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

2010-11-25 Thread Ross Bencina
Element Green wrote: I'm the author of a SoundFont instrument editing application called Swami (http://swami.sourceforge.net). A while back an interested developer added a loop finding algorithm which I integrated into the application. This feature is supposed to generate a list of start/end

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

2010-12-21 Thread Ross Bencina
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 Kaiser window) sinc in most cases. this is where you

Re: [music-dsp] resonance

2011-01-01 Thread Ross Bencina
Alan Wolfe wrote: I have a future retro revolution (303 clone) and one of the knobs it has is resonance. Does anyone know what resonance is in that context or how it's implemented? I was reading some online and it seems like it might be some kind of feedback but I'm not really sure... In

[music-dsp] New Android audio developers mailing list

2011-01-13 Thread Ross Bencina
Department for hosting the list. Ross Bencina Andraudio list admin -- 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

Re: [music-dsp] New Android audio developers mailing list

2011-01-14 Thread Ross Bencina
Dan Stowell wrote: As Oskari noted, here's hoping the angle of this new list (the low-level aspects you mention) can help speed Android towards good low-latency i/o! Yeah, well that's the main thing that got us together so I hope so. Some have already started looking at bypassing audioflinger

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

2011-01-28 Thread Ross Bencina
I oppose patent trolls and trivial patents. Beyond that I think it's a bit more murky. My a basic rule of thumb would be: If I can think of a mathematical or algorithmic solution to some random problem in my field in less than a month I don't expect that solution to be patented or patentable.

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

2011-01-29 Thread Ross Bencina
Hi Andy Andy Farnell wrote: I don't want to open up a lengthy OT debate here. But will reply privately to address some of your points in detail. Fair enough. I guess the main reason I bought into this conversation is that I do feel like it's something that affects all of us here and I'm

Re: [music-dsp] damn patents (was New patent application on uniformly partitioned convolution) [OT]

2011-01-29 Thread Ross Bencina
Hi Andy I wish I were worthy of quoting Blaise Pascal here, but instead I will just apologise for the rant... I think it has a bearing on all of us too. And thus you lure me in. But if people complain that this is getting boring, off-topic or ill-natured then let's quit it. (Subject

Re: [music-dsp] damn patents (was New patent application on uniformly partitioned convolution) [OT]

2011-01-31 Thread Ross Bencina
Hi Andy Andy Farnell wrote: AXIOM: Ideas should not be patentable. Period. Do I need to explain this? Sorry, you've lost me a bit here. Pehaps you do need to explain it.. see if I'm twisting your words below or if you find that I'm addressing your position (of course I don't expect you to

Re: [music-dsp] damn patents (was New patent application on uniformly partitioned convolution) [OT]

2011-01-31 Thread Ross Bencina
Hi Andy Are you suggesting by stating the above axiom that algorithms are _simply_ ideas and that for this reason alone they shouldn't be patentable? Yes I am, you've got it. An algorithm is unsufficiently concrete to deserve a patent, it is an abstraction, a generalisation. Ok... An

Re: [music-dsp] looking for a flexible synthesis system technically and legally appropriate for iOS development

2011-02-07 Thread Ross Bencina
Morgan wrote: simply plugging unit generators in to one another, not having to stop and think about how to, for example, go from a mono oscillator signal to a stereo reverb signal. I'd like to be able to work more like I work in SuperCollider, writing higher-level code to create a signal path,

Re: [music-dsp] good introductory microcontroller platform for audiotasks?

2011-04-19 Thread Ross Bencina
Kevin Dixon wrote: My EE friend is recommending I go PIC, but the Arduino looks promising, especially for fast return on effort :) I guess startup cost is an issue too, I'd like to be up and running for about 50 USD. Any thoughts/recommendations? Thanks, I wouldn't usually use microcontroller

[music-dsp] Fw: First SuperCollider Book sample chapter now available, 'Inside scsynth'

2011-04-22 Thread Ross Bencina
weeks, freely available. The first is actually a developer chapter, the last in the book, on the internals of scsynth, by Ross Bencina. It gives a good idea of the layout of the book for those who might not have seen inside yet, and will be particularly useful to devs exploring how SuperCollider

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

2011-05-17 Thread Ross Bencina
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 the relationship between the z-plane transfer function and

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

2011-05-20 Thread Ross Bencina
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, I was able to follow your analysis below. Previously I didn't really

Re: [music-dsp] Trapezoidal and other integration methods applied tomusical resonant filters

2011-05-20 Thread Ross Bencina
On Thursday, May 19, 2011 12:20 PM Andrew Simper wrote: I think there has been some confusion with the different responses to different questions in this thread, so sorry if I have got this wrong, but I believe Ross posted the block diagram of Hal Chamberlains filter, which, as you have shown,

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

2011-05-23 Thread Ross Bencina
On Sunday, May 22, 2011 1:27 PM robert bristow-johnson wrote: what i believe i missed is an assumption that the Q is pretty high. then all the formulae from Hal's book fall into place. does Hal explicitly make that assumption in the book? Not that I could find. I just re-read the section

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

2011-12-08 Thread Ross Bencina
Is that not like saying It is ok to use an illegal copy of software [x] because it is so expensive they cannot expect a lot of people to buy it? Or do you find this situation to be different? It might be like saying State funded research should be available free of charge to the scientific

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

2012-01-11 Thread Ross Bencina
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 the article, then the tendentious editor

Re: [music-dsp] Signal processing and dbFS

2012-01-17 Thread Ross Bencina
Hi Linda Some (possibly spurious) thoughts... I'm confused about what you're actually trying to achieve by referencing things relative to 0dBFS (which is measure of signal level relative to digital full scale). You talk about frequency responses below, which are expressed in terms of

Re: [music-dsp] music-dsp Digest, Vol 97, Issue 21

2012-01-18 Thread Ross Bencina
On 19/01/2012 9:03 AM, Linda Seltzer wrote: Why and under what circumstances is it advantageous to set up the Y axis as dbFS rather than dbV, dbSPL, out of the ones you mention, dBFS is the only one that has any meaning in the digital domain -- since as we've established, the others don't

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

2012-02-06 Thread Ross Bencina
On 7/02/2012 5:31 PM, Shashank Kumar (shanxS) wrote: But for audio applications, like EQ design, isn't the phase response supposed to be linear ? I would say not. Most audio EQ filters don't have linear phase. Additionally, the ear is very sensitive to acausality so, especially for

[music-dsp] stereo-wide pan law?

2012-02-07 Thread Ross Bencina
Hi Everyone, Does anyone know if there's a standard way to calculate pan laws for stereo-wide panning ? By stereo-wide I mean panning something beyond the speakers by using 180-degree shifted signal in the opposite speaker. For example, for beyond hard left you would output full gain signal

Re: [music-dsp] stereo-wide pan law?

2012-02-07 Thread Ross Bencina
:20 PM, Ross Bencina wrote: Hi Everyone, Does anyone know if there's a standard way to calculate pan laws for stereo-wide panning ? By stereo-wide I mean panning something beyond the speakers by using 180-degree shifted signal in the opposite speaker. For example, for beyond hard left you would

Re: [music-dsp] stereo-wide pan law?

2012-02-08 Thread Ross Bencina
On 9/02/2012 1:06 AM, Olli Niemitalo wrote: Now, it would be unreasonable if, compared to input, the output would have an opposite polarity in L or R. I'm not sure what you're getting at here, for example, the following is reasonable: Considering the left channel only (right is opposite

Re: [music-dsp] stereo-wide pan law?

2012-02-08 Thread Ross Bencina
^) This panning law agrees exactly with the panning described by HRTF methods at the low frequency limit (and only there). Jerry On Feb 7, 2012, at 11:10 PM, Ross Bencina wrote: Thanks for the responses, Seems like I may have asked the wrong question. Ralph Glasgal wrote: There is no valid

Re: [music-dsp] stereo-wide pan law?

2012-02-09 Thread Ross Bencina
On 9/02/2012 11:02 AM, Jerry wrote: (Good grief, people.) You want the *very famous* Bauer's Law of Sines: Benjamin B. Bauer, Phasor Analysis of Some Stereophonic Phenomena, IRE Transactions on Audio, January-February, 1962. This panning law is mentioned in many introductory books on stereo

Re: [music-dsp] stereo-wide pan law?

2012-02-11 Thread Ross Bencina
On 11/02/2012 2:27 PM, Jerry wrote: Glad to help. With your set-up, if you try to put a loud low frequency signal well outside the loudspeaker array, you will notice that your speakers and/or amplifiers will have melted. To the extent that sin(theta_A) = theta_A (small-angle approximation),

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

2012-02-23 Thread Ross Bencina
On 23/02/2012 6:22 PM, Oskari Tammelin wrote: Come on, it's a perfect visualization of their understanding of audio. +1 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links

Re: [music-dsp] a little about myself

2012-02-24 Thread Ross Bencina
Hi Brad, On 24/02/2012 3:01 PM, Brad Garton wrote: Joining this conversation a little late, but what the heck... Me too... On Feb 22, 2012, at 9:18 AM, Michael Gogins wrote: I got my start in computer music in 1986 or 1987 at the woof group at Columbia University using cmix on a Sun

Re: [music-dsp] a little about myself

2012-02-24 Thread Ross Bencina
Hi Charles, On 24/02/2012 10:45 PM, Charles Turner wrote: Anything else is just plugging unit generators together, which is limiting in many situations Has it escaped me that Audio Mulch supports this kind of interpretation? Hi Charles, I'm not exactly sure what you think has escaped

Re: [music-dsp] a little about myself

2012-02-24 Thread Ross Bencina
On 25/02/2012 4:50 AM, Adam Puckett wrote: Is there a minimal example of a complete working program that renders a sine wave in realtime using Kernel Streaming that will compile with just a bare MinGW install? (I have the latest GCC 4.5 on Windows XP Service Pack 3). Which DirectX SDK do I

Re: [music-dsp] a little about myself

2012-02-25 Thread Ross Bencina
Hi Andy, On 25/02/2012 5:05 AM, Andy Farnell wrote: The problem with plug unit generators languages for me is that they privilege the process (network of unit generators) over the content Some really interesting thoughts here Ross. At what level of granularity does the trade-off of control,

Re: [music-dsp] a little about myself

2012-02-25 Thread Ross Bencina
Hi Michael, I agree with you on the below within the dominant paradigm, but there are a few problematic assumptions that are central to this dicussion. What follows is a somewhat controversial rant, apologies in advance... On 25/02/2012 6:21 AM, Michael Gogins wrote: The two languages

Re: [music-dsp] a little about myself

2012-02-25 Thread Ross Bencina
On 25/02/2012 2:38 PM, Adam Puckett wrote: What is WaveRT? I don't see it in the tarball. WaveRT is a recent WDM-KS driver sub-model that was introduced in Windows Vista. It is the version of WDM-KS that people seem to get excited about as offering the lowest latency and efficiency. I can't

Re: [music-dsp] a little about myself

2012-02-25 Thread Ross Bencina
/wiki/Linguistic_relativity (see Present status section). Here's a short excerpt from a discussion on the POTAC list last year. I really liked Dan Stowell's introduction of the idea of long-term bias effects [1]: Ross Bencina wrote: In any case I am not reverting to strong-SW here.. just

Re: [music-dsp] a little about myself

2012-02-26 Thread Ross Bencina
On 26/02/2012 9:13 PM, Richard Dobson wrote: On 26/02/2012 05:48, Ross Bencina wrote: ... make this significantly easier to do. If someone wants to spend years rewriting something that already works quite well, like Csound. Depends if you think CSound works quite well. I think it is deeply

Re: [music-dsp] a little about myself

2012-02-26 Thread Ross Bencina
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 to do proper conditionals it will be really powerful. Why anyone would want to use a visual

Re: [music-dsp] a little about myself

2012-02-26 Thread Ross Bencina
On 27/02/2012 1:22 AM, Brad Garton wrote: I would like to agree with you, because I also value all these things (and am pretty much a dilettante in all four). But I see an analog with the is a DJ*really* a [computer music] composer? question that floats around (or in an earlier generation, is

Re: [music-dsp] a little about myself

2012-02-27 Thread Ross Bencina
Hi Andy, Some comments, and questions for clarification... IIRC, most Music-N line of systems are multi-rate. That means we have a fast computation rate, on which audio signals are calculated, and a slower rate (obviously some integer factor of the audio rate), usually called the control

Re: [music-dsp] a little about myself

2012-02-27 Thread Ross Bencina
Hi Richard, On 27/02/2012 3:01 AM, Richard Dobson wrote: On 26/02/2012 11:33, Ross Bencina wrote: .. Perhaps I'm not being clear. My point is about being able to execute arbitrary code at an arbitrary time based on the value of some signal(s). The zero crossing thing was a simple example

Re: [music-dsp] a little about myself

2012-02-27 Thread Ross Bencina
On 27/02/2012 3:58 AM, Andy Farnell wrote: For my sonification work for LHCsound, I used Perl to parse data files and generate Csound scores, simply because it is a task Perl is canonically optimised to do and scripts can be run up very quickly. Just a quick +1 for Perl as an event

Re: [music-dsp] a little about myself

2012-02-27 Thread Ross Bencina
On 27/02/2012 4:11 AM, Richard Dobson wrote: On 26/02/2012 05:48, Ross Bencina wrote: .. (1) An efficient way to implement a runtime compiler for block-based audio DSP chains (these are what are in software synthesizers)... which usually are based on As if block-based DSP chains are the only

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

2012-02-27 Thread Ross Bencina
On 27/02/2012 10:58 AM, robert bristow-johnson wrote: in my opinion, the optimal division of labor becomes obvious if your system modularizes specific low-level algs. I'm not sure it's what you meant, but I would say: if your system modularizes specific low-level algs there is perhaps an

Re: [music-dsp] a little about myself

2012-02-27 Thread Ross Bencina
On 28/02/2012 8:55 AM, Richard Dobson wrote: On 27/02/2012 21:00, Ross Bencina wrote: .. And as I have already said. Computer musicians, must be programmers *by definition*. Otherwise they are musicians, using computers. But there is no single universally agreed definition of Computer

Re: [music-dsp] a little about myself

2012-02-28 Thread Ross Bencina
On 29/02/2012 11:41 AM, Richard Dobson wrote: On 28/02/2012 16:03, Bill Schottstaedt wrote: I don't think this conversation is useful. The only question I'd ask is did this person make good music?, and I don't care at all about his degrees or grants. One of the best mathematicians I've known

Re: [music-dsp] very cheap synthesis techniques

2012-02-28 Thread Ross Bencina
On 29/02/2012 8:00 AM, douglas repetto wrote: Oh, come on, transistors are for babies. Real composers roll their own diodes! http://hackaday.com/2010/03/05/diy-diodes Etching your own transistors is still pretty cool: http://www.youtube.com/watch?v=w_znRopGtbE Might take a while to make

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

2012-06-14 Thread Ross Bencina
On 14/06/2012 5:29 PM, Andy Farnell wrote: Maybe this isn't the same hazard of dimensionality that Dan warns us of... I'm saying the space is definitely_warped_ with big areas of nothingness between apparently close and similar points. Does that make sense? Seems to me there are at least three

Re: [music-dsp] recommendation for VST host for dev. modifications

2012-06-29 Thread Ross Bencina
On 28/06/2012 7:05 AM, Michael Gogins wrote: Sorry, I confused this discussion with a similar one I was trying to get going on the Csound list, where I do want to get rid of the VST SDK while still being able to redistribute Csound source code from SourceForge. I have no idea how you propose

Re: [music-dsp] stuck with filter design

2012-11-17 Thread Ross Bencina
Hello Shashank, I'm interested in this stuff too, but I'm no expert. I've tried to give some pointers below. Hopefully someone else will correct me if I've made an error: On 17/11/2012 8:24 PM, Shashank Kumar (shanxS) wrote: I am a self taught Linux fanatic who is trying to teach himself

Re: [music-dsp] stuck with filter design

2012-11-18 Thread Ross Bencina
On 19/11/2012 9:24 AM, Bjorn Roche wrote: (Shashank wrote:) I have one more question: Why so many people use analog prototypes to get a digital filter ? Why not just put a few constraints on location of poles/zeros on Z plane and get done with it ? This is a really great question. Indeed.

Re: [music-dsp] stuck with filter design

2012-11-21 Thread Ross Bencina
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 of the BLT and music-dsp:

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

2012-12-09 Thread Ross Bencina
Hi Alessandro, A lot has been written about this. Google precision of summing floating point values and read the .pdfs on the first page for some analysis. Follow the citations for more. Somewhere there is a paper that analyses the performance of different methods and suggests the optimal

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

2012-12-09 Thread Ross Bencina
Hello Alessandro, On 10/12/2012 12:18 PM, Alessandro Saccoia wrote: In my original question, I was thinkin of mixing signals of arbitrary sizes. 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

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

2012-12-09 Thread Ross Bencina
On 10/12/2012 1:47 PM, Bjorn Roche wrote: There is something called double double which is a software 128 bit floating point type that maybe isn't too expensive. long double, I believe No. long double afaik usually means extended precision, as supported in hardware by the x86 FPU, and is 80

[music-dsp] mechanisms that transfer of energy between modes in acoustic systems?

2012-12-16 Thread Ross Bencina
Hi Everyone, I have a question which in a broad sense relates to physical modelling and acoustics: Under what circumstances does a resonating (acoustic) system move energy from one frequency to another? One gross example I can think of would be snares on a snare drum. But aside from

Re: [music-dsp] mechanisms that transfer of energy between modes in acoustic systems?

2012-12-16 Thread Ross Bencina
Thanks Risto, On 17/12/2012 5:05 AM, Risto Holopainen wrote: (*) Legge and Fletcher: Nonlinearity, chaos, and the sound of shallow gongs. JASA 86(6), 1989. is here: http://www.ausgo.unsw.edu.au/music/people/publications/Leggeetal1989.pdf Also at the same site: Fletcher, N. H. 1994.

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

2013-01-03 Thread Ross Bencina
On 4/01/2013 4:05 AM, Thomas Young wrote: Is there a way to modify the bandpass coefficient equations in the cookbook (the one from the analogue prototype H(s) = s / (s^2 + s/Q + 1)) such that the gain of the stopband may be specified? I want to be able I'm pretty sure that the BLT bandpass

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

2013-01-03 Thread Ross Bencina
Hi Thomas, Replying to both of your messages at once... On 4/01/2013 4:34 AM, Thomas Young wrote: However I was hoping to avoid scaling the output since if I have to do that then I might as well just change the wet/dry mix with the original signal for essentially the same effect and less

Re: [music-dsp] Starting From The Ground Up

2013-01-21 Thread Ross Bencina
Hello Jeff, Before I attempt an answer, can I ask: what programming languages do you know (if any) and how proficient are you at programming? Ross. On 21/01/2013 9:49 PM, Jeffrey Small wrote: Hello, I'm a recently new computer programmer that is interested in getting into the world of

Re: [music-dsp] Starting From The Ground Up

2013-01-21 Thread Ross Bencina
Hi Jeff, At your stage of learning with C the advice to just write some code seems most pertinent, but I guess it depends on your learning style. Coming up with an achievable project and seeing it through to completion is a good way to learn programming. Read lots of code applies, and is

Re: [music-dsp] Starting From The Ground Up

2013-01-21 Thread Ross Bencina
On 21/01/2013 9:49 PM, Jeffrey Small wrote: I'm a recently new computer programmer that is interested in getting into the world of Audio Plug Ins. I have a degree in Recording/Music, as well as a degree in Applied Mathematics. How would you recommend that I start learning how to program for

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

2013-02-07 Thread Ross Bencina
Hi Russell, So to be clear, you're creating a Linkwitz-Riley crossover? http://en.wikipedia.org/wiki/Linkwitz%E2%80%93Riley_filter On 8/02/2013 6:05 PM, Russell Borogove wrote: I have two digital 12dB/octave state-variable filters, each with lowpass/highpass/bandpass/notch outputs; I'd like

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

2013-02-10 Thread Ross Bencina
Hi Bram, A Generalization of the Biquadratic Parametric Equalizer Christensen, Knud Bank AES 115 (October 2003) https://secure.aes.org/forum/pubs/conventions/?elib=12429 Defines equations with a symmetry parameter for smoothly moving between the states you mention. There are graphs so you can

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

2013-02-10 Thread Ross Bencina
On 11/02/2013 1:37 AM, robert bristow-johnson wrote: maybe i shouldn't say this, but someone here likely has a pdf copy of the paper in case it breaks your bank to buy it from AES. Unfortunately not me. I lost the pdf in a data loss incident and only have a printout and don't have an AES

Re: [music-dsp] TR : Production Music Mood Annotation Survey 2013

2013-02-19 Thread Ross Bencina
Can someone please explain the scientific basis for this kind of study? Surely by now it is widely accepted that correlations between music and mood and emotion are culturally biased and socially acquired? Does the study below control for cultural bias? Please explain why an otherwise

Re: [music-dsp] Sound effects and Auditory illusions

2013-02-20 Thread Ross Bencina
Hi Marcelo, Just came accross this, maybe it is helpful: Rorschach Audio – Art Illusion for Sound On The Art http://rorschachaudio.wordpress.com/about/ Ross. On 19/02/2013 9:26 PM, Marcelo Caetano wrote: Dear list, I'll teach a couple of introductory lectures on audio and music

Re: [music-dsp] RE : TR : Production Music Mood Annotation Survey 2013

2013-02-21 Thread Ross Bencina
Tel: +44 (0)20 7882 7986 - Fax: +44 (0)20 7882 7997 E-mail: mathieu.bart...@eecs.qmul.ac.uk http://www.elec.qmul.ac.uk/digitalmusic/ De : music-dsp-boun...@music.columbia.edu [music-dsp-boun...@music.columbia.edu] de la part de Ross Bencina [rossb-li

Re: [music-dsp] M4 Music Mood Recommendation Survey

2013-02-21 Thread Ross Bencina
On 22/02/2013 9:54 AM, Richard Dobson wrote: Listen to each track at least once and then select which track is the best match with the seed. If you think that none of them match, just select an answer at random. Now I am no statistician, but with only four possible answers offered per test,

Re: [music-dsp] M4 Music Mood Recommendation Survey

2013-02-24 Thread Ross Bencina
, lightbulb o schoolbus? Uh, lightbulb? No! Lo siento, Schoolbus es mas macho que lightbulb. Best to you, Ross. best @ all Andy On Fri, Feb 22, 2013 at 10:19:02AM +1100, Ross Bencina wrote: On 22/02/2013 9:54 AM, Richard Dobson wrote: Listen to each track at least once and then select which

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-07 Thread Ross Bencina
Stephen, On 8/03/2013 9:29 AM, ChordWizard Software wrote: a) additive mixing of audio buffers b) clearing to zero before additive processing You could also consider writing (rather than adding) the first signal to the buffer. That way you don't have to zero it first. It requires having a

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-08 Thread Ross Bencina
On 9/03/2013 9:53 AM, ChordWizard Software wrote: Maybe you can advise me on a related question - what's the best approach to implementing attenuation? I'm guessing it is not linear, since perceived sound loudness has a logarithmic profile - or am I confusing amplifier wattage with signal

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-08 Thread Ross Bencina
On 9/03/2013 2:55 PM, Ross Bencina wrote: Note that audio faders are not linear in decibels either, e.g.: http://iub.edu/~emusic/etext/studio/studio_images/mixer9.jpg There is some discussion here: http://www.kvraudio.com/forum/viewtopic.php?t=348751 Ross. -- dupswapdrop -- the music-dsp

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-09 Thread Ross Bencina
On 10/03/2013 7:01 AM, Tim Goetze wrote: [robert bristow-johnson] On 3/9/13 1:31 PM, Wen Xue wrote: I think one can trust the compiler to handle a/3.14 as a multiplication. If it doesn't it'd probably be worse to write a*(1/3.14), for this would be a division AND a multiplication. there are

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-14 Thread Ross Bencina
On 15/03/2013 6:02 AM, jpff wrote: Ross == Ross Bencinarossb-li...@audiomulch.com writes: Ross I am suspicious about whether the mask is fast than the conditional for Ross a couple of reasons: Ross - branch prediction works well if the branch usually falls one way Ross - cmove

Re: [music-dsp] Efficiency of clear/copy/offset buffers

2013-03-14 Thread Ross Bencina
On 15/03/2013 7:27 AM, Sampo Syreeni wrote: Quite a number of processors have/used to have explicit support for counted for loops. Has anybody tried masking against doing the inner loop as a buffer-sized counted for and only worrying about the wrap-around in an outer, second loop, the way we do

Re: [music-dsp] Synth thread timing strategies

2013-03-26 Thread Ross Bencina
On 26/03/2013 4:55 PM, Alan Wolfe wrote: I just wanted to chime in real quick to say that unless you need to go multithreaded for some reason, you are far better off doing things single threaded. Introducing more threads does give you more processing power, but the communication between threads

Re: [music-dsp] Synth thread timing strategies

2013-03-26 Thread Ross Bencina
On 26/03/2013 5:28 PM, ChordWizard Software wrote: Hi Ross, Thanks, couple more questions then: - There can be significant jitter in the time at which an audio callback is called. Can you define jitter? Callbacks with different frame counts, or dropped frames? If you call

Re: [music-dsp] note onset detection

2013-08-06 Thread Ross Bencina
On 7/08/2013 2:38 AM, Theo Verelst wrote: I suppose in EE terms, if you know something about the waves you're trying to detect Strikes me that we are talking about perceptual note onset, not something you could define /easily/ in EE terms. You would need a definition of note onset that

Re: [music-dsp] note onset detection

2013-08-07 Thread Ross Bencina
On 7/08/2013 12:23 PM, charles morrow wrote: Please explain your reference Roberts transcription notes for me. Robert expressed the following requirement: On 6/08/2013 6:01 AM, robert bristow-johnson wrote: the big problem i am dealing with is people singing or humming and changing notes.

Re: [music-dsp] note onset detection

2013-08-09 Thread Ross Bencina
Hi Robert, I have a question: are you trying to output the pitch and note on/off information in a real-time streaming scenario with minimum delay? or is this an off-line process? My impression is that the MIR folk worry less about minimum-delay/causal processing than us real-time people.

Re: [music-dsp] IIR Coefficient Switching Glitches

2013-11-03 Thread Ross Bencina
On 3/11/2013 3:22 PM, Laurent de Soras wrote: Chris Townsend wrote: Any ideas? Recommendations? Probably this: http://cytomic.com/files/dsp/SvfLinearTrapOptimised.pdf Consider ramping interpolated coefficients at audio rate to smooth out parameter changes. I'm pretty sure that Andy's

Re: [music-dsp] Trapezoidal integrated optimised SVF v2

2013-11-07 Thread Ross Bencina
On 8/11/2013 4:29 AM, Theo Verelst wrote: Fine. He insulted run of the mill academic EE insights from decades ago, i merely stated facts, which should be respected, but here are still not. The theory is quite right, and I've taken the effort of correcting a lot of misinterpretations. I suppose

Re: [music-dsp] Time Varying BIBO Stability Analysis of Trapezoidal integrated optimised SVF v2

2013-11-10 Thread Ross Bencina
that the lower bound on k can approaches zero as the 2,2 entry approaches zero from below. Hopefully I'm not imagining things. Ross. On 11/11/2013 2:58 AM, Ross Bencina wrote: Hi Everyone, I took a stab at converting Andrew's SVF derivation [1] to a state space representation and followed

Re: [music-dsp] R: R: Trapezoidal integrated optimised SVF v2

2013-11-10 Thread Ross Bencina
Hi Ezra, A few comments: On 11/11/2013 3:19 PM, Ezra Buchla wrote: there seems to be some concern about distortion introduced by the trapezoidal integration. i've tried the algo in both fixed 32 ands float, and it seems to sound and look ok to but i have not done a proper analysis either

Re: [music-dsp] R: R: Trapezoidal integrated optimised SVF v2

2013-11-10 Thread Ross Bencina
On 11/11/2013 12:21 PM, robert bristow-johnson wrote: but you cannot define your current output sample in terms of the current output sample. But that, with all due respect, is what has been done for quite a while. it's been reported or *reputed* to be done for quite a while. but when the

Re: [music-dsp] Time Varying BIBO Stability Analysis of Trapezoidal integrated optimised SVF v2

2013-11-11 Thread Ross Bencina
Hi Vadim, Thanks for your feedback... On 11/11/2013 9:52 PM, Vadim Zavalishin wrote: [snip on the analog stuff] For the discrete-time case the situation is more complicated, because we can't use the continuity of the state vector function. IIRC, I also didn't manage to build the worst-case

Re: [music-dsp] Trapezoidal integrated optimised SVF v2

2013-11-12 Thread Ross Bencina
On 12/11/2013 7:40 PM, Tim Blechmann wrote: some real-world benchmarks from the csound community imply a performance difference of roughly 10% [1]. Csound doesn't have a facility for running multiple filters in parallel though does it? not even 2 in parallel for stereo. 4 biquads in

Re: [music-dsp] Implicit integration is an important term, ZDF is not

2013-11-14 Thread Ross Bencina
response which may be what matters most in audio DSP. Max On 14 November 2013 14:06, Ross Bencina rossb-li...@audiomulch.com wrote: On 14/11/2013 11:41 PM, Max Little wrote: I may have misread, but the discussion seems to suggest that this discipline is just discovering implicit finite differencing

Re: [music-dsp] Are natural sounds of minimum phase?

2013-12-10 Thread Ross Bencina
On 11/12/2013 4:29 PM, Sol Friedman wrote: minimum phase would be a likely candidate Is minimum-phase a well defined property of non-linear time-varying systems? Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book

Re: [music-dsp] Hosting playback module for samples

2014-02-26 Thread Ross Bencina
Hi Mark, I'm not really sure that I understand the problem. Can you be more specific about the problems that you're facing? Personally I would avoid managed code for anything real-time (ducks). You're need to build a simple audio engine (consider PortAudio or the ASIO SDK). And write some

Re: [music-dsp] Hosting playback module for samples

2014-02-26 Thread Ross Bencina
Hello Mark, On 27/02/2014 3:52 PM, Mark Garvin wrote: Most sample banks these days seem to be in NKI format (Native Instruments). They have the ability to map ranges of a keyboard into different samples so the timbres don't become munchkin-ized or Vader-ized. IOW, natural sound within each

Re: [music-dsp] Hosting playback module for samples

2014-02-27 Thread Ross Bencina
On 28/02/2014 12:16 AM, Michael Gogins wrote: For straight sample playback, the C library FluidSynth, you can use it via PInvoke. FluidSynth plays SoundFonts, which are widely available, and there are tools for making your own SoundFonts from sample recordings. For more sophisticated synthesis,

Re: [music-dsp] Hosting playback module for samples

2014-02-27 Thread Ross Bencina
On 28/02/2014 2:06 PM, Michael Gogins wrote: I think the VSTHost code could be adapted. It is possible to mix managed C++/CLI and unmanaged standard C++ code in a single binary. I think this could be used to provide a .NET wrapper for the VSTHost classes that C# could use. I agree. Maybe I

Re: [music-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-04 Thread Ross Bencina
On 5/03/2014 7:56 AM, Ethan Duni wrote: Seems like somebody somewhere should have already thought through the problem of matching a single biquad stage to an arbitrary frequency response - anybody? Pretty sure that the oft-cited Knud Bank Christensen paper does LMS fit of a biquad over an

Re: [music-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-04 Thread Ross Bencina
On 5/03/2014 2:27 PM, Sampo Syreeni wrote: Pretty sure that literature has to contain the relevant algorithms if used with just a single resonance. I never looked at rational function fitting, but this would be easy enough to try: http://www.mathworks.com.au/help/rf/rationalfit.html The

Re: [music-dsp] Inquiry: new systems for Live DSP

2014-03-23 Thread Ross Bencina
On 15/03/2014 1:46 AM, Richard Dobson wrote: But portaudio only states the software i/o buffer latency, it knows nothing directly of internal codec latencies. You would need to subtract the (two-way?) buffer latency portaudio reports, and then measure or compute how much of the remainder is down

Re: [music-dsp] Nyquist–Shannon sampling theorem

2014-03-27 Thread Ross Bencina
On 27/03/2014 3:23 PM, Doug Houghton wrote: Is that making any sense? I'm struggling with the fine points. I bet this is obvious if you understand the math in the proof. I'm following along, vaguely. My take is that this conversation is not making enough sense to give you the certainty you

Re: [music-dsp] Simulating Valve Amps

2014-06-19 Thread Ross Bencina
On 19/06/2014 4:52 PM, Rohit Agarwal wrote: In terms of computational complexity, most of the complexity is in modelling, tuning the parameters to fit data. However, once you're done with this offline task, running the result should not be that heavy. That process should be real-time on new

Re: [music-dsp] Simulating Valve Amps

2014-06-19 Thread Ross Bencina
On 19/06/2014 7:09 PM, Rohit Agarwal wrote: Enlighten me, does that mean faster tempo or is 10% too much delay for that? I think that this conversation is at risk of going off the rails. Make sure that you're asking the right question. There are a number of different ways that delays can

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