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Topic: The Emperor's New Sample Rate (Read 63695 times) previous topic - next topic
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The Emperor's New Sample Rate

Reply #75 –
A word of caution here: the test is being passed with an on-board sound card (and a Creative sound card) and unknown other equipment.

If you let me pick the sample and sound card, I too will pass the test, and I can't hear a thing above 17kHz!


It would be interesting to know what is reaching MLXXX's ears. He implies that A, B, C and D are so bad there's no need to ABX, and then ABXs "E" with ease.

Whereas I am not aware of a single published test where someone has ABXed a 24kHz low pass filter, unless something in the signal chain has been effectively broken.

Either we are conversing with the bionic man, or his audio equipment is faulty.

How to tell? It would be a start to try recording the analogue output of the soundcard using a decent sound card - and see what differences are present between the 48kHz, 96kHz, and 192kHz versions.

Cheers,
David.

The Emperor's New Sample Rate

Reply #76 –
Quote from: 2Bdecided on
A word of caution here: the test is being passed with an on-board sound card (and a Creative sound card) and unknown other equipment.

If you let me pick the sample and sound card, I too will pass the test, and I can't hear a thing above 17kHz!


It would be interesting to know what is reaching MLXXX's ears. He implies that A, B, C and D are so bad there's no need to ABX, and then ABXs "E" with ease.

Whereas I am not aware of a single published test where someone has ABXed a 24kHz low pass filter, unless something in the signal chain has been effectively broken.

Either we are conversing with the bionic man, or his audio equipment is faulty.

How to tell? It would be a start to try recording the analogue output of the soundcard using a decent sound card - and see what differences are present between the 48kHz, 96kHz, and 192kHz versions.

Cheers,
David.


David, I can understand your concerns. However I have now edited my post above (#72) to include links to the three files.

Anyone can download these and do their own ABX tests.  Note that as the files are all at the same sample rate, 44.1KHz, whatever sound card (or AVR etc) is used to compare the three files, will invoke the same filter for each.

Regarding my own hearing of high frequencies, it is not exceptional, one reason being that I am middle-aged.

___________________________________

Edit: I have now tested my hearing using headphones connected to the output of an Audigy 4 hub (an external sound module manufactured by Creative that interfaces with a PC).  In the ABX test below, the purpose of the 40KHz sample is to serve as a reference.  I found that at a normal gain setting for listening to music, I could perceive the test frequency up to about 19KHz fairly easily, albeit that that high frequency was very faint.

At 20KHz there was no perceivable tone (just some low level sound card noise).  When I tried 19.5KHz, I could just hear the tone again.

I recorded the test tones with Audition 3.0 at a 96KHz sample rate, and at a level of -20dB.  Here is the ABX result:

foo_abx 1.3.1 report
foobar2000 v0.9.5.1
2008/05/07 00:03:29

File A: C:\Documents and Settings\All Users\Documents\sine19500@96KHz.wav
File B: C:\Documents and Settings\All Users\Documents\sine40000@96KHz.wav

00:03:29 : Test started.
00:03:47 : 01/01  50.0%
00:04:05 : 02/02  25.0%
00:04:14 : 03/03  12.5%
00:04:32 : 04/04  6.3%
00:04:48 : 05/05  3.1%
00:04:51 : Test finished.

----------
Total: 5/5 (3.1%)

The Emperor's New Sample Rate

Reply #77 –
MLXXX,

Thanks, but that's not quite what I meant. I know pretty much what Cool Edit / Audition will do to the file.

However, I don't know what your OS, drivers, sound card, amplifier, and speakers or headphones do to the signal.


If this was a real experiment, I would put a high quality probe microphone in your ear, and compare what it picked up from the 96kHz version vs the other two. I would also re-digitise the output from the amplifier, and the output from the sound card. I would analyse all three sets of recordings to see if/where distortion crept in.

From your description, it seems very likely that the audible difference is well within the audible(!) range when it reaches your ears, and as such is being generated within your equipment.


Since you have two sound cards, would you be willing to try a further experiment?

Get one to record (maybe the creative one - will it record at 96kHz without conversion anywhere?) and the other to playback. If possible, play back the 96k, 192k, 48k, and 44.1 on the other PC, recording them all at 96k. Then post the result. Also, listen to the result and report what difference, if any, you hear between the various re-recordings.

Cheers,
David.

The Emperor's New Sample Rate

Reply #78 –
Quote from: 2Bdecided on
Since you have two sound cards, would you be willing to try a further experiment?

Get one to record (maybe the creative one - will it record at 96kHz without conversion anywhere?) and the other to playback. If possible, play back the 96k, 192k, 48k, and 44.1 on the other PC, recording them all at 96k. Then post the result. Also, listen to the result and report what difference, if any, you hear between the various re-recordings.

That strikes me as quite a project to undertake, with many variables.

At this point in time I would be pleased to hear reports from anyone inclined to ABX compare the three examples of conversion to 44.1KHz I have uploaded.  If others can hear differences too, then we appear to be on a sticky wicket as far as undertaking more advanced tests is concerned, such as comparing a 48KHz file with a 44.1KHz file.

The Emperor's New Sample Rate

Reply #79 –
Quote from: MLXXX on
At this point in time I would be pleased to hear reports from anyone inclined to ABX compare the three examples of conversion to 44.1KHz I have uploaded.  If others can hear differences too, then we appear to be on a sticky wicket as far as undertaking more advanced tests is concerned, such as comparing a 48KHz file with a 44.1KHz file.

Ok, I had a look and a listen at the files you presented. First things first, I failed to ABX any of the samples - but that doesn't prove anything - the speakers I used don't work too well above 17kHz anyways*. Second, your resampled versions are 32bit float, and the original is 24bit - are you sure your ABX software or soundcard are not treating these two depths differently? The third point is that there is obvious ringing present on the left channels Audition and Cooledit versions (in a plot of the data) - mostly just before the sound starts - this is not present on the R8brain or original samples. Level matching doesn't seem to be an issue here - the Audition, R8brain and original versions all match very closely on A weighted level. Anybody with ABX success?

* ABX Setup: AV-710 rear DAC into a LM3886 based chipamp driving full range speakers with Fostex FE167E drivers.

The Emperor's New Sample Rate

Reply #80 –
Thanks for your observations, cabbagerat.

Quote from: cabbagerat on
your resampled versions are 32bit float, and the original is 24bit - are you sure your ABX software or soundcard are not treating these two depths differently?

I could not find an option in Cooledit or Audition to create a 24-bit result when converting the sample rate.  I therefore selected  32-bit float.  At the last moment when uploading the files, I realised someone might pick me up on the fact that the r8brain conversion was at 24-bit and not at 32-bit float.  So I quickly redid the r8brain conversion so it would match the format of the other two conversions.

I cannot hear any difference between the r8brain 24-bit version and the r8brain 32-bit float version.  However, as you say, for some reason during playback my setup might have reacted differently to a 32-bit float version, compared with a 24-bit integer version, so it was as well to check this out.

As matters stand, all conversions are in the same bit format (32-bit float) and at the same bit rate (44.1KHz).

I did find the Audition and Cooledit versions extremely difficult to tell apart with headphones (and the Audigy 4 hub).  I found them slightly easier to tell apart on my main hi-fi setup and listening to my AVR doing the digital to analogue conversion, and sending the audio to speakers.

As the files contain so much high frequency energy they are quite fatiguing on the ears, I find.  It is partly for this reason I stopped at 5 tests in the ABX testing, as the doing of the testing was affecting my ears.  Even some hours after the ABXing last night, my ears were still ringing slightly and feeling a little uncomfortable.  I used normal listening gain, but the samples do contain an extraordinary amount of high frequency energy.

Quote from: cabbagerat on
there is obvious ringing present on the left channels Audition and Cooledit versions (in a plot of the data) - mostly just before the sound starts - this is not present on the R8brain or original samples. Level matching doesn't seem to be an issue here - the Audition, R8brain and original versions all match very closely on A weighted level.

The ringing in the left channel might explain why the stereo image appeared to me to shift to the left in the Audition and Cooledit conversions, compared with the original 96KHz file.

______

1. Anyone else with comments on the three conversions?

2. Is there some other downsampling method that might leave more of the original sound intact?

3. If the need arises to upload a file illustrating another conversion, there is now an upload thread titled Resampling down to 44.1KHz, Is there a method that will not colour the sound? here.

EDIT:
On further reflection I wonder whether I have reached the point of hijacking this thread towards an overspecialised technical discussion, better suited to a standalone topic.  I'd be happy if people interested in discussing the three files further do so in the related upload thread, namely:  Resampling down to 44.1KHz, Is there a method that will not colour the sound?

The Emperor's New Sample Rate

Reply #81 –
MLXXX,

I'll say it one last time. It is almost certain that your playback is colouring the sound, while the resampling is fine.

Which is why, unless you can try the experiment of re-recording the output of one sound card with another, this is a waste of time.

Or, to put it more simply, there's no point carrying out an experiment without first checking that the equipment is as described, and working correctly.

Cheers,
David.

The Emperor's New Sample Rate

Reply #82 –
Quote from: AndyH-ha on
If there is one person way outside the range encompassing everyone else, even if that one person’s score is completely valid, we have to ask if the fact has any relevance. Suppose one person in a million can really detect a difference, but the other 999999 can not? If you happen to be interested in the abnormal, then you may want to located these (relatively) few individuals so you can subject them to laboratory degradation, but if you are interested in just about any other aspect of audio, you probably could not care less about them.; they just are not relevant.
Personally. I'm not interested in either the abnormal nor the extremes, I'm simply interested in what I may be able to appreciate. Perhaps others cannot, but I'm not interested in a numbers game, nor in statistics per se. What interests me is in seeing how far I can go in improving the quality and usability of my music collection. 

Please, don't get me wrong, scepticism is a good thing, however, clearly closed-mindedness is not. I, myself, am simply trying to be open-minded, honest and inquisitive, nothing more. I'm not sure but it seems to me as if some people seem to think that if it isn't audible to a certain percentage of ABX testers, it doesn't exist. In truth, ABX testing is wholly acceptable as a method of scientific enquiry, however, how the results are being interpreted is still somewhat subjective, in my humble opinion. Now, I believe this is necessary since each subject must define what is a personally useful level of audio high fidelity. I'm simply seeking what works for me. If you have found what works for you, great! Perhaps through sharing and discussion, it will be possible for people in general to find greater joy in their own personal music experience, no?
Quis custodiet ipsos custodes?  ;~)

The Emperor's New Sample Rate

Reply #83 –
Quote from: 2Bdecided on
It is almost certain that your playback is colouring the sound, while the resampling is fine.

Normally, if three files of identical bit depth and sample rate are played back using the ABX feature in foobar but they sound different, and the ABX statistics confirm the probability of the result, then we conclude they are in fact different, by the standards of this Forum.

You have asked me to perform and upload recordings.  That seems to me to be an unusual approach which could raise as many questions as it might answer. I would prefer it if you yourself or perhaps some very young members or others with very good high frequency hearing were to ABX the three files I have provided.  This would be simple and direct.

It may be worth stressing that at this point I am not comparing the 96/24 clip with the conversions.  I am merely comparing the three conversions with each other.  Those three conversions are all at 44.1KHz 32-bit floating.

Edit: I might add that there were three different DAC devices available to me and all three gave perceptible differences as between the three files, namely: the DAC in my Audio Video Receiver, the high-definition DAC in the motherboard of my Home Theatre PC, and the DAC in the Audigy 4 external sound module.

The Emperor's New Sample Rate

Reply #84 –
Accepting the results of group ABX testing makes perfect sense to me if the individual is prepared to accept that they are highly likely to be "normal". Performing individual ABX testing allows you to draw your own conclusions regarding your own individual hearing abilities. So what's the problem?

Cheers, Slipstreem. 

The Emperor's New Sample Rate

Reply #85 –
Quote from: 2tec on
Hopefully, I'm not just beating on a dead horse here, however, I do have several more questions, please? First off, could conducting audio tests at higher than normal listening levels, reveal subtle differences being missed by current ABX testing? Secondly, doesn't the one 20/20 score in the Detmold study, merit further investigation into exceptional cases of hearing ability? Third, is there any possibility that the test equipment was simply unable to reproduce the difference?

Furthermore, I feel I must apologize in advance if these questions seem too repetitive or rudimentary for some people here.

2tec, my understanding is that a good standard of equipment was used, with a large sample of people listening at normal listening levels, to a variety of music.  If there had been a clear difference for even a small percentage of participants, that would have emerged from the testing.

What I do not know is what parts of the music were put through a 44.1Khz sampling rate bottleneck.  Unless the change occured right in the middle of particularly sparkling passages full of high frequency energy, it does not surprise me that the temporary bottleneck went unnoticed.

From my own investigations and reading, I believe that a reduction in sample depth to 16 bits is only identifiable if dithering is poor or the listening level is unrealistically high.  That to my mind leaves sample rate as the significant factor in a 'bottleneck'.

Perhaps people more familiar with the report can comment on what passages in the music got the 'special treatment'.  Was it by any chance a series of clashes of cymbals with a quick change to 44.1Khz in the middle of that series?  Unless such 'killer' episodes were included, identification of the bottlenecks could be expected to have been quite difficult, all other things being equal.

The Emperor's New Sample Rate

Reply #86 –
Quote from: 2tec on
Quote from: krabapple on

No, I think you're just not reading carefully, or not understanding the concepts involved.  I don't see anyone else here claiming to be confused by the two statements.

Sure, go ahead, think whatever you like. I see that you sure like telling us what that is! As for anyone else, why would they want to get involved in your argument?


If I really had written or even implied something as foolish as  'nobody can hear better than anyone else', as you claimed I did, it's a fair bet that someone here besides you would have taken me to task.

As for your pose as the blameless victim of mean old krabapple,  it might play better if you revised all your posts on this thread.  Doubt it, though.  Btw,  that part where you thank Pio for reminding you of what *I* wrote: priceless. 


Quote
I'll say it one last time. It is almost certain that your playback is colouring the sound, while the resampling is fine.


To nail this part down, perhaps someone can supply a couple of files of demonstrably first-rate resampling (every step approved by HA noggins),  and MLXX can see if he can ABX them?  Then at least we can narrow it down to either,  his extraordinary hearing, or his sub-extraordinary playback chain.

In the meantime, maybe I can get Arny Kruger to take a look at the thread, and offer some opinions.

The Emperor's New Sample Rate

Reply #87 –
Quote from: krabapple on
To nail this part down, perhaps someone can supply a couple of files of demonstrably first-rate resampling (every step approved by HA noggins),  and MLXX can see if he can ABX them?  Then at least we can narrow it down to either,  his extraordinary hearing, or his sub-extraordinary playback chain.

In the meantime, maybe I can get Arny Kruger to take a look at the thread, and offer some opinions.

Sounds good.

The Emperor's New Sample Rate

Reply #88 –
Quote from: MLXXX on
Quote from: 2Bdecided on

It is almost certain that your playback is colouring the sound, while the resampling is fine.
Normally, if three files of identical bit depth and sample rate are played back using the ABX feature in foobar but they sound different, and the ABX statistics confirm the probability of the result, then we conclude they are in fact different, by the standards of this Forum.
Sorry MLXXX, I wasn't referring to the ABX test itself - your other thread is subtitled "Is there a method that will not colour the sound?" - that implies a comparison with the original, which to you yields a clearly audible difference. It's that comparison that I was questioning, and that comparison that I suspect your playback is colouring.

Quote
You have asked me to perform and upload recordings.  That seems to me to be an unusual approach which could raise as many questions as it might answer. I would prefer it if you yourself or perhaps some very young members or others with very good high frequency hearing were to ABX the three files I have provided.  This would be simple and direct.
You should do a search for the discussion of "udial"...
http://www.hydrogenaudio.org/forums/index.php?showtopic=9772
... to see why this is can be anything but simple for samples that contain lots of HF/ultrasonic energy.

It's still an interesting test, but you have to be in possession of all the facts, and wary of the potential pitfalls.

Cheers,
David.

The Emperor's New Sample Rate

Reply #89 –
Quote from: 2Bdecided on
... to see why this is can be anything but simple for samples that contain lots of HF/ultrasonic energy.

Yes there are pitfalls everywhere with 44.1Khz.  These include:
[blockquote](i)  the digital filtering for the analogue to digital process required to capture an analogue source at a 44.1Khz sample rate
(ii) the digital filtering in the digital to analogue conversion required to play a recording made with a sample rate of 44.1KHz
(iii) [allied to (ii)], the fact that there may be an intermediate resampling to a card's 'native sample rate'.[/blockquote]
Digital filtering must find a compromise solution that:
* provides adequate protection against aliases
* maintains the frequency response up to a zone not far below the Nyquist frequency
* avoids excessive phase changes, or ringing, or other distortion.

The Emperor's New Sample Rate

Reply #90 –
Quote from: MLXXX on
Yes there are pitfalls everywhere with 44.1Khz.  These include:
[blockquote](i)  the digital filtering for the analogue to digital process required to capture an analogue source at a 44.1Khz sample rate
(ii) the digital filtering in the digital to analogue conversion required to play a recording made with a sample rate of 44.1KHz
(iii) [allied to (ii)], the fact that there may be an intermediate resampling to a card's 'native sample rate'.[/blockquote]
Why would (i) or (ii) be any different for 44.1kHz than 48kHz or any other sampling frequency for that matter?
lossyWAV -q X -a 4 -s h -A --feedback 2 --limit 15848 --scale 0.5 | FLAC -5 -e -p -b 512 -P=4096 -S- (having set foobar to output 24-bit PCM; scaling by 0.5 gives the ANS headroom to work)

The Emperor's New Sample Rate

Reply #91 –
Because there is very little headroom.  The 44.1KHz Nyquist limit, 22050Hz, is awkwardly close to the upper limit of human hearing, of around 20000Hz.

The Emperor's New Sample Rate

Reply #92 –
Quote from: 2Bdecided on
Quote
You have asked me to perform and upload recordings.  That seems to me to be an unusual approach which could raise as many questions as it might answer. I would prefer it if you yourself or perhaps some very young members or others with very good high frequency hearing were to ABX the three files I have provided.  This would be simple and direct.
You should do a search for the discussion of "udial"...
http://www.hydrogenaudio.org/forums/index.php?showtopic=9772
... to see why this is can be anything but simple for samples that contain lots of HF/ultrasonic energy.

It's still an interesting test, but you have to be in possession of all the facts, and wary of the potential pitfalls.

Cheers,
David.


I forgot about udial -- couldn't MLXX use that to see if his setup is resampling during playback?

Here's  udial.flac

The Emperor's New Sample Rate

Reply #93 –
Quote from: MLXXX on
Because there is very little headroom.  The 44.1KHz Nyquist limit, 22050Hz, is awkwardly close to the upper limit of human hearing, of around 20000Hz.

Yes, and the ATH curve drops off severely as it approaches 20 kHz. If only our ears respond like studio monitors...

The Emperor's New Sample Rate

Reply #94 –
Quote from: sld on
Yes, and the ATH curve drops off severely as it approaches 20 kHz. If only our ears respond like studio monitors...
It's remarkable that most recording- and mastering engineers who use (and claim audible advantages of) 96 kHz and higher sample rates, are still using rather standard bandwidth microphones and monitors (upper limit slightly over 20 kHz). Audible benefits of hi-res audio (QED) should therefore probably be searched for in the audible band (<22kHz).
Please note: 96 and 44.1 kHz versions can sound different, but this doesn't necessarily mean that the 96 version sounds better. It could be worse ! David Griesinger pointed out in this paper that high-frequency content can cause InterModulation distortion in the playback chain. It is possible that a LowPassed version reduces or eliminates this effect and therefore sounds better on some playback systems.

The Emperor's New Sample Rate

Reply #95 –
I have created an account only to post this. While a lot of helpful information is available on the forums, access to it didn't need my input. However, some people here tend to be strongly biased.

I have pretty good hearing, probably also due to my age (17). I can hear sounds as high as 23 kHz and even 24 kHz, however extremely faint. I am certain of this values, they are quoted from a medical examination and not some cheap speakers. In addition, I have Asperger's, and one direct effect is the ability to abstractize and categorize sensory input, including sound. I can clearly distinguish every instrument type in a symphony, for example.

I also do not care about the rest when it comes to sound quality. Whether they distinguish a 128 kbit/s MP3 from a SACD is not important to me, what matters is that I do. I had the occasion to compare classical music in ABX between a SACD and a version resampled to CD-A quality and I could with fairly high precision identify them. Another facet of Asperger's is that I have synestezia and I perceive some sound combinations as emotions and tastes rather than abstract vibrations. With a high degree of subjectiveness, I have found the SACD to convey a sense of serenity and trippy tranquil that the CD-A did not, to the same degree. I assume (but am not sure this is correct) that the higher harmonics were the cause for this.

I own a fairly cheap stereo at home and while it does sound clear, it peaks at about 17 kHz. When input is 20 kHz, output sounds more or less like 5-10 kHz for example. So I cannot enjoy music fully and do not afford anything more than, say, a thousand euro, which I assume is well under the price tags for pro audio.

Well so, this was my opinion on the subject. I only wished to change some biased opinions in that there are some people who can tell the difference and who most likely are among those who say CD-A is not enough. I also need to improve my non-native English, as some phrases do sound awkward...

Anyway, thank you for reading!
gantrithor

PS: By the way, I happen to have received a present consisting in a "Millenium Masterpieces" collection a few years ago. The box has the inscription "20 bit recording (DDD)" on it multiple times. I knew CD's are usually 16bit/44.1kHz/Stereo but thought 20bit is also possible (perhaps more throughput, less duration). I suspect the recording was made at 20bit, then downsampled to 16bit on CD mastering. But why do such a thing? Any ideas? Thank you again!

The Emperor's New Sample Rate

Reply #96 –
20 bits is useful for noise shaping. You can get >96db of dynamic range at frequencies the ear is most sensitive to (midrange/treble), in exchange for <96db at less sensitive frequencies (very high treble).

Have you considered scoring some headphones? You can certainly get good ultrasonic response if you know where to look. For 1000 euros you can get pretty much top-shelf headphones with change to spare for a good amp. The Sennheiser HD650s are "only" 450 euros, and they arguably have a good response out to 30khz. Hell, you could buy a used Stax electrostatic rig at that price.

The Emperor's New Sample Rate

Reply #97 –
It's not uncommon for teenagers to be able to hear up to 24kHz at high levels. Some individuals have surprisingly low thresholds. I've attached a graph of some averaged results.

See this paper for the actual results:

Henry, K. R.; and Fast, G. A. (1984).
Ultrahigh-Frequency Auditory Thresholds in Young Adults: Reliable Responses up to 24 kHz with a Quasi-Free-Field Technique.
Audiology, vol. 23, pp. 477-489.

The response at those high frequencies drops off due to noise exposure before the normal audiometry range (typically only measured up to 8kHz) shows any change.

There's some fascinating research in this area. However, few (if any) people believe that reports of audible differences between CD and other formats have anything to do with high frequency hearing.

Cheers,
David.

P.S. There are responses at 40kHz-50kHz via bone conduction. That's a whole separate topic!

The Emperor's New Sample Rate

Reply #98 –
Quote from: sld on
Quote from: MLXXX on

Because there is very little headroom.  The 44.1KHz Nyquist limit, 22050Hz, is awkwardly close to the upper limit of human hearing, of around 20000Hz.

Yes, and the ATH curve drops off severely as it approaches 20 kHz. If only our ears respond like studio monitors...

Improperly designed filters may interfere with signals far away from their roll-off frequency. The simpler (~cheaper) the filter the more likely it is to affect what it should not. The most problematic is the use of closed-loop (IIR) filters which tend to have nonlinear phase characteristics but provide steepest response and shortest start-up for the chosen order (~price).
You may not even have an idea about how many such filters has the sound actually passed between the microphone at studio and the loudspeaker/headphone at your home.
And I did not even mention the frequency/phase characteristics (~deformations) of microphones/amplifiers/cables/loudspeakers and whatnot which are unintended filters as well.

96+ kHz rate helps the software/hardware design (~reduces cost to achieve comparable results) as the "signals far away from their roll-off frequency" are much much farther away than with 44 kHz.

I think that 44 kHz is perfectly able to capture human-audible content. However, the practical results are plagued by the mentioned design/cost limits.
 

The Emperor's New Sample Rate

Reply #99 –
We keep coming back to the conclusion that 44.1/16 is perfectly adequate for distribution of the final product due to limitations in storage/bandwidth, but for any other use there are practical advantages to higher sampling rate and bit depth.