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Posts posted by 3ll3d00d


    Wonder Woman (2017, BD / Atmos) - JRiver equivalent filters

    NB: I refer to it as S below to differentiate from the original post, jriver calls this Q though.

    measurement to verify shape -> https://imgur.com/a/05Ixe

    UHF / distance (relative to flat response; see note*):

    High Shelf: f0 = 13900.0; S = 0.442082; gain = -3.501

    HF slope (see note **):

    High Shelf: f0 = 3140; S = 0.9997; gain = -2.0
    High Shelf: f0 = 7750; S = 0.552885; gain = -2.0
    EQ Peak: f0 = 1950; Q = 1.75; gain = +0.33

    MF shelf:

    Low Shelf: f0 = 1000; S = 0.720521; gain = -1.25

    LF shelf:

    Low Shelf: f0 = 265; S = 0.785708; gain = -4.0
  2. On 08/11/2017 at 9:15 PM, SME said:

    I am also thinking a bit about automation here.  I think the tricky thing about using something like MSO to find filters is that we want to only make coarse shape adjustments.  The finer peaks and dips in the peak and/or average graphs should be retained because they arise from specific effects that are meant to have characteristic sounds.

    That still may not be the most time consuming part though.  That might be the headroom check.  

    The reason I suggested using MSO is because you have complete control over the parameter range for each filter and also complete control over how many filters, and what type, it can use. Combine this with smoothing of the response if necessary and I would have thought you'd end up with a smooth filter shape.

    re headroom, is it the summed result that is the concern? 

    On 09/11/2017 at 4:13 AM, maxmercy said:

    Anywhere from 4-12 hours per film, with about 1/2 that time just waiting for processing to happen as I think about how to do the next trial solution. 

    that seems a long time to wait for processing, what are you using for that?

  3. If the approach is based on a target curve which is based on other known good bass films then t seems like this should be amenable to automation, at least as a first cut anyway.

    For example, create a minimum phase representation of the actual response and a target curve, load into MSO and create a configuration that has a load of adjustable Q shelf filters & some PEQ available to it then give it some time to work out what combination of filters produces the target response. 


  4. FWIW I used jriver convert format to generate the input wav file and then these are run through my app in a way that is equivalent to the approach used around here in speclab (which decimates it to 1000Hz and then runs it through the fft with a nuttall window of either 0.5Hz or 1Hz resolution, I forget which). I used a 0.5Hz resolution in my case, this is pretty much all done by scipy (for the fft bits) and librosa (for the resampling)

    I thought I'd double check so used eac3to to generate the wavs instead (though they both use the same underlying dts decoder anyway - https://github.com/foo86/dcadec), same result. 

    If you get a different result then it would be interesting to know how you're performing that extraction and conversion.

  5. I tried another one and got similar results, hacked together a multichannel wav with a sine wave at a different frequency for each channel and ran that through jriver then analysed it, result is a mono wav with a missing channel 4, i.e. no LFE channel in the output. I'm quite sure this used to work so will report over at jriver forum to see what is going on.


    edit: see https://yabb.jriver.com/interact/index.php/topic,111023.0.html  appears to be some random defect in this particular installation of jriver :|


    edit2: turns it out it is a bug in jriver - https://yabb.jriver.com/interact/index.php/topic,111023.msg767472.html#msg767472

  6. Any idea why your PvA looks so different from what JSS posted above?


    Yours appear to be higher resolution (longer window?) but looks smoother, so I don't think it's a resolution issue.  Indeed, the curve shape and levels both look quite a bit different.  Could the difference have to do with how the down-mixing is being done before analysis?

    dunno, I downmix using jriver as per http://www.avsforum.com/forum/113-subwoofers-bass-transducers/1333462-new-master-list-bass-movies-frequency-charts.html#post23468771


    I just doublechecked the settings, recreated the mono wav and ran it through speclab and my app. I get the same result each time.

  7. 1. Level - This is measured by digitally bass-managing the 5.1 or 7.1 signal.  The Level is a composite number, and is calculated by the average of the [1] highest peak in dB (maximum 126dB for 5.1, 128dB for 7.1), [2] the average/RMS dB level of the track (125ms integration time) and [3] the RMS peak level (loudest single 1/8th of a second of the film) in dB.  Full modulation of the waveform is considered to be 0dBFS.  The ratings are as follows:


    5 Stars - >112.5dB composite

    4 Stars - >110dB composite

    3 Stars - >107.5dB composite

    2 Stars - >105dB composite

    1 Star - <105dB composite



    @maxmercy how do you calculate [2]? I don't understand how the integration time relates to a single dB level for the entire track. The time window sounds like it relates to the length of each slice in the spectrogram but then how do you turn that into a single number?

  8. let me know if you hit any issues, don't think anyone else has used it yet :) I could easily add some options to make it easier to use for this case though, for example 


    auto detecting files dropped into a certain directory

    providing some config option for tweaking the fft length

    providing a quicker way to see peak/average on one chart (rather than having to select peak then select average each time)

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  9. fwiw the vibration app I wrote does the same sort of analysis as speclab for the target curve function - http://www.avsforum.com/forum/155-diy-speakers-subs/2681865-rpi-based-diy-vibration-meter-3.html#post52214001- and the relevant bit of user guide is http://vibe.readthedocs.io/en/latest/uiguide.html#wav-upload


    It is currently tuned to produce output with ~1Hz resolution though, it decimates to 1kHz sample rate and then performs an analysis based on a 1024 point fft. Bit easier to use than speclab as you just drag and drop wav's into it, no options exposed to configure it though.

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