Re: [Audacity-nyquist] eq-band and "width"

[Roger D. <rb...@cs...>]

On 10/24/10 10:22 PM, James Shatto wrote:
> What I'd really like to do is sort of a spline adjustment to EQ across
> the board maxing out at the high end at a specific dB level.  And
> close to 0dB on the low end.  Could that be done per frequency?  I'm
> trying to compensate for the HF loss due to fake fur.  And I've grown
> tired of entering 15 numbers for every adjustment.  Plus I'd like a
> few more bands for a smoother adjustment.  Hopefully with some control
> over how linear or dipped / curved the adjustment will be.
How would you specify the spline?

There's a file in nyquist/lib/grapheq.lsp that implements NBAND-RANGE:

|nband-rang/e//(input/, /gains/, /lowf/, /highf/)|
    A graphical equalizer applied to /input/ (a |SOUND|). The gain
    controls and number of bands is given by /gains/, an ARRAY of
    |SOUND|s (in other words, a Nyquist multichannel |SOUND|). Any sound
    in the array may be replaced by a |FLONUM|. The bands are
    geometrically equally spaced from the lowest frequency /lowf/ to the
    highest frequency /highf/ (both are |FLONUM|s).

I think you could generate an array of gains according to some 
parametric description.
> For my current setup I'm needing near 2dB to 4dB on the high end.  As
> much as 6dB on previous contraptions.  So far results are promising
> for the restoration of certain percussion instruments and certain
> wildlife.  Plus the spoken word seems to need a bit of high end for
> clarity.  My lisp is a little weak, having only used it before to do a
> bunch of notch filters to counter sox's noise reduction that seems to
> omit the 1000Hz multiples from it's filter.
>
> Ultimately I'd like the actually measure the mics response in said
> configuration and base compensation off of that information.  But for
> now, just trying to keep it simple and mostly automated.  I don't
> really care if it takes a day for an hours audio, as long as it's good
> audio.
One well-known problem with EQ is that it introduces lots of 
frequency-dependent phase shift. Esp. for percussion, you might be 
better off with a simple low-order parametric filter that just boosts 
the highs a bit rather than a big multi-band EQ. The latter will let you 
model the overall amplitude response in detail, but the phase response 
will be erratic.
> Thanks for the clarification on width.  That helps a bit.
>
> - James
>
>
> On 10/24/10, Roger Dannenberg<rb...@cs...>  wrote:
>> On 10/24/10 2:50 PM, James Shatto wrote:
>>> eq-band(signal, hz, gain, width)
>>>
>>> I'm trying to write an EQ routine for audacity and I'm having trouble
>>> understanding the width parameter.  I had thought that High Hz - Low
>>> Hz over 2 is what should be width.  But my routine doesn't work that
>>> way when it uses that on the larger Hz values.  But over 4 does seem
>>> to work.  I'd like to sort out why so I don't have any gap or overlap
>>> in the EQ bands.
>> I guess you know there will always be overlap. These are not "brick
>> wall" filters.
>>> When it does the EQ for the higher bands ~18kHz, it
>>> basically nulls out the audio if /2 is used.
>>>
>>> 20000 - 16000 = 4000
>>> 4000/2 = 2000 = width?
>>> or
>>> 4000/4 = 1000 = width?
>>> or
>>> Some other factor I seem to not quite get yet.  To be used like
>>> (eq-band s 18000 2.0 2000)
>>>
>>> Per the documentation:
>>> --- and width (a FLONUM) is the half-gain width in octaves ---
>>>
>>> Could someone give me a human-esque interpretation of that phrase.  Am
>>> I supposed to make it a factor of the gain that I'm applying or
>>> something?  If so, what does that do to the gain value?
>> The gain is measured in dB. The width is measured in octaves, e.g. if
>> the width is 2, the range of frequencies affected by the filter is from
>> hz/2 (an octave below) to hz*2 (an octave above). At these points (hz/2
>> and hz*2), the filter gain will be gain/2, measured in dB.
>>
>> To compute width, the octaves between f1 and f2, where f2>  f1, use
>> log2(f2 / f1). log2(x) is not built-in, but log2(x) = log(x) / log(2.0).
>> To hz, get the frequency in the middle, use hz = f1 * power(2.0, width /
>> 2.0).
>>
>> I wrote a little program to send a sine sweep through a filter and plot
>> the peak amplitude of the output as a function of frequency (works in
>> the Nyquist IDE, and since the output is a sound, I guess you run
>> something similar to this in Audacity:
>>
>> function test()
>>     begin
>>       ;; probe is a sinusoidal sweep at 1Hz/s
>>       with probe = hzosc(pwl(0, 0, 1000, 1000, 1000, 0)),
>>       ;; filt is a filtered probe
>>            filt = eq-band(probe, 200, -6.0, 2.0),
>>       ;; out is the rms of the input scaled to get peak
>>            out = rms(filt, 10) * sqrt(2.0)
>>       return out
>>     end
>>
>> exec s-plot(test(), 1000, 1000)
>>> Also, is there any sort of debugging tutorial, even if I have to use
>>> emacs or something.  Even though my routine is fairly simple, I hate
>>> having no ability to peak at the values (after applied arithmetic)
>>> before they get used.
>> Maybe someone has a better answer for Audacity-nyquist, but I use the
>> Nyquist IDE. -Roger
>>> Thanks,
>>> - James
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