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Upsamplers, DACs, jitter, shakes and analogue withdrawals, this is it.

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I think the issue is that it's an extra set of filtering ...

If you use a DAC with ASRC, you end up applying three filters to the audio:
1. ASRC resamples the data to try and reclock the data (to reduce incoming jitter)
2. the oversampling filter built into the DAC chip then apply oversampling on the resampled data
3. Finally, the analog reconstruction filter converts the output of the DAC (which is not entirely analog - typically it's a set of multi-level current pulses) into an analog signal

Some would argue this is one filter too many, since there is no such thing as a "perfect" filter.

ASRC would make sense if there was a way of bypassing the DAC oversampling filter (unfortunately, not really possible with today's flagship DAC chips which use a "hybrid" architecture)

Non-oversampling would make sense only if the DAC chip was "perfect" (unfortunately, the NOS type designs tend to use the TDA1541 which is notoriously non-linear at low levels - not to mention it's only a 16-bit chip anyway)

And bypassing the analog reconstruction filter is usually not a good idea (results in too much ultrasonic artefacts).

For me, KISS is the best approach - just use a modern flagship DAC and live with it's oversampling filter (which is required to convert the signal to the hybrid format required by the DAC chip), feed it a signal that's as clean and unprocessed as possible, and use an analog stage that's as simple as possible that meets the DAC chip requirements.

Sounds simple, but in practice very few designs get there. Even a lot of pro audio cards like the Lynx and RME use a PLL generated clock which in my opinion is almost as bad as ASRC. And for consumer players, we have all sorts of issues which have been discussed here ad nauseam.

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  Michael Percy Audio  

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