Wide Bit Stream on portables MZ-R 30, MZ-R 900, MZ-N 510 & Sharp MD-MT 180

[PhilippeC]

Now there we differ.

I can find no evidence that Type-R recording is inferior. I don't actually think it is.

From everything I read, Type-S is a playback enhancement. My first hand experiences confirm that.

Stephen

I know that type R is probably the best for recording, no need for a type S if you use only SP mode. So you are right, not inferior because it is not use for the same thing. BUT for those who use also the two LP modes, the type S will give the best listening quality. I suppose for what I have learned from this forum that LP is recording with the type-R part included in the type S. So type S units ARE the best for both recording and listening for LP mode, and type R is enough for SP mode. Otherwise, why everybody is still loving JA555ES and JA50ES ?

And I am still curious about the LP sound quality on a JA333ES which is type R "only"...

Stephen, you have a Sony MDS-JB980 and - if I am not wrong - a MDS-JB940. I know that you have measured (or rode somewhere) that optical out is real 24bit for the JB940. So once connected to a recent 24bit-192kHz Digital-Analog convertor (amplifier or DA), I suppose the sound is a little bit better that the JB980. But, for the two analog outs, do you ear or are aware of a difference ? I don't think so.

[sescoscuba]

I spent 95% of my time to listen to music at home,

In fact, it is difficult for me to have a good idea about the quality of the non Sony units. We need people like you who have both to tell us why Sharp, Panasonic or Aiwa are good alternatives, specially when price is very afordable.

You are a lucky man, Philippe

if I could have more time to devote to music, I would like to repeat other comparative tests of sound, between machines more homogeneous than the tests that I've done.

Approximately Kenwood and Sharp, portable, have a purchase price of 30/40% less than machines with the same features of the Sony

[MDietrich]

I think the ATRAC version of the Sharp is a few genarations behind the Sony even though the development time seems to indicate the direct opposite. The reason for this is I believe previously the version of ATRAC wasn't as important as the sound tuning of the whole device. So Sharp felt it could make a good sounding unit with a not so recent ATRAC DSP by tuning the overall unit. Sony however took a different approach by reducing / almost eliminating the quantization noise TYPE-R and -S.

Also you mentioned LP4 isn't useful to you. But you need to explain how you did your analysis for LP4.

Of course the sound of the recorder/player I use is important. But every player - good or bad - needs a perfectly encoded source I think. A good player won´t improve an encoding that hasn´t been encoded well in the first place. The player only hides possible errors in that case. And I don´t like to be fooled.

And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. But I haven´t tried anything else, Audiobook or anything else.

I have checked all the guides available on the net where we talk about the kenwood, everyone is talking about atrac 4.5.

I do not doubt that you have the 4.0, but your K. has never been brought to assitance for repairs?

No, never. It has never been repaired. The chip had a name: CXD-2652AR. That was surprising to me since it´s exactly the same ATRAC chip used in my MZ-R 30 and I thought it to be a special chip engineered for portable use only (-> energy conserving).

My impressionI repeat, my impression on the portable, is that Sony has produced machines for a more balanced listening in a quiet environment, while Sharp has the most favored listening in noisy environment.

The reason for that is simple: the Sharps have much more powerful headphone outputs. Most MD recorders or players from Sony have 5 mW on their output while many Sharp have 10 mW. With the Sharps one should be able to listen while being in a noisy environment.

[sfbp]

And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. one should be able to listen while being in a noisy environment.

"What I say three times is true" (Lewis Carroll, the Hunting of the Snark).

I utterly dispute that LP4 is of itself horrible. Perhaps when you tried it, and the way that you tried it.

[PhilippeC]

You are a lucky man, Philippe

I mean, the time I spent to listen to music is 95% at home. Not 95% of my life time is to listen to music. I would like so.

[MDietrich]

"What I say three times is true" (Lewis Carroll, the Hunting of the Snark).

I utterly dispute that LP4 is of itself horrible. Perhaps when you tried it, and the way that you tried it.

I don´t mean to be disrespectful - but this is an opinion you would have to accept as my own. I´m not making general assumptions. Furthermore, I would never have combined senctences out of context in the way you quoted them. Besides offering a sound that is unacceptable to me (for me, which doesn´t necessarily mean to anyone else) it also creates intermodulation distortions so high that they are readily audible.

[sescoscuba]

The reason for that is simple: the Sharps have much more powerful headphone outputs. Most MD recorders or players from Sony have 5 mW on their output while many Sharp have 10 mW. With the Sharps one should be able to listen while being in a noisy environment.

Hello

I'm sorry but I disagree, doubling the output power of the sound increases by 3 dB, however, listening to music for almost 40 years I am able to distinguish between sound intensity and sound quality.

Have a nice day

Sergio

[sfbp]

this is an opinion you would have to accept as my own.

Of course I accept that as your opinion. But you used the phrase "is horrible". Not "I think it's horrible". You say "is horrible" in the context of an article presenting facts in a scientific way. This in turn vitiates the undoubted strength of your careful analysis.

Furthermore, I would never have combined senctences out of context in the way you quoted them.

What on earth are you suggesting? Please - I cut and pasted exactly from what you said. No editing. Review what we each wrote and you will see I simply played back your words to you.

it also creates intermodulation distortions so high that they are readily audible.

Only if you try to go straight from a CD or so (1411kbps or better) to 66kbps. Starting from a broadcast stream (of 128kbps) using Type-R recording (at least with a deck) at 66kbps is frequently excellent. As I've stated on this board before.

[MDietrich]

Of course I accept that as your opinion. But you used the phrase "is horrible". Not "I think it's horrible". You say "is horrible" in the context of an article presenting facts in a scientific way. This in turn vitiates the undoubted strength of your careful analysis.

My articles are not scientific. I frequently address this issue during most of my articles. And I made it clear that it is my personal opinion by saying "And I mentioned LP4 is horrible (it really is, I can´t stand it)" - I think "I can´t stand it" should make it obvious that it´s my personal opinion. But I´ve just edited my article to make it more clear that a particular statement of mine is my personal opinion.

What on earth are you suggesting? Please - I cut and pasted exactly from what you said. No editing. Review what we each wrote and you will see I simply played back your words to you.

I wrote: "And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. But I haven´t tried anything else, Audiobook or anything else."

You quoted: "And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. one should be able to listen while being in a noisy environment."

The part that has been misquoted in context has been underlined by me.

Only if you try to go straight from a CD or so (1411kbps or better) to 66kbps. Starting from a broadcast stream (of 128kbps) using Type-R recording (at least with a deck) at 66kbps is frequently excellent. As I've stated on this board before.

The recording quality should be even worse then because it is basically a transcoding from one lossy format to another lossy format. But I never did such a recording so I cannot comment on this. I´ll accept your observations.

I'm sorry but I disagree, doubling the output power of the sound increases by 3 dB, however, listening to music for almost 40 years I am able to distinguish between sound intensity and sound quality.

I´m sorry, I only wanted to offer a possible explanation.

[netmduser]

MDietrich

It would be great if you can explain how to do the analysis and the software you use. No doubt you have the skills and knowledge.

A suggested analysis is to take a wav file from a cd, the same wav file can be used for SP, lp2, lp4 and do your analysis. It would be interesting to see the comparisons. lp4 is the joint stereo encoding version of lp2. lp2 sounds almost the same as SP, so lp4 really should sound great but it needs a good source to begin with to help with its strong encoding. I think today some of us appreciate lp4 more than when it was first introduced with mdlp, it's such a convenience to store 5 hr 20 min of music on a single disc.

Don't use mp3 as source that seems to be the culprit for bad sounding tracks. Use only wav / pcm from highband sources. If you really need to use mp3 make sure its 256+ and convert it to wav outside of SS.

[sfbp]

But.... direct from high bit rate to LP4 may well have been the "standard" for many comparisons... but it ain't the proper test.

My theory as to why low bitrates re-encode well to LP4 is something like this.... most of the bits in the high frequencies are encoding noise. If you already cut out the noise by mastering down to (say) 256kbps by compression, then going from there to 66kbps does NOT waste any of the precious bandwidth (thank you, Sony and Type-R) encoding those noise bits.

Make sense? (I know it's just a hand-waving attempt to explain and understand what I have observed, which at first sight does not make sense).

[sfbp]

My articles are not scientific. I frequently address this issue during most of my articles. And I made it clear that it is my personal opinion by saying "And I mentioned LP4 is horrible (it really is, I can´t stand it)" - I think "I can´t stand it" should make it obvious that it´s my personal opinion. But I´ve just edited my article to make it more clear that a particular statement of mine is my personal opinion.

I don't see the point of introducing tables and figures if they aren't even attempting to be scientific. Your blog looks a lot like someone with a lot of knowledge attempting to convey at whatever level some scientific looking measurements for us mere mortals who have mainly our ears as evidence.

Perhaps you dispute that all those tables of numbers appear to readers to represent a scientific approach?

I wrote: "And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. But I haven´t tried anything else, Audiobook or anything else."

You quoted: "And I mentioned LP4 is horrible (it really is, I can´t stand it) because I listened to it. With music at least. one should be able to listen while being in a noisy environment."

Now I see the problem. The lack of capitals on the first word of the underline phrase is a giveaway. I attempted to delete everything I didn't want and this lot missed the scalpel. My apologies. It was from another statement about something else. This was not a deliberate attempt to confuse anyone. My quote should stop after "at least".

The recording quality should be even worse then because it is basically a transcoding from one lossy format to another lossy format.

Except that, empirically, it isn't. No worries; I too abandoned LP4 right away (and for 2 years after that) when I first got an MDLP-capable device - for exactly the same reasons you cite.

Stephen

[MDietrich]

It would be great if you can explain how to do the analysis and the software you use. No doubt you have the skills and knowledge.

The software I use for testing is called RMAA. You can get it here: http://audio.rightma...index_new.shtml - in its basic version (the one I use) it´s free.

But.... direct from high bit rate to LP4 may well have been the "standard" for many comparisons... but it ain't the proper test.

My theory as to why low bitrates re-encode well to LP4 is something like this.... most of the bits in the high frequencies are encoding noise. If you already cut out the noise by mastering down to (say) 256kbps by compression, then going from there to 66kbps does NOT waste any of the precious bandwidth (thank you, Sony and Type-R) encoding those noise bits.

Make sense? (I know it's just a hand-waving attempt to explain and understand what I have observed, which at first sight does not make sense).

Well... not completely. ATRAC - just like every other lossy codec removes high frequencies by principle (depends on the codec and the bitrate of course). For the ATRAC LP4 it doesn´t make any difference if there are frequencies or not, it will remove them anyway (it cuts away everything at 13 kHz - well within audible range). Going from a lossy format (for example mp3) to ATRAC would be worse for quality in theory. When a lossy codec takes something away the thing that is left is more or less clever hidden quantization noise (beside a lowpass, joint stereo, etc.). Imagine a lossy source for ATRAC to encode: it not only has to encode the things anew that already have been encoded, it also has to encode the audible rest AND the quantization noise. We might not hear it, but ATRAC does, it even hears things we will never ever hear. When going from a lossy source to ATRAC it has to recompress all the data that was left during the first encoding. And depending on the bitrate this may not be that much. If it still sounds transparent that does only mean that us humans can be fooled easily.

I don't see the point of introducing tables and figures if they aren't even attempting to be scientific. Your blog looks a lot like someone with a lot of knowledge attempting to convey at whatever level some scientific looking measurements for us mere mortals who have mainly our ears as evidence.

Perhaps you dispute that all those tables of numbers appear to readers to represent a scientific approach?

Now I see the problem. The lack of capitals on the first word of the underline phrase is a giveaway. I attempted to delete everything I didn't want and this lot missed the scalpel. My apologies. It was from another statement about something else. This was not a deliberate attempt to confuse anyone. My quote should stop after "at least".

Except that, empirically, it isn't. No worries; I too abandoned LP4 right away (and for 2 years after that) when I first got an MDLP-capable device - for exactly the same reasons you cite.

Please, Stephen, you don´t need to apologize. Nor does anyone else. I´m the one who confuses people... You know... I don´t want to complain but it´s funny. Scientists accuse me of dumbing down my articles so that regular people (aka scientists) might understand them which is my goal. I write because I want to clean up misconceptions about vinyl, high res, lossy codecs, etc.. But these regular people state that my articles are too complex, require too much foreknowledge or are too scientific.

If I would be scientific I would do a complete study where I first would have to create a null theory to begin with, would do many measurments to have as much empirical and statistical data as possible, would incorporate probability (statistical probability), would avoid personal opinion completely, would transparently explain my setup (well, that I mostly do) - and I would do real measurments. The software I linked above has many flaws, one of them is that it isn´t entirely objective. It cannot measure everything and it interprets things wrong. The numbers are just numbers already interpreted. These numbers it presents don´t have any real expression.

Now my dilemma: I have to use it because I cannot afford true measurment equipment (which would be bulky, many different things and horribly expensive). The graphs RMAA creates are just one thing: nice looking pictures. I use them to make assumptions just because I don´t have anything else. But in reality they are only pictures. The charts are just what RMAA presents to me - and since no one knows how RMAA works on the inside (it could easily lie for all its worth) it isn´t reliable. A different set of measurments could come to a completely different outcome.

I try to work around the flaws of RMAA by stating electrical charts of the things I "measure" so that anyone knows how to compare them. The problem with RMAA is that many use it: private people, computer magazines... and those results are completely worthless because no one usually states how these measurments were created. For example: I´ve read a review for my Soundblaster X-Fi HD USB from a big magazine. They measured it using RMAA and had horrible results making the card much worse than it really is. From my experience with Windows I however knew that a misconfigured Windows audio engine (samplerates mismatch) produced these results. But they didn´t know - and they are supposed to spot this error! But RMAA didn´t tell them that they made an error - it doesn´t know that. Could also be that they were paid for the review to look bad by ASUS or another audio card manufacturer, happens all the time. The funny thing is that not one of their readers noticed the errors.

I repeat my "measurments" as much as possible to create a relatively expressive result but I cannot really know.

Sorry for answering this long. This thread has derailed and its my fault.

BTW, would anyone want to see measurments for LP2 or LP4? I can use Sound Forge to create pure ATRAC3 or ATRAC3Plus files (not ATRAC2 though).

EDIT: I will write an article about codecs. Actually I´ve been planning this but now I will do it. If every coded works the way I expect it to work I will show what has been erased and what is left - dynamically alternating quantization noise basically. With a bit of effort I will also include ATRAC2, ATRAC3, ATRAC3Plus, WMA, OGG, AAC, MP3.

EDIT II: I´m just listening with a Sony MZ-R 55 I´ve acquired a few days ago. I know everyone hates it... but I love it! It looks so beautiful and feels so nice to the touch... I hope that it doesn´t break to soon. And boy, it is fast! It sounds so well, it easily beats my MZ-R 30 and comes close to the MZ-R 900. Of course it´s WideBitStream capable during digital recording even though it cannot give it out analogue (16 bit D/A converter) I´ll plan to write an article about it and a player I now have (Sony MZ-E 60).

I just had an idea: Stephen, you have the JB980, yes? I would think that it does indeed puts out a true 20 Bit digital signal. I´ve seen the Service Manual for our Kenwood DM-5090 and there the optical output goes straight from the ATRAC chip to the output. 20 Bits seem to be the standard output for ATRAC. Or there´s something I´ve overseen. You could test it yourself with RMAA (would be good enough for that) and a Soundcard having an optical input using 24 Bit and a matching software to record that.

[sfbp]

I just had an idea: Stephen, you have the JB980, yes? I would think that it does indeed puts out a true 20 Bit digital signal. I´ve seen the Service Manual for our Kenwood DM-5090 and there the optical output goes straight from the ATRAC chip to the output. 20 Bits seem to be the standard output for ATRAC. Or there´s something I´ve overseen. You could test it yourself with RMAA (would be good enough for that) and a Soundcard having an optical input using 24 Bit and a matching software to record that.

I don't think for an instant there's actually 24-bit signal coming out the optical port on the 980. There's no setting to allow this, and if it were truly 24-bits, then the devices hooked to it would either a. register 24-bits (my receiver) or b. complain because they are expecting 16.

Tweaking it - perhaps, but at the present juncture I don't have the time and energy to delve into the electronics and find out just how the 940 (and presumably other decks) get 20- or 24-bit output. I note that all the ones which do, are Type-R only. AFAIK there's no Type-S deck with word length switchable.

[PhilippeC]

EDIT II: I´m just listening with a Sony MZ-R 55 I´ve acquired a few days ago. I know everyone hates it... but I love it! I

Why everyone hate the Sony MZ-R55 ? I love it too

[MDietrich]

I don't think for an instant there's actually 24-bit signal coming out the optical port on the 980. There's no setting to allow this, and if it were truly 24-bits, then the devices hooked to it would either a. register 24-bits (my receiver) or b. complain because they are expecting 16.

Tweaking it - perhaps, but at the present juncture I don't have the time and energy to delve into the electronics and find out just how the 940 (and presumably other decks) get 20- or 24-bit output. I note that all the ones which do, are Type-R only. AFAIK there's no Type-S deck with word length switchable.

That´s why I asked. The receiver we own is a Sony STR-DB 830. Its digital input will accept 24/96 signals but it´ll never show the bit depth, only the samplerate. So maybe your receiver just accepts it without questioning. Or maybe it´s "just" a 20 Bit signal which the receiver cannot show but will accept anyway.

[sfbp]

I did a wee bit of reading. Seems like S/PDIF Is inherently 20-bit, which I had not realised. Duh!

Presumably this can be scoped easily enough - it's just that I don't actually own one and would have to get my friend with scope to come by - this may take weeks before he has time.

Also - the specs for 980 state an SNR of "over 100dB" Clearly that's not possible with 16 bits only (theoretical maximum of 96dB). I'm intrigued. I took a look at RMAA, strange it was written in 2005 yet there is some comment about XP not yet being supported.........

[netmduser]

Y'all should read this before using this software RMAA http://nwavguy.blogs...lyzer-rmaa.html

An interesting test is the loopback test - take line out analog or optical to your minidisc recorder and let it go through the DAC to lineout and loopback

[MDietrich]

I did a wee bit of reading. Seems like S/PDIF Is inherently 20-bit, which I had not realised. Duh!

Presumably this can be scoped easily enough - it's just that I don't actually own one and would have to get my friend with scope to come by - this may take weeks before he has time.

Also - the specs for 980 state an SNR of "over 100dB" Clearly that's not possible with 16 bits only (theoretical maximum of 96dB). I'm intrigued. I took a look at RMAA, strange it was written in 2005 yet there is some comment about XP not yet being supported.........

Yes, it´s extremely difficult to assess all information about S/PDIF... what bitrate, what samplerate, etc. If I´m correct not even high samplerates are originally support. I may be wrong there though.

You´re also right about 100 dB being too much for 16 Bit. Though I have the Kenwood DP-5090 (a CD player) which measures as having almost 99 dB of SNR and dynamic while still being only 16 Bit. BTW, the best measured player yet... I mean, the best I´ve "measured" myself with RMAA. Interesting, isn´t it? The Kenwood measures better than the theoretical max. Either it does something odd or RMAA measures wrong. Or I did something wrong during measurment.

Y'all should read this before using this software RMAA http://nwavguy.blogs...lyzer-rmaa.html

An interesting test is the loopback test - take line out analog or optical to your minidisc recorder and let it go through the DAC to lineout and loopback

The article at nwavguys blog is extremely good! Very recommended reading for those who intend to use the software... why didn´t I think about the link? Thank you very much! As for the loopback test... that´s the one test I avoid like the devil avoids holy water. You don´t know what you´re measuring: the input or the output since both are connected. Which one is the noisy one? Which one distorts? One would never know...

For CD or MD players I always use just one soundcard: the E-MU 0202 USB. For the E-MU 0202 USB and the Creative Soundblaster X-Fi HD USB I´ve used the ASUS Xonar Essence ST since it shows extremely low noise and distortions. Of course, NOT doing a loopback creates the possibility of grounding errors (probably the reason for my measured differences with USB cables)... so not everything is peachy.

[sfbp]

If I've got this right, an extra 4 bits (20 vs 16) is the same as 16x (four doublings). So that would be 24dB with a theoretical max of +120 dB SNR. Check? Everything starts to make sense now... in particular I'd like to know how many bits of real resolution there are on the "average" CD, whatever that is.

[MDietrich]

If I've got this right, an extra 4 bits (20 vs 16) is the same as 16x (four doublings). So that would be 24dB with a theoretical max of +120 dB SNR. Check? Everything starts to make sense now... in particular I'd like to know how many bits of real resolution there are on the "average" CD, whatever that is.

Yeah. The scale for increased resolution is logarithmic. You´re also right about an SNR of -120 dB (more precise: maximal possible dynamic). With 24 Bit it would be -144 dB for SNR. Theoretical of course. My ASUS Xonar for example reaches an SNR of roughly -114 dB which would be roughly 18-19 Bits of resolution. Almost no microphones or microphone amps are as good. Today, digital recording resolution gets limited by the other things used to make a recording.

BTW, through proper dithering the CD can have a resolution close to that (it´s true). 96 dB are the calculated standard from times when the format was introduced. Through dithering and noiseshaping one can reach a dynamic of roughly 130 dB - all within 16 Bit and 44.1 kHz.

[PhilippeC]

For me, the audio quality of the source come first before the audio quality of the player (as a DAT or a MD recorder). But I am happy that you discuss anyway even I am already sunk.

[sfbp]

BTW, through proper dithering the CD can have a resolution close to that (it´s true). 96 dB are the calculated standard from times when the format was introduced. Through dithering and noiseshaping one can reach a dynamic of roughly 130 dB - all within 16 Bit and 44.1 kHz.

Here's my question: if you have effectively more data on the disk than the 96dB implied by the 1411/16 what's the point of programs like EAC which "merely" check that there is matching data at that data rate. Or does "dithering/noise shaping" ensure that perfectly copying the bits means that all that "extra" resolution is reproduced when a disk is ripped?

For extra credit: how do tools like Nero and/or Wave editors ensure that this extra resolution finds it way back onto a disk when burning CD's? Of course your answer may be that "home made" CD's are somehow "stuck" at 16-bit resolution.

[MDietrich]

Here's my question: if you have effectively more data on the disk than the 96dB implied by the 1411/16 what's the point of programs like EAC which "merely" check that there is matching data at that data rate. Or does "dithering/noise shaping" ensure that perfectly copying the bits means that all that "extra" resolution is reproduced when a disk is ripped?

For extra credit: how do tools like Nero and/or Wave editors ensure that this extra resolution finds it way back onto a disk when burning CD's? Of course your answer may be that "home made" CD's are somehow "stuck" at 16-bit resolution.

Don´t confuse dithering or noise shaping with ripping

Both things don´t have anything to do with each other. A CD with resolution better than the standard is like any normal CD. The signal is just contained inside the, let´s say "container", of the standard 16 bit PCM coding. To do this one needs a trick: dithering and noiseshaping. The resolution of 16 bit is basically limited by the quantization noise. The 96 dB possible dynamic were derived from the point where the quantization noisefloor starts. This noisefloor consists of nothing but quantization noise equally distributed throughout the frequency (has the same gain at 100 Hz, 1.000 Hz, 10.000 Hz etc.). In reality this noise isn´t really noise, it´s a distortion.

You see, noise (like tape noise, microphone noise etc.) is normally random and not disturbing to our ears. Quantization noise is not random and our ears are very able to pick it it up. So dithering does something very clever: it adds randomized noise to hide the quantization noise. Just like the quantization noise this noise is equally distributed. With that you reduce the dynamic of course to roughly 90 dB. And that´s where noise shaping comes into play: the noise is moved towards higher frequencies where you can´t hear it. The genius thing is that the quantization noise that´s still hidden within the dithering noise is moved along with it! So instead of equally distributed noise you now have a mountain of noise starting at 15.000 Hz - but no noise (not dither, not quantization) at lower frequencies. Et Voila: at those lower frequencies you now have a dynamic of 130 dB - and all inside 16/44.1!

You do this at the same time when decreasing bit depth from 24 to 16 Bit. If you wouldn´t you would loose any information below 96 dB as it would turn into quantization noise. Dithering and noiseshaping just avoid this by shaping the noise / distortion that would normally eat up musical information towards higher frequencies.

I´m sure you´ve seen CDs having a SBM print. SuperBitMapping is exactly that: noiseshaped dither. Just like Apogee's UV22, Telarc's 20 Bit process and countless others. A dithered and noiseshaped CD is just like any normal 16 bit with 16 bit nominal resolution - with a 24 bit information inside.

So it doesn´t matter if you use EAC, Nero or iTunes - it will always be a 16 bit CD with quasi 24 bits inside. You assume that the aforementioned programs add noise to the ripped CD - but let me assure that they don´t. A CD is a digital medium, you can copy it a million times and in theory it will always be the same (apart from jitter - but that´s something else) with no added noise, distortions or anything else.