Taiko Audio SGM Extreme : the Crème de la Crème
- Thread starter CKKeung
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All Antipodes (Oladra and any other recent model) outputs except USB are derived from a separate proprietary "Reclocking Board" that not only reclocks but is also in large part responsible for the server's signature free-flowing and organic nature. TOSlink in itself is strictly speaking an inferior format that tends to round transients. But versus the relatively cleaner/more technical-sounding USB implementation in Antipodes servers, the net effect can indeed still be more pleasing and thus subjectively better.
That is correct and I would certainly endorse using Coaxial S/PDIF over TOSlink. AES/EBU, by the way, is also very similar to S/PDIF, but just like S/PDIF, it can work very well, depending on the implementation. For Taiko Audio servers, the opposite is true and USB works best. USB is theoretically ideal, but it takes a lot of effort to actually make it ideal. It's all in the implementation.
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Indeed Toslink should be able to perform competitively to SPDIF but its implementation is often done in the cheapest possible way.
A further consideration is sample rate support which per interface is limited to:
Toslink : 96KHz, no native DSD
SPDIF : 192KHz, no native DSD
AES/EBU : 192KHz, no native DSD
USB : currently supports but technically not limited to : 1536KHz, native DSD up to DSD2048
Then there is Ethernet ofcourse but that’s a different type of interface as it requires a “streamer / endpoint” to reside inside your DAC, to quote @MarcelNL , “I have yet to hear a good implementation”. There are more advanced interfaces running on Ethernet, like Ravenna, but these are specialised interfaces and exclusive to just a handful of DACs, these can also be used for high bandwidth multichannel systems.
The new kid on the block will be our own interface launching in the next few months, designed from scratch, both hardware, firmware and drivers, supporting any sample rate. Which is CPU direct over PCIe to I2S (“DAC language”). It can be clocked from both source as destination.
Why we went through the elaborate process of designing this is all the “off the shelf” solutions, which are the “engineering blocks” everybody uses, are aging and development on those have stopped. Further to this all of them have limitations in one way or another, they’re standard building blocks needing specific surrounding circuitry to operate which you have no choice in but to follow. Things like jitter reduction, reclocking, very expensive clocks etc are really just bandaids to reduce the negative impact of the existing issues in these standard circuits/protocols.
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Yes, and also a good example by the limitations imposed by having to use off the shelf solutions on which there’s no further development. Like a 192KHz sample rate limitation and ancient drivers.
In sheer size High-end audio has become a smaller market and R&D costs have increased dramatically. Hence large companies no longer invest in developing products in the “niche market” of high-end audio as the ROI is not there. We now have a very limited choice in DAC chips supporting a wide range of samplerates for example, really only 3, ESS Sabre, Rohm and fortunately AKM is back in business. Next to that availability of the high-end versions of these chips is very low, you have to order 6-12 months in advance (Rohm & AKM)..
The high-end audio companies which do continue to design improved D/A solutions keep these proprietary to themselves and sell their products at a premium to be able to get a ROI on that.
The intended product will be modular and offer either Digital or Analog outputs. With the former, you can specify "standard" digital interface formats that still allow the use of any DAC and upsampler you like. You can also specify the new Taiko interface format which will limit the connectivity to manufacturers that have also implemented this new format. The analog-output DAC version, finally, is a closed system.
The Modular Daughter Board will provide one specific function from an array of options such as:
1. A DAC with Analog output
2. Generic S/PDIF, AES/EBU, I2s, or other digital formats
3. Custom output like MSB Pro ISL or Dual AES/EBU for DCS
4. New Taiko proprietary interface
Upsampling has more to do with DAC filtering, that difference remains. The advances are more in the interface, where uncompressed PCM 24/768 will utilise about 0.00000022% of the interface bandwidth. Therefor higher resolutions are less impacted by interface “limitations” then before. IOW the difference between 16/44.1 and 24/768 becomes more or less negligible from an interface perspective.
On my system XDMS is far away better than roon. When I come back to roon, I cannot understand how it can be so different in any point.
No then we’d be back to USB, which we already have, this interface replaced USB. We’ll have output options for spdif/aes/bnc/i2s but the benefits then become limited to the internal part inside of the server. Which can still be considerable btw.
As for the modular design, a few renders may help clarifying:
The “white block” is a solid machined piece of copper which contains and shields the optional output modules which in turn plug into the interface carrier board. In this render it contains the analogue output module.
im in Washington DC looking out on Pennsylvania Avenue and reading again about all things Taiko. Good to see this thread back on topic and read about what’s in the pipe line. Kudos Emile
I apologize but i may have mixed up my terminology of the input i was using. based on the sample rates supported i believe i was using the optical input of the spdif module and not toslnink as i was playing 192khz pcm without issue. thank you very much for the clarification and information about the best input from the taiko.
Just to make sure I’m understanding this as I follow along (not trying to nit pick) should 1) read a DAC with analog input? If not, not sure I understand,
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