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mitchco

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About mitchco

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  1. Cheers @Andyman Wrt questions about Audirvana, AU plugin and sample rates, are product questions that perhaps Flavio @flak can answer as I don't have a Mac. While one could use Dirac to measure the correction result, it is typical to use a 3rd party application for verification measurements. In my case, I have been using REW for +ten years and as a general acoustic measurement software, it has several additional acoustical analysis displays that Dirac does not have. For example, one typically uses a Step response as one of the verification measurements to view the timing response of the loudspeaker, which I show in the article. Also REW allows us to share measurements in a portable format and compare measurements in overlays. I usually put "ideally" in front of moving objects out of the path between the speakers the measurement mic. Of course, it may not be possible. The issue with the couch depends on how close the back of the couch is to the microphone and how reflective the couch surface is. For example, if the mic is 6" or 12" away from a reflective surface is right in the midrange and upper midrange frequency range, where our ears are the most sensitive. Further, because of the close proximity and how reflective the couch is, the reflected sound may be as high in amplitude as the direct sound. Producing a correction in this situation will not sound very good. If there is no choice, try and throw thick wool blankets over the couch area that the mic is moving around which will limit the comb filtering in the analysis and produce a much better sounding correction filter. I encourage folks to try it for themselves and hear what I am talking about. Re: B&W floorstanders. Take a measurement with the bungs in, then take another measurement with the bungs out Which one produces the flattest response? Go with that. As a general guide, we want to try and get the best sound possible before room correction. It is not so much about less work for the room correction software but the better the setup before hand the better the correction, meaning better sound quality. Thanks Andy, I am looking forward to putting Dirac Live Bass Management through its paces! Kind regards, Mitch
  2. Both Acourate and Audiolense have preringing compensation. There is a big section on preringing in the Audiolense help manual that I highly recommend reading. This allows you to tune the group delay without introducing preringing and therefore the step response pretty much stays the same. In my book, I was showing examples of varying the excess phase to show what that does, but I was not varying the preringing control. In the end of that example, you can see that both long and short excess low frequency excess phase correction windows, with preringing compensation engaged, shows virtually the same step response with next to no preringing.
  3. Cheers. Cracking open the Audiolense manual: "The Max boost setting determines how high a dip is allowed to be lifted by the correction filter. A too high setting will consume the dynamic range of the setup. Typical setting is below 12 dB and 6 dB is a good figure most of the time. Higher values work – and sometimes a higher value is needed to get the desired result. The correction boost will also be limited by the measurement and correction window – and quite often to an extent where a 6dB max boost setting hardly makes a difference." Not to be confused with that the correction filter never exceeds 0 dBFS. This is what I was referring to in the video. As soon as you make a correction filter and examine it in AL, you will see what I mean. It is not "boosting" in the traditional sense, like a +PEQ for example. I think you said you were playing with Audiolense? I would recommend creating a few corrections filters with different so-called boost settings and examine what the correction filter looks. And most importantly listen to the impact of increasing and decreasing this setting and what it does to the dynamic range. There usually is a "sweetspot" for any given loudspeaker/room setup, plus what each persons preference is.
  4. It is unfortunate how few people really understand Digital Room Correction 😞 While Audyssey (and many others) call their products digital room correction, they are far from SOTA. Matt did a nice job of showing that with Audyssey. However, nothing in the video applies to SOTA DRC products and this is where it goes sideways. Here I explain what the technical issues are with Audyssey, miniDSP, Trinnov, etc as the tech used is limited. https://www.audio β€œscience” review/forum/index.php?threads/audyssey-room-eq-review.12746/page-10#post-380033 EDIT: I can't get the link to paste properly. Copy the link and remove the spaces and quotes around "science" Acourate and Audiolense are SOTA DRC products, way ahead of the pack, based my comparisons of these products over the last 10 years. The simulations produced in these two products are virtually identical to the measured response. I have done this dozens of times, even here on AS if you look at my articles that have a validation section you can see for yourself. My book has several comparisons of the simulations and the measurements, down to a .25 dB tolerance match across the band. What you see is indeed what you get. Wrt on axis, versus off axis measurements. In my my book, I show a detailed example of using a single Acourate measurement analysis to design to generate a DSP correction filter. I then used REW to measure at 14 different locations over a 6' x 2' grid area like this: Here are the results: It is too bad I can't invite folks over for a listen and can hear with ones own ears how smooth and even the response is no matter where one sits on the couch. @asdf1000 I have asked you before, have you tried Acourate or Audiolense yourself? If not, there are demo versions of both that you can try with the tutorials I have written on this site. You can also validate for yourself that the simulations indeed match the measurements perfectly. You can also verify for yourself the question of a single analysis measurement versus multiple measurements... and listen. Good luck.
  5. Some of us have taken extensive measurements after correction for each of the DSP products listed... multiple times on multiple systems. It is not too hard to figure out what is going on... Dirac's technology is different using a combo of IIR and FIR filters as discussed in this article, which also links to their whitepaper on their tech: Why not ask Dirac directly? Where are you in your DSP journey?
  6. Hi Jeffrey, yes, same for AL. You can find an equivalent analysis chart in the AL manual. See page 41 of the PDF help file. Acourate and Audiolense are the only DSP software I know that employs this type of psychoacoustic filtering during the measurement/analysis process. Both software also has the capability for multiple analysis measurements. I have done both (to death!). At this level of (SOTA) performance, I encourage folks to try both (i.e. single analysis versus multiple) and let your ears decide which one you prefer, as the frequency and timing response are pretty much the same regardless of which method is chosen.
  7. Thanks @Trdat Yes, as the saying goes, "there is no replacement for displacement." πŸ™‚ At reference volume, the LS50's are nearing there max output level before audible distortion sets in, whereas the JBL's are idling. But with new tech drivers like the Purifi PTT6.5, the gap is closing... Yes, classical music concert halls where the sound is mostly diffuse, for un-amplified music. This is opposed to concert halls with monster PA systems for rock bands, etc. which are designed to cover the hall with much more direct sound versus reverberate sound. There is a concept called critical distance at play here...
  8. Hi @asdf1000 Here is a blurb from Uli from the Acourate forum: "Acourate mainly uses the measurement at the listening position. This results in the best correction at thee sweet spot. But. because Acourate applies a different approach in the calculations the result is not only valid for a single position. So you can walk around and you will notice that the sound at other positions is not getting bad. The Acourate calculation avoids over-boosts by principle.. Dirac, Audyssey, Lyngdorf use different calculations by averaging multiple frequency responses. They need to do this to to also avoid over-boosts. Despite the typical single point measurement Acourate allows you to do multiple measurements and to combine them by functions like - averaging the frequency responses - calculating the max. envelope for a bundle of frequency responses - averaging multiple pulse responses" I have a section in my book on the latter point on averaging multiple pulses called "Beamforming" and you can see the procedure and results. For more of a technical explanation on the psychoacoustic filtering applied to the measurement, see my note on transient response analysis. Also in my book... As you will note in my book, the section on design verification, I moved the measurement mic to 14 locations covering a 6ft x 2ft grid area that represents my couch listening area to verify the design of the correction filter. The correction filter design is based on a single analysis measurement. The results speaks for itself. You may want to ask Uli and other Acourate users on the Acourate forum about their experiences as well.
  9. mitchco

    HQ Player

    Yes, as @StreamFidelity shows the later versions of Acourate supports 352.8 kHz and 384 kHz correction filters.
  10. Q1 - I have used AL with single measurement and multiple measurements, both sound excellent. Personally, I would go with the single measurement. AL has a psychoacoustic filter applied which does not fill in all of dips, so a) it automatically avoids over correction and therefore does not require multiple averaged measurements b) already produces a wide sweet spot. For example, my single measurement covers my three seat sofa with uniform frequency response, which I have verified with multiple REW measurements after correction. But since the software does both, it is easy enough to try both and let your ears decide. Q2 - While my book uses Acourate as the DSP software (also excellent software!), all of the concepts and procedures apply to Audiolense and any other well designed DSP software. While the actual steps are different, they are performing the same DSP tasks.
  11. To answer your questions. Q1 - yes, it is straight ahead to do mulitiseat measurements in AL. Q2 - I highly recommend True Time Domain (TTD) correction with the added bonus to be able to use digital XO's in the future. The surround time alignment, aligns the multiple speakers, but does not do TTD from what I recall. But that is a question for Bernt. Q3 - thanks Chris! Q4 - You can download AL 6.6 in demo mode: http://juicehifi.com/download/ and then you can access the PDF help file which will answer many of your questions. When you launch the program, select the XO version. Q5- The miniDSP USB mic will work just fine. If you join the Audiolense forum: https://groups.google.com/forum/#!forum/audiolense you can search on a test I did with the UMIK-1 USB mic compared to an analog mic. Using Bernt's "clock drift correction" that is in the later versions of AL, works perfect and virtually no difference in timing between the two mics. Q6 - The Lynx Hilo has 8 digital channels, 6 channels of analog output and 2 channels of analog input. Here is a simple diagram of it working and in this case, I am taking a loopback measurement of my stereo triamp system with the digital XO and correction filters loaded in JRiver's convolution engine. PS. Always use ASIO if you can, and in my case the Hilo's ASIO driver is excellent and also happens to be multi-client, which allows me to do this:
  12. Hi @asdf1000 Not a stupid question at all. It is a complex subject area and sometimes manufacturers use the terminology in different contexts or in marketing material that can also cause confusion. Time domain correction and impulse response correction are the same thing. Time alignment and phase correction are aspects of time domain correction. Lengthy explanation ahead. Time alignment generally refers to all speakers, and the drivers in each speaker, are "time aligned" so that all of the direct sound from the drivers arrives at our ears at the same time. There are a couple of ways that is implemented. The ultimate way is to create a linear phase digital crossover for each driver in a system, including sub(s). The DSP software will digital delay the signal for each driver relative to a reference, like the tweeter for example, so that the acoustic centers of each driver are perfectly time aligned. Here is an example of that in my 3 way triamp system dual subs: The delay in milliseconds is what the DSP software measured for each driver in the system. The DSP correction will ensure all drivers are time aligned in the digital XO and correction FIR filter. For passive XO systems, the DSP software will apply an overall or global time alignment correction to align the speakers/drivers as best as possible, within a certain time window that the DSP software can do (or set by the user). An "ideal" loudspeaker is a minimum phase device. Flat frequency response and flat phase response. However, when you place speakers in rooms, we get reflections, standing waves, room resonances, etc. Because of this at low frequencies, we no longer get the ideal minimum phase response, but rather areas that are mixed phase or non minimum phase behaviour. So using long, low frequency excessphase time domain correction windows (like 600ms at 10 Hz), the DSP software will correct all channels towards the same time and phase behaviour as defined by "target" response one defines in the DSP software. We are restoring the minimum phase response of our ideal speaker at the listening position. For example, once an acoustic measurement is made, the DSP software extracts the minimum phase response. Then by inverting the amplitude response and applying it as a filter to the measured response, the result is a flat frequency response. By EQ'ing the amplitude response, the phase response is also adjusted, as it is a minimum phase system. Then the DSP software, independently of the minimum phase correction, also corrects the excess phase response towards the minimum phase response so that one ends up with the ideal low frequency and phase response across a listening area. Important note: The excess phase is the phase difference between the real signal and the minimum phase response. To learn more about minimum phase John Mulcahy, author of REW has a good explanation. To learn more about time alignment and excess phase correction, these articles using Acourate and Audiolense shows what is possible. And this article talks about an ideal speakers low frequency and phase response and what happens when placed in a room. Hope that helps.
  13. @TooSteep I have had this article half complete for some time and got distracted πŸ™‚ As far as what speakers, this is up to your personal preference. The idea is that Audiolense DSP would correct for the speakers location relative to one's near field listening position. The DSP configuration is a little different as we use a shorter time domain correction window to take care of the reflections off the front wall and desk, but not so much for the room as it is a nearfield setup. I am hoping to finish the 2nd half of the article at some point using KEF LS50's in a nearfield setup, but @Archimago has my amplifiers for an upcoming testing article. Due to Covid, I may not get those back for a while... Cheers, Mitch
  14. Hi @Tp no need to remove the carpet or other objects on walls, etc. What we are trying to avoid is anything in the direct path between the speakers and microphone like plants coffee table, etc. Also, anything around the microphone can cause odd sounding corrections, hence moving the chair or sofa from around the mic. Hi @mlknez Do you mean DiracLiveProcessor.dll? On my Win10 computer it is located at: C:\Program Files\Common Files\VST2
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