• s.elliot.perez

    Ah, yes, I see the two types of Comb Filter on Wikipedia. So the Feedback-version has higher peaks and a steeper curve because of cumulative feedback?

    I actually mostly use PD nowadays with the Hvcc-compiler, which converts patches into C-code and then wraps it for different platforms. I don't know how those plugins are affected by send~ & throw~, and the makers of hvcc disbanded and stopped answering my mails. T-T

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  • s.elliot.perez

    @seb-harmonik.ar So the "comb filter" is the system of delays with feedback? How does one specify the fundamental thereof? The length of the delays? Is the fundamental about 1.133Hz in this case?

    By the way, here's Farnell's original patch: striker.pd

    and here's a version without [send~] or [throw~] objects: telephoneBell.pd

    and they do indeed sound different, with the latter sounding louder/clearer. Did I make a mistake somewhere? Or could this indeed be due to the added delay of [send~] / [throw~] objects? I admit I stopped using them in the past because I noticed they created some kind of weird partial phase cancellation that wasn't happening with direct wire connections.

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  • s.elliot.perez

    Thanks, Sébastian, you've explained it very well and I understand.

    The comb-filter thing doesn't have anything to do with the [bp~] objects, right? So with a sound that has a lot of non-harmonic partials, like a bell, it could end up sounding more consonant (assuming some of the non-harmonic partials are about halfway between the harmonic partials) because of the casing?

    Does anyone know if that's true about send~/throw~/receive~/catch? Is it faster to use wire connections?

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  • s.elliot.perez

    Hello,

    In "Designing Sound" by Andy Farnell, p. 378-379, he gives the following description of how to approximate the resonance duration of bells inside of a telephone box:

    Making a Box
    The resonator in figure 29.14 sends its input signal into two delays that feedback into each other. Between them is a fixed filter that mimics the material properties of the box and a that serves to limit the signal and introduce a little distortion to create a brighter sound. An old telephone has a box about 20cm to 30cm (12 inches) square. From the speed of sound being 340m/s we get a resonance of 0.3/340ms or about 1.1kHz. Two delays are used with the length being slightly longer than the width. Tapping some Bakelite shows resonances somewhere between a hard wood and plastic, and for a plate about 5mm thickresonances were seen at 1kHz, 500Hz, and 300Hz. Choosing the exact values for the patch in figure 29.14 requires a bit of tweaking by hand. You want to find two values for the acoustic resonance and set the width, length, and feed-back to give a good effect, while at the same time you must pick filter valuesthat give a nice hollow radiated tone. Picking feedback values close to 0.7 givesgood results; then tune the filters to get the quality of a plastic box without accidentally sending the acoustic resonator into unstable feedback.

    image.png

    Now, I understand that 0.3/340ms = 1133Hz (about 1.1kHz) and 1/1133 = 0.88ms which is the value he used for one of the delays in his "casing" patch. What I don't understand is how he got the resonance value of 0.3/340ms to begin with. I assume the 0.3 comes from the 30 centimeters, ie. the width of the box, but what about the 340 ms? It looks related to the speed of sound, but how exactly? And is this a formula I can reuse, ie.

    w/340 = r

    where w is the width in meters and r is the resonance frequency in Hertz? I tried Googling this but the resonance formulae that came up were much more complex.

    Also, apologies if this topic is less about PD and more about Math/Acoustics. In the future, would it be better to ask questions like this on a math forum instead?

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  • s.elliot.perez

    @seb-harmonik-ar Thanks for the response, but I really wanted you to start with a "Yes" or "No" to my question "Am I on the right track?", lol. Since you talked about the impulse response as well, I take it to mean I was somewhat on the right track... but I was mistaken about the shape of the ramp being part of sine wave? Rather, it's just exponential for some reason? After that, you kind of lost me... using rpole~ instead of lop~ will just raise more questions for me, I'm afraid.

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  • s.elliot.perez

    @whale-av Let me see if I understand this... in the theoretical part of Farnell's book, "Designing Sound" (7.3 Generating Digital Waveforms), he writes that an impulse spike contains all frequencies (based on the Uncertainty Principal). So when a sudden change in value is sent into the [lop~] (replacing the [line~] in your get.pd with a [sig~] doesn't change anything noticeably), it could be said that waves of all frequencies go from 0 to 1 or 1 to 0 at their respective speeds. By filtering out all the frequencies above 1, the fastest wave to make the transition becomes 1. For 0 to 1, this transition takes the shape of cos 18π-cos 24π (0.75-1) and for 1 to 0, cos 6π-cos 12π (0.25-5).

    Am I on the right track? If so, what would be the equation to calculate the length of the transition based on a given filter pole a? I imagine it would involve dividing the wavelength/duration by 4 at some point, but I'm unsure of exactly how to do it.

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  • s.elliot.perez

    Hello,

    I've recently started studying Andy Farnell's book Sound Design and am learning a few new things about Pure Data. Sometimes the Math concepts are hard to wrap my head around but I'm mostly hanging in there. In this example link text, he shows how to use a [lop~] to in lieu of a [line~] which I'd never heard of doing before. Could someone explain to me how this works? I imagine it has something to do with the lop~ needing one full period of a 1Hz wave to check if incoming sounds are below the point of 1 or not before letting the changes through... but I really don't get it. So to clarify, my questions are:

    1. How does it work?
    2. How would you get envelope lengths other than 1? I tested, and it doesn't seem to operate on a Time to Frequency relationship. A frequency of 0.25 seems to make an envelope faster than 4 seconds.

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  • s.elliot.perez

    Hello,
    Does anyone happen to have a PD object into which you can input the Carrier Frequency, Modulator Frequency and Index of a Frequency Modulation, and get back a spectrum analysis, e.g. 4+ of it's loudest frequencies in Hz? I'm using Heavy, a platform for converting PD patches into plugins (for Unity, C, WWise, etc.), but it doesn't have (m)any options for pitch detection (not all PD objects are compatible), so I figure something like this would be the solution... but it's not simple due to FM's ability to produce infinite sidebands...
    I looked up some info, but there's an omega symbol that represents angular frequency of the carrier in radians per second and I'm not sure how to get that value: http://www.radio-electronics.com/info/rf-technology-design/fm-frequency-modulation/spectrum-bandwidth-sidebands.php
    http://literature.cdn.keysight.com/litweb/pdf/5954-9130.pdf

    Any ideas?

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  • s.elliot.perez

    Yes, I had a similar problem recently with throw~ and catch~. The signal gets quieter when two things are thrown to a catch~, but louder (the desired volume) when put directly into the dac~...

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