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Real-world comparisons of 24-bit and 16-bit music from Linn Records -
rpp3po
post Aug 19 2009, 10:22
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QUOTE (Cavaille @ Aug 19 2009, 11:16) *
Just look at the manual, you can download it for free from E-MU.


I did, that was the source of my claim. Page 12, about the E-MU USB Audio Control Panel: "5. Sample Rate - Allows you to set the system sample rate...".

It doesn't talk anywhere about being able to switch automatically.

This post has been edited by rpp3po: Aug 19 2009, 10:26
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Cavaille
post Aug 19 2009, 10:37
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QUOTE (rpp3po @ Aug 19 2009, 11:22) *
QUOTE (Cavaille @ Aug 19 2009, 11:16) *
Just look at the manual, you can download it for free from E-MU.


I did, that was the source of my claim. Page 12, about the E-MU USB Audio Control Panel "5. Sample Rate - Allows you to set the system sample rate...".

It doesn't talk anywhere about being able to switch automatically.
True. That is written there. I´m sorry for referring to the manual. They don´t even talk about automatic switching! How could they forget it? Well, with an actual application it looks like this: "Allows you to set the sample-rate" - but only when no data is being sent to the interface. With ASIO it switches automatically. I´m very sure that other owners of the 0202 could clear that up.


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rpp3po
post Aug 19 2009, 11:12
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The ASIO protocol knows commands that allow an application to request a specific sample rate. So if Foobar is doing this, what you are saying about the 0202 makes sense.
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2Bdecided
post Aug 19 2009, 11:26
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QUOTE (rpp3po @ Aug 19 2009, 10:18) *
QUOTE (2Bdecided @ Aug 19 2009, 11:12) *
1. use the provided 24/96 as a source (call that A)
2. halve the volume (23 bits! wink.gif ) (call that B)
3. resample (B) to 16/44 (call that C) using fb2k or whatever
4. ABX (B) vs ©
5. resample © back to 24/96 (call that D)
6. ABX (B) vs (D)
What's the point of repeating above procedure with the difference of halving the volume (-1db would be sufficient) and omitting dither? Both aspects are additional source of error.
Whether you dither or not is up to you - I didn't say you should or you shouldn't - it's irrelevant on the 24-bit source, and it's just one of several choices you have to make if you specify the 24/96 > 16/44 conversion parameters yourself.

Why not -1dB? Mainly because halving the volume is a very "clean" mathematical operation, with or (especially) without dither. -1dB is probably fine too, but leaves the possibility of mathematical errors in the scaling algorithm, specific systematic errors (effectively non-linear quantisation steps) if dither isn't used, and the possibility of driving certain trashier sound cards (admittedly ones completely unsuited to this test) into slightly non-linear regions.


Given that we're chasing a small (non existant?) difference, better to be safe than sorry.

...but my post was in response to Cavaille's procedure, not yours.

Cheers,
David.

P.S. In case you're about to, please don't critique these suggestions as if they come from a clueless audiophile - I've already implied what I think - i.e. ~14 bits is probably sufficient here. Even so, it's still worth doing the test properly.

(Even if the material, as I've already said, hasn't preserved the dynamics of a live performance, and uses a disappointingly high amount of DRC for Linn records - never mind for 24-bits!).


This post has been edited by 2Bdecided: Aug 19 2009, 11:30
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rpp3po
post Aug 19 2009, 13:01
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David, maybe my question got across a little rough. I don't question your expertise in digital audio.

When there's only ~14 bits above noise, I'm wondering, what Cavaille is hearing. There's no problem conserving this within the specs of Redbook. But maybe it's HF content, for which I have yet to find a paper confirming audibility in a controlled environment. Industry interest would be great, suspicious when nothing gets found anyway.

This post has been edited by rpp3po: Aug 19 2009, 13:01
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Axon
post Aug 19 2009, 15:46
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Not to put tooooo fine a point on this, but Cavaille is, statistically, predicted to have a much higher hearing limit than (most of) the rest of us.

That said, I really would like to see ABX results with 2BDecided's test.
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Cavaille
post Aug 19 2009, 18:24
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Guys, I decided not to do an upsample of the the 16 Bit material because it would have colourized the outcome when compared to the original 24/96 (and yes, it would have introduced several distortions from clipping - just like rpp3po said). I think that we all know that every resampler changes the sound (even if only slightly) and I also think that we all know that differences between 24/96 & 16/44.1 are subtle at best. By upsampling I would have added another slight subtlety - that way the outcome would not be representable.

I simply turned to editing the 24/96 file in order to have higher headroom. BTW, even with that I wouldn´t have heard quantization errors - simply because the material is too loud for this and quite a bit compressed with brickwall limiting.

2BDecided´s way of doing a test is complicated but seems to be the best method so far.

And why should I (statistically) have a higher hearing limit than others here? I´m almost 34 years old and my hearing goes up to 18.500 Hz - perfectly normal, I´d say.


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bandpass
post Aug 19 2009, 19:08
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QUOTE (2Bdecided @ Aug 19 2009, 11:26) *
2. halve the volume (23 bits!) (call that B)

If you save this as float32, you can retain all 24 bits (for the subsequent conversion at least).
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rpp3po
post Aug 19 2009, 19:42
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QUOTE (2Bdecided @ Aug 19 2009, 12:26) *
Why not -1dB? Mainly because halving the volume is a very "clean" mathematical operation, with or (especially) without dither. -1dB is probably fine too, but leaves the possibility of mathematical errors in the scaling algorithm, specific systematic errors (effectively non-linear quantisation steps)


BTW, isn't the potential quantization error when applying a -1db attenuation (much) smaller than 6db in any thinkable case?

Halving of the volume wouldn't add any mathematical error <6db but raise the noise floor by +6db instead, since you just blank the LSB. Then what would be the advantage of adding 6db uncorrelated noise in your proposal over adding <6db uncorrelated noise (the distribution up and down rounding errors should be uniform) in my proposal?

On the bottom line this is just nit-picking, though. There should be plenty of headroom left with both approaches. Also I may overlooked something, and then I'd be interested in the explanation.

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krabapple
post Aug 19 2009, 20:16
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So has anyone looked at /listened to the *other* two files, that I linked to, at Barry Diament's site?

They are mastered very 'old school' -- just a single true 'peak' sample per channel. Crest factor ~20 dB. Less than .1 dB difference between corresponding channels. Spectral content up to the limits of each format.

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MLXXX
post Aug 19 2009, 23:44
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I have downloaded them: 36MB and 118MB. In a quick informal test the two versions seem to sound slightly different on my system in terms of their frequency response, which may be a comment on my system! Haven't had the chance to analyse the files.
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2Bdecided
post Aug 20 2009, 09:55
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QUOTE (krabapple @ Aug 19 2009, 20:16) *
So has anyone looked at /listened to the *other* two files, that I linked to, at Barry Diament's site?
I hadn't, because I assumed you were linking to something I'd already seen, but these are new.

They're great recordings. I can't ABX them, but I found something interesting (well, if your "interesting" threshold is really low and skewed towards obscure audio issues!).

In fb2k 0.9.6.2, WinXP, M-audio 2496, the files play at a slightly different pitch/speed! The 44.1 / 16-bit version is fractionally too fast / high pitch.

There's some sample rate conversion going on, because I know the 2496 mutes slightly when it changes sample rate, whereas it doesn't mute when I switch from one file to the other. However, I don't have fb2k's sample rate conversion enabled - in fact, when I drop SSRC into the DSP chain to convert everything to 96kHz, the problem goes aware.

Very strange.

If I try to ABX them, the problem goes away and the 2496 mutes between the tracks as it changes sample rate - so the playback route is different within the ABX comparator than in normal playback.

I thought I was going mad and imagining it - but I've just loaded up 20 copies of the tracks in the play list, randomised them, and with my eyes closed I can pick out the 44.1kHz ones, because of the slightly higher pitch.

I don't know if this is some configuration freak on my PC, but it shows the kinds of problems you can face in such a comparison.

I've just downloaded the free mp3 extracts - it sounds like a really nice CD. Not sure about the comment on their website about the difference between CDs and CD-Rs (they charge more for a CD-R!), and the fade-in of the 24-bit file is only 16-bits (very strange!) and there's some bit-sticking during the fade out too (have a look - continuous runs of identical samples - can't happen by accident!), but the music recording seems fine.

Cheers,
David.
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MLXXX
post Aug 20 2009, 10:31
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David, I used Audacity at a project rate of 96KHz. That allowed me import the 24/96 file first and then the 16/44 file. I was the able to time align the two files despite their different sample rates.

QUOTE (krabapple @ Aug 18 2009, 04:20) *
Here's another publicly-available pair of clips , offered by the engineer for comparison of 16/44 to 24/96

http://www.soundkeeperrecordings.com/format.htm

The 24/96 file appears to be 2 or 3 samples (at 96KHz) ahead of the 16/44 file, at the beginning of the music. When time aligned, and with one file inverted, the result is inaudible at a moderate listening level (at least at the beginning; there seems to be a very slight drift in the alignment later in the files). A spectrum analysis shows some low level content below about 400Hz, and some high frequency content commencing at about 16KHz; but the null is quite good.

I imagine though that many people might prefer to downsample the 24/96 file to 16/44.1 and upsample that to 24/96, for critical ABX testing.

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2Bdecided
post Aug 20 2009, 11:26
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Yes - same in CEP - I can resample one to the other can get alignment within a fraction of a sample, but not perfect. To be expected with different resampling algorithms - there's nothing devious going on.

Cheers,
David.
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krabapple
post Aug 20 2009, 19:42
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QUOTE (2Bdecided @ Aug 20 2009, 04:55) *
I've just downloaded the free mp3 extracts - it sounds like a really nice CD. Not sure about the comment on their website about the difference between CDs and CD-Rs (they charge more for a CD-R!), and the fade-in of the 24-bit file is only 16-bits (very strange!) and there's some bit-sticking during the fade out too (have a look - continuous runs of identical samples - can't happen by accident!), but the music recording seems fine.

Cheers,
David.



Re: comments on the site, Diament adheres to a significant amount of audio woo, as witnessed by his webpage on the benefits of vibration control, and many dubious claims in his posts to Steve Hoffman's site -- but he claims to be a 'purist' in these matters which is why I thought these samples would be ideal for our purpose here. It should be possible to ask him directly why you're seeing the things you're seeing (like the 16-bit fades)

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Examiner
post Aug 21 2009, 22:06
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I gave it a try, yet I couldn't ABX them. My score after 6 runs was 3/6, which is exactly 50%, and what you get when totally guessing without any clue. I had the subjective impression that the instruments sounded more voluminous with the 24/96 file, yet I failed utterly to ABX them.

- I used the 24/96 source file to create my own master
- opened it with Audition
- normalized to 93% to prevent clipping
- saved normalized file as "Bheki 2496.wav"
- than downsampled (including dithering) the still loaded normalized file to 16/44 and saved it as "Bheki 1644.wav"

footbar2000 Asio > Asus Xonar dx > HD 555
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MLXXX
post Aug 22 2009, 13:21
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I am sorry Linn Records, but I am unable to obtain persuasive ABX results, by taking the 24/96 version, reducing it by 1dB, downsampling that to 16/44.1 with noise-shaped dither (Audacity software), upsampling back to 24/96 and comparing with the 1dB reduced version of the original.

This is both with your track referred to at post #1, and with the SoundKeeper Recordings track referred to by krabapple at post #9.

With the two tracks provided (a jazz instrumental piece, and a male vocal with guitar accompaniment), using a reasonable quality sound reproduction system, any audible difference that exists appears to be so subtle as not to be detectable with average adult human hearing.

I imagine that if some adults can detect a difference, it will be a very minor difference for those people, and perhaps noticeable only in direct A B comparisons.

I note that Cavaille's positive ABX result (though he went to the trouble of correcting for the identified amplitude disparity between the files provided for comparison) was obtained without the rigour of presenting his DAC with the same sample rate. DACs use different filter implementations at different sample rates, and there are many choices available for implementing a filter for a 44.1KHz sample rate (as there are trade-offs involved as between upper frequency response and avoiding phase artifacts). It is my understanding that these choices are likely to be greater in their potential effect on the audible sound than the effect of an upsampling from 16/44.1 to 24/96.

QUOTE (rpp3po @ Aug 18 2009, 04:07) *
3. Generate your own Redbook master from Linn's HiRez WAV (VHQ SRC with intermediate phase low-pass, noise shaped dither, 1db gain reduction to prevent clipping):
Hi rpp3po, I tried sox with the intermediate phase setting, but when I subtracted the resulting file from the original (the 24/96 files less 1dB), there was quite noticeable (to my ears) higher frequency sound. So I ended up using Audacity, with dither set to noise-shaped. This gave a very good [for my ears] null.

I still have an open mind on the 96KHz vs 44.1KHz question, but the 24/96 tracks provided do not appear to be audibly different to versions of them downsampled to 16/44.1 and then upsampled back to 24/96.
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bandpass
post Aug 22 2009, 13:50
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QUOTE (MLXXX @ Aug 22 2009, 13:21) *
QUOTE (rpp3po @ Aug 18 2009, 04:07) *
3. Generate your own Redbook master from Linn's HiRez WAV (VHQ SRC with intermediate phase low-pass, noise shaped dither, 1db gain reduction to prevent clipping):
Hi rpp3po, I tried sox with the intermediate phase setting, but when I subtracted the resulting file from the original (the 24/96 files less 1dB), there was quite noticeable (to my ears) higher frequency sound.

In general, subtracting one audio signal from another is not very enlightening, esp. where perceptual techniques (e.g. intermediate phase filtering) have been used.
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rpp3po
post Aug 22 2009, 14:05
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Using Audacity should be fine, as the noise shaper and SRC implementations are very good. The applicability of auditioning a difference file to assess the perceptibility of an effect/conversion is debatable, as it doesn't take the effects of masking into account.

My choice of an intermediate phase filter adds more post-ringing noise than, for example, a linear phase filter, but should nevertheless be less audible (because more pre-ringing is eliminated). This applies to the signal itself, but not necessarily to the difference file.

But as said, your conversion method is fine, and well suitable to produce a conclusive results for your ears.

Edit: Bandpass was faster.

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evereux
post Aug 22 2009, 14:06
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After reading this post (#16) I would very much like to see JimC perform a level matched ABX test.

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Examiner
post Aug 22 2009, 15:17
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Are there even people who are able to do that? And if there are, what do you have to lock, or rather, listen for?
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rpp3po
post Aug 22 2009, 15:43
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QUOTE (evereux @ Aug 22 2009, 15:06) *
After reading this post (#16) I would very much like to see JimC perform a level matched ABX test.


JimC seems to be a marketing employee at Linn. From that perspective how something is perceived is more important than what actually is. So don't expect him to care too much. From many audiophile (and their own) forums he is probably used to people who really want to belief that 24/96 is superior. And he just feeds them, what they want to be fed with.

I wonder how he tapped into HA, inviting to evaluate the superiority of a HiRez track with roughly only 90 db dynamic range. He probably expected this thread to evolve differently.

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2Bdecided
post Aug 24 2009, 10:36
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QUOTE (rpp3po @ Aug 22 2009, 15:43) *
I wonder how he tapped into HA, inviting to evaluate the superiority of a HiRez track with roughly only 90 db dynamic range. He probably expected this thread to evolve differently.
Doubt it - either he doesn't know what HA is for at all and is just spamming every audio forum on the internet (that's a marketing person's job), or he does, but think some people here will still be interested in nice quality recordings, whatever the sample rate / bit depth.

I ordered the http://www.soundkeeperrecordings.com/ disc - the music was quite nice too, which is more than you can say for many "audiophile" recordings.

btw, I also stumbled on...

http://www.hdtracks.com/

...which has lots of 24/96 recordings - and (more importantly for me) lossless downloads of decent quality recordings. Not cheap though! I think I'm decidedly old fashioned in my love of physical media - a FLAC download is more convenient than a DVD-V with audio, but it doesn't feel like you're buying as much!

Cheers,
David.
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Arnold B. Kruege...
post Aug 24 2009, 12:43
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QUOTE (rpp3po @ Aug 17 2009, 14:07) *
Let me first say this: This 24/96 track from Linn is an excellent recording, certainly well worth a buy!


I trust you're just being nice, because the dynamics of this recording have been compressed with a pretty heavy hand.

I couldn't find any musically-related dynamics much in excess of 30 dB.

If you discount the obviously artificial fade-out, it has minimal actual musically-related dynamic range.

If someone told me that it was a 16/44 recording upsampled to 24/96 and backfilled above 16 Khz with with shaped l noise, I'd be hard put to disprove that hypothesis.
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Arnold B. Kruege...
post Aug 24 2009, 12:49
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QUOTE (2Bdecided @ Aug 24 2009, 05:36) *
btw, I also stumbled on...

http://www.hdtracks.com/

...which has lots of 24/96 recordings - and (more importantly for me) lossless downloads of decent quality recordings. Not cheap though! I think I'm decidedly old fashioned in my love of physical media - a FLAC download is more convenient than a DVD-V with audio, but it doesn't feel like you're buying as much!


The "Dragon Boats" sample is not nearly as compressed as the Linn sampler. It might even have natural dynamics or just light compression. Still no *serious* dynamics that would tax a 16 bit system, other than one more boring artificial fade-out.

Has no one heard of recording reverb tails as an exercise in demonstrating natural dynamic range?

I think they have, but they've learned that natural reverb gets lost in room noise long before it taxes 16 bit resolution.
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