That’s a fallacious argument (the Perfect-Solution Fallacy to be precise). We do not need a “truly perfect” filter as we do not have “truly perfect”: Ears, listening abilities, amps or speakers/HPs.
I agree we almost certainly don't, but that wasn't what I was saying.
Objectively speaking, to achieve perfect reconstruction we need a perfect filter. Where the audible limit lies and for what factors (there are multiple issues/factors to different reconstruction approaches and tradeoffs) is up for debate and there is very little study on the matter.
Furthermore, we don’t need a filter to be “instant” (and that would be impossible anyway), a latency of say 200ms is literally the blink of an eye, actually a particularly fast one (most eye blinks take 300ms). And, filter attenuation to about -90dB is perfectly acceptable/inaudible under any reasonable listening conditions and most DACs, even relatively cheap ones, achieve attenuation lower than that.
Instant is not referring to delay/latency, it's referring to the filter design itself. Instant meaning that everything below 22.05khz is passed through completely unaltered and everything above that is entirely eliminated. As opposed to partial attenuation or slower rolloffs.
The MScaler doesn’t do any reconstruction, it just upscales/upsamples and outputs a digital signal. So how does no reconstruction at all “achieve better reconstruction”?
The upsampling IS the reconstruction filter. It's the same process as done internally in any oversampling DAC just done externally. The maths/process is the same ignoring filter design/performance differences.
Furthermore, by requiring a “
lot more compute power” to implement millions of taps it has particularly poor latency. And if that’s not enough, it’s upsampling filter while very abrupt, doesn’t even attenuate to -80dB (still inaudible in the vast majority of reasonable listening scenarios). It also seems to have particularly poor jitter performance (although again at inaudible levels). -
Measurement source ASR.
Latency is an inherent tradeoff to high performance reconstruction filters yes. And in many production environments where latency is key this could be an unacceptable issue, but for straight up listening, generally it doesn't matter if your music starts instantly after hitting play or a second or two later.
As to the -80dB claim, that's not true and is unfortunately a result of Amir using less than ideal testing methodology and not setting up the device correctly. At 192khz output the attenuation is over 120dB at least:
https://goldensound.audio/2022/03/17/chord-hugo-m-scaler-measurements-and-technical-evaluation/
And due to how noise shaper effectiveness is linked to conversion ratio, at 768khz output it's better, though difficult to test due to the dual-BNC output so I'm not sure what the exact figure is when in use with a Chord DAC.
Though regardless the 80dB claim is simply not true and is down to Amir's testing not the device's actual performance.
I don’t see the point of that? Of course some filters will be audibly different, especially ones designed to be audibly different, say NOS emulation filters or any other filter with a roll-off easily within the audible band. If you take a decent filter though, say the typical filters used for the last 25+ years or so, a linear phase filter with a roll-off starting around 18kHz-19kHz and a transition band of around 2kHz, good luck ABX’ing these.
I'm comparing filters that are higher performance than those inbuilt into most DACs. Not 'slow' or broken ones.
No there’s not, there’s numerous devices and there has been for years. Of course, that depends on exactly what you mean by “very high performance filters”. I take it to mean any filter that does it’s job without audible artefacts.
By high performance I'm meaning those using at least a few thousand filter coefficients. Most DACs use 128-1024 taps which is very low.
As to 'without audible artefacts', my testing has shown that the limit is surprisingly high. Video coming soon