Lyngdorf RoomPerfect signal processor

Road Runner
Road Runner Posts: 106
edited October 2008 in 2 Channel Audio
This looks pretty interesting. What you do is put a microphone in the room and have the system play test signals and do that for a bunch of different places in the room, including the listening position. Then the processor creates an algorithm for correcting the sound to match the acoustic properties of the room. I saw something else called an Accuphase Digital Voicing Equalizer which also used a microphone, but I think it just increases or decreses certain frequencies to correct for sections of the room that might resonate with those frequencies and distort the sound. This Lyngdorf thing might do more, I'm not sure. If you go to the following page and click on the link toward the bottom there a video demonstration.

http://www.lyngdorf.com/index.php?option=com_content&task=view&id=76&Itemid=35
Post edited by Road Runner on

Comments

  • SlowcarIX
    SlowcarIX Posts: 887
    edited October 2008
    my 7.(1x4) HT setup
    TV - Mitsubishi WD-65734
    AVP / Amp - Onkyo PR-SC885P / D-Sonic 2500-7
    Front - Emerald Physics CS2
    Center - JTR Triple 12LF
    Surround L/R / Back - Polk RTi4 / Polk FXi A4
    Sub - 4 X Hsu ULS15 playing nearfield
    DVD / CDP - Sony PS3/40GB / Sony SCD-XA9000ES
    Belkin PURE AV PF60 / UPS
    Buttkicker

    http://www.polkaudio.com/forums/showthread.php?t=60612
  • Road Runner
    Road Runner Posts: 106
    edited October 2008
    Yeah, maybe that one's better or there may be a lot similiar ones out there. This makes it sound like it only works with digital:
    Time and Frequency correction:


    The time domain is where many of the problems reside. Parametric and graphic equalizers can only correct for the frequency response and do so in a very coarse manner because they have limited resolution (bands).
    Further, whether they have fixed or adjustable bands it does not matter because bands cause phase problems that most people hear as "ringing" or "smearing." This is why, after thirty plus years of trying this method most people don't like the results and turn it off.

    How does MultEQ address time and frequency problems?


    MultEQ filters start in the time domain. They are not just a few parametric bands. Instead they use several hundred points to represent the room response in both the frequency and time domains. The trick is to use enough filter points to get the needed resolution, but not so many that it overwhelms the processor inside the audio component. So, we came up with a way to reduce the number of points without sacrificing accuracy and a way to provide more filter power at lower frequencies where it is needed the most. MultEQ can correct 8 channels by using only a fraction of a single DSP chip. This gives you the best of both worlds: time and frequency correction. Result--room correction that works for the first time ever.

    I'm not sure they could do all that with an analog system so if you want to use LPs with that you might be out of luck, or it could have some analog to digital converter thing in it.