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post #361 of 783

Jitter affects frequency across the board. In communication that I worked on, the voice band is limited to 3.5KHz. But since we have to pass this signal through miles of cable (digital), jitter becomes a big issue. In today's technology, packet loss and latency become a bigger issue. Interesting thing is people are so used to poor voice quality of cell phone. No one has complained about the reduced quality. This is what I mean by our ears are getting burned in.

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post #362 of 783
Quote:
Originally Posted by dvw View Post

My issue with the explanation that jitter in the cable are:

1. Description of the sound of jitter. I am not dealing with people's ability to hear jitter yet. No one has given a qualitative description of jitter. Degradation of audio is not a description. More bass/more treble is not a description. Jitter actually modulated the signal. For example, if you play an off center LP, part of the playing will be at a higher speed than the other part, this change the frequency of the signal. This is not an  accurate analogy of jitter but it is simple to use without getting into Fourier Transform.

 

 

It's a timing error (or frequency modulation, if you want to think of it that way... in the frequency domain, you get some sidebands and spread of the original frequency).  At 44.1 kHz sampling rate, you're supposed to more or less get a new sample (of course it's not an abrupt occurrence, because of the nature of the filtering and reconstruction) every 22.6757 µs or so.  Let's call that amount the time unit.  A 441 Hz tone should be 100 time units per period.  With jitter, it might be 100.001 samples per time unit, or 99.999 samples per time unit.  Or sometimes 100.002 or 999.998 samples, or less frequently 100.0008 or 99.9992 samples?  I guess this should sound like some kind of flutter / warble / vibrato effect.

 

 

 

Quote:
Originally Posted by dvw View Post

2. Second issue I have is nobody has explained to me how is the play clock derived from the USB clock (source of jitter right?). If a recording is done with a 44.1KHz +/- 1 KHz clock, How does the USB know it is 43KHz and not 45KHz? What if there are two tracks streaming and one with 43KHz and one with 45KHz?

 

By "play clock" you mean the timing of the DAC's output?  If the recording was done with a jittery ADC process, then the recording should be compromised (audibly so, if somehow at a high enough level) because information was not sampled at the correct times.  That has nothing to do with any USB transmission later on.  I'm not sure what you mean by two tracks streaming at different frequencies.  In one audio signal chain, there's only going to be one track, and its average frequency is going to be 44.1 kHz (48 kHz, 96 kHz, etc.).


Edited by mikeaj - 8/30/12 at 5:26pm
post #363 of 783
Quote:
Originally Posted by mikeaj View Post

 

 

It's a timing error (or frequency modulation, if you want to think of it that way... in the frequency domain, you get some sidebands and spread of the original frequency).  At 44.1 kHz sampling rate, you're supposed to more or less get a new sample (of course it's not an abrupt occurrence, because of the nature of the filtering and reconstruction) every 22.6757 µs or so.  Let's call that amount the time unit.  A 441 Hz tone should be 100 time units per period.  With jitter, it might be 100.001 samples per time unit, or 99.999 samples per time unit.  Or sometimes 100.002 or 999.998 samples, or less frequently 100.0008 or 99.9992 samples?  I guess this should sound like some kind of flutter / warble / vibrato effect.

 

That's a better articulation.

 

 

By "play clock" you mean the timing of the DAC's output?  If the recording was done with a jittery ADC process, then the recording should be compromised (audibly so, if somehow at a high enough level) because information was not sampled at the correct times.  That has nothing to do with any USB transmission later on.  I'm not sure what you mean by two tracks streaming at different frequencies.  In one audio signal chain, there's only going to be one track, and its average frequency is going to be 44.1 kHz (48 kHz, 96 kHz, etc.).

By play clock, I mean the sampling rate that you're playing the recording with. Let's say the recording is done with a sampling rate of 44.1 KHz with a tolerance of 1 KHz. So it is recorded with 43KHz. And you derive the clock from the USB and get a sampling rate (play clock) at 45KHz. You will have a buffer underflow and the file is virtually unplayable. If you have a file recorded at 43 KHz and another at 45KHz, how do you play them back to back from the derived clock from USB. Now if you say USB transmission has nothing to do with it. Then the jitter from the USB transmission has nothing to do with the playback at the DAC.

 

This is a long winding way of saying jitter at the USB does not affect play back. Only the jitter at the DAC clock matter.


Edited by dvw - 8/30/12 at 5:42pm
post #364 of 783
Quote:
Originally Posted by estreeter View Post

I can't wait to hear NAD's C390DD (the M2 is well above my budget) - taking the analog conversion completely out of the equation is an interesting prospect. Ironically, anyone needing to hook an analog source up to one of these amps now has to buy an optional MDC module. Ditch the DAC for an ADC and look forward to DSP updates further down the pike.Outstanding. biggrin.gif

 

Kind of like a USB cable direct to the power amp, kudos to NAD for shaking things up.

 

 

 

 

 

 

It had better be one bad@ss USB cable though wink_face.gif

post #365 of 783

grokit, I'm assuming you have read Darko's conclusion to his C390DD review ? Joining 6Moons seems to have endowed him with a touch of Srajan's legendary loqaciousness. but he gets there in the end. I get where he is coming from with the 'Power DAC' terminology. but its a misnomer for the NAD amps as your diagram clearly illustrates. 

 

As a one-box solution NAD’s C390DD seals all the magic under one roof, where DAC and amplifier play as conjoined twins. There are no interconnects to ruin the party or blow out the budget. If resolution and transparency are your top priorities, NAD’s ‘power DAC’ could have you starting again (seriously). With Class D, there will always be those that espouse the ruling classes (A and A/B) without apriori listening. To those unencumbered by such dogma, I say this: if your current DAC and amp ran you close to $3k in pursuit of a diffident neutrality, give serious though to selling them and buying this. Yes, for real.

post #366 of 783

Yes the only problem with the integrated setup is the same as all integrated components when compared to their separated counterparts. If one section fails or you want to change your sound, all of your eggs are mounted to the same chassis.

post #367 of 783
Receivers used to have jumpers in the back connecting their preamp circuitry to their power amp circuitry, so you could patch in at the last stage to equalize or do whatever you wanted to everything all in one spot. I haven't seen those lately. It's a drag. They probably dropped that feature because of multichannel sound.

There really isn't a tape loop with optical or HDMI. I don't think my Yamaha will allow you to map the output of the HDMI out and then back in. How do you patch in a 5:1 equalizer? I guess you'd need an external HDMI or optical switcher of some sort. That would be an ugly kludge.
Edited by bigshot - 8/31/12 at 11:52am
post #368 of 783
Back in the day I seem to remember a lot of pre outs, but few if any amp ins.
post #369 of 783
Quote:
Originally Posted by grokit View Post

Yes the only problem with the integrated setup is the same as all integrated components when compared to their separated counterparts. If one section fails or you want to change your sound, all of your eggs are mounted to the same chassis.

 

The NAD amps gp beyond that - unless you want to buy one of the analog modules, you may as well throw any tube pre-amp (or CDP with a tube output stage) you own in the closet - even with said module, your lovely analog signal will be digitised before it hits the power amp in the NAD. It really is an 'all or nothing' choice', but how many SET amps offer you the potential to upgrade the sound with nothing more than a firmware upgrade ?  wink.gif

post #370 of 783
Quote:
Originally Posted by Clarkmc2 View Post

Back in the day I seem to remember a lot of pre outs, but few if any amp ins.

It was a pair of U shaped metal bars on the back. I never knew what they were for until a friend showed me.
post #371 of 783
Quote:
Originally Posted by grokit View Post

 

Kind of like a USB cable direct to the power amp, kudos to NAD for shaking things up.

 

 

 

 

 

 

It had better be one bad@ss USB cable though wink_face.gif

With this configuration, they should have an WiFi option with a source selection menu. With that you can stream sound through out the house from your servers, PC, iPOD.

post #372 of 783

Announcement:

 

I think we have missed the forest for the trees. USB cables are the problem, the USB controller is the problem, I have heard the the Intel PCH controller is much better than any of the ASMedia chips, you should check your controller and make sure you have your $300 USB cable connected to the correct controller.

post #373 of 783
Do you think it beats an apogee symphony io interface? Thats my dac
post #374 of 783

Bigshot: I used to have a Luxman amp with those. I think it may still be possible with their current series, but you have to enable it via a switch.

 

TickleMeElmo: Some audiophile types buy special USB cards for computers they are using for dedicated playback. Other than that, I've always been of the mind that it simpler to build a PC using Intel boards as the likelihood of issues with chips and drivers are minimised, but I'm not that experienced in building PCs. I do recall some of the early USB and USB 2.0 controllers being absolutely awful.

post #375 of 783

I'm actually really interested in that NAD thing now, provided that I have a few of the NAD amps already. :)

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