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Is 22kHz brickwall cut off audible? - Listening test

Can you hear a difference between the files

  • I can hear a difference and I have an ABX result

    Votes: 2 11.8%
  • I can hear a difference but have no ABX result

    Votes: 3 17.6%
  • I cannot hear a difference and I have an ABX result

    Votes: 3 17.6%
  • I cannot hear a difference but have no ABX result

    Votes: 9 52.9%

  • Total voters
    17

krabapple

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If we cease usual secondary school simplifications, my humble question is - what do we really know about hearing?

Rather a lot. Do you know how much?

To those who might be interested, there is a link that summarizes results of hundreds of participants of similar tests


Ack, not Reiss again. The summarized results are actually a selection of results.
 

bennetng

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I cannot hear > 22k frequencies so I often lower the cutoff threshold to the point that I can really hear the difference when doing these kinds of tests, the threshold may not be a fixed one and could differ from file to file.

However, when I can hear the difference, very steep filters like what Chord usually use often sound worse than a gentler one with some Nyquist violation (i.e. imaging), but my preference is not as extreme as those MQA filters or NOS. Which means, something similar to what "typical" DAC filters do - some imaging up to no more than 0.6fs or so.
 

msmucr

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Pavel, that's certainly interesting experiment.
I'd love to participate with listening comparisons, but unfortunately I can't do it due to my current medical condition (middle ear issues after Covid).

I recall we've done similar tests cca 10 years ago. Although it was mainly regarding filtering and ways of downsampling (44.1 from 96k original recordings) among friends, sound engineers. Main objective was to make some sense among various options for the task. You had very expensive standalone software resamplers, opensource tools, built-in algorithms in DAWs, some people praise only analog re-capture at target rate etc. One SRC algorithm might have super steep filter curve, other have leaky slower filter, another one is fully adjustable with regards to steepness, cutoff frequency and phase response.
I contributed with some tools and processed samples. Results with comparisons of source and resampled files were pretty much non-conclusive and pretty random. Along the way we found some initially unexpected things. For example one pro audio converter had pretty different (measurably) performance with 48k and 44.1k based frequencies, when externally clocked, so difference to original files was exaggerated because 96k was 48k multiple. Also some DACs like Benchmark DAC-1 has in-line ASRC, which resamples everything to single frequency for the actual conversion, so it was felt that aspect somewhat diminish differences among different source downsampling algorithms and its settings. Possible perceived differences were very material and playback system dependent and of course we've made lot of mistakes in testing procedures along the way :) Finally, as expected, we started to argue - ranging from whether it's different, and if it really worthwhile difference in the grand scheme of things, if it really matters to match the source or if it's better to simply follow your gut feeling and pick what you feel is best sounding at the target rate. But it was lot of fun and for me definitely starting point for further technical exploration :)
I then I naturally came to conclusion, next test needs to be better prepared with ABX tools, ideally aiming to individual aspects in isolation (like your test with sources at common rate and just different filters for example, just to avoid some mentioned issues with playback systems and DACs). But in reality I haven't repeated it properly with multiple listeners, participants, as I was busy with other work.

Anyway one interesting aspect among quite a few people including myself was, those super steep filters with shortest transition band weren't preferred as generally best option for everything. Also later I found few of most used and praised external resamplers like Weiss Saracon, Izotope RX or Hepta SRC in Pyramix DAW has either hardcoded parameters or defaults to somewhat "leaky", more relaxed responses akin to common half-band filters in converters. Sure in those chips you have computational and latency constraints, so such design choice is often also necessary. However despite offline SRC can have really super steep filters with pretty much as many taps you need, those vendors opted for those slower filters with less ringing, be it with some deliberate aliasing especially in case of 1fs rates. I don't think, it was just technical decision of their programmers, as there is usually also beta testing with various sound engineers.

So if you're interested, maybe you could do another round of ABXing with gentler LPF.

Michal
 

msmucr

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Just writing about that in previous post, I've rather qucikly prepared one such gentler filter for you. It's great, nowadays we have so many great tools, which makes all those experiments easier.
Kudos to Mr. Pkane for DeltaWave. Thomas Drugeon for rePhase, where I did this FIR filter and finally applied it with ffmpeg.

Your sample filtered with another LPF
Plot from DeltaWave
Impulse - IR_LPF_192db_96k.wav
rePhase project - IR_LPF_192db_96k.rephase

If you would like to use ffmpeg for convolution with another filters, then you can start from following command.

Code:
ffmpeg -i your_input.wav -i your_impulse.wav -filter_complex "[1:a]pan=stereo|c0=c0|c1=c0[IR]; [0:a]apad=pad_len=8192[input]; [input][IR]afir=gtype=none,atrim=start_pts=8192[out]" -map "[out]" -c:a pcm_s24le output.wav

There is some slight trickery with padding and trimming. Because filter length is 16384 samples and IR is symmetrical for linear-phase response. So to compensate for this 8192 samples latency and avoid misalignment and truncation, it is necessary to append 8192 samples to the input file and discard 8192 samples at start of the output.
For possible experiments with different filter length, you'll just modify those two numbers in command accordingly.

Finally, you've had some complaints about LP recording :). I've searched for Break on Through from SACD. I recalled, it was one of my first hybrid SACDs I've ripped at Playstation 3 (it was great a challenge for me back then to cross-compile the ripping software for it ;)).

So there is first minute of DSDIFF from the song, if you would like to test also decimation to PCM (for example in SoX with DSD patch from Mansr) and possibly filtering, whether you'll find same differences like on LP recording sample.
The Doors - Break On Through SACD sample.dff
Decimated 88.2k PCM version.
The Doors - Break On Through SACD sample.flac

Have fun :)

Michal
 

restorer-john

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The two tracks sounded so dull (HF challenged) and awful, I could not see the point. A bit like disconnecting the tweeters and asking which speaker sounds better.
 

restorer-john

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I remember when I was recording cassettes in the 1980s and 'testing' my ears by switching in and out the defeatable MPX filters on my decks, listening for the difference. It was plainly obvious on particular CD or vinyl derived content with young ears and plenty of HF information. I probably wouldn't try it now, I might get a rude shock.

MPX filters were designed to kill the 19kHz FM 'pilot' tone to prevent Dolby B/C going beserk on decks that would otherwise happily record up to and beyone 20kHz.
 

tmtomh

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I remember when I was recording cassettes in the 1980s and 'testing' my ears by switching in and out the defeatable MPX filters on my decks, listening for the difference. It was plainly obvious on particular CD or vinyl derived content with young ears and plenty of HF information. I probably wouldn't try it now, I might get a rude shock.

MPX filters were designed to kill the 19kHz FM 'pilot' tone to prevent Dolby B/C going beserk on decks that would otherwise happily record up to and beyone 20kHz.

An MPX filter is an analogue notch filter, yes? I wonder what the Q is on that.
 

restorer-john

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An MPX filter is an analogue notch filter, yes? I wonder what the Q is on that.

Back when I could hear it, I certainly had no equipment to measure/sweep the filters. I think they are pretty much a straight LPF, not a notch in reality. Just kill everything above maybe 16-17kHz I'd say.

Edit: Typical switchable MPX filter for a Sony ES cassette deck heading into the Dolby IC.

1657586709756.png
 

levimax

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I don't have time to do full test now but I do have them loaded in Foobar ABX and on practice I am able to identify. I don't understand why these files are so distorted.... it isn't subtle it sounds like 10% to 20% or more distortion.... were the tracks like this to start or did the filtering ruin them? For those not trying ABX because of "high frequency hearing loss" please don't let that stop you as I don't think it matters.
Well I did an ABX test on speakers and results are below. I really struggled to hear any difference and the distortion was bugging me so I just wanted to finish but I got 8 out of 10 wrong!? I have no idea what this means and I don't think 10 trials are enough as when practicing I got up to 7 out of 9 right before I went back to random. I would be willing to do it again but hopefully on some "cleaner" tracks. I did go listen to my digital versions of this song (I actually have 4 different versions) and while distorted they all are much cleaner than this needle drop and I could ABX them 100% without any problems. Not sure much has been learned yet from this.


foo_abx 2.0.6d report
foobar2000 v1.6.7
2022-07-11 18:13:55

File A: BreakOn_1sh.wav
SHA1: 73fd67f8fb135adf5e0ec92bc9f2178bbe269709
Gain adjustment: +2.69 dB
File B: BreakOn_3sh.wav
SHA1: 80aebddcdc26e9851a9441ddc89aae59a2d099eb
Gain adjustment: +2.69 dB

Used DSPs:
Resampler-V, Convolver

Output:
DSD : Default : Speakers (4- XMOS XS1-U8 MFA (ST)) [exclusive], 24-bit
Crossfading: NO

18:13:55 : Test started.
18:17:31 : 01/01
18:17:47 : 01/02
18:18:06 : 01/03
18:18:25 : 01/04
18:18:48 : 01/05
18:19:30 : 01/06
18:19:41 : 01/07
18:19:58 : 01/08
18:20:09 : 02/09
18:20:27 : 02/10
18:20:27 : Test finished.

----------
Total: 2/10
p-value: 0.9893 (98.93%)

-- signature --
538ecf9f1aa005155fff6bd473f99786584012aa
 

fieldcar

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Sure it has! You are virtually 100% sure you cannot hear any difference! The best result so far. Hehe.
I couldn't even slightly hear a difference instantly solo auditioning between the tracks in FL studio while it played through the samples. I can't imagine ABX'ing this in a normal abx test and hoping to get any sort of result of any kind.

I'm still leaning toward something in the playback chain distorting in an unusual way other than the vinyl rip that has extremely high saturation up into 10-20%thd. It could even be the koss headphones doing something.

@pma , I would check your stylus&cartridge. It may be misaligned/defective or your anti-skate or down force could be out of wack. I usually only get this sort of distortion when my turntable has something out of sorts.
EDIT: Just gave another listen to your samples VS The Qobuz mono mastered track side-by-side (Yours is definitely not a modern stereo master that I compared it to earlier), and it seems like there is still a bit of distortion on his voice and drums during the louder parts of the vinyl rip. (I'm obviously ignoring the intentional guitar distortion.)
EDIT2: And maybe this is just vinyl's limitation. Modulation distortion during loud parts means breakup of the fine details during loud parts of the album. Heck. I dunno at this point.
 
Last edited:

krabapple

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Well I did an ABX test on speakers and results are below. I really struggled to hear any difference and the distortion was bugging me so I just wanted to finish but I got 8 out of 10 wrong!? I have no idea what this means and I don't think 10 trials are enough as when practicing I got up to 7 out of 9 right before I went back to random. I would be willing to do it again but hopefully on some "cleaner" tracks. I did go listen to my digital versions of this song (I actually have 4 different versions) and while distorted they all are much cleaner than this needle drop and I could ABX them 100% without any problems. Not sure much has been learned yet from this.


foo_abx 2.0.6d report
foobar2000 v1.6.7
2022-07-11 18:13:55

File A: BreakOn_1sh.wav
SHA1: 73fd67f8fb135adf5e0ec92bc9f2178bbe269709
Gain adjustment: +2.69 dB
File B: BreakOn_3sh.wav
SHA1: 80aebddcdc26e9851a9441ddc89aae59a2d099eb
Gain adjustment: +2.69 dB

Used DSPs:
Resampler-V, Convolver

Output:
DSD : Default : Speakers (4- XMOS XS1-U8 MFA (ST)) [exclusive], 24-bit
Crossfading: NO

18:13:55 : Test started.
18:17:31 : 01/01
18:17:47 : 01/02
18:18:06 : 01/03
18:18:25 : 01/04
18:18:48 : 01/05
18:19:30 : 01/06
18:19:41 : 01/07
18:19:58 : 01/08
18:20:09 : 02/09
18:20:27 : 02/10
18:20:27 : Test finished.

----------
Total: 2/10
p-value: 0.9893 (98.93%)

-- signature --
538ecf9f1aa005155fff6bd473f99786584012aa

If you have to try that hard to confirm an audible difference, why should it matter?
 

levimax

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If you have to try that hard to confirm an audible difference, why should it matter?
The OP apparently can ABX something that appears from the supplied measurements to be below what one would expect to be able to hear. So far he is one out of 11. Since he went to the trouble to post files I thought it would be interesting to add a data point to the discussion.
 

krabapple

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The OP apparently can ABX something that appears from the supplied measurements to be below what one would expect to be able to hear. So far he is one out of 11. Since he went to the trouble to post files I thought it would be interesting to add a data point to the discussion.

I'm referring to this

I don't think 10 trials are enough as when practicing I got up to 7 out of 9 right before I went back to random

But hey, knock yourself out if that's what you need to do.
 

daftcombo

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I listened to both tracks but can't hear a difference. Quality is not that bad for an old vinyl rip, but you should adjust gain so that the tracks peak around -1dBFS.
 
OP
pma

pma

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10 trials is not enough but 2x10 trials is. It is an equivalent to 20 trials within some time interval. You may sum it and get 8/10 + 8/10 = 16/20. This is perfectly valid in probability theory and one may find what 16/20 means. BTW, I have a further 8/10 result. Why not 16 or 20 trials in one row? Because it is difficult to keep concentration, the differences are really very small. The only important outcome to me is, that 20kHz cut may lead to a tiny sound difference and such difference is not distinguishable to everyone. Our hearing differs, and this is the fact. Humans are not measuring instruments, try to digest it. I have already posted the link that summarizes studies on about 400 subjects that confirms that some subjects are sensitive to >20kHz components that are accompanying the audio sound signals. There are no trivial answers to trivial questions, though some would like to have them. This is a bit funny, to me. Get humble, if you can.
 

dc655321

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10 trials is not enough but 2x10 trials is. It is an equivalent to 20 trials within some time interval. You may sum it and get 8/10 + 8/10 = 16/20. This is perfectly valid in probability theory and one may find what 16/20 means. BTW, I have a further 8/10 result. Why not 16 or 20 trials in one row? Because it is difficult to keep concentration, the differences are really very small. The only important outcome to me is, that 20kHz cut may lead to a tiny sound difference and such difference is not distinguishable to everyone. Our hearing differs, and this is the fact. Humans are not measuring instruments, try to digest it. I have already posted the link that summarizes studies on about 400 subjects that confirms that some subjects are sensitive to >20kHz components that are accompanying the audio sound signals. There are no trivial answers to trivial questions, though some would like to have them. This is a bit funny, to me. Get humble, if you can.

I asked previously and may have missed it, but did you post a detailed methodology for your experiment?
Filter derivation, coefficients, application, post-processing, etc
 
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