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

2010-11-19 Thread robert bristow-johnson
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.

2010-11-19 Thread robert bristow-johnson
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.

2010-11-19 Thread robert bristow-johnson
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...

2010-11-20 Thread robert bristow-johnson
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

2010-11-26 Thread robert bristow-johnson
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

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

2010-11-30 Thread robert bristow-johnson
On Nov 29, 2010, at 8:50 PM, Element Green wrote: On Thu, Nov 25, 2010 at 9:33 PM, robert bristow-johnson r...@audioimagination.com wrote: depending on how big your window is, i think a better term for this is *cross-correlation* not autocorrelation. it's a single stream of audio so

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

2010-12-05 Thread robert bristow-johnson
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 publishing it (or submitting it for publication, probably to JAES),

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

2010-12-06 Thread robert bristow-johnson
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

2010-12-06 Thread robert bristow-johnson
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

2010-12-21 Thread robert bristow-johnson
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

2010-12-21 Thread robert bristow-johnson
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

2010-12-23 Thread robert bristow-johnson
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

2010-12-23 Thread robert bristow-johnson
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 to compute the FIR when the output same lands squarely on top of an input sample when all you

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

2010-12-23 Thread robert bristow-johnson
On Dec 23, 2010, at 2:09 PM, Nigel Redmon wrote: Somehow, we are talking about different things maybe? possibly. i'll admit that i'm trying to be a little-bit anal (or OCD) with the language. In what I'm talking about, the key is that n is not integer, not even when the output sample

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

2010-12-26 Thread robert bristow-johnson
() 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

2010-12-27 Thread robert bristow-johnson
On Dec 27, 2010, at 2:05 AM, Nigel Redmon wrote: Hi Robert, hi Nigel, No need for me to address point by point, because I agree with everything you say, except for one major point (which affects a few things you said)... You seem to imply that a windowed-sinc created for 2x

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

2010-12-27 Thread robert bristow-johnson
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

2010-12-28 Thread robert bristow-johnson
: 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

2010-12-28 Thread robert bristow-johnson
so was (or is) it Line6 or was it someone else? (this is my excuse for responding when i said you could have the last word.) i sayed: was is not the same as is. On Dec 28, 2010, at 6:54 PM, Nigel Redmon wrote: It was designed in. It is in products that I could go buy at guitar

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

2010-12-28 Thread robert bristow-johnson
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

2010-12-30 Thread robert bristow-johnson
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

2011-01-04 Thread robert bristow-johnson
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

2011-01-04 Thread robert bristow-johnson
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

2011-01-04 Thread robert bristow-johnson
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

2011-01-11 Thread robert bristow-johnson
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

2011-01-11 Thread robert bristow-johnson
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

2011-01-25 Thread robert bristow-johnson
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

2011-01-28 Thread robert bristow-johnson
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] damn patents (was New patent application on uniformly partitioned convolution) [OT]

2011-02-07 Thread robert bristow-johnson
On Feb 7, 2011, at 6:54 PM, Tom Wiltshire wrote: On 7 Feb 2011, at 20:54, Andy Farnell wrote: Do a search on Yamaha Patent FM. Does that look like a widespread interpretation that is clear and unambiguous to you? My argument is simple at this point. Development was stifled. This is an

Re: [music-dsp] convolution in the frequency domain

2011-02-20 Thread robert bristow-johnson
On Feb 20, 2011, at 5:37 PM, Thomas Rehaag wrote: 1. oversample 2 times 2. multiply 3. downsample 2 times Wow, why didn't I think of this myselfe? The convolution would be much easier / better for me but you already saved me with this suggestion. I thought it goes like: convolution

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

2011-03-12 Thread robert bristow-johnson
On Mar 13, 2011, at 12:09 AM, Ross Bencina wrote: Andy Farnell wrote: How do you know these filters don't have a resonance? That could explain your results. I doubt those filters would have explicit resonance/peaking at the cutoff (it is a lowpass EQ after all). But assuming they are

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

2011-03-13 Thread robert bristow-johnson
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

2011-03-17 Thread robert bristow-johnson
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

2011-03-17 Thread robert bristow-johnson
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

2011-03-17 Thread robert bristow-johnson
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] Frequency-dependent latency of a filter

2011-03-17 Thread robert bristow-johnson
On Mar 17, 2011, at 8:05 PM, Andreas Beisler wrote: Hi. Sorry, I messed up the subject of the thread. that's whacha get fer using the digest form. that'll teach ya! -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the

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

2011-03-18 Thread robert bristow-johnson
On Mar 18, 2011, at 1:55 PM, Wen X wrote: - when considering finite duration there is the uncertainty principle, so you always deal with a pack of frequencies rather than one frequency, which makes latency dependent on the content of that pack. - however, using FT[th(t)]=j(FT[h(t)])', one

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

2011-04-27 Thread robert bristow-johnson
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 integration methodsappliedtomusical resonant filters

2011-05-17 Thread robert bristow-johnson
On May 17, 2011, at 5:09 AM, Vadim Zavalishin wrote: You mean this one? Analyzing the Moog VCF with Considerations for Digital Implementation by Tim Stilson, Julius Smith http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.50.3093 No, there was another one dealing with zero delay

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

2011-05-17 Thread robert bristow-johnson
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

2011-05-21 Thread robert bristow-johnson
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

2011-05-21 Thread robert bristow-johnson
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

2011-05-23 Thread robert bristow-johnson
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

2011-06-26 Thread robert bristow-johnson
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.

2011-07-09 Thread robert bristow-johnson
hi Olli (and others)... i was reviewing this thread because i wanted to read what Stefan Stenzel had said and realized that you had posted this response, and i don't think i or anyone had responded to it. i don't remember reading it (it must be the cannabis). i hope you're listening

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

2011-07-14 Thread robert bristow-johnson
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.

2011-07-14 Thread robert bristow-johnson
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.

2011-07-15 Thread robert bristow-johnson
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

2011-07-29 Thread robert bristow-johnson
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

2011-07-29 Thread robert bristow-johnson
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

2011-08-02 Thread robert bristow-johnson
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

2011-08-02 Thread robert bristow-johnson
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

2011-08-02 Thread robert bristow-johnson
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

2011-08-05 Thread robert bristow-johnson
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

2011-08-24 Thread robert bristow-johnson
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

2011-08-27 Thread robert bristow-johnson
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

2011-09-12 Thread robert bristow-johnson
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

2011-09-12 Thread robert bristow-johnson
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

2011-11-02 Thread robert bristow-johnson
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

2011-11-27 Thread robert bristow-johnson
On 11/27/11 12:23 PM, Dominique Würtz wrote: Hi all, I recently got interested in the approach from [1] to design of digital EQs. The main idea here is to introduce a new degree of freedom G1 in the prewarped analog prototype Hp(s) where G1 is the filter transfer gain at Nyquist frequency which

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

2011-11-27 Thread robert bristow-johnson
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

2011-12-08 Thread robert bristow-johnson
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

2011-12-08 Thread robert bristow-johnson
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?

2012-01-08 Thread robert bristow-johnson
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?

2012-01-09 Thread robert bristow-johnson
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?

2012-01-09 Thread robert bristow-johnson
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?

2012-01-10 Thread robert bristow-johnson
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?

2012-01-10 Thread robert bristow-johnson
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?

2012-01-11 Thread robert bristow-johnson
On 1/11/12 10:50 AM, Thomas Young wrote: Man I wish I hadn't gone to that wiki page now, it really is a mess and there are some pretty glaring errors (missing brackets on the summation in the Fourier series equation, and citation needed... wtf?) -Original Message- From:

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

2012-01-12 Thread robert bristow-johnson
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?

2012-01-12 Thread robert bristow-johnson
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?

2012-01-16 Thread robert bristow-johnson
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

2012-02-06 Thread robert bristow-johnson
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

2012-02-07 Thread robert bristow-johnson
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

2012-02-21 Thread robert bristow-johnson
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:

Re: [music-dsp] google's non-sine

2012-02-22 Thread robert bristow-johnson
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

2012-02-25 Thread robert bristow-johnson
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

2012-02-26 Thread robert bristow-johnson
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

2012-03-24 Thread robert bristow-johnson
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

2012-04-09 Thread robert bristow-johnson
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

2012-04-09 Thread robert bristow-johnson
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 1,2,3...

2012-04-09 Thread robert bristow-johnson
[this is a fresh message, not a reply to any other, since none of those seem to get past my SMTP server.] 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 julian_schm...@chipmusik.de wrote: I really

[music-dsp] testing different subject header

2012-04-09 Thread robert bristow-johnson
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

2012-04-10 Thread robert bristow-johnson
another day, another restarted computer, let's see if i can post to this thread (that was the weirdest of problems). here's hoping my SMTP server doesn't reject this... it *did* reject it. Thunderbird says: Alert. An error occurred while sending mail. The mail server responded: 5.7.1

Re: [music-dsp] OSC problem on STM32F4Discover

2012-04-10 Thread robert bristow-johnson
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

Re: [music-dsp] OSC problem on STM32F4Discover

2012-04-10 Thread robert bristow-johnson
On 4/10/12 1:18 PM, Nigel Redmon wrote: Excellent point, Robert. (Another way to avoid it is to mask the index with 0xff, if you want to keep the table 256, but not check for wrap in the mid-interpolation.) sure, but that masking is otherwise unnecessary (the initial index will always be 0

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

2012-04-12 Thread robert bristow-johnson
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.

2012-04-12 Thread robert bristow-johnson
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] WOLA and the phase vocoder

2012-04-20 Thread robert bristow-johnson
i wish i could program my Thunderbird client to *not* copy HTML formatting (like this font change) from the quoted post in replying. and i wish that music-dsp's Majordomo or whoever would not copy it either. that said... everythingXue says is congruent to my experience (which is no longer

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

2012-05-04 Thread robert bristow-johnson
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

2012-05-07 Thread robert bristow-johnson
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?

2012-06-08 Thread robert bristow-johnson
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?

2012-06-13 Thread robert bristow-johnson
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

2012-07-04 Thread robert bristow-johnson
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 a point in the implementation of noise gates. I

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

2012-07-05 Thread robert bristow-johnson
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

2012-07-23 Thread robert bristow-johnson
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 :)

2012-07-30 Thread robert bristow-johnson
prerecorded material, i.e. material sourced from different equipment, is streamed). A compromise would be a finite-but-very-long moving average (Randy Yates, Richard Lyons, Robert Bristow Johnson) but this one in turn introduces a very long delay (obviously not quite ideal for a live true peak meter

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

2012-07-31 Thread robert bristow-johnson
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 :)

2012-08-01 Thread robert bristow-johnson
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

2012-08-10 Thread robert bristow-johnson
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

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