Re: [music-dsp] SOLA, PSOLA and WSOLA.

2019-02-26 Thread Ross Bencina
Hi Alex, > I can't understand the difference between SOLA, PSOLA and WSOLA. I'll attempt a partial answer: I think PSOLA and WSOLA are clearly distinct. PSOLA involves identifying a time varying pitch (fundamental frequency) track for the input, segmenting the input signal into (possibly

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

2018-11-06 Thread Ross Bencina
On 7/11/2018 12:03 AM, gm wrote: A similar idea would be to do some basic wavelet transfrom in octaves for instance and then do smaller FFTs on the bands to stretch and shift them but I have no idea if you can do that - if you shift them you exceed their bandlimit I assume? and if you stretch

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

2018-11-03 Thread Ross Bencina
[resending, I think I accidentally replied off-list] On 1/11/2018 5:00 AM, gm wrote: > My question rephrased: > Lets assume a spectrum of size N, can you create a meaningfull spectrum of size N/2 > by simply adding every other bin together? > > Neglecting the artefacts of the forward

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

2018-11-02 Thread Ross Bencina
On 3/11/2018 3:41 AM, Ethan Fenn wrote: No length of FFT will distinguish between a mixture of these sine waves and a single amplitude-modulated one, because they're mathematically identitical! Specifically: sin(440t) + sin(441t) = 2*cos(0.5t)*sin(440.5t) So the question isn't whether an

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

2018-10-31 Thread Ross Bencina
Hi, Sorry, late to the party and unable to read the backlog, but: The "FFT^-1" technique that Robert mentions is from a paper by Rodet and Depalle that I can't find right now. It's widely cited in the literature as "FFT^-1" That paper only deals with steady-state sinusoids however. It won't

Re: [music-dsp] Antialiased OSC

2018-08-04 Thread Ross Bencina
Hi Robert, On 5/08/2018 8:17 AM, robert bristow-johnson wrote: In a software synthetic that runs on a modern computer, the waste of memory does not seem to be salient.  4096 × 4 × 64 = 1 meg.  Thats 64 wavetables for some instrument. The salient metric is amortized number of L1/L2/L3 cache

Re: [music-dsp] Antialiased OSC

2018-08-04 Thread Ross Bencina
Hi Kevin, Wavetables are for synthesizing ANY band-limited *periodic* signal. On the other hand, the BLEP family of methods are for synthesizing band-limited *discontinuities* (first order, and/or higher order discontinuities). It is true that BLEP can be used to synthesize SOME bandlimited

Re: [music-dsp] EQ-building with fine adjustable steepness

2018-07-02 Thread Ross Bencina
Hello Rolf, On 27/06/2018 11:31 PM, rolfsassin...@web.de wrote: Now, I like to have an EQ with most probable flat response which is adjustable in steepness and frequency. [snip] Is there an analytic function decribing this? Check this one out: Thomas Hélie, "Simulation of

Re: [music-dsp] Job: Audio and Video Project Manager at Google

2017-03-02 Thread Ross Bencina
Hi Don, > I work as an engineer in the Pro Audio team here at Google. [snip] > Mobile is the future of how people consume media There seems to be a contradiction here. Typically, "Pro Audio" includes the production and broadcast side of things. The job ad mentions only consumption and

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-23 Thread Ross Bencina
On 24/09/2016 3:01 PM, Andrew Simper wrote: > "Hard Sync Without Aliasing," Eli Brandt > http://www.cs.cmu.edu/~eli/papers/icmc01-hardsync.pdf > > But stick to linear phase as you can correct more easily for dc offsets. What's your reasoning for saying that? I'm guessing it depends on

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-23 Thread Ross Bencina
On 24/09/2016 1:28 PM, Andrew Simper wrote: Corrective grains are also called BLEP / BLAMP etc, so have a read about those. Original reference: "Hard Sync Without Aliasing," Eli Brandt http://www.cs.cmu.edu/~eli/papers/icmc01-hardsync.pdf ___

Re: [music-dsp] efficient running max algorithm

2016-09-03 Thread Ross Bencina
On 4/09/2016 1:42 PM, robert bristow-johnson wrote: i think the worst case ain't gonna be too much better than O(log2(windowSize)) per sample even with amortization over the block. You can think that if you like, but I don't think the worst case is that bad. I have given examples. If you

Re: [music-dsp] efficient running max algorithm

2016-09-03 Thread Ross Bencina
On 4/09/2016 6:27 AM, robert bristow-johnson wrote: if i were to test this out myself, i need to understand it enough to write C code (or asm code) to do it. The paper provides all of the information that you need for the basic implementation (which I recommend to start with). If there is

Re: [music-dsp] efficient running max algorithm

2016-09-03 Thread Ross Bencina
On 4/09/2016 2:49 AM, robert bristow-johnson wrote: sorry to have to get to the basics, but there are no *two* length parameters to this alg. there is only one. define the streaming real-time input sequence as x[n]. the length of this signal is infinite. output of running max alg is

Re: [music-dsp] efficient running max algorithm

2016-09-03 Thread Ross Bencina
On 3/09/2016 2:14 PM, robert bristow-johnson wrote: and in discussing iterators, says nothing about push_back() and the like. push_back(), push_front(), pop_back(), pop_front() are methods generally available on queue-like containers. from online i can get an idea, but it seems to me to

Re: [music-dsp] efficient running max algorithm

2016-09-02 Thread Ross Bencina
On 3/09/2016 3:12 AM, Evan Balster wrote: Just a few clarifications: - Local maxima and first difference don't really matter. The maximum wedge describes global maxima for all intervals [t-x, t], where x=[R-1..0]. I think it's interesting to look at the algorithm from different

Re: [music-dsp] efficient running max algorithm

2016-09-02 Thread Ross Bencina
On 2/09/2016 4:37 PM, Ross Bencina wrote: When the first difference is positive, the history is trimmed. This is the only time any kind of O(N) or O(log2(N)) operation is performed. First difference positive implies that a new local maximum is achieved: in this case, all of the most recent

Re: [music-dsp] efficient running max algorithm

2016-09-02 Thread Ross Bencina
Hello Robert, > i think i understand the algorithm. Your description seems quite distant from the algorithm as I understand it. Considering only running max: In effect, the running max keeps a history of the decreasing sub-sequences of the input. When the first difference of the input is

Re: [music-dsp] Choosing the right DSP, what things to look out for?

2016-08-24 Thread Ross Bencina
On 25/08/2016 8:44 AM, Max K wrote: How important do you reckon FFT hardware acceleration [is] when choosing the DSP? Most (all?) DSPs will be somewhat optimised for performing FFTs. They may not have special FFT hardware, but the vendor will most likely provide an optimised FFT library.

Re: [music-dsp] Faster Fourier transform from 2012?

2016-08-22 Thread Ross Bencina
Igor Carron's blog is also worth a look: http://nuit-blanche.blogspot.com.au/ On 23/08/2016 12:27 AM, Bjorn Roche wrote: In case you can't access that link, he doesn't give much info about how System Compression works ___ dupswapdrop: music-dsp

Re: [music-dsp] Faster Fourier transform from 2012?

2016-08-21 Thread Ross Bencina
[Sorry about my previous truncated message, Thuderbird is buggy.] I wonder what the practical musical applications of sFFT are, and whether any work has been published in this area since 2012? > http://groups.csail.mit.edu/netmit/sFFT/hikp12.pdf Last time I looked at this paper, it seemed

Re: [music-dsp] Faster Fourier transform from 2012?

2016-08-21 Thread Ross Bencina
On 22/08/2016 3:08 AM, Max Little wrote: indeed there are faster algorithms than the FFT if the signal is 'sparse' (or approximately sparse) in the Fourier domain. This is essentially the same idea as in compressed sensing, where you can 'beat' the Nyquist criterion for sparse signals.

Re: [music-dsp] idealized flat impact like sound

2016-07-28 Thread Ross Bencina
On 28/07/2016 3:00 AM, gm wrote: I want to create a signal thats similar to a reverberant knocking or impact sound, basically decaying white noise, but with a more compact onset similar to a minimum phase signal and spectrally completely flat. Maybe consider mixing multiple signals together:

Re: [music-dsp] Supervised DSP architectures (vs. push/pull)

2016-07-27 Thread Ross Bencina
that is likely in the local CPUs cache. Cheers, Ross. On 28/07/2016 5:38 AM, Evan Balster wrote: Hello --- Some months ago on this list, Ross Bencina remarked about three prevailing "structures" for DSP systems: Push, pull and *supervised architectures*. This got some wheels turn

Re: [music-dsp] BW limited peak computation?

2016-07-27 Thread Ross Bencina
On 28/07/2016 12:04 AM, Ethan Fenn wrote: Because I don't think there can be more than one between any two adjacent sampling times. This really got the gears turning. It seems true, but is it a theorem? If not, can anyone give a counterexample? I don't know whether it's a classical

Re: [music-dsp] BW limited peak computation?

2016-07-26 Thread Ross Bencina
On 27/07/2016 7:09 AM, Sampo Syreeni wrote: Now, what I wonder is, could you still somehow pinpoint the temporal location of an extremum between sampling instants, by baseband logic? Because I don't think there can be more than one between any two adjacent sampling times. Presumably the

Re: [music-dsp] highly optimised variable rms and more

2016-07-18 Thread Ross Bencina
can't see how it can possibly be better than O(log2(w)). :-\ r b-j Original Message Subject: Re: [music-dsp] highly optimised variable rms and more From: "Ross Bencina" <rossb-li...@audiomulch.com> Date: Mon, July 18,

Re: [music-dsp] looking for tutorials

2016-06-13 Thread Ross Bencina
>>Do everything in the recording studio Here's my first attempt at a tutorial on seekable lock-free audio record/playback: http://www.rossbencina.com/code/interfacing-real-time-audio-and-file-io Passion is a good thing Ty seems to be planning to re-implement just about everything:

Re: [music-dsp] a family of simple polynomial windows and waveforms

2016-06-13 Thread Ross Bencina
On 13/06/2016 3:01 PM, robert bristow-johnson wrote: many hours of integration by parts there's gotta be easier ways of doing it (like Euler's with binomial). I made a Python script for James' polynomial (binomial, Eulers) (sample output is at the bottom of the script). It did take a few

Re: [music-dsp] a family of simple polynomial windows and waveforms

2016-06-12 Thread Ross Bencina
On 12/06/2016 8:04 PM, Andy Farnell wrote: Great to follow this Ross, even with my weak powers of math its informative. My powers of math are still pretty weak, but I've been spending time at the gym lately ;) I did some experiments with Bezier after being hugely inspired by the sounds

Re: [music-dsp] a family of simple polynomial windows and waveforms

2016-06-11 Thread Ross Bencina
On 12/06/2016 3:05 AM, Andy Farnell wrote: Does it make any sense to talk about the "spectrum of a polynomial" over some (periodic) interval (less than infinity)?? Or is that silly talk? For the infinite interval: Expanding the definition of the Fourier transform, for polynomial p: P(w) =

Re: [music-dsp] a family of simple polynomial windows and waveforms

2016-06-11 Thread Ross Bencina
Hi Andy, On 11/06/2016 9:16 PM, Andy Farnell wrote: Is there something general for the spectrum of all polynomials? I think Robert was referring to the waveshaping spectrum with a sinusoidal input. If the input is a (complex) sinusoid it follows from the index laws: (e^(iw))^2 = e^(i2w)

Re: [music-dsp] a family of simple polynomial windows and waveforms

2016-06-10 Thread Ross Bencina
Nice! On 11/06/2016 11:31 AM, James McCartney wrote: f(x) = (1-x^a)^b Also potentially interesting for applying waveshaping to quadrature oscillators: https://www.desmos.com/calculator/vlmynkrlbs Ross. ___ dupswapdrop: music-dsp mailing list

Re: [music-dsp] High quality really broad bandwidth pinknoise (ideally more than 32 octaves)

2016-04-11 Thread Ross Bencina
On 12/04/2016 10:26 AM, Evan Balster wrote: I haven't yet come across an automated process for designing high-quality pinking filters, so if someone can offer one up I'd also love to hear about it! Last time that I checked (about a year and a half ago) the following was the best reference

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

2016-02-24 Thread Ross Bencina
On 23/02/2016 7:42 PM, Kjetil Matheussen wrote: But that's why I ask, so I don't have to do the implementation. It seems like a common task that someone, probably many, have already done. Just because many people have already done it does not mean you should not also do it from scratch.

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

2016-02-22 Thread Ross Bencina
On 23/02/2016 1:24 AM, Kjetil Matheussen wrote: On Mon, Feb 22, 2016 at 2:59 PM, Ross Bencina <rossb-li...@audiomulch.com <mailto:rossb-li...@audiomulch.com>> wrote: Hi Kjetil, On 22/02/2016 11:52 PM, Kjetil Matheussen wrote: I wonder if anyone has a tip

Re: [music-dsp] MIDI Synth Design Advice Please

2016-02-21 Thread Ross Bencina
On 2/02/2016 2:10 AM, Scott Gravenhorst wrote: Advice regarding this endeavor would be appreciated In case you haven't found it, you should research the disable_pvt config file flag. It can reduce system jitter a little. Hi again Scott, The following Linux system jitter optimisation blog

Re: [music-dsp] MIDI Synth Design Advice Please

2016-02-06 Thread Ross Bencina
Hi Scott, Interesting project! On 2/02/2016 2:10 AM, Scott Gravenhorst wrote: Advice regarding this endeavor would be appreciated In case you haven't found it, you should research the disable_pvt config file flag. It can reduce system jitter a little. Ross.

Re: [music-dsp] Anyone using Chebyshev polynomials to approximate trigonometric functions in FPGA DSP

2016-01-20 Thread Ross Bencina
On 21/01/2016 2:36 PM, robert bristow-johnson wrote: > i thought i understood Tchebyshev polynomials well. including their > trig definitions (for |x|<1), but if what you're trying to do is > generate a sinusoid from polynomials, i don't understand where the > "Tchebyshev" (with or without the

Re: [music-dsp] Anyone using Chebyshev polynomials to approximate trigonometric functions in FPGA DSP

2016-01-19 Thread Ross Bencina
Sorry, my previous message got truncated for some reason. On 20/01/2016 5:56 AM, Alan Wolfe wrote: Did you know that rational quadratic Bezier curves can exactly represent conic sections, and thus give you exact trig values? As Andrew said, the curve lies on a conic section, but the

Re: [music-dsp] Introducing: Axoloti

2015-12-14 Thread Ross Bencina
Hi Johannes, Nice to see a board with 1/4 inch jacks :) Does the board run an OS, or does the patcher compile bare-metal images? I assume there's some kind of OS if you support class-compliant MIDI over USB. Thanks, Ross. On 15/12/2015 7:05 AM, Johannes Taelman wrote: Hi, I'm pleased to

Re: [music-dsp] how to derive spectrum of random sample-and-hold noise?

2015-11-05 Thread Ross Bencina
Thanks Ethan(s), I was able to follow your derivation. A few questions: On 4/11/2015 7:07 PM, Ethan Duni wrote: It's pretty straightforward to derive the autocorrelation and psd for this one. Let me restate it with some convenient notation. Let's say there are a parameter P in (0,1) and 3

[music-dsp] how to derive spectrum of random sample-and-hold noise?

2015-11-03 Thread Ross Bencina
Hi Everyone, Suppose that I generate a time series x[n] as follows: >>> P is a constant value between 0 and 1 At each time step n (n is an integer): r[n] = uniform_random(0, 1) x[n] = (r[n] <= P) ? uniform_random(-1, 1) : x[n-1] Where "(a) ? b : c" is the C ternary operator that takes on the

Re: [music-dsp] how to derive spectrum of random sample-and-hold noise?

2015-11-03 Thread Ross Bencina
On 4/11/2015 9:39 AM, robert bristow-johnson wrote: i have to confess that this is hard and i don't have a concrete solution for you. Knowing that this isn't well known helps. I have an idea (see below). It might be wrong. it seems to me that, by this description: r[n] =

Re: [music-dsp] how to derive spectrum of random sample-and-hold noise?

2015-11-03 Thread Ross Bencina
art looking. Ross. E On Tue, Nov 3, 2015 at 9:42 AM, Ross Bencina <rossb-li...@audiomulch.com <mailto:rossb-li...@audiomulch.com>> wrote: Hi Everyone, Suppose that I generate a time series x[n] as follows: >>> P is a constant value between 0 and 1

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-20 Thread Ross Bencina
On 21/09/2015 10:34 AM, Bjorn Roche wrote: I noticed that PortAudio's API allows one to open a duplex stream with different stream parameters for each device. Does it actually make sense to open an input device and an output device with... * ...different sample rates? PA

Re: [music-dsp] sinc interp, higher orders

2015-09-11 Thread Ross Bencina
> sinc(x) := sin(x)/x On 12/09/2015 2:20 AM, Nigel Redmon wrote: I’m also aware that some people would look at me like I’m a nut to even bring up that distinction. I considered making the distinction, but it is discussed at the first link that I provided: >

Re: [music-dsp] sinc interp, higher orders

2015-09-11 Thread Ross Bencina
On 12/09/2015 1:13 AM, Nuno Santos wrote: Curiosity, by sinc do you mean sin function? sinc(x) := sin(x)/x https://en.wikipedia.org/wiki/Sinc_function https://ccrma.stanford.edu/~jos/pasp/Windowed_Sinc_Interpolation.html Cheers, Ross. ___

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-06 Thread Ross Bencina
Hello Andrew, Congratulations on libsoundio. I know what's involved. I have some feedback about the libsoundio-vs-PortAudio comparison. Most of my comments relate to improving the accuracy and clarify of the comparison page, but forgive me for providing a bit of commentary for other readers

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-06 Thread Ross Bencina
Hello Andrew, Thanks for your helpful feedback. Just to be clear: I maintain the PortAudio core common code and some Windows host API codes. Many of the issues that you've raised are for other platforms. In those cases I can only respond with general comments. I will forward the specific

Re: [music-dsp] warts in JUCE

2015-09-05 Thread Ross Bencina
On 6/09/2015 8:37 AM, Daniel Varela wrote: sample rate is part of the audio information so any related message ( AudioSampleBuffer ) should provide it, no need to extend the discursion. There's more than one concept at play here: (1) If you consider the AudioSampleBuffer as a stand-alone

Re: [music-dsp] kPlugCategShell

2015-06-18 Thread Ross Bencina
Hello Ralph, On 19/06/2015 9:18 AM, Ralph Glasgal wrote: I used to have AudioMulch 1.0 working fine with Waves IR-1 VST hall impulse responses. But after a computer crash I can't seem to get Waves working with either AudioMulch 1.0 or 2.2 due to a lack of kPlugCategShell support. How do I get

Re: [music-dsp] FFTW Help in C

2015-06-12 Thread Ross Bencina
On 12/06/2015 6:51 AM, Richard Dobson wrote: If it is purely for graphic display, the interesting aspect coding-wise will be timing, so that the display coincides closely enough with the audio it represents. The following paper might give some idea about doing that with PortAudio timestamps:

Re: [music-dsp] FFTW Help in C

2015-06-12 Thread Ross Bencina
Hey Bjorn, Connor, On 12/06/2015 1:27 AM, Bjorn Roche wrote: The important thing is to do anything that might take an unbounded amount of time outside your callback. For a simple FFT, the rule of thumb might bethat all setup takes place outside the callback. For example, as long as you do all

Re: [music-dsp] Dither video and articles

2015-02-05 Thread Ross Bencina
Hi Ethan, On 6/02/2015 1:17 PM, Ethan Duni wrote: There is just no way A/B testing on a sample of listeners, at loud, but still realistic listening levels, would show that dithering to 16bit makes a difference. Well, can you refer us to an A/B test that confirms your assertions? Personally

Re: [music-dsp] Dither video and articles

2015-02-05 Thread Ross Bencina
On 6/02/2015 1:50 PM, Tom Duffy wrote: The AES report is highly controversial. Plenty of sources dispute the findings. Can you name some? Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links

Re: [music-dsp] Efficiently modulate filter coefficients without artifacts?

2015-02-02 Thread Ross Bencina
On 2/02/2015 9:45 PM, Vadim Zavalishin wrote: One should be careful not to mix up two different requirements: - time-varying stability of the filter - the minimization of modulation artifacts True. My logic was thus: One way to minimise artifacts is to band-limit the coefficient changes.

Re: [music-dsp] Efficiently modulate filter coefficients without artifacts?

2015-02-01 Thread Ross Bencina
Hello Alan, On 1/02/2015 4:51 AM, Alan O Cinneide wrote: Dear List, While filtering an audio stream, I'd like to change the filter's characteristics. You didn't say what kind of filter, so I'll assume a bi-quad section. In order to do this without audible artifacts, I've been filtering a

Re: [music-dsp] Sallen Key with sin only coefficient computation

2014-12-21 Thread Ross Bencina
On 21/12/2014 5:12 PM, Andrew Simper wrote: and all the other papers (including the SVF version of the same thing I did a while back) are always available here: www.cytomic.com/techincal-papers Actually: http://www.cytomic.com/technical-papers -- dupswapdrop -- the music-dsp mailing list and

Re: [music-dsp] magic formulae

2014-11-27 Thread Ross Bencina
On 28/11/2014 12:54 AM, Victor Lazzarini wrote: Thanks everyone for the links. Apart from an article in arXiv written by viznut, I had no further luck finding papers on the subject (the article was from 2011, so I thought that by now there would have been something somewhere, beyond the code

Re: [music-dsp] Simulating Valve Amps

2014-06-21 Thread Ross Bencina
Hi Rich, On 22/06/2014 1:09 AM, Rich Breen wrote: Just as a data point; Been measuring and dealing with converter and DSP throughput latency in the studio since the first digital machines in the early '80's; Out of interest, what is your latency measurement method of choice? my own

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

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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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

2013-08-06 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] 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] 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] 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

2013-03-08 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

2013-03-07 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] 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] 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] 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] 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] 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] 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] 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

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