http://mixonline.com/recording/mixing/audi...ling/index.html (http://mixonline.com/recording/mixing/audio_emperors_new_sampling/index.html)
Good read. Thank you.
Some very interesting stuff in there. Two items that I can defintely concur with:
1) The higher sampling rates do not necessarily mean better sound. The primary reason IMO that DVD-A, DTS and SACD sound better is simply a better engineered recording. DTS rarely misses with their engineering quality. On the other hand I have SACDs that sound like crap and some that are excellent.
2) Room acoustics play a HUGE part in sound playback. I used to work for a company with a large RF testing chamber that was also acoustically anechoic. We put the speakers in there one weekend (JMLAB electras). the imaging was fantastic but the bass took on a very funny unnatural sound.
I have always been guided by the following 3 principles (which I borrowed)
1) Good speakers.
2) Lots of clean power amplification.
3) Good build quality.
My big disappointment is that with sagging music sales the recording companies make me pay an arm and a leg for something that should have been done right to begin with and worse yet sometimes the "audiophile" version is garbage.
I found the end of the article, talking about the sound changing by you moving just a few centimeters very interesting. It's something which most of us know and take for granted - but which is rarely taken into account in listening tests. It's the kind of argument which points out something which is forgotten because it is too obvious. Simple and trivial argument - huge consequences. Very interesting.
Previous discussion about the study the article mentions:
http://www.hydrogenaudio.org/forums/index....c=57406&st= (http://www.hydrogenaudio.org/forums/index.php?showtopic=57406&st=)
Previous discussion about the study the article mentions:
http://www.hydrogenaudio.org/forums/index....c=57406&st= (http://www.hydrogenaudio.org/forums/index.php?showtopic=57406&st=)
Sadly, but not at all unexpectedly, the AA crowd will grasp at any straw they can, and retail whatever half-remembered anecdote they can, to remain in denial about ABX testing.
http://www.audioasylum.com/forums/prophead...ages/43478.html (http://www.audioasylum.com/forums/prophead/messages/43478.html)
http://db.audioasylum.com/cgi/m.mpl?forum=...ight=EBradMeyer (http://db.audioasylum.com/cgi/m.mpl?forum=prophead&n=38080&highlight=EBradMeyer)
Very interesting article. It's nothing new, only confirming a couple of tests done here where a 'hi-definition' sample could not be ABXed against a properly dithered one.
I guess the only reason to buy a 'high-definition' media is to have a source that is mastered towards audiophiles, with full dynamic range and such, because there are no quality gains to these other formats.
What surprises me about that article is that so many people in the "pro" audio world still believes that hi-res audio is clearly better than standard reedbook CD audio.
About the author noting the audible difference possibly due to the superior time resolution of high-res audio, he and the paper he cites are just plain wrong, which again, puzzles me. Wrong in two ways. First, it has been determined experimentally that humans can detect ITDs (interaural time delays) up to 10 us, not 15 ms, as he notes. And second, time resolution on sampling systems is not just 1/fs, 1/44100 in case of CD, but more like 1/(fs*nº of discrete levels), or 1/(44100*65535), which is thousands of times smaller than the 10 us detectable under best circumstances. So CD is more than adequate in this sense.
What surprises me about that article is that so many people in the "pro" audio world still believes that hi-res audio is clearly better than standard reedbook CD audio.
About the author noting the audible difference possibly due to the superior time resolution of high-res audio, he and the paper he cites are just plain wrong, which again, puzzles me. Wrong in two ways. First, it has been determined experimentally that humans can detect ITDs (interaural time delays) up to 10 us, not 15 ms, as he notes. And second, time resolution on sampling systems is not just 1/fs, 1/44100 in case of CD, but more like 1/(fs*nº of discrete levels), or 1/(44100*65535), which is thousands of times smaller than the 10 us detectable under best circumstances. So CD is more than adequate in this sense.
Indeed. I think the whole time resolution thing is a bit of a red herring anyways. Saying "frequencies above 22kHz are audible given XdB of SNR" and "44.1kHz sampling has inadequate time resolution" are equivalent statements. Seperating the concepts of SNR and bandwidth from the concept of time resolution is not possible.
And as I showed (http://www.hydrogenaudio.org/forums/index.php?showtopic=49043&st=0)a while back, Red Book has a time resolution in the *low nanoseconds* anyway, well below the limits of audibility of time resolution.
You'd think that people who cared about such things would actually test it, or quote people who test it!
And as I showed (http://www.hydrogenaudio.org/forums/index.php?showtopic=49043&st=0)a while back, Red Book has a time resolution in the *low nanoseconds* anyway, well below the limits of audibility of time resolution.
You'd think that people who cared about such things would actually test it, or quote people who test it!
I stumbled upon this paper today, presented by
Ken Pohlmann at tape archivist's meeting a couple of years ago. In the excerpt below, isn't he presenting much the same sort of argument that you debunked, in his mentions of interaural difference and preservation of musical transient?
http://www.clir.org/activities/details/AD-...rs-Pohlmann.pdf (http://www.clir.org/activities/details/AD-Converters-Pohlmann.pdf)
Emphases mine. NB I have seen the Woszczyk 2003 preprint and he too makes the same arguments (without any new listening test data).
Sampling Frequency
Generally, sampling frequencies of 44.1, 48, 96, and 192 kHz are used in high-fidelity
recording. The usable audio bandwidth is one-half the sampling frequency, so higher
sampling frequencies provide a wider audio bandwidth. This is potentially useful because
musical instruments can generate content with wide bandwidths; for example, a cymbal
might have response of 90 dB SPL (sound pressure level) beyond 60 kHz, and a violin
might have content beyond 100 kHz.
Even so, the use of high sampling frequencies such as 96 and 192 kHz may seem
unnecessary. In rare cases, a person may be able to hear frequencies to 24 or 26 kHz, far
below the cutoff frequencies of 48 and 96 kHz. In most cases, high-frequency hearing
response is below 20 kHz. Thus, for steady-state tones, the higher-frequency response
may not be useful. However, a high sampling frequency provides additional benefits
beyond wide audio bandwidth. It can be argued that high sampling frequencies improve
the binaural time response, leading to improved imaging in multichannel recordings. For
example, if short pulses are applied to each ear, a 15-?S difference between the pulses
can be heard, and that time difference is shorter than the time between two samples at 48
kHz. Some people can hear a 5-?S difference, which corresponds to the time difference
between two samples at 192 kHz. In theory, a high sampling frequency might improve
spatial imaging.
Similarly, higher sampling frequencies provide improved temporal response. For
example, the sampling interval at 44.1 kHz is 22.7 ?S; at 192 kHz, it is 5.2 ?S. Musical
instruments can generate transients with rise times of less than 10 ?S. As another
example, room reverberation comprises a large number of reflections arriving at high
rates. For example, reverberation might comprise 500,000 arrivals per second; spaced
regularly, this time interval is less than 2 ?S. Human subjects are sensitive to interaural
time delays of between 2 and 10 ?S. Subjects have differentiated between a regular pulse
train and one with deviations of 0.2 ?S. Higher sampling frequencies clearly preserve
temporal response (Woszczyk 2003). In addition, higher sampling frequencies allow
greater latitude in the design of the anti-aliasing low-pass filter. For example, a lowerorder
slope may be employed, providing improved time-domain response; this is further
described below. Generally, high sampling frequencies can promote improved filter and
signal processing performance in the traditional audio (0 to 20 kHz) band. Ultimately,
because the limit of human hearing acuity is not yet known, the point of transparency of a
recording system cannot be known. In some cases, such as the conversion of monaural
speech recordings, a lower sampling frequency of 48 kHz may be used. However, for
highest audio fidelity, higher sampling frequencies of 96 or 192 kHz are recommended.
Human subjects are sensitive to interaural time delays of between 2 and 10 μS.
I hope they're sensitive to longer ones too!!!
Seriously 2us, where does he get that from?
And the 0.2us later on - are there any references?
10us is the usually quoted figure. I know where that came from (though don't have the reference to hand).
Of course any sampling rate is sufficient to preservse such inter-channel differences within the Nyquist bandwidth. Quantisation can impose a limit, but not anything relevant at 16-bits 44.1kHz! We've had this discussion several times before.
Cheers,
David.
First off, I am not an expert, however, it seems to me that there may be a 'flaw' in one of the conclusions of this study, if they only look at averages. One conclusion was that no one could tell between the two sample rates, but they don't mention how many particular individuals scored much higher than chance. If there were such individuals, then they, and they alone, would benefit from higher sampling rates. This makes sense if you consider that for there to be those who can't hear a difference, they must be conterbalanced by those who can hear a difference. To me all this study demonstrates is, yes, the majority of people can't tell and therefore don't need higher sampling rates. Personally, I think to satisfy my concerns, I'd like a study that tests many people looking for any who could discern a difference, and then further testing to see how good human hearing really is. Of course, if there's a flaw in my reasoning, please, don't hesitate in letting me know.
As well, I'd just like to point out that as much as some people seem to need to justify spending money on audio, others seem to need to justify not spending money on audio. Personally, I don't care about justifications or, even other peoples' preferences; I'm only looking for what quenches my thirsty ears.
First off, I am not an expert, however, it seems to me that there may be a 'flaw' in one of the conclusions of this study, if they only look at averages. One conclusion was that no one could tell between the two sample rates, but they don't mention how many particular individuals scored much higher than chance. If there were such individuals, then they, and they alone, would benefit from higher sampling rates.
If I recall Meyer and Moran's paper correctly, they did retest initial 'high-scorers', who ended up doing no better than chance in multiple tests. btw it's always better to read the actual primary research rather than dismissing it based on someone else's summary. You can buy the paper for $20 from JAES. There's also
a website supplement to the paper
http://www.bostonaudiosociety.org/explanation.htm (http://www.bostonaudiosociety.org/explanation.htm)
and Moran himself has posted here on HA in defense of the work.
Human subjects are sensitive to interaural time delays of between 2 and 10 ?S.
I hope they're sensitive to longer ones too!!!
Seriously 2us, where does he get that from?
And the 0.2us later on - are there any references?
I suspect he got them mostly from Woszczyk 2003, which itself turns out to be a review, rather than original research.
10us is the usually quoted figure. I know where that came from (though don't have the reference to hand).
Of course any sampling rate is sufficient to preservse such inter-channel differences within the Nyquist bandwidth. Quantisation can impose a limit, but not anything relevant at 16-bits 44.1kHz! We've had this discussion several times before.
Which is why I find it curious that Ken Pohlmann, of all people, would be retailing these arguments. EVen more curious is the schizophrenic nature of the paper, which offers these dubious arguments up front, but devotes a later section to emphasizing why double blind listening tests are necessary to confirm
differences.
btw it's always better to read the actual primary research rather than dismissing it based on someone else's summary. You can buy the paper for $20 from JAES.
Sure, it's better, but hardly required. As for dismissing the study, I did no such thing. I'm afraid it appears that you've completely overstated my position; perhaps you didn't read my 'comment' carefully?. Oh, and as for physically purchasing the paper, are you suggesting that unless one has purchased the study from "JAES", one has no right to comment on it in this thread?
If I recall Meyer and Moran's paper correctly, they did retest initial 'high-scorers', who ended up doing no better than chance in multiple tests....and Moran himself has posted here on HA in defense of the work.
Speaking of proof, please, can you provide a link?
Of course any sampling rate is sufficient to preservse such inter-channel differences within the Nyquist bandwidth. Quantisation can impose a limit, but not anything relevant at 16-bits 44.1kHz! We've had this discussion several times before.
Yes, we have. It's pretty well established theory, and well recognised and used in many fields (such as radar signal processing).
From http://www.bostonaudiosociety.org/explanation.htm (http://www.bostonaudiosociety.org/explanation.htm):
One of the authors, using a short repeated section of room tone on the Hartke disc mentioned above, obtained a positive result (15/15) at a gain of only 10 dB above our standard level. This setting produced sound levels clearly higher than those at the site, as the peak levels for this small vocal/percussion ensemble would have been 111 dB SPL on the loudest part of the disc.
This is an interesting result. In fact, that whole page is well worth a read.
This makes sense if you consider that for there to be those who can't hear a difference, they must be conterbalanced by those who can hear a difference.
This is an amazing use of logic and could have important application in other areas as well.
For instance, for there to be people who have never met aliens, they must be counterbalanced by those who have met aliens?
This makes sense if you consider that for there to be those who can't hear a difference, they must be conterbalanced by those who can hear a difference.
This is an amazing use of logic and could have important application in other areas as well.
For instance, for there to be people who have never met aliens, they must be counterbalanced by those who have met aliens?
Hehe.
The main flaw in 2tecs thinking is that because he isn't that experienced, he doesn't know and understand yet, that it is impossible to prove the non-existence of something anywhere in the world. But it is possible to prove the existence of something at specific locations in the world. This does NOT mean, that therefore something must exist somewhere in the world - it just means that you cannot test it. This is because we cannot look everywhere simultaneusly - we cannot test everything everywhere at the same time. Therefore, nonexistence of something regardless of location, is impossible to prove.... but it can be estimated: Probabilities. When in theory something doesn't exist, and besides of various tests and widespead awareness about the topic, no one succeeds in proving one single existence of the effect.... then it is reasonable to "asume", that it doesn't exist until proven otherwise. We have no proof that higher samplerates are unperceivable - but we also have zero evidence that it is perceivable - therefore we can ignore the issue until it starts to matter. We have no proof that the FSM doesn't exist - but we also have zero evidence that it exists - therefore we can ignore the issue until there is evidence. Thus, the burden of proof is always on the person who makes a claim about the existence of something.
And there is more to it: If apparently it is very difficult to prove the existence of something - thus, if its proposed effects seem very difficult to notice - then it is reasonable to asume, that even if it exists, its significance is very low. But if the significance of higher samplerates for listening are very low IF they exist..... then whats the point in spending all the resources for recording, storing, reproducing them? This makes higher samplerates look even more uninteresting, because it means: Higher samplerates do not seem to be perceivable - and even if they were perceivable by someone, then it is probable that in the majority of cases they are insignificant. Bummer!
btw it's always better to read the actual primary research rather than dismissing it based on someone else's summary. You can buy the paper for $20 from JAES.
Sure, it's better, but hardly required. As for dismissing the study, I did no such thing. I'm afraid it appears that you've completely overstated my position; perhaps you didn't read my 'comment' carefully?. Oh, and as for physically purchasing the paper, are you suggesting that unless one has purchased the study from "JES", one has no right to comment on it in this thread?
If I recall Meyer and Moran's paper correctly, they did retest initial 'high-scorers', who ended up doing no better than chance in multiple tests....and Moran himself has posted here on HA in defense of the work.
Speaking of proof, please, can you provide a link?
First..if you're going to attempt sarcasm, it's best not to display ignorance instead -- it's 'JAES', as I wrote, which stands for the Journal of the Audio Engineering Society.
Second, Moran posted to this HA thread about the paper (http://www.hydrogenaudio.org/forums/index.php?showtopic=57406&hl=meyer+moran) which I guess you were unable to call up by searching for 'moran' , like I just did.
Third, I misremembered. In fact, there was no retesting...because
no subject achieved a score where p< 0.05, unless levels were jacked up to abnormal levels. Quoted from the paper (emphasis mine):
The test results for the detectability of the 16/44.1 loop
on SACD/DVD-A playback were the same as chance:
49.82%. There were 554 trials and 276 correct answers.
The sole exceptions were for the condition of no signal
and high system gain, when the difference in noise floors
of the two technologies, old and new, was readily audible.
As the tests progressed, we repeatedly sorted the data
for correlations with age, sex, upper frequency hearing
limit, or experience. No such correlations have emerged.
Specifically, on music at normal levels as defined here,
audiophiles and/or working recording-studio engineers got
246 correct answers in 467 trials, for 52.7% correct. Females
got 18 in 48, for 37.5% correct. Those subjects able
to hear tones above 15 kHz got 116 in 256 trials, for 45.3%
correct; listeners aged 14–25 years old (who were, as it
turned out, the same group), also got 116 correct in 256
trials, 45.3%. The “best” listener score, achieved one
single time, was 8 for 10, still short of the desired 95%
confidence level. There were two 7/10 results. All other
trial totals were worse than 70% correct.
Furthermore, none of the more elaborate and expensive
playback systems (for which the subjects were all dedicated
amateur audiophiles, active students in a professional
recording program, and/or experienced working
professionals) revealed detectable differences on music,
again at levels as defined previously.
There is something else to take into account: To a limited degree, HA-Members have already tested ultra-high frequency perceivability en masse, simply by ABXing lossy encoders. Lossy encoders use a lowpass exactly because it is asumed that people cannot hear it - at least on normal equipment. If trained ABXers cannot hear 19-22khz, then why should they be capable of hearing stuff above 22khz? In other words, we have already tested this issue en masse, simply by ABXing between lossless and lossy. Test it yourself: just lowpass a lossless file at about 18khz... then try to ABX it on your best equipment.
There is something else to take into account: To a limited degree, HA-Members have already tested ultra-high frequency perceivability en masse, simply by ABXing lossy encoders. Lossy encoders use a lowpass exactly because it is asumed that people cannot hear it - at least on normal equipment. If trained ABXers cannot hear 19-22khz, then why should they be capable of hearing stuff above 22khz? In other words, we have already tested this issue en masse, simply by ABXing between lossless and lossy. Test it yourself: just lowpass a lossless file at about 18khz... then try to ABX it on your best equipment.
So you're saying that if you can't ABX lossy from lossless, you won't be able to ABX 'standard' digital audio from 'high-res' digital audio, either? (Or is this just one aspect of 'high-res' digital audio?)
So you're saying that if you can't ABX lossy from lossless, you won't be able to ABX 'standard' digital audio from 'high-res' digital audio, either? (Or is this just one aspect of 'high-res' digital audio?)
I am saying that if we cant ABX something which implements a lowpass, then why should we be able to ABX freqs which are even higher than that lowpass?
To simplify it:
IF for example we cannot ABX the removal of 18-22khz, then why should we be able to ABX the removal of 22-48khz? AFAIK, the human hearing curve doesnt go up the higher the freqs, but instead down.
There is something else to take into account: To a limited degree, HA-Members have already tested ultra-high frequency perceivability en masse, simply by ABXing lossy encoders. Lossy encoders use a lowpass exactly because it is asumed that people cannot hear it - at least on normal equipment. If trained ABXers cannot hear 19-22khz, then why should they be capable of hearing stuff above 22khz? In other words, we have already tested this issue en masse, simply by ABXing between lossless and lossy. Test it yourself: just lowpass a lossless file at about 18khz... then try to ABX it on your best equipment.
While the mass cannot generally ABX high-quality mp3s from source -- myself included in the mass -- a very few trusted HA posters (ones tending to be involved in LAME development) have reported the ability to ABX of the highest-quality lossy encodes on a regular basis. Don't know whether that's because of HF cut, or some other artifact.
While the mass cannot generally ABX high-quality mp3s from source -- myself included in the mass -- a very few trusted HA posters (ones tending to be involved in LAME development) have reported the ability to ABX of the highest-quality lossy encodes on a regular basis. Don't know whether that's because of HF cut, or some other artifact.
That might be a little exaggerated. I think the few posters you are referring to can regularly ABX select tracks, but I suspect that most tracks even they are not able to ABX a high-quality encode. And the ones that they are able to ABX they usually report such things as pre-echo, warbling, sandpaper sounds, things like that. I don't recall any case where they report it as being "less bright" or something similar that would indicate loss of high requencies.
IF for example we cannot ABX the removal of 18-22khz, then why should we be able to ABX the removal of 22-48khz?
I don't think you can make that leap. Lots of CD content rolls off above 20kHz, while some encoders keep everything up to about 19kHz, so the loss is tiny. The loss of 22-48kHz is huge in comparison. I'm not saying it's audible - I'm saying your argument is not safe.
IIRC there was one individual who could ABX a 19kHz low pass filter. This was back in the r3mix forum days, so you won't find the post on HA.
Cheers,
David.
While the mass cannot generally ABX high-quality mp3s from source -- myself included in the mass -- a very few trusted HA posters (ones tending to be involved in LAME development) have reported the ability to ABX of the highest-quality lossy encodes on a regular basis. Don't know whether that's because of HF cut, or some other artifact.
That might be a little exaggerated. I think the few posters you are referring to can regularly ABX select tracks but I suspect that most tracks even they are not able to ABX a high-quality encode.
Don't know about that....perhaps we should both review the HA archive. My memory is that a very few savants like gurubroolz are so attuned to mp3 artifacts that they CAN often ABX them, even at high CBR or VBR with the best LAME codecs...far more routinely than the average punter. It would not be surprising if mp3 codec tweakers were blessed/cursed with this talent. It would be surprising if a typically 40-ish mp3-denouncing 'audiophile' writing for Stereophile, could truly do the same.
And the ones that they are able to ABX they usually report such things as pre-echo, warbling, sandpaper sounds, things like that. I don't recall any case where they report it as being "less bright" or something similar that would indicate loss of high requencies.
That's why I questioned whether it was due to the HF cut.
the fact that i can hear the highest frequency available on CDs is a bit unsettling. i listen to a wide range of music, for some of which the higher frequencies often mean nothing and for others the higher frequencies are the sugar and spice of the release, including many industrial and noise albums. the fact of the matter is that i have hundreds of CDs in flac format on my harddrive (which costs about 200$) and room to spare. why not go the extra mile here and cover all frequencies audible to even the golden ears.
I don't think you can make that leap. Lots of CD content rolls off above 20kHz, while some encoders keep everything up to about 19kHz, so the loss is tiny. The loss of 22-48kHz is huge in comparison. I'm not saying it's audible - I'm saying your argument is not safe.
Good point, agreed. However, i guess you'd agree that if we had testing-material with significant content up to 22khz, speakers capable of reproducing it, and then ABX it against a 18khz-lowpassed version... that the results then do have some significance? Its no safe proof, right... but in that case, its weight would be significant, no? And its something which is much easier to test than signals >22khz, right?
IIRC there was one individual who could ABX a 19kHz low pass filter. This was back in the r3mix forum days, so you won't find the post on HA.
I think i remember hearing about him a few years ago already. I definatelly do not envy him.
the fact that i can hear the highest frequency available on CDs is a bit unsettling.
I find that statement quite unsettling as well, partially because most playback equipment isn't even capable of reproducing the full 22khz range.
why not go the extra mile here and cover all frequencies audible to even the golden ears.
Perhaps because until today, every single one of those "golden ears" failed to ABX what they can hear "easily". There isn't even one single valid and successfull >22khz DBT - nothing, zero. It is off course not impossible that you are an exception, but i guess you can understand why such stats make people suspicious unless the person can show that placebo can be excluded.
indeed, i took the test on the wrong frequency
http://www.rhintek.com/tutorial/Frequency/index.php (http://www.rhintek.com/tutorial/Frequency/index.php)
indeed, i took the test on the wrong frequency :P
http://www.rhintek.com/tutorial/Frequency/index.php (http://www.rhintek.com/tutorial/Frequency/index.php)
Eh, even if that were 22khz, this would tell you NOTHING about your hearing-capabilities, because those are test-tones. One can even engineer sounds so that you can hear even the slightest distortion regardless of your hearing capabilities (udial). So in short, test-tones are an entirely different beast than actual music - or do you listen to test-tones on your mp3-player while on the go? I can hear test-tones up to about 18khz.... but i'd never be able to perceive content that high in actual music. So in short: the true test is actual music, not testtones.
indeed, i took the test on the wrong frequency
http://www.rhintek.com/tutorial/Frequency/index.php (http://www.rhintek.com/tutorial/Frequency/index.php)
Eh, even if that were 22khz, this would tell you NOTHING about your hearing-capabilities, because those are test-tones. One can even engineer sounds so that you can hear even the slightest distortion regardless of your hearing capabilities (udial). So in short, test-tones are an entirely different beast than actual music - or do you listen to test-tones on your mp3-player while on the go? I can hear test-tones up to about 18khz.... but i'd never be able to perceive content that high in actual music. So in short: the true test is actual music, not testtones.
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
This makes sense if you consider that for there to be those who can't hear a difference, they must be conterbalanced by those who can hear a difference.
For instance, for there to be people who have never met aliens, they must be counterbalanced by those who have met aliens?
lol ... sure, and you're a master of logic. Oh, and btw, thanks, when you twist it so, you only really serve to illustrate and prove my point. The flaw with your insult, is there's no '
average' number of people who have 'met' aliens. Can you use a real example that demonstrates a real flaw in my reasoning? Now that shouldn't be so hard for someone who has got it all figured out already, eh?
I'm truly sorry, I guess it's wrong to try to discuss things around here; after all, you already seem to know everything. Oh, and ridicule is such a mature and useful tactic, isn't it?
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
LOL. I doubt those are equivalent to test tones.
Probably less musical though.
Cheers, Slipstreem.
First..if you're going to attempt sarcasm, it's best not to display ignorance instead -- it's 'JAES', as I wrote, which stands for the Journal of the Audio Engineering Society.
Well, apparently since you think it was an attempt at sarcasm, perhaps the attempt was actually completely successful? Oh, and as for the spelling thing, ya I misspelt it, however, you, apparently, still got the point.
Second, Moran posted to this HA thread about the paper (http://www.hydrogenaudio.org/forums/index.php?showtopic=57406&hl=meyer+moran) which I guess you were unable to call up by searching for 'moran' , like I just did.
Sigh, sorry, but how was I to know which "moran" post was the one that 'you' were talking about? From where I'm from, if you claim something, you should be the one to provide the reference, no?
Third, I misremembered. In fact, there was no retesting...because no subject achieved a score where p< 0.05, unless levels were jacked up to abnormal levels. Quoted from the paper (emphasis mine):
Whoops.
It seems to me that this quote illustrates my point concerning the better than average listeners, no?
Specifically, on music at normal levels as defined here,
audiophiles and/or working recording-studio engineers got
246 correct answers in 467 trials, for 52.7% correct. Females
got 18 in 48, for 37.5% correct. Those subjects able
to hear tones above 15 kHz got 116 in 256 trials, for 45.3%
correct; listeners aged 14–25 years old (who were, as it
turned out, the same group), also got 116 correct in 256
trials, 45.3%. The “best” listener score, achieved one
single time, was 8 for 10, still short of the desired 95%
confidence level. There were two 7/10 results. All other
trial totals were worse than 70% correct.
Perhaps you could explain this to me, since, clearly, I don't understand how you can believe that this study 'proves' that no one can hear any better than anyone else. My take on this study seems to be that a 16 bit path in an otherwise higher bit rate process simply produces no statistically significant auditory artifacts in the final product. Did I miss something?
First..if you're going to attempt sarcasm, it's best not to display ignorance instead -- it's 'JAES', as I wrote, which stands for the Journal of the Audio Engineering Society.
Well, apparently since you think it was an attempt at sarcasm, perhaps the attempt was actually completely successful?
Perhaps you could rethink the logic of that.
Oh, and as for the spelling thing, ya I misspelt it, however, you, apparently, still got the point.
Getting the point, and agreeing that the point is
intelligent, aren't the same thing.
Sigh, sorry, but how was I to know which "moran" post was the one that 'you' were talking about? From where I'm from, if you claim something, you should be the one to provide the reference, no?
A search for 'moran' brings up exactly 5 threads, one of which is this one. The others are:
AES conference London: High Resolution perception paper about listening test
Double-blind test of SACD and DVD-A vs. Redbook 16/44 in JAES September
SACD Ripping
Tired of MPC, I just have a question about OGG
Hmm, which one would likely contain input from the author of a paper about SACD vs Redbook that was published in JAES? Gosh, that's a real head-scratcher.
It seems to me that this quote illustrates my point concerning the better than average listeners, no?
Specifically, on music at normal levels as defined here,
audiophiles and/or working recording-studio engineers got
246 correct answers in 467 trials, for 52.7% correct.
One might note that's a performance still no better than
chance.
Females
got 18 in 48, for 37.5% correct.
Those subjects able
to hear tones above 15 kHz got 116 in 256 trials, for 45.3%
correct; listeners aged 14–25 years old (who were, as it
turned out, the same group), also got 116 correct in 256
trials, 45.3%. The “best” listener score, achieved one
single time, was 8 for 10, still short of the desired 95%
confidence level. There were two 7/10 results. All other
trial totals were worse than 70% correct.
Perhaps you could explain this to me, since, clearly, I don't understand how you can believe that this study 'proves' that no one can hear any better than anyone else. My take on this study seems to be that a 16 bit path in an otherwise higher bit rate process simply produces no statistically significant auditory artifacts in the final product. Did I miss something?
Actually it appears you've
imagined something, though at this point I wouldn't doubt you're missing something too. You seem to think that someone here (me?), or in the paper, is saying 'no one can hear any better than anyone else'. I predict few on HA would be stupid enough to make such an unqualified claim, and I am 100% certain I would never make such a claim.
As for the paper quote above -- you haven't yet read the whole paper, have you? just checking -- yes, it may demonstrate different native discriminative ability (which alas proved irrelevant to detection of difference between SACD and SACD-->Redbook, at normal levels; even the 'best' audio pro couldn't do it better than chance at p < 0.05), but you might note and think about factoring in the different number of trials for each group. 465 for the pros (and I can guess why there'd be more for them, can you?) vs 48, 256 and 256.
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
I do so as well, but this is still way more complex and organic than testtones. Not even Coils "Time Machines" is similiar to testtones, although thats probably almost as droney and pure as one can get in the genre.
I really don't have the time or inclination to separate the wheat from the chaff in this thread.
Sufficing to say: can we please keep it to adult discussion, and not a petty battle of wits? If needs be I will just remove all offending posts, whether they contain a morsel of useful input or not. It would be a shame, as it confuses responses to those points.
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
I do so as well, but this is still way more complex and organic than testtones. Not even Coils "Time Machines" is similiar to testtones, although thats probably almost as droney and pure as one can get in the genre.
Interesting, that's not a genre I am familiar with. Would you mind posting a (short, even 10s) sample for me to look at? I wasn't aware people listened to test tones for pleasure
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
I do so as well, but this is still way more complex and organic than testtones. Not even Coils "Time Machines" is similiar to testtones, although thats probably almost as droney and pure as one can get in the genre.
Time Machines owns! however Coil's ANS is much more test tone-y
thanks for the input, but i do listen to drone music which is often a single frequency fluctuating.
I do so as well, but this is still way more complex and organic than testtones. Not even Coils "Time Machines" is similiar to testtones, although thats probably almost as droney and pure as one can get in the genre.
Interesting, that's not a genre I am familiar with. Would you mind posting a (short, even 10s) sample for me to look at? I wasn't aware people listened to test tones for pleasure :)
A 10secs sample wouldn't explain much, because it isn't really music in the conventional way, but mood only... and even the word mood may imply something too complex. In short, imagine expressing aurally how it feels being "almost" in narcosis... the state where you're neither unconscious nor conscious... like a blank hypnotised stare - including that slight dizzyness.... thats how time machines sounds like. Expressed simply with rather pure soundwaves which slowly change in modulation.
Coil released a lot of their material - including AFAIK time machines - in extremely low numbers (hundreds, not thousands) on their own label, with intentionally doing no reprint. Filesharing and ebay are quite probably the only ways nowadays to get Time Machines. WP Link: http://en.wikipedia.org/wiki/Time_Machines (http://en.wikipedia.org/wiki/Time_Machines)
Actually it appears you've imagined something, though at this point I wouldn't doubt you're missing something too. You seem to think that someone here (me?), or in the paper, is saying 'no one can hear any better than anyone else'. I predict few on HA would be stupid enough to make such an unqualified claim, and I am 100% certain I would never make such a claim.
Yes, you did, right here:
If I recall Meyer and Moran's paper correctly, they did retest initial 'high-scorers', who ended up doing no better than chance in multiple tests.
So, were 'high-scorers' able to discern if a 16 bit process was used in an otherwise 32 bit process, or not?
BTW, (and only as a reply to your off-topic ad hominem)
a) When I want to use an example, personally I always try to provide the link rather than make all the readers guess at which page I'm referring too.
b) People don't actually need to completely read the actual paper, in order to add something intelligent, sadly however, the reverse doesn't appear to be necessarily true.
c) Yes, I was indeed being sarcastic, and on more than one occasion!
First off, I am not an expert, however, it seems to me that there may be a 'flaw' in one of the conclusions of this study, if they only look at averages. One conclusion was that no one could tell between the two sample rates, but they don't mention how many particular individuals scored much higher than chance. If there were such individuals, then they, and they alone, would benefit from higher sampling rates. This makes sense if you consider that for there to be those who can't hear a difference, they must be conterbalanced by those who can hear a difference.
That's right. Though the maths would be a bit more complicated in this case. Saying that one people did better than chance doesn't lead to a significant result.
Here's why : there is always a significance threshold above which a result is considered as successful. It is often set at "no more than 5% of probability that it was chance".
It means that in a set of random tests, no more than one out of 20, in average, would pass the test.
In this study, 60 listeners tried. In average, in the case all answers are random, then we should have got three listeners with "no more than 5% of probability that it was chance", in average !
Since this is the standard random result, maths are a bit more complicated. A much higher score is needed in order to really show a significant individual result among many other average ones.
Since the probabilities are small, we can say that the required result would have been around 5/60 = 0.08 % instead of 5%.
But even this result should not be taken as significant, because it can be seen as a post-experiment adjustment in order to fit personal convictions.
That's why when this case occurs, the usual practice is just to have the listener pass another test in order to confirm the result, and not bother with the maths.
It's like having 1000 people flip a coin ten times and having a small percentage getting heads 8 or 9 times out of 10. This is not beyond the realm of possibility. If they get heads 8 or 9 times out of 10 a second time around then maybe one can conclude the coin is biased.
In this study, people that got high scores were re-tested and failed. If they could truly hear a difference then they would have been able to repeat their high scores.
A 10secs sample wouldn't explain much, because it isn't really music in the conventional way, but mood only... and even the word mood may imply something too complex. In short, imagine expressing aurally how it feels being "almost" in narcosis... the state where you're neither unconscious nor conscious... like a blank hypnotised stare - including that slight dizzyness.... thats how time machines sounds like. Expressed simply with rather pure soundwaves which slowly change in modulation.
Coil released a lot of their material - including AFAIK time machines - in extremely low numbers (hundreds, not thousands) on their own label, with intentionally doing no reprint. Filesharing and ebay are quite probably the only ways nowadays to get Time Machines. WP Link: http://en.wikipedia.org/wiki/Time_Machines (http://en.wikipedia.org/wiki/Time_Machines)
very good description, the Coil drone stuff is more than feeling, it takes you to another place altogether depending on how much attention you're willing to give it. Time Machines was is one of the greatest albums i've heard to date, its very special and is much more than simply music.
if you're looking for a taste, i'd recommend this album http://thresholdhouse.greedbag.com/release...threshold-hous/ (http://thresholdhouse.greedbag.com/release/~the-remote-viewer-threshold-hous/)
it is a very close cousin to time machines.
Actually it appears you've imagined something, though at this point I wouldn't doubt you're missing something too. You seem to think that someone here (me?), or in the paper, is saying 'no one can hear any better than anyone else'. I predict few on HA would be stupid enough to make such an unqualified claim, and I am 100% certain I would never make such a claim.
Yes, you did, right here:
If I recall Meyer and Moran's paper correctly, they did retest initial 'high-scorers', who ended up doing no better than chance in multiple tests.
What I wrote there certainly is not even close to equivalent to writing 'no one can hear any better than anyone else', and if you can't see that, you're even more obtuse than I thought. Or you're trolling.
So, were 'high-scorers' able to discern if a 16 bit process was used in an otherwise 32 bit process, or not?
If you'd read the earlier thread about this, or read the paper, or even read
this thread you're on now more carefully, you'd know the answer. Which is that the inherent noise floor difference became audible when high system gain was applied in the test...a condition one would expect to highlight such differences...which in this case btw, were between DSD and Redbook, not 32 and 16 bit.
Are you suggesting that any single 16-bit process inserted into an otherwise 32-bit chain should be audible to 'golden ears' at normal levels in a blind test?
BTW, (and only as a reply to your off-topic ad hominem)
a) When I want to use an example, personally I always try to provide the link rather than make all the readers guess at which page I'm referring too.
b) People don't actually need to completely read the actual paper, in order to add something intelligent, sadly however, the reverse doesn't appear to be necessarily true.
c) Yes, I was indeed being sarcastic, and on more than one occasion!
Go do some reading. You know where the links are, and you know how to get the paper. I'm not here to be your special ed teacher.
30 second samples are available on Last.FM.
Time Machines (http://www.last.fm/music/Coil/Time+Machines)
One of the few pressed CDs of theirs I don't own.
I don't suppose that there are any musicians ‘out there’ willing to record a minute or so of music with 24-bit and 16-bit sample rates, and then present the tracks for us to ABX? It might be something as simple as playing back a pre-recorded sample (like karaoke background music), and then doing a recording in the two formats.
It would appear to be better to do a live take – but there is no way that a musician(s) could do it exactly the same way twice. If anyone is interested, I'll offer to host the tracks on my server. Lemme' know - it might go a long way towards assisting in a resolution to this discussion.
Andrew D.
www.cdnav.com
Why would you want it recorded twice (which introduces another variable) instead of just recording at 24-bit and dithering that version to 16-bit?
Also, what are the chances that mere mortals will have conditions with low enough background noise to make use of 24 bits?
I have to agree with pdq that introducing the variable of a different performance would undermine the ABX process.
However this thread appears to be not just about bit depth, such as three byte words of 24 bits or two byte words of 16 bits. It is also about sample rate, i.e. 44.1KHz vs a higher sampling rate.
A criticism sometimes raised is that it is unfair to derive a 44.1KHz sample from a higher sample rate rather than record direct from an analogue source. However, using such an 'unfair' derivation method (Audacity software), I was easily able to ABX (with foobar) the sound of a triangle being struck: see 'An easier exercise!' at post #68 (http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=17118&view=findpost&p=560190) of Listening Tests > Results for 24bit/96KHz test, vs. 16bit/44.1KHz.
An approach to address such criticism is to record one live performance but with two independent ADCs (same make and model), one set for 44.1KHz/24bits and the other for a higher sampling rate (e.g. 96KHz/24bits). [There would a slight issue of different non-linearities in the ADC process but perhaps the potential significance of that could be assessed through preliminary test recordings with both cards set at the same sample rate.]
Another issue is how filtering is implemented upon playback for different sampling rates and I think that is a variable rather difficult to evaluate the significance of. The question is: is the difference in played back sound merely a result of minor differences in actual filter implementation [recording and playback], or is it an inescapable result attributable to the use of different sampling rates, regardless of the precise filter implementation?
I would not bother with 24-bit to 16-bit conversion through noise shaped dither, but I would leave the 44.1khz recording intact at 24 bits. That would facilitate proving that a higher than 44.1KHz sampling rate can of itself make a detectable difference.
In this study, people that got high scores were re-tested and failed. If they could truly hear a difference then they would have been able to repeat their high scores.
No, there were no high scores at all, as Krabapple said above :
Third, I misremembered. In fact, there was no retesting...because no subject achieved a score where p< 0.05, unless levels were jacked up to abnormal levels.
Thus retesting was not needed. Which answers 2tec original question : both average results and individual results were taken into account. There was no positive result.
In the Detmold university listening test (http://www.hydrogenaudio.org/forums/index.php?showtopic=3390&st=75&p=374740&#entry374740), 200 listeners took the same challenge. Some scored above the significance threshold, but this was coherent with random guessing at the collective level.
However, one of them got a score of 20/20, which is significant even in a collective test with 200 listeners.
The authors said that unfortunately, a small noise at the beginning of one of the samples, though unheard by the listeners, may have biased the result.
Maybe also this listener really hears ultrasounds... I don't remember the study talking about his or her hearing ability in high frequencies.
Don't know about that....perhaps we should both review the HA archive. My memory is that a very few savants like gurubroolz are so attuned to mp3 artifacts that they CAN often ABX them, even at high CBR or VBR with the best LAME codecs...far more routinely than the average punter. It would not be surprising if mp3 codec tweakers were blessed/cursed with this talent. It would be surprising if a typically 40-ish mp3-denouncing 'audiophile' writing for Stereophile, could truly do the same
Guru ABXed artifacts of mp3, especially on low volume classical and natural instrument music. I think I have followed all of his ABX threads and I don't recall anything about him ABXing frequencies. Ability to ABX an mp3 vs CD doesn't mean it is the frequency that you can distinguish. In fact with all (pre)echo and such artifacts with mp3, higher frequencies are probably the least noticeable/contributing factor.
However, using such an 'unfair' derivation method (Audacity software), I was easily able to ABX (with foobar) the sound of a triangle being struck
All resampling is not equal. Compare the Adobe Audition Sweep tone resampling (using proper pre/post filters, or even without the filters) against that of Gold Wave for an easy explanation. Audacity (High-quality Sinc Interpolation) is significantly worse.
http://src.infinitewave.ca/ (http://src.infinitewave.ca/)
The resampled triangle sample from Audacity isn’t quite so colorful as the resampled sweep tone, but when comparing Audacity’s result to CoolEdit resampling (the precursor to Audition, for those who don’t know), the visual differences, especially in the critical midbands, are very obvious.
In situations like this I think it's worth mentioning (again) that your soundcard may be resampling during these listening tests as well.
The resampled triangle sample from Audacity isn’t quite so colorful as the resampled sweep tone, but when comparing Audacity’s result to CoolEdit resampling (the precursor to Audition, for those who don’t know), the visual differences, especially in the critical midbands, are very obvious.
The resampling quality in Audacity, when I tested it, was rather poor. I initially thought it was using libsamplerate (secret rabbit code), but apparently the use another library (http://audacityteam.org/wiki/index.php?title=Libresample), based on the same algorithm, due to some licensing issues (http://audacityteam.org/wiki/index.php?title=Libsamplerate). Libsamplerate 0.1.3 works very well, and should give excellent results, and you can build audacity on Linux to link against it.
I mentioned the visual differences due to Audacity's poor resampling, since not everyone has decent analysis software. The auditory differences in the triangel sample are so striking that anyone should be able to hear them, no special software required.
Playing with Audacity a little, I found something rather strange, or maybe not too strange as I haven’t had reason to investigate other programs since CoolEdit does such an excellent job.
Originally I generated a sweep tone in CoolEdit at 96kHz (100Hz to 48kHz Sine wave over 10 seconds). I just modified some settings I’d used some time ago. The Sine wave was modulated. I don’t remember the Modulated By value but the Modulation Frequency was 10Hz. Resampled in Audacity, this produced a very colorful Spectral View showing much harmonic distortion, aliasing, and spurious frequencies.
Afterwards I generated the same sweep tone without any modulation, as pure a sine wave as CoolEdit can generate. The Audacity resampling Spectral View of that looks almost like the CoolEdit resampling, very different than the modulated sweep tone. While neither are music, I would say the modulated tone is more representative of most music.
I don't suppose that there are any musicians ‘out there’ willing to record a minute or so of music with 24-bit and 16-bit sample rates, and then present the tracks for us to ABX? It might be something as simple as playing back a pre-recorded sample (like karaoke background music), and then doing a recording in the two formats.
It would appear to be better to do a live take – but there is no way that a musician(s) could do it exactly the same way twice. If anyone is interested, I'll offer to host the tracks on my server. Lemme' know - it might go a long way towards assisting in a resolution to this discussion.
Andrew D.
www.cdnav.com
Two different performances would definitely invalidate the test, and the only other means to compare the same recording without introducing heinous variables, is to either dither the 24 to 16, or record a live performance with two A/D converters, one set to 16bit and one to 24, from the same microphone input, at the same sample rate.
Regarding 16 vs 24 bits. I remember reading that our hearing distinguishes the differences at 18 bits resolution but no more, so 16 bits is a bit too low but 24 bits is simply an overkill. With a proper dithering, though, 16 bits is 'about as good as' 18 bits. I guess that does mean that Red Book format is sufficiently close to our hearing threshold in terms of resolution. In terms of sampling rate, 44.1Khz providing 22 Khz of bandwidth is more than enough.
You know what, I'll do this tonight or tomorrow... rip a 24-bit/48kHz DVD track (it'll be Sonic Youth, because its the only LPCM concert DVD I have) and then provide samples at:
24/48
16/48
24/44.1
16/44.1
I'll use SSRC and dither w/noise shaping for the 16-bit versions.
This high resolution track is free : http://www.hydrogenaudio.org/forums/index....showtopic=35624 (http://www.hydrogenaudio.org/forums/index.php?showtopic=35624)
Working link through Megaupload, post 31, page 2.
This high resolution track is free : http://www.hydrogenaudio.org/forums/index....showtopic=35624 (http://www.hydrogenaudio.org/forums/index.php?showtopic=35624)
Working link through Megaupload, post 31, page 2.
I had a quick listen to a 44.1KHz version (which I created with Audition 3.0) but there was no obvious difference for my ears, compared with the original 96KHz version. (I have never found the sound from guitar strings easy to detect deficencies in. I am not saying there are no differences in this laid back performance between the original 96KHz version and a 44.1KHz conversion; but merely that nothing 'stuck out' when I listened to the two versions.)
When I returned to the 96Khz/24bit sample of a triangle being struck, which I commented on at post #50 above, the differences were quite stark, so that is the sound sample I have selected for closer 'amateur analysis' ...
All resampling is not equal.
Thanks greynol, cabbagerat and AndyH for your comments.
I have now resampled the 96/24 struck triangle sound using Audition 3.0, set to maximum conversion quality (999). This has
not prevented the resampled sound sounding different. I obtained the following results:
[blockquote]A. Original 96KHz resampled to 44.1KHz:- duller, and sound appears to come more from from the left.
B. Original 96KHz resampled to 48KHz:- duller, and sound appears to come more from the left.
C. Above version A resampled back to 96KHz:- no improvement.
D. Above version B resampled back to 96KHz:- no improvement.
E. Original 96KHz resampled to 192KHz:- the 192KHz version had a slightly brighter sound.[/blockquote]
The differences between A, B, C or D and the original 96/24 sample were quite stark, and I did not perform ABX tests. Either factor -- change in position of the stereo image, or the loss of apparent high frequencies -- was quite noticeable.
The difference in situation E was relatively slight, so I ABXd, in order to satisfy forum guidelines. I was a little surprised to hear a difference in situation E. I had thought the filter performance in the audible range (and even a bit beyond that) would have been indistinguishable as between a sampling rate of 96KHz and a sampling rate of 192KHz.
Listening devices usedThe differences could be heard using the analogue outputs of the motherboard high definition audio on a pc running running Vista, feeding speakers; and using the analogue output of an Audigy 4 card on a computer running XP, feeding headphones.
Prima facie, a struck triangle is a valid test sound, as the triangle is an instrument of a symphony orchestra. However, could there be something anomalous about the particular triange sample? For example the microphones may have been so close that phase cancellations were occurring. In an auditorium, microphones could be quite some distance away from the percussion section of the orchestra.
I feel like a fish out of water writing on this particular topic. An amateur tredding down a path that others would have investigated years ago! Is there a consensus that a sample rate above 44.1KHz
can be beneficial, at least for some musical instruments?
I had always assumed there would be slight differences with a higher sampling rate, but this thread seems to challenge that.
The diffence signalAnother test I did was to subtract version C (original -> 44.1 ->96) from the original 96Kz version (using cooledit). This yielded a difference signal that sounded like a quiet version of the original file. I note that this particular type of test is independent of the precise performance of the sound card used to listen to the difference signal that cooledit computes.
I also subtracted D (original -> 48 -> 96) from the original. The result was not audible at a normal listening gain, despite the fact that when listening to the 96/24 versions
separately [version D and the original 96/24 version] I could hear a difference (confirmed with a quick ABX test).
What I wrote there certainly is not even close to equivalent to writing 'no one can hear any better than anyone else', and if you can't see that, you're even more obtuse than I thought. Or you're trolling.
All I can say is that is that's how it reads to me. Perhaps it's just your insults that make your statements seem confused?
Are you suggesting that any single 16-bit process inserted into an otherwise 32-bit chain should be audible to 'golden ears' at normal levels in a blind test?
Nope, as everyone else knows, I was just wondering if they'd retested those who scored above average.
Go do some reading. You know where the links are, and you know how to get the paper. I'm not here to be your special ed teacher.
My, aren't you being especially helpful!
What I wrote there certainly is not even close to equivalent to writing 'no one can hear any better than anyone else', and if you can't see that, you're even more obtuse than I thought. Or you're trolling.
All I can say is that is that's how it reads to me. Perhaps it's just your insults that make your statements seem confused?
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.
Are you suggesting that any single 16-bit process inserted into an otherwise 32-bit chain should be audible to 'golden ears' at normal levels in a blind test?
Nope, as everyone else knows, I was just wondering if they'd retested those who scored above average.
And you're
still wondering, even though, 'as everyone else knows', you were informed days ago that there were no scores 'above average' (at the p<.05 level)?
Btw, your statements seem confused. Are you saying you're NOT suggesting that that a proper re-test of a putative high scorer on a DSD vs Redbook test, would be to see if they could tell 16-bit from 24-bit audio?
Go do some reading. You know where the links are, and you know how to get the paper. I'm not here to be your special ed teacher.
My, aren't you being especially helpful!
You don't seem to have exploited the help you've already been given.
[blockquote]A. Original 96KHz resampled to 44.1KHz:- duller, and sound appears to come more from from the left.
B. Original 96KHz resampled to 48KHz:- duller, and sound appears to come more from the left.
C. Above version A resampled back to 96KHz:- no improvement.
D. Above version B resampled back to 96KHz:- no improvement.
E. Original 96KHz resampled to 192KHz:- the 192KHz version had a slightly brighter sound.[/blockquote]
I don't have access to Audition. Is it possible for you to make the resampled versions available for download somewhere. From online tests, it seems as though Audition's resampling is very good. Also, try out the free version of r8brain at 44.1kHz.
The differences between A, B, C or D and the original 96/24 sample were quite stark, and I did not perform ABX tests. Either factor -- change in position of the stereo image, or the loss of apparent high frequencies -- was quite noticeable.
I see three possibilities here:
- You can hear frequencies over 20kHz.
- Your soundcard's DAC is doing something different with different sampling rates, possibly performing resampling.
- Your speakers/amp do something strange with high frequencies - like excessive intermodulation distortion.
The difference in situation E was relatively slight, so I ABXd, in order to satisfy forum guidelines. I was a little surprised to hear a difference in situation E. I had thought the filter performance in the audible range (and even a bit beyond that) would have been indistinguishable as between a sampling rate of 96KHz and a sampling rate of 192KHz.
This makes me think that it's an effect of the resampler, and not your hearing. While it's possible you can hear frequencies above 20kHz, it's very unlikely you can hear above 44kHz. Maybe trying out r8brain is the way to go.
Prima facie, a struck triangle is a valid test sound, as the triangle is an instrument of a symphony orchestra. However, could there be something anomalous about the particular triange sample? For example the microphones may have been so close that phase cancellations were occurring. In an auditorium, microphones could be quite some distance away from the percussion section of the orchestra.
A struck triangle is still a legitimate music sound, wherever it's recorded from.
I feel like a fish out of water writing on this particular topic. An amateur tredding down a path that others would have investigated years ago! Is there a consensus that a sample rate above 44.1KHz can be beneficial, at least for some musical instruments?
No consensus I have come across. A lot of people have done tests (like the papers presented earlier in this thread) without statistically significant results.
Thx cabbagerat.
Did some quick tests with r8brain at its maximum quality settings. It performed more accurately than Audition 3. For example, my new file C (96 > 44.1 > 96) could be subtracted from the original struck triangle sample and leave no audible difference signal.
[I noted in my post above that Audition 3 produced quite an audible difference signal with its sample rate conversions via 44.1Khz compared with the original sample.]Am a bit pressed for time so will mention this: I found the r8brain 44.1KHz version did sound slightly duller (with my XP computer, Audigy 4 sound card, and headphones) and this difference was ABXable.
As this could have been due merely to differences in my sound card's filtering on playback, I then opened my new version C file and the original 96/24 sample, in foobar. The converted version still sounded slightly different (as if a tone control had been used to make the converted version slighter less bright). This was ABXable.
I don't have time at the moment to do uploads but may get around to that soon and can then post again.
No consensus I have come across. A lot of people have done tests (like the papers presented earlier in this thread) without statistically significant results.
MMn, that doesn't sound promising.
I don’t know about Audition 3, I still use CoolEdit200. The Help under Convert Sample Type here is quite clear. It says that quality settings of 100 to 400 give the best results. Higher quality settings can cause high frequency ringing because of the steep filters employed. Since this recording has so much energy above 22050Hz, it may be a good candidate for such problems.???
In the old Syntrillium forum, the word from the developer was to use 250 for the quality setting. Calculation times are greater at larger settings but perceived quality will not improve above 250.
C (96 > 44.1 > 96) could be subtracted from the original struck triangle sample and leave no audible difference signal.
What does this mean? If you compared the original with a resampled to 44.1 back to 96, there would have to be a major difference since nothing above 22050Hz could be in the resampled to 96kHz Do you simply mean you could not hear anything from the difference file?
I don’t know about Audition 3, I still use CoolEdit200. The Help under Convert Sample Type here is quite clear. It says that quality settings of 100 to 400 give the best results. Higher quality settings can cause high frequency ringing because of the steep filters employed. Since this recording has so much energy above 22050Hz, it may be a good candidate for such problems.???
In the old Syntrillium forum, the word from the developer was to use 250 for the quality setting. Calculation times are greater at larger settings but perceived quality will not improve above 250.
I did not read the help, but simply selected the maximum quality, assuming it would give the best result. There was also an option not to use any filtering at all but that didn't seem a good idea so I left filtering on. In light of this, I guess I'd better read up on what the r8brain help has to say about the quality setting.
C (96 > 44.1 > 96) could be subtracted from the original struck triangle sample and leave no audible difference signal.
What does this mean? If you compared the original with a resampled to 44.1 back to 96, there would have to be a major difference since nothing above 22050Hz could be in the resampled to 96kHz Do you simply mean you could not hear anything from the difference file?
Yes, simply that. During playback the volume bars on cooledit showed a burst of signal at the beginning of the difference file, but I could not hear that burst (using the same gain setting as for listening to the unaltered sample). As you suggest, there would have been high frequency content in the 22050 and above range.
I find this technique of listening to a difference file quite useful for pinpointing weaknesses (or anomalies) in digital signal processing. It enabled me to establish that Audition 3 at its maximum quality setting was fractionally altering the overall level of frequencies well down into the human audible range, when perfoming a 96KHz to 44.1KHz and then back to 96KHz conversion.
*************
I suspect that this matter of what filtering to use may ultimately prevent coming to an agreement on the resampling question in relation to 44.1KHz vs 48KHz. Any attempt to present a case that the 44.1KHz version sounds different can be dismissed by reference to filtering effects. The Nyquist limit being relatively close to the upper limit of human hearing, no doubt makes filter design difficult.
Again, perhaps you could post some lossless samples of
1) the original audio
2) the audio after your processing
'
With #1, other people could attempt to replicate what you did (resampling)...with 1 and 2 they could replicate the listening test you did and measure what, if anything, is different.
MLXXX
It'd go a long way if you could please post your ABX results.
Andrew D.
Yes I can and will do that, digital, though at this stage my main interest is receiving suggestions as to what software to use for the sample rate conversions.
I could merely upload the r8brain high quality conversions (which I have informally ABX tested and found to be distinguishable), but if someone can suggest software that will have less effect on the human audible tonality for a 44.1KHz sample rate than r8brain, then I am ... er ... all ears.
One suggestion (AndyH) was a lower quality conversion setting for Cooledit, and I guess I could try that with Audition 3, or with my expired trial version of Cooledit.
I wouldn't trust much signal substraction techniques here, because the filtering may introduce small delays in the processed signals. This delays will leave a small residue when doing the substraction, no matter how good the resampling.
To check if there is such an issue, with Audition/CoolEdit Pro take a spectral look at the difference signal with the FFT view or whatever it was called. If it has significant content (say over -120 dB) only at the ultrasonic part, then both the resampling is ok and hasn't caused any delay either.
AFAIK Audition/CoolEdit Pro resampling is very good, but make sure you have the pre/post filtering option enabled. 400 quality will make a very sharp filter, leading to a possible long ringing at half the sampling frequency (22050 Hz) if the signal has content at this exact frequency. As said, try with 250 to see if this makes a difference.
Inconclusive investigations at a target rate of 44.1Khz:
Quality settings
The wording of the Cooledit 2.00 help on conversion quality includes the following:
[blockquote]Low/High Quality
Use this slider to adjust the quality of the sampling conversion.
Higher values retain more high frequencies while still preventing aliasing of higher frequencies to lower ones, but the conversion process takes longer. A lower quality setting requires less processing time, but results in certain high frequencies being 'rolled off', leading to muffled sounding audio.[/blockquote]
I decided to give Cooledit another trial (on an old pc on which it had not previously been trialled). I found that reducing the setting from 999 to 800, 600, 400, 250 and down to 150, merely softened the clarity of the sound more and more, but did not improve it.
The behaviour was similar with r8brain - it sounded best (to my ears) at the highest quality setting. Similarly with Audition 3.0.
Subtracting converted files from each other
Cooledit produced odd results in that it was not internally consistent up to midrange frequencies. A low quality conversion subtracted from a high quality conversion did not merely yield whispers of very high frequencies but a whole swathe of sound well down into audible frequencies.
Both r8brain and Audition 3.0 produced the same level of audio output up to midrange frequencies. Subtractions between r8brain and audition 3 yielded only very high frequency audible sounds, and the sound was faint. I was inclined to reject Cooledit, based on its internal inconsistency.
No proven converter available
My difficulty in proceeding further with this exercise is that the highest quality level conversions of r8brain and Audition 3.0 are yielding slightly different sounds, and they both differ from Cooledit.*
In these circumstances, I cannot draw any definitive conclusion from any positive ABX result when comparing the 96KHz version of the struck triangle to a 44.1KHz conversion using any of these three items of software.
Any difference I heard could be explained away by reference to the filter characteristics used for the conversion.
The only tentative conclusion I can draw is that 44.1KHz may be too low a sampling rate for practical filters. If that is so, then a higher sampling rate may be called for.
I would mention that to my ears there is a greater apparent difference between the original sound sample and any of the conversions (listened to with foobar, an Audigy 4 card, and headphones) compared with differences between the conversions. All of the conversions sound a little duller than the original.
However there could be a large number of reasons why a sound card might perform differently with a 96KHz input than with a 44.1KHz input, including its own filter settings.
As for the differences in sound resulting from use of the three forms of conversion software, I have to assume the filter implementation is different in each case, and this is giving a slightly different colour to the processed sound.
______________________
* ABX report for two of the converters, r8brain 1.9 and Cooledit 2.0, operating at their highest quality settings and converting a 96Khz/24 bit file to 44.1Khz/24 bits:-
foo_abx 1.3.1 report
foobar2000 v0.9.5.1
2008/05/06 04:02:33
File A: C:\Documents and Settings\All Users\Documents\triangle-2_2496_r8brain-conversionTo44-1--HighestQuality.wav
File B: C:\Documents and Settings\All Users\Documents\triangle-2_2496--Cooledit--ConversionTo44-1--quality999.wav
04:02:33 : Test started.
04:02:54 : 01/01 50.0%
04:03:48 : 02/02 25.0%
04:04:53 : 03/03 12.5%
04:05:28 : 04/04 6.3%
04:06:41 : 05/05 3.1%
04:06:43 : Test finished.
----------
Total: 5/5 (3.1%)
And now a similar ABX report comparing the conversions of Cooledit 2.0 and Audition 3.0 with each other:-
(Note: these two sounded quite similar and were not easy to ABX!)
[blockquote]foo_abx 1.3.1 report
foobar2000 v0.9.5.1
2008/05/06 19:48:04
File A: \\action\shareddocs\triangle-2_2496--Audition3convertingTo44-1KHz-quality999.wav
File B: \\action\shareddocs\triangle-2_2496--Cooledit--ConversionTo44-1--quality999.wav
19:48:04 : Test started.
19:48:55 : 01/01 50.0%
19:49:53 : 02/02 25.0%
19:51:59 : 03/03 12.5%
19:52:39 : 04/04 6.3%
19:53:00 : 05/05 3.1%
19:53:23 : Test finished.
----------
Total: 5/5 (3.1%)[/blockquote]
LINKS TO THE THREE CONVERSIONS:
Audition 3.0 version: http://www.hydrogenaudio.org/forums/index....ost&id=4441 (http://www.hydrogenaudio.org/forums/index.php?act=Attach&type=post&id=4441)
Cooledit pro 2.0 version: http://www.hydrogenaudio.org/forums/index....ost&id=4442 (http://www.hydrogenaudio.org/forums/index.php?act=Attach&type=post&id=4442)
R8brain 1.9 version: http://www.hydrogenaudio.org/forums/index....ost&id=4444 (http://www.hydrogenaudio.org/forums/index.php?act=Attach&type=post&id=4444)
LINK TO THE ORIGINAL SOUND SAMPLE:
The original 96KHz/24-bit sample of a triangle being struck can be located on the excellent PCABX test page: http://64.41.69.21/technical/sample_rates/index.htm (http://64.41.69.21/technical/sample_rates/index.htm)
The relevant sample is the one marked "Triangle Reference Presented At 24/96".
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? Personally, I simply don't understand what you hope to gain by insulting people. However, if you feel it helps your cause, please, don't stop simply on my account.
And you're still wondering, even though, 'as everyone else knows', you were informed days ago that there were no scores 'above average' (at the p<.05 level)?
As I said, I "was" wondering. Perhaps you should try reading my post more carefully?
Btw, your statements seem confused.
There you go with the insults, again. Good luck with that!
Are you saying you're NOT suggesting that that a proper re-test of a putative high scorer on a DSD vs Redbook test, would be to see if they could tell 16-bit from 24-bit audio?
Why don't you quote me?
You don't seem to have exploited the help you've already been given.
I'm doing just fine, thanks. By the way, I feel I must compliment you on how well you've chosen your nickname. I found it amazingly appropriate.
Thus retesting was not needed. Which answers 2tec original question : both average results and individual results were taken into account. There was no positive result.
Thanks!
In the Detmold university listening test (http://www.hydrogenaudio.org/forums/index.php?showtopic=3390&st=75&p=374740&#entry374740), 200 listeners took the same challenge. Some scored above the significance threshold, but this was coherent with random guessing at the collective level. However, one of them got a score of 20/20, which is significant even in a collective test with 200 listeners. The authors said that unfortunately, a small noise at the beginning of one of the samples, though unheard by the listeners, may have biased the result.
Maybe also this listener really hears ultrasounds... I don't remember the study talking about his or her hearing ability in high frequencies.
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.
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.
There is a possibly important aspect of the test equipment in such comparisons. Many, possibly most, soundcards have somewhat different performances at different sampling rates. If no one detects any difference in the audio, the soundcard differences probably don’t matter, but if there are positive scores, we have too many variables to eadily determine why. at the very least we need to repeat the tests with different, high quality, DACs.
With higher sound pressure levels, more intrinsic audio differences will be audible. This isn’t specific to different sampling rates, it is a normal part of every day sound. Suppose something is audible (only) at very high levels. The basic question of paragraph 1 applies. Do we care? Why do we care? Will the fact ever be relevant at any time other than during such a test?
The equipment again comes into consideration. It is possible to build enormously powerful amplifiers, but the transducers for converting that electrical power into sound are another matter. Speakers without a lot of distortion get to be very expensive. I think it is probably not possible to brush aside the strong possibility that because of complex interactions in the transducers, higher frequency distortions might have effects on audible frequencies that would not occur otherwise.
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.
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%)
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.
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.
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.
Thanks for your observations, cabbagerat.
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.
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 (http://www.hydrogenaudio.org/forums/index.php?showtopic=63123).
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? (http://www.hydrogenaudio.org/forums/index.php?showtopic=63123)
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.
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?
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.
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.
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.
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.
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.
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.
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.
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 (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.
... 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.
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?
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.
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 (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 (http://www.m-ideas.com/sullivan/audio/udial.flac)
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...
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 (http://world.std.com/~griesngr/intermod.ppt) 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.
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!
20 bits is useful for noise shaping (http://en.wikipedia.org/wiki/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.
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!
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.
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.