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CHORD M-Scaler Review (Upsampler)

Rate this product:

  • 1. Poor (headless panther)

    Votes: 358 88.2%
  • 2. Not terrible (postman panther)

    Votes: 13 3.2%
  • 3. Fine (happy panther

    Votes: 7 1.7%
  • 4. Great (golfing panther)

    Votes: 28 6.9%

  • Total voters
    406

Jimster480

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I really don't see the point of this device.... Do you still have the device @amirm? Maybe you can test it with the DAVE and see if it does anything since that is the biggest claim to fame.
 

voodooless

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@PassionforSound
1658814353940.jpeg
 

Lukino

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I understand those people believe in these expensive devices for years and their manufacturers adopt them... They use it in a top system. But the measurements showed that they succumbed to the placebo effect. It hurts... He certainly won't disconnect it just like that... And that's ok. If a person is satisfied that he has it there and can listen to music with more peace, even if it's a placebo, it's OK. But he doesn't have to convince others to succumb to the placebo. He himself was worried about it when he saw the measurements. Maybe the Chord amplifier will be measured later and the measurements will show that it has been listening to big distortions for years. It's like that when you trust companies for money.;)
 

earlevel

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You don't improve the sound by upsampling. Once again, the reason you don't improve the sound by upsampling is not because I can't hear any improvement or someone else can't hear any improvement. The reason is that upsampling simply, and literally, just duplicates samples. So if you upsample from 44.1kHz to 176.4kHz, that's 4x oversampling, which means every original sample is replicated three more times - it's copy-pasted three times so that every original sample is now four samples. All four samples are identical - there is no "interpolation" like you might get with a TV that increases the frame rate by creating new frames that are combinations of the frames before and after them. That's not how upsampling works - it just copies the existing sample exactly.
Of course you're right in the first sentence—at upsampling best, upsampling doesn't lose anything significant, but it never gets better. But we may want the higher sample rate for other reasons. There is interpolation though, normally done with a filter.

And on the duplicating, you already have your answer, but I guess I had nothing better to do tonight than drawing dots and lines. This shows why insert zeros when upsampling by an integer ratio. We start with a sine wave, and a grid that show where it will be sampled:

sine.png


We sample—trust me, the samples line up with the sine, but I removed the sine to stress that information is gone. These are instantaneous measurement, mathematically equivalent to multiplying the sine by a unit pulse train:
sine sampled 1x.png


Now, we want to resample that at a grid that's 4x that. If the above were analog, there wouldn't be space between the samples, there would be 0 volts DC. (Why? because the sampling was with a pulse train, and an analog pulse train is zero in between pulses. The mathematical reason it's a pulse train is a little more complicated, see my website, but for now just trust that between space is not the equivalent of "no one knows", but precisely zero). So, at the new grid lines—yes, zeros:

sine sampled 4x.png


So, what's the difference between the weird, non-sine-looking shape we have at 4x, and the original? First, you have to understand that the 1x sampled version is not a sine wave, but a PCM-encoded sine wave—a numerically coded pulse-modulated signal. If we sampled a 3 kHz sine at 48 kHz, we'd have 3k, 45k (48-3), 51k (48+3), 93k (2*48-3), 99k (2*48+3), due to the modulation...but when we run it through the under-24k lowpass filter in the DAC, we end up with 3k analog again. However, we still have those frequencies in the 4x-sampled version. Nothing changed except the sample rate. So, after inserting the zeros, we need to filter with a ~24kHz lowpass. That gets rid of the unwanted frequencies. Or, interpolates the samples, depending on whether you'd looking at it in the time doamin or the frequency domain. It's still a PCM signal, so we end up with 3k, 189k (192-3), 195k...and again, just 3k when we run it through the DAC filter.

I hope that helps someone understand why we insert zeros. It may seem obvious to some, but I know a lot of people don't understand where the zeros come from. (20 years ago, in a discussion on DSP mailing list, a guy told me I didn't know what I was talking about on this point. I said, ok, then tell us what is the reason for the zeros. His answer was, "Serendipity." He said it was just mathematical luck. :p)
 

Clavius

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You can add data but you cannot add information, people tend to get flummoxed about the significance of the difference.
If you could simply improve the quality, as in increasing information content, of digital data by upsampling you would have no problem in recreating incomplete dna data, improve the quality of the pictures from the James Webb telescope etc, etc. If it only worked like claimed, it would lead to a scientific revolution and not a expensive hifi box.

Rather than looking for proof of the negative @PassionforSound, I would recommend that you pause for a few seconds and give reflection a chance.
 
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bboris77

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I am not sure exactly why, but reading this whole thread reminded me of that moment in everyone’s childhood when they find out that Santa does not exist.

I don’t mean this as a dis towards anyone here because I’ve been on both sides of this - subjectivist and objectivist. I wanted to believe in various magical-type thinking whether it be tubes, discrete amplifiers or exotic DACs. On the other hand, I’ve also spent money on fantastically-measuring devices in the pursuit of sonic perfection.
Ultimately, both of these approaches lead to disappointment and an empty wallet.

At this point of my audio journey, I am making my peace with the fact that audio reproduction has been solved for a very long time now, and you don’t really need to spend more than $100 for a DAC and another $100 for a headphone amp to achieve some kind of sonic nirvana. Most things that are above that price point are either about functionality or about selling dreams.

The thing that matters the most are headphones/speakers and that is where one needs to spend a bit more to get good sound. In terms of some kind of breakthrough moment in the area is sound/music enjoyment, it would have to start in the domain of sound recording before any benefits are seen/heard in the field of sound reproduction. There is simply no way to extract anything more from current digital recordings no matter how one manipulates the means of sound reproduction.
 

Jimi Floyd

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What I am not seeing is any evidence that the device isn't doing what it claims to do.
OK, let's analyse first what it claims to do. Let's read the Hugo M Scaler description page on the Chord website. I summarise here the claims.

1) The Hugo M Scaler is a highly advanced standalone upscaler capable of redefining sound quality from digital audio.
2) The Hugo M Scaler brings the unrivalled advantages of our ground-breaking FPGA-based WTA (Watts Transient Alignment) filtering technology to digitally connected audio devices, dramatically improving sound quality.
3) When partnered with either of Chord Electronics’ 768kHz-capable dual-BNC-input DACs, the Hugo M Scaler sets an astonishing technical benchmark for digital audio performance at it price point, redefining sound quality from digital audio

This is ALL Chord claims the M scaler is doing, as WRITTEN on their webpage. Other features claimed by designers or journalists in youtube talks do not count. Verba volant, scripta manent.

So, you are asking evidence that the device is not "capable of redefining sound quality" or "dramatically improving sound quality". You understand that evidence for or against something is only possible on well defined statements. Chord claims about the M scaler are so vague that any proof of anything is impossible. And, of course, that hazy and nebulous sentences just try to hide emptiness and lack of substance.

There is only one kind of information I have seen in HiFi which is more useless than the M Scaler description by Chord, and it is the M Scaler technical data supplied by Chord: FPGA: Xilinx XC7A200T, Filter tap length: 1,015,808 WTA taps, then dimensions and weight. Nothing else, enough said.
 

mansr

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This is ALL Chord claims the M scaler is doing, as WRITTEN on their webpage. Other features claimed by designers or journalists in youtube talks do not count.
I think you're being a bit too strict there. If Rob Watts (or anyone representing the company) says something in a presentation or writes it in a blog or forum post, that should count too. Statements attributed to them by magazines should perhaps count a bit less since they might have been misquoted (but did not deem it necessary to issue a correction). Clearly, miraculous qualities ascribed to the device by reviewers or random people on the internet should not be held against the manufacturer, though the fact that they encourage these people in their beliefs is telling nonetheless.
 

Jimbob54

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I suppose I shouldn't have asked if he understood what determines the timing accuracy of a digitally sampled system. He hasn't replied since I asked him the question. Maybe he is studying up on it?
Alas, no I think he got what he came for . I doubt he will be back.

It seems he wants to leave it as (paraphrased):

They didnt measure the right things . nothing they have disproves that it does what it says it does. I prefer Headfi. I must be objective because I didnt prefer the DAVE.


As of around 10pm last night UK time:

"Mike, you need to take the measurements and, more importantly the conclusions drawn, on ASR with a bucket of salt. My attempts to discuss the M-Scaler review on ASR showed an unwillingness to discuss the relevance of the measurements and conclusions drawn. Instead the emphasis was on the argument that the M-Scaler can't do anything because it can't do anything. I've got the DAVE here now (going home today) and it sounds great. If it were so flawed, it would not perform as it does in listening tests. It's not my favourite DAC and I have no desire to buy one (even if I could afford it), but ASR are doing a disservice with their butchering of these products that it seems they don't understand beyond "digital in, analog out". I'd recommend visiting the Head-Fi threads on DAVE and M-Scaler for far more balanced info with solid technical evidence being shared in both directions."
Show less
 

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Dogcoop

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Alas, no I think he got what he came for . I doubt he will be back.

It seems he wants to leave it as (paraphrased)

They didnt measure the right things . nothing they have disproves that it does what it says it does. I prefer Headfi. I must be objective because I didnt prefer the DAVE.


As of around 10pm last night UK time:

"Mike, you need to take the measurements and, more importantly the conclusions drawn, on ASR with a bucket of salt. My attempts to discuss the M-Scaler review on ASR showed an unwillingness to discuss the relevance of the measurements and conclusions drawn. Instead the emphasis was on the argument that the M-Scaler can't do anything because it can't do anything. I've got the DAVE here now (going home today) and it sounds great. If it were so flawed, it would not perform as it does in listening tests. It's not my favourite DAC and I have no desire to buy one (even if I could afford it), but ASR are doing a disservice with their butchering of these products that it seems they don't understand beyond "digital in, analog out". I'd recommend visiting the Head-Fi threads on DAVE and M-Scaler for far more balanced info with solid technical evidence being shared in both directions."
Show less
A real piece of work. His presence on this forum was to allow him to state that he was an ‘impartial’ audiophile just looking for the truth. By his actions and behavior on this site he has failed in that respect. He should never be allowed to wear the hat of ‘impartial reviewer.’
 

CapMan

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A real piece of work. His presence on this forum was to allow him to state that he was an ‘impartial’ audiophile just looking for the truth. By his actions and behavior on this site he has failed in that respect. He should never be allowed to wear the hat of ‘impartial reviewer.’
It’s important to be open minded on arriving at this forum or there is no point engaging in dialogue.

For some I guess it is hard to accept that cheap is good enough to deliver SOTA.
 

Marc v E

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Alas, no I think he got what he came for . I doubt he will be back.

It seems he wants to leave it as (paraphrased)

They didnt measure the right things . nothing they have disproves that it does what it says it does. I prefer Headfi. I must be objective because I didnt prefer the DAVE.


As of around 10pm last night UK time:

"Mike, you need to take the measurements and, more importantly the conclusions drawn, on ASR with a bucket of salt. My attempts to discuss the M-Scaler review on ASR showed an unwillingness to discuss the relevance of the measurements and conclusions drawn. Instead the emphasis was on the argument that the M-Scaler can't do anything because it can't do anything. I've got the DAVE here now (going home today) and it sounds great. If it were so flawed, it would not perform as it does in listening tests. It's not my favourite DAC and I have no desire to buy one (even if I could afford it), but ASR are doing a disservice with their butchering of these products that it seems they don't understand beyond "digital in, analog out". I'd recommend visiting the Head-Fi threads on DAVE and M-Scaler for far more balanced info with solid technical evidence being shared in both directions."
Show less
Strange that someone simply cannot understand that a device that makes things worse -and even in theory can't make things better- is not worth the money. I was pretty impressed with the patience shown here. And the wonderful explanations. Not sure what else we can do but proceed where we left. There is much more interesting stuff to do and test.
 

tmtomh

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Of course you're right in the first sentence—at upsampling best, upsampling doesn't lose anything significant, but it never gets better. But we may want the higher sample rate for other reasons. There is interpolation though, normally done with a filter.

And on the duplicating, you already have your answer, but I guess I had nothing better to do tonight than drawing dots and lines. This shows why insert zeros when upsampling by an integer ratio. We start with a sine wave, and a grid that show where it will be sampled:

View attachment 220532

We sample—trust me, the samples line up with the sine, but I removed the sine to stress that information is gone. These are instantaneous measurement, mathematically equivalent to multiplying the sine by a unit pulse train:
View attachment 220533

Now, we want to resample that at a grid that's 4x that. If the above were analog, there wouldn't be space between the samples, there would be 0 volts DC. (Why? because the sampling was with a pulse train, and an analog pulse train is zero in between pulses. The mathematical reason it's a pulse train is a little more complicated, see my website, but for now just trust that between space is not the equivalent of "no one knows", but precisely zero). So, at the new grid lines—yes, zeros:

View attachment 220535

So, what's the difference between the weird, non-sine-looking shape we have at 4x, and the original? First, you have to understand that the 1x sampled version is not a sine wave, but a PCM-encoded sine wave—a numerically coded pulse-modulated signal. If we sampled a 3 kHz sine at 48 kHz, we'd have 3k, 45k (48-3), 51k (48+3), 93k (2*48-3), 99k (2*48+3), due to the modulation...but when we run it through the under-24k lowpass filter in the DAC, we end up with 3k analog again. However, we still have those frequencies in the 4x-sampled version. Nothing changed except the sample rate. So, after inserting the zeros, we need to filter with a ~24kHz lowpass. That gets rid of the unwanted frequencies. Or, interpolates the samples, depending on whether you'd looking at it in the time doamin or the frequency domain. It's still a PCM signal, so we end up with 3k, 189k (192-3), 195k...and again, just 3k when we run it through the DAC filter.

I hope that helps someone understand why we insert zeros. It may seem obvious to some, but I know a lot of people don't understand where the zeros come from. (20 years ago, in a discussion on DSP mailing list, a guy told me I didn't know what I was talking about on this point. I said, ok, then tell us what is the reason for the zeros. His answer was, "Serendipity." He said it was just mathematical luck. :p)

Thank you - this is super-clear, and while I feel pretty dumb right now for making that error (the source of which was, I think, considering the concept of interpolation only in one sense and not the sense in which it actually gets applied in this process), I am glad to have gotten a clear understanding from you (and @danadam , as usual :) ).

After I was first notified of my mistake, I did some searching online and while I came across some good explanations, yours is the clearest I have encountered on the specific point of how ultrasonic frequencies get generated after the A/D step, requiring the low pass filter at the other end too.

Just to make sure I’m not missing anything, the effect of the 4x oversampling in your example is to increase the frequency of the unwanted ultrasonics, increasing the frequency spread between them and the desired signal/audible range, yes? My understanding has been that this is the only effect/purpose of such oversampling (easing/simplifying design of the low pass filter). Is there any additional effect besides that?

Thanks again!
 
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