Thanks to all the participants in this thread, I hope it was at least a
little educational, except maybe for some that seem to take everything
as a test to their imaginations of themselves being little computers,
and not human being with normal associations and lasting affections for
serious
This happened to me about ten days ago.
Steven Cook.
-Original Message-
From: Charles Z Henry
Sent: Thursday, August 13, 2015 8:04 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Non-linearity or filtering
On Thu, Aug 13, 2015 at 1:05 PM, Tom Duffy tdu
On 13/08/2015, Peter S peter.schoffhau...@gmail.com wrote:
Bonus experiment: try to see if you can hear the difference between
sine_fadeout16_noise.wav and sine_fadeout8_noise.wav in a blind ABX
test. If not, then having extra bits of noise make zero sense.
I did a blind ABX test between
WTF, who is trying to unsubscribe me:
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music-dsp@music.columbia.edu mailing list.
Here's an experiment that I always wanted to do:
The dither added is 1 bit (or 2 if doing TPDF),
so generating it from a PRNG is easy, you get
one bit at a time, and the bits are all completely
uncorrelated to each other - white noise spectrum.
When implemented in DSP, sometimes you get a
PRNG
On 2015-08-09, robert bristow-johnson wrote:
1) a dithered sigma-delta converter is typically better quality than
one without dithering
Correct.
there is and always had been **some** discussion and controversy about
that every time i seen it discussed at an AES convention. i remember
On Thu, Aug 13, 2015 at 1:05 PM, Tom Duffy tdu...@tascam.com wrote:
WTF, who is trying to unsubscribe me:
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On 8/10/15 10:02 AM, Peter S wrote:
On 10/08/2015, robert bristow-johnsonr...@audioimagination.com wrote:
the thing that i *think* Peter is missing in this is the same as some of
the early manufacturers when they truncated the 30-bit words (or
whatever they had in the decimation filters) to 18
On 13/08/2015, robert bristow-johnson r...@audioimagination.com wrote:
Are you *sure* truncating adds extra quantization noise?
[sigma-delta modulator][decimation filter][quantizer]---
adds extra quantization noise to:
[sigma-delta modulator][decimation
robert bristow-johnson wrote:
On 8/9/15 6:23 PM, Sampo Syreeni wrote:
1) a dithered sigma-delta converter is typically better quality than
one without dithering
Correct.
there is and always had been **some** discussion and controversy about
that every time i seen it discussed at an AES
On 8/10/2015 10:49 AM, Peter S wrote:
On 10/08/2015, Sampo Syreeni de...@iki.fi wrote:
Do notice that we're in the business of producing audio systems and
software. Not all of them are meant for pure human consumption. For
example, in audio forensics work, you'd like your signal chain to be
On 10/08/2015, Sampo Syreeni de...@iki.fi wrote:
Correct. But notice that the main dither talked about in the papers is
always done within the quantization loop, so that its power spectrum
becomes dependent on the loop's high order highpass response. Its
spectrum can and will be driven off
On 10/08/2015, robert bristow-johnson r...@audioimagination.com wrote:
the thing that i *think* Peter is missing in this is the same as some of
the early manufacturers when they truncated the 30-bit words (or
whatever they had in the decimation filters) to 18 meaningful bits.
that simply adds
What I don't understand, is why don't you guys pay attention. I showed
all these already using various demonstrations and tests, so having to
repeat the same over and over again feels quite pointless...
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On 2015-07-23, Peter S wrote:
LSB Dithering in MASH Delta–Sigma D/A Converters
http://petrified.ucsd.edu/~ispg-adm/pubs/Pamarti_TCAS1_200704.pdf
This is a nice list of references, some of which even I hadn't seen
before. Thank you!
1) a dithered sigma-delta converter is typically better
On 8/9/15 6:23 PM, Sampo Syreeni wrote:
1) a dithered sigma-delta converter is typically better quality than
one without dithering
Correct.
there is and always had been **some** discussion and controversy about
that every time i seen it discussed at an AES convention. i remember
hearing
Again, I don't respond to certain derailers I at some point don't miss
content-wise when not reading.
I started this thread, like quite a while ago I coined the basics of
sampling on this list, because as both practically and theoretically
inclined (and talented and educated) I felt a lot of
On 27/07/2015, Steven Cook stevenpaulc...@tiscali.co.uk wrote:
I *think* I've got it now, but just in case, could you explain the whole
thing again from the start because I wasn't listening?
Depends, how much will you pay?
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Said in other words, by adding extra bits of noise, you gain pretty
much nothing. You could create a 48-bit, 64-bit, 128-bit etc.
converter, and while technically you have a 48/64/128-bit converter,
in reality, all you do is just add extra bits of useless noise. Unless
you use cryogenics and
: [music-dsp] Non-linearity or filtering
Said in other words, by adding extra bits of noise, you gain pretty
much nothing. You could create a 48-bit, 64-bit, 128-bit etc.
converter, and while technically you have a 48/64/128-bit converter,
in reality, all you do is just add extra bits of useless noise
On 27/07/2015, Peter S peter.schoffhau...@gmail.com wrote:
http://morpheus.spectralhead.com/wav/noise120db_vs_sine160db_norm.wav
Sorry, this link gave a 404 error.
Fixed now.
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music-dsp@music.columbia.edu
.
-Original Message- From: Peter S
Sent: Monday, July 27, 2015 2:43 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Non-linearity or filtering
Said in other words, by adding extra bits of noise, you gain pretty
much nothing. You could create a 48-bit, 64-bit, 128-bit etc
On 27/07/2015, Peter S peter.schoffhau...@gmail.com wrote:
Mixture of -24 dB noise floor and -48 dB sine wave, at 16 bits precision:
http://morpheus.spectralhead.com/wav/noise24db_vs_sine48db.wav
Same thing converted to 8 bit:
On 25/07/2015, Tom Duffy tdu...@tascam.com wrote:
You didn't change the bandwidth.
I did. Probably you mean the 'sampling rate'.
If the target signal is max 30Hz and you have a 192kHz sampler, you low
pass
at 2x your max frequency (60Hz, but lets say 100Hz for convenience) using
a brick
On 23/07/2015, Peter S peter.schoffhau...@gmail.com wrote:
Robert, what you say, simply makes no sense. You speak about imaginary
converters with imaginary components that do not exist in the real
world.
If your imaginary converters were real, then we could simply have 138
dB dynamic range
So, just to add further spice to this already hot mixture:
127dB 768kHz/32-bit 2ch Premium ADC
http://www.akm.com/akm/en/product/detail/0055/
and, just a tad slower,
Analog Devices AD7177-2 32-Bit Sigma-Delta ADC
On 25/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
and that's not counting noise-shaping.
Since noise shaping merely changes the shape of the noise floor,
that's pretty much irrelevant. Noise shaping is equivalent to applying
a filter to the noise floor, for example a first
Peter,
I've been a lurker on this list for many years, and I've learned a lot
from the conversations. I've learned a great deal from your posts on
this topic.
Let me point out, though, that *the list doesn't waste your time*. You
choose to waste your time or to use it ways you consider
Okay, a few more thoughts:
On 23/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
okay, since there is no processing, just passing the signal from A/D to
D/A converter, there is only one quantization operation, at the A/D.
That's only true *if* it's a non-dithered converter
On 7/25/15 10:57 AM, Tom Duffy wrote:
You didn't change the bandwidth.
If the target signal is max 30Hz and you have a 192kHz sampler, you
low pass
at 2x your max frequency (60Hz, but lets say 100Hz for convenience) using
a brick wall digital filter (processed at 192kHz). Then you do a
You didn't change the bandwidth.
If the target signal is max 30Hz and you have a 192kHz sampler, you low pass
at 2x your max frequency (60Hz, but lets say 100Hz for convenience) using
a brick wall digital filter (processed at 192kHz). Then you do a
downsampling
of the signal from 192kHz to
On 25/07/2015, Peter S peter.schoffhau...@gmail.com wrote:
On 23/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
depends on what we have available for sample rates. essentially we are
only limited by the laws in Information Theory. if i have a 192 kHz
system and i only need
Since there's this contant behavioral pattern on this mailing list,
namely that irregardless of the topic, there's always some person who
basically has no clue, but tries to prove me wrong and teach me a
lesson, it's getting extremely boring.
Yes, those comments were excellent quality, if you
Awesome collection of links on dither and noise shaping, thanks.
And talking of noise, chaps, please do keep it civil and show respect to one
another, its getting a
bit noisyin the list.
best 2 all
Andy
On 23 July 2015 at 21:38 Peter S peter.schoffhau...@gmail.com wrote:
On 23/07/2015,
After checking at least the first half dozen papers I linked, it
should be apparent that
1) a dithered sigma-delta converter is typically better quality than
one without dithering
2) dithering means adding noise to the signal (usually white noise, or
a modified spectrum via noise shaping)
3)
On 7/23/15 4:38 PM, Peter S wrote:
...
https://en.wikipedia.org/wiki/The_Paradox_of_Choice
You're welcome.
http://www.imdb.com/title/tt1386011/
--
r b-j r...@audioimagination.com
Imagination is more important than knowledge.
--
dupswapdrop -- the music-dsp
On Thu, Jul 23, 2015 at 8:17 PM, Peter S peter.schoffhau...@gmail.com
wrote:
There's even literature specifically on dithering in A/D, example:
Dithering in Analog-to-digital Conversion
http://www.e2v.com/shared/content/resources/File/documents/broadband-data-converters/doc0869B.pdf
That
*snores*
Please have a civil conversation, and don't feed the trolls. If this
isn't fun, what's it good for?
A simple rhyme to remember: Make it moot and don't refute!
Nobody reads every email on a mailing list. If you have a personal
problem, consider a technical solution: email filtering
On 7/23/15 1:12 AM, Peter S wrote:
On 23/07/2015, robert bristow-johnsonr...@audioimagination.com wrote:
okay, since there is no processing, just passing the signal from A/D to
D/A converter, there is only one quantization operation, at the A/D. if
it's an old-fashioned conventional A/D, the
On 23/07/2015, Peter S peter.schoffhau...@gmail.com wrote:
sorry to point out, but also having invalid assumptions - tell me,
have you ever tried to record something with no input using a 24-bit
converter? Have you ever looked what you have at the lowest bits? Do
you understand what the term
On 23/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
Peter, do you know how a sigma-delta (or delta-sigma, people don't
always agree on the semantics) converter works?
like how a sigma-delta modulator works? oversampling? possible
dithering? noise-shaping? decimation
If noise shaping with oversampling actually worked to eliminate the
noise floor, then 24-bit sound cards would have 160 bit dynamic range.
Yet for *some* reason, they have 110-115 dB dynamic range instead, not
even approaching the theoretical 144 dB range of a 24 bit signal.
Think about that.
--
Robert, what you say, simply makes no sense. You speak about imaginary
conveters with imaginary components that do not exist in the real
world.
If your imaginary converters were real, then we could simply have 138
dB dynamic range with dithering in any 24-bit sound card, no noise
shaping would
On 23/07/2015, Theo Verelst theo...@theover.org wrote:
I'm not answering much to treacherous psychopaths (from the use of words
and the content of the communication here, anyway)
Okay, I think this was the last case when I tried to teach you
anything about digital signal processing or waste my
On 23/07/2015, Johannes Taelman johannes.tael...@gmail.com wrote:
RBJ's comments are of excellent quality, and are not about obsolete or
imaginary technologies.
Also, 16-bit converters are pretty much obsolete (if you want
high-quality audio).
Also, 24-bit converters with near zero noise floor
robert bristow-johnson wrote:
Peter...
...
it's you that are applying concepts of the old or conventional
converters here.
..
I'm not answering much to treacherous psychopaths (from the use of words
and the content of the communication here, anyway) because that doesn't
contribute much, and
Noise shaping is a filtering process that shapes the spectral energy
of quantization error, typically to either de-emphasise frequencies to
which the ear is most sensitive or separate the signal and noise bands
completely. If dither is used, its final spectrum depends on whether
it is added inside
Peter,
RBJ's comments are of excellent quality, and are not about obsolete or
imaginary technologies.
Try to learn from his comments, that would gain you more respect here
rather than trying to prove him wrong, really!
On Thu, Jul 23, 2015 at 7:16 PM, Peter S peter.schoffhau...@gmail.com
wrote:
On 7/23/15 3:36 AM, Peter S wrote:
Also if you fail to notice that the current year is 2015, and the
rules you learned 20 years ago for 8-bit and 16-bit converters do not
necessarily apply for today's typical 24-bit converters (that usually
have several bits of noise in the lowest bits),
Here is a somewhat random selection of 24-bit sound cards with SNR
data included:
http://sound-cards-review.toptenreviews.com/
Output SNR:
a) 124 dB
b) 124 dB
c) 109 dB
d) 112 dB
e) 117 dB
f) 109 dB
g) 100 dB
h) 113 dB
Input SNR:
a) 118 dB
b) 118 dB
d) 98 dB
e) 115 dB
g) 100 dB
h) 113 dB
Let's
This is painful to follow.
Can we at least go back to basics and not keep changing the
meaning of well known concepts and vocabulary.
re, sound card performance:
What is SNR, how is it measured, and why does that matter?
e.g.
http://www.cse.psu.edu/~chip/course/analog/lecture/SFDR1.pdf
There
i wrote:
the *major* component of audible noise is coming from the numerical processes
inside the codec
On 7/23/15 12:43 PM, Peter S wrote:
Seriously, where do you get that from?
well, i take it that the answer to the question i asked is no.
so there are a few docs on the web like at
On 23/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
exactly what audio codecs are you using? if they're sigma-delta (which
are, now-a-daze, nearly ubiquitous in audio), the major component of the
noise floor is actually from the operation of the converter and has a
Okay, I'll quit this discussion. It's quite pointless anyways -
there's not very much you can do to improve the reconstruction of your
soundcard, at least nothing that would be as good as buying a better
sound card.
Also if you fail to notice that the current year is 2015, and the
rules you
Hi Uli,
how well did you match the summing resistors and how much signal amplitude was
left?
Note that with a 0.1dB (~1% tolerance resistors) level difference between the
two signals, the difference would still have an amplitude about 40dB below the
original signals. You can get to -60dB with
Of course I have tried to match the resistors. But then you will also
recognize that there are gain differences between the channels. So I also
ended up with further trials of matching including trimpots. At the end I
could fine-tune by a trimpot and minimize the sum signal.
Then I tried to
On 22/07/2015, Theo Verelst theo...@theover.org wrote:
distortion figures indicate. Which should be a lot better than .1 dB
which would imply an error of over 1 percent, which wouldn't be very
good for a 50s HiFi system. But the specified harmonic distortion of a
lot of well known DACs sure
You have your signal S. When you digitize that signal, you add the
noise floor of the ADC (among other noises), let's call it N1. When
you reconstruct the signal, you add the noise floor of the DAC (among
other noises), let's call that N2. So you have
S + N1 + N2
Then you subtract the
On 23/07/2015, robert bristow-johnson r...@audioimagination.com wrote:
okay, since there is no processing, just passing the signal from A/D to
D/A converter, there is only one quantization operation, at the A/D. if
it's an old-fashioned conventional A/D, the quantization operation is,
So... if you sell your car and buy the most expensive sound card you
can get, your readings may improve by 10-20 decibels.
By doing some digital voodoo, you may increase your match between the
original and the resonstructed signal by a few decibels... But beyond
that, there's no other way than to
Uli Brueggemann wrote:
...
My simple assumption was: if the DAC is a 24 bit DAC it should be possible
to get down to e.g. -90 dB with the sum of the signals. But it seems to be
quite challenging.
It may be wideband (i.e. including sub-sonic and super-sonic) noise that
keeps the sum to about
Theo,
this reminds me on a simple test where I have never got a desired result.
Take a digital signal (a sine wave or your saw wave), send it thru a DAC.
For the second channel take the inverse wave. Add the DAC outputs e.g. by a
resistor network and try to get zero.
The digital signals add
So in short why this won't work well:
Trying to correct a highpass filter, you need an inverse filter that
has a gain of infinity at DC (since the highpass filter has gain of
-infinity at DC). Problem is, -infinity + infinity != zero, so you
likely end up with a signal that has increasingly
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