• felipe.rodriguez

    Hi, Pd people!

    I want to overlap an audio signal with a sound generated by five formants. These formants will be generated by a bank of filtered white noise. The filters I will use for this purpose will be bandpass filters, which need the frequency and Q parameters of the original signal to create a smooth overlap.

    with fiddle ~ I can get the frequency and decibels of each of the formants, but how I can get the bandwidth?

    I appreciate any advice.

    Thanks.

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  • felipe.rodriguez

    I have the same problem with ubuntu studio 12.04

    My patch has only ASCII characters

    Thanx

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  • felipe.rodriguez

    @ClaudiusMaximus said:

    You can get a continuous envelope from the audio signal with:

    in
    | \
    [*~]
    |
    [lop~ 10]
    |
    env

    then you can turn it into a digital signal with parameters for a low pass filter for attack/release:

    threshold attack coefficient
    | |
    env [dbtopow~] | release coefficient
    | | | |
    [expr~ $v1>$v2; if($v1>$v2, $v3, $v4)]
    | |
    d.env coef

    next, you can low pass filter to smooth the signal, at different rates for attack/release:

    d.env coef
    | |
    [pd low-pass-filter] -- check Miller's book on how to build filters using czero~ cpole~ etc
    |
    l.env

    then multiply the input by the envelope :)

    in l.env
    | |
    [*~]
    |
    out

    hold time is more tricky to implement, and I'm not quite sure what that means for a noise gate.

    I built a compressor/limiter in a similar way, not a very good one, but enough so that if I'm live-coding I can just feed everything into it and not have to worry too much about adjusting the volume to avoid clipping etc.

    wow, that's elegant.

    [mavg] is too cpu expensive? or at least more expensive than [*~] and [lop~]? I use [mavg] when need a smoother signal.

    well, thankyou very much Claudius Optimus et Maximus.

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  • felipe.rodriguez

    thanks, saturno.

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  • felipe.rodriguez

    anyone knows which window (hann, hamming, rectangular, gauss, etc) is the best if you want a convolution with a wide range of formants?

    Hann is not so good in real time, I noticed, because the sound generated has just one formant, and the result is very similar to a resonant filter...

    if anyone has a clue, please share it!

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