• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

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

Chester

Senior Member
Joined
May 22, 2021
Messages
442
Likes
1,068
The non-existence of it.

The test results are not the problem.. well apart from the high jitter and non standard spdif voltage. Even if they were fine, you still have a multi-thousand dollar box that essentially does nothing audible.

Basically this thing was doomed from the get go. Is that fair? Maybe not? Neither is shilling for this shiit.. ah wait, that’s a different brand :facepalm:;)

I didn’t think they were unreasonable questions, that would help me understand better.

40 odd pages on a box that was doomed from the off…..that’s all I need to know. On to something more interesting.
 

tmtomh

Major Contributor
Forum Donor
Joined
Aug 14, 2018
Messages
2,728
Likes
7,991
I didn’t think they were unreasonable questions, that would help me understand better.

40 odd pages on a box that was doomed from the off…..that’s all I need to know. On to something more interesting.

Your questions definitely were reasonable.

As @voodooless notes, the issue is twofold:

1. The M-Scaler's measured performance (jitter, SINAD, etc), while not bad enough to produce any audible problems, is notably mediocre for a digital component like a DAC (the M-Scaler isn't exactly a DAC but that's the closest analogue in terms of expectations and benchmarks for measured performance). So given its price and performance, it's not a good choice.

2. More fundamentally, the M-Scaler is claimed to "reconstruct" missing information by filling in transients lost with a digital source's original sample rate, by increasing the sample rate and thereby the "resolution" of the signal. There no evidence that it does this - and that is because it CANNOT do this, for two reasons:

(a) Once a digital recording has been made at (or downsampled to) a given sample rate, like 44.1kHz for example, that's it: any missing information is permanently gone and cannot be reconstructed. Any upscaling simply duplicates the existing samples. It's not like frame interpolation with video, where a TV or processor can create new frames in between existing ones by combining the two existing frames and therefore create the impression of smoother motion (the so-called, and much-detested, "soap opera effect" when such processing is applied to 24 frame-per-second movies). A simple way to think about why audio cannot be thought of or dealt with like video in this way is to remember that it is possible to freeze-frame a video and step through each individual frame one by one. If extra frames have been added, you can clearly see that in such a scenario. But with audio you cannot freeze-frame samples and actually listen to the sound progressing frame by frame, and have your brain be able to detect or make any sense of it, because audio plays in time while video is displayed in space. You can take advantage of time to visually inspect a frozen video frame. You cannot take advantage of space to audibly inspect a frozen audio sample. If you load up a song into Audacity or any other audio app, select a single sample, and press Play, as far as I know you will either hear nothing, or else you might hear a momentary "tick," which you cannot even begin to make any sense of as a musical sound. (Of course you can VISUALLY inspect a single audio sample in an audio editing app, but that's not the same thing as being able to LISTEN to that sample in that way.)

(b) Perhaps even more fundamentally, a higher sample rate BY DEFINITION cannot restore transients, because a sample rate of 44.1k is ALREADY capable of capturing the "fastest" transients that any human being can hear. This is a very basic and common misconception about digital audio. IMHO it's actually a quite understandable misconception for us non-experts, but for someone like Rob Watts it's either a stunning level of ignorance or a knowing act of deception. This becomes clear when we realize that the frequency of a sound literally IS its "speed." Any transient that is "too fast" to be encoded by a 44.1kHz sample rate is also, BY DEFINITION at too high of a frequency to be heard by humans.

So the test was "doomed from the start," yes - but @amirm tested it because the only way you can scientifically address claims like the ones Chord makes for the M-Scaler is to test those claims. Testing a device that could not do what it says was not a great option - but it was a better option than the alternative, which would be to refuse to test it and then leave open an opportunity for others to say, "you claim the M-Scaler can't work but you won't even test it - that seems very unscientific and close-minded."
 
Last edited:

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
16,004
Likes
36,218
Location
The Neitherlands
1 bit is 6dB... so it would depend on what bit one toggles :)

The whole -300dB story is misinterpreted and then used to make people believe he claimed he can hear down to -300dB.
The explanation is very simple. Sighted observations.
A bit like applying some EQ (adjusting sliders a bit) with the actual EQ circuit (not knowing) being disabled.
While adjusting the sliders one actually hears the sound change but in reality nothing did.
So when comparing some algorithm sighted (and knowing it differs) it is easy to 'reliably and reproducible' hear differences even when there aren't any.
The EQ thing has happened to many (it did to me a few times) and shows how 'knowing' can alter perception.
So it really is not important if some measurement showed -300 or -294dB. One can still 'hear' differences when one believes there are differences.

That's why: blind, level matched and statistically valid testing (when possible) is gold and sighted 'rigorous' testing is fooling yourself.
Also stating something like this 'shows' people that hang to his lips how great he (and his hearing) is.

In reality, even when using headphones at 120dB SPL peak he cannot hear any change -100dB lower than those peaks (and most likely even -70dB)
 
Last edited:

Shadders

Member
Joined
Jan 5, 2019
Messages
31
Likes
30
Location
Londinium
Hi,
Has anyone attempted to use the impulse response of the mscaler to determine its response and determine the filter coefficients etc ?

I am assuming that since it must be a LTI system, that it is possible.
Regards,
Shadders.
 

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
16,004
Likes
36,218
Location
The Neitherlands
You mean a needle pulse test signal being applied to the filter. A signal that cannot and does not exist in any recording and which shows the band limited character of a digital system ?
 

bidn

Active Member
Joined
Aug 16, 2019
Messages
195
Likes
821
Location
Kingdom of the Netherlands
I didn’t think they were unreasonable questions, that would help me understand better.

40 odd pages on a box that was doomed from the off…..that’s all I need to know. On to something more interesting.
Hi Chester,

you are free not to follow this thread.
What if other people have a different interest?

Personally, I was active on Head-Fi before slowly "moving" to ASR, and, while on Head-Fi, I was influenced (I might say "deceived") by all the praise for Chord products, to the point of being subjectively deluded in hearing some improvements when auditioning Chord devices, incI. the M-Scaler. (Luckily I was never entirely convinced, helped by the too high prices and poor design, and didn't purchase any except a Mojo).
So I am very interested by the present thread and the Hugo-2 one, enjoying both greatly.
Thank you so much, Amir & ASR community.
 

Shadders

Member
Joined
Jan 5, 2019
Messages
31
Likes
30
Location
Londinium
You mean a needle pulse test signal being applied to the filter. A signal that cannot and does not exist in any recording and which shows the band limited character of a digital system ?
Hi,
Yes, from memory, the impulse response of the system will be the filter coefficients (assume it is a filter) of an LTI system, and characterises the frequency response etc.
If the filter is 1milltion taps, then 1milltion samples will be the output ?
Regards,
Shadders.
 

pkane

Master Contributor
Forum Donor
Joined
Aug 18, 2017
Messages
5,668
Likes
10,299
Location
North-East
The whole -300dB story is misinterpreted and then used to make people believe he claimed he can hear down to -300dB.

Is there really a difference between "I can hear the difference between filters that differ from each other below -300dB" and "I can hear down to -300dB"? ;)
 

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
16,004
Likes
36,218
Location
The Neitherlands
It will look very much like the response of 'normal' similar style filters.
There will be more delay between the start of the song and the actual sound coming out of the analog section.
with 1M taps the first and last will have very, very little influence on the waveform. The further one is 'away' from the center value at that point for that sample the less it 'weighs' as it were.
It is the ones right before and after the actual sample that matter the most.
How many 'steps' between levels and samples is determined by the used frequencies (the x oversampling)
Of course the type of filter also matters, certainly when it come to the delay time.
 
Last edited:

Shadders

Member
Joined
Jan 5, 2019
Messages
31
Likes
30
Location
Londinium
It will look very much like the response of 'normal' similar style filters.
There will be more delay between the start of the song and the actual sound coming out of the analog section.
with 1M taps the first and last will have very, very little influence on the waveform. The further one is 'away' from the center value at that point for that sample the less it 'weighs' as it were.
It is the ones right before and after the actual sample that matter the most.
How many 'steps' between levels and samples is determined by the used frequencies (the x oversampling)
Hi,
I thought that was the case. Does the test equipment (Audio Precision) have the capability to initiate the impulse and record the response ?
Regards,
Shadders.
 

Chester

Senior Member
Joined
May 22, 2021
Messages
442
Likes
1,068
Hi Chester,

you are free not to follow this thread.
What if other people have a different interest?

Personally, I was active on Head-Fi before slowly "moving" to ASR, and, while on Head-Fi, I was influenced (I might say "deceived") by all the praise for Chord products, to the point of being subjectively deluded in hearing some improvements when auditioning Chord devices, incI. the M-Scaler. (Luckily I was never entirely convinced, helped by the too high prices and poor design, and didn't purchase any except a Mojo).
So I am very interested by the present thread and the Hugo-2 one, enjoying both greatly.
Thank you so much, Amir & ASR community.

Yeah I was talking about me personally, not that the thread should be done. You crack on ;)

I left Head-fi many years ago in a similar fashion so I get the journey.
 

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
16,004
Likes
36,218
Location
The Neitherlands
Is there really a difference between "I can hear the difference between filters that differ from each other below -300dB" and "I can hear down to -300dB"? ;)
It is both ridiculous.
But a filter could differ only -300dB at 1kHz but audibly at 10kHz.
Saying you can hear 'distortion' as low as -300dB is simply not possible.

It was marketing talk. Most 'designers' make claims about their gear though. One should see the statement as such.

Hi,
I thought that was the case. Does the test equipment (Audio Precision) have the capability to initiate the impulse and record the response ?
Regards,
Shadders.

Yes. But Amir usually doesn't post this.
 

sofrep811

Active Member
Joined
Jun 4, 2016
Messages
253
Likes
319
Your questions definitely were reasonable.

As @voodooless notes, the issue is twofold:

1. The M-Scaler's measured performance (jitter, SINAD, etc), while not bad enough to produce any audible problems, is notably mediocre for a digital component like a DAC (the M-Scaler isn't exactly a DAC but that's the closest analogue in terms of expectations and benchmarks for measured performance). So given its price and performance, it's not a good choice.

2. More fundamentally, the M-Scaler is claimed to "reconstruct" missing information by filling in transients lost with a digital source's original sample rate, by increasing the sample rate and thereby the "resolution" of the signal. There no evidence that it does this - and that is because it CANNOT do this, for two reaons:

(a) Once a digital recording has been made at (or downsampled to) a given sample rate, like 44.1kHz for example, that's it: any missing information is permanently gone and cannot be reconstructed. Any upscaling simply duplicates the existing samples. It's not like frame interpolation with video, where a TV or processor can create new frames in between existing ones by combining the two existing frames and therefore create the impression of smoother motion (the so-called, and much-detested, "soap opera effect" when such processing is applied to 24 frame-per-second movies). A simple way to think about why audio cannot be thought of or dealt with like video in this way is to remember that it is possible to freeze-frame a video and step through each individual frame one by one. If extra frames have been added, you can clearly see that in such a scenario. But with audio you cannot freeze-frame samples and actually listen to the sound progressing frame by frame, and have your brain be able to detect or make any sense of it, because audio plays in time while video is displayed in space. You can take advantage of time to visually inspect a frozen video frame. You cannot take advantage of space to audibly inspect a frozen audio sample. If you load up a song into Audacity or any other audio app, select a single sample, and press Play, as far as I know you will either hear nothing, or else you might hear a momentary "tick," which you cannot even begin to make any sense of as a musical sound. (Of course you can VISUALLY inspect a single audio sample in an audio editing app, but that's not the same thing as being able to LISTEN to that sample in that way.)

(b) Perhaps even more fundamentally, a higher sample rate BY DEFINITION cannot restore transients, because a sample rate of 44.1k is ALREADY capable of capturing the "fastest" transients that any human being can hear. This is a very basic and common misconception about digital audio. IMHO it's actually a quite understandable misconception for us non-experts, but for someone like Rob Watts it's either a stunning level of ignorance or a knowing act of deception. This becomes clear when we realize that the frequency of a sound literally IS its "speed." Any transient that is "too fast" to be encoded by a 44.1kHz sample rate is also, BY DEFINITION at too high of a frequency to be heard by humans.

So the test was "doomed from the start," yes - but @amirm tested it because the only way you can scientifically address claims like the ones Chord makes for the M-Scaler is to test those claims. Testing a device that could not do what it says was not a great option - but it was a better option than the alternative, which would be to refuse to test it and then leave open an opportunity for others to say, "you claim the M-Scaler can't work but you won't even test it - that seems very unscientific and close-minded."
While I knew most of this--you explained it very well for a guy like me. Nice job!
 

mansr

Major Contributor
Joined
Oct 5, 2018
Messages
4,685
Likes
10,703
Location
Hampshire
Yes, from memory, the impulse response of the system will be the filter coefficients (assume it is a filter) of an LTI system, and characterises the frequency response etc.
If the filter is 1milltion taps, then 1milltion samples will be the output ?
I'd also like to see this done with the M-scaler. In fact, it's the FIRST thing I'd do if testing one. I can't understand why nobody has done it. Then we'd all get to see what that Rob's magic window function actually looks like.
 

voodooless

Grand Contributor
Forum Donor
Joined
Jun 16, 2020
Messages
10,371
Likes
18,287
Location
Netherlands
I'd also like to see this done with the M-scaler. In fact, it's the FIRST thing I'd do if testing one. I can't understand why nobody has done it. Then we'd all get to see what that Rob's magic window function actually looks like.
Looks like there is some mechanism in the M-Scaler that makes it not output an impulse. It detects it and output something else. You’ll need to get the impulse by other means.
 

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
16,004
Likes
36,218
Location
The Neitherlands
Then we'd all get to see what that Rob's magic window function actually looks like.

1115hugo.Chugofig01.jpg

Hugo TT impulse response 44.1kHz sample freq. Looks really nice for a test signal that can not exist in a recording :).
No reason to think this will improve/change much with the M-scaler... LOL... maybe at -300dB we might see some differences.. :D
Technically the Chord filter is about as good as it gets in physical DACs.
Audible ?
 
Last edited:

Shadders

Member
Joined
Jan 5, 2019
Messages
31
Likes
30
Location
Londinium
1115hugo.Chugofig01.jpg

Hugo TT impulse response 44.1kHz sample freq. Looks really nice for a test signal that can not exist in a recording :).
No reason to think this will improve/change much with the M-scaler... LOL... maybe at -300dB we might see some differences.. :D
Technically the Chord filter is about as good as it gets in physical DACs.
Audible ?
Hi,
The lead in and the tail of the impulse response is not like a low pass filter i have seen. Is the high values at the start of the impulse and at the end, due to noise shaping ?
Regards,
Shadders.
 

bearcatsandor

Member
Joined
Nov 11, 2020
Messages
59
Likes
98
Hi Chester,

you are free not to follow this thread.
What if other people have a different interest?

Personally, I was active on Head-Fi before slowly "moving" to ASR, and, while on Head-Fi, I was influenced (I might say "deceived") by all the praise for Chord products, to the point of being subjectively deluded in hearing some improvements when auditioning Chord devices, incI. the M-Scaler. (Luckily I was never entirely convinced, helped by the too high prices and poor design, and didn't purchase any except a Mojo).
So I am very interested by the present thread and the Hugo-2 one, enjoying both greatly.
Thank you so much, Amir & ASR community.
For what it's worth (a lot), the fine folks over at Hydrogen Audio Forums have a similar no-bullshit claims bent to you folks. They even have a warning system in place should you claim that one component sounds better than another, should you postulate without abx testing. They helped cure me when I was an audiophool. We probably have some members in common.
 
Top Bottom