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Sabaj A30a announced

What’s the point of upsampling on a device that resamples back to at 44.1 kHz?
 
Just curious , specs say it should but it doesn’t
Don’t you have the option, or do you not get sound? In the first case, you may need a driver to make it work. Default USB audio drivers may not expose all sample rates.
 
There is no sound or very distorted sound , I’m using a Mac so no drivers
That's just down to your upsampling software or settings.
Have you checked the settings in your Audio-MIDI-Setup.app?

Real 768 kHz files are displayed that way when played.
But as @voodooless said, the A30a only plays 44.1 kHz. Upsampling has a rather negative effect there.
 
That's just down to your upsampling software or settings.
Have you checked the settings in your Audio-MIDI-Setup.app?

Real 768 kHz files are displayed that way when played.
But as @voodooless said, the A30a only plays 44.1 kHz. Upsampling has a rather negative effect there.
Audio midi says 768 and 768 is on the dac
 
Audio midi says 768 and 768 is on the dac
Do you mean that 768 kHz is displayed on A30a, or on another DAC?

Are you using the original cable from the A30a or another USB cable.
 
Sorry i meant the A30a, i have tried a few different cables.
My advice is to stop worrying about it.

Anything above 48kHz is a waste of bits. Anything above 96kHz is a massive waste of bits, and a waste of the effort you are putting in trying to make it work.
 
Anything above 96kHz is a massive waste of bits, and a waste of the effort you are putting in trying to make it work.
Are you suggesting this because there is an up sampling then down sampling to 44.1 kHz PCM, 2 processes/transcodings, which means the original signal is changed/transcoded with the up sampling (which is not good and as such, a massive waste of bits), therefore unnecessary especially as down sampling to 44.1 kHz would require much more processing effort by the A30a and the original signal is further changed/transcoded by the down sampling process?

Perhaps it is better if you can explain it better in your words?
 
Are you suggesting this because there is an up sampling then down sampling to 44.1 kHz PCM, 2 processes/transcodings, which means the original signal is changed/transcoded with the up sampling (which is not good and as such, a massive waste of bits), therefore unnecessary especially as down sampling to 44.1 kHz would require much more processing effort by the A30a and the original signal is further changed/transcoded by the down sampling process?

Perhaps it is better if you can explain it better in your words?
No, I'm stating (not suggesting) that 48Khz is fully capable of delivering the full bandwidth that the human ear is capable of detecting.

Higher than that only delivers ultrasonics - which - first of all we cannot hear, second of all is mostly noise shaped out of the audio band, and worse - if there are significant levels of ultrasonics, they can (theoretically) inter-modulate in the speaker to create audible band distortion products that might become audible.

No audible benefit - and worse case an actual audible** degradation of sound.

This applies to all applications of higher sample rate audio - whether upsampled from 44.1kHz (which is particularly redundant), or delivered natively in 96kHz and above, (or eg DSD)


** though I confess to never having heard an audible degradation - I suspect it is a vanishingly rare effect simply because of the very low level of ultrasonics present.
 
Sorry i meant the A30a, i have tried a few different cables.
Let's make it simple: If you can play original files at 768 kHz perfectly on the A30a, then the problem must be with the upsampling.

You can download some free samples from sound liaison to test. The 768 kHz track plays perfectly on the A30a and is displayed as such.
 
@antcollinet, thanks for the insight! I understand your point of view. So SACD/DSD is then also an analog signal, according to your logic?

... And worst of all, for all the Axign fans, that chip is just an (extra)ordinary DAC, just one that it outputs PWM ;)
This is how proper Gentlemen interact, well done chaps.

Perfect example of discourse as it should be, I have hope for the Whippersnappers yet:cool:

Bravo antcollinet and voodooless for setting the example!
 
Let's make it simple: If you can play original files at 768 kHz perfectly on the A30a, then the problem must be with the upsampling.
Good point. If real files play, and the upsampled version does not, it may be a CPU load issue or simply a bug in JRiver...
 
@antcollinet, thanks for the insight! I understand your point of view. So SACD/DSD is then also an analog signal, according to your logic?

... And worst of all, for all the Axign fans, that chip is just an (extra)ordinary DAC, just one that it outputs PWM
It could suggested that until PWM can be played through passive speakers, and enjoyed (edit: enjoyable experience), that it is yet to be defined as analogue but a type of digital (even like FM (called digital) or even like DSD (called digital)).... that is reasonable, isn't it?

Yes, the Axign AX5689 chip is an (extra)ordinary DAC (perhaps DPAC), just one that it outputs PWM (utilising PFFB, similarity to a good classB design like the B100), and the A30a could be called a type of Devialet implementation.... that is reasonable, isn't it?
 
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I would like to change the American power cord/plug that was sent to me from sabaj (I live in the United Kingdom). It is rated at 110v 10amp. Obviously the rated voltage here in the uk is 240v, but do I buy one with a 10 amp or 13 amp, as that's what most kettle cables come with.
 
110v 10A will do 1100W. Michael more than the amp needs. The 240V version would need about half of that to deliver the same amount of power safely. Given that the amp delivers way less power, I would say that any cord will be able to deliver enough power to the amp.
 
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