[ProAudio] [ProAud] The High-Resolution Challenge

Tue Feb 11 10:56:07 EST 2020

I would concur that it’s the converters.

It’s really difficult to design a listening test that changes one variable however, so I’m not 100% sure.

We use an SRC in front of a DAC chip, and the DAC chip runs at 216KHz sampling rate.  With the same source material at 44.1 the sound changes as I change the DAC chip input frequency. This obviously includes the SRC filters, but it indicates to me that there is an approximation occurring in the maths somewhere.

I would welcome somebody who’s looked in more detail into this sharing their findings/ opinion with us :)

Cheers

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From: ProAudio [mailto:proaudio-bounces at bach.pgm.com] On Behalf Of Bob Katz via ProAudio
Sent: 11 February 2020 18:28
To: proaudio at bach.pgm.com
Subject: Re: [ProAudio] [ProAud] The High-Resolution Challenge

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:

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. 

In Bob Stuart's paper, which is open access so you do not have to be an AES member to download this:

  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: https://doi.org/10.17743/jaes.2019.0011

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.... 

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? 

Bob Stuart's explanation for this phenomenon is the design of the converters themselves. 

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. 

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. 

BK

On 2/11/20 1:28 AM, James Johnston via ProAudio wrote:

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.

On Mon, Feb 10, 2020 at 10:15 PM mark whitehouse via ProAudio <proaudio at bach.pgm.com <mailto:proaudio at bach.pgm.com> > wrote:

Following on from our discussion of around 18 months ago, 

I know in Australia there were moves to get some kind of "truth in marketing" 

in regards to High sample rate recordings.

And promoting the differences in standards that are now available. 

Essentially a standard that could be understood by and promoted to consumers, musicians etc.

It doesnt seem to have gone far and when you see things like this

http://archimago.blogspot.com/2013/07/list-suspected-44-or-48khz-pcm.html  

It makes you wonder if things will improve.

Is there any consensus on getting this acknowledged? 

regards

Mark  

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On Fri, Jun 15, 2018 at 8:25 AM Mark Waldrep <mwaldrep at aixmediagroup.com <mailto:mwaldrep at aixmediagroup.com> > wrote:

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.

You can find the article at http://www.realhd-audio.com/?p=6197

Mark Waldrep, Ph.D.
AIX Records and author of
Music and Audio: A User Guide To Better Sound

On 6/14/18, 3:13 PM, "Stephen Morley" <proaudio-bounces at bach.pgm.com <mailto:proaudio-bounces at bach.pgm.com>  on behalf of stephenmorley at iprimus.com.au <mailto:stephenmorley at iprimus.com.au> > wrote:

    I can record DXD and the downsample in Pyramix, or else at 192/24 and then downsample to 44/24
    -------- Original message --------From: James Johnston <audioskeptic at gmail.com <mailto:audioskeptic at gmail.com> > Date: 14/6/18  8:56 pm  (GMT+10:00) To: proaudio at bach.pgm.com <mailto:proaudio at bach.pgm.com>  Subject: Re: [ProAud] Wow. 384/32 LPCM! 
    Just make sure the content is identical on both.

    On Thu, Jun 14, 2018 at 2:36 AM, Stephen Morley <
    stephenmorley at iprimus.com.au <mailto:stephenmorley at iprimus.com.au> > wrote:

    > Hi James,I could send something, but not until next week when I return
    > from leave.Stephen
    > -------- Original message --------From: James Johnston <
    > audioskeptic at gmail.com <mailto:audioskeptic at gmail.com> > Date: 13/6/18  8:53 am  (GMT+10:00) To:
    > proaudio at bach.pgm.com <mailto:proaudio at bach.pgm.com>  Subject: Re: [ProAud] Wow. 384/32 LPCM!
    > I'll repeat my request:
    >
    > Send me some recordings at 24/192 and a similar one at 44/16. I can and
    > will measure the actual "information" present, via SFM and bit depth.
    >
    > _______________________________________________
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    > ProAudio at bach.pgm.com <mailto:ProAudio at bach.pgm.com> 
    > http://bach.pgm.com/mailman/listinfo/proaudio
    >

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    James D. (jj) Johnston
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