Maybe the fact that this has been on his channel for almost two years?What’s your true motivation for defending a device that has no defense unless it was priced strictly for the chassis and parts?
Sounds amazing - where can I buy one …. LolMaybe the fact that this has been on his channel for almost two years?
I know I’d struggle to drop that overnight.
That, or he’s running just one more DBT.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?
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.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.
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.What I am not seeing is any evidence that the device isn't doing what it claims to do.
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.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.
Alas, no I think he got what he came for . I doubt he will be back.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?
It is possible, though slightly convoluted:But if we do zero-order hold instead of zero samples then this happens:
Code:No sox command this time :( If it is possible to do in sox, then I don't know how
sox in.wav -r 16k out.wav upsample 2 remix 1 1 delay 0 1s remix -v1 trim 0 -1s
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.’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."
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It’s important to be open minded on arriving at this forum or there is no point engaging in dialogue.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.’
I have no idea who the guy is. Big world out there on the wild web.Maybe the fact that this has been on his channel for almost two years?
I know I’d struggle to drop that overnight.
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.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
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. )