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Ryan Berry

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About Ryan Berry

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  1. Hi Csaba, Hard to say until we're done, really. USB is easy. The rest is more complicated. As far as timeline, the more time we get to spend on it, the better we get to make it. Give us some time and we'll have something good in the end. Ayre has a long history of making changes in units available to existing owners of our products. It's simply not right to add something and then have no path for someone who already owns the unit to be able to have the same cool features, so it will be available to all once we're happy with it. Cheers, Ryan
  2. Hey Ari, that's great news! Thanks for such a quick update. If there's anything you ever need from us, just give me a call and we'll be happy to provide it. I really appreciate the kind words. Cheers, Ryan
  3. With the QB-9 being asynchronous, you're likely getting little benefit from external reclocking solutions. However, I think people are seeing an improvement with devices like these as they add an extra level of isolation on the incoming power. Your QB-9 uses the PC's power and ground to power the USB PHY chip prior to isolation from the rest of the digital circuit. We then send it over to the DAC after separating the ground and adding our own power supply for the rest of the digital circuit. Having a separate power supply from the PC going into that PHY chip would very likely have varying levels of improvement depending on the PC. Of course, as you said, the best way to tell what you prefer is to try it out and let me know what you think. I may want to pick up one for myself! As far as improvement, I think you'll be pretty blown away going from a 24/96 QB-9 to the Twenty. Each upgrade we've done on this particular unit has been pretty spectacular in how much it improved from the previous version, so jumping straight to the Twenty is going to sound a lot different for you, I'd wager.
  4. Thanks for the report. We'll be in touch and see what needs to be done. Cheers, Ryan
  5. In some ways we agree. Many of the processes you're talking about is not done in the DAC chip in our units. Allowing the DAC to handle everything is easy to do, and I'm sure it's really tempting for some to slap a chip on a board, feed it power, clock, and signal and call it a day, but we've found that it doesn't sound particularly good. In other ways we're probably not going to see eye to eye (or perhaps ear to ear) here. I think one of the tricky things is that it can be difficult to understand what is actually happening inside of a unit without knowing the designer's implementation. We're fortunate enough to be able to turn off various functions of the DACs we use, disable or enable filtering, listen to various external programs handling the upsampling, etc. before we make decisions. Ultimately, we picked what sounded the best to us that also gives listeners the most consistent results after very exhaustive testing. Happily, we can do all that we're doing AND focus on the items you mentioned, so it's a win-win. I'll agree on one part with R-2R DACs...they're damned expensive to do well. Charley roughed together an idea that cost well over $1000 in parts alone just to put together. It's not whether or not it was doable at that point, but a question what would people actually gain in the process vs. what would be gained elsewhere in the design. Like I said, we don't completely disagree on several points, just feel there are better gains in different approaches.
  6. And impedes manufacturers from doing what we know makes a large positive change in the units. I fail to see the advantages of attempting to drive the DSD standard on listeners. All it's really doing in the end is bypassing filters, which means people are often comparing apples to oranges and not realizing it. Add in times when things are getting "dressed up" in DSP on the software end, that even tricks people into thinking the DSD is somehow better.
  7. Unfortunately, yes. In a way, it's worse than before as people continue to redefine DSD to give it new life with standards never defined by Sony.
  8. I can at least give you an old one! http://www.the-ear.net/features/dsd-under-fire
  9. The more I read, the more questions I end up with. Are the files being tested conversions of a PCM file or rerecorded with a DSD ADC? What if the ADC being used for PCM had a filter like the QA-9 instead of a brick wall filter? How is the DSD vs. PCM data handled inside the DAC? The list goes on and on and every test is likely flawed in some way.
  10. That's a fair point that I wasn't thinking about in DSD terms, Miska. Good catch. DSD passes directly through to the ESS untouched, so it wouldn't be subject to another round of oversampling like pre-upsampling PCM would. It doesn't eliminate some of our other issues with the format as a whole, but you're right that it wouldn't be subject to double-oversampling.
  11. That's a big topic full of a lot of opinions. Some groups studied could tell no difference while a study in Tokyo suggested people could hear the differences between DSD and PCM with a preference toward DSD (though no differences heard between different rates of DSD, interestingly). I've not heard much benefit of doing a conversion to DSD myself and I know that Charley really didn't care for the format, but that was probably more on principle than anything. Either way, I tell people to trust their own ears and just enjoy the music. The argument on what is right or wrong would go on endlessly and we'd all be sitting here debating on forums instead of listening to music.
  12. I know HQPlayer pretty well, we've used it in the past for testing and agree that it's a nice piece of software. However, you will still run into a second pass of oversampling being done at the FPGA to get to 16X and apply our Minimum Phase filter, so you'll run into what I mentioned before.
  13. Hey Chris, I see. To be honest, we've never been big proponents of upsampling outside of the FPGA, where we can do everything in a single pass to minimize rounding errors as well as ensure that the original data is preserved and not rounded out by some software post-process. We only can guarantee what WE do internally, so this gives us a bit more control on making sure the end product is as true to the original as possible. I'm sure you already know how we tend to think of DSD based on posts from Charley over the years, but I still think that converting a 192kHz sample to DSD is a fair argument for a 48kHz-based DSD rate being available if you're really set on doing it. Let me discuss that aspect with Ariel. It's just a matter of adding a line to the code and figuring out if there's a good way to display what it's doing with the characters available on the display. Well, it CAN operate at 100MHz. I wouldn't ever recommend using it that way.
  14. H Hey Chris, The QB-9 has handlers written for DSD256 at 11.2896 MHz but not 12.288MHz right now. We looked into this before releasing the QB-9 and could not find any source material that was recorded in a DSD format with a multiple of 48kHz, so it came off as more of a marketing gimmick to us than actually a useful feature and only adds confusion to what DSD256 is. Adding a handler is REALLY trivial, as I imagine it would be for most DAC manufacturers. The only hesitation is if there's any actual gain doing so. If there's some source material out there we're missing, let me know and I'll talk with Ariel some more about it.
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