44.1 vs 96khz music - double blind study conducted...

[sergef50]

The benefits come during the processing at 96000hz

thats why just converting the already processed and mixed song wont make much of a discernable different.

Id like to see a test that double blind tested EQ and comp settings done on a piece of audio, processing at 44 and 96...thats when you would hear the difference because its during the processing the aliasing would become apparent.

thats why a lot of developers over sample there plugins as it moves the aliasing out of the hearable spectrum of sound.

THE above results are most likely true and completely valid...for a consumer looking at an end product purchase....

..but for an engineer it CAN be benefitial to process or synthesise when done digitally at doublerate samplerates.

(i suggest you check this out for yourself on a simple one track with either a synth or an eq curve onto an audio track....export both and use DBX)

[Stupid American Pig]

while I find the science of furthering hi fidelity fascinating, I still dont mind listening to a well recorded mono tape based recording if the tune moves me. It really isnt necessary to make me enjoy a song. In fact I think that the lo finess of some music totally adds to the appeal. Plastic ono band is one record that is gloriously lo-fidelity and I wouldnt want to hear it any other way.

[jeffb01]

For the most part, I get what Per Lichtman is saying and I agree 100%... great point. That's why people record at 24 bit when they know it's going to go on a 16 bit compact disk.

to emdot_ambient: 60 people would definately make this statistically significant... I've read a lot of scientific papers with an (n=9) or even less... the lower the n is (number of people) the harder it is to get the p-value below .05 which is widely accepted in scientific circles as being statistically significant. The p-value wasn't disclosed and I'm not going to look up the calculation (if I could find them), but it is safe to say that 60 people is a good amount of people for a study such as this. You could maybe argue if those 60 people weren't a representative population, but it seems they were of people that would more likely hear the difference (being experts in the field).

[Echo Voodoo]

THE above results are most likely true and completely valid...for a consumer looking at an end product purchase....

As far as I can tell from reading the article, this is the only thing they were testing - how the final product sounds to the end consumer. Whether sample rate makes a difference in production was not tested.

[ughnonumus]

I saw this in Mix magazine this month. Check it out:

http://mixonline.com/recording/mixing/a ... _sampling/

Here's the VERY brief summary. An extremely thorough, properly-conducted double blind test was done wherein several groups of people (audiophiles, professional audio engineers, average college students) were presented with 96khz audio, and that same audio converted to 44.1khz. The goal was to see if people could hear a difference. The findings of the study indicated that not even the audiophiles or the engineers could get above a 50% success rate (or at most, 1 or 2 points) - meaning they would have done just as well simply flipping a coin.

The authors of the study encourage others to try to reproduce it and get the same results, but if you read the methodology, it was pretty damn foolproof (IMO). Another strike against audiophile BS. 44.1khz is just fine for a final delivery format.

Thanks, for that.

[rifftrax]

http://www.avsforum.com/avs-vb/showthread.php?t=981557

That forum thread continues the discussion with a lot more being added on both sides. It includes comments by several very experienced engineers, some of whom output multi-channel recordings at 24/96. Some of them mention that DVD-A and SACD have an advantage over the compressed surround tracks used with DTS or Dolby Digital even if the higher samplerates are not an issue. As for myself, my post in that thread (#57) also addresses the fact that even if the study is taken as representative, it does not provide evidence that there is no benefit to working at higher samplerates during production but rather only that the final output can be oversampled without the listener losing anything.

Here is an excerpt from my post there:

Old 04-08-08, 03:03 PM #57 | Link
Per Lichtman
New Member

Join Date: Apr 2008
Posts: 1

I am surprised that there hasn't been more discussion about the impact of different samplerates/bitdepths prior to the delivery stage.

Working under the premise that the original study cited is accurate about the end delivery format of Redbook audio being indistinguishable to listeners from 24/96 PCM or other high quality audio formats, that does not mean that it irrelevant at other stages of the production process, whether tracking or mixing. Note, I am not saying that they do make a difference either, just that it is a different question. Fair enough?

Let's imagine for a moment that using the same, high quality A/D converter at 48 kHz and 96 kHz that there is a difference in the digital files created. Let us imagine that the difference is too minute for us to perceive under normal circumstances. What about when that mono or stereo signal is mixed with the other tracks recorded at the same time? Do the these minute differences start to add up as the signals are combined/summed?

As an analog, it is sometimes said the average listener will not consciously perceive less than 3db drop in the volume of sine wave. However, as any mixing engineer can tell you, dropping one or more signals by 2db can have a significant effect on how a mix is perceived or how they relate to each other, and we don't go around arguing that boosting an EQ band by 2db is the same as leaving it at unity.

If the analog applies here (I have not done sufficient empirical testing to confirm or disprove that at this point) the higher sample rates or bitrates could be useful during production even if they were not required in the final delivery. That is before even taking into account digital processing.

A lot of mixing and mastering engineers use some form of digital processing. Not all digital processing applies the same rate (if any) of oversampling. Now that is an area I have done a lot of testing in and I came to the conclusion (as did Massenberg) that there is can be a clear difference when working at higher sample rates for digital signal processing such as EQ and compression. This is comparitively easy to demonstrate because the output files can both be included on the same CD and listened to sequentially.

It seems, as has been discussed here earlier, that one way to test higher samplerate and bit depth differences in the final output that would rule out differences at other stages would be to create the file at 24/96 or 24/192 and then resample that file to Redbook specifications. Of course there are arguments about different resamplers being superior or inferior to each other so maybe it would have to be done multiple times. Thoughts?

you're missing the very rudimentary fundamental fact that all mathematical calculations that occur within audio dsp processing with the ONLY exception being pitch-shifting and resampling will effect ONLY THE AMPLITUDE OR BIT DEPTH OF THE WAVEFORM. You cannot intermodulate the frequency response of a waveform via equalization or compression to create a higher sample rate (or extra samples within the time-domain) anymore than you can upconvert 480i video to 1080p and call it "better".

When you use an equalizer to adjust a 16 bit waveform - if the equalizer operates with an internal precision of 32 bits - then you will actually have resultant waveforms with a higher bitrate than before the process took place. This is...how do you say...impossible to replicate with samplerates unless (again) you are pitch-shifting, resampling, or time-streching.

Look at it this way. When you import files of any samplerate into a DAW project - the individual sample points are either going to be misaligned and require upsampling and conversion to achieve a downmixed output at 1 specified sample rate or they are going to be snapped to line up with each other in the case multiplicatively related samplerates (22.05, 44.1 88.2, 176.4, etc.). The smartest thing (duh) would be to avoid the conversion process altogether and record anything with sample rates that are multiples of each other so that conversion simply involves sample-point doubling. Herein - you've not created ANY new information AT ALL in the sampling or time-related domain. Period. End of story. I mean honestly - you are saying that by using a high-precision equalizer that oversamples I have created new and valuable sample data that 'should' be used as part of the math in future signal processing to ensure quality of the "mix"? Bullcrap. Equalizers and compressors and flangers and phasers don't operate to alter the y-axis extension of the audio. That's not the function of them (they are not pitch-shifters if I remember correctly) and if that's what is happening then I'm pretty sure they are not doing their job...are they?

This is math guys. Not science fiction. Keep your x and y axes (yes, that's the plural) separate and you'll do okay.

[hifiboom]

the problem with this superbe test is, that there is no proof at all which and more important how many people are sensitive enough to capture the difference.

I mean take 99 colour-blinded people and one with normal seeing. Then do a test if there is a difference between two same pictures but with different colours.
You won't get significant results...
That does not proove that there is no differences at all. there is just no significant difference.

such a test simply fails in that case.
that test just prooves that the crowd of testpersons cannot hear a difference in average.

another important factor is: how much time does a user need to get used to/get sensitive to a certain speaker system to hear small audible differences.
The same analogy applies to when somebody buys new monitor speakers. He'll need some weeks to get used to the new system. So if you invite people to listen to different music on some high end audio speakers, that doesn't guarantee that he is able to hear a difference on that speaker system, as he is not used to the system itself and may be fooled at all.

[jeffb01]

I mean take 99 colour-blinded people and one with normal seeing. Then do a test if there is a difference between two same pictures but with different colours.
You won't get significant results...
That does not proove that there is no differences at all. there is just no significant difference.

but i think if 99% of a representative population was color blind, you would be wasting money printing up your product in color instead of black and white. As long as you agree that this is a representative population or even an ideal population (audiophiles and studio engineers), then that's the whole point - that it's not worth using 96 khz. Of course there is a technical difference - 96,000 as opposed to 44,000 samples per second - but if even studio engineers can't tell the diff, then why bother.

[gwahorton]

Just because something is inaudible doesn't mean it has no effect:
"Psychological evaluation indicated that the subjects felt the sound containing an HFC to be more pleasant than the same sound lacking an HFC." (HFC = High Frequency Component, defined as the set of frequencies above 22kHz, i.e those missed at a 44.1kHz sample-rate.)

And here's another study:
"Increasing the intensity of the inaudible HFC resulted in a significant increase in the comfortable listening level, and the subjective impression of sounds."

[The Chase]

The benefits come during the processing at 96000hz

thats why just converting the already processed and mixed song wont make much of a discernable different.

Id like to see a test that double blind tested EQ and comp settings done on a piece of audio, processing at 44 and 96...thats when you would hear the difference because its during the processing the aliasing would become apparent.

thats why a lot of developers over sample there plugins as it moves the aliasing out of the hearable spectrum of sound.

THE above results are most likely true and completely valid...for a consumer looking at an end product purchase....

..but for an engineer it CAN be benefitial to process or synthesise when done digitally at doublerate samplerates.

(i suggest you check this out for yourself on a simple one track with either a synth or an eq curve onto an audio track....export both and use DBX)

bingo!

Of course down-sampling post production will make practically no difference: 44.1Khz reconstructs the audible range completely (which I don't understand the reason for, but despite my tendency not to take authority's word for it, all scholarly sources seem to say so).

However, the results from, say, bit-crushing, distortion, synthesis, et cetera, carry quite a vast difference from 44.1 to 96.

[nollock]

Id like to see a test that double blind tested EQ and comp settings done on a piece of audio, processing at 44 and 96...thats when you would hear the difference because its during the processing the aliasing would become apparent.

thats why a lot of developers over sample there plugins as it moves the aliasing out of the hearable spectrum of sound.

Yes but that should be done by the plugin designer, it should be part of the design process of the algorythm. We shoudnt need DAW users to run their whole projects at high rates to compensate for shitty dsp coding.

[hifiboom]

jeff thats why the test is senseless,

- if you are using such a format (SACD or something like that), its an INDIVIDUAL DECISION, you have to do for your own. If you think you can hear a difference, it makes sense, doesn't matter if 4999 other people cannot hear a difference.

It has a value for yourself, because you either think you can hear a difference or you really do.

- From a marketing view, the test may have a value: they know now that they have to do a good marketing campaign to fool the mass of people (placebo effect) and to sell the 96khz stuff to average crowd, because the majority seems to not hear any difference.
-----

Finally the testbed and the results may be correct, but the conclusion "there is no audible difference" or "people cannot hear a difference" is simply wrong.
you can say at best: "in this test, the number of people that could hear a difference were not high enough to prove that there is a audible difference at a certain level of significance."

you don't need to do a such a test at all, because it is a fact that 96khz holds more information than 44,1 if recorded with proper equipment.

I don't doubt that, even with proper replay equipment, only a minority will be able to hear the difference at all.
But its just a question of time to find one with proper ears and sensitive enough to hear the difference. Humans are different. There are some that have eyes with 150% instead of the normal 100%, I don't doubt that there people around with "dog ears" up to 192khz. who knows.

A minority compared to a mass will liekly always be non-significant.


So this tests does not show any new or interesting information. Its more an interesting way, how people waste their life time imo.

[zircon]

- if you are using such a format (SACD or something like that), its an INDIVIDUAL DECISION, you have to do for your own. If you think you can hear a difference, it makes sense, doesn't matter if 4999 other people cannot hear a difference.

Evidently, no one could hear the difference, including audiophiles and audio engineers.

Obviously it's always up to the individual whether they want to do something or not. If I want to rub dirt on my body next time I get the flu, because I think it will make me get better, I am free to do that. MAYBE I will even convince myself that I AM getting better and very slightly improve my condition. But in the end, it's just a stupid thing to do. It's smoke and mirrors. Snake oil.

Your argument could be applied to any scientific study which studies a sample of the population in order to make a statistical inference. It's a very bad argument; if we followed your reasoning we would make no medical progress.

[Sonic Assault]

The benefits come during the processing at 96000hz

thats why just converting the already processed and mixed song wont make much of a discernable different.

Id like to see a test that double blind tested EQ and comp settings done on a piece of audio, processing at 44 and 96...thats when you would hear the difference because its during the processing the aliasing would become apparent.

thats why a lot of developers over sample there plugins as it moves the aliasing out of the hearable spectrum of sound.

THE above results are most likely true and completely valid...for a consumer looking at an end product purchase....

..but for an engineer it CAN be benefitial to process or synthesise when done digitally at doublerate samplerates.

(i suggest you check this out for yourself on a simple one track with either a synth or an eq curve onto an audio track....export both and use DBX)

Good point.
All the test shows is the premise of the Nyquist theory is roughly correct in relation to the minimum sample rate required for high quality audio. A true measure of difference would be a piece processed and recorded at each sample rate throughout the entire recording chain (as sergef50 points out) and then played back on source equipment operating at a higher rate than either of them. The claim to use high quality analogue equipment in the playback chain, although valid in providing the best possible analogue signal, is somewhat misleading as it could be perceived as increasing the validity of what the test claims to demonstrate - which it doesn't.

[rifftrax]

Just because something is inaudible doesn't mean it has no effect:
"Psychological evaluation indicated that the subjects felt the sound containing an HFC to be more pleasant than the same sound lacking an HFC." (HFC = High Frequency Component, defined as the set of frequencies above 22kHz, i.e those missed at a 44.1kHz sample-rate.)

And here's another study:
"Increasing the intensity of the inaudible HFC resulted in a significant increase in the comfortable listening level, and the subjective impression of sounds."

this is a totally different and esoteric realm of the idea though...