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CHORD M-Scaler Review (Upsampler)

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Blumlein 88

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Repeated for emphasis.

Using a filterless NOS DAC is broken by design. the world filterless implies this. The fact that you can circumvent the broken part by adding a filter and MUST upsample in order to make the filter undo the 'broken' aspect says it all.
 

voodooless

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Using a filterless NOS DAC is broken by design. the world filterless implies this. The fact that you can circumvent the broken part by adding a filter and MUST upsample in order to make the filter undo the 'broken' aspect says it all.
Adding filters and upsampling still leaves it broken. Nothing is undone. you'll just make it a little bit worse. The fact that ones assumes a sample value is valid for the complete sample period makes it broken. All else is collateral.

So can we please remove the "filterless"? NOS is broken by design, whatever implementation..
 

solderdude

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Adding filters and upsampling still leaves it broken. Nothing is undone. you'll just make it a little bit worse. The fact that ones assumes a sample value is valid for the complete sample period makes it broken. All else is collateral.

So can we please remove the "filterless"? NOS is broken by design, whatever implementation..

I don't agree, assuming bit steps are correct.
When the DAC in question is a filterless design that can handle 706kHz 16 bits then the quantization errors are very, very small and the post analog slow filter that is always present will do the last bit of smoothing the steps.
So... a Chord upsampler can definitely ensure a proper reconstruction filter is in place and a 'steps free' analog output waveform can be created from a filterless NOS DAC.
In essence the Chord + DAC no longer is filterless NOS but simply a properly filtered OS DAC.

Using a filterless NOS DAC as is, at sample frequencies below 88.2kHz, is silly. regardless if goldeneared (and less goldeneared) folks think it sounds fine or even 'better'.
Above that I don't think filterless NOS is very broken when considering music.

That said, the only NOS CDP's were first generation not Philips. After the first generation CDP all of them were at least 2 or 4x ovesampling. This was because it was difficult to make good enough analog reconstruction filters. So NOS has not really been used for over 30 years before some idiot decided to use 'old' chips without their digital filtering.
So in that light... yes all real NOS R2R DACs are broken when used without upsampling/filtering.
 
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Blumlein 88

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I don't agree, assuming bit steps are correct.
When the DAC in question is a filterless design that can handle 706kHz 16 bits then the quantization errors are very, very small and the post analog slow filter that is always present will do the last bit of smoothing the steps.
So... a Chord upsampler can definitely ensure a proper reconstruction filter is in place and a 'steps free' analog output waveform can be created from a filterless NOS DAC.
In essence the Chord + DAC no longer is filterless NOS but simply a properly filtered OS DAC.

Using a filterless NOS DAC as is, at sample frequencies below 88.2kHz, is silly. regardless if goldeneared (and less goldeneared) folks think it sounds fine or even 'better'.
Above that I don't think filterless NOS is very broken when considering music.
And at those higher rates your speakers and your hearing are the filters.
 

voodooless

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I don't agree, assuming bit steps are correct.
What does that mean,"correct bit steps"?
When the DAC in question is a filterless design that can handle 706kHz 16 bits then the quantization errors are very, very small and the post analog slow filter that is always present will do the last bit of smoothing the steps.
So... a Chord upsampler can definitely ensure a proper reconstruction filter is in place and a 'steps free' analog output waveform can be created from a filterless NOS DAC.
In essence the Chord + DAC no longer is filterless NOS but simply a properly filtered OS DAC.

Using a filterless NOS DAC as is, at sample frequencies below 88.2kHz, is silly. regardless if goldeneared (and less goldeneared) folks think it sounds fine or even 'better'.
Above that I don't think filterless NOS is very broken when considering music.
None of this means that it's no longer broken, even though you may get a halfway decent result.
 

solderdude

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What does that mean,"correct bit steps"?

Linearity, which can be an issue for R2R.

None of this means that it's no longer broken, even though you may get a halfway decent result.

The upsampler + NOS R2R DAC, providing linearity is good and 16 bit resolution can be had is not just halfway decent it can be as good as perfect.
Nothing even remotely broken about it, certainly when properly dithered.
 

voodooless

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Linearity, which can be an issue for R2R.
Thanks for the clarification
The upsampler + NOS R2R DAC, providing linearity is good and 16 bit resolution can be had is not just halfway decent it can be as good as perfect.
Nothing even remotely broken about it, certainly when properly dithered.
A good outcome doesn't mean it's still fundamentally broken. It's like using a hammer to drive in a screw. If you hit it hard and long enough, it works, and probably it will hold perfectly well. But it's still not a good idea.

But yes, if you would have a filterless NOS DAC, the best you can do is oversample to a high rate and still have an acceptable result. It won't give you any timing advantages though ;)
 
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solderdude

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. It's like using a hammer to drive in a screw.

More like a modified hammer where one grinds away the head of a hammer into a perfect screw-head is what an upsampler basically does.
Upsampling (properly with filter) modifies the DAC's behavior to near perfect performance. Basically removing the 'broken part'.

It won't give you any timing advantages though
It certainly wont. It will merely remove the 'broken' part only.
 

HP9000

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non-existent audible difference
We cannot hear or "sense" 100kHz.
I have hesitated from making an explicit admission of being wrong in referencing a study about spatial hearing to back up my claim of 10us/100khz monaural time sensitivity. This is because I have (now had) been skeptical of why the brain would prove to have a 10us capability for two ears and not keep that capability when hearing with only one ear/the same way through both ears. To spare some words I also excluded pointing out the trend toward finer values that the range of microsecond-unit results take on.

"The shortest auditory click I was able to find in the literature, and which was used in a psychophysical context (i.e., audible to a human) was 10 microseconds..."
"The upper edge of human hearing is 20 kHz so it is unlikely that we could detect a difference between a 25 us impulse and a 10 us impulse (or even a 1 us impulse), but assuming sufficient energy, all would be detectable.", As in my previous post, where I talked about energy.
From: https://psychology.stackexchange.co...re-the-temporal-limits-of-the-auditory-system

I think it's no coincidence that this number is the same between left/right differential and amplitude/time differential.

~

The tractor will only ever be useful for field work in the mud.
As delta-sigma dacs/oversampling will only ever be useful as long as high res music is not played into it. You are attributing the issue of scarcity of higher bit and sample rate material to the dac.
With actual music material, you won't see any of it.
The picture comparisons show differences in the waveforms. Together with that, the sensitivity of the ear is higher than the visual proportional differences.
NOS audio looks like a staircase that doesn't even represent the actual signal very well
a NOS DAC will ring on EVERY single sample: its step response will also not be perfect. It will always overshoot. It's clearly visible in the pictures I posted earlier.
Either you are taking issue with PCM/digital audio in general ('staircase'), are saying that the bit/sample rate is not sufficient (quantization error), or that no reconstruction filter is in place - not that one can't be added.
What? Class D amps are not digital...
Hence my use of parentheses around the words, other people call them that.
we can have class D amps capable of distortion products below -120 dB, all the way up to 20 kHz:
My first comment on this thread was about how the square wave test is usually excluded. I'd like to see a class d amp tested in that way.
prove that NOS DAC's are in any way superior at the timing bit.
It is already proven. By default they have an unaffected impulse response.

Only when you upsample (so not make it NOS) you can have an influence on the filter response.
No. There are R2R dacs (an example of a dac architecture that is filter-less and NOS) that have analogue low pass filters, digital filters, and oversampled-prior material played into them. There are various configurations.
One can do the exact same using DS (and upsampling) and decide to use different reconstruction filters than the one (or more) that come with it.
Yet you don't get to determine that it won't oversample in the case that it doesn't need to be doing that. It is in this way inflexible.
A filterless NOS DAC is broken by design, the world filterless implies this a vital component is missing. The fact that you can circumvent the broken part by adding a filter and MUST upsample in order to make the filter undo the 'broken' aspect says it all.
"Broken by design" has a specific meaning. It is to state that there is something innately wrong versus there being something exteriorly wrong.
In the case of a filter-less NOS dac, you can circumvent the problems posed by circumstances exterior to the architecture before having to swap architectures.
And there is nothing exclusively advantageous about upsampled material vs natively hi-res material. It is not something that must be done.
You have a very weird idea of how class-D operates
A triangle wave reads or probes the input signal, a power transistor works by it/is not doing that itself, and is acting as a switch and not a linear amplifier. Explain what is weird about this understanding.
as well as about a lot of other things.
Like what?

So in that light... yes all real NOS R2R DACs are broken when used without upsampling/filtering.
So they are broken when playing natively high bit and sample rate material into them while having no upsampling/filtering?
if you would have a filterless NOS DAC, the best you can do is oversample to a high rate and still have an acceptable result.
This sentence is a non-sequitur. It would read correctly if you said "if you have a ex. a Redbook file, the best you can do is oversample to a high rate and still have an acceptable result."
 

Sir Sanders Zingmore

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have hesitated from making an explicit admission of being wrong in referencing a study about spatial hearing to back up my claim of 10us/100khz monaural time sensitivity. This is because I have (now had) been skeptical of why the brain would prove to have a 10us capability for two ears and not keep that capability when hearing with only one ear/the same way through both ears. To spare some words I also excluded pointing out the trend toward finer values that the range of microsecond-unit results take on.

"The shortest auditory click I was able to find in the literature, and which was used in a psychophysical context (i.e., audible to a human) was 10 microseconds..."
"The upper edge of human hearing is 20 kHz so it is unlikely that we could detect a difference between a 25 us impulse and a 10 us impulse (or even a 1 us impulse), but assuming sufficient energy, all would be detectable.", As in my previous post, where I talked about energy.
From: https://psychology.stackexchange.co...re-the-temporal-limits-of-the-auditory-system

I think it's no coincidence that this number is the same between left/right differential and amplitude/time differential.
I honestly don't understand your point. Thing is, I don't need to.
I'm talking about your very detailed explanations about why the M-Scaler makes an audible difference. The reason I don't need to understand your explanations is that they can be ignored completely due to the fact that no-one has ever shown that the M-scaler makes an audible difference.
 

Blumlein 88

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I have hesitated from making an explicit admission of being wrong in referencing a study about spatial hearing to back up my claim of 10us/100khz monaural time sensitivity. This is because I have (now had) been skeptical of why the brain would prove to have a 10us capability for two ears and not keep that capability when hearing with only one ear/the same way through both ears. To spare some words I also excluded pointing out the trend toward finer values that the range of microsecond-unit results take on.

"The shortest auditory click I was able to find in the literature, and which was used in a psychophysical context (i.e., audible to a human) was 10 microseconds..."
"The upper edge of human hearing is 20 kHz so it is unlikely that we could detect a difference between a 25 us impulse and a 10 us impulse (or even a 1 us impulse), but assuming sufficient energy, all would be detectable.", As in my previous post, where I talked about energy.
From: https://psychology.stackexchange.co...re-the-temporal-limits-of-the-auditory-system

I think it's no coincidence that this number is the same between left/right differential and amplitude/time differential.

~


As delta-sigma dacs/oversampling will only ever be useful as long as high res music is not played into it. You are attributing the issue of scarcity of higher bit and sample rate material to the dac.

The picture comparisons show differences in the waveforms. Together with that, the sensitivity of the ear is higher than the visual proportional differences.


Either you are taking issue with PCM/digital audio in general ('staircase'), are saying that the bit/sample rate is not sufficient (quantization error), or that no reconstruction filter is in place - not that one can't be added.

Hence my use of parentheses around the words, other people call them that.

My first comment on this thread was about how the square wave test is usually excluded. I'd like to see a class d amp tested in that way.

It is already proven. By default they have an unaffected impulse response.


No. There are R2R dacs (an example of a dac architecture that is filter-less and NOS) that have analogue low pass filters, digital filters, and oversampled-prior material played into them. There are various configurations.

Yet you don't get to determine that it won't oversample in the case that it doesn't need to be doing that. It is in this way inflexible.

"Broken by design" has a specific meaning. It is to state that there is something innately wrong versus there being something exteriorly wrong.
In the case of a filter-less NOS dac, you can circumvent the problems posed by circumstances exterior to the architecture before having to swap architectures.
And there is nothing exclusively advantageous about upsampled material vs natively hi-res material. It is not something that must be done.

A triangle wave reads or probes the input signal, a power transistor works by it/is not doing that itself, and is acting as a switch and not a linear amplifier. Explain what is weird about this understanding.

Like what?


So they are broken when playing natively high bit and sample rate material into them while having no upsampling/filtering?

This sentence is a non-sequitur. It would read correctly if you said "if you have a ex. a Redbook file, the best you can do is oversample to a high rate and still have an acceptable result."
I could make a big post refuting all the mis-guided ideas you parrot here. I say parrot because you clearly lack understanding of your own. The easiest one underlying everything else is you have the wrong idea that PCM is a stair-case signal. It isn't. The idea is one of those that "is not even wrong". Ideas so incomplete you cannot even make useful predictions. A filter-less DAC is broken by design so much it isn't even wrong. It is broken. And you don't understand that.

Plenty of class D amps that do a fine job with squarewaves. Plenty of oversampled DACs that border on theoretical perfection for high sample rate material. But you are so lost you cannot even understand it at all. Don't even know how off the reservation you are in what you are writing. As an example you can get at least 23 bits correctly out of oversampled Delta-Sigma converters. Few R2R converters can get close to 16 bit performance. You are wandering in the wilderness with a head full of mythical conceptions. Ideas you parrot, yet you don't understand enough to see when people have exposed your wrong-headed ideas and unwilling to see actual results contradicting EVERY SINGLE ASSERTION in your posts. Your posts are an embodiment of Brandolini's law.
 

voodooless

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The picture comparisons show differences in the waveforms.
Yes, there are some, but only where actually the signal is no longer band limited and actually invalid (things like clipping)
Together with that, the sensitivity of the ear is higher than the visual proportional differences.
Obviously not. Visually I can represent any discrepancy, however small it is. No way all of these are audible.
Either you are taking issue with PCM/digital audio in general ('staircase')
:facepalm: Staircase != PCM. It's even in the name: Pulse Code Modulation. It's a train of pulses, not a staircase! I'll refer to the Monty video for that. Plus you clearly haven't understood anything of my previous post detailing why NOS is broken by design. It has everything to do with this.
It is already proven. By default they have an unaffected impulse response.
I asked for actual audio data that proves you are right. I guess you have none?
Yet you don't get to determine that it won't oversample in the case that it doesn't need to be doing that. It is in this way inflexible.
There is no point, is there? You need oversampling to make a delta-sigma DAC work. Flexibility? What for? We get performance that is totally SOTA and beyond what our ears can differentiate.
This sentence is a non-sequitur. It would read correctly if you said "if you have a ex. a Redbook file, the best you can do is oversample to a high rate and still have an acceptable result."
Sure, If you already have high-res material, that works as well. The reality is though, that the vast majority of audio I play is not high-res, and I suspect that that is the same for most other people as well.
 

BDWoody

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I have hesitated from making an explicit admission of being wrong

Until then, you've taken up enough of this thread.

I hope you do more reading and learning, but until then...
 

solderdude

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There are R2R dacs (an example of a dac architecture that is filter-less and NOS) that have analogue low pass filters, digital filters, and oversampled-prior material played into them.

Please show me a least 1 example of a NOS DAC that has analog low pass filters that are steep enough to act as a reconstruction filter and shifts frequency depending on sample rate.


you don't get to determine that it won't oversample in the case that it doesn't need to be doing that. It is in this way inflexible.

Set Chord to bypass ?

A triangle wave reads or probes the input signal, a power transistor works by it/is not doing that itself, and is acting as a switch and not a linear amplifier. Explain what is weird about this understanding.

A lot, yes, a triangular wave can be used, yes output devices do switch between PS rails... yes... it is the rest of your theories that are a bit ... well... You only seem to have read some basics somewhere so it seems.


"Broken by design" has a specific meaning. It is to state that there is something innately wrong versus there being something exteriorly wrong.
In the case of a filter-less NOS dac, you can circumvent the problems posed by circumstances exterior to the architecture before having to swap architectures.

In this case a filterless NOS DAC is broken as it does not adhere to the sampling theorem and also can't either.
The fact that you do not understand says enough.

And there is nothing exclusively advantageous about upsampled material vs natively hi-res material. It is not something that must be done.

In order to properly reproduce 44.1 and 48kHz there is a difference between upsampled and native in filterless NOS DACs.
The vast majority of available music is 44.1kHz. The fact that some folks use filterless NOS DACs natively and even like/prefer it is because our transducers and hearing is so crappy.

So they are broken when playing natively high bit and sample rate material into them while having no upsampling/filtering?

I already stated that they aren't 'broken' above 88.2kHz didn't I so clearly that is NOT what I have been saying ?
Using a filterless NOS DAC as is, at sample frequencies below 88.2kHz, is silly.


Besides... it also highly depends on whether or not the drivers or amp(s) are creating IM products within the audible band which also are not masked by music.
 

Geert

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I have hesitated from making an explicit admission of being wrong in referencing a study about spatial hearing to back up my claim of 10us/100khz monaural time sensitivity. This is because I have (now had) been skeptical of why the brain would prove to have a 10us capability for two ears and not keep that capability when hearing with only one ear/the same way through both ears. To spare some words I also excluded pointing out the trend toward finer values that the range of microsecond-unit results take on.

"The shortest auditory click I was able to find in the literature, and which was used in a psychophysical context (i.e., audible to a human) was 10 microseconds..."

The study you referred to determined our sensitivity for a temporal discontinuity (the 'Two‐Click Threshold', meaning a break in a sound). It showed we can discriminate 2 pulses with a 10 microseconds pause in between from a single pulse with a 20 microsecond pulse length. This says nothing about our ability to discriminate timing differences (timing resolution), and certainly not absolute time resolution (not a difference between 2 concurrent impulses where's one a bit early or late like in ITD but hearing timing of a single pulse in a pulse train is off). Your conclusion is nothing more than science fiction.
 
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voodooless

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Please show me a least 1 example of a NOS DAC that has analog low pass filters that are steep enough to act as a reconstruction filter and shifts frequency depending on sample rate.
I have a better experiment: Take a 44.1 kHz file. Do a NOS "upsample" to 705.6 kHz (Zero Order Hold style). Now use a nice brick wall filter at ~21.5 kHz, and then downsample that back to 44.1 kHz. Let's put the difference in Deltawave :)

I was figuring out how to do this in sox, but it looks like the rabbit filter that should do the upsampling is deprecated and removed from my version of sox.
 

solderdude

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I just wanted HP9000 to point me to such R2R DAC he claims exist that is not inherently 'broken' for 44.1 and 48kHz and less broken for 88.1 and 96kHz.
AFAIK there aren't any of the ones he claims in existence.

HP9000 never claimed that one should downsample the output of the Chord back to 44.1 so see no value in that (thought) experiment.
Every downsample will undo the gains one obtains when upsampling 44.1/48 when played through filterless NOS DAC.
 
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