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<div class="moz-cite-prefix">On 2020-02-11 1:28 p.m., Bob Katz via
ProAudio wrote:<br>
</div>
<blockquote type="cite"
cite="mid:2fc517e1-69bb-adf3-2893-d987a409b389@digido.com">
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<p>Come on, guys.... We've been down this road in this reflector
several times. You may recall at least one of the tests I worked
on with some of you many years ago. I've subsequently performed
differently-designed tests designed to try to settle the issue
of "bandwidth", and each time the listening tests lead to the
same conclusions:<br>
</p>
<p>Thus I am 99.9999999% convinced that the sonic differences
between sample rates are not due to the bandwidth, but rather to
the performance of the converters themselves. Unfortunately, the
bandwidth mafia at HD Tracks keeps maintaining the illusion that
what we can see has anything to do with what we can hear. And I
hope that JJ does not make his application because it will
continue to mislead the public. <br>
</p>
</blockquote>
Bob, I thank you for saying that. For some reason, the archival
community -- based mostly on the premise that 96 sounded better than
44.1 back in 2001 or so and I told them exactly that it was the
converters' filtering and topology that was making the difference,
not the actual bandwidth -- decided on the minimum standard of 96/24
for all archival work. When I get in large (for me) collections of
oral history on cassettes recorded on the $40 book sized 5-C-cell
recorders, I really, really try and talk them out of 96/24 and
suggest 48/24.<br>
<br>
Part of this trend for the last two decades has been that archivists
learn 96/24 in archiving school and they're not audio engineers.
Sadly, 96 kHz is too low to capture bias, but it does provide a more
gentle rolloff and some tape machines can go out to 35-40 kHz, so
for high-quality music I don't argue, but for the aforementioned
cassettes I tell people, "you want to use twice the data space for
the project for no benefit?" I get one of two responses, "OK, I
understand, use 48" or "No, our standard is 96 and we want you to
use that."<br>
<br>
Goran Finnberg also agrees with this premise and has been saying it
for two decades as well.<br>
<blockquote type="cite"
cite="mid:2fc517e1-69bb-adf3-2893-d987a409b389@digido.com">
<p> </p>
<p>In Bob Stuart's paper, which is open access so you do not have
to be an AES member to download this:</p>
<p> J. R. Stuart and P. G. Craven, “The Gentle Art of Dithering”
J. Audio Eng. Soc., vol. 67, no. 5, pp. 278–299, (2019 May.).
DOI: <a class="moz-txt-link-freetext"
href="https://doi.org/10.17743/jaes.2019.0011"
moz-do-not-send="true">https://doi.org/10.17743/jaes.2019.0011</a></p>
<p>he points out on page 290 the number of decimators and
upsampling filters that occur in typical chip-based converters.
And that these stages are not dithered. And that there are fewer
of these stages when the converters are used at a higher sample
rate. Smoking gun.... <br>
</p>
In a conversation I had with him at the New York AES last year, I
told him that I discovered that audio sounds superior in many
current converters if you upsample it and reproduce it at the
higher rate. I also told him of my experiments showing that if you
start with, for example, a 96 kHz recording, downsample it to 44.1
k and then reupsample it to 96k, that it sounds identical to the
original, but the 44.1 k intermediate stage sounds worse, smaller
and less resolved. In my book, I point out: How can a second
generation in a chain sound worse than the third generation? <br>
<p>Bob Stuart's explanation for this phenomenon is the design of
the converters themselves. <br>
</p>
<p>Thus my conclusion that the DACs perform better at the higher
sample rate. Stuart explained that when the converters work at a
higher sample rate, the audio goes through fewer stages of
either decimation or upsampling, and that these stages are
typically not dithered. The fewer of these stages, the better
the audio sounds. So, folks, it's not the bandwidth that makes
the higher sample rates sound better, it's the internal design
of the converters themselves. <br>
</p>
<p> </p>
<p>Note that in one or more of his MQA papers Stuart describes the
processes' restoration of high frequency information "just in
case" but acknowledges that it may not be necessary. Like
chicken soup, keeping the extra high frequency information
couldn't hurt (except for wasting storage space and processing
time). But I am resentful that many of my great-sounding masters
that I have worked so carefully to make them sound better,
including upsampling before processing --- have to be
downsampled in order to be released on HD tracks because of the
high res mafia. <br>
</p>
<br>
<p>BK<br>
</p>
<br>
<div class="moz-cite-prefix">On 2/11/20 1:28 AM, James Johnston
via ProAudio wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAO2qRdPAazFEjNCLjXNfMiY38_2Hy+DF6tMo6z2PD4cJ6zBVjg@mail.gmail.com">
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<div dir="ltr">
<div>I must say that I am sorely tempted to see if I can get
somebody to whip up an app that measures effective bandwidth
of a PCM track at any common sampling rate.<br>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Mon, Feb 10, 2020 at
10:15 PM mark whitehouse via ProAudio <<a
href="mailto:proaudio@bach.pgm.com" moz-do-not-send="true">proaudio@bach.pgm.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex">
<div dir="ltr">Following on from our discussion of around 18
months ago,
<div>I know in Australia there were moves to get some kind
of "truth in marketing" </div>
<div>in regards to High sample rate recordings.</div>
<div>And promoting the differences in standards that are
now available. </div>
<div><br>
</div>
<div>Essentially a standard that could be understood by
and promoted to consumers, musicians etc.</div>
<div><br>
</div>
<div>It doesnt seem to have gone far and when you see
things like this</div>
<div><a
href="http://archimago.blogspot.com/2013/07/list-suspected-44-or-48khz-pcm.html"
target="_blank" moz-do-not-send="true">http://archimago.blogspot.com/2013/07/list-suspected-44-or-48khz-pcm.html</a> </div>
<div><br>
</div>
<div>It makes you wonder if things will improve.</div>
<div>Is there any consensus on getting this acknowledged? </div>
<div><br>
</div>
<div>regards</div>
<div>Mark <br clear="all">
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<div dir="ltr" class="gmail_attr">On Fri, Jun 15, 2018 at
8:25 AM Mark Waldrep <<a
href="mailto:mwaldrep@aixmediagroup.com"
target="_blank" moz-do-not-send="true">mwaldrep@aixmediagroup.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex">In the interest of
exploring the issue of perceptibility of high-resolution
vs. Redbook versions of the exact same file, I've posted
on my blog page today "The High-Resolution Audio
Challenge". I've prepared 6 of my AIX Records native 96
kHz/24-bit PCM masters as A and B versions. One is the
original high-res master and the other is a Redbook
downconversion. You are welcome to download the files
and play them. Please do not share them outside of the
group. I'm conducting a casual survey to see if people
can detect differences. I don't claim this will be
definitive. However, I've always complained that
previous tests failed because the source materials
weren't actual high-resolution files. This removes that
flaw. These are the real deal. The conversion was done
using triangular dither and noise shaping.<br>
<br>
You can find the article at <a
href="http://www.realhd-audio.com/?p=6197"
rel="noreferrer" target="_blank"
moz-do-not-send="true">http://www.realhd-audio.com/?p=6197</a><br>
<br>
Mark Waldrep, Ph.D.<br>
AIX Records and author of<br>
Music and Audio: A User Guide To Better Sound<br>
<br>
<br>
On 6/14/18, 3:13 PM, "Stephen Morley" <<a
href="mailto:proaudio-bounces@bach.pgm.com"
target="_blank" moz-do-not-send="true">proaudio-bounces@bach.pgm.com</a>
on behalf of <a
href="mailto:stephenmorley@iprimus.com.au"
target="_blank" moz-do-not-send="true">stephenmorley@iprimus.com.au</a>>
wrote:<br>
<br>
I can record DXD and the downsample in Pyramix, or
else at 192/24 and then downsample to 44/24<br>
-------- Original message --------From: James
Johnston <<a href="mailto:audioskeptic@gmail.com"
target="_blank" moz-do-not-send="true">audioskeptic@gmail.com</a>>
Date: 14/6/18 8:56 pm (GMT+10:00) To: <a
href="mailto:proaudio@bach.pgm.com" target="_blank"
moz-do-not-send="true">proaudio@bach.pgm.com</a>
Subject: Re: [ProAud] Wow. 384/32 LPCM! <br>
Just make sure the content is identical on both.<br>
<br>
On Thu, Jun 14, 2018 at 2:36 AM, Stephen Morley <<br>
<a href="mailto:stephenmorley@iprimus.com.au"
target="_blank" moz-do-not-send="true">stephenmorley@iprimus.com.au</a>>
wrote:<br>
<br>
> Hi James,I could send something, but not until
next week when I return<br>
> from leave.Stephen<br>
> -------- Original message --------From: James
Johnston <<br>
> <a href="mailto:audioskeptic@gmail.com"
target="_blank" moz-do-not-send="true">audioskeptic@gmail.com</a>>
Date: 13/6/18 8:53 am (GMT+10:00) To:<br>
> <a href="mailto:proaudio@bach.pgm.com"
target="_blank" moz-do-not-send="true">proaudio@bach.pgm.com</a>
Subject: Re: [ProAud] Wow. 384/32 LPCM!<br>
> I'll repeat my request:<br>
><br>
> Send me some recordings at 24/192 and a similar
one at 44/16. I can and<br>
> will measure the actual "information" present,
via SFM and bit depth.<br>
><br>
> _______________________________________________<br>
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<br>
-- <br>
James D. (jj) Johnston<br>
Independent Audio and Electroacoustics Consultant<br>
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-- <br>
<div dir="ltr" class="gmail_signature">
<div>James D. (jj) Johnston</div>
<div>Independent Audio and Electroacoustics Consultant</div>
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If you want good sound on your album, come to
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