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Nice criticism of modern digital audio

M

martin griffith

Jan 1, 1970
0
I've been a long time meddler in (pro) audio, and over the last few
years I've been getting irritated by the stupid specmanship on audio
IC's.

My annoyance really started a year or so ago, when I was mucking
around with TI's PGA2310, a digitally controlled audio fader, with
the late Baz Porter, on a digitally controlled analogue parametric
equaliser. The PGA is a reasoable little IC for audio, but the stupid
way it was designed..... 0.5dB steps down to -95.5dB.

Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible. When I was a spotty
lad, -60dB was a tolerable level for distortion in a system!

On rec.audio.pro, ( a very busy NG,but very little wheat in the chaff
for me ) there was a thread about 384KHz PCM audio systems, and one
reply really hit the mark
I think it's this:
Message-ID: <[email protected]>
but I'm not too good on googling, so I post the whole message here,

Quote "

My name is Dan Lavry, and I hope you do not mind that I barge in. I do
want to say a few words on that 192Khz. As a designer, I have long
realized that faster sampling means less accuracy. You can get 1 bit
at a GHz, 8 bits at 100MHz, 12 bits at 30Mhz…. Nearly 24 bits at a few
Hz. Well, I am not suggesting to sample at 10Hz. We need to cover the
audio bandwidth. Going too slow is bad. Going too fast is also bad.
Let's face it, you want to say charge a cap, or settle an amp, and it
does dot happen at zero time. Nothing does. Weather you get there
exponentially or you have that step with some ringing… whatever it is,
if you wait till it settled you get it closer and more accurate. These
are practical limitations. The thing that gets me is that all the EE's
I talk to make jokes about 192Khz – a common one is "my dog can not
hear it". But many in the recording and mastering side are a part of a
different crowd. There seem to be a gap between the EE designer types
and the recording types. I am not suggesting that EE's are better or
worst. There have been times when the good ear is right, and the EE
learns new things. But there should be times such as right here and
now for the recording types to be open-minded. And folks, that 192KHz
is a crock.

I have wondered about how is it possible that a whole industry can go
in the wrong direction. After all, Few musical instruments will under
special cases yield a -50dB harmonic at 50Khz, and almost nothing at
100KHz. Experiments showing that special case required special mics
and gear. Most recording mics will not pick up a thing at 60khz… The
speakers will not play it, the ear will not hear it, no matter how big
the ego is!

So the sounds are negligible at those high frequencies, the mics will
not pick them, the speaker will not play it, and you can not hear it.
So how can such a crock take place? And let me assure you the folks
that made it happen at the semiconductor houses and workstation houses
did not include the EE department... I think it was and is about
making money. I talked to many engineers that stated privately that
they are not talking because they need their jobs! Other "did not want
to make waves", "stir the pot"…

So is everyone trying to cheat? Of course not. It started with 48Khz
better than 44.1, which is true. Also, while a bit excessive, 88.2 and
96KHz are in some cases a better compromise than 44.1. That is due to
the old FIR pre shooting (pre echo) that can happen under some cases
with a lot of processing. By the time you are at 60KHz it is so far
down you will not hear it, so 70Khz is a pretty reasonable place to
be. So after we have improved some as we went higher, the expectation
was than to continue upwards. Where does it stop? 384? Why not 1Ghz?
Oh well, the will be pretty bad, will it not?

But there is more to that than that "trendy upwards worked before so
lets continue". The common sense tells you that more is better. For
example, more pixels yield more detail. More bits gives better sample
accuracy. So if we take that analog wave, made out infinite points
connecting into a line, will we not benefit from a better "tracking"
of the wave? Well here is the news, and it is pretty old stuff, though
it may not be easy to grasp with simple common sense. More pixels for
better picture- yes! More bits for better accuracy (assuming
theoretical case - no noise) – yes! More sample density (higher sample
rate) for more accuracy –NO! This is the beauty of the sampling
theory. Nyquist did not say: We take more points and we will get
closer approximation. What he said – and it is a FUNDUMENTAL THEORY,
is that once we agree to deal with a limited bandwidth (called the
Nyquist bandwidth), all we need to do is sample at a tiny amount
greater than twice that bandwidth. This will yield 100% of the
waveform information in the data stream. We may need to filter a
signal (anti alaising) to make sure we do not have energy over
Nyquist, but than we are home free. Taking 4 times as many samples
does not yield 400% of the information. You can only have 100%! How do
you retrieve the information? You use a filter and it connects the
sample points in such a way that that you get the original wave shape.
A filter does not connect the dots (sample points) with straight
lines, or parabola… It recreates the original wave! You do not need to
help things with extra point in between. It buys you nothing!

I also see a lot of confusion regarding that Nyquist, upsampling,
oversampling, gradual filters… Some folks think that a 96KHz AD will
require a sharper anti aliasing filter than say 192KHz. This is
typically very wrong. The 96KHz or 192KHz AD refers to the OUTPUT RATE
of the converter. The antialias requirement is determined by the INPUT
SAMPLING RATE which is usually way beyond 192Khz. This days, most
modern AD's are running at input sampling rates of 3-12Mhz!!! DSD is
64fs and many mulibit IC's run even faster… So even with 50Khz audio,
Nyquist is so high the a gradual 3 pole will yield 120dB at the input
rate Nyquist. The outcome of the high rate modulator (input) is than
down-sampled to whatever – 44.1, 96, 192… Of course, when the sales
guys try to stick you with it they tell you need more bandwidth, but
they also ALL I saw regarding semiconductor and gear makers alike:
specs for 192KHz device are with A weighting – which states
(indirectly?) that you do not even need to measure flat to 20Khz. So
is it a crock? It is!

Theoretically, there is "no harm" in more points, and there is "no
added good". But as I pointed out, faster is less accurate! And yes,
you double the sample rate and the processing power requirement, and
so is the file size. These are serious draw backs! Don't you say you
do not care about file size: The DVD audio has 12 to 1 compression
(Dolby AC), We do not even get near a 1 to 1, and we want to push it
higher?

I realize that with all that reasoning and science and engineering,
someone is going to tell me that they hear it and like it. In fact,
someone told me that they still hear that high frequency in the
44.1KHz CD. I will not dignify that impossibility. If you hear some
distortion you like on the 44.1K CD, you did not need to go any faster
than 44.1KHz to generate it. I am not arguing against controlled
distortions (such as tube sound and what not). If you like it is fine.
It may be artistic decision. If you feel like you need to go to 1Mhz
than down to like it, fine. I think you are letting the gear control
you instead of the other way around, but fine! Just as long as I get
you to realize that you can get those distortions with a 44Khz… And we
do not all need to double the file size and processing power, and buy
new gear that is less accurate.

192 is a crock! 382 is a super crock! 88.2/96Khz is a bit excessive,
but not too far from a good rate. I too can glue a faster IC on the
board and make more money. My 192 DA prototype is not bad, but the
96KHz bits it by a lot.
Anyone telling you that more points will give better aproximation is
lacking lacking some know how.

Thanks for your patience.

Dan Lavry
Lavry Engineering
"
end quote
martin



"When all else fails, digitize everything, use fiber optic cable and enter a
whole new realm of problems."
<Found on the Rane tech page>
 
F

Frank Bemelman

Jan 1, 1970
0
martin griffith said:
I've been a long time meddler in (pro) audio, and over the last few
years I've been getting irritated by the stupid specmanship on audio
IC's.

My annoyance really started a year or so ago, when I was mucking
around with TI's PGA2310, a digitally controlled audio fader, with
the late Baz Porter, on a digitally controlled analogue parametric
equaliser. The PGA is a reasoable little IC for audio, but the stupid
way it was designed..... 0.5dB steps down to -95.5dB.

Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible. When I was a spotty
lad, -60dB was a tolerable level for distortion in a system!

Why is it stupid? You can still make the step from -60 to
infinity in the software that controls it. Perhaps there
are other applications (other than audio, but within audio
frequencies) where the -95.5db is welcome. Do you really
need smaller steps than 0.5 db? You can put 2 in series ;)

[snip]
 
A

Allan Herriman

Jan 1, 1970
0
Why is it stupid? You can still make the step from -60 to
infinity in the software that controls it. Perhaps there
are other applications (other than audio, but within audio
frequencies) where the -95.5db is welcome. Do you really
need smaller steps than 0.5 db? You can put 2 in series ;)

Martin's application was a parametric equaliser rather than a straight
fader.
I guess the eq was quite sensitive to the fader step size.

Regards,
Allan.
 
B

Ban

Jan 1, 1970
0
martin griffith wrote:
|| I've been a long time meddler in (pro) audio, and over the last few
|| years I've been getting irritated by the stupid specmanship on audio
|| IC's.
||
|| My annoyance really started a year or so ago, when I was mucking
|| around with TI's PGA2310, a digitally controlled audio fader, with
|| the late Baz Porter,

I love the neteq, sad to hear Mr. Porter has passed away. Do you still
continue this work and do you need some support? I'm ready to collaborate!
Drop me a mail in case.

snipped quote
||
|| "When all else fails, digitize everything, use fiber optic cable and
|| enter a whole new realm of problems."
|| <Found on the Rane tech page>

My feeling is that despite a few exaggerated specs digital audio has finally
reached an acceptable quality level, allowing all-digital amps and a
reproduction superiour to analog gear. This is mainly due to 24bit and 88.2
or 96kHz sampling rate. Hopefully the high-end SACD will take off within the
next years in contrast to this awful MP3 hype, the kids are spreading.
As much as you don't need the accuracy of a 99.0 to 99.5dB attenuation or
the 24 vs. 16bit, it doesn't cost anything extra and it is nice to have it
for maybe some unexpected use.

In an all digital amp also the volume pot is realized digitally and with low
settings you would usually loose information because if you have speakers
with high sensitivity you might need to adjust your volume pot to -60dB and
10bits are gone, so the remaining 14 will be better than only 6 in a 16bit
system, and as you see here we better had 32bits available.

With high sampling rates I do agree, but again a counter example: a constant
directivity horn requires an equalisation of +24dB at 20kHz of shelving
6dB/Octave. Now with a 44.1 sampling rate you will have problems to
implement this digitally and even at 96kHz it is done only approximatly. So
there is always a pro and a con.
Todays progress in DSP will get boosted with newer standards and we have to
create more demanding situations to keep the engine running, as Bill Gates
does it with Windows.
Keep on rocking...
 
I

Ian Stirling

Jan 1, 1970
0
martin griffith said:
I've been a long time meddler in (pro) audio, and over the last few
years I've been getting irritated by the stupid specmanship on audio
IC's.

My annoyance really started a year or so ago, when I was mucking
around with TI's PGA2310, a digitally controlled audio fader, with
the late Baz Porter, on a digitally controlled analogue parametric
equaliser. The PGA is a reasoable little IC for audio, but the stupid
way it was designed..... 0.5dB steps down to -95.5dB.

Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible. When I was a spotty
lad, -60dB was a tolerable level for distortion in a system!

A volume control can need that dynamic range.
Say 110dB on the top end, that's only 50db on the bottom.
Distortion is different.
I want to be able to adjust volume from somewhere around 90db, all the way
down to almost audible in a quiet room (say 5-10db).
 
R

raymund hofmann

Jan 1, 1970
0
I agree.
There are not that many applications for intelligence as one might
think. (quote from Dilbert)

Raymund Hofmann
 
T

Tim Shoppa

Jan 1, 1970
0
martin griffith said:
My annoyance really started a year or so ago, when I was mucking
around with TI's PGA2310, a digitally controlled audio fader, with
the late Baz Porter, on a digitally controlled analogue parametric
equaliser. The PGA is a reasoable little IC for audio, but the stupid
way it was designed..... 0.5dB steps down to -95.5dB.

Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible. When I was a spotty
lad, -60dB was a tolerable level for distortion in a system!

The PGA2310 is a part. The designer of the system including the part
is responsible for using the part appropriately.

You may as well complain that a resistor manufacturer would let you put
two resistors together and make a -95.5dB divider, and that if they
were a responsible resistor company they wouldn't sell that combination :).

Tim.
 
T

Tim Hubberstey

Jan 1, 1970
0
Frank said:
Do you really
need smaller steps than 0.5 db? You can put 2 in series ;)

I know there's a smiley there, but still . . .

That will give you a fader with a theoretical range of 0 to -191 dB, but
still in 0.5 dB steps. Decibels in series add, they don't multiply. To
get finer steps, you need a fader with finer steps.

I believe Martin's point is that the chip has includes all the circuitry
to give 192 steps, so why not space them over a reasonable range instead
of wasting them to get a "better" spec?
 
D

Dbowey

Jan 1, 1970
0
martin posted, among other things:
...The PGA is a reasoable little IC for audio, but the stupid
way it was designed..... 0.5dB steps down to -95.5dB.

Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible.

That coming from an Engineer appears shallow; why use finer steps when one
cannot perceive the change of even 0.5 db? I imagine the ability to attenuate
to -95.5dB enables it's use as a mute device.

The rest of your post is mostly rant. Try to focus a bit.

Don
 
W

Walter Harley

Jan 1, 1970
0
martin griffith said:
Who on earth would want to fade something that low. Most mixing
console faders tend to cut to infinity at -60dB... dumb, why not finer
steps, down to -60. It just seems so sensible. When I was a spotty
lad, -60dB was a tolerable level for distortion in a system!

Most of the consoles I've worked with don't have 0 as the top of the faders.
Rather, they go from somewhere around +15dB down to somewhere around -60dB
and thence -infinity.
 
W

Walter Harley

Jan 1, 1970
0
Dbowey said:
That coming from an Engineer appears shallow; why use finer steps when one
cannot perceive the change of even 0.5 db? I imagine the ability to attenuate
to -95.5dB enables it's use as a mute device.

Realistically, you need a -inf setting (just like you get from a real pot) -
you want to be able to turn something OFF, not just down to the point where
it's theoretically inaudible, because you might be adding a bunch of
nominally "off" channels together. But if -inf is really -120dB (or
whatever the i/o bleedthrough amounts to), it doesn't follow that the next
notch up has to be one notch higher than -120. It is fine to go from "the
lowest thing you need to fade to" down to "off".

If I were designing a digital audio attenuator (as opposed to a digital
potentiometer, which is the OP's problem), I'd want it to have an uneven
scale: fine gradations around 0, down to about -40, and then coarser
gradations down to -inf. That would correspond to how pro mixing desks seem
to work: the physical distance between -30 and -40 on one desk I regularly
work is about the same as the distance between 0 and +3, for instance.

0.5dB is a noticeable jump if you're trying to do a mix. (I know, the OP
was about a parametric EQ, not a channel fader.) I guess that just as with
pitch, it's easier to tell slight level changes in comparison to a fixed
reference than by themselves. If you need to get the lead vocal just a tad
hotter than the guitar, 0.5dB gradations are awkwardly coarse - workable but
undesirable.
 
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