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Everything posted by SwissBear

  1. The only thing you have to do is, when generating the correction impulses, to decide the width of the window you want to use to correct the excess phase of the system. This is part of Macro 4. You can refer to the web site of Acourate (http://www.audiovero.de/en/acourate.php), and look at the description of the product: You will notice that the possibility to create digital (active) cross-overs is part of the 'Further functions' paragraph, whereas the Timing errors correction via phase correction is part of the 'Solution', ie standard features.
  2. You have nothing to do. This is part of the standard Acourate correction process.
  3. @tboooe, Acourate allows you to make time domain corrections even if you do not change the passive cross-overs of your speakers for active cross-overs. Here are the changes I made to my speakers in this field, without touching anything to the HW. The impulses have considerably been improved by Acourate: and the phase has also been improved:
  4. Not sure. Phase issues are difficult. But this is definitely worth a trial
  5. Hi Keith, I probably misunderstood the statement which was quoted by tboooe. Glad we are on the same page
  6. Hi Kugel, MathAudio looks nice, but I never tried it. You have a multiple dimensions problem to solve: - one dimension is the functionalities: are you after room correction in amplitude only, or are you trying to correct the time alignment of your speakers as well. I you are in the amplitude only, REW is an option. If you are interested in amplitude+time alignment, you would need to consider REW+rePhase, or Acourate or Dirac as possible choices. Dirac is a no brainer (works for everyone), Acourate is more versatile (you can use it as a profane as well as an expert). REW+rePhase requir
  7. Hi Kugel, You can try LAConvolver (http://audio.lernvall.com/) to apply correction impulses inside Audirvana. But this is an old piece of SW which does not work well in Audirvana with large correction impulses. Damien Plisson, the developer of Audirvana is aware of this and might work on it later, after he has completed the 2.6 version of his SW.
  8. Hi tboooe, I do not quite share the comments of Keith_W regarding Acourate: this product is, in my opinion, very versatile and can accommodate the needs/expectations of the beginner as well as those of the expert. Just follow step by step the tutorial of Mitch and you will be safe. Do not try to innovate with the settings; just keep the default ones and you will produce very good corrections. Also, Uli, the developer of Acourate, is very helpful and friendly. It might be worth noting that the corrections brought by Acourate are a little more 'micro-managed' than the ones brought by REW
  9. Two other tips: - after the inversion macro (Ctrl+F3), Acourate gives you an indication of the level adjustment which is done by the correction applied. You should compensate for this correction by adjusting the volume of your system accordingly - you might need to make progressive adjustments, starting with a flat target curve, just to get used to the correction process, and gradually adjust to correct further problems; this way you will gradually get used to the new sound of your system
  10. Hi Kilroy, A few answers to your questions: - yes the sound of DSP is somewhat flatter than what your are used to as it is supposed to get rid of the sound signature of your system/room to reproduce what is on the recording; you will get used to it and recognize later that your former signature was erroneous - I would advise you to start as proposed by Mitch. His tutorial/e-book are very informative - to test different impulses, create different directories one per target curve - pre-ringing is not necessarily audible - do not hesitate to contact Uli Brueggemann from AudioVero. He is
  11. Hi tboooe, I concur with kilroy's comment that your measurement curve is very good. What you should look in the low range is if you have resonance. This is the conjunction of high levels with long energy dispersion time. See in the low frequency what happens in my room on the attached waterfall. On the corrected measurement (with RePhase+REW, but you can achieve the same with Acourate), you can notice that the resonance have been drastically reduced. Another thing to look at is the time domain. In the first measurement, you will notice that the three monitors of my th
  12. Hi tboooe, The example I gave with 48kHz was on purpose as this is the frequency most often used for measurements. Now I guess you have to learn how to use REW and make your first measurements. Here is the home forum of REW where you will probably find the best support in this venture: REW Forum - Home Theater Forum and Systems - HomeTheaterShack.com
  13. You need to write, for each and every sampling frequency, a config file with the following structure (please note the directory used in this example is under MacOS): And this is the file that you will reference in the JRiver convolution window. So you will have a different correction impulse for each channel. You could have 8 different correction impulses if you had a 7+1 system.
  14. Well, it seems you are running on Windows. So you can use a functionality of JRiver under Windows, which is to activate JRiver as a WDM driver (Options->General->Features->WDM). You can select the convolution options inside JRiver: Then, you can select JRiver as the driver for your outputs in REW and the corrections will be applied to the measures you are making. So you will be able to measure the results of your corrections.
  15. Hi tboooe, I guess we all have a lot to learn. So if we can help, don't hesitate to ask
  16. And here is a guide how to use free software to make frequency and time (phase) room corrections: Guide to Speaker/Room Correction Using Freeware and JRiver
  17. If you use filters made with REW to correct your room, you will probably only adjust things in the frequency domain and not in the time domain. As very well explained by Mitch in this article (Computer Audiophile - Acourate Digital Room and Loudspeaker Correction Software Walkthrough), as well as in his book (https://www.amazon.com/Accurate-Sound-Reproduction-Using-DSP-ebook/dp/B01FURPS40/ref=sr_1_1?ie=UTF8&qid=1476908004&sr=8-1&keywords=accurate+dsp), time domain adjustment with Acourate is a very easy solution to implement. This will provide you with the additional benefits of he
  18. Hi Keith, Thanks for the answer. I was probably not precise enough. I meant using a 96kHz sampling rate, in order to capture whatever was in the 20-25kHz frequency area, and which might be altered when measuring with a 48kHz sampling rate. Thanks for detailing your measurement setting.
  19. Hi Keith, Jumping in on this subject. You are mentioning the Focusrite 2i2 as an option for external USB card. Isn't this device limited in bandwidth to 20Hz-20kHz ? Wouldn't it be worth buying a device which has a larger bandwidth and make measurements at 96kHz for better results in the high frequency range ? Thanks.
  20. The µ-Rendu is supposed to be exempt of HF noise, as long as you feed it with an LPSU. I chose an HDPlex one, which is adjustable to 7V where the µ-Rendu sounds very well, and it also saves it for a longer term usage. If there is no HF noise here, why would I want to add an additional item, and more cables ? What I would like from Mutec is the option to feed it with a LPSU (just an additional plug), which would be very useful for the audiophile market, and avoid dangerous tricks to suppress the standard switching PSU.
  21. Well, if you really want to be on the safe side and avoid all the pollution which is generated by PCs and transferred to audio equipment via USB, you could rely on : PC -> RJ45 -> Sonore µ-Rendu (with a linear power supply) -> USB -> Mutec MC-3+USB -> DAC. This would really be a close to optimal solution. I tested it personally and was enchanted. The µ-Rendu has a mode which is compatible with HQPlayer (HQPlayer NAA) as well as a mode which is compatible with JRiver (UPnP). The former will probably be better as far as SQ is concerned, and can be interfaced with Roon, the
  22. Hi Julian, Could you please update us on the Ref10 (high precision clock from Mutec) project ?
  23. Hi RoB, It seems that resampling, upsampling aso are complex operations which require a lot of computation power and that the PCs are in this field more appropriate to do the job. Here is what is said on this subject by the developer of HQPlayer (Signalyst): "All modern DACs employ oversampling and delta-sigma modulation, however the hardware implementations are more or less resource constrained. Higher quality oversampling and delta-sigma modulation can be done by utilizing vast mount of processing power available in modern PCs. Many AVRs also resample int
  24. SwissBear

    HQ Player

    Thanks Keith for the answer
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