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Everything posted by mitchco

  1. Awesome!! Mathematically speaking, aside from one's preferred tonal response, there is an optimal or "ideal" convolution filter for each set of speakers and room combination. You can see an example of that it in the charts for the "ideal" response in this article: Let's have a look at the "ideal" step or timing response in particular: Preringing is mostly a benign artefact. You have to know what to listen for in order to hear it, even though there maybe some preringing showing in the chart, you may not hear it. How "vertical" the spike
  2. Chris, a nice technical upgrade with lower noise, lower distortion, more resolution and selectable sample rates, plus an ASIO driver for Windows! I think this would be a good upgrade for diyAudio folks designing/building loudspeakers, measuring distortion profiles, etc. But for measuring speakers for digital room correction purposes, unlikely to see any differences in the measurements as the UMIK-1 already has very good resolution. Hope you are enjoying the music!
  3. That target is from Bob Katz and James Johnston's research and listening tests. It is a (small) variation of the ITU-R BS.1116-3 standard. But falls within the range of operational room response curves.
  4. "Marketing" is the answer to both questions :-) Sorry Mani, I could not resist. But on a more serious note, we are talking about LTI systems is the real answer. Some basics: https://en.wikipedia.org/wiki/Linear_time-invariant_system#Overview
  5. @eternaloptimist in the software world, Dirac is a completely closed source solution... Glad you are enjoying the music! @ecwlDirac has no user controls other than target curve. Thus it is a general purpose solution. Both Acourate and Audiolense XO have full user control over frequency dependent windowing. Both low and high frequency window widths can be independently adjusted for both magnitude and excess phase correction, along with the amount of amplitude correction applied. The physics is understanding frequency dependant windowing (FDW) and how that app
  6. @eternaloptimistI don't disagree one bit. I spend the majority of my time troubleshooting measurement setups and as the song goes, nobody knows the trouble I have seen :) You are talking about systems design and this is what I did with my miniPC DSP offering. Everything is already configured, turn it on, take measurements, use my service or design/generate your own filters, install and enjoy! Guaranteed to work out of the box. No fuss, no muss. The issue is very few people are starting from scratch. They already have a computer or a converter or a measurement mic or an
  7. Yes, if using a USB mic with Audiolense, make sure in the measurement window, Advanced Settings menu that "Use Clock Drift Correction" is enabled. This will compensate for two different clocks and works extremely well. I have a post on the Audiolense forum that compares this to an analog mic and the difference was 0.02ms worst case. If you are using a USB mic like the UMIK-1, and an ASIO capable DAC, then you need to download and install: https://www.asio4all.org/ Make sure you have installed it with "offline" settings checked on. Then open it up and ensure only the UMIK-1 inp
  8. Hi @Gavin1977Sorry, I missed your first note. Sound stage depth and imaging are excellent. A little different dispersion characteristics then the LS50, i.e. a bit narrower beamwidth due to the ribbon polar pattern, but other that, a very wide sweetspot. Of course, there is no comparison to bottom end extension or SPL output, the Purifi's are in a league of there own.
  9. Wow @hulkss that is some engineered system you have there! Like I say, I would love to hear it! 14 x 15" woofers, that must be some slam! Just in case you did not see it, JRiver 27 will have VST3 support: https://yabb.jriver.com/interact/index.php/topic,126706.0.html Still likely need the metaplugin... Thanks for sharing!
  10. @hulkss very cool man! Would love to hear your system! Very interesting about DTS Neural Surround UpMix, will have to look into that... Curious about what your JRiver DSP Studio looks like and the order of your plugins... How is the MSO filter added?
  11. HI @hulkss Cool! Any photos of the setup to share? Or a graph or two of the result? Cheers, Mitch
  12. Hi @deneb thanks for the kind words. Perhaps Thierry @fresponse can point us to the sweep signals...
  13. @Cazzesman Thanks! The listening is a lot of fun, but the writing is a lot of work as I am more familiar with programming languages than English 🙂 The Kii Three's you and @firedog have are something special for sure. The least coloured speaker I have listened to yet. As I understand it, Bruno had control from analog/digital input to acoustic output. The corrective feedback loop encompassed the entire signal chain and acoustic output devices. There are a number of patents that cover the details of some of the pieces. It will be exciting to see what Purifi does next!
  14. Hi @R1200CL Dirac is a bit different than other DSP software in that it uses a combination of IIR and FIR filters. This paper covers how it works, Controlling the impulse responses and the spatial variability in digital loudspeaker-room correction." Note Dirac does not have many user adjustable controls... This short primer on FIR Filter for Audio Practitioners answers your questions on calculating the number of taps and frequency resolution. If you really want to get into it, then I suggest this accessible text on, "The Scientist and Engineer's Guide to Digital Signal Proces
  15. Hi @skatbelt Both Acourate and Audiolense DSP provide sophisticated digital XO capabilities. One can choose the type of digital XO, slopes and number of XO points. Here is an article on Acourate for a triamp system and here is another article on Audiolense that integrates and time aligns dual subwoofers with stereo mains (which also happens to be biamped). Since the latter article, I upgraded to dual Rythmik F18 subs crossed at 45 Hz with a steep digital XO, as you can see in the Audiolense article. Further, these XO's sum perfectly in the frequency and time domains. You can see the frequency
  16. Hi @R1200CL room correction and Room Shaper are two different technologies solving two different types of room acoustic problems. Room correction, in the context of this article, corrects both the frequency and time domain. Room Shaper reduces the low frequency "decay time" in a room. Room correction filters, in the context of this article, are static FIR filters that are convolved with the music in real time. Hence requiring a convolver to work. Room Shaper is not a static filter and processes the incoming signal based on a threshold level and strength that the user sets.
  17. Hi @Bob Stern, it is proprietary psychoacoustic smoothing.
  18. Hi @ALLDIGITAL as @The Computer Audiophile and @firedog have mentioned, flexibility is key. Another technical point I would like to bring out is that all h/w DSP solutions, whether they be Lyngdorf, miniDSP, Trinnov, DEQX or Audyssey that is used in so many AVR's, is that their filtering capabilities are very limited. How much "filtering" that can be done is based on the number of FIR filter taps available. In these h/w devices the number of filters taps available is typically limited to 1024 taps or up to 4096 and at the extreme, 8192 taps. Whereas the filters that can
  19. Thanks @blue2 Other than a UMIK-1 mic, a Mac, and REW to take the measurements, the hardware is listed under the picture of John's @Olesno system. I can export the REW impulses and import them into Audiolense. On this site, you will find a number of "walkthroughs" I wrote using Audiolense, Acourate, and Dirac. These should provide you with most of the details to your question.
  20. Hi @R1200CL Here is my review of Thierry's Room Shaper SW. Note Thierry's RS is based on events detection and modifications of only some portions of the input signal. It requires a VST plugin to operate. Re: Audiolense XO - this version has the time domain correction capabilities which I recommend. I use commercial DSP software like Acourate, Audiolense, and Dirac designed for loudspeaker and room correction. Kind regards, Mitch
  21. I see this question asked often on audio forums and there are many varying answers depending on one’s viewpoint. Viewpoints range from being able to recreate the performance, to its just entertainment folks, and everything in between. For this article, the context for what is accurate sound is the closeness to which our systems reproduce the incoming signal. To qualify the context even further, I am discussing the last component in the sound reproduction chain: loudspeakers in rooms. The intent of this article is to be educational. Digital Signal Processing (DSP) in au
  22. Wow, Josh! Way to capture the spirit of Bill Withers! Re: I told Bill, ‘That’s a hit!’ Man, I remember hearing this tune when it first came on the radio and it just blew my mind! Thanks for bringing back an awesome memory. And now thanks to you I can get an awesome version of it! Keep up the great writings Josh! Kind regards, Mitch
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