Filburt

Happy Birthday Al!

So...what is MQA then? The nearly-contentless descriptions in the links make it sound like a method of dynamically switching bit depth and sample rate based on analyzing the parametric requirments of the input, utilizing a psychoacoustic model. In which case, it's doing a less sophisticated form of what so-called "lossy" codecs do. I don't really care to pay AES $20 to read the white paper, though, in order to find out what they're actually up to.

Happy birthday Jack!

With modern off-the-shelf parts it really isn't as challenging to make something that sounds pretty good, than was the case several years ago, and it can be done for cheaper and in a smaller factor than was practicable back then. Just following the reference design is enough in many cases.

I'd pass on it.

"I knew we shouldn't have trusted the dingoes to assemble them! Our reputation will never recover!"

Oh...yeah, I figured... It was just ironic enough that I thought I'd mention it anyway *shrug*

Ah, OK. So here's the worst part... Since the Sabre32 runs at 32 bit precision internally, it doesn't have to truncate in order to attenuate even a 24 bit input. I don't recall if they use floating or fixed-point math for this operation but even at fixed point one should be able to get at least around 50dB before truncation. Also note, everything will be attenuated, including the noise floor of the recording. In practice, you'll be limited by the analog noise floor of your system long before you're limited by internal precision on the DAC. Thanks to the exceptional SINAD of the ES9012/18, you could probably get at least 10-20dB of attenuation without significant loss in dynamic performance at its output (so, then, it's up to the rest of the system to keep up). On the other hand, with an analog volume control, additive noise will go up as you attenuate in addition to other potential maladies (e.g. crosstalk, distortion, and so forth).

I'm not familiar with the LCD-X. I like the LCD-2 with GS-X. Not sure how similar it is to the X, though. Also...why did Oppo use an analog volume control on the HA-1? There is no "re-digitizing" involved using the internal ("digital") volume control on the Sabre32; it's just a multiply on the incoming data. This is more or less the sort of operation you're doing with an analog control too (well, ideally), except in practice you'll pick up some additional noise and distortion (e.g. phase and harmonic). I also don't recall pots typically having great crosstalk specs, but maybe that's changed since I last looked.

I have a smaller model from Zojirushi that I received as a gift recently. So far, my bread has come out really nicely. I'd love to try sourdough, but I don't know much about how to make that one happen successfully at the moment. On another note, sourdough pancakes sounds amazing. Steve - very interesting info re: starters! I'd not heard that before.

Whoa. Uh...I'm at least a bit surprised that kickstarter permits that.

They seem to be pretty good at delivering new indiegogo campaigns. Maybe they should have an indiegogo campaign to deliver future indiegogo campaigns, to show that they can deliver something promptly.

I'm still not particularly clear on what is supposed to be the advantage to transmitting DSD ("1 bit" PCM) over 24 bit LPCM in these cases. Why not just use the foobar plugin (or one of the other media players that plays this format)?

The plan with your friend sounds reasonable. With regard to the more technical question, I guess it's a matter of whether you want the short(er) or the long answer. The short(er) answer is that I would estimate that the use of 'audiophile' cables that substantially depart from the design and construction of ordinary commercial-grade designs would generally correlate with reduced (rather than improved) performance if any change in system dynamics were observed at all. However, equipment destined for working primarily within audio-band frequencies, at least those with well-designed power supplies, should generally be able to handle less than ideal conditions with regard to the design of your power interconnect. So, you may be able to use the device pictured without degrading the performance of your equipment to a degree that would be noticed in ordinary use cases. As for the longer answer, I'm not sure I feel like writing it. However, probably the most germane aspect of that to note would be that the "weakest link" aphorism is apt to mislead intuition and mischaracterize the dynamics of systems, so I don't recommend it when thinking about how to improve the performance of one's audio equipment.

...looks like the DSD cargo cult is really taking off. Who wrote this "white paper"?