subject:"Re\: \[music\-dsp\] Dither video and articles"


On 2/10/15 8:49 AM, Didier Dambrin wrote:
What are you talking about - why would phase not matter? It's 
extremely important (well, phase relationship between neighboring 
partials).





well, it's unlikely you'll be able to hear the difference between this:

x(t) = cos(wt) - 1/3*cos(3wt) + 1/5*cos(5wt) - 1/7*cos(7wt)

and this:

x(t) = cos(wt) + 1/3*cos(3wt) + 1/5*cos(5wt) + 1/7*cos(7wt)

yet the waveshapes are much different.

so if you have MATLAB or Octave, try this file out and see what you can 
hear.  look at the waveforms and see how different they are.


%
%   square_phase.m
%
%   a test to see if we can really hear phase changes
%   in the harmonics of a Nyquist limited square wave.
%
%   (c) 2004 r...@audioimagination.com mailto:r...@audioimagination.com
%

if ~exist('Fs', 'var')
 Fs = 44100  % sample rate, Hz
end

if ~exist('f0', 'var')
 f0 = 110.25 % fundamental freq, Hz
end

if ~exist('tone_duration', 'var')
 tone_duration = 2.0 % seconds
end

if ~exist('change_rate', 'var')
 change_rate = 1.0   % Hz
end

if ~exist('max_harmonic', 'var')
 max_harmonic = floor((Fs/2)/f0) - 1
end

if ~exist('amplitude_factor', 'var')
 amplitude_factor = 0.25 % this just keeps things from 
clipping

end

if ~exist('outFile', 'var')
 outFile = 'square_phase.wav'
end


   % make sure we don't uber-Nyquist 
anything

max_harmonic = min(max_harmonic, floor((Fs/2)/f0)-1);

t = linspace((-1/4)/f0, tone_duration-(1/4)/f0, Fs*tone_duration+1);

detune = change_rate;

x = cos(2*pi*f0*t);  % start with 1st harmonic

n = 3;   % continue with 3rd harmonic while 
(n = max_harmonic)
 if ((n-1) == 4*floor((n-1)/4))   % lessee if it's an even or 
odd term

 x = x + (1/n)*cos(2*pi*n*f0*t);
  else
 x = x - (1/n)*cos(2*pi*(n*f0+detune)*t);
 detune = -detune;% comment this line in an see some
 end  % funky intermediate waveforms
 n = n + 2;   % continue with next odd harmonic
end

x = amplitude_factor*x;

% x = sin((pi/2)*x);   % toss in a little soft clipping

plot(t, x);  % see
sound(x, Fs);% hear
wavwrite(x, Fs, outFile);% remember






16 bits is just barely enough for high-quality audio.


So to you, that Pono player isn't snake oil?


well, Vicki is the high-res guru here.

i certainly don't think we need 24-bit and 192 kHz just for listening to 
music in our living room.  but for intermediate nodes (or intermediate 
files), 24-bit is not a bad idea.  and if you have space or bandwidth to 
burn, why not, say, 96 kHz.  then people can't complain about the 
scrunching of the bell curve near Nyquist they get with cookbook EQ.


for a high-quality audio and music signal processor, i think that 16-bit 
pre-emphasized files (for sampled sounds or waveforms) is the minimum i 
want, 16-bit or more ADC and DAC, and 24-bit internal nodes for 
processing is the minimum i would want to not feel cheap about it.  if 
i were to use an ADI Blackfin (i never have) to process music and 
better-than-voice audio, i would end up doing a lot of double-precision 
math.


BTW, at this: 
http://www.aes.org/events/125/tutorials/session.cfm?code=T19 i 
demonstrated how good 7-bit audio sounds in a variety of different 
formats, including fixed, float (with 3 exponent bits and 4 mantissa 
bits), and block floating point (actually that was 7.001 bits per 
sample), dithered and not, noise-shaped and not.  but i still wouldn't 
want to listen to 7-bit audio if i had CD.


well dithered and noise-shaped 16-bits at 44.1 kHz is good enough for 
me.  i might not be able to hear much wrong with 128 kbit/sec MP3, but i 
still like CD audio better.





Besides, if it had mattered so much, non-linear (mu/A-law) encoding 
could have applied to 16bit as well..




naw, then you get a sorta noise amplitude modulation with a signal of 
roughly constant amplitude.  and there are much better ways to do 
optimal bit reduction than companding.  companding is a quick and easy 
way they did it back in the old Bell System days.  and, even in 
companding, arcsinh() and sinh() would be smoother mapping than either 
mu or A-law.



--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



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Re: [music-dsp] Dither video and articles

Interestingly, I wasn't gonna suggest that a possible cause could have been 
a compressor built-in the soundcard, because.. why would a soundcard even do 
that..


However.. I've polled some people in our forum with this same test, and one 
guy could hear it. But it turns out that he owns an X-Fi, and it does 
feature automatic gain compensation, which was on for him. Owning the same 
soundcard, I turned it on, and yes, that made the noise at -80dB rather 
clear.


I'm not saying it's what's happening for you, but are you 100% sure of 
everything the signal goes through in your system?



This said, the existence of a built-in compressor in a soundcard.. that 
alone might be a point for dithering, if the common end listener leaves that 
kind of thing on.





-Message d'origine- 
From: Andrew Simper

Sent: Tuesday, February 10, 2015 6:52 AM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

Hi Didier,

I count myself as having good hearing, I always wear ear protection at
any gigs / loud events and have always done so. My hearing is very
important to me since it is essential for my livelihood.

I made a new test, a 440 hz sine wave with three 0.25 second white
noise bursts -66 dB, -72 dB and -75 dB below the sine (which is at -6
dBFS). I can hear the first one very clearly, then just hear the
second one. I can't actually hear the hiss of the third one but I can
hear the amplitude of the sine wave fractionally lowering when the
actual amplitude of the test sine remains constant, I don't know why
this is but that's how I hear it.

You will clearly see where the white noise bursts are if you use some
sort of FFT display, but please just have a listen first and try and
pick where each (3 total) are in the file:

www.cytomic.com/files/dsp/border-of-hearing.wav

For the other way around, a constant noise file and with bursts of 440
hz sine waves, the sine has to be very loud before I can hear it, up
around -28 dB from memory. Noise added to a sine wave is much easier
to pick, which is why I think low pass filtered tones that are largely
sine like in nature are the border case for dither.

All the best,

Andy


-- cytomic -- sound music software --


On 10 February 2015 at 10:56, Didier Dambrin di...@skynet.be wrote:

I'm having a hard time finding anyone who could hear past the -72dB noise,
here around.

Really, either you have super-ears, or the cause is (technically) 
somewhere
else. But it matters, because the whole point of dithering to 16bit 
depends

on how common that ability is.




-Message d'origine- From: Andrew Simper
Sent: Saturday, February 07, 2015 2:08 PM

To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

On 7 February 2015 at 03:52, Didier Dambrin di...@skynet.be wrote:


It was just several times the same fading in/out noise at different
levels,
just to see if you hear quieter things than I do, I thought you'd have
guessed that.

https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing
(0dB, -36dB, -54dB, -66dB, -72dB, -78dB)

Here if I make the starting noise annoying, then I hear the first 4 
parts,

until 18:00. Thus, if 0dB is my threshold of annoyance, I can't hear
-72dB.

So you hear it at -78dB? Would be interesting to know how many can, and 
if

it's subjective or a matter of testing environment (the variable already
being the 0dB annoyance starting point)



Yep, I could hear all of them, and the time I couldn't hear the hiss
any more as at the 28.7 second mark, just before the end of the file.
For reference this noise blast sounded much louder than the bass tone
that Nigel posted when both were normalised, I had my headphones amp
at -18 dB so the first noise peak was loud but not uncomfortable.

I thought it was an odd test since the test file just stopped before I
couldn't hear the LFO amplitude modulation cycles, so I wasn't sure
what you were trying to prove!

All the best,

Andy





-Message d'origine- From: Andrew Simper
Sent: Friday, February 06, 2015 3:21 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.

All the best,

Andrew Simper



On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:



On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:



Just out of curiosity, until which point

Re: [music-dsp] Dither video and articles


On 2/9/15 10:19 PM, Nigel Redmon wrote:

But it matters, because the whole point of dithering to 16bit depends on how 
common that ability is.

Depends on how common? I’m not sure what qualifies for common, but if it’s 1 in 
100, or 5 in 100, it’s still a no-brainer because it costs nothing, effectively.


i have had a similar argument with Andrew Horner about tossing phase 
information outa the line spectrum of wavetables for wavetable 
synthesis.  why bother to do that?  why not just keep the phase 
information and the waveshape when it costs nothing to do it.


regarding dithering and quantization, if it were me, for 32-bit or 
24-bit fixed-point *intermediate* values (like multiple internal nodes 
of an algorithm), simply because of the cost of dithering, i would 
simply use fraction saving, which is 1st-order noise shaping with a 
zero at DC, and not dither.  or just simply round, but the fraction 
saving is better and just about as cheap in computational cost.


but for quantizing to 16 bits (like for mastering a CD or a 16-bit 
uncompressed .wav or .aif file), i would certainly dither and optimally 
noise-shape that.  it costs a little more, but like the wavetable phase, 
once you do it the ongoing costs are nothing.  and you have better data 
stored in your lower resolution format.  so why not?


16 bits is just barely enough for high-quality audio.  and it wouldn't 
have been if Stanley Lipshitz and John Vanderkooy and Robert Wanamaker 
didn't tell us in the 80's how to extract another few more dB outa the 
dynamic range of the 16-bit word.  they really rescued the 80 minute Red 
Book CD.


--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



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links
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Re: [music-dsp] Dither video and articles

What are you talking about - why would phase not matter? It's extremely 
important (well, phase relationship between neighboring partials).





16 bits is just barely enough for high-quality audio.


So to you, that Pono player isn't snake oil?

Besides, if it had mattered so much, non-linear (mu/A-law) encoding could 
have applied to 16bit as well..






-Message d'origine- 
From: robert bristow-johnson

Sent: Tuesday, February 10, 2015 2:37 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Dither video and articles

On 2/9/15 10:19 PM, Nigel Redmon wrote:
But it matters, because the whole point of dithering to 16bit depends on 
how common that ability is.
Depends on how common? I’m not sure what qualifies for common, but if it’s 
1 in 100, or 5 in 100, it’s still a no-brainer because it costs nothing, 
effectively.


i have had a similar argument with Andrew Horner about tossing phase
information outa the line spectrum of wavetables for wavetable
synthesis.  why bother to do that?  why not just keep the phase
information and the waveshape when it costs nothing to do it.

regarding dithering and quantization, if it were me, for 32-bit or
24-bit fixed-point *intermediate* values (like multiple internal nodes
of an algorithm), simply because of the cost of dithering, i would
simply use fraction saving, which is 1st-order noise shaping with a
zero at DC, and not dither.  or just simply round, but the fraction
saving is better and just about as cheap in computational cost.

but for quantizing to 16 bits (like for mastering a CD or a 16-bit
uncompressed .wav or .aif file), i would certainly dither and optimally
noise-shape that.  it costs a little more, but like the wavetable phase,
once you do it the ongoing costs are nothing.  and you have better data
stored in your lower resolution format.  so why not?

16 bits is just barely enough for high-quality audio.  and it wouldn't
have been if Stanley Lipshitz and John Vanderkooy and Robert Wanamaker
didn't tell us in the 80's how to extract another few more dB outa the
dynamic range of the 16-bit word.  they really rescued the 80 minute Red
Book CD.

--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



--
dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp 
links

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Re: [music-dsp] Dither video and articles

I'm talking about simple initial phase offsets, nothing dynamic. It's an old 
subject, you will find it back as ghost thone in this mailing list, with 
audio examples.


I'll redo an audio demo if you insist, but simply randomizing the *initial* 
(yes, nothing dynamic) phases of all partials of a sawtooth, will give a 
pretty distinctive metallic tone, absolutely nothing like a pure sawtooth, 
and only differing in partial phases.





-Message d'origine- 
From: robert bristow-johnson

Sent: Tuesday, February 10, 2015 7:47 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Dither video and articles

On 2/10/15 1:22 PM, Didier Dambrin wrote:
Of course, a lot of visually different waveshapes sound the same, as soon 
as the phase relationship between neighboring partials is shifted by the 
same amount.

they can be shifted by *any* amount, as long as it's static.

in fact, what do you mean by same amount?  same amount of time?  then
that's just a delay.

same amount of phase?  well that *does* change the waveshape, but it can
be any amount of phase for a perfectly periodic waveform.  when things
get less than perfectly periodic, then you have changing harmonic
coefficients, both in amplitude and phase.



That doesn't mean it's always the case


i agree.  if the phase changes rapidly enough, you'll hear it as a
detuned or slightly non-harmonic partial.

i can't argue Andrew Horner's case for him (he just didn't think he
needed to deal with changing relative phases in all of his wavetable
synthesis papers he had in the JAES).  i know you can construct all-pass
filters with long delay times inside (and sufficient feedback
coefficient) and you'll *definitely* hear a difference.  APFs only
change the phase and nothing else.

and my argument to Andrew was that it costs nothing to preserve the
phase in wavetable synthesis, so why not?  my own work (which is now
about 2 and 3 decades old) didn't even use what is commonly called the
heterodyne oscillator to get the wavetables.  i yanked this
time-domain waveforms directly outa the time-domain data.  Andrew would
do something like a sinusoidal modeling analysis, get both amplitude and
phase of each harmonic, and then throw the phase away before creating
the wavetables.

and I've once posted here examples of how shifting the phase of 1 harmonic 
of a sawtooth sounded very different.

I think you were even part of the debate.


probably.  perhaps i posted the same MATLAB file for discussion.

Pretty easy to check the obvious difference between a pure low sawtooth, 
and the same sawtooth with all partials starting at random phases.




all partials?  it's a bandlimited saw, no?  the harmonic numbers stop
at some finite number.

well, it's a (bandlimited) square wave in the example below and the
partials are changing phase in some reasonable goofy manner.  some
partials are slightly detuned up and others are slightly detuned down.
and both in such a way that the waveform slowly changes from square wave
to something unrecognizable and slowly back to square.  and, if your
playback system is nice and linear, it's unlikely you'll hear it do that
(and you keep the number of harmonics low, don't uber-Nyquist it).

it can be rewritten to do it for saw.

BTW, because of word-wrapping that i cannot turn off, be sure to unwrap
some of the comment lines in the MATLAB program.

--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



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Re: [music-dsp] Dither video and articles

Of course 24bit isn't a bad idea for intermediate files, but 32bit float is 
a better idea, even just because you don't have to normalize  store gain 
information that pretty much no app will read from the file. And since the 
price of storage is negligible these days..





-Message d'origine- 
From: robert bristow-johnson

Sent: Tuesday, February 10, 2015 6:11 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Dither video and articles


i certainly don't think we need 24-bit and 192 kHz just for listening to
music in our living room.  but for intermediate nodes (or intermediate
files), 24-bit is not a bad idea. 


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Re: [music-dsp] Dither video and articles

So to you, that Pono player isn't snake oil?

It's more the 192kHz sampling rate that renders the Pono player into snake
oil territory. The extra bits probably aren't getting you much, but the
ridiculous sampling rate can only *hurt* audio quality, while consuming
that much more battery and storage.
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Re: [music-dsp] Dither video and articles


On 2/10/15 1:30 PM, Didier Dambrin wrote:
Of course 24bit isn't a bad idea for intermediate files, but 32bit 
float is a better idea, even just because you don't have to normalize 
 store gain information that pretty much no app will read from the 
file. And since the price of storage is negligible these days..


can't disagree with that.  now, even with float, you can dither and 
noise shape the quantization (from double to single-precision floats), 
but the code to do so is more difficult.  and i dunno *what* to do if 
adding your dither causes, for a single sample, the exponent to change.  
it's kinda messy.  i guess you just accept that this particular sample 
will not be perfectly dithered correctly and, whatever quantization 
error *does* result, use that in the noise-shaping feedback.


--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



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Re: [music-dsp] Dither video and articles

Of course, a lot of visually different waveshapes sound the same, as soon as 
the phase relationship between neighboring partials is shifted by the same 
amount.


That doesn't mean it's always the case and I've once posted here examples of 
how shifting the phase of 1 harmonic of a sawtooth sounded very different.

I think you were even part of the debate.
Pretty easy to check the obvious difference between a pure low sawtooth, and 
the same sawtooth with all partials starting at random phases.






-Message d'origine- 
From: robert bristow-johnson

Sent: Tuesday, February 10, 2015 6:11 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Dither video and articles

On 2/10/15 8:49 AM, Didier Dambrin wrote:
What are you talking about - why would phase not matter? It's extremely 
important (well, phase relationship between neighboring partials).





well, it's unlikely you'll be able to hear the difference between this:

x(t) = cos(wt) - 1/3*cos(3wt) + 1/5*cos(5wt) - 1/7*cos(7wt)

and this:

x(t) = cos(wt) + 1/3*cos(3wt) + 1/5*cos(5wt) + 1/7*cos(7wt)

yet the waveshapes are much different.

so if you have MATLAB or Octave, try this file out and see what you can
hear.  look at the waveforms and see how different they are.

%
%   square_phase.m
%
%   a test to see if we can really hear phase changes
%   in the harmonics of a Nyquist limited square wave.
%
%   (c) 2004 r...@audioimagination.com mailto:r...@audioimagination.com
%

if ~exist('Fs', 'var')
 Fs = 44100  % sample rate, Hz
end

if ~exist('f0', 'var')
 f0 = 110.25 % fundamental freq, Hz
end

if ~exist('tone_duration', 'var')
 tone_duration = 2.0 % seconds
end

if ~exist('change_rate', 'var')
 change_rate = 1.0   % Hz
end

if ~exist('max_harmonic', 'var')
 max_harmonic = floor((Fs/2)/f0) - 1
end

if ~exist('amplitude_factor', 'var')
 amplitude_factor = 0.25 % this just keeps things from
clipping
end

if ~exist('outFile', 'var')
 outFile = 'square_phase.wav'
end


   % make sure we don't uber-Nyquist
anything
max_harmonic = min(max_harmonic, floor((Fs/2)/f0)-1);

t = linspace((-1/4)/f0, tone_duration-(1/4)/f0, Fs*tone_duration+1);

detune = change_rate;

x = cos(2*pi*f0*t);  % start with 1st harmonic

n = 3;   % continue with 3rd harmonic while
(n = max_harmonic)
 if ((n-1) == 4*floor((n-1)/4))   % lessee if it's an even or
odd term
 x = x + (1/n)*cos(2*pi*n*f0*t);
  else
 x = x - (1/n)*cos(2*pi*(n*f0+detune)*t);
 detune = -detune;% comment this line in an see some
 end  % funky intermediate waveforms
 n = n + 2;   % continue with next odd harmonic
end

x = amplitude_factor*x;

% x = sin((pi/2)*x);   % toss in a little soft clipping

plot(t, x);  % see
sound(x, Fs);% hear
wavwrite(x, Fs, outFile);% remember






16 bits is just barely enough for high-quality audio.


So to you, that Pono player isn't snake oil?


well, Vicki is the high-res guru here.

i certainly don't think we need 24-bit and 192 kHz just for listening to
music in our living room.  but for intermediate nodes (or intermediate
files), 24-bit is not a bad idea.  and if you have space or bandwidth to
burn, why not, say, 96 kHz.  then people can't complain about the
scrunching of the bell curve near Nyquist they get with cookbook EQ.

for a high-quality audio and music signal processor, i think that 16-bit
pre-emphasized files (for sampled sounds or waveforms) is the minimum i
want, 16-bit or more ADC and DAC, and 24-bit internal nodes for
processing is the minimum i would want to not feel cheap about it.  if
i were to use an ADI Blackfin (i never have) to process music and
better-than-voice audio, i would end up doing a lot of double-precision
math.

BTW, at this:
http://www.aes.org/events/125/tutorials/session.cfm?code=T19 i
demonstrated how good 7-bit audio sounds in a variety of different
formats, including fixed, float (with 3 exponent bits and 4 mantissa
bits), and block floating point (actually that was 7.001 bits per
sample), dithered and not, noise-shaped and not.  but i still wouldn't
want to listen to 7-bit audio if i had CD.

well dithered and noise-shaped 16-bits at 44.1 kHz is good enough for
me.  i might not be able to hear much wrong with 128 kbit/sec MP3, but i
still like CD audio better.




Besides, if it had mattered so much, non-linear (mu/A-law) encoding could 
have applied to 16bit as well..




naw, then you get a sorta noise amplitude modulation with a signal of
roughly constant amplitude.  and there are much better ways to do
optimal bit reduction than companding.  companding is a quick and easy
way they did it back in the old Bell System days.  and, even in
companding, arcsinh() and sinh() would be smoother mapping

Re: [music-dsp] Dither video and articles


On 2/10/15 1:51 PM, Ethan Duni wrote:

So to you, that Pono player isn't snake oil?

It's more the 192kHz sampling rate that renders the Pono player into snake
oil territory. The extra bits probably aren't getting you much, but the
ridiculous sampling rate can only *hurt* audio quality, while consuming
that much more battery and storage.


that's interesting.  why does higher-than-needed sample rate hurt audio 
quality?  might not be necessary, but how does it make it worse 
(excluding the increased computational burden)?  i always think that 
analog (or continuous-time) is like having an infinite sample rate.



--

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Imagination is more important than knowledge.



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Re: [music-dsp] Dither video and articles

2015-02-10 Thread gwenhwyfaer

On 10/02/2015, Didier Dambrin di...@skynet.be wrote:
 Pretty easy to check the obvious difference between a pure low sawtooth, and

 the same sawtooth with all partials starting at random phases.

Ah, this again? Good times. I remember playing. I made 7 sawtooth
waves with random (static) phases and one straightforward sawtooth
wave, with all partials in phase. I just listened to it again, to
check my memory. On a half-decent pair of headphones, the difference
between the all-partials-in-phase sawtooth and the random-phase ones
is readily audible, but it was rather harder to tell the difference
between the various random-phase waves; they all kind of sounded
pulse-wavey. On a pair of speakers through the same amp and soundcard,
though, I can still *jst about* pick out the in-phase sawtooth -
but I couldn't confidently tell the difference between the 7 other
waves. Which I'm guessing has something to do with the difference
between the fairly one-dimensional travel of sound from headphone to
ear, vs the bouncing-in-from-all-kinds-of-directions speaker-ear
journey.

I'm only a data point, though, so I'm not brave enough to actually
conclude anything. At least, not any more. ;)
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Re: [music-dsp] Dither video and articles

2015-02-10 Thread Tom Duffy


The only comment in that page that actually tells the story is buried:

--
Different media, different master

I've run across a few articles and blog posts that declare the virtues 
of 24 bit or 96/192kHz by comparing a CD to an audio DVD (or SACD) of 
the 'same' recording. This comparison is invalid; the masters are 
usually different.



The benefit end users get from Pono / Hi resolution files is exactly
this - the master was prepared without the usual requirements for
radio-play-ready compression or filtering.

Sure you can do the same in 44.1kHz/16bit or MP3 or MP4, but packaging
is everything, and it's a lot easier to market something that requires
a bigger file as being better.

Everyone gets hung up on the but you can't hear the extra bits / extra
FS but ignore the advantage of less GI-GO.

Tom.


On 2/10/2015 12:46 PM, Ethan Duni wrote:
why does higher-than-needed sample rate hurt audio quality?
might not be necessary, but how does it make it worse (excluding
the increased computational burden)?

The danger is that you are now including a bunch of out-of-band content in
your output signal, which can be transformed into in-band aliasing by any
nonlinearities in your playback chain. It's generally not a big deal, but
it is measurable and does hurt quality:

http://xiph.org/~xiphmont/demo/neil-young.html

This is an excellent example of the tension between audiophile
perfectionism (i.e., more sample rate must always be at least as good,
because digital audio is some kind of terrifying bogeyman) and actual
engineering quality control (i.e., overspec-ing systems drives up costs,
compromises the quality in other components, and generally creates more
headaches than it solves).

E


On Tue, Feb 10, 2015 at 10:54 AM, robert bristow-johnson 
r...@audioimagination.com wrote:

On 2/10/15 1:51 PM, Ethan Duni wrote:

So to you, that Pono player isn't snake oil?

It's more the 192kHz sampling rate that renders the Pono player into snake
oil territory. The extra bits probably aren't getting you much, but the
ridiculous sampling rate can only *hurt* audio quality, while consuming
that much more battery and storage.


that's interesting.  why does higher-than-needed sample rate hurt audio
quality?  might not be necessary, but how does it make it worse (excluding
the increased computational burden)?  i always think that analog (or
continuous-time) is like having an infinite sample rate.


--

r b-j  r...@audioimagination.com

Imagination is more important than knowledge.



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Re: [music-dsp] Dither video and articles

2015-02-10 Thread Tom Duffy

So you like the bar being raised, but not the way that Neil Young has
attempted?

Whether the higher resolution actually degrades the quality is a
topic up for future debate.

From the ponomusic webpage:
...and now, with the PonoPlayer, you can finally feel the master in all
its glory, in its native resolution, CD quality or higher, the way the
artist made it, exactly

Even they are not saying it has to be higher than CD quality, just
that it has to have been made well in the first place.

I don't get why so many people are trying to paint this as
a snake oil pitch.

---
Tom.

On 2/10/2015 1:13 PM, Ethan Duni wrote:
I'm all for releasing stuff from improved masters. There's a trend in my
favorite genre (heavy metal) to rerelease a lot of classics in full
dynamic range editions lately. While I'm not sure that all of these
releases really sound much better (how much dynamic range was there in an
underground death metal recording from 1991 anyway?) I like the trend.
These are regular CD releases, no weird formats (demonstrating that such is
not required to sell the improved master releases).

But the thing is that you often *can* hear the extra sampling frequency -
in the form of additional distortion. It sounds, if anything, *worse* than
a release with an appropriate sample rate! Trying to sell people on better
audio, and then giving them a bunch of additional intermodulation
distortion is not a justified marketing ploy, it's outright deceptive and
abusive. This is working from the assumption that your customers are
idiots, and that you should exploit that to make money, irrespective of
whether audio quality is harmed or not. The fact the Neil Young is himself
one of the suckers renders this less objectionable, but only slightly.
Anyway Pono is already a byword for audiophile snake oil so hopefully the
damage will mostly be limited to the bank accounts of Mr. Young and his
various financial backers in this idiocy. Sounds like the product is a real
dog in industrial design terms anyway (no hold button, awkward shape,
etc.). Good riddance...

E
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Re: [music-dsp] Dither video and articles

2015-02-10 Thread Michael Gogins

What I am interested in, regarding this discussion, is quite specific.
I make computer music using Csound, and usually using completely
synthesized sound, and so far only in stereo. Csound can run at any
sample rate, can output floating-point soundfiles, and can dither. My
sounds are not necessarily simple and cover the whole frequency range
and a wide dynamic range.

My only real question is, since the signal path right up to the point
where the soundfile is written is likely to be the same in all cases,
what kind of differences if any can I try to hear in CD audio versus
say 96 KHz floating-point?

These differences (if any) will be caused by the different Csound
sampling rate, the different soundfile sample word size/dynamic range,
and of course the different things that might happen to these two
kinds of soundfiles on their way out of a high-quality
DAC/amplifier/monitor speaker rig.

At times, my pieces have fortunately been presented in nice quiet
concert halls with really good amplifiers and speakers. I have also
been able to listen a few times in high-end recording studios designed
for this kind of music (this is a very different listening
experience).

Regards,
Mike

-
Michael Gogins
Irreducible Productions
http://michaelgogins.tumblr.com
Michael dot Gogins at gmail dot com


On Tue, Feb 10, 2015 at 4:13 PM, Ethan Duni ethan.d...@gmail.com wrote:
 I'm all for releasing stuff from improved masters. There's a trend in my
 favorite genre (heavy metal) to rerelease a lot of classics in full
 dynamic range editions lately. While I'm not sure that all of these
 releases really sound much better (how much dynamic range was there in an
 underground death metal recording from 1991 anyway?) I like the trend.
 These are regular CD releases, no weird formats (demonstrating that such is
 not required to sell the improved master releases).

 But the thing is that you often *can* hear the extra sampling frequency -
 in the form of additional distortion. It sounds, if anything, *worse* than
 a release with an appropriate sample rate! Trying to sell people on better
 audio, and then giving them a bunch of additional intermodulation
 distortion is not a justified marketing ploy, it's outright deceptive and
 abusive. This is working from the assumption that your customers are
 idiots, and that you should exploit that to make money, irrespective of
 whether audio quality is harmed or not. The fact the Neil Young is himself
 one of the suckers renders this less objectionable, but only slightly.
 Anyway Pono is already a byword for audiophile snake oil so hopefully the
 damage will mostly be limited to the bank accounts of Mr. Young and his
 various financial backers in this idiocy. Sounds like the product is a real
 dog in industrial design terms anyway (no hold button, awkward shape,
 etc.). Good riddance...

 E
 --
 dupswapdrop -- the music-dsp mailing list and website:
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Re: [music-dsp] Dither video and articles

I like the trend of releasing remastered material, where there is scope for
improved quality. Which isn't always, but there's an entire generation of
albums that were victims of the loudness wars, and various early work by
artists that hadn't access to quality mastering at the time, and so on,
that can benefit. This has been happening totally independent of Pono.

I don't like the Pono music scam because it confounds that (legitimate)
aspect with the snake oil about 24 bits and high sampling rates - while
charging a premium. There is zero meaningful test results that back Pono's
quality claims (and note how frequently their marketing adds caveats about
comparing to low-res MP3s, as if it's 1998 or something). And while there
isn't a definitive formal test showing that Pono sucks, there are multiple
informal tests without obvious methodological flaws which show that Pono is
inferior to your regular iTunes downloads. Neil Young says he's going to
give you better quality (for 2-3 times the price), and instead delivers
*lower* quality (or, maybe, the same, at best).

The fact that their own marketing material can't even seem to keep their
story straight regarding what the high resolution is or is not supposed to
provide you, seems to me to go to the point that this is all a marketing
exercise in bullshitting the consumer with a bunch of ill-founded claims.
For that matter, Pono's implication that one can't get improved masters via
other routes is itself deceptive.

I'm also somewhat bemused by Neil Young being the poster boy for this
high-resolution snake oil. While I admittedly haven't listened to his
entire catalogue, his whole style features low dynamic range, non-extreme
spectrum, and quite high noise floors (typically easily audible at even
moderate volume). Which is fine, nothing wrong with the crunchy/vintage
rock sound. It just doesn't fit with the whole we need to be able to hear
stuff at 35kHz and -130dB delusions.

That said, this statement seems problematic:

Whether the higher resolution actually degrades the quality is a topic up
for future debate.

I mean, if you personally don't want to debate it right here and now that's
fine. But nobody is obliged to set this stuff aside. It's immediately
topical, and the test files for evaluating it have been provided in the
xiph link.

E



On Tue, Feb 10, 2015 at 1:25 PM, Tom Duffy tdu...@tascam.com wrote:

 So you like the bar being raised, but not the way that Neil Young has
 attempted?

 Whether the higher resolution actually degrades the quality is a
 topic up for future debate.

 From the ponomusic webpage:
 ...and now, with the PonoPlayer, you can finally feel the master in all
 its glory, in its native resolution, CD quality or higher, the way the
 artist made it, exactly

 Even they are not saying it has to be higher than CD quality, just
 that it has to have been made well in the first place.

 I don't get why so many people are trying to paint this as
 a snake oil pitch.

 ---
 Tom.


 On 2/10/2015 1:13 PM, Ethan Duni wrote:
 I'm all for releasing stuff from improved masters. There's a trend in my
 favorite genre (heavy metal) to rerelease a lot of classics in full
 dynamic range editions lately. While I'm not sure that all of these
 releases really sound much better (how much dynamic range was there in an
 underground death metal recording from 1991 anyway?) I like the trend.
 These are regular CD releases, no weird formats (demonstrating that such is
 not required to sell the improved master releases).

 But the thing is that you often *can* hear the extra sampling frequency -
 in the form of additional distortion. It sounds, if anything, *worse* than
 a release with an appropriate sample rate! Trying to sell people on better
 audio, and then giving them a bunch of additional intermodulation
 distortion is not a justified marketing ploy, it's outright deceptive and
 abusive. This is working from the assumption that your customers are
 idiots, and that you should exploit that to make money, irrespective of
 whether audio quality is harmed or not. The fact the Neil Young is himself
 one of the suckers renders this less objectionable, but only slightly.
 Anyway Pono is already a byword for audiophile snake oil so hopefully the
 damage will mostly be limited to the bank accounts of Mr. Young and his
 various financial backers in this idiocy. Sounds like the product is a real
 dog in industrial design terms anyway (no hold button, awkward shape,
 etc.). Good riddance...

 E
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book reviews,
 dsp links
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 NOTICE: This electronic mail message and its contents, including any
 attachments hereto (collectively, this e-mail), is hereby designated as
 confidential and proprietary. This e-mail may be viewed and used only by
 the person to whom it

Re: [music-dsp] Dither video and articles

why does higher-than-needed sample rate hurt audio quality?
might not be necessary, but how does it make it worse (excluding
the increased computational burden)?

The danger is that you are now including a bunch of out-of-band content in
your output signal, which can be transformed into in-band aliasing by any
nonlinearities in your playback chain. It's generally not a big deal, but
it is measurable and does hurt quality:

http://xiph.org/~xiphmont/demo/neil-young.html

This is an excellent example of the tension between audiophile
perfectionism (i.e., more sample rate must always be at least as good,
because digital audio is some kind of terrifying bogeyman) and actual
engineering quality control (i.e., overspec-ing systems drives up costs,
compromises the quality in other components, and generally creates more
headaches than it solves).

E


On Tue, Feb 10, 2015 at 10:54 AM, robert bristow-johnson 
r...@audioimagination.com wrote:

 On 2/10/15 1:51 PM, Ethan Duni wrote:

 So to you, that Pono player isn't snake oil?

 It's more the 192kHz sampling rate that renders the Pono player into snake
 oil territory. The extra bits probably aren't getting you much, but the
 ridiculous sampling rate can only *hurt* audio quality, while consuming
 that much more battery and storage.


 that's interesting.  why does higher-than-needed sample rate hurt audio
 quality?  might not be necessary, but how does it make it worse (excluding
 the increased computational burden)?  i always think that analog (or
 continuous-time) is like having an infinite sample rate.


 --

 r b-j  r...@audioimagination.com

 Imagination is more important than knowledge.



 --
 dupswapdrop -- the music-dsp mailing list and website:
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Re: [music-dsp] Dither video and articles

2015-02-10 Thread Zhiguang Zhang

Re:Pono, what about the DAC in the device?  That could make an audible and real 
difference.  Also, there is undeniably more information in high res downloads, 
if the original master was recorded to tape or to hi-res in Pro Tools.  So, has 
anyone ever considered the sample-level ‘phase’ effect of listening to properly 
mastered hi-res audio if the playback chain is of a quality that diminishes 
intermodulation artifacts?




-EZ

On Tue, Feb 10, 2015 at 4:45 PM, Ethan Duni ethan.d...@gmail.com wrote:

 I like the trend of releasing remastered material, where there is scope for
 improved quality. Which isn't always, but there's an entire generation of
 albums that were victims of the loudness wars, and various early work by
 artists that hadn't access to quality mastering at the time, and so on,
 that can benefit. This has been happening totally independent of Pono.
 I don't like the Pono music scam because it confounds that (legitimate)
 aspect with the snake oil about 24 bits and high sampling rates - while
 charging a premium. There is zero meaningful test results that back Pono's
 quality claims (and note how frequently their marketing adds caveats about
 comparing to low-res MP3s, as if it's 1998 or something). And while there
 isn't a definitive formal test showing that Pono sucks, there are multiple
 informal tests without obvious methodological flaws which show that Pono is
 inferior to your regular iTunes downloads. Neil Young says he's going to
 give you better quality (for 2-3 times the price), and instead delivers
 *lower* quality (or, maybe, the same, at best).
 The fact that their own marketing material can't even seem to keep their
 story straight regarding what the high resolution is or is not supposed to
 provide you, seems to me to go to the point that this is all a marketing
 exercise in bullshitting the consumer with a bunch of ill-founded claims.
 For that matter, Pono's implication that one can't get improved masters via
 other routes is itself deceptive.
 I'm also somewhat bemused by Neil Young being the poster boy for this
 high-resolution snake oil. While I admittedly haven't listened to his
 entire catalogue, his whole style features low dynamic range, non-extreme
 spectrum, and quite high noise floors (typically easily audible at even
 moderate volume). Which is fine, nothing wrong with the crunchy/vintage
 rock sound. It just doesn't fit with the whole we need to be able to hear
 stuff at 35kHz and -130dB delusions.
 That said, this statement seems problematic:
Whether the higher resolution actually degrades the quality is a topic up
 for future debate.
 I mean, if you personally don't want to debate it right here and now that's
 fine. But nobody is obliged to set this stuff aside. It's immediately
 topical, and the test files for evaluating it have been provided in the
 xiph link.
 E
 On Tue, Feb 10, 2015 at 1:25 PM, Tom Duffy tdu...@tascam.com wrote:
 So you like the bar being raised, but not the way that Neil Young has
 attempted?

 Whether the higher resolution actually degrades the quality is a
 topic up for future debate.

 From the ponomusic webpage:
 ...and now, with the PonoPlayer, you can finally feel the master in all
 its glory, in its native resolution, CD quality or higher, the way the
 artist made it, exactly

 Even they are not saying it has to be higher than CD quality, just
 that it has to have been made well in the first place.

 I don't get why so many people are trying to paint this as
 a snake oil pitch.

 ---
 Tom.


 On 2/10/2015 1:13 PM, Ethan Duni wrote:
 I'm all for releasing stuff from improved masters. There's a trend in my
 favorite genre (heavy metal) to rerelease a lot of classics in full
 dynamic range editions lately. While I'm not sure that all of these
 releases really sound much better (how much dynamic range was there in an
 underground death metal recording from 1991 anyway?) I like the trend.
 These are regular CD releases, no weird formats (demonstrating that such is
 not required to sell the improved master releases).

 But the thing is that you often *can* hear the extra sampling frequency -
 in the form of additional distortion. It sounds, if anything, *worse* than
 a release with an appropriate sample rate! Trying to sell people on better
 audio, and then giving them a bunch of additional intermodulation
 distortion is not a justified marketing ploy, it's outright deceptive and
 abusive. This is working from the assumption that your customers are
 idiots, and that you should exploit that to make money, irrespective of
 whether audio quality is harmed or not. The fact the Neil Young is himself
 one of the suckers renders this less objectionable, but only slightly.
 Anyway Pono is already a byword for audiophile snake oil so hopefully the
 damage will mostly be limited to the bank accounts of Mr. Young and his
 various financial backers in this idiocy. Sounds like the product is a real
 dog in industrial design terms anyway (no hold

Re: [music-dsp] Dither video and articles

How do the crest factors of these different sawtooth waveforms compare?
I'd expect one with randomized phase to have a much lower crest factor.
Which is to say that I'd expect the in-phase sawtooth to activate a lot
more nonlinearity in the playback chain, which explains why that one is
easy to pick out but the various randomized ones all sound similar. It also
implies that we'd need a very fancy playback system with excellent
linearity to draw any conclusions about the underlying audibility of the
sawtooth partial phases as such.

E

On Tue, Feb 10, 2015 at 1:52 PM, gwenhwyfaer gwenhwyf...@gmail.com wrote:

 On 10/02/2015, Didier Dambrin di...@skynet.be wrote:
  Pretty easy to check the obvious difference between a pure low sawtooth,
 and
 
  the same sawtooth with all partials starting at random phases.

 Ah, this again? Good times. I remember playing. I made 7 sawtooth
 waves with random (static) phases and one straightforward sawtooth
 wave, with all partials in phase. I just listened to it again, to
 check my memory. On a half-decent pair of headphones, the difference
 between the all-partials-in-phase sawtooth and the random-phase ones
 is readily audible, but it was rather harder to tell the difference
 between the various random-phase waves; they all kind of sounded
 pulse-wavey. On a pair of speakers through the same amp and soundcard,
 though, I can still *jst about* pick out the in-phase sawtooth -
 but I couldn't confidently tell the difference between the 7 other
 waves. Which I'm guessing has something to do with the difference
 between the fairly one-dimensional travel of sound from headphone to
 ear, vs the bouncing-in-from-all-kinds-of-directions speaker-ear
 journey.

 I'm only a data point, though, so I'm not brave enough to actually
 conclude anything. At least, not any more. ;)
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Re: [music-dsp] Dither video and articles

2015-02-10 Thread Eric Brombaugh


Here's the guts of the Pono:

http://mikebeauchamp.com/2014/12/pono-player-teardown/

DAC is an ESS ES9018K2M

http://www.esstech.com/PDF/ES9018-2M%20PB%20Rev%200.8%20130619.pdf

32-bit - Wonder what the actual ENOB is...

Output driver is a discrete design.

Main MCU is apparently a TI OMAP similar to those found on the Beagleboard.

Eric

On 02/10/2015 03:27 PM, Zhiguang Zhang wrote:

Actually scratch that 2nd thought.  It would be good to know what DAC the Pono 
device contains.

-EZ

On Tue, Feb 10, 2015 at 5:20 PM, Zhiguang Zhang ericzh...@gmail.com
wrote:


Re:Pono, what about the DAC in the device?  That could make an audible and real 
difference.  Also, there is undeniably more information in high res downloads, 
if the original master was recorded to tape or to hi-res in Pro Tools.  So, has 
anyone ever considered the sample-level ‘phase’ effect of listening to properly 
mastered hi-res audio if the playback chain is of a quality that diminishes 
intermodulation artifacts?
-EZ


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Re: [music-dsp] Dither video and articles

2015-02-09 Thread Vicki Melchior

Nigel, I looked at your video again and it seems to me it's confusing as to 
whether you mean 'don't dither the 24b final output' or 'don't ever dither at 
24b'.  You make statements several times that imply the former, but in your 
discussion about 24b on all digital interfaces, sends and receives etc, you 
clearly say to never dither at 24b.  Several people in this thread have pointed 
out the difference between intermediate stage truncation and final stage 
truncation, and the fact that if truncation is done repeatedly, any distortion 
spectra will continue to build.   It is not noise-like, the peaks are coherent 
peaks and correlated to the signal.  

You don't say in the video what the processing history is for the files you are 
using.  If they are simple captures with no processing, they probably reflect 
the additive gaussian noise present at  the 20th bit in the A/D, based on 
Andy's post, and are properly dithered for 24b truncation.   My point is that 
at the digital capture stage you have (S+N) and the amplitude distribution of 
the S+N signal might be fine for 24b truncation if N is dither-like.  After 
various stages of digital processing including non-linear steps, the (S+N) 
intermediate signal may no longer have an adequate amplitude distribution to be 
truncated without 24b dither.  

I think the whole subject of self dither might be better approached through FFT 
measurement than by listening.   Bob Katz shows an FFT of truncation spectra at 
24b in his book on 'Itunes Music, Mastering for High Resolution Audio Delivery' 
 but he uses a generated, dithered pure tone that doesn't start with added 
gaussian noise.  Haven't thought about it but I can imagine extending his 
approach into a research effort.  

Offhand I don't know anything that would go wrong in your difference file ( 
...if the error doesn't sound wrong).  It's a common method for looking at 
residuals.

Vicki


On Feb 8, 2015, at 6:11 PM, Nigel Redmon wrote:

 Beyond that, Nigel raises this issue in the context of self-dither”...
 
 First, remember that I’m the guy who recommended “always” dithering 16-bit 
 (no “always” as in “alway necessary”, but as in “do it always, unless you 
 know that it gives no improvement”), and to not bother dithering 24-bit. So, 
 I’m only interested in this discussion for 24-bit. That said:
 
 ...In situations where there is a clear external noise source present, 
 whether the situation is analog to digital conversion or digital to digital 
 bit depth change, the external noise may, or may not, be satisfactory as 
 dither but at least it's properties can be measured.
 
 For 24-bit audio, could you give an example of when it’s likely to not be 
 satisfactory (maybe you’ve already given a reference to determining 
 “satisfactory)? Offhand, I’d say one case might be with extremely low noise, 
 then digitally faded such that you fade the noise level below the dithering 
 threshold while you still have enough signal to exhibit truncation 
 distortion, and the fade characteristics allow it to last long enough to 
 matter to your ears—if we weren’t talking about this distortion being down 
 near -140 dB in the first place. I’d think that, typically, you’d have 
 gaussian noise at a much higher level that is needed to dither 24-bit; that 
 could change with digital processing, but I think that in the usual recording 
 chain, it seems pretty hard to avoid for your analog to digital conversion” 
 case.
 
 I’m still interested in what you have to say about my post yesterday (“...if 
 the error doesn’t sound wrong to the ear, can it still sound wrong added to 
 the music?”). Care to comment?
 
 
 On Feb 8, 2015, at 8:09 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 I have no argument at all with the cheap high-pass TPDF dither; whenever it 
 was published the original authors undoubtedly verified that the moment 
 decoupling occurred, as you say.  And that's what is needed for dither 
 effectiveness.   If you're creating noise for dither, you have the option to 
 verify its properties.  But in the situation of an analog signal with added, 
 independent instrument noise, you do need to verify that the composite noise 
 source actually satisfies the criteria for dither.  1/f noise in particular 
 has been questioned, which is why I raised the spectrum issue.  
 
 Beyond that, Nigel raises this issue in the context of self-dither.  In 
 situations where there is a clear external noise source present, whether the 
 situation is analog to digital conversion or digital to digital bit depth 
 change, the external noise may, or may not, be satisfactory as dither but at 
 least it's properties can be measured.  If the 'self-dithering' instead 
 refers to analog noise captured into the digitized signal with the idea that 
 this noise is going to be preserved and available at later truncation steps 
 to 'self dither' it is a very very hazy argument.   I'm aware of the various 
 caveats that are often

Re: [music-dsp] Dither video and articles

2015-02-09 Thread Vicki Melchior

That's a clear explanation of the self-dither assumed in A/D conversion, thanks 
for posting it. 

Vicki
 
On Feb 8, 2015, at 9:11 PM, Andrew Simper wrote:

 Vicki,
 
 If you look at the limits of what is possible in a real world ADC
 there is a certain amount of noise in any electrical system due to
 gaussian thermal noise:
 http://en.wikipedia.org/wiki/Johnson%E2%80%93Nyquist_noise
 
 For example if you look at an instrument / measurement grade ADC like
 this: 
 http://www.prismsound.com/test_measure/products_subs/dscope/dscope_spec.php
 They publish figures of a residual noise floor of 1.4 uV, which they
 say is -115 dBu. So if you digitise a 1 V peak (2 V peak to peak) sine
 wave with a 24-bit ADC then you will have hiss (which includes a large
 portion of gaussian noise) at around the 20 bit mark, so you will have
 4-bits of hiss to self dither. This has nothing to do with microphones
 or noise in air, this is in the near perfect case of transmission via
 a well shielded differential cable transferring the voltage directly
 to the ADC.
 
 All the best,
 
 Andy
 -- cytomic -- sound music software --
 
 
 On 9 February 2015 at 00:09, Vicki Melchior vmelch...@earthlink.net wrote:
 I have no argument at all with the cheap high-pass TPDF dither; whenever it 
 was published the original authors undoubtedly verified that the moment 
 decoupling occurred, as you say.  And that's what is needed for dither 
 effectiveness.   If you're creating noise for dither, you have the option to 
 verify its properties.  But in the situation of an analog signal with added, 
 independent instrument noise, you do need to verify that the composite noise 
 source actually satisfies the criteria for dither.  1/f noise in particular 
 has been questioned, which is why I raised the spectrum issue.
 
 Beyond that, Nigel raises this issue in the context of self-dither.  In 
 situations where there is a clear external noise source present, whether the 
 situation is analog to digital conversion or digital to digital bit depth 
 change, the external noise may, or may not, be satisfactory as dither but at 
 least it's properties can be measured.  If the 'self-dithering' instead 
 refers to analog noise captured into the digitized signal with the idea that 
 this noise is going to be preserved and available at later truncation steps 
 to 'self dither' it is a very very hazy argument.   I'm aware of the various 
 caveats that are often postulated, i.e. signal is captured at double 
 precision, no truncation, very selected processing.  But even in minimalist 
 recording such as live to two track, it's not clear to me that the signal 
 can get through the digital stages of the A/D and still retain an unaltered 
 noise distribution.  It certainly won't do so after considerable processing. 
  So the sho
 r
 t
 answer is, dither!  At the 24th bit or at the 16th bit, whatever your output 
 is.  If you (Nigel or RBJ) have references to the contrary, please say so.
 
 Vicki
 
 On Feb 8, 2015, at 10:11 AM, robert bristow-johnson wrote:
 
 On 2/7/15 8:54 AM, Vicki Melchior wrote:
 Well, the point of dither is to reduce correlation between the signal and 
 quantization noise.  Its effectiveness requires that the error signal has 
 given properties; the mean error should be zero and the RMS error should 
 be independent of the signal.  The best known examples satisfying those 
 conditions are white Gaussian noise at ~ 6dB above the RMS quantization 
 level and white TPDF noise  at ~3dB above the same, with Gaussian noise 
 eliminating correlation entirely and TPDF dither eliminating correlation 
 with the first two moments of the error distribution.   That's all 
 textbook stuff.  There are certainly noise shaping algorithms that shape 
 either the sum of white dither and quantization noise or the white dither 
 and quantization noise independently, and even (to my knowledge) a few 
 completely non-white dithers that are known to work, but determining the 
 effectiveness of noise at dithering still requires examining the 
 statistical properties of the error signal and showi
 n
 g
 
 th
 at the mean is 0 and the second moment is signal independent.  (I think 
 Stanley Lipschitz showed that the higher moments don't matter to 
 audibility.)
 
 but my question was not about the p.d.f. of the dither (to decouple both 
 the mean and the variance of the quantization error, you need triangular 
 p.d.f. dither of 2 LSBs width that is independent of the *signal*) but 
 about the spectrum of the dither.  and Nigel mentioned this already, but 
 you can cheaply make high-pass TPDF dither with a single (decent) uniform 
 p.d.f. random number per sample and running that through a simple 1st-order 
 FIR which has +1 an -1 coefficients (i.e. subtract the previous UPDF from 
 the current UPDF to get the high-pass TPDF).  also, i think Bart Locanthi 
 (is he still on this planet?) and someone else did a simple paper back in 
 the 90s about the possible benefits of

Re: [music-dsp] Dither video and articles

2015-02-09 Thread Nigel Redmon

But it matters, because the whole point of dithering to 16bit depends on how 
common that ability is.

Depends on how common? I’m not sure what qualifies for common, but if it’s 1 in 
100, or 5 in 100, it’s still a no-brainer because it costs nothing, effectively.

But more importantly, I don’t think you’re impressed by my point that it’s the 
audio engineers, the folks making the music, that are in the best position to 
hear it, and to do something about it. There are the ones listening carefully, 
in studios built to be quiet and lack reflections and resonances that might 
mask things, on revealing monitors and with ample power. I don’t think that you 
understand that it’s these guys who are not going to let their work go out the 
door with grit on it, even if it’s below -90 dB. You wouldn’t get many 
sympathetic ears among them if you advocated that they cease this dithering 
nonsense :-) I get enough grief about telling them that dither at 24-bit is 
useless.

How common it is for for the average listener is immaterial. It’s not done for 
the average listener.


 On Feb 9, 2015, at 6:56 PM, Didier Dambrin di...@skynet.be wrote:
 
 I'm having a hard time finding anyone who could hear past the -72dB noise, 
 here around.
 
 Really, either you have super-ears, or the cause is (technically) somewhere 
 else. But it matters, because the whole point of dithering to 16bit depends 
 on how common that ability is.
 
 
 
 
 -Message d'origine- From: Andrew Simper
 Sent: Saturday, February 07, 2015 2:08 PM
 To: A discussion list for music-related DSP
 Subject: Re: [music-dsp] Dither video and articles
 
 On 7 February 2015 at 03:52, Didier Dambrin di...@skynet.be wrote:
 It was just several times the same fading in/out noise at different levels,
 just to see if you hear quieter things than I do, I thought you'd have
 guessed that.
 https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing
 (0dB, -36dB, -54dB, -66dB, -72dB, -78dB)
 
 Here if I make the starting noise annoying, then I hear the first 4 parts,
 until 18:00. Thus, if 0dB is my threshold of annoyance, I can't hear -72dB.
 
 So you hear it at -78dB? Would be interesting to know how many can, and if
 it's subjective or a matter of testing environment (the variable already
 being the 0dB annoyance starting point)
 
 Yep, I could hear all of them, and the time I couldn't hear the hiss
 any more as at the 28.7 second mark, just before the end of the file.
 For reference this noise blast sounded much louder than the bass tone
 that Nigel posted when both were normalised, I had my headphones amp
 at -18 dB so the first noise peak was loud but not uncomfortable.
 
 I thought it was an odd test since the test file just stopped before I
 couldn't hear the LFO amplitude modulation cycles, so I wasn't sure
 what you were trying to prove!
 
 All the best,
 
 Andy
 
 
 
 
 -Message d'origine- From: Andrew Simper
 Sent: Friday, February 06, 2015 3:21 PM
 To: A discussion list for music-related DSP
 Subject: Re: [music-dsp] Dither video and articles
 
 Sorry, you said until, which is even more confusing. There are
 multiple points when I hear the noise until since it sounds like the
 noise is modulated in amplitude by a sine like LFO for the entire
 file, so the volume of the noise ramps up and down in a cyclic manner.
 The last ramping I hear fades out at around the 28.7 second mark when
 it is hard to tell if it just ramps out at that point or is just on
 the verge of ramping up again and then the file ends at 28.93 seconds.
 I have not tried to measure the LFO wavelength or any other such
 things, this is just going on listening alone.
 
 All the best,
 
 Andrew Simper
 
 
 
 On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:
 
 On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:
 
 Just out of curiosity, until which point do you hear the noise in this
 little test (a 32bit float wav), starting from a bearable first part?
 
 
 https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing
 
 
 I hear noise immediately in that recording, it's hard to tell exactly
 the time I can first hear it since there is some latency from when I
 press play to when the sound starts, but as far as I can tell it is
 straight away. Why do you ask such silly questions?
 
 All the best,
 
 Andrew Simper
 
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 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book reviews, dsp
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Re: [music-dsp] Dither video and articles

2015-02-09 Thread Didier Dambrin

I'm having a hard time finding anyone who could hear past the -72dB noise, 
here around.


Really, either you have super-ears, or the cause is (technically) somewhere 
else. But it matters, because the whole point of dithering to 16bit depends 
on how common that ability is.





-Message d'origine- 
From: Andrew Simper

Sent: Saturday, February 07, 2015 2:08 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

On 7 February 2015 at 03:52, Didier Dambrin di...@skynet.be wrote:
It was just several times the same fading in/out noise at different 
levels,

just to see if you hear quieter things than I do, I thought you'd have
guessed that.
https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing
(0dB, -36dB, -54dB, -66dB, -72dB, -78dB)

Here if I make the starting noise annoying, then I hear the first 4 parts,
until 18:00. Thus, if 0dB is my threshold of annoyance, I can't 
hear -72dB.


So you hear it at -78dB? Would be interesting to know how many can, and if
it's subjective or a matter of testing environment (the variable already
being the 0dB annoyance starting point)


Yep, I could hear all of them, and the time I couldn't hear the hiss
any more as at the 28.7 second mark, just before the end of the file.
For reference this noise blast sounded much louder than the bass tone
that Nigel posted when both were normalised, I had my headphones amp
at -18 dB so the first noise peak was loud but not uncomfortable.

I thought it was an odd test since the test file just stopped before I
couldn't hear the LFO amplitude modulation cycles, so I wasn't sure
what you were trying to prove!

All the best,

Andy





-Message d'origine- From: Andrew Simper
Sent: Friday, February 06, 2015 3:21 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.

All the best,

Andrew Simper



On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:


On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:


Just out of curiosity, until which point do you hear the noise in this
little test (a 32bit float wav), starting from a bearable first part?


https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing



I hear noise immediately in that recording, it's hard to tell exactly
the time I can first hear it since there is some latency from when I
press play to when the sound starts, but as far as I can tell it is
straight away. Why do you ask such silly questions?

All the best,

Andrew Simper


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Re: [music-dsp] Dither video and articles

2015-02-09 Thread Nigel Redmon

OK, I don’t want to diverge too much from the practical to the theoretical, so 
I’m going to run down what is usual, not what is possible, because it narrows 
the field of discussion.

Most people I know are using recording systems that bussing audio at 32-bit 
float, minimum, and use 64-bit float calculations in plug-ins and significant 
processing. They may still be using 24-bit audio tracks on disk, but for the 
most part they are recorded and are dithered one way or another (primarily 
gaussian noise in the recording process). They may bounce things to tracks to 
free processor cycles. I think in large majority of cases, these are 
self-dithered, but even if it doesn’t happen for some, I don’t think it will 
impact the audio. And if people are worried about it, I don’t understand why 
they aren’t using 32-bit float files, as I think most people have that choices 
these days.

Some of the more hard core will send audio out to a convertor (therefore 
truncated at 24-bit), and back in. Again, I think the vast majority of cases, 
these will self dither, but then there’s the fact error is at a very low level, 
will get buried in the thermal noise of the electronics, etc.

Maybe I left out some other good ones, but to cut it short, yes, I’m mainly 
talking about final mixes. At 24-bit, that often goes to someone else to 
master. The funny thing is that some mastering engineers say “only dither 
once!”, and they want to be the one doing it. Others point out that they may 
want to mess with the dynamic range and boost frequencies, and any error from 
not dithering 24-bit will show up in…you know, the stereo imaging, depth, etc. 
I think it would be exceptional to actually have truncation distortion of 
significant duration, except for potential situations with unusual fades, so 
I’m not worried about saying don’t dither 24-bit, even heading to a mastering 
engineer (but again, do it if you want, it’s just no big deal for final 
outputs–in contrast to the pain in the rear it is to do it at every point for 
the items I mentioned in previous paragraphs).

Down the more theoretical paths, I’ve had people argue that this is a big deal 
because things like ProTools 56k plug-ins need to be dithered internally…but 
why argue legacy stuff that “is what it is”, and secondly, these people usually 
don’t think through how many 24-bit truncations occur in a 56k algorithm, and 
you only have so many cycles. The other thing I sometimes get is the specter of 
the cumulative effect (but what if you have so many tracks, and feedback, 
and…)—but it seems to me that the more of this you get going on, to approach a 
meaningful error magnitude, the more it’s jumbled up in chaos and the less easy 
it is for your ear to recognize it as “bad”.



 On Feb 9, 2015, at 7:54 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 Nigel, I looked at your video again and it seems to me it's confusing as to 
 whether you mean 'don't dither the 24b final output' or 'don't ever dither at 
 24b'.  You make statements several times that imply the former, but in your 
 discussion about 24b on all digital interfaces, sends and receives etc, you 
 clearly say to never dither at 24b.  Several people in this thread have 
 pointed out the difference between intermediate stage truncation and final 
 stage truncation, and the fact that if truncation is done repeatedly, any 
 distortion spectra will continue to build.   It is not noise-like, the peaks 
 are coherent peaks and correlated to the signal.  
 
 You don't say in the video what the processing history is for the files you 
 are using.  If they are simple captures with no processing, they probably 
 reflect the additive gaussian noise present at  the 20th bit in the A/D, 
 based on Andy's post, and are properly dithered for 24b truncation.   My 
 point is that at the digital capture stage you have (S+N) and the amplitude 
 distribution of the S+N signal might be fine for 24b truncation if N is 
 dither-like.  After various stages of digital processing including non-linear 
 steps, the (S+N) intermediate signal may no longer have an adequate amplitude 
 distribution to be truncated without 24b dither.  
 
 I think the whole subject of self dither might be better approached through 
 FFT measurement than by listening.   Bob Katz shows an FFT of truncation 
 spectra at 24b in his book on 'Itunes Music, Mastering for High Resolution 
 Audio Delivery'  but he uses a generated, dithered pure tone that doesn't 
 start with added gaussian noise.  Haven't thought about it but I can imagine 
 extending his approach into a research effort.  
 
 Offhand I don't know anything that would go wrong in your difference file ( 
 ...if the error doesn't sound wrong).  It's a common method for looking at 
 residuals.
 
 Vicki
 
 
 On Feb 8, 2015, at 6:11 PM, Nigel Redmon wrote:
 
 Beyond that, Nigel raises this issue in the context of self-dither”...
 
 First, remember that I’m the guy who recommended “always”

Re: [music-dsp] Dither video and articles

2015-02-09 Thread Nigel Redmon

I’m thankful for Andy posting that clear explanation too. Sometimes I 
understate things—when I said that it would be “pretty hard to avoid” having 
ample gaussian noise to self-dither in the A/D process, I was thinking 
cryogenics (LOL).


 On Feb 9, 2015, at 7:54 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 That's a clear explanation of the self-dither assumed in A/D conversion, 
 thanks for posting it. 
 
 Vicki
 
 On Feb 8, 2015, at 9:11 PM, Andrew Simper wrote:
 
 Vicki,
 
 If you look at the limits of what is possible in a real world ADC
 there is a certain amount of noise in any electrical system due to
 gaussian thermal noise:
 http://en.wikipedia.org/wiki/Johnson%E2%80%93Nyquist_noise
 
 For example if you look at an instrument / measurement grade ADC like
 this: 
 http://www.prismsound.com/test_measure/products_subs/dscope/dscope_spec.php
 They publish figures of a residual noise floor of 1.4 uV, which they
 say is -115 dBu. So if you digitise a 1 V peak (2 V peak to peak) sine
 wave with a 24-bit ADC then you will have hiss (which includes a large
 portion of gaussian noise) at around the 20 bit mark, so you will have
 4-bits of hiss to self dither. This has nothing to do with microphones
 or noise in air, this is in the near perfect case of transmission via
 a well shielded differential cable transferring the voltage directly
 to the ADC.
 
 All the best,
 
 Andy
 -- cytomic -- sound music software --
 
 
 On 9 February 2015 at 00:09, Vicki Melchior vmelch...@earthlink.net wrote:
 I have no argument at all with the cheap high-pass TPDF dither; whenever it 
 was published the original authors undoubtedly verified that the moment 
 decoupling occurred, as you say.  And that's what is needed for dither 
 effectiveness.   If you're creating noise for dither, you have the option 
 to verify its properties.  But in the situation of an analog signal with 
 added, independent instrument noise, you do need to verify that the 
 composite noise source actually satisfies the criteria for dither.  1/f 
 noise in particular has been questioned, which is why I raised the spectrum 
 issue.
 
 Beyond that, Nigel raises this issue in the context of self-dither.  In 
 situations where there is a clear external noise source present, whether 
 the situation is analog to digital conversion or digital to digital bit 
 depth change, the external noise may, or may not, be satisfactory as dither 
 but at least it's properties can be measured.  If the 'self-dithering' 
 instead refers to analog noise captured into the digitized signal with the 
 idea that this noise is going to be preserved and available at later 
 truncation steps to 'self dither' it is a very very hazy argument.   I'm 
 aware of the various caveats that are often postulated, i.e. signal is 
 captured at double precision, no truncation, very selected processing.  But 
 even in minimalist recording such as live to two track, it's not clear to 
 me that the signal can get through the digital stages of the A/D and still 
 retain an unaltered noise distribution.  It certainly won't do so after 
 considerable processing.  So the sho
 r
 t
 answer is, dither!  At the 24th bit or at the 16th bit, whatever your 
 output is.  If you (Nigel or RBJ) have references to the contrary, please 
 say so.
 
 Vicki
 
 On Feb 8, 2015, at 10:11 AM, robert bristow-johnson wrote:
 
 On 2/7/15 8:54 AM, Vicki Melchior wrote:
 Well, the point of dither is to reduce correlation between the signal and 
 quantization noise.  Its effectiveness requires that the error signal has 
 given properties; the mean error should be zero and the RMS error should 
 be independent of the signal.  The best known examples satisfying those 
 conditions are white Gaussian noise at ~ 6dB above the RMS quantization 
 level and white TPDF noise  at ~3dB above the same, with Gaussian noise 
 eliminating correlation entirely and TPDF dither eliminating correlation 
 with the first two moments of the error distribution.   That's all 
 textbook stuff.  There are certainly noise shaping algorithms that shape 
 either the sum of white dither and quantization noise or the white dither 
 and quantization noise independently, and even (to my knowledge) a few 
 completely non-white dithers that are known to work, but determining the 
 effectiveness of noise at dithering still requires examining the 
 statistical properties of the error signal and showi
 n
 g
 
 th
 at the mean is 0 and the second moment is signal independent.  (I think 
 Stanley Lipschitz showed that the higher moments don't matter to 
 audibility.)
 
 but my question was not about the p.d.f. of the dither (to decouple both 
 the mean and the variance of the quantization error, you need triangular 
 p.d.f. dither of 2 LSBs width that is independent of the *signal*) but 
 about the spectrum of the dither.  and Nigel mentioned this already, but 
 you can cheaply make high-pass TPDF dither with a single (decent) uniform 
 p.d.f. random

Re: [music-dsp] Dither video and articles

2015-02-08 Thread Vicki Melchior

I have no argument at all with the cheap high-pass TPDF dither; whenever it was 
published the original authors undoubtedly verified that the moment decoupling 
occurred, as you say.  And that's what is needed for dither effectiveness.   If 
you're creating noise for dither, you have the option to verify its properties. 
 But in the situation of an analog signal with added, independent instrument 
noise, you do need to verify that the composite noise source actually satisfies 
the criteria for dither.  1/f noise in particular has been questioned, which is 
why I raised the spectrum issue.  

Beyond that, Nigel raises this issue in the context of self-dither.  In 
situations where there is a clear external noise source present, whether the 
situation is analog to digital conversion or digital to digital bit depth 
change, the external noise may, or may not, be satisfactory as dither but at 
least it's properties can be measured.  If the 'self-dithering' instead refers 
to analog noise captured into the digitized signal with the idea that this 
noise is going to be preserved and available at later truncation steps to 'self 
dither' it is a very very hazy argument.   I'm aware of the various caveats 
that are often postulated, i.e. signal is captured at double precision, no 
truncation, very selected processing.  But even in minimalist recording such as 
live to two track, it's not clear to me that the signal can get through the 
digital stages of the A/D and still retain an unaltered noise distribution.  It 
certainly won't do so after considerable processing.  So the short 
 answer is, dither!  At the 24th bit or at the 16th bit, whatever your output 
is.  If you (Nigel or RBJ) have references to the contrary, please say so.

Vicki

On Feb 8, 2015, at 10:11 AM, robert bristow-johnson wrote:

 On 2/7/15 8:54 AM, Vicki Melchior wrote:
 Well, the point of dither is to reduce correlation between the signal and 
 quantization noise.  Its effectiveness requires that the error signal has 
 given properties; the mean error should be zero and the RMS error should be 
 independent of the signal.  The best known examples satisfying those 
 conditions are white Gaussian noise at ~ 6dB above the RMS quantization 
 level and white TPDF noise  at ~3dB above the same, with Gaussian noise 
 eliminating correlation entirely and TPDF dither eliminating correlation 
 with the first two moments of the error distribution.   That's all textbook 
 stuff.  There are certainly noise shaping algorithms that shape either the 
 sum of white dither and quantization noise or the white dither and 
 quantization noise independently, and even (to my knowledge) a few 
 completely non-white dithers that are known to work, but determining the 
 effectiveness of noise at dithering still requires examining the statistical 
 properties of the error signal and showing
  
 th
  at the mean is 0 and the second moment is signal independent.  (I think 
 Stanley Lipschitz showed that the higher moments don't matter to audibility.)
 
 but my question was not about the p.d.f. of the dither (to decouple both the 
 mean and the variance of the quantization error, you need triangular p.d.f. 
 dither of 2 LSBs width that is independent of the *signal*) but about the 
 spectrum of the dither.  and Nigel mentioned this already, but you can 
 cheaply make high-pass TPDF dither with a single (decent) uniform p.d.f. 
 random number per sample and running that through a simple 1st-order FIR 
 which has +1 an -1 coefficients (i.e. subtract the previous UPDF from the 
 current UPDF to get the high-pass TPDF).  also, i think Bart Locanthi (is he 
 still on this planet?) and someone else did a simple paper back in the 90s 
 about the possible benefits of high-pass dither.  wasn't a great paper or 
 anything, but it was about the same point.
 
 i remember mentioning this at an AES in the 90's, and Stanley *did* address 
 it.  for straight dither it works okay, but for noise-shaping with feedback, 
 to be perfectly legitimate, you want white TPDF dither (which requires adding 
 or subtracting two independent UPDF random numbers).  and i agree with that.  
 it's just that if someone wanted to make a quick-and-clean high-pass dither 
 with the necessary p.d.f., you can do that with the simple subtraction trick. 
  and the dither is not white but perfectly decouples the first two moments of 
 the total quantization error.  it's just a simple trick that not good for too 
 much.
 
 -- 
 
 r b-j  r...@audioimagination.com
 
 Imagination is more important than knowledge.
 
 
 
 
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Re: [music-dsp] Dither video and articles

2015-02-08 Thread robert bristow-johnson

On 2/7/15 8:54 AM, Vicki Melchior wrote:
Well, the point of dither is to reduce correlation between the signal and quantization noise. Its effectiveness requires that the error signal has given properties; the mean error should be zero and the RMS error should be independent of the signal. The best known examples satisfying those conditions are white Gaussian noise at ~ 6dB above the RMS quantization level and white TPDF noise at ~3dB above the same, with Gaussian noise eliminating correlation entirely and TPDF dither eliminating correlation with the first two moments of the error distribution. That's all textbook stuff. There are certainly noise shaping algorithms that shape either the sum of white dither and quantization noise or the white dither and quantization noise independently, and even (to my knowledge) a few completely non-white dithers that are known to work, but determining the effectiveness of noise at dithering still requires examining the statistical properties of the error signal and showing

at the mean is 0 and the second moment is signal independent. (I think
Stanley Lipschitz showed that the higher moments don't matter to audibility.)

but my question was not about the p.d.f. of the dither (to decouple both
the mean and the variance of the quantization error, you need triangular
p.d.f. dither of 2 LSBs width that is independent of the *signal*) but
about the spectrum of the dither. and Nigel mentioned this already, but
you can cheaply make high-pass TPDF dither with a single (decent)
uniform p.d.f. random number per sample and running that through a
simple 1st-order FIR which has +1 an -1 coefficients (i.e. subtract the
previous UPDF from the current UPDF to get the high-pass TPDF). also, i
think Bart Locanthi (is he still on this planet?) and someone else did a
simple paper back in the 90s about the possible benefits of high-pass
dither. wasn't a great paper or anything, but it was about the same point.

i remember mentioning this at an AES in the 90's, and Stanley *did*
address it. for straight dither it works okay, but for noise-shaping
with feedback, to be perfectly legitimate, you want white TPDF dither
(which requires adding or subtracting two independent UPDF random
numbers). and i agree with that. it's just that if someone wanted to
make a quick-and-clean high-pass dither with the necessary p.d.f., you
can do that with the simple subtraction trick. and the dither is not
white but perfectly decouples the first two moments of the total
quantization error. it's just a simple trick that not good for too much.

r b-j r...@audioimagination.com

Imagination is more important than knowledge.

Re: [music-dsp] Dither video and articles

2015-02-08 Thread Andrew Simper

Vicki,

If you look at the limits of what is possible in a real world ADC
there is a certain amount of noise in any electrical system due to
gaussian thermal noise:
http://en.wikipedia.org/wiki/Johnson%E2%80%93Nyquist_noise

For example if you look at an instrument / measurement grade ADC like
this: 
http://www.prismsound.com/test_measure/products_subs/dscope/dscope_spec.php
They publish figures of a residual noise floor of 1.4 uV, which they
say is -115 dBu. So if you digitise a 1 V peak (2 V peak to peak) sine
wave with a 24-bit ADC then you will have hiss (which includes a large
portion of gaussian noise) at around the 20 bit mark, so you will have
4-bits of hiss to self dither. This has nothing to do with microphones
or noise in air, this is in the near perfect case of transmission via
a well shielded differential cable transferring the voltage directly
to the ADC.

All the best,

Andy
-- cytomic -- sound music software --


On 9 February 2015 at 00:09, Vicki Melchior vmelch...@earthlink.net wrote:
 I have no argument at all with the cheap high-pass TPDF dither; whenever it 
 was published the original authors undoubtedly verified that the moment 
 decoupling occurred, as you say.  And that's what is needed for dither 
 effectiveness.   If you're creating noise for dither, you have the option to 
 verify its properties.  But in the situation of an analog signal with added, 
 independent instrument noise, you do need to verify that the composite noise 
 source actually satisfies the criteria for dither.  1/f noise in particular 
 has been questioned, which is why I raised the spectrum issue.

 Beyond that, Nigel raises this issue in the context of self-dither.  In 
 situations where there is a clear external noise source present, whether the 
 situation is analog to digital conversion or digital to digital bit depth 
 change, the external noise may, or may not, be satisfactory as dither but at 
 least it's properties can be measured.  If the 'self-dithering' instead 
 refers to analog noise captured into the digitized signal with the idea that 
 this noise is going to be preserved and available at later truncation steps 
 to 'self dither' it is a very very hazy argument.   I'm aware of the various 
 caveats that are often postulated, i.e. signal is captured at double 
 precision, no truncation, very selected processing.  But even in minimalist 
 recording such as live to two track, it's not clear to me that the signal can 
 get through the digital stages of the A/D and still retain an unaltered noise 
 distribution.  It certainly won't do so after considerable processing.  So 
 the shor
 t
  answer is, dither!  At the 24th bit or at the 16th bit, whatever your output 
 is.  If you (Nigel or RBJ) have references to the contrary, please say so.

 Vicki

 On Feb 8, 2015, at 10:11 AM, robert bristow-johnson wrote:

 On 2/7/15 8:54 AM, Vicki Melchior wrote:
 Well, the point of dither is to reduce correlation between the signal and 
 quantization noise.  Its effectiveness requires that the error signal has 
 given properties; the mean error should be zero and the RMS error should be 
 independent of the signal.  The best known examples satisfying those 
 conditions are white Gaussian noise at ~ 6dB above the RMS quantization 
 level and white TPDF noise  at ~3dB above the same, with Gaussian noise 
 eliminating correlation entirely and TPDF dither eliminating correlation 
 with the first two moments of the error distribution.   That's all textbook 
 stuff.  There are certainly noise shaping algorithms that shape either the 
 sum of white dither and quantization noise or the white dither and 
 quantization noise independently, and even (to my knowledge) a few 
 completely non-white dithers that are known to work, but determining the 
 effectiveness of noise at dithering still requires examining the 
 statistical properties of the error signal and showin
 g

 th
  at the mean is 0 and the second moment is signal independent.  (I think 
 Stanley Lipschitz showed that the higher moments don't matter to 
 audibility.)

 but my question was not about the p.d.f. of the dither (to decouple both the 
 mean and the variance of the quantization error, you need triangular p.d.f. 
 dither of 2 LSBs width that is independent of the *signal*) but about the 
 spectrum of the dither.  and Nigel mentioned this already, but you can 
 cheaply make high-pass TPDF dither with a single (decent) uniform p.d.f. 
 random number per sample and running that through a simple 1st-order FIR 
 which has +1 an -1 coefficients (i.e. subtract the previous UPDF from the 
 current UPDF to get the high-pass TPDF).  also, i think Bart Locanthi (is he 
 still on this planet?) and someone else did a simple paper back in the 90s 
 about the possible benefits of high-pass dither.  wasn't a great paper or 
 anything, but it was about the same point.

 i remember mentioning this at an AES in the 90's, and Stanley *did* address 
 it.  for straight

Re: [music-dsp] Dither video and articles

2015-02-07 Thread Andrew Simper

32-bit internal floating point is not sufficient for certain DSP tasks
and will be plainly audible as causing all sorts of problems, a DF1 at
low frequencies is the classic example of this, it causes large
amounts of low frequency rumble. This is a completely different thing
to the final bit depth of an audio file to listen to.

Andy

-- cytomic -- sound music software --

On 7 February 2015 at 02:24, Michael Gogins michael.gog...@gmail.com wrote:

 Do not believe anything that is not confirmed to a high degree of
 statistical signifance (say, 5 standard deviations) by a double-blind
 test using an ABX comparator.

 That said, the AES study did use double-blind testing. I did not read
 the article, only the abstract, so cannot say more about the study.

 In my own work, I have verified with a double-blind ABX comparator at
 a high degree of statistical significance that I can hear the
 differences in certain selected portions of the same Csound piece
 rendered with 32 bit floating point samples versus 64 bit floating
 point samples. These are sample words used in internal calculations,
 not for output soundfiles. What I heard was differences in the sound
 of the same filter algorithm. These differences were not at all hard
 to hear, but they occurred in only one or two places in the piece.

 I have not myself been able to hear differences in audio output
 quality between CD audio and high-resolution audio, but when I get the
 time I may try again, now that I have a better idea what to listen
 for.

 Regards,
 Mike



 -
 Michael Gogins
 Irreducible Productions
 http://michaelgogins.tumblr.com
 Michael dot Gogins at gmail dot com


 On Fri, Feb 6, 2015 at 1:13 PM, Nigel Redmon earle...@earlevel.com wrote:
 Mastering engineers can hear truncation error at the 24th bit but say it is 
 subtle and may require experience or training to pick up.
 
  Quick observations:
 
  1) The output step size of the lsb is full-scale / 2^24. If full-scale is 
  1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths 
  of a volt). Hearing capabilities aside, the converter must be able to 
  resolve this, and it must make it through the thermal (and other) noise of 
  their equipment and move a speaker. If you’re not an electrical engineer, 
  it may be difficult to grasp the problem that this poses.
 
  2) I happened on a discussion in an audio forum, where a highly-acclaimed 
  mastering engineer and voice on dither mentioned that he could hear the 
  dither kick in when he pressed a certain button in the GUI of some beta 
  software. The maker of the software had to inform him that he was mistaken 
  on the function of the button, and in fact it didn’t affect the audio 
  whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a 
  great source of info to people and is clearly excellent at what he does, 
  and everyone who works with audio runs into this at some point.) The 
  mastering engineer graciously accepted his goof.
 
  3) Mastering engineers invariably describe the differences in very 
  subjective term. While this may be a necessity, it sure makes it difficult 
  to pursue any kind of validation. From a mastering engineer to me, 
  yesterday: 'To me the truncated version sounds colder, more glassy, with 
  less richness in the bass and harmonics, and less front to back depth in 
  the stereo field.’
 
  4) 24-bit audio will almost always have a far greater random noise floor 
  than is necessary to dither, so they will be self-dithered. By “almost”, I 
  mean that very near 100% of the time. Sure, you can create exceptions, such 
  as synthetically generated simple tones, but it’s hard to imagine them 
  happening in the course of normal music making. There is nothing magic 
  about dither noise—it’s just mimicking the sort of noise that your 
  electronics generates thermally. And when mastering engineers say they can 
  hear truncation distortion at 24-bit, they don’t say “on this particular 
  brief moment, this particular recording”—they seems to say it in general. 
  It’s extremely unlikely that non-randomized truncation distortion even 
  exists for most material at 24-bit.
 
  My point is simply that I’m not going to accept that mastering engineers 
  can hear the 24th bit truncation just because they say they can.
 
 
  On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
  The following published double blind test contradicts the results of the 
  old Moran/Meyer publication in showing (a) that the differences between CD 
  and higher resolution sources is audible and (b) that failure to dither at 
  the 16th bit is also audible.
 
  http://www.aes.org/e-lib/browse.cfm?elib=17497
 
  The Moran/Meyer tests had numerous technical problems that have long been 
  discussed, some are enumerated in the above.
 
  As far as dithering at the 24th bit, I can't disagree more with a 
  conclusion that

Re: [music-dsp] Dither video and articles

2015-02-07 Thread Andrew Simper

On 7 February 2015 at 03:52, Didier Dambrin di...@skynet.be wrote:
 It was just several times the same fading in/out noise at different levels,
 just to see if you hear quieter things than I do, I thought you'd have
 guessed that.
 https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing
 (0dB, -36dB, -54dB, -66dB, -72dB, -78dB)

 Here if I make the starting noise annoying, then I hear the first 4 parts,
 until 18:00. Thus, if 0dB is my threshold of annoyance, I can't hear -72dB.

 So you hear it at -78dB? Would be interesting to know how many can, and if
 it's subjective or a matter of testing environment (the variable already
 being the 0dB annoyance starting point)

Yep, I could hear all of them, and the time I couldn't hear the hiss
any more as at the 28.7 second mark, just before the end of the file.
For reference this noise blast sounded much louder than the bass tone
that Nigel posted when both were normalised, I had my headphones amp
at -18 dB so the first noise peak was loud but not uncomfortable.

I thought it was an odd test since the test file just stopped before I
couldn't hear the LFO amplitude modulation cycles, so I wasn't sure
what you were trying to prove!

All the best,

Andy




 -Message d'origine- From: Andrew Simper
 Sent: Friday, February 06, 2015 3:21 PM
 To: A discussion list for music-related DSP
 Subject: Re: [music-dsp] Dither video and articles

 Sorry, you said until, which is even more confusing. There are
 multiple points when I hear the noise until since it sounds like the
 noise is modulated in amplitude by a sine like LFO for the entire
 file, so the volume of the noise ramps up and down in a cyclic manner.
 The last ramping I hear fades out at around the 28.7 second mark when
 it is hard to tell if it just ramps out at that point or is just on
 the verge of ramping up again and then the file ends at 28.93 seconds.
 I have not tried to measure the LFO wavelength or any other such
 things, this is just going on listening alone.

 All the best,

 Andrew Simper



 On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:

 On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:

 Just out of curiosity, until which point do you hear the noise in this
 little test (a 32bit float wav), starting from a bearable first part?


 https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing


 I hear noise immediately in that recording, it's hard to tell exactly
 the time I can first hear it since there is some latency from when I
 press play to when the sound starts, but as far as I can tell it is
 straight away. Why do you ask such silly questions?

 All the best,

 Andrew Simper

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Re: [music-dsp] Dither video and articles

2015-02-07 Thread Vicki Melchior

Hi RBJ,

Well, the point of dither is to reduce correlation between the signal and 
quantization noise.  Its effectiveness requires that the error signal has given 
properties; the mean error should be zero and the RMS error should be 
independent of the signal.  The best known examples satisfying those conditions 
are white Gaussian noise at ~ 6dB above the RMS quantization level and white 
TPDF noise  at ~3dB above the same, with Gaussian noise eliminating correlation 
entirely and TPDF dither eliminating correlation with the first two moments of 
the error distribution.   That's all textbook stuff.  There are certainly noise 
shaping algorithms that shape either the sum of white dither and quantization 
noise or the white dither and quantization noise independently, and even (to my 
knowledge) a few completely non-white dithers that are known to work, but 
determining the effectiveness of noise at dithering still requires examining 
the statistical properties of the error signal and showing th
 at the mean is 0 and the second moment is signal independent.  (I think 
Stanley Lipschitz showed that the higher moments don't matter to audibility.)

Probably there are papers around looking at analog noise in typical music 
signals and how well it works as self dither (because self dither is assumed in 
some A/D conversion) but I don't know them and would be very happy to see them. 
 The one case I know involving some degree of modeling was a tutorial on dither 
given last year in Berlin that advised against depending on self dither in 
signal processing unless the noise source was checked out thoroughly before 
hand.  Variability of amplitude, PDF and time coherence were discussed if I 
recall.

Best,
Vicki 

On Feb 6, 2015, at 9:27 PM, robert bristow-johnson wrote:

 
 
 
 
 
 
 
  Original Message 
 
 Subject: Re: [music-dsp] Dither video and articles
 
 From: Vicki Melchior vmelch...@earthlink.net
 
 Date: Fri, February 6, 2015 2:23 pm
 
 To: A discussion list for music-related DSP music-dsp@music.columbia.edu
 
 --
 
 
 
 The self dither argument is not as obvious as it may appear. To be effective 
 at dithering, the noise has to be at the right level of course but also 
 should be white and temporally constant.
  
 why does it have to be white?  or why should it?
 
 
 
 
 
 --
  
 r b-j   r...@audioimagination.com
  
 Imagination is more important than knowledge.
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book reviews, dsp 
 links
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Re: [music-dsp] Dither video and articles

2015-02-07 Thread Nigel Redmon

why does it have to be white?  or why should it?

A common and trivial dither signal for non-shaped dither is rectangular PDF 
noise through a one-pole highpass filter. In other words, instead of generating 
two random numbers and adding them together for the dither signal at each 
sample, one random number is generated, and the random number for the previous 
sample is subtracted. The idea is that it biases the noise toward the highs, 
less in the body of the music, and is a little faster computationally (which 
typically doesn’t mean a thing).


 On Feb 6, 2015, at 6:27 PM, robert bristow-johnson 
 r...@audioimagination.com wrote:
 
 
  Original Message 
 
 Subject: Re: [music-dsp] Dither video and articles
 
 From: Vicki Melchior vmelch...@earthlink.net
 
 Date: Fri, February 6, 2015 2:23 pm
 
 To: A discussion list for music-related DSP music-dsp@music.columbia.edu
 
 --
 
 
 The self dither argument is not as obvious as it may appear. To be effective 
 at dithering, the noise has to be at the right level of course but also 
 should be white and temporally constant.
  
 why does it have to be white?  or why should it?
 
 
 --
  
 r b-j   r...@audioimagination.com
  
 Imagination is more important than knowledge.

--
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Re: [music-dsp] Dither video and articles

2015-02-07 Thread Nigel Redmon

Hi Vicki,

My intuitive view of dither is this (I think you can get this point from my 
video):

After truncation, the error introduced is the truncated signal minus the 
original high resolution signal. We could analyze it statistically, but our 
ears and brain do a real good job of that. And after all, the object here is to 
satisfy our ears and brain.

Listening to the original, high-resolution signal, plus this error signal, is 
equivalent to listening to the truncated signal.

So, my question would be, given such an error signal that sounds smooth, 
pleasant, and unmodulated (hiss-like, not grating, whining, or sputtering, for 
instance): Under what circumstances would the result of adding this error 
signal to the original signal result in an unnecessarily distracting or 
unpleasant degradation of the source material? (And of course, we’re talking 
about 16-bit audio, so not an error of overpowering amplitude.)

I’m not asking this rhetorically, I’d like to know. Measurable statistical 
purity aside, if the error doesn’t sound wrong to the ear, can it still sound 
wrong added to the music? I’ve tried a bit, but so far I haven’t been able to 
convince myself that it can, so I’d appreciate it if someone else could.

Nigel


 On Feb 7, 2015, at 5:54 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 Hi RBJ,
 
 Well, the point of dither is to reduce correlation between the signal and 
 quantization noise.  Its effectiveness requires that the error signal has 
 given properties; the mean error should be zero and the RMS error should be 
 independent of the signal.  The best known examples satisfying those 
 conditions are white Gaussian noise at ~ 6dB above the RMS quantization level 
 and white TPDF noise  at ~3dB above the same, with Gaussian noise eliminating 
 correlation entirely and TPDF dither eliminating correlation with the first 
 two moments of the error distribution.   That's all textbook stuff.  There 
 are certainly noise shaping algorithms that shape either the sum of white 
 dither and quantization noise or the white dither and quantization noise 
 independently, and even (to my knowledge) a few completely non-white dithers 
 that are known to work, but determining the effectiveness of noise at 
 dithering still requires examining the statistical properties of the error 
 signal and showing th
 at the mean is 0 and the second moment is signal independent.  (I think 
 Stanley Lipschitz showed that the higher moments don't matter to audibility.)
 
 Probably there are papers around looking at analog noise in typical music 
 signals and how well it works as self dither (because self dither is assumed 
 in some A/D conversion) but I don't know them and would be very happy to see 
 them.  The one case I know involving some degree of modeling was a tutorial 
 on dither given last year in Berlin that advised against depending on self 
 dither in signal processing unless the noise source was checked out 
 thoroughly before hand.  Variability of amplitude, PDF and time coherence 
 were discussed if I recall.
 
 Best,
 Vicki 
 
 On Feb 6, 2015, at 9:27 PM, robert bristow-johnson wrote:
 
 
 
 
 
 
 
 
  Original Message 
 
 Subject: Re: [music-dsp] Dither video and articles
 
 From: Vicki Melchior vmelch...@earthlink.net
 
 Date: Fri, February 6, 2015 2:23 pm
 
 To: A discussion list for music-related DSP music-dsp@music.columbia.edu
 
 --
 
 
 
 The self dither argument is not as obvious as it may appear. To be 
 effective at dithering, the noise has to be at the right level of course 
 but also should be white and temporally constant.
 
 why does it have to be white?  or why should it?
 
 
 
 
 
 --
 
 r b-j   r...@audioimagination.com
 
 Imagination is more important than knowledge.
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book reviews, dsp 
 links
 http://music.columbia.edu/cmc/music-dsp
 http://music.columbia.edu/mailman/listinfo/music-dsp
 
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book reviews, dsp 
 links
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Re: [music-dsp] Dither video and articles


mmh, Affiliation: Meridian Audio Ltd?




-Message d'origine- 
From: Vicki Melchior

Sent: Friday, February 06, 2015 2:21 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

The following published double blind test contradicts the results of the old 
Moran/Meyer publication in showing (a) that the differences between CD and 
higher resolution sources is audible and (b) that failure to dither at the 
16th bit is also audible.


http://www.aes.org/e-lib/browse.cfm?elib=17497

The Moran/Meyer tests had numerous technical problems that have long been 
discussed, some are enumerated in the above.


As far as dithering at the 24th bit, I can't disagree more with a conclusion 
that says it's unnecessary in data handling.  Mastering engineers can hear 
truncation error at the 24th bit but say it is subtle and may require 
experience or training to pick up.  What they are hearing is not noise or 
peaks sitting at the 24th bit but rather the distortion that goes with 
truncation at 24b, and it is said to have a characteristic coloration effect 
on sound.  I'm aware of an effort to show this with AB/X tests, hopefully it 
will be published.  The problem with failing to dither at 24b is that many 
such truncation steps would be done routinely in mastering, and thus the 
truncation distortion products continue to build up.  Whether you personally 
hear it is likely to depend both on how extensive your data flow pathway is 
and how good your playback equipment is.


Vicki Melchior

On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote:


On 6/02/2015 1:50 PM, Tom Duffy wrote:

The AES report is highly controversial.

Plenty of sources dispute the findings.


Can you name some?

Ross.
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Re: [music-dsp] Dither video and articles

It was just several times the same fading in/out noise at different levels, 
just to see if you hear quieter things than I do, I thought you'd have 
guessed that.

https://drive.google.com/file/d/0B6Cr7wjQ2EPub2I1aGExVmJCNzA/view?usp=sharing
(0dB, -36dB, -54dB, -66dB, -72dB, -78dB)

Here if I make the starting noise annoying, then I hear the first 4 parts, 
until 18:00. Thus, if 0dB is my threshold of annoyance, I can't hear -72dB.


So you hear it at -78dB? Would be interesting to know how many can, and if 
it's subjective or a matter of testing environment (the variable already 
being the 0dB annoyance starting point)





-Message d'origine- 
From: Andrew Simper

Sent: Friday, February 06, 2015 3:21 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.

All the best,

Andrew Simper



On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:

On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:

Just out of curiosity, until which point do you hear the noise in this
little test (a 32bit float wav), starting from a bearable first part?

https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing


I hear noise immediately in that recording, it's hard to tell exactly
the time I can first hear it since there is some latency from when I
press play to when the sound starts, but as far as I can tell it is
straight away. Why do you ask such silly questions?

All the best,

Andrew Simper

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Re: [music-dsp] Dither video and articles

Hi Michael,

I know that you already understand this, and comment that this is for internal 
calculations, but for the sake of anyone who might misinterpret your 32-bit vs 
64-bit comment, I’ll point out that this is a situation of error feedback—the 
resulting error is much greater than the sample sizes you’re talking about, and 
can result in differences far above the 24-bit level. A simple example is the 
ubiquitous direct form I biquad, which goes all to hell in lower audio 
frequencies with 24-bit storage (unless you noise shape or increase resolution).

Nigel


 On Feb 6, 2015, at 10:24 AM, Michael Gogins michael.gog...@gmail.com wrote:
 
 Do not believe anything that is not confirmed to a high degree of
 statistical signifance (say, 5 standard deviations) by a double-blind
 test using an ABX comparator.
 
 That said, the AES study did use double-blind testing. I did not read
 the article, only the abstract, so cannot say more about the study.
 
 In my own work, I have verified with a double-blind ABX comparator at
 a high degree of statistical significance that I can hear the
 differences in certain selected portions of the same Csound piece
 rendered with 32 bit floating point samples versus 64 bit floating
 point samples. These are sample words used in internal calculations,
 not for output soundfiles. What I heard was differences in the sound
 of the same filter algorithm. These differences were not at all hard
 to hear, but they occurred in only one or two places in the piece.
 
 I have not myself been able to hear differences in audio output
 quality between CD audio and high-resolution audio, but when I get the
 time I may try again, now that I have a better idea what to listen
 for.
 
 Regards,
 Mike
 
 
 
 -
 Michael Gogins
 Irreducible Productions
 http://michaelgogins.tumblr.com
 Michael dot Gogins at gmail dot com
 
 
 On Fri, Feb 6, 2015 at 1:13 PM, Nigel Redmon earle...@earlevel.com wrote:
 Mastering engineers can hear truncation error at the 24th bit but say it is 
 subtle and may require experience or training to pick up.
 
 Quick observations:
 
 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 
 1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths 
 of a volt). Hearing capabilities aside, the converter must be able to 
 resolve this, and it must make it through the thermal (and other) noise of 
 their equipment and move a speaker. If you’re not an electrical engineer, it 
 may be difficult to grasp the problem that this poses.
 
 2) I happened on a discussion in an audio forum, where a highly-acclaimed 
 mastering engineer and voice on dither mentioned that he could hear the 
 dither kick in when he pressed a certain button in the GUI of some beta 
 software. The maker of the software had to inform him that he was mistaken 
 on the function of the button, and in fact it didn’t affect the audio 
 whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a 
 great source of info to people and is clearly excellent at what he does, and 
 everyone who works with audio runs into this at some point.) The mastering 
 engineer graciously accepted his goof.
 
 3) Mastering engineers invariably describe the differences in very 
 subjective term. While this may be a necessity, it sure makes it difficult 
 to pursue any kind of validation. From a mastering engineer to me, 
 yesterday: 'To me the truncated version sounds colder, more glassy, with 
 less richness in the bass and harmonics, and less front to back depth in 
 the stereo field.’
 
 4) 24-bit audio will almost always have a far greater random noise floor 
 than is necessary to dither, so they will be self-dithered. By “almost”, I 
 mean that very near 100% of the time. Sure, you can create exceptions, such 
 as synthetically generated simple tones, but it’s hard to imagine them 
 happening in the course of normal music making. There is nothing magic about 
 dither noise—it’s just mimicking the sort of noise that your electronics 
 generates thermally. And when mastering engineers say they can hear 
 truncation distortion at 24-bit, they don’t say “on this particular brief 
 moment, this particular recording”—they seems to say it in general. It’s 
 extremely unlikely that non-randomized truncation distortion even exists for 
 most material at 24-bit.
 
 My point is simply that I’m not going to accept that mastering engineers can 
 hear the 24th bit truncation just because they say they can.
 
 
 On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 The following published double blind test contradicts the results of the 
 old Moran/Meyer publication in showing (a) that the differences between CD 
 and higher resolution sources is audible and (b) that failure to dither at 
 the 16th bit is also audible.
 
 http://www.aes.org/e-lib/browse.cfm?elib=17497
 
 The Moran/Meyer tests had numerous technical

Re: [music-dsp] Dither video and articles

I SO agree with 4), that when it comes to recorded  not synthesized (but 
even synthesized in some cases actually - I've made additive synths and it's 
a big CPU saver to avoid processing inaudible partials) audio, room noise is 
so much above the levels we're debating, that it's a bit silly.





-Message d'origine- 
From: Nigel Redmon

Sent: Friday, February 06, 2015 7:13 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Dither video and articles

Mastering engineers can hear truncation error at the 24th bit but say it is 
subtle and may require experience or training to pick up.


Quick observations:

1) The output step size of the lsb is full-scale / 2^24. If full-scale is 
1V, then step is 0.000596046447753906V, or 0.0596 microvolt (millionths 
of a volt). Hearing capabilities aside, the converter must be able to 
resolve this, and it must make it through the thermal (and other) noise of 
their equipment and move a speaker. If you’re not an electrical engineer, it 
may be difficult to grasp the problem that this poses.


2) I happened on a discussion in an audio forum, where a highly-acclaimed 
mastering engineer and voice on dither mentioned that he could hear the 
dither kick in when he pressed a certain button in the GUI of some beta 
software. The maker of the software had to inform him that he was mistaken 
on the function of the button, and in fact it didn’t affect the audio 
whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a 
great source of info to people and is clearly excellent at what he does, and 
everyone who works with audio runs into this at some point.) The mastering 
engineer graciously accepted his goof.


3) Mastering engineers invariably describe the differences in very 
subjective term. While this may be a necessity, it sure makes it difficult 
to pursue any kind of validation. From a mastering engineer to me, 
yesterday: 'To me the truncated version sounds colder, more glassy, with 
less richness in the bass and harmonics, and less front to back depth in 
the stereo field.’


4) 24-bit audio will almost always have a far greater random noise floor 
than is necessary to dither, so they will be self-dithered. By “almost”, I 
mean that very near 100% of the time. Sure, you can create exceptions, such 
as synthetically generated simple tones, but it’s hard to imagine them 
happening in the course of normal music making. There is nothing magic about 
dither noise—it’s just mimicking the sort of noise that your electronics 
generates thermally. And when mastering engineers say they can hear 
truncation distortion at 24-bit, they don’t say “on this particular brief 
moment, this particular recording”—they seems to say it in general. It’s 
extremely unlikely that non-randomized truncation distortion even exists for 
most material at 24-bit.


My point is simply that I’m not going to accept that mastering engineers can 
hear the 24th bit truncation just because they say they can.



On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net 
wrote:


The following published double blind test contradicts the results of the 
old Moran/Meyer publication in showing (a) that the differences between CD 
and higher resolution sources is audible and (b) that failure to dither at 
the 16th bit is also audible.


http://www.aes.org/e-lib/browse.cfm?elib=17497

The Moran/Meyer tests had numerous technical problems that have long been 
discussed, some are enumerated in the above.


As far as dithering at the 24th bit, I can't disagree more with a 
conclusion that says it's unnecessary in data handling.  Mastering 
engineers can hear truncation error at the 24th bit but say it is subtle 
and may require experience or training to pick up.  What they are hearing 
is not noise or peaks sitting at the 24th bit but rather the distortion 
that goes with truncation at 24b, and it is said to have a characteristic 
coloration effect on sound.  I'm aware of an effort to show this with AB/X 
tests, hopefully it will be published.  The problem with failing to dither 
at 24b is that many such truncation steps would be done routinely in 
mastering, and thus the truncation distortion products continue to build 
up.  Whether you personally hear it is likely to depend both on how 
extensive your data flow pathway is and how good your playback equipment 
is.


Vicki Melchior

On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote:


On 6/02/2015 1:50 PM, Tom Duffy wrote:

The AES report is highly controversial.

Plenty of sources dispute the findings.


Can you name some?

Ross.
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Re: [music-dsp] Dither video and articles

2015-02-06 Thread Victor Lazzarini

Yes, but note that in the case Michael is reporting, all filters have 
double-precision coeffs and data storage. It is only when passing samples 
between unit generators that the difference lies (either single or
double precision is used). Still, I believe that 
there can be audible differences.

Victor Lazzarini
Dean of Arts, Celtic Studies, and Philosophy
Maynooth University
Ireland

 On 6 Feb 2015, at 18:43, Ethan Duni ethan.d...@gmail.com wrote:
 
 Thanks for the reference Vicki
 
 What they are hearing is not noise or peaks sitting at the 24th
 bit but rather the distortion that goes with truncation at 24b, and
 it is said to have a characteristic coloration effect on sound.  I'm
 aware of an effort to show this with AB/X tests, hopefully it will be
 published.
 
 I'm skeptical, but definitely hope that such a test gets undertaken and
 published. Would be interesting to have some real data either way.
 
 The problem with failing to dither at 24b is that many such truncation
 steps would be done routinely in mastering, and thus the truncation
 distortion products continue to build up.
 
 Hopefully everyone agrees that the questions of what is appropriate for
 intermediate processing and what is appropriate for final distribution are
 quite different, and that substantially higher resolutions (and probably
 including dither) are indicated for intermediate processing. As Michael
 Goggins says:
 
 In my own work, I have verified with a double-blind ABX comparator at
 a high degree of statistical significance that I can hear the
 differences in certain selected portions of the same Csound piece
 rendered with 32 bit floating point samples versus 64 bit floating
 point samples. These are sample words used in internal calculations,
 not for output soundfiles. What I heard was differences in the sound
 of the same filter algorithm. These differences were not at all hard
 to hear, but they occurred in only one or two places in the piece.
 
 Indeed, it is not particularly difficult to cook up filter
 designs/algorithms that will break any given finite internal resolution. At
 some point those filter designs become pathological, but there are plenty
 of reasonable cases where 32 bit float internal precision is insufficient.
 Note that a 32-bit float only has 24 bits of mantissa, which is 8 bits less
 than is typically used in embedded fixed-point implementations (for
 sensitive components like filter guts, I mean). So even very standard stuff
 that has been around for decades in the fixed-point world will break if
 implemented naively in 32 bit float.
 
 E
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Re: [music-dsp] Dither video and articles

Isn't it generally agreed that truncation noise is correlated with the signal?

“Is correlated”? No, but it can be.

First, if there is enough noise in the signal before truncation, then it’s 
dithered by default—no correlation.

Second, if the signal is sufficiently complex, it seems, then there is no 
apparent correlation. See my video (https://www.youtube.com/watch?v=KCyA6LlB3As 
https://www.youtube.com/watch?v=KCyA6LlB3As) where I show a 32-bit float mix, 
truncated to 8-bit, nulled, and boosted +24 dB. There is no apparent 
correlation till the very end, even though the noise floor is not sufficient to 
self-dither.


 On Feb 6, 2015, at 10:42 AM, Tom Duffy tdu...@tascam.com wrote:
 
 Isn't it generally agreed that truncation noise is correlated with the
 signal?
 The human ear is excellent at picking up on correlation, so a system
 that introduces multiple correlated (noise) signals may reach a point
 where it is perceptual, even if the starting point is a 24 bit signal.
 
 I would believe this to be an explanation for why ProTools early hardware 
 mixers were regarded as having problems - they used 24bit
 fixed point DSPs, coupled with fixed bit headroom management may
 have introduced truncation noise at a level higher than the 24 bit
 noise floor.
 
 Also, the dither noise source itself needs to be investigated.
 Studies have shown that a fixed repeated buffer of pre-generated white
 noise is immediately obvious (and non-pleasing) to the listener up to
 several hundred ms long - if that kind of source was used as a dither
 signal, the self correlation becomes even more problematic.
 Calculated a new PRDG value for each sample is expensive, which
 is why a pre-generated buffer is attractive to the implementor.
 
 ---
 Tom.
 
 On 2/6/2015 10:32 AM, Victor Lazzarini wrote:
 Quite. This conversation is veering down the vintage wine tasting alley.
 
 Victor Lazzarini
 Dean of Arts, Celtic Studies, and Philosophy
 Maynooth University
 Ireland
 
 On 6 Feb 2015, at 18:13, Nigel Redmon earle...@earlevel.com wrote:
 
 Mastering engineers can hear truncation error at the 24th bit but say it is 
 subtle and may require experience or training to pick up.
 
 Quick observations:
 
 1) The output step size of the lsb is full-scale / 2^24. If full-scale is 1V, 
 then step is 0.000596046447753906V, or 0.0596 microvolt (millionths of a 
 volt). Hearing capabilities aside, the converter must be able to resolve 
 this, and it must make it through the thermal (and other) noise of their 
 equipment and move a speaker. If you’re not an electrical engineer, it may be 
 difficult to grasp the problem that this poses.
 
 2) I happened on a discussion in an audio forum, where a highly-acclaimed 
 mastering engineer and voice on dither mentioned that he could hear the 
 dither kick in when he pressed a certain button in the GUI of some beta 
 software. The maker of the software had to inform him that he was mistaken on 
 the function of the button, and in fact it didn’t affect the audio 
 whatsoever. (I’ll leave his name out, because it’s immaterial—the guy is a 
 great source of info to people and is clearly excellent at what he does, and 
 everyone who works with audio runs into this at some point.) The mastering 
 engineer graciously accepted his goof.
 
 3) Mastering engineers invariably describe the differences in very subjective 
 term. While this may be a necessity, it sure makes it difficult to pursue any 
 kind of validation. From a mastering engineer to me, yesterday: 'To me the 
 truncated version sounds colder, more glassy, with less richness in the bass 
 and harmonics, and less front to back depth in the stereo field.’
 
 4) 24-bit audio will almost always have a far greater random noise floor than 
 is necessary to dither, so they will be self-dithered. By “almost”, I mean 
 that very near 100% of the time. Sure, you can create exceptions, such as 
 synthetically generated simple tones, but it’s hard to imagine them happening 
 in the course of normal music making. There is nothing magic about dither 
 noise—it’s just mimicking the sort of noise that your electronics generates 
 thermally. And when mastering engineers say they can hear truncation 
 distortion at 24-bit, they don’t say “on this particular brief moment, this 
 particular recording”—they seems to say it in general. It’s extremely 
 unlikely that non-randomized truncation distortion even exists for most 
 material at 24-bit.
 
 My point is simply that I’m not going to accept that mastering engineers can 
 hear the 24th bit truncation just because they say they can.
 
 
 On Feb 6, 2015, at 5:21 AM, Vicki Melchior vmelch...@earthlink.net wrote:
 
 The following published double blind test contradicts the results of the old 
 Moran/Meyer publication in showing (a) that the differences between CD and 
 higher resolution sources is audible and (b) that failure to dither at the 
 16th bit is also audible.

Re: [music-dsp] Dither video and articles

2015-02-06 Thread Michael Gogins

This was done before John ffitch (I believe it was he) changed the
filter samples in even the single-precision version of Csound to use
double-precision. And I think this change may have been made as a
result of my report.

Regards,
Mike

-
Michael Gogins
Irreducible Productions
http://michaelgogins.tumblr.com
Michael dot Gogins at gmail dot com


On Fri, Feb 6, 2015 at 2:04 PM, Victor Lazzarini
victor.lazzar...@nuim.ie wrote:
 Yes, but note that in the case Michael is reporting, all filters have 
 double-precision coeffs and data storage. It is only when passing samples 
 between unit generators that the difference lies (either single or
 double precision is used). Still, I believe that
 there can be audible differences.

 Victor Lazzarini
 Dean of Arts, Celtic Studies, and Philosophy
 Maynooth University
 Ireland

 On 6 Feb 2015, at 18:43, Ethan Duni ethan.d...@gmail.com wrote:

 Thanks for the reference Vicki

 What they are hearing is not noise or peaks sitting at the 24th
 bit but rather the distortion that goes with truncation at 24b, and
 it is said to have a characteristic coloration effect on sound.  I'm
 aware of an effort to show this with AB/X tests, hopefully it will be
 published.

 I'm skeptical, but definitely hope that such a test gets undertaken and
 published. Would be interesting to have some real data either way.

 The problem with failing to dither at 24b is that many such truncation
 steps would be done routinely in mastering, and thus the truncation
 distortion products continue to build up.

 Hopefully everyone agrees that the questions of what is appropriate for
 intermediate processing and what is appropriate for final distribution are
 quite different, and that substantially higher resolutions (and probably
 including dither) are indicated for intermediate processing. As Michael
 Goggins says:

 In my own work, I have verified with a double-blind ABX comparator at
 a high degree of statistical significance that I can hear the
 differences in certain selected portions of the same Csound piece
 rendered with 32 bit floating point samples versus 64 bit floating
 point samples. These are sample words used in internal calculations,
 not for output soundfiles. What I heard was differences in the sound
 of the same filter algorithm. These differences were not at all hard
 to hear, but they occurred in only one or two places in the piece.

 Indeed, it is not particularly difficult to cook up filter
 designs/algorithms that will break any given finite internal resolution. At
 some point those filter designs become pathological, but there are plenty
 of reasonable cases where 32 bit float internal precision is insufficient.
 Note that a 32-bit float only has 24 bits of mantissa, which is 8 bits less
 than is typically used in embedded fixed-point implementations (for
 sensitive components like filter guts, I mean). So even very standard stuff
 that has been around for decades in the fixed-point world will break if
 implemented naively in 32 bit float.

 E
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Re: [music-dsp] Dither video and articles

So you hear all 6 too?

-Message d'origine- 
From: Richard Dobson

Sent: Friday, February 06, 2015 4:10 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Dither video and articles

On 06/02/2015 14:21, Andrew Simper wrote:

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.

Its a series of six smoothly enveloped noise bursts (slowish rise/
slower decay) the first peaking at max amplitude (so you have to be
ready to hear it as very loud!), then successively softer repeats until
at some point it is (presumably?) too quiet to be heard. Very visible in
Audacity using the Waveform (dB) display mode. So the word until is
entirely appropriate. I do recommend visual inspection of waveforms in
such situations to minimise guessing (or at least, to confirm the
guesses or otherwise). In any case, I would expect people to hear all
six, give a suitably quiet listening environment and an appropriately
generous overall playback level etc.

Richard Dobson

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Re: [music-dsp] Dither video and articles

 to choose, between the two 16-bit ones I would prefer the one
 with dither but put through a make mono plugin, as this sounded the
 closest to the float version.
 
 All the best,
 
 Andy
 
 -- cytomic -- sound music software --
 
 
 On 5 February 2015 at 16:46, Nigel Redmon earle...@earlevel.com wrote:
 Hmm, I thought that would let you save the page source (wave file)…Safari 
 creates the file of the appropriate name and type, but it stays at 0 
 bytes…OK, I put up and index page—do the usual right-click to save the 
 field to disk if you need to access the files directly:
 
 http://earlevel.com/temp/music-dsp/
 
 
 On Feb 5, 2015, at 12:13 AM, Nigel Redmon earle...@earlevel.com wrote:
 
 OK, here’s my new piece, I call it Diva bass—to satisfy your request for 
 me to make something with truncation distortion apparent. (If it bother 
 you that my piece is one note, imagine that this is just the last note of 
 a longer piece.)
 
 I spent maybe 30 seconds getting the sound—opened Diva (default 
 “minimoog” modules), turn the mixer knobs down except for VCO 1, set 
 range to 32’, waveform to triangle, max release on the VCA envelope.
 
 In 32-bit float glory:
 
 http://earlevel.com/temp/music-dsp/Diva%20bass%2032-bit%20float.wav
 
 Truncated to 16-bit, no dither (Quan Jr plug-in, Digital Performer), 
 saved to 16-bit wave file:
 
 http://earlevel.com/temp/music-dsp/Diva%20bass%2016-bit%20truncated.wav
 
 You’ll have to turn your sound system up, not insanely loud, but loud. (I 
 said that this would be the case before.) I can hear it, and I know 
 engineers who monitor much louder, routinely, than I’m monitoring to hear 
 this. My Equator Q10s are not terribly high powered, and I’m not adding 
 any other gain ahead of them in order to boost the quiet part.
 
 If you want to hear the residual easily (32-bit version inverted, summed 
 with 16-bit truncated, the result with +40 dB gain via Trim plug-in):
 
 http://earlevel.com/temp/music-dsp/Diva%20bass%2016-bit%20truncated%20residual%20+40dB.wav
 
 I don’t expect the 16-bit truncated version to bother you, but it does 
 bother some audio engineers. Here's 16-bit dithered version, for 
 completeness, so that you can decide if the added noise floor bothers you:
 
 http://earlevel.com/temp/music-dsp/Diva%20bass%2016-bit%20dithered.wav
 
 
 
 On Feb 4, 2015, at 1:10 PM, Didier Dambrin di...@skynet.be wrote:
 
 Yes, I disagree with the always. Not always needed means it's 
 sometimes needed, my point is that it's never needed, until proven 
 otherwise. Your video proves that sometimes it's not needed, but not 
 that sometimes it's needed.
 
 
 
 -Message d'origine- From: Nigel Redmon
 Sent: Wednesday, February 04, 2015 6:51 PM
 To: A discussion list for music-related DSP
 Subject: Re: [music-dsp] Dither video and articles
 
 I totally understood the point of your video, that dithering to 16bit 
 isn't always needed - but that's what I disagree with.
 
 Sorry, Didier, I’m confused now. I took from your previous message that 
 you feel 16-bit doesn’t need to be dithered (dithering to 16bit will 
 never make any audible difference”). Here you say that you disagree with 
 dithering to 16bit isn't always needed”. In fact, you are saying that 
 it’s never needed—you disagree because “isn’t always needed” implies 
 that it is sometimes needed—correct?
 
 
 On Feb 4, 2015, at 5:06 AM, Didier Dambrin di...@skynet.be wrote:
 
 Then, it’s no-win situation, because I could EASILY manufacture a bit 
 of music that had significant truncation distortion at 16-bit.
 
 Please do, I would really like to hear it.
 
 I have never heard truncation noise at 16bit, other than by playing 
 with levels in a such a way that the peaking parts of the rest of the 
 sound would destroy your ears or be very unpleasant at best. (you say 
 12dB, it's already a lot)
 
 I totally understood the point of your video, that dithering to 16bit 
 isn't always needed - but that's what I disagree with.
 
 
 
 -Message d'origine- From: Nigel Redmon
 Sent: Wednesday, February 04, 2015 10:59 AM
 To: A discussion list for music-related DSP
 Subject: Re: [music-dsp] Dither video and articles
 
 Hi Didier—You seem to find contradictions in my choices because you are 
 making the wrong assumptions about what I’m showing and saying.
 
 First, I’m not steadfast that 16-bit dither is always needed—and in 
 fact the point of the video was that I was showing you (the viewers) 
 how you can judge it objectively for yourself (and decide whether you 
 want to dither). This is a much better way that the usual that I hear 
 from people, who often listen to the dithered and non-dithered results, 
 and talk about the soundstage collapsing without dither, “brittle” 
 versus “transparent , etc.
 
 But if I’m to give you a rule of thumb, a practical bit of advice that 
 you can apply without concern that you might be doing something wrong 
 in a given circumstance, that advice is “always dither 16-bit 
 reductions

Re: [music-dsp] Dither video and articles

2015-02-06 Thread Vicki Melchior

The self dither argument is not as obvious as it may appear.  To be effective 
at dithering, the noise has to be at the right level of course but also should 
be white and temporally constant.  The noise floors present in music data 
normally come from the self noise of the analog components used in recording 
and are composites of a number of noise PDFs.  For example, a graph in a second 
paper by the same group (cited below if wanted) shows spectra of the measured 
noise floors from around a dozen recordings.  The noise spectra are composites 
with the lower frequencies clearly 1/f noise and the upper frequencies summing 
closer to flat.  Whether composite noise of this sort is both temporally 
continuous and white enough to be relied on for dither needs to be shown; it's 
been shown under at least some circumstances (not in these papers) that a 
truncation distortion spectrum can be produced and measured when signals are 
truncated to 24b.  

I'm not saying the self dither argument is necessarily wrong; but it needs 
verification as to when and where it is reliably valid.   If 24b truncation 
turns out to be demonstrably audible in an AB/X, then the self dither idea 
clearly needs to be rethought.

Vicki Melchior

(graph mentioned is fig 8 in this paper:   
http://www.aes.org/e-lib/browse.cfm?elib=17501)

On Feb 6, 2015, at 2:20 PM, Nigel Redmon wrote:

 First, if there is enough noise in the signal before truncation, then it’s 
 dithered by default—no correlation.

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Re: [music-dsp] Dither video and articles

2015-02-06 Thread robert bristow-johnson








 Original Message 

Subject: Re: [music-dsp] Dither video and articles

From: Vicki Melchior vmelch...@earthlink.net

Date: Fri, February 6, 2015 2:23 pm

To: A discussion list for music-related DSP music-dsp@music.columbia.edu

--



 The self dither argument is not as obvious as it may appear. To be effective 
 at dithering, the noise has to be at the right level of course but also 
 should be white and temporally constant.
�
why does it have to be white?� or why should it?





--
�
r b-j � � � � � � � � � r...@audioimagination.com
�
Imagination is more important than knowledge.
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Re: [music-dsp] Dither video and articles

2015-02-06 Thread Andrew Simper

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.

All the best,

Andrew Simper



On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote:
 On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:
 Just out of curiosity, until which point do you hear the noise in this
 little test (a 32bit float wav), starting from a bearable first part?

 https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing

 I hear noise immediately in that recording, it's hard to tell exactly
 the time I can first hear it since there is some latency from when I
 press play to when the sound starts, but as far as I can tell it is
 straight away. Why do you ask such silly questions?

 All the best,

 Andrew Simper
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Re: [music-dsp] Dither video and articles

2015-02-06 Thread Andrew Simper

On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote:
 Just out of curiosity, until which point do you hear the noise in this
 little test (a 32bit float wav), starting from a bearable first part?

 https://drive.google.com/file/d/0B6Cr7wjQ2EPucjFCSUhGNkVRaUE/view?usp=sharing

I hear noise immediately in that recording, it's hard to tell exactly
the time I can first hear it since there is some latency from when I
press play to when the sound starts, but as far as I can tell it is
straight away. Why do you ask such silly questions?

All the best,

Andrew Simper
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Re: [music-dsp] Dither video and articles

2015-02-06 Thread Vicki Melchior

The following published double blind test contradicts the results of the old 
Moran/Meyer publication in showing (a) that the differences between CD and 
higher resolution sources is audible and (b) that failure to dither at the 16th 
bit is also audible.  

http://www.aes.org/e-lib/browse.cfm?elib=17497

The Moran/Meyer tests had numerous technical problems that have long been 
discussed, some are enumerated in the above.  

As far as dithering at the 24th bit, I can't disagree more with a conclusion 
that says it's unnecessary in data handling.  Mastering engineers can hear 
truncation error at the 24th bit but say it is subtle and may require 
experience or training to pick up.  What they are hearing is not noise or peaks 
sitting at the 24th bit but rather the distortion that goes with truncation at 
24b, and it is said to have a characteristic coloration effect on sound.  I'm 
aware of an effort to show this with AB/X tests, hopefully it will be 
published.  The problem with failing to dither at 24b is that many such 
truncation steps would be done routinely in mastering, and thus the truncation 
distortion products continue to build up.  Whether you personally hear it is 
likely to depend both on how extensive your data flow pathway is and how good 
your playback equipment is.  

Vicki Melchior

On Feb 5, 2015, at 10:01 PM, Ross Bencina wrote:

 On 6/02/2015 1:50 PM, Tom Duffy wrote:
 The AES report is highly controversial.
 
 Plenty of sources dispute the findings.
 
 Can you name some?
 
 Ross.
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Re: [music-dsp] Dither video and articles

2015-02-06 Thread Richard Dobson


On 06/02/2015 14:21, Andrew Simper wrote:

Sorry, you said until, which is even more confusing. There are
multiple points when I hear the noise until since it sounds like the
noise is modulated in amplitude by a sine like LFO for the entire
file, so the volume of the noise ramps up and down in a cyclic manner.
The last ramping I hear fades out at around the 28.7 second mark when
it is hard to tell if it just ramps out at that point or is just on
the verge of ramping up again and then the file ends at 28.93 seconds.
I have not tried to measure the LFO wavelength or any other such
things, this is just going on listening alone.




Its a series of six smoothly enveloped noise bursts (slowish rise/ 
slower decay) the first peaking at max amplitude (so you have to be 
ready to hear it as very loud!), then successively softer repeats until 
at some point it is (presumably?) too quiet to be heard. Very visible in 
Audacity using the Waveform (dB) display mode. So the word until is 
entirely appropriate. I do recommend visual inspection of waveforms in 
such situations to minimise guessing (or at least, to confirm the 
guesses or otherwise). In any case, I would expect people to hear all 
six, give a suitably quiet listening environment and an appropriately 
generous overall playback level etc.


Richard Dobson


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Re: [music-dsp] Dither video and articles