# Optical Digital vs analog from HU to DSP



## Dan750iL (Jan 16, 2016)

Is there any advantage for one over the other? I read the post below and am thinking about getting that head unit. The advantages I can think of for optical right off the top of my head are running one thin cable vs up to 6 thicker ones and no chance of the cable itself picking up any RF interference. Are there any other advantages or disadvantages?

http://www.diymobileaudio.com/forum/general-car-audio-discussion/251066-clarion-nx706-out.html


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## Gonadman2 (Jul 25, 2008)

If you use a digital source like a CD or FLAC and an analog output then it will pass through a DAC before heading to the Processor. Once it enters the processor the signal will pass through an ADC where the signal can be manipulated via the processor functionality. Once it reaches the end of the chain, the signal is converted back to analog via another DAC, and is then output to the amplifiers. So it looks something like this:

Source (digital) > Source DAC (analog) > Processor ADC (digital) > Processor DAC (analog) > amplifier

Using a digital output on the source removes the DAC > ADC part of the chain as the signal is passed to the processor straight from the source, which looks more like this:

Source (digital) > Processor DAC (analog) > amplifier.

Typically, the DAC's in the Processors are better quality/sounding units so there is a nett benefit in that the lowest quality analog part is the processor.


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## oabeieo (Feb 22, 2015)

Going from digital to analog is a very complicated mathematical equation for every sample going into a digital to analog converter or DAC

a sample is basically a impulse response meaning a big chart of all the frequencies we hear in a file...think of it like an excel file containing every frequency, every and every frequencies phase and amplitude attached to each frequency on the chart. Depending on the sample rate will determine which frequencies to skip or how far past 20000hz it will extend to or it's depth...samples are processed in clusters (lots of them) depending on the sample rate. So a whole lotta samples of data is processed in an instant. If you have a sample rate of 96khz , than 96 thousand samples are processed every second. If the file itself has a bit depth of 24bit than each sample will contain 24bits of data. (Bit depth) so go back to that excel file, the sample rate determines the frequency the system will measure volume or amplitude, the bit depth will determine how many volume measurements the system has to work with .....

So I probably explained that partially wrong because I'm tired but that's not the point it's just an example ( but it's at least 90% right) or so...

Point is the DAC will convert a TON of numbers to electrical signals , depending on sample rate and bit depth some frequencies may be completely skipped , for example;
A 16bit recording may have have frequencies listed 0hz, 1hz,2hz,3hz on and on and gets to 35hz,36hz,37hz, than skip to 40hz leaving out 38 and 39hz completely. Not a big deal play connect the dots and the system can recreate that just fine....the problem is after you re create that and go back through a analog to digital converter all the missed frequencies that were a part of the original sample are going to be created as something else...so electrically it's not an exact representation of the original sample it's now a new copy of it with its own new "excel file" or impulse response ~if you will. 

Now times that across the entire spectrum and things get messy... the large values , or the volumes that have the highest volumes will have the best chance of an exact copy because the numbers have to equal that value to make that frequency..so on an Oscilloscope clean some waves will look the same...on music which has many many sinewaves intertwined the small details or very small changes can have different outcomes. And even very very good analog to digital converters will have some degree of change to the original sample..

Now that's just the analog to digital section, go back to the very beginning, the original digital file, the 1st DAC would have to make a perfect representation of the numbers to even make that senecio possible, dacs are attached to many other components in the output chain to make the pre-amp signal to your RCA cords. There's capacitors in the output of the dac that block DC from passing to your RCas , no capacitor does a perfect job at this , than there's op-amps that actually turn the ever so faint analog signal into a pre-amp signal that your amp can use...depending on different aspects of the op-amp will also change the outcome, some op-amps have different slew rates and speeds that make the audio signal have a distinct sound. That sound is a change to the original sample that was a at the very very beginning coming from a analog to digital converter when your music was recorded (unless you listen purely to synth pop with no vocals...you 80s butt rocker!) okay anyway...so that ADC had to make a perfect copy of the musitian and we now know it's not a perfect copy so the guy in the control room adds different effects and crap to try to mask it's deficiencies ..

So now you have a digital file that's not an exact copy of the musitian, now you want to rumor through a DAC through caps , op-amps and other electronics than send it back into a ADC.....and than expect it to come out the other side just as good as the analog signal that it went into....crazy right? Yeah it's pretty crazy

On the contrary tho, there's some pretty dam good DACs and ADCs out there that crunch all that back into a sound that is so close to the original most people can't even tell...except if your system is good enough to hear those differences than it raises a problem....and you have to have a pretty dam good system to hear those ever so small details..

To me , those differences sound robotic or kinda like a crappy 96kbps Sirius xm audio broadcast claiming to be cd quality..it's that bad...

So it's a fantastic idea to NOT have that 2nd ADC conversion. You will largely have a much more authentic version of what the musicians analog signal was by half or more depending of the quality of conversion...

We now have ADCs and DACs that run at speed much higher than the recording was originally recorded making things much much much better, but even at double or triple sample rate you will never get away with the problems in the analog side like op-amp sounds or DC at the output or any electrical abnormalities that that circuit will surly induce..largely NOISE. 


So if you have a way to only go Digital to analog once. Do it. I'm not saying everyone who does two conversions stereos are bad, not in any way am I implying that at all, after all so so so many changes from speakers to amps and rooms and all kinds of things will also affect the analog or acoustical waves, after all music really is numbers. But if it can be avoided kinda thing...sorta like , the thickness of cables is of ZERO concern as you expressed concern... it's not about proffered cables it's about keeping the music sounding the way it was meant to sound.


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## lurch (Jan 20, 2014)

WOW !
nerd out indeed !
thanks for that oabeieo.


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## t3sn4f2 (Jan 3, 2007)

Just to add to what oabeieo mentioned........There's also jitter to worry about when going digital from one component to the other. Well I mean if you are one to worry about an extra high quality D to A A to D conversion then you most certainly should be worried about jitter. 

Jitter is the timing errors (ie fluctuations) between digital sources that occur due to the physical separation between the clocks as well as the inherent timing differences of the clocks themselves. As the samples are ticked off at the source they aren't sampled at precisely that same point in time on the receiving end, that cause phase (ie timing) distortions.

It manifest itself as added noise to the already present noise floor. Which comes to be at worst about the same amount as added noise from that additional D to A A to D.

One way to eliminate it completely is to do that conversion to the analog domain. That way the signal is resolved completely and then resampled. eliminating any chance of timing errors. 

Do I worry about any of the two in a competent component set when installed correctly? Not at all. That being said I would go digital over analog if I had the chance to and there weren't any annoying complications that come from it.


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## Dan750iL (Jan 16, 2016)

Okay there are some awesome reads here. Thanks for all of the input. I really do appreciate the education!

I went ahead and ordered the HU last night and am going to run the optical cable back to the DSP.


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## BlackHHR (May 12, 2013)

Optical is only going to give you (1) left and (1) right signal. No front, rear and sub.


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## oabeieo (Feb 22, 2015)

Thanks for filling in that very large blank t3snf42 

Yeah that's exactly right, jitter and clock speed can be problematic in the digital realm 
Especially when trying to make two digital devices talk to each other through a optical or co-axial cable (or i2s or aeu whatever) that sync between devices absolutely will cause those issues described.

I was actually reading about timing errors on a single dac and how it translates into math errors in the process.... it blows my mind how far man has come and who invented this ****.


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## SSSnake (Mar 8, 2007)

Ummm, I have to disagree with most of the responses in this post.

First,


> a sample is basically a impulse response meaning a big chart of all the frequencies we hear in a file.


 is not correct and I can't seem to figure out a way that would even make this seem partially correct. A sample is nothing more than a number. The number represents the amplitude of the sampled signal. It is NOT a large group of numbers or anything remotely similar. Yes, it takes more than one sample to reproduce any meaningful audio signal.



> Point is the DAC will convert a TON of numbers to electrical signals


 AGREED.



> depending on sample rate and bit depth some frequencies may be completely skipped


 I think you are trying to describe quantization error but I am not real sure. It is important to note that the D/A does not skip frequencies. There is a limit to the resolution a digital signal can provide based upon the bit depth (assuming the sample frequency is high enough to capture the initial waveform accurately) but that isn't the same as skipping frequencies. 

Rather than going through the entire discussion I would suggest a review of this paper http://www.diva-portal.org/smash/get/diva2:395/FULLTEXT02.pdf Chapter 2, beginning on page 7, has a good concise discussion of sampling and reconstruction.

I say all of this to make one point. Do additional D/A and A/D conversions degrade the signal quality. Yes. Is the degradation audible in a moving car? No. Is it audible in a competition system while parked? Yes, but likely only by a trained ear. IMO the real advantage to TOSLINK interfaces is the noise immunity. IME fiber connections are TYPICALLY audibly better than copper even while driving. Differential copper connections are TYPICALLY better than single ended copper even while driving. Now can you make these copper connections very high SNR? Yes. Can you reduce the noise in copper so that you can't hear the difference between copper and fiber? Yes, but it will usually require some fairly unique approach such as adjusting the gain in the final gain stage (the power amp) vice using volume adjustment in the HU.

Again not trying to attack anyone just trying to provide accurate information.


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## oabeieo (Feb 22, 2015)

SSSnake said:


> Ummm, I have to disagree with most of the responses in this post.
> 
> First, is not correct and I can't seem to figure out a way that would even make this seem partially correct. A sample is nothing more than a number. The number represents the amplitude of the sampled signal. It is NOT a large group of numbers or anything remotely similar. Yes, it takes more than one sample to reproduce any meaningful audio signal.
> 
> ...


Your right, and I explicitly said "basicly" 
I didn't want to go too in depth for what the point I was trying to make, 
Sorry for the confusion...

I figured there would be at least one person to knit pick the details and disregard the over arching theme. But yes your correct , and that is why I kept my language so vague for time sakes to make a point simply because I was typing on a iPhone and that can be tedious  
And to clairify its many many samples that make a impule responce . Minus aliasing 
y(T)->Ty(nT)->y 
Pretty sure that will sample a Ir , however an IR could be anything, hell it could be a complete audio track if one wanted to use it as such...so your right a distinction must be made...


"Is degradation audiable in a moving car? No"

I either disagree or you have a pretty amazing ADC, but we both know it takes a long time of careful listening to be able to hear that degradation at all. So eeh , I'll give that a maybe ...the ADC I use is considered good and I can hear it ...moving . 



imagur


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## Kazuhiro (Apr 28, 2015)

BlackHHR said:


> Optical is only going to give you (1) left and (1) right signal. No front, rear and sub.


Doesn't optical support full dolby 5.1 7.1?

Aside - I've been considering going digital to dsp, via headunit modification. 
Just unsure if the benefits outweigh the efforts.


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## t3sn4f2 (Jan 3, 2007)

Kazuhiro said:


> *Doesn't optical support full dolby 5.1 7.1?*
> 
> Aside - I've been considering going digital to dsp, via headunit modification.
> Just unsure if the benefits outweigh the efforts.


Yes, only multichannel media though. Not for stereo, downmixed, or processed outputs.


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## BlackHHR (May 12, 2013)

Kazuhiro said:


> Doesn't optical support full dolby 5.1 7.1?


Not coming out of the Clarion head unit. You will only get (1) left and (1) right stereo pair. 

Your processor will then work with the 2 channel signal and route the signal as you have programmed it to do so.


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## Kazuhiro (Apr 28, 2015)

BlackHHR said:


> Kazuhiro said:
> 
> 
> > Doesn't optical support full dolby 5.1 7.1?
> ...


Didn't realise there was a new clarion unit. The old clarion dts processors used full 5.1 over optical so I assumed the new units would too


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## Rrrrolla (Nov 13, 2008)

oabeieo said:


> Your right, and I explicitly said "basicly"
> I didn't want to go too in depth for what the point I was trying to make,
> Sorry for the confusion...
> 
> ...


An A to D converter is pretty simple. It looks at the signal voltage and outputs a number. That's it. It does it 44,000 times a second if that's what your sample rate is. 

I don't understand why you're talking about impulse responses and information in the frequency domain... A to D and D to A don't do any complicated math like that...


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## Babs (Jul 6, 2007)

Without going into the science that's already been eluded to on the subject, my opinion is now full circle on this. 

With analog, yes, you have that extra AD then DA conversion in the DSP instead of remaining digital from source to DSP output DAC. However, with strong crispy clean source signal, preferably volume controlled at DSP so always sending the DSP ADC cleanest full tilt, you suffer negligible and inaudible degradation. 

Talking to Subterfuse recently on this subject, as he just returned from super cool Systune training, he had some discussion with some pretty good pro guys there. The take away on this subject stated what I mention above. Sending the processor good, strong and clean analog signal, makes the extra AD/DA inconsequential.

I tested this kind of haphazardly... 
iPhone - HDMI - Toslink - Helix DSP
verses...
iPhone - 99RS - analog - Helix DSP

The head unit analog actually spanks the HDMI method, which rather surprised me. The previous 80PRS vs HDMI was much closer to same fidelity. So the question becomes why isn't the toslink run all that and a bucket of chicken awesome sauced? 

I suspect there's a bunch of factors involved though. One main suspect being the AV adapter with a board in it doing some mysterious airport/airplay apple voodoo chicanery. Another suspect is possibly poor clocking either at adapter or toslink converter. Points being just cuz it's toslink, don't make it all essqueey gooey good.  A bunch of junk in the signal chain is still never good recipe, anny or digy.

The comparison and a few days of quiet basement listening has convinced me I'm done.. Don't need to go digital. Makes me pretty much a candidate customer for only two head unit models on the planet currently available then. Okidokie.. 

All that said, from a cabling convenience and noise prevention perspective, I'm all for a single digital signal if done well. I'll repeat what I've heard others say. They need to embrace HDMI from source to DSP like the rest of the world.


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## Dan750iL (Jan 16, 2016)

Babs said:


> ...I'll repeat what I've heard others say. They need to embrace HDMI from source to DSP like the rest of the world.


That would be awesome. Still a single cable and a little more durable than an optical.


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## oabeieo (Feb 22, 2015)

Rrrrolla said:


> An A to D converter is pretty simple. It looks at the signal voltage and outputs a number. That's it. It does it 44,000 times a second if that's what your sample rate is.
> 
> I don't understand why you're talking about impulse responses and information in the frequency domain... A to D and D to A don't do any complicated math like that...


 Because it's all relevant , you can't have a digital to analog converter without some sort of output stage at least I've never seen one .... every output stage as a whole including the digital to analog converter has a certain sound to it ...
If you gonna take that sound as a whole the whole thing and turn it back to digital you're also going to turn back the sound of the noise and problems in the da as an inherent sound all together, Then you're going to have the output stage again in the second time around and it's going to have his own distinct sound and problems with extra emphasis on problems and combined theres Signal degradation . The ADC by it self Shirley does just fine . So I'm not sure where the breakdown is in understanding the concept ...

No the A.D. And D.A. Don't participate in impulse manipulation but I guarantee the processor in between does . That's what we're doing right ? Who in their ****ing right mind goes from digital to analog back to digital and then back to analog just for fun ? There's always a purpose right ? So in the IIR and FIR Side of things the impule matters. And it has its role in the entire signal chain ... even if you didn't modify anything in IIr or Fir an impule has a sample rate. I was referring to an impule responce as an excel file because you can open ia impule in excel and see how different impulses with different sample rates have different frequency charts, I don't know how I could describe sample rate any other way except use an example that is something that is at the sample level, we wouldn't be talking about opening samples in excel :/ now we could open a MP3 in sample level but it would be chaotic, while an ir is clean in frequency and makes sence because how we make our Ir's is one continuous sweep in frequency...is it a direct correlation to what we're talking about?...probably not the biggest, but the point I made wasn't strictly talking about frequency domain I was talking about Da ad . 

It sounds like you don't understand what an impulse response is. It's not just a measurement that you can take in your car , it can be Many more things than a measurement, The measurement is called a impulse response because we are breaking it down into mathematics, that just so happen to be at the sample level . And guess what, sample rate MATTERS at that point silly  otherwise it would just be called a measurement, my wiener could have a impule responce when I see a hot chick and I could measure its growth .  



Wikipedia/impulse responce said:


> In all these cases, the dynamic system and its impulse response may be actual physical objects, or may be mathematical systems of equations describing such objects.


https://en.m.wikipedia.org/wiki/Impulse_response


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## oabeieo (Feb 22, 2015)

And Babs is right except the problem is most people control the volume upstream  

But it is inconsequential to a large large degree but going back to the very very first post if you have the opportunity to skip a conversion it is a tiny bit better ....I'm just sayin not a whole lot of people can tell the difference if you can great ,*If you can't that's fantastic go for it !* Personally in my car I can tell . It is a very very very very very very very very very very very very very very very very very very very very very very very small difference


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## SSSnake (Mar 8, 2007)

> I figured there would be at least one person to knit pick the details and disregard the over arching theme.


I wasn't trying to "nitpick" but rather convey an accurate message. Rrrrolla's statement



> An A to D converter is pretty simple. It looks at the signal voltage and outputs a number.


is much simpler and correct. This is why I am not sure why the impulse response discussion even entered the conversation. If your point is the subsequent processing is dependent upon the input signal, we can agree but the processing doesn't necessarily exacerbate the problem. This is also true with analog "processors" so I don't see where you are going.



> They need to embrace HDMI from source to DSP like the rest of the world.


This I would love to see. Even most of the PC sound cards don't do HDMI (likely because it is in the video card most often). A single cable that provides digital (and therefore noise resistant), high fidelity, multi channel audio and video is tough to beat.


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## SSSnake (Mar 8, 2007)

> Can you reduce the noise in copper so that you can't hear the difference between copper and fiber? Yes, but it will usually require some fairly unique approach such as adjusting the gain in the final gain stage (the Power amp) vice using volume adjustment in the HU.





> Without going into the science that's already been eluded to on the subject, my opinion is now full circle on this.
> 
> With analog, yes, you have that extra AD then DA conversion in the DSP instead of remaining digital from source to DSP output DAC. However, with strong crispy clean source signal, preferably volume controlled at DSP so always sending the DSP ADC cleanest full tilt, you suffer negligible and inaudible degradation.





> But it is inconsequential to a large large degree but going back to the very very first post if you have the opportunity to skip a conversion it is a tiny bit better ....I'm just sayin not a whole lot of people can tell the difference if you can great ,If you can't that's fantastic go for it ! Personally in my Car I can tell . It is a very very very very very very very very very very very very very very very very very very very very very very very small difference



^^^^sounds like a WHOLE lot of agreement.


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## t3sn4f2 (Jan 3, 2007)

Babs said:


> Without going into the science that's already been eluded to on the subject, my opinion is now full circle on this.
> 
> With analog, yes, you have that extra AD then DA conversion in the DSP instead of remaining digital from source to DSP output DAC. However, with strong crispy clean source signal, preferably volume controlled at DSP so always sending the DSP ADC cleanest full tilt, you suffer negligible and inaudible degradation.
> 
> ...


The AV adapter also does a sample rate conversion to the HDMI standard 48kHz. I wonder just how good of a converter they are using.


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## Babs (Jul 6, 2007)

t3sn4f2 said:


> The AV adapter also does a sample rate conversion to the HDMI standard 48kHz. I wonder just how good of a converter they are using.



Hmm. That right there in and of itself makes me say nope to the AV adapter method. I wonder if the Apple camera adapter does a similar conversion for USB?


Sent from my iPad using Tapatalk


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## grinkeeper (Jun 26, 2015)

oabeieo really got the ball rolling here, I learned a thing or three. Thank you


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## subterFUSE (Sep 21, 2009)

Optical cables are probably the worst of the digital connection types on the market. The cables themselves are delicate, and they are also limited to 96kHz. If you want to use higher resolution music than 96k, you'll need a coaxial digital or other connection.

With analog, as long as you have a good quality DAC then you'll be able to play the highest resolution files and have a nice, clean signal coming out. The new Sony head unit can ONLY playback DSD files over the analog outputs. It has an optical out, but that is limited to 96k due to the nature of optical cables.

Jitter is also a potential issue when going from digital to digital. If the sample rates are different there can be issues.


I was really surprised when the instructors at my SysTune training were so adamant about going analog into a DSP. A lot of my previously held notions of digital always being better, and fewer DA conversions being better, were pretty much shattered. But they said that if you have a good quality DAC, put that DAC in front of your DSP. I was really shocked because I would have thought it better to put the high quality DAC after the DSP, but they both said no. Put it in front.


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## AyOne (Sep 24, 2016)

subterFUSE said:


> I was really surprised when the instructors at my SysTune training were so adamant about going analog into a DSP. A lot of my previously held notions of digital always being better, and fewer DA conversions being better, were pretty much shattered. But they said that if you have a good quality DAC, put that DAC in front of your DSP. I was really shocked because I would have thought it better to put the high quality DAC after the DSP, but they both said no. Put it in front.


So when you say "a good quality DAC" are you referring to the head unit or adding another one to the signal path?


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## subterFUSE (Sep 21, 2009)

AyOne said:


> So when you say "a good quality DAC" are you referring to the head unit or adding another one to the signal path?




I'm speaking in a general sense.

If you have a high end head unit like the Sony GS9, then you want to go analog into the DSP. Take advantage of the high quality DAC by putting the good conversion before the DSP.

If you don't have a head unit, but instead use an outboard DAC like an iStreamer then you want to put that in front of the DSP.


I know what you are thinking... it's completely opposite what everyone has always assumed. I had the same reaction at first.


I'm installing a Sony deck in my car as we speak. One of the things I plan to test is a blind audition between analog vs digital connection for it.


Sent from my iPhone using Tapatalk


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## ErinH (Feb 14, 2007)

It used to boil down simply to: optical (toslink) is more impervious to noise as opposed to analog. This was useful if you required a long signal run from headunit to dsp/amp(s). Less of a concern if the runs were short. That was practically the sole reason people ran digital out of their headunits (and went through the trouble to mod some headunits for digital output). 

Now that components seem to be better with higher SNR/dynamic range specs and we as a community tend to have a better sense of building a system to be more immune to those noise issues, the area of concern boils down to sampling rate compatibility and jitter as John covered above. I've noticed with the home audio crowd that a lot more people are going *back* to analog output from their source for these reasons.


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## cmusic (Nov 16, 2006)

Back in the early '00s when Nick Wingate was competing with his black WV Bettle, he said running analog into his Alpine F#1 PXA-H900 sounded better than using the optical inputs. I think Nick told me he told the guys at Alpine (Steve Brown and Chris Yato) about the analog input, and when they did their listening tests, they agreed with Nick that the analog input sounded better. Nick was using a Denon head unit but I hooked my F#1 CDA-7990 head unit to my H900 via twisted pair RCA cables and it sounded just as good or perhaps just slightly better than the optical inputs.


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## ErinH (Feb 14, 2007)

subterFUSE said:


> I was really surprised when the instructors at my SysTune training were so adamant about going analog into a DSP. A lot of my previously held notions of digital always being better, and fewer DA conversions being better, were pretty much shattered. But they said that if you have a good quality DAC, put that DAC in front of your DSP. I was really shocked because I would have thought it better to put the high quality DAC after the DSP, but they both said no. Put it in front.


Interesting. Did they give rationale for that? I'm curious of the 'science' behind that... or if it is more of a "no real sonic advantage with digital, and better interface with analog" type logic (since, DVC _can _be a PITA).


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## subterFUSE (Sep 21, 2009)

ErinH said:


> Interesting. Did they give rationale for that? I'm curious of the 'science' behind that... or if it is more of a "no real sonic advantage with digital, and better interface with analog" type logic (since, DVC _can _be a PITA).



I wish we could have spent more time talking about the subject, but we were in a class setting and it would have hijacked the time. But the general points made were:

Digital to Digital connections are not always the best because of the issues we have covered above. Sample frequency mismatches, clock sync, jitter, bandwidth limitations, etc... All of these can make the connection flaky or problematic. Taking a 192k digital signal and converting it to 96 or 48 in the digital realm is not necessarily better than allowing a good DAC and then a good ADC to do their thing.

The sound degradation that occurs in a DSP is actually quite minimal, assuming it is of good quality itself. Of course, these guys are most familiar with using pro-audio DSP units that cost about $6000 a piece and they might have a rack with 8 of them.  Their idea of quality processing and our idea might be on a different plane.


Another thing which they did not bring up, but which is possibly true, is that in the pro-audio realm they are dealing with a much more complicated string of equipment than we are. Instruments and microphones feeding a mixing board, in turn feeding a rack of DSPs, and a rack of amplifiers. (Actually, most of the pro-audio amps have DSPs integrated into them these days) It might simply by that when trying to interface so much gear together, analog presents a more stable solution across the devices. Then again, most pro-audio gear uses industry-standard communication protocols like Dante where they can communicate over Ethernet.

From my personal experience as a DJ, I can say that the best nightclub sound systems in the world have generally preferred analog outputs from the DJ mixer into the house mixer. Twilo in New York, which had possibly the best club sound system that ever existed in the world, had an old, vintage Urei analog mixer from the 1970s even though the sound system was state of the art and used lots of digital processing.


I might shoot Bruce an email to get more details from him.


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## Shapin (Jun 23, 2015)

What about BT vs optical?
Did something like AMAS by mosconi will have a good signal like optical cable to dsp? (same source)


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## subterFUSE (Sep 21, 2009)

Bluetooth is a lossy compressed transmission format. It is not as good as optical or coax digital.


Sent from my iPhone using Tapatalk


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## bertholomey (Dec 27, 2007)

Good info here - looking forward to reading more.


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## Kazuhiro (Apr 28, 2015)

Sorry to throw this in here, but while on the topic; 
Car length optical from tablet to dsp, or wifi airplay->optical to dsp. 

Thanks to whoever 
I'm sure in the end it won't really matter in a car.


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## Babs (Jul 6, 2007)

Kazuhiro said:


> Sorry to throw this in here, but while on the topic;
> Car length optical from tablet to dsp, or wifi airplay->optical to dsp.
> 
> Thanks to whoever
> I'm sure in the end it won't really matter in a car.


I've done the wifi airport express.. Wired optical from device would be better. And it will matter in a car, IMHO.  Plus the airport wifi is wonky in a car.


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## Kazuhiro (Apr 28, 2015)

Well you just saved me some money thank you 

Yall know if the $30 turtle beach adapter is worthy of the task?


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## Shapin (Jun 23, 2015)

So stick to tablet>optical>dsp
I wanted to do tablet>BT>amas>dsp


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## Babs (Jul 6, 2007)

Kazuhiro said:


> Well you just saved me some money thank you
> 
> Yall know if the $30 turtle beach adapter is worthy of the task?


On methods or rather what gear to use to get there, there are a few threads in here on the topic and one that keeps popping up is iFi. iLink or something like that. Found it.. iLink You're not going to like the price.  

But to make a point, in order to be on topic, that you can do many super $30 cheap methods to run digital between A and B. Some might introduce all kinds of aforementioned compromises to the signal such as mediocre clocking, sample rate conversions etc.


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## Kazuhiro (Apr 28, 2015)

Ouch 

The other option I had in mind is the popular x-mos chip, one notable model is the SMSL X-USB which seems to be 89.99 everywhere.

SMSL X-USB DAC Digital Audio Interface with DSD Support


Another thing, will any of this work on standard android 7.0 devices?


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## Velozity (Jul 6, 2007)

t3sn4f2 said:


> The AV adapter also does a sample rate conversion to the HDMI standard 48kHz. I wonder just how good of a converter they are using.


Actually HDMI 2.0b goes up to 1536kHz. The converter must not be capable of higher sample rates.




Babs said:


> Hmm. That right there in and of itself makes me say nope to the AV adapter method. I wonder if the Apple camera adapter does a similar conversion for USB?
> Sent from my iPad using Tapatalk



Nope, the camera kit doesn't limit to 48kHz. I've tested USB at least up to 192kHz.





Kazuhiro said:


> Sorry to throw this in here, but while on the topic;
> Car length optical from tablet to dsp, or wifi airplay->optical to dsp.
> 
> Thanks to whoever
> I'm sure in the end it won't really matter in a car.



Depends on the quality of components used at the transmitter and receiver. When you say tablet to dsp, is that direct over USB? To my knowledge there are no DSPs out yet (Zapco and Helix pending) that accept direct input of a USB source. If you're talking about using one long optical cable vs. airport express plus a shorter optical cable, it may just come down to what's more convenient for your install. Both wired and wireless methods take the I2S digital signal and convert it to a transmission medium and back again (either via RF or USB to S/PDIF). In general though the fewer conversions and connections the better.




Babs said:


> On methods or rather what gear to use to get there, there are a few threads in here on the topic and one that keeps popping up is iFi. iLink or something like that. Found it.. iLink You're not going to like the price.
> 
> But to make a point, in order to be on topic, that you can do many super $30 cheap methods to run digital between A and B. Some might introduce all kinds of aforementioned compromises to the signal such as mediocre clocking, sample rate conversions etc.


I use the iFi products. I use the iFi Nano iDSD in my car system and the iFi Micro iDSD Black Label for headphone listening. Their stuff is hard to beat at any price. But you can certainly find cheaper options.


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## SSSnake (Mar 8, 2007)

> Optical cables are probably the worst of the digital connection types on the market. The cables themselves are delicate, and they are also limited to 96kHz. If you want to use higher resolution music than 96k, you'll need a coaxial digital or other connection.


A couple of points here. You can find absolutely bulletproof optical cables but they are definitely few and far between and expensive (I haven't looked at these in a while but someone like Canare used to have a kevlar jacketed cable that was nearly indestructible. You could tie it into a not and it would still work.) You do have to protect the ends no matter what so definitely delicate from that standpoint. While the toslink implementation is bandwidth limited, the fiber itself is not the limiting factor (other standards could be developed using fiber that are not as bandwidth limited - however electrical digital seems to be the way of the future for anything but large data transmission lines). In most cases you will be using two channel source data in a car anyway...



> Of course, these guys are most familiar with using pro-audio DSP units that cost about $6000 a piece and they might have a rack with 8 of them. Their idea of quality processing and our idea might be on a different plane.


This is a completely different environment. The big problem with noise in this environment is usually ground loop or something similar (lighting and other effects can generate some interesting noise issues as well). Pro audio doesn't have to deal with an extremely noisy power supply (usually) like the car audio stuff. Having the alternator emitting within a couple of feet of an analog source is tough to overcome from a noise standpoint.



> From my personal experience as a DJ, I can say that the best nightclub sound systems in the world have generally preferred analog outputs from the DJ mixer into the house mixer. Twilo in New York, which had possibly the best club sound system that ever existed in the world, had an old, vintage Urei analog mixer from the 1970s even though the sound system was state of the art and used lots of digital processing.


Keep in mind that in live sound you are not so much trying to faithfully reproduce the original content but develop an appealing sound for the artist in that venue. There are some venues where the venue has its own "signature" sound that has developed over the years.



> Now that components seem to be better with higher SNR/dynamic range specs and we as a community tend to have a better sense of building a system to be more immune to those noise issues, the area of concern boils down to sampling rate compatibility and jitter as John covered above. I've noticed with the home audio crowd that a lot more people are going *back* to analog output from their source for these reasons.


Says the guy with his volume control AT THE AMPS  Things have definitely gotten better but it is tough to get away from the analog noise floor if you actually output a varying voltage analog signal from the HU (analog volume control in the HU). IMO sampling rate compatibility is one of those things you consider from a design standpoint and design around (which means it shouldn't be a noise source in a properly designed system). The multitude of sources we have available today can make that difficult but again IMO it is part of the system design process. On the home audio crowd going back to analog output from their sources. I have seen this trend as well but I have also seen a lot of people incorporating turn tables back into their systems as well. I have been all digital in for quite some time and can't see myself going back.

Erin you have a horn rig at home. IME these are some of the most sensitive to noise. In the past (it has been a while) when I would do an A/B comparison on my horn system there was a noticeable difference in the noise floor between inputs with analog outs and digital outs. More subtle was the difference in the noise above the floor. In my system the edge went to the digital inputs there as well. I will say that it was close enough that I would have been ok with either, however, the noise floor edge with digital made it the implementation of choice.

This has been my experience and I will say it is somewhat dated as I have been out of the hobby for a while. I am definitely open to discussion so please don't take this post as argumentative as is not intended that way (I am just naturally a jerk  )


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## oabeieo (Feb 22, 2015)

Maybe sssnaky can explain this;
1st off I would like to make a point about most people using volume upstream and preprocessing upstream before the first DAC , that's where i hear the most amount of problems in ADC or the whole process of it .

If I apply EQ to the upstream source as I'm applying EQ I can definitely hear that it doesn't sound as good as that same exact EQ applied if I took the DSP out of the loop . 

I could set the DSP to have zero effect and basically be a pass-through ADC with unity gain , use EQ in the deck and when I do the A.B. between the two I can definitely absolutely without a doubt tell there is something that makes it sound not very good compared.

I'll let you explain 
Cheers


Ps I'm a jerk too


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## hella356 (Dec 11, 2016)

So, is the idea of using the Clarion NX706 with a DSP via Toslink not a good one? It would appear, on the surface, to be ideal. Being limited to 24bit/96kHz isn't a deal killer for me, and my assumption has been that essentially bypassing the D/A inside the deck and passing the audio signal to a higher quality external DSP would be a superior solution to using analog outputs into a DSP. But from reading this thread, it sounds like going with the digital-out is more limiting, functionally, than the RCA outputs. This is crucial information to me, as I recently (with the exception my Rainbow ProFi front speakers) had my entire system stolen, and have been trying to map out my best path towards a fully new system. I figured the NX706 + DSP should be clearly better than using a 4100/4200NEX + DSP, but now I'm thoroughly confused!  

I'm ultimately looking to go with either a 2-way active front stage + coax rear + sub, or 3-way active front stage + sub. But if there are limitations of the Toslink connection in terms of how the system is controlled, perhaps I'll stick with a cheaper NEX unit. Any thoughts?


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## subterFUSE (Sep 21, 2009)

Cars are WAY easier to control noise than a home or live venue. 60 Hz noise is a huge problem. Recording studios have to get special electricians to wire up their buildings to prevent noise problems. It's such a pain in the ass.


Sent from my iPhone using Tapatalk


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## oabeieo (Feb 22, 2015)

hella356 said:


> So, is the idea of using the Clarion NX706 with a DSP via Toslink not a good one? It would appear, on the surface, to be ideal. Being limited to 24bit/96kHz isn't a deal killer for me, and my assumption has been that essentially bypassing the D/A inside the deck and passing the audio signal to a higher quality external DSP would be a superior solution to using analog outputs into a DSP. But from reading this thread, it sounds like going with the digital-out is more limiting, functionally, than the RCA outputs. This is crucial information to me, as I recently (with the exception my Rainbow ProFi front speakers) had my entire system stolen, and have been trying to map out my best path towards a fully new system. I figured the NX706 + DSP should be clearly better than using a 4100/4200NEX + DSP, but now I'm thoroughly confused!
> 
> I'm ultimately looking to go with either a 2-way active front stage + coax rear + sub, or 3-way active front stage + sub. But if there are limitations of the Toslink connection in terms of how the system is controlled, perhaps I'll stick with a cheaper NEX unit. Any thoughts?



^^^reguarding this 

Who has that video of that recording engineer that makes the case that anything more than 44.1 is damn near useless, as far as format goes not dsp sample rate


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## ErinH (Feb 14, 2007)

SSSnake said:


> Says the guy with his volume control AT THE AMPS


Not anymore. 



SSSnake said:


> Erin you have a horn rig at home. IME these are some of the most sensitive to noise. In the past (it has been a while) when I would do an A/B comparison on my horn system there was a noticeable difference in the noise floor between inputs with analog outs and digital outs. More subtle was the difference in the noise above the floor. In my system the edge went to the digital inputs there as well. I will say that it was close enough that I would have been ok with either, however, the noise floor edge with digital made it the implementation of choice.


Totally. There's always been a noise floor in my home system with the horns. I could turn the gain down but that just lowers the output capability. I'm running Oppo in to my AVR which then sends the signal to the external amps which are used to actively cross the 15" woofer and the CD. I've also gone from the Oppo straight to the amps (analog) and used the setup with the Rane DSP. Always a noise floor. Sucks, but unless I pay for some really nice amps with a super low noise floor, it's going to be an issue. And even then, the issue may not be completely resolved. 

How are you running your horns via a digital signal? Are they passive off an AVR or active off a separate amp/dsp?


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## felix509 (Dec 17, 2006)

oabeieo said:


> ^^^reguarding this
> 
> Who has that video of that recording engineer that makes the case that anything more than 44.1 is damn near useless, as far as format goes not dsp sample rate


It seems There are a whole lot of professional audio guys that will tell you the same. With the big push for HiRES audio right now, there is alot more info on why it really is not necessary. It seems that 24bit/48KHz may be the ideal, but it really depends on whom you want to believe.....

Being the nerd that I am I went all in trying to get 24bit/192KHz, but have recently decided that I will aim for 24/48 as my target bit depth/sampling rate..

We could start a thread with all of the links to the different views on HiRES audio..


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## nineball76 (Mar 13, 2010)

felix509 said:


> It seems There are a whole lot of professional audio guys that will tell you the same. With the big push for HiRES audio right now, there is alot more info on why it really is not necessary. It seems that 24bit/48KHz may be the ideal, but it really depends on whom you want to believe.....
> 
> Being the nerd that I am I went all in trying to get 24bit/192KHz, but have recently decided that I will aim for 24/48 as my target bit depth/sampling rate..
> 
> We could start a thread with all of the links to the different views on HiRES audio..


Same for me. I went hard in the 24/192 realm for a little while, later decided I didn't need it. Converting all down to 24/48 and 16/48 for iPod usage.


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## SSSnake (Mar 8, 2007)

> Totally. There's always been a noise floor in my home system with the horns. I could turn the gain down but that just lowers the output capability. I'm running Oppo in to my AVR which then sends the signal to the external amps which are used to actively cross the 15" woofer and the CD. I've also gone from the Oppo straight to the amps (analog) and used the setup with the Rane DSP. Always a noise floor. Sucks, but unless I pay for some really nice amps with a super low noise floor, it's going to be an issue. And even then, the issue may not be completely resolved.
> 
> How are you running your horns via a digital signal? Are they passive off an AVR or active off a separate amp/dsp?


I was afraid my generalization would be too vague - my bad. First, to be accurate I am not now running horns although I am migrating back to them. The WAF became problematic but I now have a method to make her happy as well. When I was running the horns I did quite a bit of evaluation trying to get the last little bits of noise out of the system. The best setup ended up as digital output from source to AVR, then digital from the AVR/preamp, to my processor that provided and active xover and equalization (I tried several), then differential analog out to a class A amp on the horn section with an AB on the bass unit. You could definitely tell when when you went analog out of the source into the AVR/preamp and it was a change for the worse.



> Cars are WAY easier to control noise than a home or live venue. 60 Hz noise is a huge problem. Recording studios have to get special electricians to Wire up their buildings to prevent noise problems. It's such a pain in the ass


Agree to disagree on this one. 60 hz is definitely a problem but it is a specific freq which makes filtering easy. I have implemented several. Filtering car power sources is, IMO, more difficult because of the spread spectrum nature AND the current requirements. Big lowpass filters that don't saturate with current are large and expensive. 

The worst case I got into was in the Pentagon. It is a national historic landmark and therefore the outside appearance can't be changed without LOTS of red tape. Since the outside can't be changed much we couldn't bring in fresh power distribution. Therefore, you are stuck with 70 yr old base systems that have been hacked and patched for decades. Once we got the 60hz stuff cleaned up with our standard solutions we began systematically chasing about 2 dozen other sources coming in on different inputs. Some were ground loops, some were emi from crossing power lines, ballasts, and other such nonsense but in the end it got done.

Maybe I just haven't seen enough of the fixed venue issues but all so far have been a lot easier to deal with than noise in an automotive environment (PS I don't consider turning the gain down to be a solution here - I know a LOT of people do). Now if you expand this to outdoor festivals and such I COMPLETELY AGREE. Those things can be nightmares.



> Maybe sssnaky can explain this;
> 1st off I would like to make a point about most people using volume upstream and preprocessing upstream before the first DAC , that's where i hear the most amount of problems in ADC or the whole process of it .
> 
> If I apply EQ to the upstream source as I'm applying EQ I can definitely hear that it doesn't sound as good as that same exact EQ applied if I took the DSP out of the loop .
> ...


You and I have obviously had very different experiences. I have not had problems with "upstream" equalization and do it frequently. As the numbers are just modified by an algorithm to produce a second set of numbers I am unsure how you induce any consistently negative effects. 

Think about this scenario. I have a PC with digital output to an AVR hooked to a pair of speakers. The AVR has a processing loop that lets you loop the analog output of the AVR through some processor then back into the AVR for amplification. The processor we put in the loop is a digital EQ (it takes in analog input converts it to digital and then back to analog again for amplification by the AVR).

With your statement if I preprocess the signal to some equalization curve and then feed it through the system it sounds worse than if I do the exact same equalization in the processor that is in the AVR's processor loop (If this is not what you are saying let me know). I personally have never experienced this.

So now take this example. The pre-processing is done to a digital audio file on your PC. You can use equations to apply an equalization curve and then save the file. Unless you did something wrong LIKE exceeded the magnitude afforded by the encoding schemes bit depth I am unsure how you would degrade the SQ. The actions are causal and repeatable so you can then apply the inverse of the eq curve and then compare the digital data files. Again, unless you did something wrong the two files should be identical. So how could it be possible that the upstream eq damaged the sq of the signal.

But lets take it further. I listen the the altered file with zero eq applied in the processor. Then I listen to the original un-eq'd file with the eq applied in the processor. It seems to me that your position is that the second scenario would sound appreciably better. If that is what you say you are experiencing then I think there must be something you are missing. IME jitter and other noise sources have never been of higher magnitude than the conversion to analog then re-sampling (which include the added noise found in the analog signal) and then converting back to digital. 

Think of it this way. We quite frequently make perfect copies of digital files. They match in all appreciable ways, shapes, forms, and fashions. We have never been able to make perfect copies of analog files. If you add more analog stages to any system the SNR is ALWAYS lower than the lowest SNR component in the system. Now if you put multiple digital stages in a system it is POSSIBLE to retain bit perfect copies of the data throughout the entire chain. Do you always end up with bit perfect copies? Nope - but it is possible. Can you poorly design/implement a digital transmission system so that you would have been better off with an analog transmission system? Sure - but it is much easier to design a high quality digital transmission system and an analog one.


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## oabeieo (Feb 22, 2015)

SSSnake said:


> I was afraid my generalization would be too vague - my bad. First, to be accurate I am not now running horns although I am migrating back to them. The WAF became problematic but I now have a method to make her happy as well. When I was running the horns I did quite a bit of evaluation trying to get the last little bits of noise out of the system. The best setup ended up as digital output from source to AVR, then digital from the AVR/preamp, to my processor that provided and active xover and equalization (I tried several), then differential analog out to a class A amp on the horn section with an AB on the bass unit. You could definitely tell when when you went analog out of the source into the AVR/preamp and it was a change for the worse.
> 
> 
> 
> ...


Very interesting, 

Maybe it's my equipment, 
That is a possibility,
Dexp99 going into minidsp 2x4HDs at 96k 
The p99 eq sounds good clean by itself 
Through the mini it's good ,just not as good 
Optical into mini same eq settings it's great no issues 


And I get what you're saying about modifying files in audacity or whatever program you use and then burning it to a disk or whatever media .
That would be the same as the CD unaltered 
What about the effects of the IIR eq in the pioneer that definitely plays a tiny little bit in the EQd sound , maybe it's more of a combination of all different components leaving their own little breadcrumb of problems that we discount as not big enough to make a difference , compounded. Ya know, if the processing upstream never went through a DAC and it would be less and of those little problems that all add up. 
Maybe it's the amplitude change along with phase changes that the ADC has to deal with now converting a signal that has problems in it I've heard a ADC wants a strong signal, what if the eqd portion causes the ADC to output in a funny way. Maybe it's my gain structure , input is as high as possible without any clipping and amps are all way down and I usually listen with the p99 all way up.(it won't clip lots eq cuts-nothing boosted) 
So I think there is a distinction that needs to be made for pre-processing, yeah altering the Music file itself vs. Built in dsp before dsp with an ADC in the middle.

And as usual , very interesting read, thanks for the time. I enjoy it a lot.


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## oabeieo (Feb 22, 2015)

Well sssnake 

I finally got it all installed. The spdif made a huge difference, I can absolutely hear more depth into recording. It seems like there's more ambiance but I think it's just being confused with plain old resolution. It's noticeably cleaner. When I was testing before I was using one of my minis in my house and going USB in. Now I have same system same dsp just toslink vs analog 

Again maybe it's just the ADC I was using because it's the first system I've been able to A/B
it could be my super sensitive HLcDs that give it all away, however I think for now I'm going to have to respectfully disagree. ADC And back IMO is detrimental to SQ. I'm pretty firm on this now and would take a pretty amazing ADC to convince me otherwise


So the part I said it's only very very very very very very very very very very very very very small difference. I withdraw that statement and would like to now say ; it's a pretty good upgrade. I won't say it's totally awesome diffrances just a pretty dang good upgrade for sure  and worth the hassle of using spdif


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