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Topic: 'Normalization' of PCM audio - subjectively benign? (Read 140815 times) previous topic - next topic
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'Normalization' of PCM audio - subjectively benign?

Reply #75
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.)

'Normalization' of PCM audio - subjectively benign?

Reply #76
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.

'Normalization' of PCM audio - subjectively benign?

Reply #77
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.

'Normalization' of PCM audio - subjectively benign?

Reply #78
hi all (again) 

why cd players have LPF in the output? (seems off topic but it's not)

'Normalization' of PCM audio - subjectively benign?

Reply #79

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.

'Normalization' of PCM audio - subjectively benign?

Reply #80
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.

'Normalization' of PCM audio - subjectively benign?

Reply #81
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 (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.

'Normalization' of PCM audio - subjectively benign?

Reply #82
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.



You were saying?

'Normalization' of PCM audio - subjectively benign?

Reply #83
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.

'Normalization' of PCM audio - subjectively benign?

Reply #84
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.

'Normalization' of PCM audio - subjectively benign?

Reply #85
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.

'Normalization' of PCM audio - subjectively benign?

Reply #86

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.)

'Normalization' of PCM audio - subjectively benign?

Reply #87
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.

'Normalization' of PCM audio - subjectively benign?

Reply #88


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.

'Normalization' of PCM audio - subjectively benign?

Reply #89
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.


'Normalization' of PCM audio - subjectively benign?

Reply #91
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"?

'Normalization' of PCM audio - subjectively benign?

Reply #92
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).

'Normalization' of PCM audio - subjectively benign?

Reply #93
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
and back to the beginning of the thread if needed.

regards.

'Normalization' of PCM audio - subjectively benign?

Reply #94
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.

'Normalization' of PCM audio - subjectively benign?

Reply #95
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

'Normalization' of PCM audio - subjectively benign?

Reply #96

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

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.

'Normalization' of PCM audio - subjectively benign?

Reply #97


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
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 (!)

'Normalization' of PCM audio - subjectively benign?

Reply #98
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.

'Normalization' of PCM audio - subjectively benign?

Reply #99
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?