Topic: Multi-interface Operation at high sampling rates >192kHz (PCM/DSD) ?

With a Tascam DA-3000 recorder I can share the Word Clock with a standard BNC cable to synchronize 4, 6, 8, ... audio channels for A/D conversion like DSD128.

The ADI-2 Pro does not have a Word Clock connection, only AES, ADAT, SPDIF which are limited to 192kHz.
Does this mean that you cannot use DSD256/PCM768 (USB only) when using a RME ADI-2 PRO FS multi-interface configuration?




from pdf manual https://www.rme-audio.de/download/adi2profs_e.pdf

33.5 Digital
- Clocks: Internal, AES In, SPDIF In, ADAT In
- Supported sample rates for external clocks: 32 kHz up to 200 kHz
- Internally supported sample rates: 44.1 kHz up to 768 kHz

27.3 Multi-interface Operation
OS X supports the usage of more than one audio device within an audio software. This is done
via the Core Audio function Aggregate Devices, which allows to combine several devices into
one. All units have to be in sync, i.e. have to receive valid sync information via a digital input
signal, then all channels can be used at once.
If one of the devices is set to clock mode Master, all others have to be set to clock mode Slave,
and have to be synced from the master by feeding AES, SPDIF, Word or ADAT. The clock
modes of all units have to be set up correctly in their Settings dialog.

2

Re: Multi-interface Operation at high sampling rates >192kHz (PCM/DSD) ?

cauldron wrote:

Does this mean that you cannot use DSD256/PCM768 (USB only) when using a RME ADI-2 PRO FS multi-interface configuration?

Exactly.

Regards
Matthias Carstens
RME

Re: Multi-interface Operation at high sampling rates >192kHz (PCM/DSD) ?

Thank you for the confirmation...

It is not the most professional and simple way but fortunately RME ADI-2 (PRO) FS is a USB class compliant 2 interface and even if it is not supported by RME we can use linux to maintain a pseudo master clock through an adaptive SRC resampling using also frequencies greater than 192kHz (obviously only PCM SRC).

I satisfactorily used 2 PCI Express Asus Essence II 7.1 cards with Texas Instrument PCM 1792A chip and TCXO clock with an adaptive pseudo-sync ( https://kokkinizita.linuxaudio.org/linu … guide.html ). Approximately every 50 seconds one sound card exceeded the other of a single sample and the jitter was very stable.

In the case of various RME ADI-2 DAC or RME ADI-2 DAC PRO it will be sufficient to set each individual sound card with the internal master clock (at 768kHz for example). Apart from the first sound card all the others will be adapted with the SRC through the zita-resampler library on the jack audio linux infrastructure.

To contain the transmission jitter on the USB bus it might be useful to use some USB3 controllers and increase the sound cards buffer. We should try but only if the jitter of the whole system remains stable then we have a good quality transparent resampling.

When I have a chance to verify it, I will update the page ...

Regards,