Currently have a Prisma NP5 streamer (bought used) connected digitally to Digital Input 1 of my NDX - small experiment in streaming Qobuz etc. NP5 internet connection is Wi-Fi through an EE 4g router connected to an AirPort Extreme Gen 6 as a wireless access point and switch.
All working Ok - but I have a question regarding re-clocking of the digital input to the NDX - when the input is set to ‘stable’; I see that the NDX ‘cannot re-clock audio’. When set to ‘unstable’, the message disappears but the instructions suggest a small degradation in SQ.
What does this actually mean? I’m unable to fully assess the SQ at the moment since its all playing temporarily through a Ruark Audio R4i whilst the main system is being stored.
Naim’s buffer or memory method of jitter removal relies on a simple concept: the audio data is clocked into the memory at the incoming, inconsistently timed rate and is then clocked out of the memory and into the DAC chips using a precise clock. The rate at which the memory fills and empties is controlled by selecting the master clock that best matches the average incoming clock frequency. In this way the data entering the DAC chips is completely isolated from the incoming jitter. Only in rare cases will none of the Naim NDX’s selectable master clocks be closely enough matched to the incoming data rate. To cope with this eventuality we have also implemented, as a backup, an asynchronous sample rate converter (ASRC) that will accept any sample rate from 32kHz –10% to 192kHz +10%.
When the NDX is playing audio via S/PDIF, the source of the S/PDIF signal is the master and the NDX’s master clock must, on average, match the frequency of the source clock. Some products achieve this either by using an asynchronous sample rate converter (ASRC) or a voltage-controlled crystal oscillator (VCXO) together with a Phase Locked Loop (PLL). ASRCs can work well but they rely on considerable mathematic processing, which will cause problems if not implemented very carefully. There are no ASRC chips on the market that can convert arbitrary input sample frequency to high enough frequencies that the analogue output filter can be kept fairly simple. This means that you still have to oversample the data before the DAC, so you might as well do that properly from the outset and avoid the ASRC entirely. Moreover, since ASRC relies on averaging the incoming sample rate, not all jitter will be removed. The NDX uses the built-in ASRC in the SHARC DSP only when the incoming data rate is outside the S/PDIF spec. The SHARC’s ASRC has an arithmetic noise level of –128dB and so it is only used in these exceptional circumstances.
Thanks James - not entirely sure I follow all that but my understanding then is that in essence; the NDX only re-clocks if the incoming signal frequency cannot closely match one of the NDX clocking freqs.
Does that mean that the NP5 falls outside of these parameters and the ‘unstable’ input setting forces the NDX to re-clock?
Just curious - as I say I can’t tell the difference through the limitations of the system I’m listening through at the moment.
Worth a try. It’s not going to be a big degradation and for trialing Qobuz it should be fine. I would recommend looking at the later players if you get on with Qobuz. I’ve had both NDX and NDX2, and the later is a nice step up with all the decent streaming services built in. Qobuz is great too.
Just to close this off, after having a play around I fixed the output freq of the NP5 to 96khz (as opposed to streaming ‘natively’) and the NDX locks on without issue…
I don’t think I’ve ever listened to music above this sample rate so can’t imagine I’m missing anything (not tried fixing at 192khz yet).
Looking forward to getting it all plumbed in ‘proper’ when I get some speakers to play it all through in the next few months - the NDX2 will have to wait a year or so!