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SwissBear

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Everything posted by SwissBear

  1. SwissBear

    HQ Player

    Hi Miska, Trying to plan for my next move: NAA will be on a small fanless computer, with a very low noise/low impedance LPSU server will be outside the room : how important is it to go fanless for the server ? Is it better to have a powerfull server with a good GPU card or to put more money in a fanless one with expensive LPSU ?
  2. SwissBear

    HQ Player

    Checked the configurator and it seems you are right. Thanks for the heads-up. The video is misleading
  3. SwissBear

    HQ Player

    Just a short complement to my previous answer: the Devialet is not a digital amplifier but a hybrid one. I will not develop further as this is not the topic here, but if you are interested, you can have a look at the technology here: https://vimeo.com/93141227
  4. SwissBear

    HQ Player

    Yes it does. Thank you very much for your time
  5. SwissBear

    HQ Player

    Thank you ecwl. You have a good point here. This is the way the Devialet is handling the signal. Having said that, some users of Devialet have invested in other DACs, totaldac for instance, and reported positive impressions. But these were PCM 24/192 DACs and not DSD ones. Not sure is this is very different as the DAC is producing an analog signal. Thanks anyway.
  6. SwissBear

    HQ Player

    Thanks bogi. I understand that HQPlayer is upsampling in one tap. My question was probably not well formulated. I meant: if upsampling from 16/44 to DSD 512 is a 100 segment, how big are the 16/44 to 24/192 resp. 24/192 to DSD512 sub-segments ? Are we talking 50/50, which would mean to I am contemplating a further considerable improvement compared to what I have, or 80/20 or even 95/5 in which last case I should not consider this further investment.
  7. SwissBear

    HQ Player

    Hi, I have been using HQPlayer for a few months. I am currently running it together with Roon on a MacMini Late 2014 with i7 processor. My setup is MacMini -> SOtM sMS-200 -> Mutec MC3+ USB -> Devialet D900. HQPlayer is used to upsample eveything in 24/192. I am very happy so far. I have read very enthusiastic reports of users upsampling to DSD 512. Having said that, Miska often pointed out the fact that upsampling from Redbook, the first step (I guess from 44.1kHz to 88.2 or 96kHz) was the most audible. So my first question to the community here would be: is there a real up
  8. Hi Miska, Is it possible to use a Windows box as you described as a NAA device, and still use a MacOS box as a server to have Roon/HQPlayer ? Thanks.
  9. Hi Alex, When announcing the LPS-1, you wrote: "That is cool, because over time (definitely NOT this year) we could readily scale the core tech of John's efforts to larger supplies, at higher voltages and higher currents." Do you still have plans to make such a larger LPSU ? Any plan to have a version with two different outputs, like an upgraded JS-2 ? I would love to use this to power a µ-rendu/SOtM sMS-200 on one output and a Mutec MC-3+ USB on the other output. Thanks.
  10. If you have a DAC which has a capable AES/EBU or S-P/DIF input, I would like to add that the µ-rendu, combined with a Mutec MC-3+ USB, is an absolutely fantastic combination. I have this combination configured with the recommendations of this thread: http://www.computeraudiophile.com/f10-music-servers/novel-way-massively-improve-sq-sms-200-and-microrendu-31110/#post618454 ie µ-rendu connected on a bridged MacMini Ethernet interface instead of going through a switch. I am feeding this with HQPlayer in NAA mode. HQplayer is doing a tremendous job in the fields of upsampling and corre
  11. Hi everyone, How far is it necessary to go in the optimisation process of the PC/Mac which is used as a server, when using an SOtM sMS-200 ? Is there still value in having a fanless PC, with a LPSU or, given the design of the sMS-200, is a standard PC/Mac convenient enough ? Thanks in advance.
  12. So you switched to Ralf's hardware. Good choice IMHO I have kept a very good impression from my tests of this piece of equipment. Congratulations ! What is surprising though is the need you felt to have an Intona before the Acousence, which is superbly equipped in terms of galvanic isolation. Any comment on this ?
  13. Hi Fujak, Glad to see you here Are you using a Mutec MC-3+USB at the output of the SOtM smS-200 ?
  14. tbooe, I would sincerely invite you to spend 10 USD and buy this short e-book, which is very instructive and will give you a lot of simple answers: https://www.amazon.com/Accurate-Sound-Reproduction-Using-DSP-ebook/dp/B01FURPS40/ref=sr_1_1?ie=UTF8&qid=1481324367&sr=8-1&keywords=accurate+dsp Having said that, what you are looking for adjusting this excess phase window value is to lower the Group Delay, which is a measure of the excess phase. Here, measured with REW, an illustration between two measures, one without correction, and one with correction. Also, what is
  15. Hi Jud, You are right. But if we want to be extensive, there is also this time masking theme (http://www.bodziosoftware.com.au/pre_post_ringing_ir_and_pulses.pdf) which makes pre and post-ringing not always audible. So to be more into the details, one should not only look at the step, but also at the ETC (energy time curve) and make sure it keeps below thresholds. Then it might get a little bit too complicated for this thread. My point was just to explain how one should set-up some parameters in Acourate Just to illustrate, this is post-ringing:
  16. To illustrate, the green step response is showing pre-ringing, whereas the red one is exempt of pre-ringing. It can be ripples, or smoother variations before the time = 0 of the impulse. It is not always audible though, but should be avoided... And this is a real life measurement: Probably not audible though.
  17. Hi tboooe, the values of the ExcessPhase Window are the size of the FDW (frequency dependent window) you are using to make the time domain correction. The excess phase is the phase in excess of the minimum phase. Acourate tries to reduce this excess phase to match the minimum phase as closely as possible. The larger the value, the larger the size of the FDW, and the closer your corrected phase will be from the minimum phase. If you start with a value of 3, you are taking only 3 cycles into consideration. This is a moderate value. If you take 6 cycles, you are probably taking a reason
  18. Hi everyone, Then last version of rePhase, which is implementing the interface with REW, is available: rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 135 - diyAudio Thanks again to Pos and John for making this happen.
  19. Hello, Today, I would like to illustrate a slightly different use of REW / rePhase, in the field of simulation, adapted to pedagogy or design of systems. But how can we achieve simulations with rePhase ? It's very simple starting from a blank page, we create a filter, for example high pass like this: This filter is a minimum phase filter, and we can generate a pulse from it, for example, with the characteristics on the left of the screen. We will do the same with a linear phase filter of the same characteristics: The corresponding pulses are generated and im
  20. A short update to mention that, starting with REW 5.17 Beta 14, an XML interface has been substituted to the text file format. To save the filters to the .xml format, just use this function: The rest of the procedure is unchanged, Pos having changed in rePhase the format of the expected file from REW.
  21. Hi everyone, I wanted to share a new feature of REW/rePhase which makes combining the two pieces of software to make amplitude corrections in the frequency domain very easy. Since v5.17 Beta 12, REW incorporates a rePhase EQ filters set, which makes it easy to generate amplitude correction filters aimed at being used with rePhase. In the EQ tab of REW, select rePhase as an equalizer and generate your filters. You can then export these filters as text. You filters file should look like this: Then, in rePhase, you can easily import these filters: in order to apply them as your correc
  22. John has agreed to develop an interface between REW and rePhase (rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 128 - diyAudio). This might be the sign that he does not disagree with the interest of making time domain corrections as well
  23. Hi Michael, Could you please elaborate on this ? Here are the links to rePhase: - tutorial (Guide to Speaker/Room Correction Using Freeware and JRiver) - forum (rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 130 - diyAudio) I do not have any experience replacing passive cross-overs with active ones. But my experience using rePhase as part of a room correction suite with REW as an alternative to Acourate has taught me that aiming at minimum phase corrections was the only way to get rid of pre-ringing and that linear filters were a significant source o
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