Re: [music-dsp] Thoughts on DSP books and neural networks
On Thu, Feb 5, 2015 at 5:10 AM, Peter S peter.schoffhau...@gmail.com wrote: On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: I would probably listen to both, if both were sharing information, and in a helpful way. I’m not implying that you aren’t (I don’t know—I’ve seen some of your posts here, but don’t have a lot of experience with you). I do know that Robert has published many helpful and thought-provoking papers, and has been doing the same in this forum and comp.dsp for many years. I’m always interested in what he has to say. My only problem is - when I say something that is not a traditional approach, why is the first response of some guys here No, you're an idiot! That's not how we do it! Crackpot! Idiot! Go home and read some beginner books! All your mail will be sent to thrash! Why do you always take things the wrong way? Don't take this the wrong way. I don't see how anyone's trying to offend you. Have fun, why else do we do this if it's not, at least, fun. -- 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] Thoughts on DSP books and neural networks
On 2/5/15 6:10 AM, Peter S wrote: ... as I'll die some day and those things that I invented and are in my head will go to the grave with me, welcome to the club. and future generations will need to reinvent all that knowledge. i wouldn't assume that for me. them future generations might not give a rat's ass what i think. -- 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
Re: [music-dsp] Thoughts on DSP books and neural networks
On 2/5/15 2:25 PM, Ethan Duni wrote: 'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. Sorry, I should have written at z=0, not infinity. Typing too fast at the end of the day. Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. Right. So can the cookbook filters be made to put one of the zeros at the origin, or do they always produce responses with two non-trivial zeros? i don't think any of the cookbook filters have a zero at the origin. sometimes zeros at DC or Nyquist. the cookbook EQ is not general. even if you toss in an additional constant gain, it's 4 knobs. Knud Christensen tossed in this additional symmetry parameter an that made it 5 knobs controlling independent things. an Orfanidis parametric EQ is a subset of the Christensen model. of course, the cookbook parametric (or the cookbook shelves) is a subset. but just because of the 5 degree of freedom concept, i am pretty sure that the generalized biquad http://www.google.com/patents/WO2004054099A1?cl=en can have identical transfer function and frequency response of any LTI 2nd-order digital filter. Knud's five parameters injectively map to five coefficients. -- 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
Re: [music-dsp] Thoughts on DSP books and neural networks
'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. Sorry, I should have written at z=0, not infinity. Typing too fast at the end of the day. Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. Right. So can the cookbook filters be made to put one of the zeros at the origin, or do they always produce responses with two non-trivial zeros? E On Thu, Feb 5, 2015 at 11:00 AM, Peter S peter.schoffhau...@gmail.com wrote: On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: To your first comment—well, of course. But maybe you’ve lost site of the context—here’s your comment: My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. Ethan pointed out that they can be, because one-zero is equivalent to a special case of two-zero. (Some would argue that if it’s a second-order filter, with two poles, then it always has matching zeros, even if one or both at at the origin. shrug.) Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. On 05/02/2015, Ethan Duni ethan.d...@gmail.com wrote: You just stick the extra zero(s) off at z=infinity. Does 'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. - Peter -- 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] Thoughts on DSP books and neural networks
On 2/5/15 5:00 PM, Ethan Duni wrote:
P.S. Anyone who knows how to effectively turn ideas into money while
everyone can benefit, let me know. Patenting stuff doesn't sound like
a viable means to me.
Well, that's exactly what patents are for. I'm not sure why you don't
consider that viable. Is it to do with the costs and time required to file
a patent?
Absent a patent filing, you're stuck choosing between keeping your work
undisclosed as a trade secret, or publicizing it without any means of
collecting licensing or other revenues. That's why patents were introduced,
to cut through that knot and allow inventors to profit while still
disclosing their findings.
Note that if you decide to retain your work as a trade secret, there is no
legal barrier for others to reverse engineer it and use it without paying
you a dime. IANAL but someone could even come up with it independently and
patent it themselves.
patents are expensive. getting a patent attorney to help costs
something like $20K. enforcing infringements of the patent is laborious
(discovering the infringement) and costly (legal action). publishing in
the patent can teach a competitor how you do it and this competitor uses
that information and adds his/her own twist to it and gets something
better and different enough that you can't sue them for infringement. i
won't tell you how many patented IVL ideas i have made better after
first learning them. ('cuz it's a trade secret!) some of those IVL
ideas i had already had. IVL *did* patent some obvious ideas and some
prior art, IMHO. i think their Vocalist patent has run out, i dunno.
maybe even the independent tuning on pitch shift and formant shift,
maybe that one has also run out. maybe the turn the keyboard player's
voice into a harmonizing background group of singers, maybe that patent
has run out. they try new patents as a way to extend the old ones.
--
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
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Re: [music-dsp] Thoughts on DSP books and neural networks
P.S. Anyone who knows how to effectively turn ideas into money while everyone can benefit, let me know. Patenting stuff doesn't sound like a viable means to me. Well, that's exactly what patents are for. I'm not sure why you don't consider that viable. Is it to do with the costs and time required to file a patent? Absent a patent filing, you're stuck choosing between keeping your work undisclosed as a trade secret, or publicizing it without any means of collecting licensing or other revenues. That's why patents were introduced, to cut through that knot and allow inventors to profit while still disclosing their findings. Note that if you decide to retain your work as a trade secret, there is no legal barrier for others to reverse engineer it and use it without paying you a dime. IANAL but someone could even come up with it independently and patent it themselves. E On Thu, Feb 5, 2015 at 5:15 AM, Peter S peter.schoffhau...@gmail.com wrote: The sad fact is 1) I really don't have time to write papers or books. I know from experience, that both takes a lot of time, even writing a single DSP paper properly will take days to complete. 2) Writing a book to a very small audience is simply not worth it financially (again, I know this from experience). Small niche markets are not profitable, and realistically, the DSP market is maybe just a few hundred people (or a few thousand, at max). So it's very time consuming but gives you very little profit. 3) What I would effectively be doing, is giving away my algorithms to all my competitors. Sadly, I am not an academic who gets paid an hourly rate to write papers and books, so I also have to keep business considerations in mind (that's the sad reality). Best, Peter P.S. Anyone who knows how to effectively turn ideas into money while everyone can benefit, let me know. Patenting stuff doesn't sound like a viable means to me. -- 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] Thoughts on DSP books and neural networks
What do you guys use to turn your impulse responses into fancy FFT diagrams? If you can recommend some software, I'll post some transfer curves of the 2 pole 1 zero biquad filter. - P -- 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] Thoughts on DSP books and neural networks
Ian, Thanks for the suggestion. I'll try to make some fancy graphs. I think I have Octave and Scilab installed (hm, even Mathematica). - Peter On 06/02/2015, Ian Esten i...@ianesten.com wrote: Octave or Matlab. Or even Mathematica. It would be very interesting to see the transfer function of your filter on the same graph as the 'ideal' analog filter. Ian On Thursday, February 5, 2015, Peter S peter.schoffhau...@gmail.com wrote: What do you guys use to turn your impulse responses into fancy FFT diagrams? If you can recommend some software, I'll post some transfer curves of the 2 pole 1 zero biquad filter. - P -- 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] Thoughts on DSP books and neural networks
Octave or Matlab. Or even Mathematica. It would be very interesting to see the transfer function of your filter on the same graph as the 'ideal' analog filter. Ian On Thursday, February 5, 2015, Peter S peter.schoffhau...@gmail.com wrote: What do you guys use to turn your impulse responses into fancy FFT diagrams? If you can recommend some software, I'll post some transfer curves of the 2 pole 1 zero biquad filter. - P -- 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] Thoughts on DSP books and neural networks
Fair enough, Peter. But remember that the question you posed was whether people, after hearing your algorithms, want to hear your ideas, or Robert’s. But in your reply to me, and several of your other messages to the list this morning, you explain why you won’t divulge your algorithms. That pretty much settles it that people will listen Robert then, no? Don’t get me wrong—I don’t care that you don’t divulge your work, that is your choice and right. I admire people who give their time and knowledge, but I don’t look down upon people who don’t. I give away some of my knowledge via my website, but I don’t expect anything in return—I’m just trying to give a little toe-hold for some people who are otherwise overwhelmed by the technical nature of DSP. But beyond having a good grasp of filters that people before me figured out, I have no experience with making my own. Your stuff sounds good, I’m impressed. But you are posting to a group of technical people, few of which are likely to buy your designs. So, I think the value of you showing how good your stuff sounds, then not saying why or how, is somewhat limited on this mailing list. I understand that the point was to say that you are worth being listened to, but that will only get you so far—once stated, people either decide that’s sufficient, or not, and there’s not much point in arguing it. Regards, Nigel On Feb 5, 2015, at 3:10 AM, Peter S peter.schoffhau...@gmail.com wrote: On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: I would probably listen to both, if both were sharing information, and in a helpful way. I’m not implying that you aren’t (I don’t know—I’ve seen some of your posts here, but don’t have a lot of experience with you). I do know that Robert has published many helpful and thought-provoking papers, and has been doing the same in this forum and comp.dsp for many years. I’m always interested in what he has to say. My only problem is - when I say something that is not a traditional approach, why is the first response of some guys here No, you're an idiot! That's not how we do it! Crackpot! Idiot! Go home and read some beginner books! All your mail will be sent to thrash! Don't get me wrong, my point is not that Robert is not a smart guy - I certainly know that he has a lot of knowledge and insight, a have absolutely zero doubt about that. (And he certainly knows a lot more than me about certain topics.) My point is - why that arrogant responses? I believe, that is unfair. I think I *could* publish a lot of thought-provoking papers or information, things that you currently won't find in any book or paper, but if I face such arrogance, why would I even waste my time writing those papers? Let me make this clear - techically I am not an academic, that is, no one pays me a hourly rate to write papers and books, so if there's not much incentive for me to share any of my information, and I am also faced with such high levels of arrogance, then I don't feel like sharing any of that information at all. Then I'll say - okay, if you're so smart, then just go and figure it out on your own. Which would be a pity, as I'll die some day and those things that I invented and are in my head will go to the grave with me, and future generations will need to reinvent all that 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] Thoughts on DSP books and neural networks
On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: To your first comment—well, of course. But maybe you’ve lost site of the context—here’s your comment: My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. Ethan pointed out that they can be, because one-zero is equivalent to a special case of two-zero. (Some would argue that if it’s a second-order filter, with two poles, then it always has matching zeros, even if one or both at at the origin. shrug.) Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. On 05/02/2015, Ethan Duni ethan.d...@gmail.com wrote: You just stick the extra zero(s) off at z=infinity. Does 'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. - Peter -- 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] Thoughts on DSP books and neural networks
http://morpheus.spectralhead.com/demos/ Great sounds! Keep up the good work. --- David Akbari, Au.D., CCC-A, F-AAA, F-ADA Doctor of Audiology and Systems Development Specialist IntriCon Corporation On Thu, Feb 5, 2015 at 12:24 PM, Peter S peter.schoffhau...@gmail.com wrote: On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: A two-zero filter with one fixed at the origin of the z plane is the same as a one-zero filter. It should be obvious that in the transfer function, a coefficient of 0 for a term is equivalent to that term no being there. Well, that is true, you can add as many 0-coefficient terms to the 1-zero filter as you want, since 0*x = 0. So yes, that is trivially true, you can add even a hundred more zeros at the origin that do nothing. But why would you do that, instead of simply removing all 0-coefficient terms that effectively do nothing? Realistically, anyone who uses two zeros in a filter, uses nonzero coefficients for both zeros, otherwise it's just wasted computation. At least I've seen no design which always gives a biquad coefficient of zero for one of the zeros; in the lowpass case, both zeros are always at Nyquist. Your stuff sounds good, I'm impressed. But you are posting to a group of technical people, few of which are likely to buy your designs. So, I think the value of you showing how good your stuff sounds, then not saying why or how, is somewhat limited on this mailing list. I understand that the point was to say that you are worth being listened to, but that will only get you so far--once stated, people either decide that's sufficient, or not, and there's not much point in arguing it. Fair enough, that was the only purpose. I'm not trying to sell these algorithms here (places like KVR Audio etc. are more suited to that). And maybe later I'll reveal some of my designs and/or write a few papers, but currently I don't have enough time nor incentive to do that. - Peter -- 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] Thoughts on DSP books and neural networks
The cookbook filters put the zeros at -1, the Nyquist frequency, for the low pass filter. So it’s true that the filters can’t be made the same for the cookbook formula, just not true that a two-zero filter can’t have the same transfer function as a one-zero. On Feb 5, 2015, at 11:25 AM, Ethan Duni ethan.d...@gmail.com wrote: 'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. Sorry, I should have written at z=0, not infinity. Typing too fast at the end of the day. Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. Right. So can the cookbook filters be made to put one of the zeros at the origin, or do they always produce responses with two non-trivial zeros? E On Thu, Feb 5, 2015 at 11:00 AM, Peter S peter.schoffhau...@gmail.com wrote: On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: To your first comment—well, of course. But maybe you’ve lost site of the context—here’s your comment: My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. Ethan pointed out that they can be, because one-zero is equivalent to a special case of two-zero. (Some would argue that if it’s a second-order filter, with two poles, then it always has matching zeros, even if one or both at at the origin. shrug.) Well, I think that would be rather a semantic distinction or an 'implied' zero at the origin - in practice, no one would actually calculate 0*x in an algorithm (as that would be pointless), though you're right that during calculation and analyis of the filter, you can treat the whole thing as a 2 pole / 2 zero filter with one or both zeros at the origin, as a special case of the general 2 pole biquad filter. On 05/02/2015, Ethan Duni ethan.d...@gmail.com wrote: You just stick the extra zero(s) off at z=infinity. Does 'z=infinity' mean it's at the origin? I'm not 100% sure of the terminology used here. - Peter -- 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] Thoughts on DSP books and neural networks
On 05/02/2015, Nigel Redmon earle...@earlevel.com wrote: The cookbook filters put the zeros at -1, the Nyquist frequency, for the low pass filter. So it’s true that the filters can’t be made the same for the cookbook formula, just not true that a two-zero filter can’t have the same transfer function as a one-zero. Correct. -- 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] Thoughts on DSP books and neural networks
On 2/4/15 8:25 PM, Peter S wrote: On 04/02/2015, robert bristow-johnsonr...@audioimagination.com wrote: i'm only saying that with a 2nd-order filter, there are only 5 degrees of freedom. only 5 knobs. so when someone says they came up with a different or better method of computing coefficients for *whatever* 2nd-order filter, my first curiosity will be what is the transfer function and from that, we usually find out it is equivalent to the Cookbook transfer function except Q or bandwidth is defined differently. My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. your 2-pole 1-zero filter is equivalent to a 2-pole 2-zero filter with the possible difference of 1 sample delay (a zero at z=0). I think that automatically assumes, the transfer function cannot be equivalent. it *can* be made equivalent to the 5-parameter generalized biquad, essentially the standard peaking EQ with this tilt or symmetry parameter. take a look at http://dsp.stackexchange.com/questions/19225/audio-eq-cookbook-without-frequency-warping/19253?s=1|0.1113#19253 (And strictly speaking, the transfer curve of my filter is not biquadratic, just BLT.) it's biquadratic with b0 or b2 set to zero. if it's second order in the denominator, even if b2 in the numerator is 0, it still can be mapped to the Knud Christensen model with 5 knobs on it. if that tilt or symmetry parameter that Knud introduces is 0, it's the old-fashioned peaking EQ that's a BPF in parallel to a wire, which is 3 knobs (one of which is the bandwidth knob) with possibly with a constant gain, a 4th knob. That also means, that my filter is also faster to process. 4 coefficients is quicker than 5. -- 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
Re: [music-dsp] Thoughts on DSP books and neural networks
On 04/02/2015, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote: Due to the current problems wit Adobe flash player, I’d prefer a de-flashed website. Or do you have links to the demoes to circumvent the flash stuff? No problem: http://morpheus.spectralhead.com/demos/mp3/ Here's the whole thing as a single .ZIP, in case you prefer that [100 MB]: http://morpheus.spectralhead.com/demos/demos.zip - Peter -- 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] Thoughts on DSP books and neural networks
Due to the current problems wit Adobe flash player, I’d prefer a de-flashed website. Or do you have links to the demoes to circumvent the flash stuff? Steffan On 04.02.2015|KW6, at 16:57, Peter S peter.schoffhau...@gmail.com wrote: DSP Algorithms of the Future http://morpheus.spectralhead.com/demos/ http://morpheus.spectralhead.com/demos/ -- 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] Thoughts on DSP books and neural networks
A few months ago someone suggested me to go home and read some beginner books. Actually, nearly a decade ago, I co-authored a book on modular audio processing titled Visual VSTi Programming, teaching people how to implement all the classical DSP algorithms. And before you say that modular audio is only for beginners and lamers, let me say that I've hand-written more than 200 DSP modules in C++, and that book represents my knowledge 10 years ago. By education, I'm an IT teacher and linguist. Which knowledge do you think is worth more? 1) have read a book about it 2) implemented it in code 3) implemented it in code, and also taught others how to do it Over the years, I invented and implemented several new DSP algorithms, ones that you won't find in any book, anywhere. To show you the work I have done over the years, I uploaded some audio demos of the musical filters, effects and instruments that I invented. I think that is all 100% on-topic here, these are strictly *musical* filters. Before you dismiss my theories, listen to my DSP work: DSP Algorithms of the Future http://morpheus.spectralhead.com/demos/ For comparison, I also included audio examples of Robert Bristow-Johnson's cookbook biquad filters, who argued that my theories are wrong, and I should do things by the book. I'm not telling you which sounds better. You be the judge - which sounds better? The classic, old-school algorithms that you find in textbooks and old papers, or the ones that came out of my head? After listening to my demos, if you wanted to learn digital filters and synthesis, who would you ask? Robert, or me? (You can also compare them to your own audio algorithms, if you have any). Personally, I prefer to invent the future that yet doesn't exist in books, instead of re-iterating the past the 1000th time. Most of these filter algorithms you won't find in any book, anywhere, as they're entirely my own inventions, not following any 'traditinal' designs. I'm a pragmatic person - I do what works in any given situation, even if it contradicts someone's theory. Shannon's work is titled A Mathematical *Theory* of Communication - which is, as the title says, just a theory (although a very inspiring one). It's not a Bible or a One Universal Truth. As soon as you start applying Shannon entropy to certain real-world scenario outside some 1950s telecommunications context, it doesn't fully make sense. Real-world example: Imagine that I download the Titanic movie from the internet (during which, it already became a 'message', more precisely, fragmented over several TCP/IP packets). [And if you prefer to stay strictly on-topic, you could substitute the Titanic with your favourite musical album, say, encoded as PCM or MP3.] Now imagine that I burn that movie to a DVD, and send it to you in a message (say, in the mail). So I've sent the string of bits that represent the Titanic movie to you over a non-noisy physical carrier on an optical media. (More precisely, the optical media is noisy from the dust and scratches, but it has its own built-in error correction.) Question: what is the entropy of my message? Now, this question doesn't make much sense. First of all, how do you even define that at all in this context? Second, what would even be the *purpose* of assigning some real-valued number between 0-1 to a movie? What? You say, the Titanic movie cannot be a message? Who said that? Seriously, did Shannon say, that a message *cannot* be 4.5 gigabytes long? And if it *can* be 4.5 GB long (why not?), then why couldn't it be the Titanic movie, encoded as x264 MPEG4? Why would one make such arbitrary restriction? (Remember: *all* digital messages consist of bits, so any digital message is effectiely a string of bits.) [Food for thought: the usage of the 0-1 range to express measure of information, is an entirely arbitrarily chosen metric, which is followed merely by convention. In practice, it could be any range. Someone, somewhere in history, once said: Hmm... I like the number '1' Let's use that as upper bound for amount of information! And since, it's followed by tradition. Now, when I do practical computing on fixed point numbers, I may find it more practical to use a fixed point integer number instead, as the samples or pixels I am processing are often in fixed point anyways. So why do int-float conversion in the first place, when it's not necessary? Normalization is costy, division being the *most expensive* operation, like, costing literally 30x as much as a single addition (which may make normalization extremely costy if you want to do it a billion times), and is often redundant, and may actually decrease precision or introduce quantization errors. This should be trivial for anyone being intimately familiar with floating point representation of numbers. Since IIRC the originally suggested problem was that you cannot compute so many correlations - in response to that, I gave the *simplest*, dumbest possible decorrelation
Re: [music-dsp] Thoughts on DSP books and neural networks
[…] On 04 Feb 2015, at 16:57 , Peter S peter.schoffhau...@gmail.com wrote: After listening to my demos, if you wanted to learn digital filters and synthesis, who would you ask? Robert, or me? Robert. -- 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] Thoughts on DSP books and neural networks
My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. No, that does not follow. A filter with two zeros can produce all of the transfer functions that a filter with one zero can, and many more besides. You just stick the extra zero(s) off at z=infinity. That also means, that my filter is also faster to process. First order filters are typically cheaper to process than second order filters. I wouldn't say that is a good argument that they are preferable to second order filters, though. E On Wed, Feb 4, 2015 at 5:25 PM, Peter S peter.schoffhau...@gmail.com wrote: On 04/02/2015, robert bristow-johnson r...@audioimagination.com wrote: i'm only saying that with a 2nd-order filter, there are only 5 degrees of freedom. only 5 knobs. so when someone says they came up with a different or better method of computing coefficients for *whatever* 2nd-order filter, my first curiosity will be what is the transfer function and from that, we usually find out it is equivalent to the Cookbook transfer function except Q or bandwidth is defined differently. My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. (And strictly speaking, the transfer curve of my filter is not biquadratic, just BLT.) That also means, that my filter is also faster to process. Best regards, Peter -- 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] Thoughts on DSP books and neural networks
On 04/02/2015, robert bristow-johnson r...@audioimagination.com wrote: i don't know precisely what you want to compare. i found something at the bottom of the list (#71) that says it came from cookbook filters. what should we be comparing it to? I'll post some comparisons later (if I ever find the time to make some fancy FFT graphs for it, I'm quite busy) - Peter -- 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] Thoughts on DSP books and neural networks
On 04/02/2015, robert bristow-johnson r...@audioimagination.com wrote: i'm only saying that with a 2nd-order filter, there are only 5 degrees of freedom. only 5 knobs. so when someone says they came up with a different or better method of computing coefficients for *whatever* 2nd-order filter, my first curiosity will be what is the transfer function and from that, we usually find out it is equivalent to the Cookbook transfer function except Q or bandwidth is defined differently. My filter has 2 poles and 1 zero. Unlike the Cookbook filter, which has 2 poles and 2 zeros. I think that automatically assumes, the transfer function cannot be equivalent. (And strictly speaking, the transfer curve of my filter is not biquadratic, just BLT.) That also means, that my filter is also faster to process. Best regards, Peter -- 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
