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Author Topic: 16/44.1 vs. 24/176.4  (Read 89925 times)
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manisandher
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« Reply #60 on: August 21, 2010, 05:32:00 pm »

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

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

Mani.
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« Reply #61 on: August 21, 2010, 06:56:33 pm »

>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 Happy
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)
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PeterSt
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« Reply #62 on: August 21, 2010, 07:24:41 pm »

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 Happy).

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
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« Reply #63 on: August 21, 2010, 07:28:27 pm »

To be clear :

Quote
but, rather, as suspected, just kept every 4th sample - right?

Totally impossible.
Happy
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manisandher
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« Reply #64 on: August 21, 2010, 07:29:31 pm »

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?

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.
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« Reply #65 on: August 21, 2010, 07:44:53 pm »

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
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A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

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« Reply #66 on: August 21, 2010, 07:56:29 pm »

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
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« Reply #67 on: August 21, 2010, 08:56:22 pm »

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" Happy ) and I stand corrected: there are no less than 7 different dithering algorithms, lol Happy

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)...




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« Reply #68 on: August 21, 2010, 08:59:28 pm »

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?
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« Reply #69 on: August 21, 2010, 09:11:21 pm »

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 Happy
 
Let's keep this discussion open no need to get excited (and especially not over some product Happy )

>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 Happy   
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« Reply #70 on: August 21, 2010, 09:24:42 pm »

Is this just marketing again (i.e. dithering has nothing with HDCD encoding)...

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.
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« Reply #71 on: August 21, 2010, 09:29:39 pm »

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?

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.
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« Reply #72 on: August 21, 2010, 09:33:46 pm »

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. Happy Happy

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.
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XXHighEnd Mach III Stealth LPS PC -> Xeon Scalable 14/28 core with Hyperthreading On (set to 14/28 cores in BIOS and set to 10/20 cores via Boot Menu) @~660MHz, 48GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/0/0/*1*/ Q1Factor = *4* / Dev.Buffer = 4096 / ClockRes = *10ms* / Memory = Straight Contiguous / Include Garbage Collect / SFS = *10.13*  (max 10.13) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / Stop Desktop, Remaining, WASAPI and W10 services / Use Remote Desktop / Keep LAN - Not Persist / WallPaper On / OSD Off (!) / Running Time Off / Minimize OS / XTweaks : Balanced Load = *62* / Nervous Rate = *1* / Cool when Idle = n.a / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filtering *Low* (16x) / Always Clear Proxy before Playback = On -> USB3 from MoBo -> Lush^3
A: W-Y-R-G, B: *W-G* USB 1m00 -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (16ms) -> B'ASS Current Amplifier -> Blaxius*^2.5* A:B-G, B:B-G Interlink -> Orelo MKII Active Open Baffle Horn Speakers. ET^2 Ethernet from Mach III to Music Server PC (RDC Control).
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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manisandher
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« Reply #73 on: August 21, 2010, 09:43:02 pm »

... then we are running in circles...

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.
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Main System:
Phasure Mach III (Win 14393.0 on RAM-OS / controlled by RDC, / connected directly to music server / XXHighEnd 2.11 / Minimize OS / Engine#4 Adaptive / DB=4096 / Q1=10 / xQ1=15 / Q3,4,5=1 / SFS=4.00 / XTweaks = 34, 10, 0, 0, 0 / Straight Contiguous / Clock Resolution = 15ms / Scheme 3-5 (low/realtime) / 8x Arc Prediction / switch #5 'up/off' / Unattended) mobo USB3 port -> Lush^3 -> Phasure NOS1a B75 G3 -> 8m Blaxius^2 -> First Watt F5 mono amps -> Tune Audio Anima horn speakers
Office System:
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GerardA
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« Reply #74 on: August 21, 2010, 10:04:33 pm »

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??


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