### Re: [music-dsp] Is beating the same thing as flanging?

On Nov 19, 2010, at 12:28 PM, Theo Verelst wrote: Of course digital filtering and processing is often not resampled (for the obvious consideration that that process is far less than causal, computation intensive and hard even when the Niquist filtering is done properly), so that filter deleys

### [music-dsp] who else needs a fractional delay.

On Nov 19, 2010, at 3:42 PM, Alan Wolfe wrote: i fear to post a question being the OP of this huge 100+ message thread but... it was mentioned here and in a previous email that for digital flangers you want to interpolate between samples for best results. Would you want to do this for all

### Re: [music-dsp] who else needs a fractional delay.

On Nov 19, 2010, at 6:33 PM, Scott Gravenhorst wrote: https://ccrma.stanford.edu/~jos/Interpolation/ Lagrange_Interpolation.html Linear interpolation over 1 sample delay time. two notes: 1. linear interpolation while not sounding as sophisticated as first-order Lagrange interpolation,

### Re: [music-dsp] [OT] vinyl? No, thanks...

On Nov 20, 2010, at 2:34 PM, Victor Lazzarini wrote: This is because they have probably not experienced building a 1000+ vinyl collection only to see it disintegrate along the years, with crackles, pops and scratches. Every time I picked up one of favorite albums and discovered a new

### Re: [music-dsp] Algorithms for finding seamless loops in audio

On Nov 26, 2010, at 2:21 AM, Ross Bencina wrote: robert bristow-johnson wrote: you can have a periodic (or quasi-periodic) signal with absolutely no energy at harmonic #1 (what i would call the fundamental), and as long as it has energy in most other odd harmonics, the autocorrelation

### [music-dsp] A theory of optimal splicing of audio in the time domain.

a few mistakes are spotted and corrected before i forget This is a continuation of the thread started by Element Green titled: Algorithms for finding seamless loops in audio As far as I know, it is not published anywhere. A few years ago, I was thinking of writing this up and

### Re: [music-dsp] FIR filter question

On Dec 6, 2010, at 12:33 PM, Nigel Redmon wrote: If I understand correctly, you want to take an arbitrary one-cycle wav and build mip-map tables, dropping out upper harmonics successively. ... But, it seems like this might be a better fit for the frequency domain--why not do an FFT, and

### Re: [music-dsp] Interpolation for SRC, applications and methods

On Dec 21, 2010, at 11:22 AM, Nigel Redmon wrote: As for things like distortion modeling of guitars, I can tell you that windowed sinc is involved, at least on the upsampling leg where you likely want to preserve phase. ... As long as you lowpass filter the signal first, then you're only

### Re: [music-dsp] Interpolation for SRC, applications and methods

On Dec 21, 2010, at 10:45 PM, Ross Bencina wrote: robert bristow-johnson wrote: one thing i might point out is that, when comparing apples-to- apples, an optimal design program like Parks-McClellan (firpm() in MATLAB) or Least-Squares (firls()) might do better than a windowed (i presume

### Re: [music-dsp] Bandlimiting, Aliasing and Reconstructed Signals

On Dec 23, 2010, at 8:56 AM, Charles Turner wrote: Here's my limit case: let's assume some typical laptop with CD- quality sound generation capability with a sample rate of 44.1khz and sample size of 16 bits. I create a sinusoidal waveform on the computer with a period of 4,410hz. I choose

### Re: [music-dsp] Interpolation for SRC, applications and methods

() function and is not directly because of half- band filter, but it *happens* to be the case that for 2x upsampling, this windowed-sinc is *also* a half-band filter. On Dec 24, 2010, at 5:16 AM, Nigel Redmon wrote: On Dec 23, 2010, at 9:58 PM, robert bristow-johnson wrote: In what I'm talking

### Re: [music-dsp] Interpolation for SRC, applications and methods

will confess that when i said early on: On Dec 23, 2010, at 1:18 PM, robert bristow-johnson wrote: On Dec 23, 2010, at 12:46 PM, Nigel Redmon wrote: On Dec 21, 2010, at 8:36 PM, robert bristow-johnson wrote: and trying to point to an obvious advantage to any windowed sinc (that you don't have

### Re: [music-dsp] Interpolation for SRC, applications and methods

: On Dec 27, 2010, at 11:03 AM, robert bristow-johnson wrote: On Dec 27, 2010, at 2:05 AM, Nigel Redmon wrote: ... Ideally, you would want everything from 0.50 to 1.00 to be clear to a reasonable degree. It's not. It's down 6 dB at .50, and hits the -90dB stop-band at about 0.70. (You can get

### Re: [music-dsp] Interpolation for SRC, applications and methods

On Dec 28, 2010, at 11:51 PM, Nigel Redmon wrote: Would it have been better if I said, I can tell you that windowed sinc is used? Hmm, I have a feeling that you might read that as ... is exclusively used, not sure... depends on what the meaning of is is. when Bill Clinton, when first

### Re: [music-dsp] Interpolation for SRC, applications and methods

On Dec 29, 2010, at 9:10 PM, Nigel Redmon wrote: i think we skeered 'em, Robert ;-) my driver's license photo looks pretty scary (but my facebook, linked- in, whatever isn't so scary). i got accosted once by plain-clothes NYC cops about a year ago. they said they stopped me because i

### Re: [music-dsp] resonance

On Jan 4, 2011, at 11:03 PM, Didier Dambrin wrote: My new additive synth features full control on the filter, and I learnt a lot about good sounding resonance. Since I can control pretty much anything, I can shift the resonance point around the cutoff point, it's very useful musically.

### Re: [music-dsp] Bandlimiting, Aliasing and Reconstructed Signals

an alias, Rick's book refers--clearly--to multiple images as aliasing. Not saying right or wrong (I probably don't qualify to make that call). Like you, I use aliasing to mean the thing we don't like, not the images, which just exist. On Dec 23, 2010, at 8:41 AM, robert bristow-johnson

### Re: [music-dsp] resonance

On Jan 5, 2011, at 12:02 AM, Didier Dambrin wrote: I said additive :) I was talking fully in the freq domain, it's really nice to be free of the restrictions of IIRs (which I never really understood). i get it. no post-filtering. so are you applying to the additive components some

### Re: [music-dsp] Approaches to multiple band EQ

On Jan 11, 2011, at 1:23 PM, Thomas Young wrote: I need to develop a real-time multiple band EQ DSP effect, but I am unsure about how to approach it. do you mean a graphic EQ? My preferred approach would be to FFT- Modify Spectrum- IFFT, if you do that, better look up the concepts

### Re: [music-dsp] Approaches to multiple band EQ

On Jan 11, 2011, at 7:01 PM, Tom Wiltshire wrote: I'd approach this from a analogue-thinking angle and design a tunable parametric EQ stage and then parallel a load of them up, like Robert suggested. that's not exactly what i meant to suggest. what goes in parallel are not simply these

### Re: [music-dsp] Autocorrelation - probably a daft question

On Jan 25, 2011, at 7:34 PM, Jan Baumgart wrote: When the two signal portions are alike, they are strongly correlated - so you get a maximum value for the correlation. If they have nothing in common you get a correlation value near zero.\ he said he was using periodic function generation.

### Re: [music-dsp] New patent application on uniformly partitioned convolution

On Jan 28, 2011, at 4:47 PM, Nigel Redmon wrote: I've been on a number of patent cases (as software expert, sometimes electronics), big players, on both sides... First, patents are important, and help progress. Non-obvious advances often come from expensive and lengthy research. Imagine

### Re: [music-dsp] digital EQ (passive) adding gain

another way to think about it is to pretend that your filter, whatever it is, is a matched filter. matched to what? you say. it's matched to a signal that looks just like a time-reversed copy of the filter's impulse response. so whatever the impulse response of the filter is, if there

### Re: [music-dsp] Frequency-dependent latency of a filter

On Mar 17, 2011, at 9:21 AM, Wen X wrote: As far as causality is concerned it's the *group* delay that should be non-negative. well, even group delay is negative with the peaking filters, for *some* frequencies. with group delay, there is no issue of phase unwrapping since the phase

### Re: [music-dsp] Frequency-dependent latency of a filter

On Mar 17, 2011, at 12:00 PM, Wen X wrote: From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson well, even group delay is negative with the peaking filters, for *some* frequencies. Yes, but only if the filter

### Re: [music-dsp] Frequency-dependent latency of a filter

On Mar 17, 2011, at 2:27 PM, xue wen wrote: Yes, but only if the filter has high (negative?) dispersion at that frequency. i'm not sure what that means. my understanding of dispersion would be a rapid change of phase or delay vs. frequency. my understanding is if different

### Re: [music-dsp] Sinewave generation - strange spectrum

On Apr 27, 2011, at 1:38 AM, Ross Bencina wrote: eu...@lavabit.com wrote: *out++ = data-amplitude[0] * sinf( (2.0f * M_PI) * data-phase[0] ); *out++ = data-amplitude[1] * sinf( (2.0f * M_PI) * data-phase[1] ); /* Update phase, rollover at 1.0 */ data-phase[0] += (data-frequency[0] /

### Re: [music-dsp] Trapezoidal and other integrationmethodsappliedtomusical resonant filters

On May 17, 2011, at 6:27 PM, Ross Bencina wrote: robert bristow-johnson wrote: even though the cookbook yields coefficients for Direct 1 or Direct 2 forms, it's pretty easy to translate that to the state- variable design if that is the form you wanna use. I've often wondered about

### Re: [music-dsp] Trapezoidal and other integration methods appliedtomusical resonant filters

On May 20, 2011, at 7:43 AM, Ross Bencina wrote: robert bristow-johnson wrote: i don't have time now to complete the analysis, but here is my first pass at getting the z-plane transfer function (something to compare to the DF1 or DF2). Thanks very much Robert, yer welcome. i think i

### Re: [music-dsp] Trapezoidal and other integration methods appliedtomusical resonant filters

On May 21, 2011, at 11:27 PM, robert bristow-johnson wrote: t-1t y(t) = integral{ x(u) du} + integral{ x(u) du} -inf t-1 ~= t(t-1) + x(t) this should

### Re: [music-dsp] Trapezoidal and other integration methodsappliedtomusical resonant filters

On 5/22/2011 5:27 AM, robert bristow-johnson wrote: [...] which might be what Hal gets, i think. it's the only way to make the claim that the Qc coefficient is independent of w0 and depends only on Q. but if the resonant frequency is closer to Nyquist, you need to scale Q with a sinc

### Re: [music-dsp] Hardware Sampler Timestretch

On Jun 26, 2011, at 7:52 PM, Didier Dambrin wrote: pretty sure that on a piece of hardware 20 years ago, it couldn't be anything else / anything in the freq domain As an introduction to time stretching I thought I'd try and emulate how some of the older hardware samplers used to do it.

### Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

On Jul 13, 2011, at 9:29 AM, Olli Niemitalo wrote: On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson r...@audioimagination.com wrote: On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote: [I] chose that the ratio a(t)/a(-t) [...] should be preserved by preserved, do you mean constant

### Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

On Jul 14, 2011, at 5:36 PM, Olli Niemitalo wrote: On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson r...@audioimagination.com wrote: g(t) = 1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 ) might this result match what you have? Yes! I only derived the formula for the linear ramp, p(t

### Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

On Jul 15, 2011, at 12:46 AM, Sampo Syreeni wrote: What are you trying to accomplish here, really? Optimum splicing, sure, but against which precise criterion? the precise criterion is how well the two signals being spliced correlate to one another. i tried to set that up with the

### Re: [music-dsp] Reverb removal

On Jul 29, 2011, at 5:00 AM, Alexandros Tsilfidis wrote: In the dereverberation context, room reverberation is regarded as the combination of early reﬂections and late reverberation. It is well known that early reflections produce a spectral degradation which is perceived as coloration,

### Re: [music-dsp] Reverb removal

p6TMjFnh006345 for music-dsp@music.columbia.edu; Fri, 29 Jul 2011 22:45:16 GMT Message-Id: 47043a22-b58b-4d04-81f5-369a4bd0f...@audioimagination.com From: robert bristow-johnson r...@audioimagination.com To: A discussion list for music-related DSP music-dsp@music.columbia.edu In-Reply

### Re: [music-dsp] Precise, Real Time Pitch Shift with Formant Control

On 8/2/11 9:32 AM, Igor Brkic wrote: On Tue, Aug 2, 2011 at 2:28 PM, Conley, Dylan dylan.con...@marquette.edu wrote: Is anyone aware of an open source pitch-shift algorithm implementation that is quick ( 2ms) precise (to within 0.5 cents) and leaves the formant intact? ... you can do that

### Re: [music-dsp] Precise, Real Time Pitch Shift with Formant Control

On 8/2/11 12:01 PM, Wen Xue wrote: This might be purely theoretical - but can you pitch-shift something below 500Hz with2ms delay at reasonable precision? no, not in a meaningful way. i didn't realize in my earlier response that the OP spec'd that. it's an unreasonable spec. There doesn't

### Re: [music-dsp] Precise, Real Time Pitch Shift with Formant Control

On 8/2/11 2:04 PM, Steffan Diedrichsen wrote: Since you implement for a synthesizer, you may look into the option for an off-line pitch detection and real-time grain-synthesis. Grain synthesis has a nice formant control and is fairly easy to implement. i think that this grain synthesis is

### Re: [music-dsp] Precise, Real Time Pitch Shift with Formant Control

On 8/4/11 8:32 AM, Conley, Dylan wrote: I did have a couple follow up questions that I hope aren't too irrelevant. Because we are working with the VST spec (and temporarily within an implementation of the Java MIDI Interface) we will have access to all MIDI information. Assuming the

### Re: [music-dsp] Electrical Engineering Foundations

well, the math for the sampling and reconstruction theorem (from where we understand the zero-order-hold effect on frequency response from a conventional D/A converter and from where we understand the basis of bandlimited interpolation, resampling or sample-rate conversion) is pretty

### Re: [music-dsp] music-dsp Digest, Vol 92, Issue 7

On 8/24/11 1:51 PM, Andy Farnell wrote: ... So my question for you Theo ... put on the profs hat... How would you make these very powerful and (to me) wonderful and mind boggling things in signals theory interesting and relevant in an age where we have to compete with autotune and facebook?

### Re: [music-dsp] FM Synthesis

i am not a Java programmer, but i think i can read this code. where does the symbol buffer[] get declared? i resume you're getting opBuffer[] operator.buffer. private void modulate( final int numFrames ) { clear( numFrames ); // zero buffer for( @NotNull final Link

### Re: [music-dsp] FM Synthesis

what Brad Smith points out (that at least 1 sample delay is necessary for feedback) is true for any discrete-time processing alg. and we know that if block processing or chunk processing (whatever you wanna call the technique) would require a minimum delay of BLOCK_SIZE samples for any signal

### Re: [music-dsp] Splitting audio signal into N frequency bands

On 11/2/11 2:37 PM, David Reaves wrote: 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. this might be when it's useful to look up

### Re: [music-dsp] Orfanidis-style filter design

On 11/27/11 3:17 PM, Dominique Würtz wrote: Any ideas? Knud Christensen A Generalization of the Biquadratic Parametric http://www.aes.org/e-lib/browse.cfm?elib=12429 Hmm, reading the abstract I'm not 100% sure if it really addresses what I'm aiming at. Sorry for being sceptical,

### Re: [music-dsp] Orfanidis-style filter design

On 12/8/11 4:36 PM, Theo Verelst wrote: robert bristow-johnson Sun Nov 27 17:29:14 EST 2011 wrote: On 11/27/11 3:17 PM, Dominique Würtz wrote: Any ideas? Knud Christensen A Generalization of the Biquadratic Parametric http://www.aes.org/e-lib/browse.cfm?elib=12429 Hmm, reading

### Re: [music-dsp] Orfanidis-style filter design

On 12/9/11 12:55 AM, Michael Olsen wrote: Robert, well, since, i have received a pdf copy of the Christensen paper. i am willing to send it along to any small quantity of people who ask. i realize the AES would rather that people get the paper from them and pay for it, but if the cost is

### [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

there's a guy there with handle Clusternote (who might be lurking here for all's i know) who is slugging it out with an IP (can't imagine who that is) about the math that goes into additive synthesis. if you ever bother to edit the en WP, it might be a good time to examine the article and

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

On 1/9/12 11:00 AM, Victor Lazzarini wrote: Wouldn't it be nice if all of the knowledge embodied in this list could find its way into Wikipedia, fixing the howlers and myths that exist in some of the audio, synthesis, effects, computer music, etc pages? I know that some of us have at time

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

On 1/9/12 11:58 AM, Scott Nordlund wrote: I looked at it a bit, and it's a lot to juggle, looking at diffs and the back and forth. Maybe it's just getting late, and I played a lot of basketball earlier, but the final thing that told me it's bed time was, in skimming the article, Its [RMI]

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

On 1/10/12 9:31 PM, Alen Koebel wrote: I get paid to write, so I'm no stranger to research. I have edited the work of others and had my work edited. Many here can say the same, I'm sure. With that background I have tried to edit articles on Wikipedia. IMO, Wikipedia is fundamentally a bad

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

On 1/10/12 11:29 PM, Scott Nordlund wrote: On January 9, 2012 at 3:02:04 PM Veronica Merryfield veronica.merryfield@shaw.cawrote: My feel is that to make it right, it probably needs more than a bit of adjustment. If this is to be fixed, I think it needs to be an organized effort. I scan down

### Re: [music-dsp] anyone care to take a look at the Additivesynthesis article at Wikipedia?

On 1/12/12 2:41 AM, Ross Bencina wrote: On 12/01/2012 4:01 AM, robert bristow-johnson wrote: well, i cannot tell that the WP admins are going to do anything about this other than wait for the page protection to expire (about 26 hours) and then see what happens. if enough of us converge upon

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

hey, i appreciate the help from folks here (namely Olli and Ross) dropping in on that Wikipedia article, now that it has been released from protection. please don't go away, there is lotsa stuff to do and we have time to do it. it appears that this editor who wanted to rewrite everything

### Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?

On 1/16/12 1:16 AM, Nigel Redmon wrote: Nice improvements. This may seem like nitpicking, but the Timeline of additive synthesizers section seems to choose keeping the instrument name as the start of the sentence over proper grammar. For instance: Hammond organ, invented in 1934[26], is

### Re: [music-dsp] choice of Q for graphic equalizers

On 2/6/12 3:28 PM, Nils Pipenbrinck wrote: A quick question: I am writing a little 31 band graphical equalizer (three bands per octave), and I want to use the peaking-eq biquads from Roberts excellent filter cookbook. Everything is working fine so far, but I wonder what Q should I choose for

### [music-dsp] test

test. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- 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

### Re: [music-dsp] a little about myself

you're not related to Miller Puckett, are you? just curious. and you're still welcome to the group no matter the answer. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website:

On 2/22/12 9:20 AM, douglas repetto wrote: This is driving me nutz: http://www.google.com And now an image search for Hertz features lots and lots of pictures of a non-sinewave! Arrg! i was wondering if it was the same Hertz. i guess it is. sometimes Google's authority is dubious.

### Re: [music-dsp] a little about myself

On 2/20/12 10:28 AM, douglas repetto wrote: Hi Adam, Welcome to the list. It's slow right now, but no doubt it'll flare up again soon! no shit -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing

### [music-dsp] high-level vs. low-level coding of algs

changed the subject line to something more accurate... On 2/26/12 9:25 AM, Ross Bencina wrote: On 27/02/2012 1:11 AM, Brad Garton wrote: We're fooling around with the new Max/MSP gen~ stuff in class, it seems an interesting alternative model for low-level DSP coding. Once they figure out how

### Re: [music-dsp] More on the job opening

On 3/24/12 4:45 PM, Linda Seltzer wrote: Kindly allow me to provide further information on the job ad. The experience requires advanced degrees in engineering or physics (this is not a position for a music major unless the music major double majored in engineering or physics). The areas of

### Re: [music-dsp] OSC problem on STM32F4Discovery

i hadn't heard of this dev board before. at http://www.st.com/internet/evalboard/product/252419.jsp it says that the single unit prices is US\$14.9 . is that right? that's nearly free. where do the software tools (the compiler/linker/loader/etc) come from? regarding wavetable indexing,

### [music-dsp] testing 1,2,3

i dunno why, but i can no longer reply to the thread that Julian started. if this post gets to the list, then i think there is some damaged header or something. this has happened to me before and it only happens with this mailing list. after hitting Send, Thunderbird tries sending it and

### [music-dsp] testing different subject header

testing 1,2,3... this is identical to a previous message (that would not get past my SMTP) with this sentence added and the subject header changed.. On 4/9/12 5:25 PM, Julian Schmidt wrote: Am 09.04.2012 23:22, schrieb Olli Niemitalo: On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt

### Re: [music-dsp] OSC problem on STM32F4Discover

now that i think of it, if you're doing linear interpolation and you forget to add that extra repeated point at the end of the wavetable: float wavetable[257]; // one extra point for doing linear interpolation // make sure that wavetable[256] = wavetable[0] you will

### [music-dsp] okay, so i got my STM32F4-Discovery board in the mail today.

it was pretty spare in the mail. essentially just the board and a cute little card in a bubble box. the card has some Getting started instructions and number 5. says to got to http://www.st.com/stm32f4-discovery tutorial, and i'll do that soon. it also mentions dev toolchains: Altium

### Re: [music-dsp] okay, so i got my STM32F4-Discovery board in the mail today.

On 4/12/12 10:06 PM, Eric Brombaugh wrote: On 04/12/2012 05:53 PM, robert bristow-johnson wrote: it was pretty spare in the mail. essentially just the board and a cute little card in a bubble box. Yes, that's pretty much all you get. Bring your own mini-USB cable. the card has some Getting

### Re: [music-dsp] To EE or not to EE (Was: Job at Waldorf and Possible Job Opportunity)

are pretty good that your early attempts are/were crap. Many of you know Robert Bristow-Johnson. oh jeepers. He is a bit famous in this group because in part, he did the rb-j cookbook. one-hit wonder. I think it is obvious that Robert needed his engineering education to jump start his

### Re: [music-dsp] Wavetable interpolation

On 5/7/12 5:45 PM, ChordWizard Software wrote: I am working on a new project using PortAudio and testing it with a waveform stored in a buffer. This could be generated myself (sine, square, sawtooth, etc) or a more complex waveform loaded from a file. I want to be able to render the waveform

### Re: [music-dsp] Pointers for auto-classification of sounds?

On 6/8/12 1:36 PM, Charles Turner wrote: I was initially hesitant to post to the list as I haven't explored this topic very deeply, but after a second thought I said what the hell, so please forgive if my Friday mood is more lazy than inquisitive. nothing wrong with posting this. nothing at

### Re: [music-dsp] _ Pointers for auto-classification of sounds?

On 6/13/12 3:22 PM, Andy Farnell wrote: I would second that. My research in the 1990s led to the same conclusion, in essence the parametric space is vast while the perceptually useful space is very small and sparsely dotted around in the param space. Upshot: needle in a haystack i dunno about

### Re: [music-dsp] Noise gate for guitar amplifiers and hysteresis

On 7/4/12 12:44 PM, robert bristow-johnson wrote: On 7/4/12 11:06 AM, Ivan Cohen wrote: Hello rbj ! What do you mean by slew ? Is it a filtering applied on the VCA attenuation ? specifically, *low-pass* filtering. I think the answer to your question is obviously no :) I may have missed

### Re: [music-dsp] compensating for LPF phase delay in damped karplus-strong/comb filter

On 7/23/12 4:52 AM, Oli Larkin wrote: Can anyone here advise me how I can precisely compensate for pitch dependant detuning when my damping filter is active in a tuned comb filter? I'm trying to implement a damping control that doesn't alter the fundamental frequency of the comb filter. I'm

### Re: [music-dsp] DC blocking (again :)

On 7/31/12 4:45 AM, Domagoj Saric wrote: On 30.7.2012. 20:51, robert bristow-johnson wrote: i didn't have anything to do with the subtract-the-moving-average DC block filter. I apologize...at least I attributed too much rather than too little ;) no sweatsky. i generally try to actively

### Re: [music-dsp] DC blocking (again :)

On 8/1/12 5:25 AM, Domagoj Saric wrote: On 1.8.2012. 6:29, robert bristow-johnson wrote: if DC is slowly varying, small displacements of a windowed section of DC (which is what comes out of any weighted moving-average filter) does not change it much. the difference between the IIR vs FIR

### Re: [music-dsp] i need a knee

On 8/10/12 6:23 AM, Bastian Schnuerle wrote: ok, i got it by myself, took a while .. but a small hint would have been nice, you guys have all those books i can not afford and i am only a ee dipl.ing. and they wanted me to build bombs and instead i am coding musical instruments, you should

### Re: [music-dsp] DAFx 2012 - Software Development for Audio and Music Researchers Tutorial

On 10/20/12 10:30 AM, Andy Farnell wrote: Great to see prof Mark Plumbley talking some sense to the train wreck of the present academic trajecory in those slides. On Sat, Oct 20, 2012 at 04:17:29PM +0100, Victor Lazzarini wrote: What do you mean? On 10/20/12 1:29 PM, Andy Farnell wrote:

### [music-dsp] need help with gnuplot and octave on my mac.

say, any among you using a Mac and Octave and gnuplot? i used to be able to plot with Octave, it would start up X11 and if i set the variable GNUTERM=x11 before starting Octave this would work. now it doesn't :-( anybody know what i'm doing wrong? i could use some help. thanks for any.

### Re: [music-dsp] need help with gnuplot and octave on my mac.

file) best Andy On Fri, Oct 26, 2012 at 05:03:14AM -0700, robert bristow-johnson wrote: say, any among you using a Mac and Octave and gnuplot? i used to be able to plot with Octave, it would start up X11 and if i set the variable GNUTERM=x11 before starting Octave this would work. now

### Re: [music-dsp] need help with gnuplot and octave on my mac.

On 10/27/12 2:25 AM, gwenhwyfaer wrote: On 27/10/2012, gwenhwyfaergwenhwyf...@gmail.com wrote: On 26/10/2012, robert bristow-johnsonr...@audioimagination.com wrote: say, any among you using a Mac and Octave and gnuplot? i used to be able to plot with Octave, it would start up X11 and if i

### Re: [music-dsp] stuck with filter design

On 11/18/12 2:33 PM, Shashank Kumar (shanxS) wrote: @ RBJ: Thanks for doing amazing stuff. :) meat and potatoes. check out Julius Smith and CCRMA or companies like Melodyne for the amazing. I have one more question: Why so many people use analog prototypes to get a digital filter ?

### Re: [music-dsp] stuck with filter design

On 11/21/12 8:41 AM, Ross Bencina wrote: On 19/11/2012 6:33 AM, Shashank Kumar (shanxS) wrote: Why so many people use analog prototypes to get a digital filter ? Further to this question, I just came accross this brief but enlightening piece by Ken Steiglitz, it discusses the dawn of the use

### Re: [music-dsp] Call for Papers - DAFx13, Maynooth, Ireland Sept. 2013

On 12/6/12 8:34 AM, Victor Lazzarini wrote: Apologies for cross-posting. DAFX 13 === CALL FOR PAPERS there is absolutely no need to apologize for posting DAFx to music-dsp (or the comp.dsp newsgroup, if you want). -- r b-j r...@audioimagination.com

### Re: [music-dsp] Precision issues when mixing a large number of signals

um, a sorta dumb question is, if you know that all signals are mixed with equal weight, then why not just sum the fixed-point values into a big long word? if you're doing this in C or C++, the type long long is, i believe, 64 bits. i cannot believe that your sum needs any more than that.

### Re: [music-dsp] Precision issues when mixing a large number of signals

On 12/10/12 11:18 AM, Bjorn Roche wrote: On Dec 10, 2012, at 4:41 AM, Alessandro Saccoia wrote: I don't think you have been clear about what you are trying to achieve. Are you trying to compute the sum of many signals for each time point? Or are you trying to compute the running sum of a

### Re: [music-dsp] Lerping Biquad coefficients to a flat response

looks like i came here late. someone tell me what it was about. admittedly, i didn't completely understand from a cursory reading. the only difference between the two BPFs in the cookbook is that of a constant gain factor. in one the peak of the BPF is always at zero dB. in the other,

### Re: [music-dsp] Lerping Biquad coefficients to a flat response

put in the parenths where you should, i think these are the same. r b0j -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 17:58 To: A discussion list for music-related DSP

### Re: [music-dsp] Lerping Biquad coefficients to a flat response

must be early onset alzheimer's. bestest, r b-j -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 18:25 To: A discussion list for music-related DSP Subject: Re: [music-dsp

### Re: [music-dsp] ITU 1770 RLB filter coefficients and biquad IIR filter

On 1/11/13 2:27 PM, Thomas Rehaag wrote: Hi Uli, have you solved the RLB filter problem in between? It looks as if the a coefficients have exactly the values they should have if they were not normalized to b0. Cheers, Thomas Am 26.04.2011 08:13, schrieb Uli Brueggemann: Hi, I'm wondering

### Re: [music-dsp] Calculating the gains for an XY-pad mixer

On 1/17/13 11:59 PM, Aengus Martin wrote: This may be a fairly idiosyncratic issue, but I think someone here might be able to comment on the correctness of what I've done. I am implementing a mixer in which the gains of four sounds are controlled using a single XY-pad. There is one sound

### Re: [music-dsp] Calculating the gains for an XY-pad mixer

On 1/18/13 8:20 AM, Wen Xue wrote: Somehow I feel it's the correlated case that deserves more attention. Things being uncorrelated simply means their correlation coefficients are zero; but things being correlated these can be anything from -1 to 1 but zero. You probably don't want to handle

### Re: [music-dsp] Calculating the gains for an XY-pad mixer

On 1/21/13 5:36 AM, Aengus Martin wrote: i don't think this has anything to do with barycentric coordinates, but i thought it might deal with your mixing gain issue: http://music.columbia.edu/pipermail/music-dsp/2010-December/069419.html for me, the issue was splicing more than mixing, but i

### Re: [music-dsp] 24dB/oct splitter

On 2/8/13 2:15 AM, Ross Bencina wrote: There are a at least two linear SVFs floating round now (the Hal Chamberlin one and Andy Simper's [1] ) [1] http://www.cytomic.com/files/dsp/SvfLinearTrapOptimised.pdf i've analyzed Hal's SVF to death, and i was exposted to Andy's design some time

### Re: [music-dsp] filter smoothly changeable from LP-BP-HP?

On 2/10/13 7:37 AM, Ross Bencina wrote: A Generalization of the Biquadratic Parametric Equalizer Christensen, Knud Bank AES 115 (October 2003) https://secure.aes.org/forum/pubs/conventions/?elib=12429 maybe i shouldn't say this, but someone here likely has a pdf copy of the paper in case it

### Re: [music-dsp] filter smoothly changeable from LP-BP-HP?

On 2/10/13 12:13 PM, Johannes Kroll wrote: On Sun, 10 Feb 2013 03:23:54 -0800 Bram de Jongbram.dej...@gmail.com wrote: does anyone know of a filter design that can smoothly be changed from LP to BP to HP with a parameter? IIRC LP/AP/HP could be done simply by perfect reconstruction LP/HP

### Re: [music-dsp] filter smoothly changeable from LP-BP-HP?

On 2/11/13 11:43 AM, Johannes Kroll wrote: On Mon, 11 Feb 2013 10:28:17 -0500 robert bristow-johnsonr...@audioimagination.com wrote: On 2/10/13 12:13 PM, Johannes Kroll wrote: On Sun, 10 Feb 2013 03:23:54 -0800 Bram de Jongbram.dej...@gmail.com wrote: does anyone know of a filter design

### Re: [music-dsp] filter smoothly changeable from LP-BP-HP?

On 2/11/13 2:47 PM, Johannes Kroll wrote: On Mon, 11 Feb 2013 12:52:00 -0500 robert bristow-johnsonr...@audioimagination.com wrote: On 2/11/13 11:43 AM, Johannes Kroll wrote: On Mon, 11 Feb 2013 10:28:17 -0500 robert bristow-johnsonr...@audioimagination.com wrote: On 2/10/13 12:13 PM,