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Posts posted by mitchco

  1. @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 applies to ones speakers and room. There are a number of math calculations to make relative to how much amplitude and excess phase windowing is applied for any given speaker in any given room. Usually at low frequencies one wants quite a long window to take care of room modes/reflections and then above the rooms transition frequency, a lot less excess phase correction, but still some direct sound correction if the speaker is not the smoothest.


    As pointed out in this article, there is an "ideal response" or textbook response one wants to shoot for the most accurate sound. Of course ones ears count :-)


  2. 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 input "blue pin" is enabled and then the DAC playback output blue pin is enabled. Then make sure all other pins are disabled. In Audiolense, for playback and input devices, just select ASIO4ALL in each dropdown and that should map the the settings made earlier. One can verify be checking speaker connections. If still no sound, then in the Advanced Settings menu, select "use separate play and record streams:" and all shall be well :-)


    Kind regards,


  3. 18 hours ago, R1200CL said:

    Could one (Dirac) limit the frequency band below a certain frequency ? Like you concentrate the frequency taps in restricted area, and then achieve a more narrow frequency bin. 

    I think no, cause this is about frequency resolution, so you can’t shrink somehow. 

    Also some HW manufacturers is using 96 kHz, and then you actually need double the number  of taps in in order to achieve same resolution. Correct ?


    What would be an acceptable frequency bin or resolution ?

    I probably don’t ask the correct question, but I hope you somehow understand what I’m thinking about 😀

    Like adjusting more taps to lover frequencies. And less to the higher ones. Don’t think this make sense. 

    96000/65536 = 1.46 Hz. Would you like to see a lower number, or what is acceptable In your opinion ?

    I notice Dirac now have a basic version below 500 Hz for 3.x, and then you pay extra in order to get access to adjust frequency above that. 
    I still would think Dirac requires some sort of available processing power from the HW manufacturers. Unless Dirac supplies the board to the manufacturers. Do you happen to know anything about how this works ?




    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 Processing."


    I say this as your note on more "filtering" down low and less up high is doable, but it involves more than just the number of taps... We get into Frequency Dependant Windowing and controlling how much amplitude and excessphase correction is applied at low frequencies versus high frequencies. This takes into account how much loudspeaker plus room gets corrected at low frequencies with how much direct sound is corrected at higher frequencies and by how much as it is all adjustable in other DSP software. It gets complicated quickly as there many variables to take into account, including why we hear what we hear in small room acoustics. My DSP book tries to explain this as simply as possible with examples.


    Wrt to what DSP chips are used in what h/w devices, I could not say.


    Happy listening!

  4. 7 hours ago, R1200CL said:

    I’m trying to understand if your service includes the use of DSP produced by Rom Shaper, or if one “must” add this in addition to files received back from you.

    I think I understand your very nice review of Rom Shaper, that Room Shaper is something that’s added to Audiolence/Acourate DSP result.

    I’m asking all this, cause both your and HAF services produce an outcome, that’s is very easy to apply to Roon or HQPlayer. No extra computer needed. Present, not so with Room Shaper. (And Dirac). 

    Not sure if you’re the right person to ask, but if your answer is that end user have to purchase Room Shaper in addition, why is it not possible to mix to DSP processes in one one process, (zip-file). After all it’s only bits 😀


    Another way of looking at DSP in general, is that it seems all these top 3 products, can be further developed in order to achieve what Room Shaper does.

    I assume nothing forces us (the developer) to use technology like VST plugin to achieve this.



    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.


    If you require further information, and why RS requires a VST, I suggest you reach out to Thierry directly.

  5. 12 hours ago, blue2 said:

    Excellent article! Could you provide some more details about how you achieved these results? It appears the filters were measured and applied via Audiolense? What hardware was used for DSP playback? I appreciate a full answer would be another full length article so brief details of the hardware, software and measurements would be fantastic.



    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.

  6. 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,




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