There is zero proof of that..
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Actually, there is, but as I hadn't indicated what would be better or why, it's clear you are merely looking for an argument.
-Good luck with that.
I now see Guerilla was correct.
Are you kidding me? Someone reacts to an almost 2 year old post of mine with an ad hominem and now I’m looking for an argument?
If you think I’m wrong, tell me how and why with reason and logic, not with assertions or pointless ad hominems.
And yes, my answer to you was also just an assertion.. see how annoying that is?
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I do think you’re wrong about there being “zero proof” that higher is better.
I do think your statement (that Guerilla referenced) showed ignorance of the basis of sampling theory, and I thought simply pointing out the concept of the Nyquist-Shannon theorem would at least help. (I didn’t recognize how old that statement was).
But think your response of declaring “zero proof” to a minor point (meant to be helpful) actually belies your attitude toward intercourse in this subject. Seems like you’re primed for a fight – and I don’t care. So, I’m inclined to avoid any further interaction with you.
Still, I at least wanted to respond, for clarity, a few of the reasons that I said ‘higher is better’ in terms of sampling frequency.
First of all, a slightly higher sampling rate should be used to be sure you don’t cut off too low, because clocks are not perfect.
Also, I believe the modern interpretation of the Whittaker-Nyquist-Shannon theorem is frequently stated explicitly that there must be no sinusoidal content at exactly 1/2 the sampling rate. In other words, the sampling frequency must be higher than the 2x the highest frequency to be replicated.
Of course, anything excessively high can allow aliasing, so 'higher is better' is not an open-ended statement.
Nyquist-Shannon theorem applies to bandwidth-limited signals.
The ‘Nyquist criteria’ is typically discussed assuming sinusoidal signals, for which it works quite well, no argument. But non-sinusoidal signals appear in real life (much less in music and recorded audio, but still. Particularly in electronic music).
In a system where there may be a goal of replicating audio that might include non-sinusoidal content, the 2x sampling rate still applies, strictly speaking. All good, so far. But for some non-sinusoidal waveforms, such as triangle waves, or square waves, you need to consider the highest frequency component of the signal. That’s practically impossible, depending on your fidelity goals, because triangle waves and square waves are not bandwidth-limited in nature. In these cases, ‘higher is better’ – within reason.
-I know, an argument can be made that non-sinusoidal wave replication is outside the area of interest in an audio forum, but that’s an opinion, not a fact.
Even in sinusoidal, bandwidth-limited replication, oversampling can reduce approximation error in many modern DACs that use zero-order-hold for example. In these cases ‘higher is better’.
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This is not because of non-perfect clocks, it’s to give the filter time to attenuate enough while maintaining enough passband.
Yes, that’s what the theorem says.
Not with proper filtering.
Well, that’s my main point really. Adding more samples doesn’t add more information in the bandwidth you need it. That’s what the theorem is all about. 192 kHz doesn’t give more information below 20 kHz than 44.1 kHz sampling.
I don’t think (non)-sinusoidal is a proper term for this. It’s about capturing the information of a bandwidth limited signal, and for that the sampling theorem applies, no matter the wave shape. That’s why you have to band-limit the signal before you do the capturing. If you want to capture a square wave, choose how much bandwidth you want to spend on it, and pick a sampling frequency accordingly. Higher rates in this case, may yield a better result in your scope. And you can argue that it better represents the original. That’s fine, no argument there.
But note that the argument was about an already band-limited signal! If you lowpass some audio signal halfway it’s spectrum, you logically also only need half the original sample rate to capture all the information contained in the low-passed signal, as the sampling theorem proves.
I view zero order hold broken by design
. Yes, oversampling makes it perform better, but why bother.. because it can do better square waves that aren’t even in the source signal?Was it? I didn't see you make that point. You didn't make any point.
Yes, that was the topic almost 2 years ago.
I wasn't reading those 2 year old messages (just the one Guerilla referred to, because you invited education...)
-but now that I go back and check your reference.....
and what do I find?
- Is this voodooless preferring to see higher sampling rate than the 2x maximum frequency of interest?
- but why, even with "zero proof"? /s
You seem to be self-contradicting, arguing for the sake of arguing...
No I don’t, and the reason is already explained in my previous posts: leave some room for the filter to do its task and move all effects of it out of the audible band.
This is not higher is better. If so I would think 96 kHz would be better. I do not think it is.
I'm fairly confident they can make a firmware version that does add a full-range output to the SW, but I seriously doubt they would. It seems like a bad business decision, to invest time & resources (testing) for a f/w update, for an almost non-existent market demand. A full-range mono output may be useful to you for an outdoor sound reinforcement, but it's not likely to be a very popular option on a stereo amp, even a 2.1.
That said, it never hurts to ask - try to contact someone at SMSL/Sabaj.
Roland68
Major Contributor
I've already asked twice and even offered help the second time, but there was no response.
Due to the ICs used in the device, it is very likely that full range sub-out is not possible, as are other changes to the 2.1 functionality.
Unfortunately, the ICs used in addition to the Axign AX5689 indicate that Sabaj and SMSL neither understand the capabilities of the Axign AX5689 nor are using them correctly. As a result, from a purely functional perspective, the A30a and VMV A2 remain far below the realizable possibilities.
It's a real shame.
A quick look at the datasheet shows that it basically should have most of the basic bass management features of a modern AVR, including proper channel mixing, crossover selection and large/small settings for the main speakers. Highest crossover seems to be 160Hz by default, but I suspect you can do stuff with custom biquads as well.
So it’s safe to say that the potential not understanding can be extended to the STA309A as well. Well, that’s probably only half of the story. The second parts is simply not bothering to implement these features.
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Its not a problem or a limitation. As long as sub does what its supposed to, there's no reason for the mains to go lower. When low cut they play cleaner with less distortion. Keeps its woofers further away from their xmax and their coils in the magnet field. You bought speakers that goes unnessary low, but that's just a luxury-problem.
Cheers!
GiBo61
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Another very positve review of a Sabaj product here on ASR (Sabaj A20h Balanced Headphone Amp Review).
It's a pity that our beloved A30a has not been tested by Amir.
It's a pity that our beloved A30a has not been tested by Amir.
D
Deleted member 68952
Guest
Hi to all,
I ended up buying this Sabaj. Way better than the SMSL. Everything is better. It drives my speakers with authority and refinement. I am glad I bought it !
Zek
Major Contributor
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Which amp specifically, SMSL has a very wide range of amp products.
D
Deleted member 68952
Guest
I quoted myself, I had the ao300
You can't go wrong with the A30a.
A30A gets quite hot after a few hours, even if not working, but last weekend I was away from home and forgot to turn it off so I was very concerned about it.
Surprisingly, after three full days the temperature was apparently the same, it's hot but not untouchable, if I must guess I'll say like 45 to 50 Celsius degrees, maybe 60-ish at worst.
Just for my curiosity, is there anyone with engineering skills who can explain to me the source of heat in this kind of digital amp?
My former class AB amps were not this warm when they were switched on but not in use, the only one that had similar behaviour was heavily biased towards class A, so I sold it because I don't like to waste so much energy (and paying for it).
By the way, when the A30A is hot, it sounds way better, and I'm still impressed by the quality of this amp, so impressed that I suddenly stopped reading reviews of other equipments in the last months, it was a long journey, almost 20 years since my first encounter with a Tripath TA2024, but I think I've finally found my ideal cheap way of listening to music with pleasure.
Surprisingly, after three full days the temperature was apparently the same, it's hot but not untouchable, if I must guess I'll say like 45 to 50 Celsius degrees, maybe 60-ish at worst.
Just for my curiosity, is there anyone with engineering skills who can explain to me the source of heat in this kind of digital amp?
My former class AB amps were not this warm when they were switched on but not in use, the only one that had similar behaviour was heavily biased towards class A, so I sold it because I don't like to waste so much energy (and paying for it).
By the way, when the A30A is hot, it sounds way better, and I'm still impressed by the quality of this amp, so impressed that I suddenly stopped reading reviews of other equipments in the last months, it was a long journey, almost 20 years since my first encounter with a Tripath TA2024, but I think I've finally found my ideal cheap way of listening to music with pleasure.
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