XXHighEnd

For anyone who is interested:

I've recorded Dave Brubeck's 'Time Out' from vinyl at 16/44.1 and 24/176.4 resolutions. These are separate recordings (i.e. the first was not derived from the second).

At the time of writing, there are 9 downloads left, and they will expire on the 15th of August. You can get them here:

http://www.filemail.com/dl.aspx?id=XJRYNNLTBXQOBAJ

Of course, if you do download them, please share your thoughts on how they compare with each other. Is the hires version really better-sounding? Can upsampling make the 16/44.1 version sound better than the native hires? Etc...

Notes:
1) Please disable HDCD encoding in XX - although the 16/44.1 file contains the HDCD flag (actually, so does the 24/176.4!), I haven't used any of the HDCD processes.
[EDIT: This is actually not necessary, as the 16/44.1 file has been saved as a 24-bit file by the recording software. It will be treated by XX as a regular 16-bit file though, and will play back perfectly fine on a 16-bit DAC.]
2) I have NOT applied any sort of DSP on these files to remove noise or pops - you will get a sense of how 'clean' vinyl can sound...
3) This album was recorded in 1959!!! Which means that the copyright has expired (in Europe at least) - this file share is totally legal.

Cheers,
Mani.

Downloading... I'm a fan of 16/44 rather than upsampled material, but native 24/176.4 is another thing. I'll get back here...

G'day Mani,

I am unable to play back the 176K stuff currently, but I did take this opportunity to compare your 16/44.1 sample to my Columbia 20 bit redbook remaster from 1997 ( 7464-65122-2 )( CK 65122 ) and the first obvious difference was the diminished microdetail ( eg. air flutter ) in the reed from Desmond's sax. It was quite subdued in the vinyl rip. Although your vinyl rip was a lot cleaner than most needle drops I have listened too, it still had a noise floor well above the redbook copy, and that probably contributed to a little of the sparkle missing from the performance I am used to hearing from the album.

I went through an exercise a while back of transferring a number of my old reel to reel tapes to digital and I came to the realisation that whenever an intermediate copy is introduced, it will never be as good as if you used the original master tapes. Your vinyl rip just reinforces that point.

IMHO, needle drops are a worthwhile pursuit if you wish to casually listen to a favourite and often rare and unobtainable album in the car or when mobile, but will rarely equal a redbook CD derived from the same original masters.

BTW, if you enjoy Take Five as a tune in it's right, and I presume you do, Joe Morello released an album called Morello Standard Time in which he plays a very nice rendition. It's released on the DMP label.

Cheers,

Russ

Mani, XXHE indicates 24bit source file with your 16/44 track.

Hi Russ,

I have the same CD as you from 1997 (although mine seems to be 01-065122-10). I also have the SACD (7464-65122-6).

A few things:

1) the sax is a LOT hotter on the CD/SACD than my version of the vinyl
2) my version of the vinyl is the single-sided 45rpm, remastered by Bernie Grundman from the original master tapes
3) did you know that your CD is actually HDCD-encoded?

Will definitely look into Joe Morello's 'Morello Standard Time' - thanks for the reference.

Cheers,
Mani.

Oops, Mani ... True !

Back to the drawing board I guess.

I must say, I already noticed the files beging quite large, saw the minutes not corresponding to that, but never payed real attention to it.

Or was it on purpose ?

Edit : I guess not :

:)

You're right!

BUT... it's definitely a 16 bit recording. I've just verified this with RME's Digicheck.

I'm really puzzled why the 24/176.4 file is only 4 times the size of the 16/44.1 file, and not 6 times the size. This implies that the 16/44.1 really is 24 bits!

What's going on???

[EDIT: I attach screen prints.]

Mani.

Just saved as 24 bits files. :yes:

Ok, as far as I can tell this is totally harmless. Thus, for XXHighEnd it really won't matter, as long as only 16 bits are utilized (and Mani showed just that). The result for SQ would be 100% the same as if it were a 16 bit file.

Peter

Mani, could you explain why are you playng 16 bit track with 24, actually 32 bit setting? It's pointless IMHO and from my experience it sounds worse.

I don't really understand what you mean...

I use an RME AES-32 PCI interface to connect to my ADC/DAC. In XX, I have to set the DAC to '32-bit', or else it won't work. As for the recordings - I use RME's Digicheck software. It has just now transpired that this automatically creates 24-bit files, even with a 16-bit input.

In other words, there is NOTHING I can do!!! But I don't think the sound is suffering :)

Mani.

Of course not ...


Yea, maybe if your DAC isn't at its best with 24 bits. That can be the case. But I'm rather sure this is not what you mean.

If you think this is so, set your DAC Is to 16 bits, and see what all won't work anymore (this is indirectly what Mani is referring to, but he doesn't get what you don't get :swoon:).

I must honestly say, there's nothing much to explain. And/but with that, I can hardly imagine that you don't understand it ...
So, what's up ?

Peter

Oh, I see what you mean. For playing back 16/44.1 files, why in the 'DAC Settings' section don't I just choose 'DAC is 16 bits and 44.1KHz', right? Well, this means I can't use AP/PeakExtend or use the XX vol. control (in a lossless way) or play back hires files. Of course, if you don't use these anyway, then I guess you're OK.

Mani.

OK, but could you give me one advantage of playing 16bit track with DAC set at 32bits assuming that I don't use volume control, upsamling etc?

This is maybe stupid question, but I'm trying to understand what's going on. If the track is 16bit than other 16bits are just zeroes? What's good with that? For me it's just unnecessary processing for CPU.

Nothing much. But you're asking for the known answer. You know it was a little mistake (by Mani).


I can't. So, set your DAC Is to 16.
With a little luck it also plays the 24/176.4.

Dear Manisandher,

All 10 was gone before I got a chance to download it.  :cry:
Can you pls make it available one more time? I would LOVE to hear a quality rip of the Take Five LP in 24/176.

Which turntable/playback did you use?

Best regards

Done. You can download them here: http://www.filemail.com/dl.aspx?id=ZPOOCCHMGRMLXGT

But, I wouldn't get too excited - I used a pretty modest vinyl setup:

- Technics SP-10 MkII turntable on a solid metal plinth (wall-mounted on a solid brick wall)
- SME 3009 S2 (pre-improved) arm
- Denon 103 cartridge
- Cardas armtube cable (single run from cartridge to phono stage in balanced configuration, terminated with XLR plugs)
- AQVOX phono stage (balanced MC input, balanced output)

(For those of you who are interested, the cartridge resonance occurs below 10Hz with this setup.)

On the digital side:

- Pacific Microsonics Model Two
- AES/EBU cable (for signal) and BNC-terminated coaxial cable (for wordclock)
- RME AES-32 PCI interface
- RME Digicheck 'Global Record' software

(No analogue or digital gain/attenuation was used whatsoever in the AQVOX, the PMII or the RME.)

Cheers,
Mani

Thanks!

I allways enjoy listening to vinyl. I have heard this track several times on my friend Leif's super turntable and the big stomp in the kickdrum is hard to get from the CD. It is probably certainly a vinyl artifact (resonances etc), but never the less the drumkit SOUNDS "better" than I have heard from my CD.

I will revert later on!

OK. 24/176 track is much better from the first seconds. It strikes you with wider soundstage with 'lot of air'. The sound is fuller, especially in the mid-range. (sax)

For those of you who downloaded these files, I'd really love to hear your thoughts on the sound of the 16/44.1 vs. 24/176.4 file.

Cheers,
Mani.

Yes, I agree on your points. And also there is more texture to the instruments and it sounds more resolved. The 16/44 track sounds canned ("CD") in comparison, more dry and boring to listen to. (This is my impression after a quick A-B today).

Afterwards, I had to try my CD rip. It sounds cleaner, but I miss the extra low end kick which the vinyl has. The CD bass is tighter (and probably more faithful to the mastertape) but never the less I prefer the vinyl bass on this track. (Which often is the case with recordings from 50s and 60s).

It was a nice vinyl rip, although hearing the same LP at "leif" (he is a forum member, too) is even more impressive, through his $$$ vinyl rig.

Thanks Marcin, thanks pedal for your views.

I'm still looking for others' experiences before I share my own...

Cheers,
Mani.

So am I ...

But I already feel the need to say that might it be so that people thought "hey nice, let's download some hires" ... there are other sites for that. So if that is the case, notice you took away the download facility for others with more real intentions and I already feel the urge to track down who the other gang is. So if you see your name popup outside your own posts, you know what the origin is.

But not yet !  :) :)

Hello,
First of all I have to admit that my intention when I downloaded the archives was only to make a comparison between them and my own vinyl recordings just to confirm how good all of them are for me.
 I don´t make vinyl recordings very often because is a tedious job,  the record has to be in good conditions and then you have to cut the archive and generate a cue. My opinions at this respect are not relevant because I have not the so called "golden ears". I have a syntetic ears instead an analytical ones.  I like it or I don´t. I am so simple. Furthermore my Dac doesn't admit 176,4 so I can't compare to 16/44.1
Well, said that I must confess I feel sorry about the inconvenience caused to the forum and to Mani in particular whose question was very clear. But honestly I didn´t expect the number of downloads were so conclusive.
To amend this I have the intention of make a recording at 16/44,1 ( as I use to do ) and at 24/96 (maximun admited on my DAC) but as I have explained above, don't expect very clear conclusions.

Maxi

Ok, ok, if you start to threat...

I listened to the 44 first and thought, well it sounds good. I don't expect the hires to sound much better, and got a bit bored of the one-note hihat, one-note sax and one-note drum.
But than the 174 started off and sounded almost completely different.
The first thing you hear is the acoustic off the recording place, it sounded warmer, less upfront and not one-note anymore.
Not boring at all anymore and now I can understand why you liked to record this song.

But I used no oversampling/ QAP on 44 so maybe that changes the picture?

Edit: Just tried 44 QAP and I think now it is more detailed, bass drum more clear, more space but not the aha-feeling.

If I understand correctly, 16/44 version was really recorded like that?

If so, IMHO this is not a fair comparison: it would be much more interesting if 16/44 was derived from 'master' 24/176. To save you the trouble I did just that - get it here:
http://www.filemail.com/dl.aspx?id=ADMNTWIBWBZXRRS

As I don't have hardware that can play 24/176 I'd like all to please compare and share your thoughts again!

Josef, the comparison wasn't meant to be just between the native 16/44.1 and the native 24/176.4. I was hoping that people would try comparing an AP-upsampled 16/44.1 to the native 24/176.4. I know of one person on this forum who actually prefers the former to the latter.

As for downsampling/decimating the 24/176.4 to 16/44.1 and comparing it to the native 16/44.1, well this is exactly what my ADC does internally when recording at 16/44.1 - internally it works at 24/176.4 (or 24/192) and downsamples/decimates to 16/44.1. So here you will really be testing whether your decimation DSP is better than the decimation DSP in my ADC. Still, an interesting test (for me at least) which I will do.

Which program did you use for the downsampling/decimation?

Cheers,
Mani.

I didn't use the ArcP eigther, only straight 16/44. (Actually ArchP doesnt work on the XX version I have been using the last month. I havnt updated to the current version, because I have only played hi-rez files lately - so no need/hurry). I have some time tomorrow to do an software update and re-listen to the 16/44 with 4xArc!

It is not all that easy to make apples and apples from this;
AP is supposed to mimic the original higher res file, but it will mangle with the upsampling algorithms in the DAC. The latter is already doubtful (for me) because once upsampled already, the DAC('s algorithms) may leave it alone. That is at least what I expect for some DACs (but can't be prooved by me).
The other way around, if you do nothing (no AP), it's for sure left to the DAC what it makes of it, and I don't think it can be anything good. But then there are also the NOS DACs which do *nothing* with it, and now it will be wrong because of too much harmonic distortion.

Notice that under the hood this test is about what may go wrong at recording 176.4 files (what may be applied to it which mangles in the first place) vs. what is *not* applied to it when recorded at 16/44.1. But then of course ...


... such a thing shouldn't be happening. I must honestly say though that I don't know whether it will be normal for any digital recording device to approach it like this. I mean, it can just as well work so that during the recording the sampler runs at 44.1, and thus really nothing is done to the recorded samples further. I wouldn't even know how to check for that, otherwise than connecting the scope to ... yeah, what ? I know about the "output" lines (at playback), but input lines ? I may not even have ever heard of it ! So FYI :
At playback there's a bit clock and word clock running, and they change per sample rate used at playback (both derived from the physical clock which always runs at the same speed). But at recording ? as said, I wouldn't even know where to look for. Sure not those same "i2s" lines ...

Peter

I wrote this in another thread a while ago:


Mani

I hope this stays on-topic, but this would proove what I am saying all the time ? (maybe not in here, but for others : I do).

Why in the world would this be necessary, when during playback this is also done ?

Also interesting (for me) : what would be "dynamic decimation" ? This, about a possibly clue that 16/44.1 sounds less mangled with (this is a very indirect derival of 8xAP possibly outbettering "native" 176.4).

Peter


PS: I too will tell about my findings later, hoping for some others to be first.

Wow we created some controversy here, great fun :)

Like Peter, I am a bit skeptical about how ADC is treating 16/44 - if I wanted to be really, really nasty (no hard feelings please:) I might surmise that 'dynamic decimation' is a marketing term for 'just chop 3 out of 4 samples' :)

But seriously: Version I posted (let's call it 'DND' - see below) is simply a more realistic scenario because software is more likely to be better at what it does (because it's easier to experiment & update) and also likely to _do more_ than hardware - Specifically, version I posted has been processed minimally (& maximally at same time :)) like this:

1. Downsampling from 176 to 44: There are many complex algorithms here i.e. this is much more complicated than simply keeping every 4th sample (not saying your ADC does that - just illustrating the point of complexity and likelihood that sw will beat hw every time)
2. Noise Shaping - this is theoretically not needed but in my experience I found it to be essential when 'downsizing' to 44/16. Again, there is a multitude of algorithms to choose from...
3. Dithering - i.e. cutting 24 to 16 bits. This one was to me always more impressive (even magical!) than first 2 as, in theory, it affects only the last 16th bit (hey, how important can it be whether it's 0 or 1??)  and yet dithering method can impact a definitive sound signature - And of course, there are a zillion algorithms to choose...

Point being - to get _good_ 16/44 you need to choose from many, many different algorithms which is easy (although time-consuming) to do with software but often impossible to do with hardware alone, and results are bound to be waaay betetr with software approach.

Anyway - who cares about all this technical mumbo jumbo stuff - let's discuss how this sounds to us!

24/176 does not work on my hw but I am fairly certain that 16-bit 'DND' version is superior to 'raw' 16/44.

It is hard to put it in words but while listening I remembered an occasion when I was experimenting with manipulating WAVs I was loading to my iPod. I was using Foobar and turned off both Noise Shaping & Dithering controls (no changes were being made to music apart from ReplayGain).

Initially, I was surprised how much more immediate everything sounded - vocals were 'in my head' and music was very much 'in your face'. After listening for longer time however, I realized that effect of no NS & D was, in a way, similar to comparing 'flat' & 'heavily compressed' track. Compressed track always sounds 'better' because there is 'more there' until one gets a headache and starts seeing square waves in sleep :)

'Raw' 16/44 sounds eerily similar: It's like a rough diamond - all information, bits (pun intended :) & pieces, atoms & molecules, are there but it's not cut yet and certainly not polished - I assume 24/176 will sound similar: more flow, more ease, more polish - simply more 'music' as opposed to 'sounds'.

But maybe I'm mistaken: would people with capable hardware care to share their thoughts of both 'native' 24/176, 'raw' & 'DND' 16-bit and, especially interesting for us fans of XX, AP upsampled 16-bits?

PS.
>Which program did you use for the downsampling/decimation?

Mani, I'm not sure it's OK to talk about other software on this forum so I'll send you an email!

Firstly, let me ask a favour of you - please take what comes below in the spirit in which it’s meant. I’m not here to convince you of anything. I’m just here to share my experiences that you may or may not find useful or interesting.

18 months ago, I bought a couple of ‘HRx’ 24/176.4 WAV albums (burned on DVD-Rs) from Reference Recordings and was really taken aback by their SQ. There was a ‘quality’ that they possessed that I hadn’t heard before from any of my ripped or downloaded 16/44.1 files. This quality is hard to put into words. Extra low-level information? Smoother? More dynamic? I’m not sure that any of these manage to describe the particular quality that I heard. The only really accurate description that I can offer is ‘life’ – the music ebbs and flows as if it’s alive. (But please don’t mistake this description for ‘life-like’ – you would never mistake these files being played back on my system for the real thing. A better analogy would be with a painting – great paintings have so much ‘life’ in them, but they’re not necessarily ‘life-like’ at all. Do you understand what I mean?)

And this got me thinking about 16/44.1 vs. 24/176.4. Are the former inherently inferior to the latter in an audible way? If so, is there anything that can be done about this?

I wanted to be as objective as possible. Of course, it’s impossible to be totally objective, but I wanted to eliminate as many of my own biases as possible. The best way seemed to be to get an analogue source, record it at the two different resolutions and compare the recordings with the original analogue source. An assumption that I’m making here is the absolute quality of the playback system (preamp, amp, speakers, cables, etc) becomes irrelevant because it is the same system used in all cases. Of course, this is only valid if the system is resolving enough to allow any differences to heard – and I believe it is.

Method

Before making the comparison between the vinyl, native-24/176.4 and native-16/44.1 files, I wanted to ensure that my digital chain was totally transparent.
I tested the following two routes:
1. phono stage -> AD -> DA (i.e. internal)
2. phono stage -> AD -> AES cable -> RME PCI card -> RME mixer -> AES cable -> DA
Irrespective of sample rate, I found that there is absolutely NO difference in sound between these two routes. (Incidentally for 2, I knew that I really was passing through the RME mixer because I could hear any changes I made to levels, muting, etc in the RME mixer.) From this, I concluded that the AES cables, the RME PCI card, the RME mixer and the RME driver are all transparent enough for the purposes of this assessment.

Objective Assessment

(Yes once again, I know that this cannot be truly objective if I am making the assessment...) I played the vinyl and was looking for the digital file that sounded closest to it. IN COMPARISON TO THE VINYL, here is my ranking:
1) native-24/176.4
- sounds identical
- cannot distinguish in double blind listening tests
2) (XX) QAI of native-16/44.1 = (PM) 1:4 Interpolation of native-16/44.1 = (Software) 4:1 Decimation of native-24/176.4
- tonal balance maintained
- less depth
- rounded transients
- less low-level detail
3) (XX) QAP of native-16/44.1
- sharper
- leading edges (over?) emphasised
- cymbals sound thicker and much less delicate – the initial strike is more emphasised but the shimmer/decay is attenuated
- less body and weight to instruments – they sound smaller and ‘cheaper’
4) native-16/44.1
- brighter
- more edgy
- more forward

I suspect that most people would come to a similar conclusion here. I’ve played the vinyl to a number of (non-audiophile) people and then these digital files. Asking them which digital file sounds most similar to the vinyl, I get very similar results to above, although they often find it difficult to articulate exactly why they think what they do.

Subjective Assessment

Here, I consider which digital ‘sound’ I most like and can most happily live with. My ranking is:
1) native-24/176.4
- I don’t long for anything else, this has it all
- it is detailed and yet totally smooth and easy on the ear too
- it has the all-important ‘life’ quality
2) native-16/44.1
- this has ‘life’
- it breathes
- although it is brighter and more edgy than the native-24/176.4, I can live with this
- it doesn’t annoy (too much)
- good boogie factor
3) (XX) QAP of native-16/44.1
- the ‘life’ of the native-16/44.1 is maintained
- sounds very impressive on first playing
- tight as hell
- all instruments/voices are absolutely delineated and focused
- phenomenal boogie factor
- but gets fatiguing after a while
4) (XX) QAI of native-16/44.1 = (PM) 1:4 Interpolation of native-16/44.1 = (Software) 4:1 Decimation of native-24/176.4
- very smooth and easy to listen to
- all the ‘life’ has been sucked out of the music
- great for innocuous background music

Conclusion

No, I couldn’t get 16/44.1 to sound as good as the vinyl source or 24/176.4. But I can get it to sound ‘good enough’ to listen to, and more importantly, to ENJOY. Playing the native-16/44.1 file hits this spot. Using QAP with my setup doesn’t. But I’m holding out for OAP on a NOS1... which I'm hoping will do it.

Cheers.
Mani.

Hey Gerard,

I suppose this is nitpicking, but I never heard a hihat in there ! What's used there is a ride cymbal which really sounds different. Just for fun : http://en.wikipedia.org/wiki/Ride_cymbal

But let's say this won't bother you anymore in near future. Haha.

Peter

This is going to be complicated ...

Let me first paste in what I wrote to Mani earlier via email (I had the files somewhat earlier than they were posted here) :

---

First listened to the 44.1. It sounded just gorgious to me. I tried to imagine what the 176 could bring more. Listened through the whole track ... no complaints.

Next the 176. Hmm ... Right from the start I seemed (!) to notice more air. Later (listening a second time) the same perception. But, somehow I can't listen this one through. It doesn't fit. The piano at the right (assumed I have my L/R cables right) wouldn't be the leading "rythm", while I think it is/should. Later (maybe the 4th time) I perceived it as a stupid cut "ploink" thing. Besides, the sax (is it a sax ?) clearly annoyes with the 176.

---

Notice : I was just comparing again (Josef's decimated version incorporated) and after listening several times to pieces of the 44.1 versions, I compared with the 176.4, and the exact same as before : the piano doesn't want to work.

For me the 44.1 version(s) win (8x AP), although it is hard to find more reasons than the piano. I have some allright, but I'm not sure what they tell :

Firstly I got the focus somehow on the sax after 56 seconds (up to 1:20 or so) in the track. It seems that it contains a distortion in the lower regions in Josef's version. I was comparing this to Mani's 44.1 version, and thought that one sounded more natural.

When I lastly compared it to the 176.4 "original", I found the same "distortion" to be in there. So, was Josef's version a better copy ?

In order to get better merits of this all, I listened to 2:50 up to 3:20. This is right after the drums start to be soloish. I think it might be important that I listened to this part first with the 176.4 version. All sounded very clean and as should. And, this is exactly why I wanted to hear that very part in the 44.1 versions. Here's is the strange part :
Both 44.1 versions contained an echoish sound on the drums. A sound of which "you" most probably would say it doesn't belong there. If you'd hear what I mean, you would be convinced that it must be something strange, because the lower tom hits this is about, don't contain any higher frequency "noises" you perceive here (and which were not there in the 176.4). And, if I hadn't been recoding my own drum set the other day I would never have heard it ... this is the sound of the drum kit itself. Partly it is the snare of the snare drum, but if you were next to such a kit and tick against the poles and sides of drums and anything, you'd have a metalish sound just of "loose" things.

From that moment on I thought that the distortion in the lower regions of the sax just are details which the Mani-44.1 coudldn't bring. However, they are there in the 176.4 version as well (as in Josef's 44.1 version).

Conclusions ? none. Or none other than that everything makes a difference.
For me the 176.4 version shows more air, but nothing which makes the happening better. The contrary.

Keep in mind the context in which this is to be interpreted :

1. What exactly happens to the resulting file when the 176.4 is recorded;
2. What will exactly happen LESS to the resulting file when the 44.1 is recorded (which internally still is a 176.4 according to Mani's description);
3. What will exactly happen otherwise when the 176.4 is decimated manually (which can go per a 1000 means as per Josef's description).

Ad 1.
It is quite important to know that very many of the CDs we listen to, are recorded by the same device Mani used to take this recording. And, what at least I am trying to, is how to benefit (with playback means) from the knowledge of what really happened to our CDs when they were produced.

Peter


PS: In the mean time I have another idea to look at this all. Give me a few minutes ... :)

Haha, yes thinking of it.. Never to old to learn/think!
So much for the rest of my remarks!
One-note is not true too, and is it just an ordinary sax?

One way it is interesting to find out what instruments you can hear, maybe a new hobby..

Ok, I wanted to do this more extensively, but it really takes too much time. Anyway, below (also see next posts) you see a few pictures.

The first one here shows Josef's 44.1 take at the top, and Mani's at the bottom;

To me it is clear that Josef's version is better. It shows more detail, and it better follows the original;
Hard to explain by means of just this one picture, but what I see, I see back everywhere. Look at the bottom channel of the bottom picture (Mani's version) for an example. You there see two situations (one at 1/3 of the screen, and one at 1/2) where two adjacent samples have no difference in amplitude. With this in mind, already in this one picture you can see that the amplitude (volume) steps generally are smaller in Mani's version.

Notice : It is not said at all that larger steps are better, but it is about following the original better (see next posts).

What we see here is Mani's version compared with the 176.4 "original". The original has 4 steps sideways where the 44.1 has 1.

Look at the top channel just before the middle; The original shows the tendency of the downgoing slope being higher in amplitude than the upgoing slope preceeding it; You can see that the samples at the other side (the right side) of the slope are higher in amplitude than at the left side. However, Mani's version shows it the other way around, and a downwards going tendency.
If you look more to the right, you see similar happening at the top channel (Mani's version) where the wave disappears under the horizontal line. It shows an upwards tendency, while the original clearly shows a donwards tendency (compare with the left side of that wave).

Try to see more of this yourself ...

Here we see Josef's version against the "original". Read the text from the previous post again, and see that here things are as expected.
Well, as how I see it. :)

Also look at the second peak from the left, and what happens in front of it. Look at the down - plus upgoing small slope and compare it with the original. Yes, you can sense Mani's version (previous post) as better matching, but in the mean time that shows nothing of a smooth going up again, where this one does. But, doubtful in this case.

But, just look, and "feel" that this is al a better representation;
I have done it for several of these pictures, and they all give me this same feeling.

Lastly a more zoomed out comparison between Josef's (top) and Mani's (bottom).

Without seeing the original (which I couldn't locate in 30 minutes for this position, so I gave up), again you can sense the better representation of the top one. If you look at the 2s138.6ms position you see a downgoing slope with no sense in the bottom picture. No guarantee when looking at this only, but seeing more of these picture really makes me say this.

Here too, you can clearly see (if not dizzy by now) that many of the slopes go just the other way around. So, one of them must be wrong, and that this is Mani's is clear to me, but also follows from the few examples in the before posts.

But :
... see last post about this ...

Haha... this is what they meant by 'dynamic decimation' :)

Looking forward to your next post before I respond...

Mani.

... it is not said that the one "decimator" is better than the other. I didn't look for it (takes too much time), but what Josef should have done, and probably has not, is starting his "recording" at the very same sample Mani did with his 44.1 take. Now, already that is impossible, because Mani took two recordings, and the sampler with its discrete fixed sample rate will never have started at the same (analogue) wave position as the previous time. But would that have happened by pure coincidence (with a change of 1:174000), then Josef should have started his recording at a common denominator sample of the 44.1 version of Mani's. When you don't do that, all is shifted a little, and where Mani's 44.1 virtual sampler would have started at the first sample, Josef's will have started om e.g. the 2nd, and all will be totally different because of it.
Look again at one of the 44.1 versions against the 176.4 version, and you'll get what I mean. Just look one 176400 sample more sideways, and up going slopes will be downgoing slopes.

Is this all too stupidly looking at details ? No, I don't think so. I hear a difference between the two versions (Mani, Josef), and the files *are* different. But also I heard similarities between 176400 and Josef's take (see earlier post), which just as well is explained now.


Allright. Don't try to understand this. But if we are spending our time on subjects like this anyway, we must do it well. I don't say I just did, but it really comes down to these kind of details;
If files are different they should sound different. If they do not, better wonder what can be outbettered on your system.

And oh, nothing tells me that Josef's file should sound different from the 176.4 original. This depends on many other things (like the analogue workout and how *that* compares). But all sure tells me that Mani's 44.1 MUST sound different.

Peter

WOW! Great analysis Peter. Thanks.


I agree with this 100%. What I call the native-16/44.1 sounds the most different from the vinyl. And the native-24/176.4 sounds the most similar to the vinyl (actually identical, as far as I can tell on my system). Josef's 16/44.1 definitely sounds more similar to the vinyl than my 16/44.1.

Mani.

BUT... I prefer the sound of my 16/44.1...

EDIT: Pacific Microsonics strongly recommends using the 1:4 Interpolation in the PM2 for 16/44.1, but I don't like the way that sounds either.

Peter, as Josef's 16/44.1 has been derived directly from the original 24/176.4, would it be possible to create another graph comparing QAP applied to his 16/44.1 vs. the original 24/176.4? This would give us an excellent idea of how accurately AP can reconstruct a hires file.

Mani.

Yes, that's the difficult part of it all. I too like your version better, though technically (listening) the 176.4 sounds better. So, this even goes further.

Maybe it is more important to recognize that either version sounds very good to begin with, and that somehow we are (suddenly, as found out) very close to letting whatever it was sound very close to the original - by means of digital recording and playback.

Assuming that these details already matter (and why not), I could for fun make an AI version of your 176.4 and show that. But I already know that won't work, because no common points can be found then. It will be totally different (I already tried that once).

So, in the far end it comes down to what happens beyond what we just looked at, and this will be our DACs. I can reason out what it will be with my DAC (namely, nothing as far as it concenrns the digital part), but e.g. yours ?
But careful, because "nothing" is not necessarily good, and all now depends on how good Arc Prediction is (of which I can *not* make a file without programming something for it (opposed to AI this happens in memory, and not as a pre-pre process which goes to disk first)).

But ok, since I made that 16/44.1 recording of a live drum session (people in here may not know it) which can't be distinguished from real at playing it back, I'm quite confident.

Now Mani, when are you in the neighbourhood again ? It sure looks like we can have some interesting times;
I'm not obsessed with this (you are I think, haha), but I really think if we can go down to the bottom of this al, and learn how to avoid mistakes, playback can be better again. So yes, that would come down to playback anticipating on recording devices, but since it can be done in software (and the NOS1), why not !
Better plan more than an ordinary one night. :)

Peter

Well, work has been very quiet over the summer, but will kick off in earnest in about 2 weeks time. I've just been trying to get as much done as possible around the house and on the hifi before I start living on a plane and in hotels again. My dream is to come back home after a work trip, spend time with my family and then relax in front of the hifi, just listening to music... without obsessing over how it sounds!

One thing I've been thinking about though. Your recordings - you're using the FF800, right? This uses delta-sigma ADC chips. Now, I have no idea how these things work on the ADC side, but they must be OK. I mean, they can't be mangling up the sound, because you're getting such good results from your live recordings. So, would there be any advantage in using a multi-bit ADC chip, like the one in the PM2?

Mani.

I really have no idea. I think I can only know when I really measured. One way or the other. Also see next posts ...

Ok. With my own statement in mind "an AI upsampled file can't be compared", I thought "but wait, this never was an AI upsampled file ... this was the result of an OS DAC with something "like that" in it. So, let's see ...

Was I wrong on thinking AI couldn't be compared ...
What you see here is just that; At the top is the original 176.4, and below it you see the AI upsampled file, with Josef's 16/44.1 as the base.
Remember, the 16/44.1 was the decimated version from the same 176.4 you see here at the top ...

Yes, in the top channel (at 1/3, but notice the scrollbar at the bottom !) you see an anomaly, the AI version showing a bump, the original just showing a slightly upwards going slope.
Notice that I can not know whether this is cause by my AI upsampling, or whether it is caused by the decimation process from Josef. And no, I am not going to find this position in his file. :)
But these anomalies are very very rare; I have been browsing through this step by step for 45 minutes, and found only two of them (see later for the other one). Can this change sound ? maybe, if only enough are there. My 45 minutes of browsing was worth 6 seconds of music ... (I started at 1 minute, going backwards).

Also see next posts for some clues ... :yes:

A little zoomed out. What we see here, I only saw after a long time as a pattern ... what do you see ?

Look at each of the peaks and compare the both takes ...

Do you see it ?

Somehow there is a consistent "opposite" balance in how the peaks are formed. Just find a peak where one sample bumps out. It doesn't matter where you find it. Now look at its counterpart. See ? the one with one sample jumping out has 3 samples to *let* that sample jump out, where it counterpart has an even number of samples for the width concerned, and it is impossible for one sample to jump out.

If you try to reason where the "base" of this emerges, at least it is hard to see, but it must be at the base of the wave somewhere, where the one is wider as the other.

This is a most consistent thing throughout the both files !

Will this change sound ? ha, you bet it will.

Two other examples, and nothing new. The first one contains an anomaly again (at 2/3, but notice the scrollbar at the bottom !), and further you again see what I talked about in the before post.

Notice that it is not in 100% of cases like the rightmost two pointing down ones (the second one is just visible in the upper right corner of the bottom half (screenshot) of the first picture).

But generally this upsampling means is very very much alike the original. What ? it is 1000 times better than I ever expected.
I should even listen to it ! haha

Peter

EDIT, PS: I'm not sure whether it can be the cause, but it looks like the upsampled version is one sample shifted opposed to the original (like skipping the first sample). In the end I don't think this is true, but if you look at the first post of this last sequence, you can see that the bottom part is not time alined with the top part. Still it is ... So, if the upsampler shifts one sample, then you'd have something like this. I think.

EDIT2, PPS: I just saw that the 2nd picture in this post, at the very right (use the scrollbar for that) also contains an anomaly. I didn't see it at first, so there may be more than I have noticed.

QAI applied to my native 16/44.1 came 2nd in my 'Objective Assessment', after the native 24/176.4. I'm going to give QAP and QAI applied to Josef's 16/44.1 file a good listen... might be interesting.

Mani.

If you could repeat QAI with 'my' 16/44.1, you could see if these bumps are there also. If so, it's likely to be AI doing this, no?

EDIT: No, of course you can't - they're not aligned, and you're not going to do that, right?

Mani.

Apart from the odd anomaly, and the strange behaviour at the peaks, these graphs look remarkably alike, don't they? Yeah, 1000 times better than one would expect from AI!

Now, is there a way to speed the processing of AI up???

Mani.

If I had to look for that same spot ? undoable. This merely has to go like seeing a recognizeable point (a strange shape when zoomed out) and find that in both files by slowly zooming more and more. Can take 30 minutes for such a recognizeable spot. But without such a spot (zoomed out) ? even if you had a day ...


If there was that I could think of, I had applied it.
Yea, applying a more simple filter. Haha.

Peter

Try this:
http://www.filemail.com/confirmation.aspx?id=HAHCNWRYCNBXSTE

I am curious if you find this version more similar to your 'native 16/44'. (it is also derived from your 24/176 master but in a different way)

If you do find it 'more similar' then we might have some interesting food for thought :)

Yes, definitely 'more similar'... but still different...

Firstly, I like the sound of the your new 16/44.1 - it's got 'life'. I listened to it all the way through first off and really enjoyed it. However, it's not as close as your original 16/44.1 is to the 24/176.4. The main difference seems to be in a 'thinning' of the sound. Everything seems to sound clearer, but at the expense of body to the instruments. In this respect, your new 16/44.1 is remarkably similar to the effect that QAP has on my native-16/44.1.

If I had to choose only one 16/44.1 file, it would be either your new 16/44.1 or my native-16/44.1. BUT... not because they sound like the 24/176.4 - they don't. But rather because they have 'life' and are just so enjoyable to listen to. My one caveat with your new 16/44.1 is that on prolonged listening, it might get a bit fatiguing (again, like QAP on my native-16/44.1).

So please, let us in on it - what did you change? I'm intrigued...

Mani.

Maybe it is a little off topic, but after the "ages ago" I created the AI upsampling, I now judged it again.

Oh boy ...

I remember that after I created it I used it for two months or so, just because I was very satisfied with it. But did time proceed ...

AI TO_TAL_LY destroys the sound. It just makes complete rubbish of it. But, nothing much different from the Weiss Minerva I listened to the other day. So please keep in mind - it is just what one is used to (like me liking AI before, which was on the same NOS/Filterless DAC, though in a much much lesser (SQ) version of it).

But maybe you can squeeze out the same as I did;

Maybe you have Avishai Cohen - AT Home. Well, you should get it if you don't have it anyway. :)
I listened to track 04-Remembering;
This is a rather sensitive track, which first of all now doesn't "work" at all. But besides that all the piano notes seem cut, and there's nothing like a real bass around. But more interestingly, while the guy plays the bass in a to me strange fashion - like plucking two strings while the same note (frequency) comes out (can that be done with one very large left hand ?) - this interacts in AI in a way that additional harmonics emerge which make the man play totally off-key. It sounds interesting, but it shouldn't be there (later I checked with AP and there it's just two the same notes).

I played some ambient piece you won't have anyway, and I already learned from the Minerva that when things become the most fragile, it will bring the most distortion. And indeed it did. Nicely sweeping sounds just become shattering noise. Yes, I guess it is hard to believe, but it just is so. Funnily enough, when you are not used to how it should sound, you could accept it as normal, but then to the sense of "some lousy band playing a lousy synthesizer lousily".

I played "The Weight" from Chris Barber. It sounded the most normal of them all, but I noticed it was very loud. Actually so loud that the trumpets shattered a few tads too much to my likings. Later, with AP everything appeared to be just normal.

So ...
AI may then be able to follow the waves in an unexpected good manner, it is and remains a filter which should ring as hell. Why I can't find this in the data I don't know, but most probably that will be a matter of just not being able to see it in those kazillion samples. And now I think of it, I just heard a toing-toing-toing echo in one of the tracks with AI of which I'm sure I never heard it. But I forgot to check it back with AP, so not sure about that. What I do now is that the shattering distortion I talked about, *always* is something which repeats/smears. So, when there would be an original sound of say 1 second, when it is turned into a shattering noise this will be 3 seconds.

Am I glad I did this. Because, as it now turns out "I" am lightyears further than I knew myself. It just needs recalling the time I used AI for two months. How can I ever have done *that* ... It is unlistenable.

Spoiled me.

>So please, let us in on it - what did you change? I'm intrigued...

OK, here it comes but please be forewarned: you may not like what you'll hear so no shooting the messenger, OK? :)

Idea was this: I suspected that 'dynamic decimation' _was_ indeed a marketing term to just keep 1 out of 4 samples or some very, very simple (=poor quality) downsampling.
More importantly, I suspected that ADC also did some limited/poor quality dithering or no dithering at all i.e. it just chopped last 8 of 24 bits.

So if this theory was correct, by taking your master and doing the same thing to it we should be getting close to the sound of your native 16/44 version!

Now, I don't have software that can do _exactly_ that - perhaps somebody else does - but I was convinced if I just did something _similar_ we should be getting closer to your native 16/44 sound.
  
So, for step 1. what I did is to take the worst resampler I had at hand: Foobar's PPHS. PPHS still does much more than simply dropping 3/4 samples but is clearly inferior in quality to professional algorithm used for previous version. This is important as downsampling creates all kinds of side-effects and lower-quality algorithms create more of that (think it's called aliasing or quantization noise or somesuch :).  BTW - I could not get as close to your native 16/44 sound by using high-quality resampler!

Then, I simply skipped steps 2. & 3. from first version (Noise Shaping & Dithering) and just 'cut' the bits to first 16. Note that this will add _even more_ quantization noise to result!

While this step also is not exactly the same as taking 24 bits and discarding last 8 it is close enough: strictly technically speaking when PPHS dowsampled from 176 to 44.1 it created 64-bit floats and then those 64-bit floats were just cut by Foobar to 16-bit integer in the simplest manner = without any processing.

So what does all this mean? Here's one explanation I believe is at least close to truth:

Initially, I agreed with your assesment of 'native 16/44': it  _seemed_ to have more 'buggy factor' or 'life' as you put it. But when I compared it with my manually downsampled version (note that some amazing software was used as I explained in email!) I suddenly found it too 'rough'.
Then I remembered where I heard that pattern before: as mentioned in my post, some time ago when I was processing files for Ipod (=applying Replay Gain when converting to WAVs), I turned off Noise Shaping & Dithering to experiment (you can try the same!)
And the track I used also had a leading piano :) And it also initially sounded like it had 'more life' and I thought I heard the piano screeching & making all kinds of funny noises which I thought uncovered more detail & realism that previously was masked! So I converted bunch of music this way!
Long story short, after some time (and it wasn't minutes, it was probably days - I'm getting old, lol:)) I realized that I was getting fatigued after relatively short listening sessions - just like you found out in your latest post!

So, there wasn't more 'realism' or more 'life' there - it was just more quantization noise that fooled me back then and, if I may respectfully suspect, you too this time:)

Cheers,
Josef

Hahaha ... I'm not sure what the conclusion is now ... but listen to both your versions Josef, and Mani's (the 44.1) ... about quantization noise ...

A few minutes ago I already typed "but the quantization noise should be audible ?" ... and then I thought to listen for it. Josef, your both versions are full of it. Mani's has NONE.

You can hear it right in the beginning, when the needle is still in the lead-in.

Sorry ... (but maybe it was intentional ? ... I can't be sure readig your last post)

(and besides, Mani's version keeps on sounding the best to me)

Peter

No, no chance of this. I'm always happy to learn...


There is one thing that I'm absolutely certain of - the Pacific Microsonics Model Two does NOT have poor quality downsampling capabilities. I know of at least one highly respected mastering engineer who prefers its downsampler to ANY software downsampler - and he's tried them all, including the one you use Josef.


This is not the case. The Model Two provides a number of 16-bit dither options, which I understand are pretty sophisticated. You may or may not be aware that all HDCD-encoded CDs are mastered on the Model One/Two and have all been decimated down from its internal resolution of 24/176.4 or 24/192. In any event, I haven't come across an HDCD-encoded CD that doesn't sound anything other than superb.


Maybe. But remember that my strong preference is for the analogue vinyl and the native-24/176.4 file, neither of which has this issue.

So, how is it that I can live with my native-16/44.1 and your new 16/44.1, even though they clearly sound very different from the vinyl and the native-24/176.4? (And by the way, don't get the impression that they sound the same as each other, as they really don't.)

If I had to hazard a guess, I'd say that I generally prefer as little DSP as possible. The more there is, the more the 'life' seems to get sucked out of the music. For example, I can't listen to your original 16/44.1 - I simply switch off.

I'm not sure where this leaves us really. But I'll just reiterate that I firmly believe that 16/44.1, although not perfect, can be made to sound really enjoyable.

Mani.

I've just tried it again, and agree with you (and with my initial 'Subjective Assessment') - AI kills the sound. (It also takes way too long to load.)

Mani.

To me too... not as good as the vinyl or the 24/176.4... but very engaging and enjoyable, with zero fatigue.

Mani.

>A few minutes ago I already typed "but the quantization noise should be audible ?" ... and then I thought to listen for it. Josef, your both versions are full of it. Mani's has NONE.

Yes, of course :)
Both 'my' versions were downsampled from 176 to 44 - AFAIK By definition, downsampling will create quantization noise, won't it? The question is not whether it is there (it has to be) but does it negatively affect the sound?

Now, if you say Mani's version has no quantization noise that would also make sense if ADC did _not_ dynamically downsample 176 to 44 but, rather, as suspected, just kept every 4th sample - right?

>You can hear it right in the beginning, when the needle is still in the lead-in.

Lead-in lasts maybe 2 secs on Mani's 16/44 but only maybe 0.5 sec on 'my' version - Sorry, but I could not hear any weird noises in 0.5 secs, it's just too short a time before drum starts hitting - Or did you mean something else?

(see also next post)

Hi Josef,

Awaitening your "next post" ...

Still not sure yet, but I think I start to understand better what to make of your earlier post (the one from 6 hours back or so);

Yes, quantization noise will be there, but therefore is the Anti Alias filter. So, in your decimation you didn't apply that, right ? (or a bad one perhaps ?). Anyway, this is not about dither ...

Again, I may understand you wrongly, but taking out 1 out of 4 samples (at going from 176.4 to 44.1) doesn't work, as it doesn't work the other way around (inject samples at positions by "normal" math); That needs an Anti Imaging filter ... (AI). One exception : inject the samples at the proper position and it's ok (this is AP ... hopefully :)).

I once started out with decimation myself, and did this for DXD (which is 24/352.8 ) as provided by 2L. I didn't have a 24/352.8 DAC at that time, so I couldn't check the original. And thus, the quantization noise (I know now) was all over. However, instead of interpreting it like that I called 2L and told them their classical samples were quite noise ...
I'm still red on the cheeks from that conversation. So, they created a properly decimated version especially for me, and it was completely out of noise.
An AA filter just does the trick.

Yes, Mani's version has a longer lead in to listen to, but at comparing this afternoon I could notice the relatively loud noise from (indeed) your half of a second from any distance. Careful here, because I have horn speakers, and they unveil quite some more than any "normal" speaker. But moreover, you can listen "in" the horn with your ear, and there you hear everything.
Also one must know how quantization noise sounds. This is a whizzling quite unlike the ticking ans "sliding" (which you can also hear) of the needel over empy space (which is not empty, because it has to be lead). In dB's I would estimate your "general noise" at 12-15 (over two times more noise).

I don't think this is a matter of "will it be audible or not", because it is nothing like dither (needed). It's just plain distortion which digs itself into anything. As a matter of fact, before listening to the lead in with my ear in the horn, I first listened to the piano, and some later in the track when the drums get more heavy you can hear how that distorts (notice the piano is in the right speaker, so listening to the drum which is in the left, but halls over to the right is quite easy to do (without hurting)). This is not the case with Mani's version.

Lastly (but you may know it), dither is about the decimation in bits (not sample rate) and making more of it than actually there. The more really random the dither is, the better it works. If not really random, patterns will emerge, and it just takes away the sound.
Notice that with proper dither even with 1 bit you can still recognize music. RME has (or had) a test file for it.

Peter

To be clear :


Totally impossible.
:)

Josef, why do you 'suspect' that Keith Johnson and Michael Ritter (of former Pacific Microsonics) would be lying about the decimation processes in the Model Two?

I can create a 16/44.1 file from the native-24/176.4 file using the Model Two's digital-to-digital DSP engine. To do this, I will have to 'record' the Model Two's digital output as it downsamples the native-24/176.4 file to 16/44.1, so the two files will not be aligned or of quite the same length. Let me know if this will likely help us understand the Model Two's decimation process.

Mani.

Wait ...

I didn't look it up, nor will I ever try, but there exists at least some "dynamical" filtering, invented by some Japanese guru. IIRC it was implemented in some Denon model.

This meant that different filtering was applied to higher frequencies opposed to lower. Or better : that lower weren't subject to it, or something like that.

Also, I don't think that Josef was "accusing" as such, but merely has an idea about what "dynamically" could ever mean, besides commercial talk. So, I only want to point out that such a thing really existed.

Btw, that Japanese guy himself has been talking about it in DIYAudio IIRC (not working at the company concerned anymore).

2c

OK, here's more on 'dynamic decimation' from Michael Ritter:

"A filter designer who has to make a 'brick wall' filter at 22 kHz is confronted with conflicting requirements. You want to have flat frequency response out to at least 20 kHz, but you can't have any energy above 22 kHz or you will get alias distortion. This requires a very sharp multipole filter with a very steep transition between the passband and the stopband, which has a number of distortive effects on the signal. It smears transients and causes significant ripples in the passband. If you try to simplify the filter, then to avoid totally unacceptable aliasing you have to start rolling off at 13 to 15 kHz, and even then the signal will not be completely cut off by 22 kHz.

We slightly delay the 88.2kHz [or 176.4KHz] signal, not enough to cause any sync problems but enough that we can do a continuous Fast Fourier Transform. The resultant information is digitally analyzed in real time by an algorithm that determines, based upon a model of the mechanics of hearing and psychoacoustics, what is perceptually dominant in the signal from instant to instant. And that information is used to optimize the decimation filter. One moment you might have a sudden sharp transient, so it uses a filter with minimum time dispersion to pass the transient cleanly. The next instant, there might be a cymbal crash, so it uses a filter that minimizes alias distortion. All the filters are the same length, so you are not getting a phase shift as this is going on.

We never simply truncate. We have a palette of four 16-bit dither options. The reduced distortion-sharper transient response and reduced aliasing-becomes part of the digital recording and will be heard on any player, whether it has HDCD decoding or not."

HTH.

Mani

Guys "next post" is hopefully coming quickly but in meantime:

>>>More importantly, I suspected that ADC also did some limited/poor quality dithering or no dithering at all i.e. it just chopped last 8 of 24 bits.

>This is not the case. The Model Two provides a number of 16-bit dither options, which I understand are pretty sophisticated.

Yes - I found the Instruction Manual (that's why you need to wait for "next post" :) ) and I stand corrected: there are no less than 7 different dithering algorithms, lol :)

Here however, we have a small problem: you mentioned that you did NOT use any HDCD processing but looking at manual under dither it is called "HDCD 16-bit Dithers"? It also says that DITHER1 algorithm is default. Is this just marketing again (i.e. dithering has nothing with HDCD encoding) or does this imply that _some_ HDCD internal processing is still being done if dithering is used?

Because if it is then we are running in circles - of course we cannot HDCD encode via software (at least I can't)...

And, just to clarify: did you use mode AD_44.1 or AD+DD44 or something else for 16/44 version? And I assume word length was set to HDCD16?

And oh:

>Josef, why do you 'suspect' that Keith Johnson and Michael Ritter (of former Pacific Microsonics) would be lying about the decimation processes in the Model Two?

Please, where did I state they were lying? People use marketing terms 'freely' all the time and these guys would not be the first so it does not hurt being suspicious - just look at that guy's Goodwin's site - PM Model 2 has processing power of '200 million instructions per second' - Wow that's as fast as 3x 486/66MHz computers or almost half of Pentium Pro from 1996 :)
 
Let's keep this discussion open no need to get excited (and especially not over some product :) )

>Yes, quantization noise will be there, but therefore is the Anti Alias filter. So, in your decimation you didn't apply that, right ? (or a bad one perhaps ?). Anyway, this is not about dither ...

Alias was used - but only the default, I'll try something else.
And you may be right that we should focus on aliasing filter: I was just thinking - if version 2 without dithering but with different aliasing is sounding closer to native then maybe dither is not important.
Still - I think it will be important at least to a point: I think I know how I can test this (if I can find a piece of software I bought long time ago but never used, lol :)   

My understanding is that dithering has nothing to do with HDCD decoding, so shouldn't be an issue here.

I didn't use any of the three HDCD 16-bit amplitude encoding options.

Mani.

AD_44.1 with wordlength set to HDCD16. BUT... Peak Extension and Low Level Extension were NOT applied.

(As explained earlier, my native-16/44.1 file was saved as a 20-bit file by the recording software - but the signal is 16-bit.)

Mani.

Let's keep this discussion open no need to get excited

The most important, but I'm also the most sure we all are.

Both open and excited. :) :)

Well, I am. I really think it is important.
Just wait until you heard what I have been up to all the time. And, I think I'm finished with it, but I am not.
Not by a mile. I'm fairly sure of that.

I really don't think there is an issue with comparing the native-16/44.1 file (encoded with 'HDCD dither') with the native-24/176.4 file. However, if anyone believes that this is not totally fair, then fine, just use Josef's original 16/44.1 file for the comparison.

Mani.

Well, the technical part is over my head, although I get reminiscences of the mp3/atrac/dcc explanations from 20 years ago.

The piano still intrigues me, is the rhytm missing on 44 kHz too?
To me it sounds like the piano is missing some of it's impact because of resonances from the recordplayer-arm.
I don't have this record but on my player I like the pianosound a lot.
If only I could record at 192/176, not even 44 at the moment.

A nice explanation can be found here: http://www.thefunkfirm.co.uk/arms/fxr_II.html

If this is the case, a very good record player could give us a perfect digital source??

The Denon 103 is a very low compliance cartridge, so resonance could be an issue. I've tried to mitigate this by using a high-mass arm. The resonance peaks at ~9Hz, which is considered 'ideal'. But this does not say anything about the magnitude of the resonance. I have no idea how low it is in my setup.

Does anyone else hear any anomalies in the recordings that might be attributable to arm resonance?

Mani.

Mani,

It's not the 9 Hz resonance but the ones at the audible frequencies like the page about the Funk Firm Fxr arm shows.
If you see that, your heart skips a beat, but of course help is on it's way.
Just buy their arm.  ;)
BTW. Funk Firm is the new name for the restarted Pink Triangle firm, known for it's very good=neutral sounding record players.

>My understanding is that dithering has nothing to do with HDCD decoding, so shouldn't be an issue here.

Yeah - but it has to do a lot with encoding...
Or maybe not a lot as Peter suspects and as I will try hopefully try to test soon :)

I've finally found an old licensed copy of secret rabbit resampler I knew I had somewhere but never really needed before now :) Since discussion seems to indicate that SRC is more important than dither I made couple 90 secs samples from master 24/176 with secret rabbit (with 2 different dithers - just to be sure), sox (http://sox.sourceforge.net/SoX/Resampling) and same pro-resampler used for v1 but this time set to max aka 'ideal' filter steepness.

So, for those still not totally fatigued :) :
http://www.filemail.com/dl.aspx?id=SWAJCATOPTNNITT

Peter, can you put your head again in your horns and check if distortion ('quantization noise') in <0.5 sec is still there? (I honestly can't hear it: maybe because I'm on the road and putting my head into 10cm portable speakers does not quite work, lol :)
If it is, then it can't be aliasing filter can it? (with 4 different algorithms?)

On a separate note: Did I understand correctly that AP is performed in real-time? If so, doesn't that go against the principle of no processing during playback? (or did you compare to AP pre-processed files and found no impact?)

Josef - Nothing is done in real time. But maybe I confused you with calling AI a "pre-pre" process.
AP is a "pre" process. As how I call it, a pre-process happens all in memory, while a pre-pre is happening even before the sound engine starts (FLAC conversion would also be an example).

So, a pre process happens in memory, but not during playback. Before it.


When I can use the room again (family is watching some TV right now), I will look for quantization noise !
... and hope it is not there this time ... :)

>As how I call it, a pre-process happens all in memory, while a pre-pre is happening even before the sound engine starts

Aaah, gotcha...

In sequence of "least noise" :

1. Sox.
2. Mani's (but difficult to compare because of the longer lead in; could be equal so Sox just as well).
3. Rabbit Dither2
4. Rabbit Dither1
5. v1-Max
6. V2
7. DND (actually equal with V2).

Peter

Of the four new 16/44.1 files posted by Josef, my strong preference is for v1-Max. I don't think much of the two Rabbits or the Sox.

Mani.

As an interesting aside (and a welcome 'Time Out' from 'Take Five'... :)), Blue Coast Records are offering some free downloads here: http://www.bluecoastrecords.com/cas-2010-downloads.

My understanding is that these were originally recorded in DSD on a Korg. They have been converted to PCM in two resolutions: 16/44.1 and a 24/96. It's quite interesting to compare these two versions.

(But I have to say that I really don't like DSD. I wish they'd just have recorded in hires PCM in the first place.)

Mani.