[sdiy] vocoder filters

Sun Sep 1 22:49:53 CEST 2019

Den sön 1 sep. 2019 21:22 Richie Burnett <rburnett at richieburnett.co.uk>
skrev:

> One way to control the spectral broadening due to the amplitude modulation
> is obviously to band-limit the VCA's control signals.  I guess most VCA
> CV's
> have a faster attack than release, so it's mostly limiting the attack
> speed
> that we're talking about here.  High order LPFs applied to the VCA CVs
> give
> a more favourable trade-off between settling time and residual ripple.
>

Yes, and it's possible that a simple RC link after the ordinary envelope
follower is sufficient.

Another way to control the spectral broadening due to amplitude modulation
> in the VCA, and stop it bleeding into the adjacent bands, is to put the
> VCAs
> in-between the filter stages that make up each of the synthesis band
> filters.  This is what Roland did in their SVC-350 Vocoder.  It was
> probably
> done as much to filter out the PWM carrier frequency, but putting some of
> the vocoder's synthesis band filtering after the VCA helps to control
> out-of-band leakage caused by any fast VCA modulation.
>

Sennheiser VSM-201 did the same, IIRC.

On a related note, the conundrum I always face when thinking about Vocoder
> analysis and synthesis filter banks is how to deal with the bottom few
> bands...  The job of the analysis filter bank is to determine the spectral
> profile of the speech at any particular instant in time.  So ideally I
> want
> shed loads of closely spaced bands so that I get a fine-resolution picture
> of what the spectral profile is doing across the full speech range.  For
> instance, I might choose to use 24 bands spaced 1/4 octave exponentially
> to
> cover the 6 octave range from 80Hz up to 5.12 kHz.  This seems sensible as
> there probably aren't many vocal tract resonances outside this range that
> noticeably affect speech intelligibility.  However a problem arises with
> the
> bottom few bands.  All of the bands have the same Q and the same
> fractional
> bandwidth, however the actual bandwidth of the lower bands becomes only a
> handful of Hertz.  For example the bottom bands in my setup would be:
>
> 80-95Hz,
> 95-113Hz,
> 113Hz-135Hz,
> 135Hz-160Hz,
> 160-190Hz, *
> 190-226Hz,
> 226-269Hz,
> 269-320Hz,
> 320-381Hz, *
> 381-453Hz,
> 453-538Hz,
> 538-640Hz, *
> etc...
>
> This can lead to the frustrating situation where harmonics of the speech
> signal only fall into certain analysis bands at the low-frequency end of
> the
> speech range.  For example a female voice might easily have a fundamental
> frequency of 180Hz.  This would only put energy into the analysis bands
> centred around 180, 360, 540, 720 Hz, etc.  marked with the stars above,
> with nothing in the intervening bands.  This situation persists until
> eventually we get to a point where more than one harmonic of the voice
> signal starts to fit into each analysis band at some higher frequency.
> Then
> the holes in the analysis spectrum disappear.
>
> Clearly the analogue filter bank in this example is doing a poor job of
> estimating the spectral shaping of the vocal tract when the bands are made
> too narrow because they start to resolve the actual discrete spectral
> harmonics of the vocal signal.  This then leads to the situation where the
> output signal from the vocoder is very quiet unless the pitches of the
> voice
> and carrier inputs are related, so that harmonics of the speech and
> carrier
> inputs both land in the same frequency bands (>.<)  This isn't right.
>
> The problem is that I don't know the best way around this.

Frequency tracking of the voice signal, controlling a blending of the lower
bands in order to avoid the gaps?

I could make the
> analysis and synthesis bands wider and use less bands, but that seems a
> waste when modern DSPs have got enough grunt to run 40 bands easily.  Or I
> could make the lower frequency bands wider in an attempt to make sure that
> some harmonics of a given voice signal end up in every band at the lower
> end
> of the spectrum.  This is what Sennheiser seem to have done in at least
> one
> of their Vocoders

Did they make more models?

and it makes sense. However, I liked the idea of having

filter bands that are evenly spaced in octaves because I can then shift
> formants up or down easily by routing the outputs from analysis bands to
> adjacent synthesis bands across the audio range.  This isn't as simple if
> I
> alter the fractional bandwidths of the filters across the audio spectrum.
> The only other idea I had was to try to come up with some sort of
> interpolation algorithm to work around the "holes" in the spectral at the
> bottom end generated by the analysis filter bank which are particularly
> bad
> with voices that have a relatively high fundamental.
>
> I guess ultimately I need to accept that the analogue analysis filter-bank
> paradigm is a compromise.  Ultimately the best way to estimate the vocal
> tract resonances is probably to window one complete pitch period of the
> speech signal and then FFT it to get the spectrum.  That way you get a
> continuous spectrum, instead of a line spectrum out of the analysis
> process.
>
> Any thoughts and comments welcome.
>

Thanks for some interesting discussion.

/mr

>
> -Richie,
>
>
>
> -----Original Message-----
> From: Mattias Rickardsson
> Sent: Sunday, September 01, 2019 4:54 PM
> To: David G Dixon
> Cc: Synth DIY
> Subject: Re: [sdiy] vocoder filters
>
>
> The next level is to ponder on the unwanted higher-frequency AM effects
> from
> controlling a vocoder band VCA with the "best" (fastest) envelope
> follower,
> and whether a slower response could be more optimal. So much fun! :-)
>
> /mr
>
>
>
> Den lör 31 aug. 2019 23:23David G Dixon <dixon at mail.ubc.ca> skrev:
>
>
> Not really.  It’s a full-wave rectifier followed by a standard LP filter
> stage tuned to about 1/4 of the expected incoming frequency followed by a
> notch filter tuned to twice the incoming frequency.  I also use a full
> wave
> rectifier instead of the normal half wave rectifier, because I figure this
> gives faster integration.  This is why the ripple is at twice the incoming
> frequency, and a notch filter knocks it out nicely.  Through the judicious
> choice of gain at the LP filter, the envelope follows the waveform tops
> more
> or less exactly, and comes up to full strength within two periods of the
> incoming waveform, with ripple which is inconsequential.  For a 10Vpp
> waveform coming in, the envelope rides at 5V, which will turn on my
> favored
> linearized 2164 VCA design to unity gain.
>
>
>
>
>
> From: David Moylan [mailto:dave at expeditionelectronics.com]
> Sent: Saturday, August 31, 2019 5:26 AM
> To: David G Dixon
> Cc: synth-diy at synth-diy.org
> Subject: Re: [sdiy] vocoder filters
>
>
>
>
>
>
>
> Curious about this envelope follower you mention. Trade secret?
>
>
>
>
>
>
>
> On Aug 31, 2019 03:48, David G Dixon <dixon at mail.ubc.ca> wrote:
>
>
>
> Well, I know that the higher-Q filters have a longer delay, so that they
> take longer to respond to the incoming waveform.  I’m thinking that a Q of
> about 3 is probably about right, and with that, only a 4-pole filter is
> required.
>
>
>
> I’ve got a nice design for an envelope follower which responds quickly and
> has little or no ripple, so that’s not a problem.
>
>
>
> On a related note, does anyone here have problems getting the Bode plotter
> in Multisim to work consistently?  I am finding with this simulation that
> sometimes if I change the component values, the Bode plotter doesn’t work
> at
> all.  Also, for some simulations, changing the component values doesn’t
> change the filter response at all.  Multisim is sure glitchy.  It’s very
> frustrating.  I can sometimes fix it if I erase all of the passive
> components and load new ones with the new values, rather than just
> changing
> them, but that sort of thing is just complete bullshit.  Multisim is a sad
> excuse for a professional program.  There must be something better out
> there
> (?).
>
>
>
>
>
> From: Paul Perry [mailto:paulfrancisperry at gmail.com]
> Sent: Friday, August 30, 2019 10:22 PM
> To: David G Dixon
> Subject: Re: [sdiy] vocoder filters
>
>
>
>
>
>
>
> I don't think there is a "right" answer. To my mind, it depends on what
> one
> wants to do with the unit. Think about what will happen when a single
> swept
> tone is used to modify white noise. The low pass filter on the VCAs
> probably
> has a significant effect as well.
>
>
>
> paul perry Melbourne Australia
>
>
>
>
>
>
>
> On Sat, 31 Aug 2019 at 14:50, David G Dixon <dixon at mail.ubc.ca> wrote:
>
>
>
> Well, I think I might have answered my own question.  Looking again at the
> JH Living Vocoder, since Jurgen Haible’s filter responses overlap at about
> the 8dB mark, it really should not matter at all what’s going on around
> the
> skirt of the response, and higher-Q filters with two 2-pole stages should
> give very similar results to low-Q filters with four 4-pole stages (and be
> much much cheaper to build).
>
>
>
> I’d still appreciate if anyone has any specific insights into this
> problem.
> Cheers.
>
>
>
>
>
> From: Synth-diy [mailto:synth-diy-bounces at synth-diy.org] On Behalf Of
> David
> G Dixon
> Sent: Friday, August 30, 2019 9:05 PM
> To: synth-diy at synth-diy.org
> Subject: [sdiy] vocoder filters
>
>
>
>
>
> Hey SDIY Team!
>
>
>
> I’m thinking about building a vocoder, and I have a general question about
> the bandpass filters.
>
>
>
> I’ve looked at Jurgen Haible’s Living Vocoder, and he used 8-pole filters
> with low Q.  These give a reasonably broad band with fairly steep slopes.
> He makes the filters from two pairs of LP and HP.
>
>
>
> I was thinking about using BP filter sections, but just 4-pole, and with
> higher Q (around 10).  This gives a somewhat narrower band, and the slope
> is
> steep near the corner, but fairly shallow around the skirt.  This idea
> uses
> a lot fewer components (about half as many).
>
>
>
> What I’m asking is, does anybody here have any insight into what the
>  “proper” approach to vocoder filters would be?  What is the design goal?
> Do you want significant overlap from one band to the next, or should they
> be
> fairly distinct?  I guess I’m just looking for some general guidelines and
> conventional wisdom.
>
>
>
> Cheers,
>
> Dave Dixon
>
>
>
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