone' getting Shannon and Nyquist crossed, with what 2bdecided said about 'several thousands' (not '10000s') of taps....conflating at least two people, or possibly three if SebastianG's objections are included, into one)

Anyway, I suggested that rather than me arguing by proxy for those on HA that he considers incompetent and wrong about the 'Mother of Tone' article, that he address you directly here, but as you see he demurs.  So I'm bringin' it here, for said parties to respond ...or not.

Btw, I of course have no issue with either Shannon or Nyquist, and my issue on that AVSF thread was with people pointing to the dreaded 'stairsteps' views as an indication of how supposedly 'problematic' Redbook is at various frequencies, and also with the lack of audibility test data from people who expect DACs that measure differently, to routinely sound different.

EDIT: fixed broken URLs in the quote
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: SebastianG on 2008-09-08 09:36:58
As I understand it DT acknowledges the sampling theorem and goes on to say that ideal reconstruction is impossible because the ideal reconstruction filter's impulse response (normalized sinc curve (http://en.wikipedia.org/wiki/Sinc_function)) extends infinitly to both sides. So far, so good. But in practice this is not an issue. You can always design a practical reconstruction filter that is good enough. Take CDDA for example: it's not a problem to design a practical filter that faithfully reconstructs the audible band (0-20 kHz) and rejects image frequencies (those above 22.05 kHz) by 100 dB or more. The impulse response of a filter like this can be as short as 2 milliseconds:
[a href="http://img158.imageshack.us/my.php?image=reconstruct44uq7.png" target="_blank"]

Cheers,
SG
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2009-07-15 19:13:38
A heads up:

uart's 'scope screen cap (http://www.hydrogenaudio.org/forums/index.php?showtopic=47827&view=findpost&p=427738) on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums)  of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm (http://www.physics.sc.edu/kunchur/Acoustics-papers.htm)

Woodinville's (and Axon's)  initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464 (http://forum.stereophile.com/forum/showflat.php?Cat=0&Number=69464&an=0&page=0#Post69464)

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf (http://www.physics.sc.edu/kunchur/papers/FAQs.pdf)
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Woodinville on 2009-07-16 00:47:40
A heads up:

uart's 'scope screen cap (http://www.hydrogenaudio.org/forums/index.php?showtopic=47827&view=findpost&p=427738) on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums)  of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm (http://www.physics.sc.edu/kunchur/Acoustics-papers.htm)

Woodinville's (and Axon's)  initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464 (http://forum.stereophile.com/forum/showflat.php?Cat=0&Number=69464&an=0&page=0#Post69464)

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf (http://www.physics.sc.edu/kunchur/papers/FAQs.pdf)


The equivocation and attempt to deflect is really pretty astonishing to me.

The unwillingness to actually address the facts of the frequency response difference implied by a two-tap FIR filter with the two taps 5 microseconds apart is, for me, the final straw. Referring to a mechanism that creates an audible difference within the 20kHz band as mathematical nitpicking is simply not a reasonable response.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: uart on 2009-07-16 02:16:30
A heads up:

uart's 'scope screen cap (http://www.hydrogenaudio.org/forums/index.php?showtopic=47827&view=findpost&p=427738) on this thread. has been referenced (negatively) in a reply by Dr. Milind Kunchur to critiques (from Woodinville on Stereophile's forum, and, apparently, from other critics on other forums)  of his published claims about the insufficient 'time resolution' of Redbook, briefly summarized here

http://www.physics.sc.edu/kunchur/Acoustics-papers.htm (http://www.physics.sc.edu/kunchur/Acoustics-papers.htm)

Woodinville's (and Axon's)  initial and new critiques are here

http://forum.stereophile.com/forum/showfla...age=0#Post69464 (http://forum.stereophile.com/forum/showflat.php?Cat=0&Number=69464&an=0&page=0#Post69464)

Dr. Kunchur's reply is here:

http://www.physics.sc.edu/kunchur/papers/FAQs.pdf (http://www.physics.sc.edu/kunchur/papers/FAQs.pdf)


That's interesting, I hadn't seen that pdf-file reply before. Just one quick question, in the reply he repeatedly makes reference to the "5 microsecond or better temporal resolution of the human hearing". Does anyone know where he got that figure from or how that was measured? It seems a bit extreme to me.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2009-07-16 04:55:40
He measured it himself, in some (peer-reviewed!) papers, and actually, it's a figure that nobody involved (jj or krab or I) really disagree with. What this whole spat is about is the meaning of such a result, especially in the context of the fidelity of redbook, and in a larger context, the meaning of the term "resolution".

There's a really long backstory/flamewar behind all of this that you have not been introduced to (and I won't even try to do so). krab only posted because I used your scope shots as a trivial demonstration of the ability of mainstream sound cards to reproduce 14khz sine waves without distortion. Kunchur's response to your plot is a little bizarre to me, but I'm still crunching on everything before I will comment any further.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Woodinville on 2009-07-16 21:11:51
He measured it himself, in some (peer-reviewed!) papers, and actually, it's a figure that nobody involved (jj or krab or I) really disagree with. What this whole spat is about is the meaning of such a result, especially in the context of the fidelity of redbook, and in a larger context, the meaning of the term "resolution".

There's a really long backstory/flamewar behind all of this that you have not been introduced to (and I won't even try to do so). krab only posted because I used your scope shots as a trivial demonstration of the ability of mainstream sound cards to reproduce 14khz sine waves without distortion. Kunchur's response to your plot is a little bizarre to me, but I'm still crunching on everything before I will comment any further.


Specifically, he plays one pulse, and then two half-amplitude pulses separated by 5 microseconds, and listeners with unimpaired hearing in good settings can hear the difference.

Given that this is a comb filter with its first zero at 100kHz, being able to hear the difference at 20kHz isn't too surprising, now.

So, this shows a mechanism for which purely in-band (i.e. dc - 20kHz) signal can allow the distinction.

Let's see....

Down about .45dB is just at the DL.  In really good conditions, for direct, immediate comparison, with no level roving, etc.

So, what did the doctor actually discover?

Nothing that I can tell.

This, of course, after misdirection upon misdirection from the people 'defending' his work, starting with quotes that made things look like it was interaural resolution (which is also audible, just barely, in the best conditions, at that level), defending his claim that you couldn't get that kind of time resolution out of a PCM signal, rejecting plots showing otherwise, changing the claims and the goalposts repeatedly and at the same time accusing the people trying to have a dialog of "shifting their excuses", and repeated, exhaustive references to Dr. K's PhD and how publishing this in a conference record was the full endorsement of the organization, etc.

The people who are saying this do, I think, know better, too.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Woodinville on 2009-07-17 04:43:02
Oh, and the best joke of all this was that the intentionally ignorant over there think I was being rude.

Oh, baby, they have no idea what RUDE is. Apparently Gauss, Fourier, Fletcher and Zwicker are also rude, in any case.

It's just like creationists, they only accept confirmatory data.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: 2Bdecided on 2009-07-17 10:10:05
It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits.

Hopefully that would be simple enough to show how silly this is.

Cheers,
David.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: instaud on 2009-07-17 13:56:06
It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits.

Not only fun, but essential to prove his claim that CDDA is not good enough, and furthermore a test that his recommended 192kHz is the way to go. Without that, where is the relation between his artificial laboratory tests (5 years long and so very PhD-like theoretical), and reality of music reproduction?

Prof. Kunchur talks about the big picture which his results have to be viewed within. He himself writes in his FAQ about errors that are acceptable in consumer audio, but not in his experiment. So even if he found an error in CDDA, is it reason enough to replace it? He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: krabapple on 2009-07-17 15:03:49
He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.


Wait, Dr. Kunchur funded his own research??  Nice for those who can!

Actually the audio-related research reported in his three journal publications (http://www.physics.sc.edu/kunchur/Acoustics-papers.htm) appears to have been funded (well, "partially') intramurally  -- i.e., by his own university.  I don't see any indication of industry bias.  This is in contrast to the vast evil CONSPIRACY involving the AES, the low-end audio industry, Hydrogenaudio, the Freemasons, and our secret alien lizard overlords, as uncovered by certain brave Stereophile forum posters!     

Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Axon on 2009-07-17 18:30:51
Specifically, he plays one pulse, and then two half-amplitude pulses separated by 5 microseconds, and listeners with unimpaired hearing in good settings can hear the difference. Given that this is a comb filter with its first zero at 100kHz, being able to hear the difference at 20kHz isn't too surprising, now. So, this shows a mechanism for which purely in-band (i.e. dc - 20kHz) signal can allow the distinction. Let's see.... Down about .45dB is just at the DL.  In really good conditions, for direct, immediate comparison, with no level roving, etc. So, what did the doctor actually discover? Nothing that I can tell.
Um, no?

His experiments only involve pulses by implication - ie, that playing two time-misaligned sine waves is equivalent to one sine wave plus two rectangular pulses, or that a first-order-filtered square wave can be roughly treated as a square wave plus two exponentially decaying pulses. Neither of which actually involve pulses spaced 5us apart. Granted, everybody involved (including Dr. Kunchur) seem to be all too happy to reduce this down to the spaced pulses idea, but let's keep it real on what is actually in the papers.

Combing does occur with his speaker-alignment experiment but he goes to lengths to measure the headphone output and assert that the DL is not reached at any frequency, and his method appears more or less satisfactory to me (at least as far as this particular part of the procedure is concerned).

Quote
This, of course, after misdirection upon misdirection from the people 'defending' his work
Yeah, there's no excuse for that, but seriously, what did you expect out of a gaggle of audiophiles like that? It's not all that productive to drag sasaudio etal into a fight that really should just be between you and Kunchur.

Oh, and the best joke of all this was that the intentionally ignorant over there think I was being rude. Oh, baby, they have no idea what RUDE is. Apparently Gauss, Fourier, Fletcher and Zwicker are also rude, in any case. It's just like creationists, they only accept confirmatory data.
The impression I got was that Fourier (at the least) was a gasbag of a reasonably high order. I would not call him a role model.

It would be fun to repeat the experiment, but hook a CD recorder into the loop, testing if (proving that!) the "effect" was still audible after passing through 44.1kHz / 16-bits. Hopefully that would be simple enough to show how silly this is.
Curiously enough Kunchur never does run this control (ie, run the tests with 7khz sine instead of 7khz square, and/or 7+21khz sines). I'm unsure how important this is.

He says he is an audiophile, and that's the reason his answer is yes. I come to this conclusion because he has "experienced hearing loss ... from riding sports motorcycles", listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance. So the big picture for me is biased research which is funded by those having an interest in a certain result.


Wait, Dr. Kunchur funded his own research??  Nice for those who can!

Actually the audio-related research reported in his three journal publications (http://www.physics.sc.edu/kunchur/Acoustics-papers.htm) appears to have been funded (well, "partially') intramurally  -- i.e., by his own university.  I don't see any indication of industry bias.  This is in contrast to the vast evil CONSPIRACY involving the AES, the low-end audio industry, Hydrogenaudio, the Freemasons, and our secret alien lizard overlords, as uncovered by certain brave Stereophile forum posters!     
Also note that Hi-Fi Critic did a feature on Kunchur (as an example of one of the "good guys", meaning somebody who justifies/apologizes for high end audio).

There's an extremely critical point to be made related to all of this, but I'm saving it to address to the good doctor himself.
Title: 'Normalization' of PCM audio - subjectively benign?
Post by: Ethan Winer on 2009-07-17 19:04:40
Prof. Kunchur ... listens to cassettes and LPs, but on the other hand was not happy with the sound of his CD transport/DAC combination, so he needed a "jitter buster" appliance.

That says an awful lot IMO.

I wonder if he has any bass traps and other room treatment.

--Ethan