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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

Jomungur

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Everything Rob Watts says is at best a misrepresentation and often outright lies.
I wish this were true, it'd make it easier to dismiss him. Chord's mixing truths with lies and half-lies, or even just reckless claims, is worse.

His DACs do work well, which is part of the problem. People enjoy his DACs and trust his expertise. Doesn't hurt when everyone and their mother in the audio world raves about Chord. Then when he says stuff that is false or reckless, people defer to him because they lack the knowledge to evaluate it properly.
 

DSJR

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I'm thick (moreso than usual) and still don't understand - so apologies. My 1988 Philips based Micro Seiki CD player (based more on the Marantz CD94 version of the donor chassis) has the revered or dreaded 1541 dac chip with matching filter chip (all spread out on boards where a single chip seems to do it all today). I'm sure these old players used 4x oversampling of the 16/44 red book discs played on it - is that not a similar if far simpler method of old doing it the same way? As said, please excuse the repeated over-simplistic question..

My local dealer sold an M-Scaler a while ago to the owner of a set of Kii Threes and controller (can't remember the source here) and the owner apparently loves the combionation, again claiming an improvement in 'sound quality.' If he's the chap I met once, he's intelligent and kindly with a very keen interest in music rather than the gear as such and the system is used to play music rather than be adored as 'the gear.'

Head and heart, how rarely they seem to come together cohesively...
 

voodooless

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This is impossible, right? Through upscaling or upsampling (not sure what the correct term is so I'll use them interchangeably), you cannot recreate the original analog performance from a digital file that was sampled at 44.1hz/16 bit. That is, the best you can do is remove (through sampling, filters, etc.) audible artifacts that are an inevitable byproduct of the analog to digital conversion.
Within the signal bandwidth, the reconstruction is basically perfect (enough), just look at the THD+N over frequency of a SOTA DAC. For that, we don't need the M-Scaler. Basically any modern DAC oversamples and applies a digital filer. The difference here is the number of filter taps that the M-Scaler claims.
This might be true, but not for the reasons stated? That is, the problem with sampling at 44,100khz is that you introduce aliasing from the folding in of frequencies above (.5 * 44,100khz), which distorts the recreated analog waveform. But the distortion is not created by the gaps between samples, the gaps are mathematically filled in by reconstructing the wave form through the inverse Fourier transform? And you can address the aliasing to a large degree through filters and perhaps upsampling.
The bit depth determines how much timing resolution you have. If you need more, apply (noise-shaped) dither. Transients that are smaller than the time between two samples are illegal signals and should be filtered out, and are therefore irrelevant.
This is impossible? You cannot "repair" a digital file and add back information that was never there; what you can do through upscaling is mitigate sonic artifacts by shifting the Nyquist threshold to higher frequencies.
Indeed, you cannot repair anything. It just does what any DAC does, just with a lot of wasted computational resources.
Perhaps a little unfair because this was written a few years ago, but HQ Player can match or exceed the number of taps listed here?
We did the math a few pages back. You'll need a very serious system to do that on generic computer hardware with traditional convolution. On a GPU or using FFT, this becomes much more feasible. I have no idea how HQ Player does it though.

Edit: looks like HQPlayer has options of up to 16 million taps ;) That will definitely not be done the traditional way.
That is, can filter "quality" matter vs. filter quantity?
Well yes. How many bits are used for the calculations, the accumulation, and how to transform that back to the target bit depth all matter. But why make a 1G taps filter is you think the quality is important? Why not make a 1K taps filter with 256-Bit floating point math? Should be a fun experiment ;)
This is simply false, and perhaps absurd?
Well, it looks roughly 500k samples in the past and 500k samples into the future, so you could say it looks "deep into the data". Nice story, nothing more.
This is just plain false, based on testing but also Rob's own posts in the Headfi forums.
Well, evidence that it does not work with "everything" is in #1 ;)
 
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Rednaxela

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I'm thick (moreso than usual) and still don't understand - so apologies. My 1988 Philips based Micro Seiki CD player (based more on the Marantz CD94 version of the donor chassis) has the revered or dreaded 1541 dac chip with matching filter chip (all spread out on boards where a single chip seems to do it all today). I'm sure these old players used 4x oversampling of the 16/44 red book discs played on it - is that not a similar if far simpler method of old doing it the same way? As said, please excuse the repeated over-simplistic question..

My local dealer sold an M-Scaler a while ago to the owner of a set of Kii Threes and controller (can't remember the source here) and the owner apparently loves the combionation, again claiming an improvement in 'sound quality.' If he's the chap I met once, he's intelligent and kindly with a very keen interest in music rather than the gear as such and the system is used to play music rather than be adored as 'the gear.'

Head and heart, how rarely they seem to come together cohesively...
My kingdom for Bruno’s opinion on this upgrade.
 

mansr

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If I believed having the steepest possible filter cut-off was important, as seems to be Rob Watts' selling point, I'd simply resample my files offline using the nifty "interpft" function in Matlab. Here's the transition band with a 48 kHz white noise input:

1657807747917.png


That's 150 dB attenuation, twice as much as the M-scaler, over about 0.5 Hz. Upsampling a 10-minute input file 16x takes about 10 seconds on my computer, so it's pretty fast, and that's using 64-bit floating-point.
 

voodooless

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That's 150 dB attenuation, twice as much as the M-scaler
You're comparing digital domain vs analog domain. The digital attenuation is more than 150 dB:
regular-wide-WM-1.png
 

mansr

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You're comparing digital domain vs analog domain.
I was looking at the graph in the first post here. There, the Topping native filter reaches 100 dB attenuation while with the M-scaler it reaches only around 75 dB. That suggests that analogue performance isn't the limiting factor.

The digital attenuation is more than 150 dB:
regular-wide-WM-1.png
That looks more like 130 dB (190 - 60) to me.
 

voodooless

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That looks more like 130 dB (190 - 60) to me.
Ah yes, forgot the -60 dB. Still, it's more than the 75 that comes out at the analog end... strange.

Also, we're dealing with 24-bit audio. How would it ever be lower than 144 dB?
 

bennetng

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Ah yes, forgot the -60 dB. Still, it's more than the 75 that comes out at the analog end... strange.

Also, we're dealing with 24-bit audio. How would it ever be lower than 144 dB?
Using white noise as input, when taking clipping prevention into account, should be less than 140dB.

However with a single tone, depends on the tone's frequency, dither type and how the FFT plot is configured, can get the attenuation well below 144dB.
PS: The foobar resampler tests could be outdated, for those who are curious, do a similar test with the latest version.
 

earlevel

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I've ordered a Topping D90SE DAC and get it on Saturday. I want to compare it side by side with the DAVE now after more research in my audio systems since it's less than 1/10 of the price. This is just subjective listening tests for myself, but it shows how much I am getting disturbed by Chord's marketing claims.

I revisited the official Chord white paper, which really reads more like a marketing brochure. I've included some quotes from it, and then below them why they seem problematic. Please correct me if I'm not understanding the concepts correctly, as this stuff is new to me (I'm a lawyer, not an engineer).

1. Thirty-five years after the introduction of the CD, some would think that huge leaps forward would be improbable at this point. The Chord Hugo M Scaler proves without doubt that is not the case. It may in fact be the single greatest advancement in digital sound reproduction ever. The Hugo M Scaler offers the world’s most advanced upscaling technology, taking digital files from any source and transforming them into audio that’s virtually indistinguishable from the original analog performance.

This is impossible, right? Through upscaling or upsampling (not sure what the correct term is so I'll use them interchangeably), you cannot recreate the original analog performance from a digital file that was sampled at 44.1hz/16 bit. That is, the best you can do is remove (through sampling, filters, etc.) audible artifacts that are an inevitable byproduct of the analog to digital conversion.

2. At 44,100 samples per second, there’s a gap between samples—22 microseconds to be exact. The problem is musical timing and transient information also occurs in these gaps and whatever information that exists in the gaps is lost when creating the digital file. DACs can’t recover missing timing information—they simply miss the start of the transient. Blurring of transients is the result. That gets confusing for the ear and the brain. Which means we won’t perceive timbre or soundstage or the pitch of bass instruments properly.

This might be true, but not for the reasons stated? That is, the problem with sampling at 44,100khz is that you introduce aliasing from the folding in of frequencies above (.5 * 44,100khz), which distorts the recreated analog waveform. But the distortion is not created by the gaps between samples, the gaps are mathematically filled in by reconstructing the wave form through the inverse Fourier transform? And you can address the aliasing to a large degree through filters and perhaps upsampling.

3. The Hugo M Scaler acts like a “pre-DAC”. It takes the digital file and repairs it, adding back the information lost between the samples, then it sends the repaired file to the DAC. The M Scaler increases the sampling rate from 44,100 times per second (44.1 KHZ) by a multiple of 16, to 705,600 times per second (706.6 KHz). With 705,600 samples per second, a huge amount of important information that was lost when creating the 44.1 digital file is now recovered. The more samples, the closer you get to the original analog signal. In essence the Hugo M Scaler places 15 additional new musical samples in between each original musical sample resulting in an astounding improvement in the recreation of the original music signal.

This is impossible? You cannot "repair" a digital file and add back information that was never there; what you can do through upscaling is mitigate sonic artifacts by shifting the Nyquist threshold to higher frequencies.

4. Rob Watts, Chord’s Digital Design Consultant, has developed his exclusive WTA (Watts Transient Alignment) technology, which incorporates the most advanced interpolation filter of its kind in the world. That mammoth processing power allows for a huge breakthrough in what’s known as tap length of the filter—to a previously unimaginable 1,015,808 taps.

Perhaps a little unfair because this was written a few years ago, but HQ Player can match or exceed the number of taps listed here? More importantly, this implies that the *way* the interpolation filter works matters more than the simple number of taps, and yet it boasts about the number of taps, not sure which factor is meaningful (or both). That is, can filter "quality" matter vs. filter quantity?

5. Simply put, taps are a measure of device’s capacity to reproduce the original waveform. The longer the tap length of the filter, the closer it gets to the original analog signal.

This is incorrect, right? Tap lengths help to a point but they yield diminishing or no returns on reconstruction of the analog waveform?

6. With this ingenious technology, the M Scaler doesn’t do a crude interpolation like all other filters or “guess” to fill in the dots between each step. It peers deep into the actual data itself, as if looking under a microscope, and reconstructs the missing waveform in its exact original form thus creating an almost perfect new digital version of the original analog performance.

This is simply false, and perhaps absurd?

7. The Hugo M Scaler quite literally represents the realization of a lifelong dream for Rob Watts. Half a million lines of code and hundreds of listening tests later, listeners can now experience something they never could before— a huge difference in resolution, bass definition, sound staging, instrument separation and focus and more varied instrument timbre.

Perhaps (it's a subjective claim), but it seems the hundreds of listening tests where done on one (or a few) persons who have an interest in promoting the device?

8. Works With Everything. The Hugo M Scaler is inserted in your system ahead of the DAC— it is not a DAC and it does not replace your DAC. The M Scaler improves the sound of all digital audio systems. It works with all digital files and streaming services and all digital source components; streamers, smart devices, computers, CD/DVD players and video systems. The M Scaler works with all DAC brands so you don’t need a Chord Electronics DAC to get the benefits of upscaling. Even if you have an older DAC that only accommodates 4 or 8 times upscaling, you will still get a very worthwhile improvement in sound quality when adding the M Scaler to your audio system.

This is just plain false, based on testing but also Rob's own posts in the Headfi forums.

I was going to include the testimonials in the white paper but decided not to because it feels like kicking a dog at this point. They are pretty outrageous though.
That was painful...instead of hitting point by point, I'll give some basics that cover them:

Sampling is exact—nothing gets left out. And if it did get left out, nothing Rob Watts could do would bring it back.

More on "sampling is exact": First, there are errors in due to jitter and quantization, but for 24-bit converters the errors are small and less than errors in the analog audio electronics components, so I'll disregard them for this discussion. Second, sampling is exact only for a signal band-limited to under half the sample rate. So we're basically talking about a low pass filter.

More on that lowpass filter: The fact is we're really sampling the lowpass filtered audio, so if the the stop band at half the sample rate and above is low enough to to not be hearable, then we're good. At that point, we can consider the sampled signal to be exactly equivalent to our source signal run through that same lowpass filter in the analog domain. So that's the full definition of exact digitization—the info in the samples is exactly the lowpass filtered signal.

So, those caveats aside, there is no smearing or loss of transients. More precisely, there is no loss of smearing of the lowpass filtered signal that gets digitized. It's all there. This is mathematically irrefutable for a PCM digitized signal. (Go to my website if you want the math.)

On playback, we need another lowpass filter, and we're again somewhat at the mercy of its qualities. The reconstruction is theoretically exact, again at the mercy of jitter and quantization—though here we can completely ignore quantization for 24-bit or greater because we're already quantized to 24-bit, so that error is baked—and the quality of the lowpass filter. Again, we can ignore everything but the lowpass filter.

So, forget about transients—the only way to have more information is to sample at a higher rate to start with (discussions elsewhere about whether it matters). If something is already sampled at a lower rate, that is what you're stuck with. You can only hope to reproduce it without degrading it, noticeably, further. Which basically means using a lowpass filter that good enough that further improvements make no difference in sound quality.

Rob Watt's device, at best, hopes to offload some of the filter work, so that the DAC has an easier task. A reasonable idea, but only worthwhile if you end up with an audibly better result. And it is possible to end up with a worse result, or no audible difference. The question would be if you're spending a lot of money trying to fix something that's not broken. But the part about restoring the true transients is fantasy.
 

charleski

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2. At 44,100 samples per second, there’s a gap between samples—22 microseconds to be exact. The problem is musical timing and transient information also occurs in these gaps and whatever information that exists in the gaps is lost when creating the digital file. DACs can’t recover missing timing information—they simply miss the start of the transient. Blurring of transients is the result. That gets confusing for the ear and the brain. Which means we won’t perceive timbre or soundstage or the pitch of bass instruments properly.

This might be true, but not for the reasons stated? That is, the problem with sampling at 44,100khz is that you introduce aliasing from the folding in of frequencies above (.5 * 44,100khz), which distorts the recreated analog waveform. But the distortion is not created by the gaps between samples, the gaps are mathematically filled in by reconstructing the wave form through the inverse Fourier transform? And you can address the aliasing to a large degree through filters and perhaps upsampling.
Chord's claim about time resolution here displays fundamental misconceptions that should be embarrassing. It's worth taking a look at @mansr's article on this:
 

Fourlegs

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Apparently the manual was written by a kid on ‘work experience’ perhaps he also designed the M-Scaler while he was there.
Keith
I am the originator of that bit about the manual being written by a kid on ‘work experience’ and posted it on another forum. It is nice to know that you read my posts there. I said it by way of expressing my frustration in how difficullt it is to understand the input and output compatibility charts in the Mscaler manual. In fact that is an understatement, they as near to incomprhensible as makes no difference. To be clear though as far as I am aware there is no truth whatsoever in saying the charts were created by a work experience kid (least your post gives any credibility to that notion).
 
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amirm

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4. Rob Watts, Chord’s Digital Design Consultant, has developed his exclusive WTA (Watts Transient Alignment) technology, which incorporates the most advanced interpolation filter of its kind in the world. That mammoth processing power allows for a huge breakthrough in what’s known as tap length of the filter—to a previously unimaginable 1,015,808 taps.
I sure hope Rob himself didn't write this. WTA is a windowing function, not a filter. Windowing is there because you have discontinuity at either end of the filter taps if you just stop (nothing abruptly stops and ends that way). So convention is to roll each end off (think of it as fade up and fade down). There are different windowing options and I guess he has invented his own. This is just preprocessing of the audio samples to then be followed up by the actual filter. It involves no "interpolation" on its won. To mix them up shows no understanding of the basics of signal processing.
 
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amirm

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3. The Hugo M Scaler acts like a “pre-DAC”. It takes the digital file and repairs it, adding back the information lost between the samples, then it sends the repaired file to the DAC.
I guess we are wasting money on large telescopes. We can just use a postage stamped frame and create massive images from it! All we have to do is filter them! Oh wait, filter takes things away, not add them! Only in going after layman can you get away with such nonsense.
 

rkt31

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No, it doesn't, and that's what makes it really ridiculous.
I don't think any other hardware upsampler was measured here. If it was then you may please post the link for filter quality.
 

rkt31

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If I believed having the steepest possible filter cut-off was important, as seems to be Rob Watts' selling point, I'd simply resample my files offline using the nifty "interpft" function in Matlab. Here's the transition band with a 48 kHz white noise input:

View attachment 218265

That's 150 dB attenuation, twice as much as the M-scaler, over about 0.5 Hz. Upsampling a 10-minute input file 16x takes about 10 seconds on my computer, so it's pretty fast, and that's using 64-bit floating-point.
Isn't it digital? Digital results are much higher than analog !
 

voodooless

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I don't think any other hardware upsampler was measured here. If it was then you may please post the link for filter quality.
It doesn't do better than a DAC without an upsampler, so why would any other upsampler be any different? The DAC itself already does a perfectly fine state-of-art job.
 
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