[music-dsp] Auto-tune sounds like vocoder
I’m wondering about why the ever-prevalent auto-tune effect in much of today's (cough!) music (cough!) seems, to my ears, to have such a vocoder-y sound to it. Are the two effects related? Just curious. David Reaves ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] What is resonance?
In the physical world, resonance can generally be observed as the frequency-dependent cyclic exchanging of energy between potential (stored) and kinetic (active) forms. A pendulum is a basic example: at either end of its swing a pendulum exhibits purely its maximal stored potential energy while momentarily there is zero (kinetic) motion; at mid-swing there is no storage (potential) and all energy is kinetic, in motion. Each form of energy has two opposing modes: potential has the two opposite extremes of the swing; kinetic, the two different directions of swing. The amplitudes of the two forms of energy are in quadrature, a 90° cyclic relationship; when charted, the two energy levels trace out sine and cosine forms. In the real world, when continually given new energy with proper timing, a pendulum will swing indefinitely. If the outside source of energy is taken away, the resonance will decay: friction (resistance; damping) will slow and eventually stop the swinging. Within the limits of swing amplitude, the frequency of the pendulum's motion remains the same. If you see such a back-and-forth, energy-trading relationship, there is likely resonance going on. Freezing can possibly be seen as energy storage and stretching can possibly be seen as activity, but unless one ‘feeds' the other and vice-versa, it’s probably not resonance. (I will be pleased if someone corrects any false assumption I have made.) David Reaves On Sun, 22 Jul 2018 22:05:48 -0400 charles morrow mailto:c...@cmorrow.com>> wrote: > > How about freezing and stretching moments. Is this resonance? ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] bandsplitting strategies (frequencies) ?
What I meant when I explained how I derived my crossover frequencies, was that I used actual, typical program audio and **its own inherent spectral energy density distribution** to determine empirically (over many months work) generally where to set the frequencies. Much typical full-frequency-range music tends to have equal amounts of RMS energy above and below a dividing point in the range of roughly 500-600Hz. Further splitting those two bands into four equal-program-energy bands resulted experientially in the 150 and 1800 splits. This determination is approximate; these frequencies are not hard and fast, never mind “magic,” and there will be exceptions, but the product I designed was known for its natural-ness on all music types: classical, pop, jazz, etc., and also worked well on voice. The fact that single-pole subtractive filters are extremely broad AND sum back to the original with extremely low transient distortion, was also helpful. If what you do involves material with an unusual spectral balance, and/or if you use aggressive filter roll offs and/or you use something other than RMS detection, then my assumptions may not be useful. David Reaves Sent from my iPad On Tue, 27 Mar 2018 15:10:12 +0200, gm <g...@voxangelica.net> wrote: > > > i keep dividing into equal bands on a log2 scale, > > I believe thats equal energy on a -6dB/octave spectrum and gives figures > very close > > to what David Reaves suggested the other day for 4 bands when you set > 6300 Hz as the upper limit > > and 150 Hz corner frequency for the bass band (or 45 Hz for the lower limit) > ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] bandsplitting strategies (frequencies) ?
I designed a four-band AGC for broadcast about 20 years ago, using single-pole subtractive band split filters. I used RMS detection, and wanting each band’s processing to be doing roughly equal ‘work,’ determined the crossover frequencies on the basis of typical energy distribution. This worked out to around 150 Hz, 500 Hz and 1800 Hz, and the processor sounded extremely natural on pretty much all sources. Kind Regards, David Reaves Recklinghausen, German > On Mar 23, 2018, at 5:01 PM, music-dsp-requ...@music.columbia.edu > <mailto:music-dsp-requ...@music.columbia.edu> wrote: > > Date: Fri, 23 Mar 2018 15:05:47 +0100 > From: gm <g...@voxangelica.net <mailto:g...@voxangelica.net>> > To: music-dsp@music.columbia.edu <mailto:music-dsp@music.columbia.edu> > Subject: > Message-ID: <ba3456f4-1b94-3fb6-a5a6-1702cb485...@voxangelica.net > <mailto:ba3456f4-1b94-3fb6-a5a6-1702cb485...@voxangelica.net>> > Content-Type: text/plain; charset=utf-8; format=flowed > > > The purpose is multiband compression and distortion. > > So I only have a few bands, 2 to 5. > > I use ERB scale in my vocoder, which worked slightly better than Bark > scale for me (it seems better defined at the low range) > > I was wondering if I should use it here too or if it's better on a log2 > scale. > > Also I cant decide what upper and lower frequency I should use when I > divide evenly on a log scale. > > I chose 100 Hz cause thats the lowest Bark band I think. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?
Fascinating thread! I find the “oar in the water” example to be very useful in visualizing the subject at hand. This reminds me of how difficult it is to get really good low frequency response: An oar moving very slowly simply will not move a boat at all. A loudspeaker with its cone fully extended to one extreme does not permanently change the room pressure; the pressure change collapses soon after the movement ends. So we typically do not record absolute pressure, but rather the *change* in this pressure. In either digital or analog, we can record down to DC, if we like. But typically we don’t. The complex physics behind all this is probably more than I can conceive, but it’s still really interesting to poke around the edges. I wonder whether there has ever been a microphone that records absolute pressure, as opposed to pressure changes... Kind Regards, David Reaves Recklinghausen, Germany ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] idealized flat impact like sound
What you are describing sounds a bit like a description of thunder: a sharp, wideband pulse followed by reverberant randomness, (though not spectrally flat due to environmental absorption). Perhaps you can use that as a model? David Reaves Recklinghausen, Germany On Wed, 27 Jul 2016 19:00:02 +0200, gm <g...@voxangelica.net> wrote: > Hi > > I want to create a signal thats similar to a reverberant knocking or > impact sound, > basically decaying white noise, but with a more compact onset similar to > a minimum phase signal > and spectrally completely flat. > > I am aware thats a contradiction. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Trapezoidal integrated optimised SVF v2
I think the distinction is that SCIENCE is open-minded. Scientists, OTOH, are only open-minded if they choose to be. But then, the closed-minded ones aren't really scientists, now are they? ;-) David Reaves Recklinghausen, Germany On Fri, 08 Nov 2013 22:34:51 +, David Hoskins cont...@quikquak.com wrote: snip As with all disciplines, when someone doesn't speak the 'correct' technical language they are shunned, mainly because the academics have invested all that time and money into said work, an automatic defensive stance clicks in. As an aside, in Universities people can be shut out for decades simply for disagreeing with the current paradigms, taking years to prove themselves correct all along - so much for scientists being open minded. Dave. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Effects paradigms
Stephen, I have two books which I have found to be indispensable, Digital Audio Signal Processing by Udo Zölzer, and DAFX: Digital Audio Effects edited by Zölzer, with numerous authors. They're my 'go-to' books when I need background and ideas on the subject, offering both conceptual and implementational info. Perhaps others on the list will have other books or sources they can recommend. David Reaves Recklinghausen, Germany On Fri, 04 Oct 2013 15:58:07 +1100 ChordWizard Software corpor...@chordwizard.com wrote: Hi all, I'm wondering if someone can point me to some good background articles that illuminate what is happening to the signal with common effects such as reverb, chorus, flanger, etc. I'm not specifically talking about algorithm strategies here - although I am also interested in them, if you can recommend anything. Rather I am generally curious about the conceptual processes that are being implemented with these effects. For example, I imagine that reverb is adding a series of duplicates of the original signal at regular delays with descending amplitudes. Or is there something more to it? Obviously overdrive involves clipping, but there are so many varieties around, there must be a lot more to it than that. And I have very little idea what is happening with chorus, flanger, phaser, etc. Why does chorus often have a stereo output, does this naturally arise from the effect design? Any pointers much appreciated. Regards, Stephen Clarke Managing Director ChordWizard Software Pty Ltd corpor...@chordwizard.com http://www.chordwizard.com ph: (+61) 2 4960 9520 fax: (+61) 2 4960 9580 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Boulez
Douglas, I think my wife Ariane, who plays First Violin in the Neue Philharmonie Westfalen, can relate to your description of the activities of an orchestra musician. A few years back she was reprimanded when an audience member told management they had seen (from up in a balcony) a female violinist playing Sudoku during a slow section of a musical. Ari had hoped no one would notice her, tucked away down in the pit, LOL. Better to just put a book on the music stand. ;-) Kind Regards, David Reaves On Sat, 25 Feb 2012 20:43:04, douglas repetto doug...@music.columbia.edu wrote: snip But that's not really how live musicians tend to think of it. It's not like a violinist keeps her bow moving at all times and only touches it to the strings when there's a note to be played. But that's kinda what sending zeros to a buffer when there's no sound is like. On the other hand, if you work directly in hardware (say using an analog synth, hooking up logical oscillators, or programming a microcontroller) you can take a very different approach. You twiddle some output pins when you want sound and when you don't want sound you can just go off and do other things. In many ways I think that's a lot more like what many musicians do -- when you're not playing (either because you've got a bunch of rests, or maybe you're playing improvised music and you're just sitting out for awhile, or whatever) you don't really sit there counting off the beats. You stop playing. You might think about other things. After awhile hopefully you'll notice that the conductor is about to cue you in, or you get an idea and decide to join the improvisation, etc. I've seen people reading books in Broadway orchestra pits... snip -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Splitting audio signal into N frequency bands
Thilo, I would ask the question as to if your ultimate goal is really to totally and completely separate the bands, or, rather, to control levels in a subtle, musical manner. If the former, then ask yourself what happens when an instrument or voice straddles the crossover frequency, moving from one band to another. If the band borders are abrupt, when the gains are very different from band-to-band the result will be very obvious, and unnatural. On the other hand when using gradual filters for crossovers, such as first order IIR, it will be extremely unusual to notice any such effect. That's been my experience, and I have successful commercial product designs in everyday use that exploit this as a feature, not a bug. For my purposes, since a multi-band compressor is ultimately designed to be listened to, the solution that is least objectionable sonically is the one I choose. And if it is simpler, so much the better. :-) Kind Regards, David Reaves On Monday, October 31, 2011 10:47 AM Thilo K?hler koehlerth...@gmx.de wrote: Hello all! I have implemented a multi-band compressor (3 bands). However, I am not really satisfied with the splitting of the bands, they have quite a large overlap. What I do is taking the input singal, perfoming a low pass filter (say 250Hz) and use the result for the low band#1. Then I subtract the LP result from the original input and do a lowepass again with a higher frequency (say 4000Hz). The result is my mid band#2, and after subtracting again the remaining signal is my highest band#3. I assume this proceedure is appropriate, please tell me otherwise The question is now the choise of the filter. I have tried various filters from the music-dsp code archive, but i still havent found a satisfiying filter. I need a steep LP filter (12db/oct or more), without resonance and fewest ringing possible. The result subtracted from the input must works as a HP filter. Are there any concrete suggestions how such a LP filter should look like, or is there even a different, better way to split the audio signal into 3 bands (or N bands)? I know I can use FFT, but for speed reasons, I want to avoid FFT. Regards, Thilo Koehler -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Splitting audio signal into N frequency bands
Thilo, If you use traditional (analog-similar) IIR filters, you can create a simple, first-order low-pass, then subtract that signal from the original to create a perfectly-summing, complementary high-pass. This is what I usually do. To create more bands, I just do the same thing again to those signals. The crossovers in Aphex's (analog) Dominator multiband limiter use this method. My analog and digital Ariane AGC designs use it for four bands. Keep in mind that even a basic first-order filter is ultimately 20dB per decade, which really is pretty useful. When you use two-pole (second-order) filters, not only is the design more complex, you also risk phase anomalies around the crossover point, usually requiring you to invert the polarity of one of the bands. For perfect summation, you can use the same trick i.e., subtracting the two-pole low-pass from the original signal, but the resultant high-pass created will only be single-pole. If delay time is no issue, and your computing power plentiful, you can do ANYthing with FIR filters. :-) I've never had that luxury. Kind Regards, David Reaves Recklinghausen, Germany On 02 Nov 2011 14:21:21, Thilo K?hler koehlerth...@gmx.de wrote: Hello David! I would ask the question as to if your ultimate goal is really to totally and completely separate the bands, or, rather, to control levels in a subtle, musical manner. The latter case. Total seperation is NOT wanted, because a single sin-tone should not jump between two bands, if it is close to the splitting frequency. So a reasonable overlap is desired. If the former, then ask yourself what happens when an instrument or voice straddles the crossover frequency, moving from one band to another. If the band borders are abrupt, when the gains are very different from band-to-band the result will be very obvious, and unnatural. Yes. I fully agree. For my purposes, since a multi-band compressor is ultimately designed to be listened to, the solution that is least objectionable sonically is the one I choose. And if it is simpler, so much the better. :-) So, may I ask what kind of filter did you use and how your splitting strategy looks like? First order IIRC seems too flat to me, I liked second order better because the seperation is more clear. But I havent played around enough to tell if that also works best for the acutal multi-band compression. Regards, Thilo Koehler -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
