[music-dsp] http://musicdsp.org/
http://musicdsp.org/ seems to be down, does anyone know if the webmaster can be contacted to fix it? CONFIDENTIALITY NOTICE: This e-mail message (including any attachments) is for the sole use of the intended recipient and may contain confidential, privileged and/or trade secret information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] [admin] list etiquette
Thank you Douglas. Some of the toxic replies here have really harmed the friendly spirit and approachability of this mailing list. -Original Message- From: music-dsp [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Douglas Repetto Sent: 22 August 2015 16:22 To: A discussion list for music-related DSP Subject: [music-dsp] [admin] list etiquette Hi everyone, Douglas the list admin here. I've been away and haven't really been monitoring the list recently. It's been full of bad feelings, unpleasant interactions, and macho posturing. Really not much that I find interesting. I just want to reiterate a few things about the list. I'm loathe to make or enforce rules. But the list has been pretty much useless for the majority of subscribers for the last year or so. I know this because many of them have written to complain. It's certainly not useful to me. I've also had several reports of people trying to unsubscribe other people and other childish behavior. Come on. So: * Please limit yourself to two well-considered posts per day. Take it off list if you need more than that. * No personal attacks. I'm just going to unsub people who are insulting. Sorry. * Please stop making macho comments about first year EE students know this and blahblahblah. This list is for anyone with an interest in sound and dsp. No topic is too basic, and complete beginners are welcome. I will happily unsubscribe people who find they can't consistently follow these guidelines. The current list climate is hostile and self-aggrandizing. No beginner, gentle coder, or friendly hobbyist is going to post to such a list. If you can't help make the list friendly to everyone, please leave. This isn't the list for you. douglas ___ music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp ___ music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] [admin] another HTML test
Thanks Doug :) -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of douglas repetto Sent: 11 November 2013 17:19 To: A discussion list for music-related DSP Subject: Re: [music-dsp] [admin] another HTML test Okay, it seems to be working. You can now post to the list in HTML but your message will be converted to plain text. This seems like a good first step to accommodate people who need plain text and people for whom sending plain text is a pain. I tried to respond to some message from my phone while I was out of town and found that it's impossible to send plain text email that way! best, douglas On 11/11/13 12:16 PM, douglas repetto wrote: Hi Dave, I have the list set to convert HTML mail to plain text. So it's a good sign that the email went through. I'm composing this in red text and changing fonts around. That should all disappear when this message arrives... Here's a list: 1. * taco 2. * truck Wheee! -- ... http://artbots.org .douglas.irving http://dorkbot.org .. http://music.columbia.edu/cmc/music-dsp ...repetto. http://music.columbia.edu/organism ... http://music.columbia.edu/~douglas -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] family of soft clipping functions.
This reminds me of experimenting with polynomials as an amplitude enveloping
function for a soft synthesiser. There was something rather alluring about the
idea of a one-line-of-code amplitude envelope - unfortunately it made creating
the envelopes pretty tiresome, and when I thought about the performance I
realised it was probably slower to do that maths than the few conditionals you
would use for a standard piecewise envelope. So basically a rubbish idea unless
you have some strange need to have your amplitude envelope be a single line of
code :I
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
Sent: 29 October 2013 01:56
To: music-dsp@music.columbia.edu
Subject: [music-dsp] family of soft clipping functions.
at the last AES in NYC, i was talking with some other folks (that likely hang
out here, too) about this family of soft clipping curves made outa polynomials
(so you have some idea of how high in frequency any generated images will
appear).
these are odd-order, odd-symmetry polynomials that are monotonic from -1 x
+1 and have as many continuous derivatives as possible at +/- 1 where these
curves might be spliced to constant-valued rails.
the whole idea is to integrate the even polynomial (1 - x^2)^N
x
g(x) = integral{ (1 - v^2)^N dv}
0
you figger this out using binomial expansion and integrating each power term.
normalize g(x) with whatever g(1) is so that the curve is g(x)/g(1) and splice
that to two constant functions for the rails
{ -1x = -1
{
f(x) = { g(x)/g(1) -1 = x = +1
{
{ +1 +1 = x
you can hard limit (at +/- 1) before passing through this soft clipper and it
still works fine. but it has some gain in the linear region which is g(0).
if you want to add some even harmonic distortion to this, add a little bit of
(1 - x^2)^M
to f(x) for |x| 1 and it's still smooth everywhere, but there is a little DC
added (which has to be the case for even-symmetry distortion).
M does not have to be the same as N and i wouldn't expect it to be.
you can think of f(x) as a smooth approximation the sign or signum
function
sgn(x) = lim f(a*x)
a - +inf
or
sgn(x) = lim f(x)
N - +inf
which motivates using this as a smooth approximation of the sgn(x)
function as an alternative to
(2/pi)*arctan(a*x) or tanh(a*x). from the sgn(x) function, you can
create smooth versions of the unit step function and use that for
splicing. as many derivatives are continuous in the splice as possible.
and it satisfies conditions of symmetry and complementarity that are
useful in our line of work.
u(x) = 1/2 * (1 + sgn(x)) =approx 1/2*(1 + f(x))
you can run a raised cosine (Hann) through this and get a more flattened
Hann. in some old stuff i wrote, i dubbed this window:
w(x) = 1/2 + (9/16)*cos(pi*x) - (1/16)*cos(3*pi*x)
as the Flattened Hann Window but long ago Carla Scaletti called it the
Bristow-Johnson window in some Kyma manual. i don't think it deserves
that label (i've seen that function in some wavelet/filterbank lit since
for half-band filters). you get that window by running a simple Hann
through the biased waveshaper:
1/2 * ( 1 + f(2x-1) )
with N=1. you will get an even more pronounced effect (of smoothly
flattening the Hann) with higher N.
below is a matlab file that demonstrates this as a soft clipping function.
BTW, Olli N, this can be integrated with that splicing theory thing we
were talking about approximately a year ago. it would define the
odd-symmetry component. we should do an AES paper about this. i
think now there is nearly enough meat to make a decent paper. before
i didn't think so.
FILE: softclip.m
line_color = ['g' 'c' 'b' 'm' 'r'];
figure;
hold on;
x = linspace(-2, 2, 2^18 + 1);
x = max(x, -1); % hard clip for |x| 1
x = min(x, +1);
for N = 0:10
n = linspace(0, N, N+1);
a = (-1).^n ./ (factorial(N-n) .* factorial(n) .* (2*n + 1));
a = a/sum(a);
y = x .* polyval(fliplr(a), x.^2); % stupid MATLAB puts the coefs
in the wrong order
plot(x, y, line_color(mod(N,length(line_color))+1) );
end
hold off;
have fun with it, if you're so inclined.
L8r,
--
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] Analog 24dB lowpass filter emulation perfected. (Roland, Oberheim etc)
Interesting, thanks -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Ove Karlsen Sent: 28 October 2013 11:57 To: music-dsp@music.columbia.edu Subject: [music-dsp] Analog 24dB lowpass filter emulation perfected. (Roland, Oberheim etc) Peace Be With You. I have perfected the analog 24dB lowpass filter in digital form. It is a generalized and fast implementation without obscurities. It takes from analog simply what is good, and otherwise is a perfect implementation one would expect from a digital filter. I wrote about it here: http://ovekarlsen.com/Blog/abdullah-filter/ Interested should read, and there is a soundexample. Peace Be With You. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] My latest computer DSP signal pathfor audioimprovements
apparently that is more emotional and personal for some people to be able to neutrally communicate about This from the man who wants to recapture the sound of records from his lost youth :P I think Sampo had a lot of good points personally. I would not dismiss the value of simplification, your existing audio setup is bafflingly complicated. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo Verelst Sent: 03 May 2013 00:12 To: A discussion list for music-related DSP Subject: Re: [music-dsp] My latest computer DSP signal pathfor audioimprovements Sampo Syreeni wrote: On 2013-05-02, Rob Belcham wrote: Agreed. There is a lot more low end in the processed but the loss of the high frequencies is distracting. ... Well, I wanted to make clear modern mixes on consumer materials are often messed up, and discuss that. I f your monitoring makes all the CD and well known blurays and such sound fine to your ears, fine. I'm heavy enough edified univ. EE with serious theoretical physics knowledge, and well developed practical skills, so I don't feel I need new theories, when the 1d and 2d year Electrical Engineering university years offer a complete, mathematically closing, and theoretically accurate theory for electrical network theory, signal processing, information theory and such. I mean I know what an FFT will do with a normal electrical signal transient, what poles and zeros are and do, and what the difference is between a control loop and a filter+delay, and that such systems have certain mathematical properties. I don't know about you, but I had absolutely great records and radio to listen to as a kid and teenager, and I can't stand the garbage that nowadays is made of the materials of good, serious musicians (which I am too), so that's what I wanted to discuss, and to an extend offer solutions for, but apparently that is more emotional and personal for some people to be able to neutrally communicate about. You mathematically oriented language doesn't do justice to the basis of EE, which is much harder than you appear to think, those different signal paths can be followed I think by audio engineers with some experience (and have to an extend), and there is such thing as loudness perception and studio processing that can work with it. So apart from some new boys and girls wanting to use music for power instead of enjoyment, there's a whole world of studio processing, and some of that I'm quite accurately aware of, so maybe that's interesting for some people to hear about. I suppose if I'd put a schematic of a digital TV before you, you might be tempted to think it's a mess, but hey... T.V. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] basic trouble with signed and unsigned types
As someone pointed out what you really want here is c++ templates, those compile out completely and can be specialised for different types - which you need for speed. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Sampo Syreeni Sent: 01 May 2013 23:08 To: A discussion list for music-related DSP Subject: Re: [music-dsp] basic trouble with signed and unsigned types On 2013-05-01, Bjorn Roche wrote: Not sure I completely understand what you are after, but I think most audio folks don't use types like int, long and short, but rather types like int32_t from types.h (or sys/types.h). There are various guarantees made about the sizes of those types that you can look up. Correct, and for now I'm doing something similar by calculating all of my widths, maximum values and whatnot beforehand into #defines. In the end I'd like an implementation that is fully parametrizable, though, and with as few adjustable parameters as at all possible. That's because while the idea I'm trying to put into code is primarily useful wrt audio, the basic framework of dithering I'm working within admits plenty of environments beyond *just* that. So, ideally, I'd want to make the code just work, even within certain highly unusual environments. (Say, you receive a single bit integer type from the outside which you have to use to process your samples; there's way to go towards that but it might eventually happen. And, by the way, if you've ever done anything with DSD/SACD, treating the 1-bit channel as a modulo-linear one actually seems to make it too perfectly ditherable, breaking the Lipshitz and Vanderkooy critique. Thus, audio relevance even here. :) What I'm after is ways of reimplementing at least right shifts for unsigned quantities within standard C99, so that my new operator is fully portable and guaranteedly implements right shifts without sign extension, regardless of what integer type you use them on, and without any parametric polymorphism or reference to built-in constants describing your types at the meta level. There are bound to be other problems I have to worry about in the future, like how to deal with the possibility of one's complement arithmetic where I'm now taking parities. And whatnot. But at least the right shift would help me a lot right now. Also, I assume you've given C++ templates and operator overloading consideration. Yes. Immediately after I wanted to die. ;) -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] My latest computer DSP signal path for audioimprovements
There is quite an audible loss of high end, it's especially noticeable on the 'double processed' example which sounds very low passed. Is that intentional? -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo Verelst Sent: 01 May 2013 17:57 To: A discussion list for music-related DSP Subject: Re: [music-dsp] My latest computer DSP signal path for audioimprovements Rob Belcham wrote: Hi Theo, Some wet / dry example audio would be interesting to hear. Cheers Rob .. OK, that should be reasonable. For this limited club (little over a hundred world wide ?) of people I suppose there's no real problem with a few very short (7 secs) examples from varied musical sources: http://www.theover.org/Musicdspexas/dsptst1.mp3 BEWARE: it's a nice few examples, but it isn't produced very nicely because I forgot to put off a 10 dB boost+perfect limiter (1.2 Sec) in the recording... I didn't want to start the whole startup and mic mixing again, so beware it sounds a bit like radio-compressed, but the idea is still ok-ish. For people who are interested in playing/learning/??? with my actual signal path, I by now have a complete set of tcl scripts to start the particular chain settings up, completely automatically, with settings from the last example, and all the jack-rack and other files, which I can make downloadable per individual. You'll need a strong Linux machine with a 192kHz soundcard and the ladspa plugins, jamin and jack installed. Greetings Theo V. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] meassuring the difference
Your mean square error procedure is slightly incorrect. You should take the final signals from both processes, say A[1..n] and B[1..n], subtract them to get your error signal E[1..n], then the mean square error is the sum of the squared error over n. Sum( E[1..n]^2 ) / n This (MSE) is a statistical approach though and isn't necessarily a great way of measuring perceived acoustical differences. It depends on the nature of your signal but you may want to check the error in the frequency domain (weighted to specific frequency band if appropriate) rather than the time domain. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of volker böhm Sent: 07 March 2013 15:10 To: A discussion list for music-related DSP Subject: [music-dsp] meassuring the difference dear all, i'm trying to meassure the difference between two equivalent but not identical processes. right now i'm feeding some test signals to both algorithms at the same time and subtract the output signals. now i'm looking for something to quantify the error signal. from statistics i know there is something like the mean squared error. so i'm squaring the error signal and take the (running) average. mostly i'm getting some numbers very close to zero and a gut feeling tells me i want to see those on a dB scale. so i'm taking the logarithm and multiply by 10, as i have already squared the values before. (as far as i can see, this is equivalent to a RMS meassurement). is there a correct/better/preferred way of doing this? next to a listening test, in the end i want to have a simple meassure of the difference of the two processes which is close to our perception of the difference. does that make sense? thanks for any comments, volker. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Lerping Biquad coefficients to a flat response
Ah, I wondered what you meant about those parenthesis! The better efficiency comes from avoiding all the divisions (I don't normalise out a0 in my implementation) - it's a pretty trivial performance difference in reality but the coefficient equations are a little bit neater as well - less of those confusing parenthesis ;) -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 20:59 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response On 1/4/13 1:29 PM, Thomas Young wrote: Er.. yes sorry I transcribed it wrong, well spotted a1: ( - 2 * cos(w0) ) * A^2 ooops! i'm embarrassed! i was thinking it was -2 + cos(), sorry! can't believe it. chalk it up to being a year older and losing one year's portion of brain cells. So yea it's the same, I just rearranged it a bit for efficiency dunno what's more efficient before normalizing out a0, but if it's after, you change only the numerator feedforward coefs and you would multiply them by the reciprocal of what the linear peak gain, 1/A^2. sorry, that parenth thing must be early onset alzheimer's. bestest, r b-j -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 18:25 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response On 1/4/13 1:11 PM, Thomas Young wrote: someone tell me what it was about In a nutshell... Q: What is the equation for the coefficients of a peaking EQ biquad filter with constant 0 dB peak gain? A: This (using cookbook variables): b0: 1 + alpha * A b1: -2 * cos(w0) b2: 1 - alpha * A a0: A^2 + alpha * A a1: - 2 * cos(w0) * A^2--- are sure you don't need pareths here? a2: A^2 - alpha * A dragged over a lot of emails ;) yeah, i guess that's the same as b0: (1 + alpha * A)/A^2 b1: (-2 * cos(w0)/A^2 b2: (1 - alpha * A)/A^2 a0: 1 + alpha / A a1: -2 * cos(w0) a2: 1 - alpha / A if you put in the parenths where you should, i think these are the same. r b0j -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 17:58 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response looks like i came here late. someone tell me what it was about. admittedly, i didn't completely understand from a cursory reading. the only difference between the two BPFs in the cookbook is that of a constant gain factor. in one the peak of the BPF is always at zero dB. in the other, if you were to project the asymptotes of the skirt of the freq response (you know, the +6 dB/oct line and the -6 dB/oct line), they will intersect at the point that is 0 dB and at the resonant frequency. otherwise same shape, same filter. the peaking EQ is a BPF with gain A^2 - 1 with the output added to a wire. and, only on the peaking EQ, the definition of Q is fudged so that it continues to be related to BW in the same manner and so the cut response exactly undoes a boost response for the same dB, same f0, same Q. nothing more to it than that. no Orfanidis subtlety. if the resonant frequency is much less than Nyquist, then there is even symmetry of magnitude about f0 (on a log(f) scale) for BPF, notch, APF, and peaking EQ. for the two shelves, it's odd symmetry about f0 (if you adjust for half of the dB shelf gain). the only difference between the high shelf and low shelf is a gain constant and flipping the rest of the transfer function upside down. this is the case no matter what the dB boost is or the Q (or S). if f0 approaches Fs/2, then that even or odd symmetry gets warped from the BLT and ain't so symmetrical anymore. nothing else comes to mind. -- 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 http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Lerping Biquad coefficients to a flat response
Hi Nigel, which analogue prototype are you referring to when you suggest multiplying denominator coefficients by the gain factor, the peaking one? -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Nigel Redmon Sent: 04 January 2013 09:26 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response 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 messing about. I read quickly this morning and missed this...with something like a lowpass, you do get irregularities, but with something like a peaking filter it stays pretty smooth when summing with the original signal (I guess because the phase change is smoother, with the second order split up between two halves). So that part isn't a problem, BUT... Consider a wet/dry mix...say you have a 6 dB peak that you want to move down to 0 dB (skirts down to -6 dB). OK, at 0% wet you have a flat line at 0 dB. At 100% wet you have your 6 dB peak again (skirt at 0 dB). At 50% wet, you have about a 3 dB peak, skirt still at 0 dB. There is no setting that will give you anything but the skit at 0 dB. Again, as Ross said earlier, you could have just an output gain-set it to 0.5 (-6 dB), and now you have your skirt at - 6 dB, peak at 0 dB. But a wet/dry mix know is not going to do it. Ross said: There is only a difference of scale factors between your constraints and the RBJ peaking filter constraints so you should be able to use them with minor modifications (as Nigel suggests, although I didn't take the time to review his result). Assuming that you want the gain at DC and nyquist to be equal to your stopband gain then this is pretty much equivalent to the RBJ coefficient formulas except that Robert computed them under the requirement of unity gain at DC and Nyquist, and some specified gain at cf. You want unity gain at cf and specified gain at DC and Nyquist. This seems to me to just be a direct reinterpretation of the gain values. You should be able to propagate the needed gain values through Robert's formulas. Actually, it's more than a reinterpretation of the gain values (note that no matter what gain you give it, you won't get anything like what Thomas is after). The poles are peaking the filter and the zeros are holding down the shirt (at 0 dB in the unmodified filter); obviously the transfer function is arranged to keep that relationship at any gain setting. So, you need to change it so that the gain is controlling something else-changing the relationship of the motion between the poles and zeros (the mod I gave does that). On Jan 3, 2013, at 10:34 PM, Ross Bencina rossb-li...@audiomulch.com wrote: 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 messing about. Someone else might correct me on this, but I'm not sure that will get you the same effect. Your proposal seems to be based on the assumption that the filter is phase linear and 0 delay (ie that the phases all line up between input and filtered version). That's not the case. In reality you'd be mixing the phase-warped and delayed (filtered) signal with the original-phase signal. I couldn't tell you what the frequency response would look like, but probably not as good as just scaling the peaking filter output. On 4/01/2013 6:03 AM, Thomas Young wrote: Additional optional mumblings: I think really there are two 'correct' solutions to manipulating only the coefficients to my ends (that is, generation of coefficients which produce filters interpolating from bandpass to flat): The first is to go from pole/zero to transfer function, basically as you (Nigel) described in your first message - stick the zeros in the centre, poles near the edge of the unit circle and reduce their radii - doing the maths to convert these into the appropriate biquad coefficients. This isn't really feasible for me to do in realtime though. I was trying to do a sort of tricksy workaround by lerping from one set of coefficients to another but on reflection I don't think there is any mathematical correctness there. The second is to have an analogue prototype which somehow includes skirt gain and take the bilinear transform to get the equations for the coefficients. I'm not really very good with the s domain either so I actually wouldn't know how to go about this, but it's what I was originally thinking of. In the end you're going to have a set
Re: [music-dsp] Lerping Biquad coefficients to a flat response
Aha, success! Multiplying denominator coefficients of the peaking filter by A^2 does indeed have the desired effect. Thank you very much for the help -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thomas Young Sent: 04 January 2013 10:33 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response Hi Nigel, which analogue prototype are you referring to when you suggest multiplying denominator coefficients by the gain factor, the peaking one? -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Nigel Redmon Sent: 04 January 2013 09:26 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response 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 messing about. I read quickly this morning and missed this...with something like a lowpass, you do get irregularities, but with something like a peaking filter it stays pretty smooth when summing with the original signal (I guess because the phase change is smoother, with the second order split up between two halves). So that part isn't a problem, BUT... Consider a wet/dry mix...say you have a 6 dB peak that you want to move down to 0 dB (skirts down to -6 dB). OK, at 0% wet you have a flat line at 0 dB. At 100% wet you have your 6 dB peak again (skirt at 0 dB). At 50% wet, you have about a 3 dB peak, skirt still at 0 dB. There is no setting that will give you anything but the skit at 0 dB. Again, as Ross said earlier, you could have just an output gain-set it to 0.5 (-6 dB), and now you have your skirt at - 6 dB, peak at 0 dB. But a wet/dry mix know is not going to do it. Ross said: There is only a difference of scale factors between your constraints and the RBJ peaking filter constraints so you should be able to use them with minor modifications (as Nigel suggests, although I didn't take the time to review his result). Assuming that you want the gain at DC and nyquist to be equal to your stopband gain then this is pretty much equivalent to the RBJ coefficient formulas except that Robert computed them under the requirement of unity gain at DC and Nyquist, and some specified gain at cf. You want unity gain at cf and specified gain at DC and Nyquist. This seems to me to just be a direct reinterpretation of the gain values. You should be able to propagate the needed gain values through Robert's formulas. Actually, it's more than a reinterpretation of the gain values (note that no matter what gain you give it, you won't get anything like what Thomas is after). The poles are peaking the filter and the zeros are holding down the shirt (at 0 dB in the unmodified filter); obviously the transfer function is arranged to keep that relationship at any gain setting. So, you need to change it so that the gain is controlling something else-changing the relationship of the motion between the poles and zeros (the mod I gave does that). On Jan 3, 2013, at 10:34 PM, Ross Bencina rossb-li...@audiomulch.com wrote: 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 messing about. Someone else might correct me on this, but I'm not sure that will get you the same effect. Your proposal seems to be based on the assumption that the filter is phase linear and 0 delay (ie that the phases all line up between input and filtered version). That's not the case. In reality you'd be mixing the phase-warped and delayed (filtered) signal with the original-phase signal. I couldn't tell you what the frequency response would look like, but probably not as good as just scaling the peaking filter output. On 4/01/2013 6:03 AM, Thomas Young wrote: Additional optional mumblings: I think really there are two 'correct' solutions to manipulating only the coefficients to my ends (that is, generation of coefficients which produce filters interpolating from bandpass to flat): The first is to go from pole/zero to transfer function, basically as you (Nigel) described in your first message - stick the zeros in the centre, poles near the edge of the unit circle and reduce their radii - doing the maths to convert these into the appropriate biquad coefficients. This isn't really feasible for me to do in realtime though. I was trying to do a sort of tricksy workaround by lerping from one set of coefficients
Re: [music-dsp] Lerping Biquad coefficients to a flat response
someone tell me what it was about In a nutshell... Q: What is the equation for the coefficients of a peaking EQ biquad filter with constant 0 dB peak gain? A: This (using cookbook variables): b0: 1 + alpha * A b1: -2 * cos(w0) b2: 1 - alpha * A a0: A^2 + alpha * A a1: - 2 * cos(w0) * A^2 a2: A^2 - alpha * A dragged over a lot of emails ;) -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 17:58 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response looks like i came here late. someone tell me what it was about. admittedly, i didn't completely understand from a cursory reading. the only difference between the two BPFs in the cookbook is that of a constant gain factor. in one the peak of the BPF is always at zero dB. in the other, if you were to project the asymptotes of the skirt of the freq response (you know, the +6 dB/oct line and the -6 dB/oct line), they will intersect at the point that is 0 dB and at the resonant frequency. otherwise same shape, same filter. the peaking EQ is a BPF with gain A^2 - 1 with the output added to a wire. and, only on the peaking EQ, the definition of Q is fudged so that it continues to be related to BW in the same manner and so the cut response exactly undoes a boost response for the same dB, same f0, same Q. nothing more to it than that. no Orfanidis subtlety. if the resonant frequency is much less than Nyquist, then there is even symmetry of magnitude about f0 (on a log(f) scale) for BPF, notch, APF, and peaking EQ. for the two shelves, it's odd symmetry about f0 (if you adjust for half of the dB shelf gain). the only difference between the high shelf and low shelf is a gain constant and flipping the rest of the transfer function upside down. this is the case no matter what the dB boost is or the Q (or S). if f0 approaches Fs/2, then that even or odd symmetry gets warped from the BLT and ain't so symmetrical anymore. nothing else comes to mind. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. On 1/4/13 11:23 AM, Nigel Redmon wrote: Great! On Jan 4, 2013, at 2:40 AM, Thomas Youngthomas.yo...@rebellion.co.uk wrote: Aha, success! Multiplying denominator coefficients of the peaking filter by A^2 does indeed have the desired effect. Thank you very much for the help -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thomas Young Sent: 04 January 2013 10:33 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response Hi Nigel, which analogue prototype are you referring to when you suggest multiplying denominator coefficients by the gain factor, the peaking one? -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Nigel Redmon Sent: 04 January 2013 09:26 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response 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 messing about. I read quickly this morning and missed this...with something like a lowpass, you do get irregularities, but with something like a peaking filter it stays pretty smooth when summing with the original signal (I guess because the phase change is smoother, with the second order split up between two halves). So that part isn't a problem, BUT... Consider a wet/dry mix...say you have a 6 dB peak that you want to move down to 0 dB (skirts down to -6 dB). OK, at 0% wet you have a flat line at 0 dB. At 100% wet you have your 6 dB peak again (skirt at 0 dB). At 50% wet, you have about a 3 dB peak, skirt still at 0 dB. There is no setting that will give you anything but the skit at 0 dB. Again, as Ross said earlier, you could have just an output gain-set it to 0.5 (-6 dB), and now you have your skirt at - 6 dB, peak at 0 dB. But a wet/dry mix know is not going to do it. Ross said: There is only a difference of scale factors between your constraints and the RBJ peaking filter constraints so you should be able to use them with minor modifications (as Nigel suggests, although I didn't take the time to review his result). Assuming that you want the gain at DC and nyquist to be equal to your stopband gain then this is pretty much equivalent to the RBJ coefficient formulas except that Robert computed them under the requirement of unity gain at DC and Nyquist, and some specified gain at cf. You want unity gain
Re: [music-dsp] Lerping Biquad coefficients to a flat response
Er.. yes sorry I transcribed it wrong, well spotted a1: ( - 2 * cos(w0) ) * A^2 So yea it's the same, I just rearranged it a bit for efficiency -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 18:25 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response On 1/4/13 1:11 PM, Thomas Young wrote: someone tell me what it was about In a nutshell... Q: What is the equation for the coefficients of a peaking EQ biquad filter with constant 0 dB peak gain? A: This (using cookbook variables): b0: 1 + alpha * A b1: -2 * cos(w0) b2: 1 - alpha * A a0: A^2 + alpha * A a1: - 2 * cos(w0) * A^2--- are sure you don't need pareths here? a2: A^2 - alpha * A dragged over a lot of emails ;) yeah, i guess that's the same as b0: (1 + alpha * A)/A^2 b1: (-2 * cos(w0)/A^2 b2: (1 - alpha * A)/A^2 a0: 1 + alpha / A a1: -2 * cos(w0) a2: 1 - alpha / A if you put in the parenths where you should, i think these are the same. r b0j -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 04 January 2013 17:58 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response looks like i came here late. someone tell me what it was about. admittedly, i didn't completely understand from a cursory reading. the only difference between the two BPFs in the cookbook is that of a constant gain factor. in one the peak of the BPF is always at zero dB. in the other, if you were to project the asymptotes of the skirt of the freq response (you know, the +6 dB/oct line and the -6 dB/oct line), they will intersect at the point that is 0 dB and at the resonant frequency. otherwise same shape, same filter. the peaking EQ is a BPF with gain A^2 - 1 with the output added to a wire. and, only on the peaking EQ, the definition of Q is fudged so that it continues to be related to BW in the same manner and so the cut response exactly undoes a boost response for the same dB, same f0, same Q. nothing more to it than that. no Orfanidis subtlety. if the resonant frequency is much less than Nyquist, then there is even symmetry of magnitude about f0 (on a log(f) scale) for BPF, notch, APF, and peaking EQ. for the two shelves, it's odd symmetry about f0 (if you adjust for half of the dB shelf gain). the only difference between the high shelf and low shelf is a gain constant and flipping the rest of the transfer function upside down. this is the case no matter what the dB boost is or the Q (or S). if f0 approaches Fs/2, then that even or odd symmetry gets warped from the BLT and ain't so symmetrical anymore. nothing else comes to mind. -- 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 http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Lerping Biquad coefficients to a flat response
One for RBJ if he's back from his hols :) or anyone kind enough to answer of course... 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 My thought was to interpolate the coefficients towards a flat response, i.e. b0=1 b1=0 b2=0 a0=1 a1=0 a2=0 I tried some plots and this does basically work except that the characteristics of the filter are affected (namely the peak gain exceeds 0dB). Thomas Young -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Lerping Biquad coefficients to a flat response
Thanks Nigel - I have just been playing around with the pole/zero plotter (very helpful app for visualising the problem) and thinking about it. You guys are probably right the simplest approach is just to scale the output and using the peaking filter. Additional optional mumblings: I think really there are two 'correct' solutions to manipulating only the coefficients to my ends (that is, generation of coefficients which produce filters interpolating from bandpass to flat): The first is to go from pole/zero to transfer function, basically as you (Nigel) described in your first message - stick the zeros in the centre, poles near the edge of the unit circle and reduce their radii - doing the maths to convert these into the appropriate biquad coefficients. This isn't really feasible for me to do in realtime though. I was trying to do a sort of tricksy workaround by lerping from one set of coefficients to another but on reflection I don't think there is any mathematical correctness there. The second is to have an analogue prototype which somehow includes skirt gain and take the bilinear transform to get the equations for the coefficients. I'm not really very good with the s domain either so I actually wouldn't know how to go about this, but it's what I was originally thinking of. Thanks for the help -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Nigel Redmon Sent: 03 January 2013 18:48 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response Thomas-it's a matter of manipulating the A and Q relationships in the numerator and denominator of the peaking EQ analog prototypes. I'm not as good in thinking in the s domain as the z, so I'd have to plot it out and think-too busy right now, though it's pretty trivial. But just doing the gain adjustment to the existing peaking EQ, as Ross suggested, is trivial. Not much reason to go through the fuss unless you're concerned about adding a single multiply. (To add to the confusion, my peaking implementation is different for gain and boost, so that the EQ remains symmetrical, a la Zolzer). On Jan 3, 2013, at 9:34 AM, Thomas Young thomas.yo...@rebellion.co.uk wrote: I'm pretty sure that the BLT bandpass ends up with zeros at DC and nyquist Yes I think this is essentially my problem, there are no stop bands per-se just zeros which I was basically trying to lerp away - which I guess isn't really the correct approach. The solution you are proposing would work I believe; along the same lines there is a different bandpass filter in the RBJCB which has a constant stop band gain (or 'skirt gain' as he calls it) and peak gain for the passband - so a similar technique would work there by scaling the output. 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 messing about. I feel in my gut there must be some way to do it by just manipulating coefficients. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Ross Bencina Sent: 03 January 2013 17:16 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response 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 ends up with zeros at DC and nyquist so I'm not sure how you're going to define stopband gain in this case :) Maybe start with the peaking filter and scale the output according to your desired stopband gain and then set the peak gain to give 0dB at the peak. peakGain_dB = -stopbandGain_dB (assuming -ve stopbandGain_dB). Does that help? Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links
Re: [music-dsp] Ghost tone
1) The low frequencies are audible 2) It's not speaker distortion, the low frequencies are present in the signal I think the spectrum of the first signal can be a bit misleading, if you are a bit more selective about where you take the spectrum (i.e. between the asymptotic sections) the low frequency contribution is easier to see. The unpleasant pressure effect is exactly that, sound pressure waves. The strength will be dependent on the acoustics of your environment, it will be particularly objectionable if your ears happen to be somewhere where a lot of the wavefronts collide. The proximity of headphones to your ears is no doubt exacerbating the effect, especially in the very low frequencies which would otherwise bounce all over the place and diffuse. Mics generally won't pick up very low frequencies - or more accurately their sensitivity to lower frequencies is very low. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Didier Dambrin Sent: 06 December 2012 05:50 To: A discussion list for music-related DSP Subject: [music-dsp] Ghost tone Hi, Here's something to listen to: http://flstudio.image-line.com/help/publicfiles_gol/GhostTone.wav It's divided in 2 parts, the same bunch of sine harmonics in the upper range, only difference is the phase alignment. (both will appear similar through a spectrogram) Disregarding the difference in sound in the upper range, 1. anyone confirms the very low tone is very audible in the first half? 2. (anyone confirms it's not speaker distortion?) 3. anyone knows about litterature about the phenomenon? While I can understand where the ghost tone is from, I don't understand why it's audible. I happen to have hyperacusis can't stand the low traffic rumbling here around, and I was wondering why mics weren't picking it, as I perceive it very loud. I hadn't been able to resynthesize a tone as nasty until now, mainly because I was trying low tones alone, and I can't hear simple sines under 20Hz. The question is why do we(?) hear it, why is so much pressure noticable (can anyone stand it through headphones? I find the pressure effect very disturbing). Strangely enough, I find the tone a lot more audible when (through headphones) it goes to both hears, not if it's only left or right. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] i need a knee
Not getting a response on a mailing list doesn't mean people are disrespecting
you, we can do without the self righteousness thank you.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Bastian Schnuerle
Sent: 10 August 2012 11:24
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] i need a knee
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 respect that .. thanks
Am 20.07.2012 um 08:17 schrieb robert bristow-johnson:
On 7/19/12 4:19 PM, Bastian Schnuerle wrote:
hey everybody,
i am trying to compute a knee value in peak limiting. my code below
works quite nice for very loud singals (which i like), but if not so
loud values are processed this code introduces distortion. in the
mycode there a some way-off alterations, which made the signal being
quite smooth for loud signal, but as soon as it come to low peak,
they/i are/am failing.
i would be very happy if you guys could post me some suggestions why
this beaviour for low peaked signals appear and maybe your are
willing to share altererations to my code to make it universal
working ?!
.//...
static const float log6dB = 6.02059991327962f;
20*log10(2.0)
the dB equivalent to one octave.
const float kNee = log6dB-GetKneeDBfromGUI();
GetKneeDBfromGUI() returns the knee in dB relative to what? the
rails?
mFinalEnv = mProcessSignal;
how is mProcessSignal defined?
float kneeGain = 1.0;
if( mFinalEnv 1.0f ) gain = 1.0f/mFinalEnv;
and where is gain used?
const float outLimit = 2.0f*1.0f;
float curOutLimit6dBBelow = curOutLimit/2.0f; const float
maxUnequalAtt = -10.0f;
if( mFinalEnv curOutLimit6dBBelow ) //6db below limit {
floatinv = curOutLimit6dBBelow/mFinalEnv;
// do knee computation
floatkneePos1 = 0;
floatkneePos2 = 0;
if( mFinalEnv = curOutLimit )
{
kneePos1 = 1.0f;
kneePos2 = 1.0f;
}
else
{
// create two knee curves
floatxPos = (mFinalEnv - curOutLimit6dBBelow)/
(curOutLimit-curOutLimit6dBBelow);
kneePos1 = xPos*xPos*xPos*xPos;
kneePos2 = xPos;
}
// xfade between the knees
float kneeFactor = kNee/log6dB;
is kNee ever anything other than 1?
kneeFactor = ((1.0f-kneeFactor)*kneePos1 +
kneeFactor*kneePos2)/7;
i see the xfade. i don't see why you are dividing by 7.
.
floatoverallAttDb = FastLinToDb(inv);
kneeAtt = FastDbToLin(get_max(gain*(overallAttDb*
(kneeFactor)),maxUnequalAtt));
okay, gain is used here.
kneeGain = kneeAtt;
}
}
for( int fc = 0; fc channels; ++fc ) {
float val = mFinalVectors[fc][mFinaTail[fc]] * kneeGain; }
..//..
can you define mathematically you're trying to do? i can see that you
have some sorta mix between linear and x^4. i realize this is for a
soft-knee limiter of some sort. but i cannot grok the intended math
to be accomplished with this code. can you just state the math (with
if statements, where needed)?
--
r b-j r...@audioimagination.com
Imagination is more important than knowledge.
--
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Re: [music-dsp] Pointers for auto-classification of sounds?
You haven't really explained which aspect of the timbre you want to use to organise the sounds, and timbre is such a catch-all word that you need to specify the characteristics you are looking for in more detail to have any chance of producing something useful. Categorising by amplitude envelope would be pretty straight forward since amplitude over time is something you can easily measure and process before judging it against some metric or heuristic. You would need to pick similar aspects of the timbre, things you can quantatively measure, in order to develop a scheme for categorisation. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Charles Turner Sent: 08 June 2012 18:36 To: music-dsp@music.columbia.edu Subject: [music-dsp] Pointers for auto-classification of sounds? Hi all- 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. Here's my project: say I have a collective of sound files, all short and the same length, say 1 second in length. I want to classify them according to timbre via a single characteristic that I can then organize along one axis of a visual graph. The files have these other properties: . Amplitude envelope. I don't need to classify by time characteristic, but samples could have different characteristics, ranging from complete silence, to a classical ADSR shape, to values pegged at either +-100% or 0% amplitude. . Timbre. Samples could range in timbre from noise to (hypothetically) a pure sine wave. Any ideas on how to approach this? I've looked at a few papers on the subject, and their aims seem somewhat different and more elaborate than mine (instrument classification, etc). Also, I've started to play around with Emmanuel Jourdain's zsa.descriptors for Max/MSP, mostly because of the eazy-peazy environment. But what other technology (irrespective of language) might I profit from looking at? Thanks so much for any interest! Best, Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] maintaining musicdsp.org
Maybe submissions should be added to a moderation queue rather than added directly (i.e. they need to be manually whitelisted). I don't think a super quick turnaround on new algorithm submissions is really important for something like musicdsp.org. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Bram de Jong Sent: 04 April 2012 16:07 To: A discussion list for music-related DSP Subject: Re: [music-dsp] maintaining musicdsp.org On Wed, Apr 4, 2012 at 3:31 PM, Bastian Schnuerle bastian.schnue...@silberstein.de wrote: what is exactly the roadmap and tasks to do ? i think i could find some helping hands for you, including mine .. maybe altogether we find a way to get some work away from you ? oh there is absolutely no roadmap! it just needs some love to stop the spammers from submitting DSP algorithms (about well known drugs and handbags I won't describe here lest I end up in spam filters). and, if anyone has some new/fresh ideas for it and someone else feels like implementing those, always welcome! - bram -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] maintaining musicdsp.org
lol wow -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Bram de Jong Sent: 04 April 2012 16:15 To: A discussion list for music-related DSP Subject: Re: [music-dsp] maintaining musicdsp.org On Wed, Apr 4, 2012 at 5:11 PM, Thomas Young thomas.yo...@rebellion.co.uk wrote: Maybe submissions should be added to a moderation queue rather than added directly (i.e. they need to be manually whitelisted). I don't think a super quick turnaround on new algorithm submissions is really important for something like musicdsp.org. they ARE added to a queue. the queue now contains about 500 spam submissions. that's the whole (current) problem. some kind of report as spam thing for the comments would be nice too as there are SOME (but few) spam comments. - bram -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a little about myself
Ringing in your ears due to exposure to loud noise is the stereocilia (small hair cells) being damaged and falsely reporting to your brain that there is still sound vibration present. The frequency of the ringing is not a function of the sound that damaged your ears (a super loud bassy sound doesn't cause a bassy ringing in your ears). Tom -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of David Olofson Sent: 23 February 2012 16:15 To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] a little about myself On Thursday 23 February 2012, at 17.10.52, David Olofson da...@olofson.net wrote: On Thursday 23 February 2012, at 16.17.38, Adam Puckett adotsdothmu...@gmail.com wrote: Interesting. How would you make an ear ringing sound? [...] A more realistic approach might be to FFT the offending audio snippet that triggers the effect, remove all bands but the too loud peaks and then IFFT that to create the ring waveform. Oh, wait. Probably throw in some heavy distortion before the FFT, approximating the mechanical distortion in the ear...? Also, the whole thing needs to take in account that the too loud level is highly frequency dependent. Nothing is ever nearly as simple as it may seem at first. ;-) -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '-' -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Introducing myself (Alessandro Saccoia)
float pan = sin(2 * PI * frequency * time++ / 44100); As 'time' increases, changes to 'frequency' will result in larger and larger discontinuities. You should offset (add) the change in time rather than multiplying by it. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Bill Moorier Sent: 23 February 2012 18:05 To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] Introducing myself (Alessandro Saccoia) Thanks Alessandro! Unfortunately I don't think this is the problem though. I added a simple moving average on the parameter and it didn't make the nasty artifacts go away. So I rewrote the whole thing as a VST so I can post more code without having to reveal my messy in-progress javascript framework ;) Here it is: http://abstractnonsense.com/23-feb-2012-cc.html So if I sweep the parameter smoothly with this VST loaded in Ableton, I get nasty artifacts as the sweep is happening. I can see from the trace that the parameter isn't changing too quickly, and the resulting frequency variable always changes by less than 0.01Hz between samples. But the pan variable ends up changing by a lot - often nearly as much as 0.1 between samples! Thinking about it some more, I'm getting suspicious of the code that writes to the pan variable - it's supposed to be an LFO. The thing that makes me suspicious is, if we change the frequency, even very slightly, between samples then we're gluing together two different sine waves. But we're not taking into account phase! If the time variable gets big (which it will), then there doesn't seem to be any reason to believe that two slightly-different-frequency sine waves would have almost the same values at the current time. The ends of the sine waves might not match up when we glue them together! Am I on the right track with this line of thinking? Is there a better way to write a variable-frequency LFO than just: float pan = sin(2 * PI * frequency * time++ / 44100); Thanks again, Bill. Hello Bill, I take your question as a chance to introduce myself. When you sweep the input parameter you are introducing discontinuities in the output signal, and that sounds awful. The simplest case to figure that out in your code is imagining that you have the input variable set at 0 (pan = 0), and then abruptly you change the input parameter to a value that will make the value of the pan variable jump to 1. Both of the channels will generate a square wave, that will sound badly because of the aliasing. One solution is to control the slew rate of your parameter lowpass filtering your parameter. A simple moving average filter should do the job correctly. I have been reading this newsletter for a couple of years now, and I think that it's the best place to learn about the practical applications of musical dsp. I have been working in the digital audio field since 3 years now, even though I have been interested in computer music since my first years at the university. Now I am freelancing in this field, and I also get to play music more often. This is really stimulating my imagination, and I hope that in the next months I will have the time to implement some new effect or instruments. Thank you for all the nice things that I have learnt here, Alessandro -- Bill Moorier abstractnonsense.com | @billmoorier -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a little about myself
I've tried that before, basically writing a soft synth and a sequencer. I won't pretend I made anything that sounded very good but it was fun. In the demoscene there is a whole field of 'programmed music' which is just this, although generally they use a sequencer for the composition and just code the instruments. I have to say I've never really seen the appeal of CSound since it seems as complicated as just writing your own code in C/C++, maybe I am missing some great use case for it. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Adam Puckett Sent: 22 February 2012 13:45 To: A discussion list for music-related DSP Subject: Re: [music-dsp] a little about myself It's nice to see some familiar names in Csound's defense. Here's something I've considered since learning C: has anyone (attempted to) compose music in straight C (or C++) just using the audio APIs? I think that would be quite a challenge. I can see quite a bit more algorithmic potential there than probably any of the DSLs written in it. On 2/21/12, Michael Gogins michael.gog...@gmail.com wrote: It's very easy to use Csound to solve idle mind puzzles! I think many of us, certainly myself, find ourselves becoming distracted by the technical work involved in making computer music, as opposed to the superficially easier but in reality far more difficult work of composing. Regards, Mike On Tue, Feb 21, 2012 at 7:53 PM, Emanuel Landeholm emanuel.landeh...@gmail.com wrote: Well. I need to start using csound. To actually do things in the real world instead of just solving idle mind puzzles. On Tue, Feb 21, 2012 at 10:02 PM, Victor victor.lazzar...@nuim.ie wrote: i have been running csound in realtime since about 1998, which makes it what? about fourteen years, however i remember seeing code for RT audio in the version i picked up from cecelia.media.mit.edu back in 94. So, strictly this capability has been there for the best part of twenty years. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- Michael Gogins Irreducible Productions http://www.michael-gogins.com Michael dot Gogins at gmail dot com -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Signal processing and dbFS
Most people seem to be overthinking this, Uli has posted the important equation here: Y [dBFS] = 20*Log10(X/FS) That is all you need for converting normalised peak values to dbFS. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Uli Brueggemann Sent: 18 January 2012 08:20 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Signal processing and dbFS My simple point of view about dBFS (full scale): The full scale FS of a 16 bit soundcard is 2^15=32768, of a 24 bit soundcard it is 2^23 = 8388608. The dB number Y of a value X represents a relation to the full scale: Y [dBFS] = 20*Log10(X/FS) So with X=32786 and a 16 bit soundcard you get Y = 20*Log10(32768/32768) = 20*Log10(1) = 20*0 = 0 dBFS The result 32768/32768 = 1.0 (or 8388608/8388608) explains, why the DSP maths often use values in the range -1.0 .. +1.0. Of course with floating point numbers you can also use higher values. At least until you try to send the result to a soundcard. Converting a float number to an integer value by multiplication with the resolution of the soundcard will lead to clipping if you have values exceeding the full scale. All this consideration has nothing to do with the output voltage of the soundcard. Thus 0 dBFS may result in a level of -10 dBV or + 4dBu or whatever the analog reference level is. In practical applications it is always necessary to avoid clipping in the digital domain and in the analog domain. This means to leave some headroom. E.g if you define a headroom of 12 dB for yourself as optimal then you may arbitrarily define -12 dBFS as 0 dBLS (Linda S). That's why there is all the confusion with different definitions around. Especially if you forget to mention LS and talk about 0 dB only instead of 0 dBLS :-) Uli On Wed, Jan 18, 2012 at 7:34 AM, Linda Seltzer lselt...@alumni.caltech.edu wrote: Good evening from snowy Washington State, Those who know me personally know that my background in music DSP is via the ATT Labs / Bell Labs signal processing culture. So for the first time I have encountered the way recording studio engineers think about measurements. They plot frequency response graphs using dbSF on the Y axis. Recording engineers are accustomed to the idea of dbSF because of the analog mixes where the needle looked like it was pegging at 0dB. The problem is what this means in signal processing. My understanding is that different manufactureres have different voltage levels that are assigned as 0dB. In signal processing we are not normally keeping track of what the real world voltage level is out there. When I am looking at a signal in Matlab it is a number in linear sampling from the lowest negative number to the highest negative number. Or it is normalized to 1. I don't know how to translate from my plot of a frequency response in Matlab to a graph using dbFS. Visually a signal processing engineer wouldn't think about the peaks as much as a recording engineer sitting at a mixing console does. I can see that from their point of view they are worrying about the peaks because of worries about distortion. A signal processing engineer assumes that was already taken care of by the times the numbers are in a Matlab data set. The idea of plotting things based on full scale is a bit out of the ordinary in signal processing because many of our signals never approach full scale. Telephone calls are different from rock music. Any comments on this issue would be greatly appreciated. Linda Seltzer -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Signal processing and dbFS
I'm just highlighting Uli's post which I think contains the answer to her main question of how to convert a normalised matlab plot into dbFS. I don't mean to denigrate what other people have said. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Nigel Redmon Sent: 18 January 2012 15:58 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Signal processing and dbFS If we're overthinking it, it's probably because we're not sure what Linda's after. I'm sure she knows the formula for voltage conversion to dB. On Jan 18, 2012, at 2:34 AM, Thomas Young wrote: Most people seem to be overthinking this, Uli has posted the important equation here: Y [dBFS] = 20*Log10(X/FS) That is all you need for converting normalised peak values to dbFS. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia?
I'd like to say well done to everyone who has edited this so far, it looks massively better :) -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert bristow-johnson Sent: 16 January 2012 16:16 To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] anyone care to take a look at the Additive synthesis article at Wikipedia? On 1/16/12 1:16 AM, Nigel Redmon wrote: Nice improvements. This may seem like nitpicking, but the Timeline of additive synthesizers section seems to choose keeping the instrument name as the start of the sentence over proper grammar. For instance: Hammond organ, invented in 1934[26], is an electronic organ that uses nine drawbars to mix several harmonics, which are generated by a set of tonewheels. This should either read The Hammond organ, invented in 1934[26], is an electronic organ that uses... or something like Hammond organ-invented in 1934[26], the Hammond organ is an electronic organ that uses... or Hammond organ: Invented in 1934[26], the Hammond organ is an electronic organ that uses... (Note that one entry, EMS Digital Oscillator Bank (DOB) and Analysing Filter Bank: According to..., does it this way already.) You have enough cooks working on that page right now, so I'd rather leave it up to you guys what route you go. But if you use a sentence, it should read like one. well, there is evidence that Clusternote is from Japan. dunno if these sentences were written by him. i wouldn't discourage you from editing at all. L8r, -- 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 http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Splitting audio signal into N frequency bands
'Biquad' is a filter topology meaning the transfer function is expressed as a quadratic divided by another quadratic (biquadratic). They are pretty common in the world of digital filters because there are very simple to convert into an efficient algorithm (Direct form 1 etc...). See http://en.wikipedia.org/wiki/Digital_biquad_filter A Butterworth filter has a particular transfer function, see http://en.wikipedia.org/wiki/Butterworth_filter. I think there are a few implementations in the music DSP archives. I think David made a good point that you don't really want an aggressive cut off with your filter, since sounds which straddle the crossover frequencies can suffer quite badly. As he suggests using simpler, less aggressive filters (e.g. pretty much any first or second order iir filter) will probably be acoustically a lot nicer. Thomas -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thilo Köhler Sent: 02 November 2011 12:10 To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] Splitting audio signal into N frequency bands Hello Thomas, Wen! Thank you for the quick input on this. 1. I found that in the 3-band case, splitting up the low and high band from the input and then generating the mid band by subtracting them works much better than the salami stategy (chopping off slices with a LP). Thanks! 2. Subtracting the LP part makes sense only if the LP filter is zero-phase. I dont know if my filters are zero phase, I am not that deep into the filter math to tell you straight away. It is an IIRC taken from here: http://www.musicdsp.org/showArchiveComment.php?ArchiveID=259 This one seems to work best for my purposes, but that is just from subjective listening wihtout any mathematical evidence. Is this a butterworth filter like Thomas suggests? (sorry if the question sounds like a noob...) In the comment they call it biquad, i dont know if a biquad can be butterworth or this is mutual exclusive. I have also tried: http://www.musicdsp.org/showArchiveComment.php?ArchiveID=266 Doesnt work well for low cutoff frequencies, like 150Hz. I am using single precision. http://www.musicdsp.org/showArchiveComment.php?ArchiveID=117 Seems to be too flat, not steep enough. http://www.musicdsp.org/showArchiveComment.php?ArchiveID=237 Seems to be too flat, not steep enough. I think in the use case of a mulit-band compressor, perfect reconstruction is important. That is my I want to create the band by subtracting and not with independent filters. I assume this is a good strategy, no? Regards, Thilo I believe the typical way is to directly construct a series of steep band-pass filters to cover the whole frequency range. This is very flexible but usually means the individual parts do not accurately add up to the original signal. On the other hand, if perfect sum is desirable you may wish to take a look at mirror filters, such as QMF. These are pairs of LP and HP filters designed to guarantee perfect reconstruction. -- From: Thilo K?hler koehlerth...@gmx.de Sent: Monday, October 31, 2011 10:47 AM To: music-dsp@music.columbia.edu Subject: [music-dsp] Splitting audio signal into N frequency bands Hello all! I have implemented a multi-band compressor (3 bands). However, I am not really satisfied with the splitting of the bands, they have quite a large overlap. What I do is taking the input singal, perfoming a low pass filter (say 250Hz) and use the result for the low band#1. Then I subtract the LP result from the original input and do a lowepass again with a higher frequency (say 4000Hz). The result is my mid band#2, and after subtracting again the remaining signal is my highest band#3. I assume this proceedure is appropriate, please tell me otherwise The question is now the choise of the filter. I have tried various filters from the music-dsp code archive, but i still havent found a satisfiying filter. I need a steep LP filter (12db/oct or more), without resonance and fewest ringing possible. The result subtracted from the input must works as a HP filter. Are there any concrete suggestions how such a LP filter should look like, or is there even a different, better way to split the audio signal into 3 bands (or N bands)? I know I can use FFT, but for speed reasons, I want to avoid FFT. Regards, Thilo Koehler -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Splitting audio signal into N frequency bands
I think you will get better results doing a lowpass highpass, then subtracting those from your original signal to get a middle band. For the filter I would have thought you want something as 'clean' as possible, i.e. no ripple, flat as possible in the passband and with a linear falloff (it shouldn't need to be particularly steep, but 12db/oct does sound about right). For me a Butterworth springs straight to mind, second order will give you 12db/oct. It's pretty cheap as well if that is a concern. I am interested to know how other people do the splitting into bands as well though, I plan on implementing a multiband compressor shortly myself. Multiple bandpass filters would seem logical, but there must be a bit of an issue with colouring the signal. Thomas Young -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thilo Köhler Sent: 31 October 2011 10:47 To: music-dsp@music.columbia.edu Subject: [music-dsp] Splitting audio signal into N frequency bands Hello all! I have implemented a multi-band compressor (3 bands). However, I am not really satisfied with the splitting of the bands, they have quite a large overlap. What I do is taking the input singal, perfoming a low pass filter (say 250Hz) and use the result for the low band#1. Then I subtract the LP result from the original input and do a lowepass again with a higher frequency (say 4000Hz). The result is my mid band#2, and after subtracting again the remaining signal is my highest band#3. I assume this proceedure is appropriate, please tell me otherwise The question is now the choise of the filter. I have tried various filters from the music-dsp code archive, but i still havent found a satisfiying filter. I need a steep LP filter (12db/oct or more), without resonance and fewest ringing possible. The result subtracted from the input must works as a HP filter. Are there any concrete suggestions how such a LP filter should look like, or is there even a different, better way to split the audio signal into 3 bands (or N bands)? I know I can use FFT, but for speed reasons, I want to avoid FFT. Regards, Thilo Koehler -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] FM Synthesis
I don't think musicians mind tweaking and poking presets (what's the worst that can happen?), so it's really up to the makers to provide plenty of decent preset sounds. NI FM8 for example has a pretty good selection of presets and is a popular FM synth these days, I rate it pretty highly myself. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Didier Dambrin Sent: 14 September 2011 11:19 To: A discussion list for music-related DSP Subject: Re: [music-dsp] FM Synthesis I believe FM sounds got back in fashion at the time everyone had forgotten that shitty GM FM bank in Windows, when FM meant cheesy MIDI files. ..but the problem with FM is that no one can program presets, it's very unpredictable when you deal with more than 3 operators, and I'm not sure today's musicians wanna deal with such complexity anymore. Actually, if the story that most DX7s never got their patches tweaked is true, maybe no musician ever wanted to deal with such complexity. On 13/09/2011 21:06, Theo Verelst wrote: Hi .. Remember that Frequency Modulation of only two operators already has theoretically (without counting sampling artifacts!) a Bessel-function guided spectrum which is in every case infinite, although at low modulation indexes the higher components are still small. Also think about the phase accuracy: single precision numbers are not good at counting more samples than a few seconds for instance. Not too much of a problem if you use table lookup, which is what I assume the DX 7 did. Phase errors are a problem in single precision if you compute and accumulate. .. Oh, there are Open Source FM synths maybe worth looking at: a csound script (or what that is called there) Csound has the foscil two-operator self-contained opcode, and of course you can roll your own operator structures ad lib. Somewhere there is a full DX 7 emulation complete with patches (poss in the Csound book; not to hand right now). Have we now reached the point where FM sounds are back in fashion? Richard Dobson -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] FM Synthesis
Interesting. I was referring to 'propper' synths really, which I wouldn't really group with romplers and black boxes personally, but I guess end users don't necessarily make the same distinction. I can definitely Personally I'm not making black boxes, because to do this you can't just be a programmer, you also have to be an artist That's part of the fun of working with music DSP though, you need a bit of a creative touch, I would have thought most people on this list would agree. Obviously making black boxes is taking it to the extreme though. preset designers [...] Probably not very well paid Heh, well yea, that's the problem with being a programmer, you can never really move into a 'creative' job without suffering major pay decimation :P Out of curiosity do you program soft or hard synths? I'd be interested to know if it is easy to get into professional soft synth/effect industry and if it pays well. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Didier Dambrin Sent: 14 September 2011 13:49 To: A discussion list for music-related DSP Subject: Re: [music-dsp] FM Synthesis It's totally ok to make presets yourself with a beta team, as long as the full editor is given to the user (so that he or others can do presets himself), but when it's a rompler or a black box, you really need artists to be using your private tools. Also it's pretty hard to keep backwards compatibility during the first 90% of the dev of a plugin, so yes you make a lot of sounds during that process, but they will be lost. Well that's in my case. Personally I'm not making black boxes, because to do this you can't just be a programmer, you also have to be an artist because you will be making hard-coded choices, so you'd better know what you're doing (I mean: a parameter that doesn't end up on the GUI is a choice that the programmer had to make. If it's a bad choice, the artist using it will have to do with it). I prefer to give as much power as possible, even if some features end up not being used. Then for the next synth/effect I can still only implement what has been used in the previous stuff. But what's sure is that a lot of musicians don't like complex synths. I always suspected most of the presets just fell out of the development process, you are always going to need to test that your engine sounds good (synth construction is 50% science and 50% magic after all) and that surely generates a whole load of sounds. Polish those up and you already have a load of presets, let your QA/Interface designers have a bash while they are developing as well and you have yourself a whole suite! I could be totally wrong of course. some kind of middleware guys, in-between musicians engineers, who program presets for engines And where can I apply for this job? ;) -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Didier Dambrin Sent: 14 September 2011 11:32 To: A discussion list for music-related DSP Subject: Re: [music-dsp] FM Synthesis The evolution seems to be going towards some kind of middleware guys, in-between musicians engineers, who program presets for engines put in black box instruments, so that the musican can get access to non-sampled FM presets, while not having access to FM at all. Well, seems to be what NI is doing as well. It already happened with samplers, in the past the musician got a sampler, that he could either program himself or feed with soundbanks. These days a musician buys a rompler, doesn't have access to the editor (which obviously has to exist for the middleware guy who made the soundbank). I don't think musicians mind tweaking and poking presets (what's the worst that can happen?), so it's really up to the makers to provide plenty of decent preset sounds. NI FM8 for example has a pretty good selection of presets and is a popular FM synth these days, I rate it pretty highly myself. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Didier Dambrin Sent: 14 September 2011 11:19 To: A discussion list for music-related DSP Subject: Re: [music-dsp] FM Synthesis I believe FM sounds got back in fashion at the time everyone had forgotten that shitty GM FM bank in Windows, when FM meant cheesy MIDI files. ..but the problem with FM is that no one can program presets, it's very unpredictable when you deal with more than 3 operators, and I'm not sure today's musicians wanna deal with such complexity anymore. Actually, if the story that most DX7s never got their patches tweaked is true, maybe no musician ever wanted to deal with such complexity. On 13/09/2011 21:06, Theo Verelst wrote: Hi .. Remember that Frequency Modulation of only two operators already has theoretically (without counting
Re: [music-dsp] FM Synthesis
Regarding: buffer[i] = Math.sin( ( phase * ratio * frequency + buffer[ i ] ) * PI_TWO ); // FM (PM whatever) phase += 1.0 / 44100.0; // advance phase in normalized space Your oscillator will be prone to drifing out of phase due to the way you are adding the reciprocal of the sample rate (a recurring decimal) to a variable each sample. phase += 1.0 / 44100.0; ( 0.226757369...) This error will accumulate and become significant after a large number of samples, it may not be your problem here but it would still be worth using a different mechanism for calculating it (i.e. use (i / SampleRate)) Don't you also want + buffer[ i ] after the * PI_TWO? I might just be missing something there though. Tom Young -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Andre Michelle Sent: 12 September 2011 14:59 To: music-dsp@music.columbia.edu Subject: [music-dsp] FM Synthesis Hi all, I just started an implementation of a FM-Synth and I am struggling with a certain effect, which I am not sure, if it is caused by a bug or conceptional error. The structure is straight-forward. I have N operators, which can be freely routed including feedback. Each Oscillator has one buffer to write its processing result. The next operators in the processing chain can use the buffer as modulation input. I compare the output with NI FM8 to check if I get similar output. So far very close. The issue occurs, when I choose a routing with feedback. A B A, where B's output will be send to the sound card. If I choose a blocksize (samples each computation) of one, I get reasonable audio at my output (similar to sawtooth). However running a blocksize of e.g. 64 samples or even just more than 1 sample results in some nasty glitchy sound. Looks to me like a phase problem, but I am pretty sure, that my code covers all that. My question is: Is it even possible to let the operators process entire audio-blocks independently? I guess, later processing of LFOs, envelopes and everything each sample might not lead to readable code and could not be easily optimized. my Java code of a single Operator: http://files.andre-michelle.com/temp/OperatorOscillator.txt For the routing above (ABA, out:B) the linear process sequence is A,B,Output (while B,A,Output would also be nice) Any hints? Thanks! -- André Michelle http://www.audiotool.com -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Electrical Engineering Foundations
If the goal here is education then I would suggest formatting your work as a syllabus or structured as lessons with exercises. The raw information is already available through books and sites like Wikipedia (where the wording and presentation is already carefully reviewed), so I would have thought the focus would be on presenting the reader with the information in what you consider to be the 'correct' order, from the fundamentals upward. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo Verelst Sent: 27 August 2011 23:11 To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] Electrical Engineering Foundations I've made a small beginning at http://www.theover.org/Dsp . Feel free to comment/request/fire question, etc, in fact I don't even mind making it a Wiki page (to let other contribute), and I don't know yet how many linked pages and example materials I'll make. Theo -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Multichannel Stream Mixer
Is there something wrong with just summing the buffers for each channel?
It's not really clear what you are trying to achieve, do you want to downmix
your channels?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of StephaneKyles
Sent: 30 August 2011 14:05
To: 'A discussion list for music-related DSP'
Subject: Re: [music-dsp] Multichannel Stream Mixer
FramesRecorder(i,0) is a class pointer with access to i the sample number,
0 is the channel.
Variable vol, is just an example to assign volume scale to the samples.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Wen Xue
Sent: Tuesday, August 30, 2011 2:18 PM
To: 'A discussion list for music-related DSP'
Subject: Re: [music-dsp] Multichannel Stream Mixer
Is FramesRecorder(...) a macro or a function call?
And why multiply vol=1.0 when everything is floating-point already?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of StephaneKyles
Sent: 30 August 2011 12:13
To: 'A discussion list for music-related DSP'
Subject: [music-dsp] Multichannel Stream Mixer
Hi, I am trying to implement a multi channel mixer in c++, very basic. I
have multiple streams and their associated buffer.
im using rtaudio and stk to work with the samples and audio devices.
I gather the buffers in a tick method, and fetch all the samples to produce
the sum :
FramesRecorder is the final output buffer. -1.0f to 1.0f
Frames[numplayers] are the stream buffers. -1.0f to 1.0f
int numplayers, the stream index.
the sines are normalized to -1.0 to 1.0 , float32.
for ( unsigned int i=0; i nBufferFrames ; i++ ) {
Float vol = 1.0f;
FramesRecorder(i,0) = ( FramesRecorder(i,0) + (
Frames[numplayers](i,0) * vol )) ;
FramesRecorder(i,1) = ( FramesRecorder(i,1) + (
Frames[numplayers](i,1) * vol )) ;
}
// and something like giving the sum of streams.
FramesRecorder[0] /= numplayers;
FramesRecorder[1] /= numplayers;
what would be the best method to mix multi channel buffers together ? I know
it's a newbie question, but I would like to know about your experience with
this. I would like to implement an efficient method as well. Thx !
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Re: [music-dsp] Electrical Engineering Foundations
What resources would you recommend Theo? -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo Verelst Sent: 24 August 2011 17:01 To: music-dsp@music.columbia.edu Subject: [music-dsp] Electrical Engineering Foundations Just to maybe put some people at ease and hopefully arousing some discussions, I'd like to point the attention of a lot of people in the DSP corners of recreation and science, hobby and serious research to the general foundations for Sampling Theory and Digital Signal Processing and possibly information theory and filter knowledge. Of course there is little against a lot of gents (and possibly gals) plowing around in the general subject of doing something with audio and a computer. It appears however that for the reasonably reasons such as required IQ and education (or for fun) but also for less reasonable reasons such as power pyramids and greed, the theoretical foundations of even the simplest thing of all, playing a CD or soundfile through a DA converter for many remain rather shaky or plain wrong or simply absent. For analog electronics filters and such are well known subjects applicable from the simple to the quite complex and the electronics world generally tends to have a solid foundation for dealing with the issues and theories involved, though not necessarily all electronicists are on par with the required network theoretical foundations, I've understood MNA methods in the US are even taught at high school level. For EEs (electrical engineers) it will usually be the case they've heard about sampling theory somewhere in their education, and idem ditto for some filter or circuits and systems type of theory. Many others, like in IT (informatics), even physics and other disciplines or people making their hobby a profession this might all just as well be abacadabra, and it seems to me this should change. Regards, Theo Verelst -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] New patent application on uniformly partitioned convolution
In principle a patent protects the investment that a company makes in order to develop a new technology; companies are unlikely to invest large amounts on research if their ideas are going to be copied and their product beaten to market by an opportunistic competitor. In practice, as we all know, things like software patents are just food for patent trolls. -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Victor Lazzarini Sent: 28 January 2011 18:07 To: A discussion list for music-related DSP Subject: Re: [music-dsp] New patent application on uniformly partitioned convolution I would have thought that the whole point of a patent is to make money. A scientific paper, IMHO, is the way to move the field forward. Victor On 28 Jan 2011, at 18:02, Dave Hoskins wrote: The whole point of a Patent is to help engineers move forward, so it's completely legitimate to take a previous invention and add to it, to make a new Patent. It's a shame they are not seen like this at all. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Factorization of filter kernels
Hi Uli I don't know if this will be useful for your situation, but a simple method for decomposing your kernel is to simply chop it in two. So for a kernel: 1 2 3 4 5 6 7 8 You can decompose it into two zero padded kernels: 1 2 3 4 0 0 0 0 0 0 0 0 5 6 7 8 And sum the results of convolving both of these kernels with your signal to achieve the same effect as convolving with the original kernel. You can do this because convolution is distributive over addition, i.e. f1*(f2+f3) = f1*f2 + f1*f3 For signals f1,f2 f3 (* meaning convolve rather than multiply). Obviously all those zero's do not need to be evaluated, meaning the problem is changed to one of offsetting your convolution algorithm (which may or may not be practical in your situation), but does allow you to use half the number of coefficients. Thomas Young Core Technology Programmer Rebellion Developments LTD -Original Message- From: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Uli Brueggemann Sent: 19 January 2011 14:56 To: A discussion list for music-related DSP Subject: Re: [music-dsp] Factorization of filter kernels Hi, thanks for the answer so far. A polyphase filter is a nice idea but it does not answer the problem. The signal has to be demultiplexed (decimated), the different streams have to be filtered, the results must be added to get the final output signal. My question has a different target. Imagine you have two system (e.g. some convolution boards with DSP). Each system can just run a 512 tap filter. Now I like to connect the two systems in series to mimic a desired 1024 tap filter. The 1024 kernel is known and shall be generated by the two 512 tap filters. So what's a best way to decompose the known kernel into two parts ? Is there any method described somewhere? Uli 2011/1/19 João Felipe Santos joao@gmail.com: Hello, a technique that allows something similar to what you are suggesting is to use polyphase filters. The difference is that you will not process contiguous vectors, but (for a 2-phase decomposition example) process the even samples with one stage of the filter and the odd samples with another stage. It is generally used for multirate filter design, but it makes sense to use this kind of decomposition if you can process the stages in parallel... or at least it is what I think makes sense. You can search for references to this technique here [1] and here [2]. A full section on how to perform the decomposition is presented on Digital Signal Processing: a Computer-based approach by Sanjit K. Mitra. [1] http://www.ws.binghamton.edu/fowler/fowler%20personal%20page/EE521_files/IV-05%20Polyphase%20FIlters%20Revised.pdf [2] https://ccrma.stanford.edu/~jos/sasp/Multirate_Filter_Banks.html -- João Felipe Santos On Tue, Jan 18, 2011 at 5:46 AM, Uli Brueggemann uli.brueggem...@gmail.com wrote: Hi, a convolution of two vectors with length size n and m gives a result of length n+m-1. So e.g. two vectors of length 512 with result in a vector of length 1023. Now let's assume we have a vector (or signal or filter kernel) of size 1024, the last taps is 0. How to decompose it to two vectors of half length? The smaller vectors can be of any arbitrary contents but their convolution must result must be equal to the original vector. It would be even interesting to factorize given kernel into n smaller kernels. Again the smaller kernels may have any arbitrary but senseful contents, they can be identical but this is not a must. Is there a good method to carry out the kernel decomposition? (e.g. like calculating n identical factors x of a number y by x = Exp(Log(y)/n) with x^n = x*x*...*x = y) Uli -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
