[sdiy] Fast envelope follower circuit needed..

cheater cheater cheater00 at gmail.com
Sat Mar 30 19:24:54 CET 2013


Hi Tom,
just like I said at the very beginning:

> Do the rectification in analog.

Then, use one of Julius Smith's filter cook-books to get a good
asymmetric low-pass filter, 2nd-order, and feed that directly from the
ADC. Set the rise time and fall time separately, call them "attack"
and "release".

If your PIC's ADC does not have one, use a brickwall filter chip in
front of it to prevent aliasing when digitizing the signal.

To improve quality you can up-sample the signal from the ADC, run the
asymmetric filter at that higher sample rate, and then lower the
sample rate.

With some thought (beyond my current rusty knowledge of filters) it
should be possible to have the upsampling, filtering, and downsampling
in one single filter, with very nice numerical properties when it
comes to stability and cycle efficiency. Listen to Richie's advice if
he offers any.

In fact, digital filters are a very good fit for such analysis tasks -
using the best of both worlds (rectifying in analog, filtering in
digital) you might get a superb envelope follower, unlike anything on
the market right now.

Of course, make sure to have a nice high-current output on the DAC side.

It doesn't hurt to use a dedicated ADC, since onboard PIC stuff would
be more suited to encoders than to audio. This means reduced quality
when it comes to fast (audio-rate) signals and usually cross-over
distortion.

Cheers,
D.

On Sat, Mar 30, 2013 at 5:22 PM, Tom Wiltshire <tom at electricdruid.net> wrote:
> So how would *you* solve the problem, Damian?
>
> So far, you've spent a lot of effort telling everyone who's made a suggestion about the problem why they're wrong, and why it couldn't possibly work, but I'm not seeing much in the way of better suggestions.
>
> Tom
>
>
> On 30 Mar 2013, at 14:19, cheater cheater <cheater00 at gmail.com> wrote:
>
>> Hi guys,
>> there are several points from Tom, Scott, and Brian here I'd like to
>> address. I hope it's ok, I would like to point out some errors to
>> spare you frustration in the future.
>>
>> 1. "the filter does not care about the spikes in the valleys"
>>
>> that's incorrect. Bear in mind that this high-frequency content is not
>> removed in a low-pass filter, it is attenuated. Quite often the
>> rejection is less than the theoretical maximum for many reasons
>> (numerical error, incorrect algorithm implementation, etc). In effect,
>> that "rejected" signal is still there.
>>
>> Especially if you use non-linearities later in your signal path (that
>> is, always), whatever program you modulate with such a tainted
>> envelope will sound worse than it could otherwise.
>>
>> 2. "analog rectification creates wide-band energy bursts anyways"
>>
>> not an accurate picture of the situation. In analog, this sound will
>> be harmonically related to your input, which is a good thing. In
>> digital, it will work exactly against it, which is very bad. Besides,
>> the turn-on and turn-off speed is limited, it's not when using abs()
>> in a digital signal, but I don't know what the exact difference in
>> bandwidth limitation is in analog - it's unlimited in digital (with
>> the caveat that it gets mirrored around fs/2 and 0Hz, perhaps several
>> times)
>>
>> 3. "the peak hold doesn't care about the little valleys"
>>
>> not true at all. I believe this comes from the conceptual
>> simplification of the filter, which could go like this:
>> (i) when the input signal is above the current output, raise the output to match
>> (ii) when the input signal is below the current output, drop the
>> output at a constant rate
>>
>> that is extremely inaccurate. What happens is more similar to this:
>> (iii) measure the distance between the current output and current
>> input. Move so that during a period of time, say 1 second, you cover a
>> fixed percentage of this distance, directly related to a number called
>> the time constant. For example, cover 90% in 1 second.
>>
>> What does this mean? Say your output is at 1V, and your input is a
>> sinewave oscillation between 0V and 0.01V. As your output is falling,
>> it'll fall relatively faster when the input is near 0V (because the
>> distance is higher), and it'll fall relatively slower when the input
>> signal is *closer* to the output signal (i.e. when the input is at its
>> 0.01V maximum).
>>
>> This is easily described by saying that the peak hold is just a filter
>> as well, but it doesn't quite picture it, that's why I described the
>> operation above.
>>
>> You might ask: "But can't I just use max() of the input and output and
>> somehow use that?"
>>
>> One might think this code would work:
>> if (output < input) {
>>  output = input
>> }
>> else {
>>  output -= 0.01
>> }
>>
>> This is a beginner's mistake. This sort of design will create awful
>> amounts of zipper noise. You don't want that. You want to use a
>> filter.
>>
>> Brian, you first correct Tom on his mistake, then make the same
>> mistake yourself elsewhere, then agree with Scott who has made the
>> same mistake :)
>>
>> 4. "because peak detection allows the output to suddenly change if the
>> input is greater than the hold value, and any time a digital algorithm
>> allows a sudden discontinuity in its output you have the potential for
>> non-band-limited square wave type aliasing"
>>
>> Not true. You just use a normal (asymmetric) filter for the peak hold,
>> like you usually would, and set its time constant for rising signals
>> so that it reacts instantly.
>>
>> Your logic is flawed because you're thinking in terms of the "if/else"
>> model I described above. It doesn't hold. The transition you describe
>> will not be abrupt. There will still be phase delay like in any usual
>> filter (that's a good thing), and this phase delay will give the
>> filter just enough time buffer for things to happen relatively
>> smoothly. It's a different question, though, whether your filter
>> aliases *as a whole*.
>>
>> 5. "Assuming your audio signal is limited to 20 kHz maximum, and your
>> sample rate is 44.1 kHz, then your envelope signal need only have no
>> aliases above 2.05 kHz. Bingo, no audio aliasing"
>>
>> That's not how it works. If your signal picked up aliasing somewhere
>> earlier, it's not going to magically get rid of it by becoming
>> band-limited later on. If you have frequencies within your 2.05 kHz
>> that do not agree with the harmony of your program, you're going to
>> hear it, simple as that. If you said...
>>
>> "multiplying two signals, such that the sum of their bandwidth is less
>> than fs/2, will not create new aliases"
>>
>> ...then I might be partial to agreeing, although due to numerical
>> inaccuracies that's not really true either. Depending on what you're
>> working with, YMMV.
>>
>> 6. "I do not see how that is a problem with an envelope detector. The
>> goal is to produce a control signal with basically "no" frequency
>> components, or at least they should be as close to DC as possible. In
>> practice, there will always be undesired low-frequency components in
>> the envelope. My point here is that aliased components are no more
>> problematic than non-aliased frequency components."
>>
>> Only if you revel in muddy sound that becomes more and more occluded
>> and unclear the more processing you apply. It's a fact that
>> harmonically unrelated sounds present in your program will deteriorate
>> its quality. They're much worse than e.g. white noise, because they're
>> concentrated in single, discernible tones, which play melodies that go
>> completely against the sounds that you are playing on the keyboard.
>> I'm not sure how to better say this; if anyone here is not convinced
>> that having this is bad, please read the previous sentence 50 times.
>>
>> 7. "The remaining frequency components are all equally undesired, and
>> it doesn't really matter whether they're aliases or not, so long as
>> they are low enough in frequency."
>>
>> It's exactly the other way around: aliased low-frequency components
>> are very undesirable and are glaring problems in the final sound.
>> Harmonically related frequency components don't work against the sound
>> you're building, and are often very desirable. At worst they are
>> harmless because they become invisible when surrounded by lots of
>> other stuff playing in the same harmony.
>>
>> 8. "Perhaps I don't even need to bother rectifying the signal. Why not
>> just keep a check on the negative peaks like the positive ones, and
>> then use some average of the two as the final output. JP said his
>> signal might well be asymmetric (which is not at all uncommon), so we
>> can't assume the two envelopes will be the same."
>>
>> This is DSP, not analog fx design. DSP is a harsh mistress. You can do
>> a lot of learning by experimenting in analog electronics and it works
>> out well. The signal degradation which happens when you make errors is
>> very musical: nth harmonic distortion, homogenic noise, cross-talk of
>> content that is harmonically related. It's all nice and very
>> forgiving. In DSP, you have very, very bad degradation modes: i)
>> aliasing - takes your melody and makes it sound like it's just random
>> notes; ii) through-zero distortion - adds mud and reduces clarity in a
>> way that's not related to your sound in any way; iii) time delay -
>> makes sure that you will never be able to mix your sound correctly
>> again after you've split it; iv) numerical errors - similar to
>> aliasing, those create junk in your sound that is harmonically
>> unrelated to the sound you're trying to achieve (examples include bit
>> errors, zipper noise, clicks, hard clipping, etc).
>>
>> Be warned: you really have to go by the book on DSP. There are very
>> good books online; Julius Orion Smith's website is the best,
>> most-recommended classic. You simply can't "common-sense" your way
>> through digital audio. It's daunting. You'll be making errors, and the
>> worst thing is that in analog when you make errors they're obvious
>> whereas in digital, when you make errors, you might not notice until
>> after you've shipped the box; in the end people figure it out though
>> and just don't like your design. Granted that doesn't matter if you're
>> selling for the masses that buy microcorgi and other junk like that;
>> but let's maybe leave this kind of talk off this list and try to think
>> about what it takes to do an honest job. As someone said, if
>> something's worth doing, it's worth doing well.
>>
>> Cheers,
>> D.
>>
>> On Sat, Mar 30, 2013 at 7:24 AM, Paul Perry <pfperry at melbpc.org.au> wrote:
>>> A little philosophy on envelope followers:
>>> It's easy to see what the envelope should look like - after the original
>>> signal has been! But if you take a typical input signal, graph it, and then
>>> cover it with a card, and try to draw an envelope as you slide the card to
>>> the right........... you might see that is is not just 'difficult', but
>>> actually 'impossible' to have a circuit that behaves as you would wish.
>>>
>>> In my experience, there is no envelope follower that suits all applications.
>>> In particular, percussion and guitar are very different. I don't doubt that
>>> Harry's circuit is ideal for a guitar.. and that if he could have got away
>>> with fewer components, he would have (remember, a hex guitar has SIX
>>> outputs!)
>>>
>>> On the bright side, I have also found that if you are using the derived
>>> envelope to control a filter or VCA, the actual audio effect can be quite
>>> musically useful, even though the envelope itself looks really unpromising
>>> on the scope. In particular, full wave rectification was not much better
>>> than half wave.
>>>
>>> Real problems arise when the frequencies being enveloped are very low, in
>>> this case the low pass filtering is in conflict with the envelope being
>>> derived. Again, it is a matter of knowing what the results are going to be
>>> with a practical envelope, rather than knocking oneself out trying to
>>> produce an 'ideal' envelope.
>>>
>>> paul perry Melbourne Australia
>>>
>>>
>>>
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