Jump to content

andyc56

Members
  • Content count

    9
  • Joined

  • Last visited

  • Days Won

    1

andyc56 last won the day on October 20 2016

andyc56 had the most liked content!

Community Reputation

6 Neutral

About andyc56

  • Rank
    Newbie

Contact Methods

  • Website URL
    http://andyc.diy-audio-engineering.org/mso/html/

Profile Information

  • Gender
    Male
  1. Here is a link to a Harman study about target curves.Hopefully it will work.
  2. I noticed when reading this article on room gain that there appears to be a few character encoding or decoding errors on the page. It' looks like what was intended as an apostrophe or similar character ended up undergoing some sort of illegal conversion somewhere along the line. It is in the text: "The difference between the two measurements will show the effect the boundaries and acoustics of the space has on the subwoofer�s basic raw response as delivered to the listening position." and "Based on these results, we can then gauge what other subwoofers� maximum headroom and basic response shape would be in the same placement, either through looking at outdoor groundplane results such as those presented here or inside of a simulation program." Looking at the HTML in a hex editor, the black diamond with question mark is the three-byte hex sequence \xef\xbf\xbd, which is the black diamond with white question mark character in a UTF-8 encoding. I noticed the character encoding of the HTML file is listed as iso-8859-1, which seems unusual to me. If the CMS is using UTF-8 in its database, errors can occur with some characters if trying to convert them to ISO-8859-1 for display. UTF-8 can encode all possible Unicode characters, but ISO-8859-1 can only encode less than 256 of them. Anyway, I just thought I'd mention it in case nobody had seen it.
  3. I recently got into a discussion with REW author John Mulcahy about this, and it turned out to be very fruitful. There are several different conventions for "Q" for PEQ filters of different hardware vendors, and some question of how this parameter relates to bandwidth, and even how bandwidth should be defined in the first place. I wrote an article about it to hopefully clear up some of the mystery about different conventions for different DSP vendors.
  4. I haven't tried to do any simulation of finite word length effects on the coefficients. If they were 32-bit floating-point, I guess I could use Matlab's single-precision floating-point arithmetic to get an idea of the error. But with oddball-sized integer coefficients, it seems that would be a PITA to figure out. I'm not a digital filters guy anyway, so I had to read a book on them by Antoniou (the op-amp gyrator guy) to understand what was going on. I had to shake out some mental cobwebs going back to the 1970s. I seem to recall that someone on AVS, it might have been notnyt, had some problems at low frequencies due to word length effects with one of the 8-channel miniDSP boxes when he used a 96 kHz plugin. I believe it was for a high-pass filter for a vented-box sub with a pretty low tune. He had to go with a 48 kHz plugin, which improved things, but the errors were still pretty easy to see as I recall. It would be nice to find that thread again. Bristow-Johnson has some amazingly compact formulas for the conversion of the analog filters to z-domain form for different filter types, but it is a different set of formulas for each filter type. I wanted to have as much code shared between the filter types as possible for debugging reasons, so I ended up with formulas that are a lot messier than Bristow-Johnson's, but it's the same formula each time. Only the center frequency pre-distortion technique varies from filter to filter. I've attached the derivation below.
  5. I'm having trouble splitting up quotes, so I'll just paste them into regular quote blocks. That's really weird. So just applying a single EQ to a single channel doesn't give the expected measured response? I don't know what could be causing that. Edit: Can you try this with just a PEQ and no all-pass filter in the path? I'll go back and re-check the biquad coefficients for the all-pass filters using Matlab. Internally, MSO computes the filter responses as analog filters. When it computes the biquad coefficients, it uses the bilinear transformation to get the z-domain transfer function. There is a nonlinear scale factor that "predistorts" e.g. the center frequency of the PEQ such that after the bilinear transformation the z-domain transfer function has the exact same center frequency as the analog one. There is some "bandwidth warping" going on due to the nonlinearity of the frequency transformation, but when the critical frequencies of the filter are so much less than the Nyquist frequency of the DSP, there is negligible error, on the order of a couple hundredths of a dB. In order to ensure the integrity of the calculated biquad coefficients, I compared my results with the spreadsheet by Charlie Laub at the miniDSP site. When I was checking out the biquad coefficient code, the results agreed with Charlie's spreadsheet down to 14 digits for all filter types except the PEQ. To chase the difference down further, I wrote a Matlab script and a set of functions to compute the response of the analog PEQ, the coefficients of the digital PEQ, and the response of the digital PEQ. Center frequency and boost/cut at the center frequency were dead on, as expected by the "pre-distortion" of the analog filter center frequency. Maximum error between the analog filter response and the computed digital filter response was a couple hundredths of a dB over the frequency band to 200 Hz. (Edit: I just went back and reinstalled Matlab and ran that script again. Maximum error between analog and digital filter is 1.4 * 10^(-4) dB.) I'm not saying Charlie's spreadsheet for the PEQ is wrong though. There is potential ambiguity in the definition of "Q" for the analog PEQ. I use the one that Bruno Putzeys uses in his Hypex sub amps. For those, the "Q" is assigned to the analog filter "pole Q" when there is a boost, and to the analog filter "zero Q" when there is a cut. This gives the result that two PEQs with the same Q and center frequency, one with a boost of X dB and the other with a cut of X dB will exactly cancel each other out when combined. My guess, although I have not proven it, is that the discrepancy between my results and Charlie's for the PEQ may be due to a possibly different definition of "Q" being used for the analog PEQ filter. At any rate, I got excellent agreement between the analog and digital filter responses, so I am satisfied the calculations are good for the PEQ. Let's hope that was the cause of the problem. Is there any dynamic EQ sneaking into the AVR? That should give an error message. I think the 4-way advanced plugin can have 6 PEQs, but the 2x4 advanced is limited to 5. When you tell MSO how many biquads your DSP has per channel, and you use more than that, it will still give you all the coefficients in order to not lose information. I believe it prints out a warning in the filter report when that happens. I'll check that, and if it doesn't, I'll fix it so it does. I should probably display a warning message when this happens when exporting biquad text as a file too. The purpose of specifying the maximum number of biquads is that if you use less, it will pad them with "through connection" biquads up to the maximum number in the exported text and the filter report. It could be that they're using a small number of frequencies and lots of smoothing so that the peak or dip is missed and smoothed over. That's just a guess on my part. With regard to MSO PEQ biquad computations, the center frequency and gain/attenuation of a single PEQ digital filter at the center frequency ought to be dead nuts on. I'm glad to hear that at least, but sorry to hear that you're having these other problems.
  6. At the moment, there's no way to constrain the maximum allowable total PEQ boost in a given channel, just individual boost. I've always used MSO with all PEQs set to a maximum individual boost of 0 dB per the default. In this way, if there are any notches in a combined response, it must try to fill them in by minimizing cancellation. I'd suggest giving that a try. You can often fill in response notches by a surprising amount without any boost in any PEQ at all.
  7. This is the first I've heard of FreeDSP, so I'll have to spend some time going through the web site. I can export biquad coefficients in the same format that REW does for miniDSP. These have the miniDSP oddity that the a1 and a2 coefficients in the file are actually the negatives of the coefficients in the transfer function. I haven't used SigmaStudio, so i don't know if it expects the inverted a1 and a2 the way the miniDSP software does.
  8. You're welcome! A while back, you were talking about raising the max allowable center freq of PEQ filters for horn sub EQ. I can look at how to do that while discouraging the high max center frequency usage for more typical applications. Maybe have a special PEQ type that allows it.
  9. Hi, I've had some strange problems when trying to do a loopback measurement of the frequency response of my UMC-200 pre-pro at line level with ASIO4ALL on a Windows 7 machine with AMD processor using HDMI. I was able to get a good sweep again (but only temporarily) by changing the REW output channel to a different one, then changing back to the desired channel again. I don't know if this is the same problem you were having, and I don't know what the sweep would have sounded like, since it was a line-level measurement without a mic. All I know is that it gave me a really crazy looking frequency response when it happened, which for a line-level device is clearly wrong. I'm the MSO author, so feel free to ask about it here or in the main MSO thread at AVS.
×