From douglas at music.columbia.edu Thu May 1 00:00:00 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Thu May 1 00:00:13 2008 Subject: [music-dsp] [admin] music-dsp FAQ Message-ID: <20080501040000.D6D8B22DE7748@music.columbia.edu> Hi, Just a reminder that if you are new to the list you should read the music-dsp FAQ. It contains answers to both technical _and_ adminstrative questions that often come up on the list. If your question appears in the FAQ it is safe to assume that it has been discussed on the list many times in the past, and you should probably have a look through the list archives before posting your question to the list. http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html Also of interest to new and not-so-new list members: The music-dsp list archives http://music.columbia.edu/cmc/music-dsp/musicdsparchives.html The music-dsp source code archive http://www.musicdsp.org music-dsp books and reviews http://music.columbia.edu/cmc/music-dsp/dspbooks.html All this and more at: http://music.columbia.edu/cmc/music-dsp Hasta la pasta, douglas (this is an automated message sent out on the 1st and 15th of each month) From kaleja at estarcion.com Fri May 2 02:07:06 2008 From: kaleja at estarcion.com (Russell Borogove) Date: Fri May 2 02:18:16 2008 Subject: [music-dsp] Questions about python References: <20080428230614.6F1C9224C957A@music.columbia.edu> <1080.69.226.74.140.1209432906.squirrel@mail.alumni.caltech.edu> Message-ID: <015f01c8ac1c$3f81ebc0$c701000a@jaseroque> It's non-free, but I've used the WingIDE development environment for Python, which does have a good debugger. However, it seemed to bog down somewhat on non-trivial programs, so I mostlydebug python by assert and print now. After half a lifetime of working in C and C++, I find python amazingly more productive even without a debugger. Programming is fun again! http://xkcd.com/353/ ----- Original Message ----- Some members of this list have recommended python. I downloaded it and have some questions. (1) I downloaded the version for Windows. The installer made what seem to be two executable files, python and pythonw. What is the difference? (2) The documentation talks about ttys and interactive programming. Is there an IDE for python, with a debugger. (I don't see the advantage of python if there is no IDE / workspace and debugger.) From music-dsp at musemagic.com Fri May 2 13:10:45 2008 From: music-dsp at musemagic.com (Bob Grove) Date: Fri May 2 13:11:18 2008 Subject: [music-dsp] Questions about python In-Reply-To: <015f01c8ac1c$3f81ebc0$c701000a@jaseroque> References: <20080428230614.6F1C9224C957A@music.columbia.edu><1080.69.226.74.140.1209432906.squirrel@mail.alumni.caltech.edu> <015f01c8ac1c$3f81ebc0$c701000a@jaseroque> Message-ID: <000201c8ac77$77514680$6401a8c0@acer> It's an interpreted language, so in terms of usefulness, think shell scripts, PERL initially (at least that's what I did). IDE/workspace/debugger here: http://pydev.sourceforge.net/ This is a plug-in for Eclipse (www.eclipse.org). If you do not use Eclipse, that's probably at this point a good IDE to learn overall. It's open source. Learning curve can be a little sticky for some, it's not quite the same as VC 6.0 although I find VS 2005/2008 team suite obnoxious myself and Eclipse way superior as a IDE. pythonw.exe doesn't automatically generated a MS-DOS console, or inherit one. In your downloaded on windows Python, under Python25\Doc is a compiled HTML help. Within there is a pretty good tutorial (a few things out of place but overall pretty good) If you know C and Matlab, Python is a piece of cake frankly to learn for the basics. Just go through the tutorial using the command line first and then move onto the more heady topics. For myself, looking to see what I could use it for, I started with running scripts where I would have used a shell commands or a .bat file. -----Original Message----- From: Russell Borogove [mailto:kaleja@estarcion.com] Sent: Thursday, May 01, 2008 11:07 PM To: music-dsp@music.columbia.edu Subject: Re: [music-dsp] Questions about python It's non-free, but I've used the WingIDE development environment for Python, which does have a good debugger. However, it seemed to bog down somewhat on non-trivial programs, so I mostlydebug python by assert and print now. After half a lifetime of working in C and C++, I find python amazingly more productive even without a debugger. Programming is fun again! http://xkcd.com/353/ ----- Original Message ----- Some members of this list have recommended python. I downloaded it and have some questions. (1) I downloaded the version for Windows. The installer made what seem to be two executable files, python and pythonw. What is the difference? (2) The documentation talks about ttys and interactive programming. Is there an IDE for python, with a debugger. (I don't see the advantage of python if there is no IDE / workspace and debugger.) -- 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 -- No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.7/1409 - Release Date: 5/1/2008 8:39 AM No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.7/1409 - Release Date: 5/1/2008 8:39 AM From banzaii at gmx.net Fri May 2 13:26:19 2008 From: banzaii at gmx.net (banzaii) Date: Fri May 2 13:26:39 2008 Subject: [music-dsp] Questions about python In-Reply-To: <000201c8ac77$77514680$6401a8c0@acer> References: <20080428230614.6F1C9224C957A@music.columbia.edu><1080.69.226.74.140.1209432906.squirrel@mail.alumni.caltech.edu> <015f01c8ac1c$3f81ebc0$c701000a@jaseroque> <000201c8ac77$77514680$6401a8c0@acer> Message-ID: <02478A7C-9DC4-44C7-8671-ED767419C369@gmx.net> Hi! Yesterday I stumbled across a nice math project while searching for a free alternative to mathlab: Have a look at sage (http:// www.sagemath.org/). As far as I know up to now, it seems to be also python based and combines some math programs in one program. So, you can write functions in python (it seems to be possible to compile them also to c code from that program and immediately use them in your equations...nice...). There are also plugins so that you can use functions from external programs like mathlab etc. I am sorry, I cannot tell more about it - I am quite new to dsp programming and especially sage is absolutley new to me. What I haven't found yet is a possibility to load a soundfile into this and apply calculations to it, but I think this should be possible via python. Hope it helps and is not too off-topic. greetings, Kai-Philipp Schoellmann From mark.plumbley at elec.qmul.ac.uk Mon May 5 18:15:11 2008 From: mark.plumbley at elec.qmul.ac.uk (Mark Plumbley) Date: Mon May 5 18:15:31 2008 Subject: [music-dsp] Call for Papers: ICArn International Workshop 2008 Message-ID: <3399496864F99445B051FD9556FF3B6FF28790@staff-mail1.vpn.elec.qmul.ac.uk> Call for Papers ICA Research Network International Workshop 25-26 September, 2008, Liverpool, U.K. www.icarn.org The ICA Research Network is sponsored by the Engineering and Physical Sciences Research Council (EPSRC) in the U.K., and is aimed at improving communications in the area of Blind Source Separation (BSS) and Independent Component Analysis (ICA). The 2008 ICA Research Network Workshop will be held at the University of Liverpool covering the latest developments and techniques in the area of source separation and ICA. Submissions from international participants are most welcome. Topics The workshop will feature keynote addresses and technical presentations (oral and poster), which will be included in the registration. Papers are solicited on topics in the area of source separation or ICA, including but not limited to: Algorithms and Architectures (Non-linear ICA, Probabilistic Models, Sparse Coding, etc), Theory (Optimization, Complex Methods, Time-Frequency Representations, etc), Applications (Audio, Bio-Informatics, Biomedical Engineering, Communications, Finance, Image Processing, Psychology, etc), and novel methods (compressed sensing, non-negative matrix factorization). A special feature of this workshop will be a special poster session where authors will have the opportunity to present their current work in progress. Authors should indicate their preference at the submission stage and submit short papers. Full Paper Submission Procedure Prospective authors are invited to submit camera-ready papers of no more than four A4-size pages in the PDF format. Please use the template and the electronic submission procedure described at the workshop homepage (www.icarn.org). At least one author of each accepted paper must undertake to attend the workshop. Prospective authors can seek clarification using the email address icarnw08@liverpool.ac.uk. Accepted papers will be published in a bound volume. Authors of the most innovative papers will be invited to submit substantially extended and updated versions of their papers for further review and possible publication in the IET Signal Processing journal. There will be one Best Student Paper Award for the best paper presented at the workshop by a student. Registration costs will include attendance in all the sessions, a copy of the bound proceedings, mid-morning and mid-afternoon refreshments as well as buffet lunches on both days, and the Workshop dinner on the evening of 25 September 2008. Important Dates/Deadlines Submission of papers: 17 June, 2008 Notification of acceptance : 17 July, 2008 Submission of camera-ready accepted paper : 07 August, 2008 Early registration and author registration : 07 August, 2008 Final date for registration : 08 September, 2008 Workshop : 25-26 September, 2008 Website: www.icarn.org Chair A K Nandi Vice-Chair X Zhu Organising Committee A K Nandi X Zhu W Al-Nuaimy M E Davies J Gao M D Plumbley Programme Committee National P Baxter M E Davies R Everson C Fyfe M Girolami C J James A K Nandi M D Plumbley S Sanei X Zhu International J F Cardoso A Cichocki P Comon C Jutten E Oja P Smaragdis -- Dr Mark D Plumbley Centre for Digital Music Department of Electronic Engineering Queen Mary University of London Mile End Road, London E1 4NS, UK Tel: +44 (0)20 7882 7518 Fax: +44 (0)20 7882 7997 Email: mark.plumbley@elec.qmul.ac.uk http://www.elec.qmul.ac.uk/people/markp/ From mark.plumbley at elec.qmul.ac.uk Mon May 5 18:18:43 2008 From: mark.plumbley at elec.qmul.ac.uk (Mark Plumbley) Date: Mon May 5 18:18:58 2008 Subject: [music-dsp] Call for Papers: ICArn International Workshop 2008 Message-ID: <3399496864F99445B051FD9556FF3B6FF28792@staff-mail1.vpn.elec.qmul.ac.uk> Dear List, This workshop may be of particular interest to those of you working in audio signal separation or auditory scene analysis. Best wishes, Mark Plumbley ------------------------------------------------------------------- Call for Papers ICA Research Network International Workshop 25-26 September, 2008, Liverpool, U.K. www.icarn.org The ICA Research Network is sponsored by the Engineering and Physical Sciences Research Council (EPSRC) in the U.K., and is aimed at improving communications in the area of Blind Source Separation (BSS) and Independent Component Analysis (ICA). The 2008 ICA Research Network Workshop will be held at the University of Liverpool covering the latest developments and techniques in the area of source separation and ICA. Submissions from international participants are most welcome. Topics The workshop will feature keynote addresses and technical presentations (oral and poster), which will be included in the registration. Papers are solicited on topics in the area of source separation or ICA, including but not limited to: Algorithms and Architectures (Non-linear ICA, Probabilistic Models, Sparse Coding, etc), Theory (Optimization, Complex Methods, Time-Frequency Representations, etc), Applications (Audio, Bio-Informatics, Biomedical Engineering, Communications, Finance, Image Processing, Psychology, etc), and novel methods (compressed sensing, non-negative matrix factorization). A special feature of this workshop will be a special poster session where authors will have the opportunity to present their current work in progress. Authors should indicate their preference at the submission stage and submit short papers. Full Paper Submission Procedure Prospective authors are invited to submit camera-ready papers of no more than four A4-size pages in the PDF format. Please use the template and the electronic submission procedure described at the workshop homepage (www.icarn.org). At least one author of each accepted paper must undertake to attend the workshop. Prospective authors can seek clarification using the email address icarnw08@liverpool.ac.uk. Accepted papers will be published in a bound volume. Authors of the most innovative papers will be invited to submit substantially extended and updated versions of their papers for further review and possible publication in the IET Signal Processing journal. There will be one Best Student Paper Award for the best paper presented at the workshop by a student. Registration costs will include attendance in all the sessions, a copy of the bound proceedings, mid-morning and mid-afternoon refreshments as well as buffet lunches on both days, and the Workshop dinner on the evening of 25 September 2008. Important Dates/Deadlines Submission of papers: 17 June, 2008 Notification of acceptance : 17 July, 2008 Submission of camera-ready accepted paper : 07 August, 2008 Early registration and author registration : 07 August, 2008 Final date for registration : 08 September, 2008 Workshop : 25-26 September, 2008 Website: www.icarn.org Chair A K Nandi Vice-Chair X Zhu Organising Committee A K Nandi X Zhu W Al-Nuaimy M E Davies J Gao M D Plumbley Programme Committee National P Baxter M E Davies R Everson C Fyfe M Girolami C J James A K Nandi M D Plumbley S Sanei X Zhu International J F Cardoso A Cichocki P Comon C Jutten E Oja P Smaragdis -- Dr Mark D Plumbley Centre for Digital Music Department of Electronic Engineering Queen Mary University of London Mile End Road, London E1 4NS, UK Tel: +44 (0)20 7882 7518 Fax: +44 (0)20 7882 7997 Email: mark.plumbley@elec.qmul.ac.uk http://www.elec.qmul.ac.uk/people/markp/ From douglas at music.columbia.edu Mon May 5 19:16:05 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Mon May 5 19:16:18 2008 Subject: [music-dsp] FAQ is BAQ! Message-ID: <481F9535.1000703@music.columbia.edu> Whew, I finally got one of our workstudy students to go through the musicdsp.org wiki-based FAQ and extract all of the questions into a text file. They're now re-html-ized and returned to their original home at: http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html As you can see, they look horrible, are difficult to read, are poorly formatted, and are terribly out of date. But at least they're accessible now and they'll be properly crawled by search engines. Who wants to have a FAQ updating party??? Please send revisions, addition, etc, to me and I'll gladly post them. Sorry this took so long! douglas -- ............................................... 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 From rossb-lists at audiomulch.com Tue May 6 04:12:00 2008 From: rossb-lists at audiomulch.com (Ross Bencina) Date: Tue May 6 04:12:34 2008 Subject: [music-dsp] FAQ is BAQ! References: <481F9535.1000703@music.columbia.edu> Message-ID: <002501c8af50$dc5c7370$0300a8c0@rossmacbook> Hi Douglas The FAQ links to http://music.columbia.edu/music-dsp/dspbooks.html but that page doesn't exist. Under the question "is there a good online dsp guide?" You could link to Miller Puckette's book as an online book "The Theory and Technique of Electronic Music" http://crca.ucsd.edu/~msp/techniques.htm Might be useful to number the FAQs so we can refer to them by number. Under the question "What is the situation with MPEG patents? " There is info here: http://en.wikipedia.org/wiki/MP3 Apparently mp3 codec licencing is handled via http://www.mp3licensing.com/ Under the question "how do i make an equalizer? how do i make guitar pedal effect xxx? " If you're serious about sound processor design, you should research existing algorithms published by the various conferences and the Journal of the AES (audio engineering society) (available online http://www.aes.org/) You could also look in publications of the DAFX conference (available free online at http://www.dafx.de/) and the International Computer Music Conference (http://www.computermusic.org/, not free online but available in many university libraries) That's as far as I got for now.. Thanks! Ross. ----- Original Message ----- From: "douglas repetto" To: Sent: Tuesday, May 06, 2008 9:16 AM Subject: [music-dsp] FAQ is BAQ! > > Whew, I finally got one of our workstudy students to go through the > musicdsp.org wiki-based FAQ and extract all of the questions into a text > file. They're now re-html-ized and returned to their original home at: > > http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html > > > As you can see, they look horrible, are difficult to read, are poorly > formatted, and are terribly out of date. But at least they're accessible > now and they'll be properly crawled by search engines. > > Who wants to have a FAQ updating party??? > > Please send revisions, addition, etc, to me and I'll gladly post them. > > > Sorry this took so long! > > > douglas > > > > -- > ............................................... 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 > From douglas at music.columbia.edu Wed May 7 16:15:14 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Wed May 7 16:15:32 2008 Subject: [music-dsp] FAQ is BAQ! In-Reply-To: <002501c8af50$dc5c7370$0300a8c0@rossmacbook> References: <481F9535.1000703@music.columbia.edu> <002501c8af50$dc5c7370$0300a8c0@rossmacbook> Message-ID: <48220DD2.4000001@music.columbia.edu> Great start Ross, thanks. I made your changes. Does someone want to take a crack at redoing the organization and adding a numbering scheme? Right now there are three main sections: meta FAQs code/theory FAQs development FAQs It'd be great to have an index up top with links to the sections and questions, like most FAQs. douglas Ross Bencina wrote: > Hi Douglas > > The FAQ links to http://music.columbia.edu/music-dsp/dspbooks.html but > that page doesn't exist. > > > Under the question "is there a good online dsp guide?" You could link to > Miller Puckette's book as an online book > "The Theory and Technique of Electronic Music" > http://crca.ucsd.edu/~msp/techniques.htm > > > Might be useful to number the FAQs so we can refer to them by number. > > > Under the question "What is the situation with MPEG patents? " > There is info here: > http://en.wikipedia.org/wiki/MP3 > Apparently mp3 codec licencing is handled via http://www.mp3licensing.com/ > > Under the question "how do i make an equalizer? how do i make guitar > pedal effect xxx? " > > If you're serious about sound processor design, you should research > existing algorithms published by the various conferences and the Journal > of the AES (audio engineering society) (available online > http://www.aes.org/) You could also look in publications of the DAFX > conference (available free online at http://www.dafx.de/) and the > International Computer Music Conference (http://www.computermusic.org/, > not free online but available in many university libraries) > > > That's as far as I got for now.. > > Thanks! > > Ross. > > > > > ----- Original Message ----- From: "douglas repetto" > > To: > Sent: Tuesday, May 06, 2008 9:16 AM > Subject: [music-dsp] FAQ is BAQ! > > >> >> Whew, I finally got one of our workstudy students to go through the >> musicdsp.org wiki-based FAQ and extract all of the questions into a >> text file. They're now re-html-ized and returned to their original >> home at: >> >> http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html >> >> >> As you can see, they look horrible, are difficult to read, are poorly >> formatted, and are terribly out of date. But at least they're >> accessible now and they'll be properly crawled by search engines. >> >> Who wants to have a FAQ updating party??? >> >> Please send revisions, addition, etc, to me and I'll gladly post them. >> >> >> Sorry this took so long! >> >> >> douglas >> >> >> >> -- >> ............................................... 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 -- ............................................... 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 From padawan12 at obiwannabe.co.uk Wed May 7 19:06:36 2008 From: padawan12 at obiwannabe.co.uk (Andy Farnell) Date: Wed May 7 19:06:48 2008 Subject: [music-dsp] FAQ is BAQ! In-Reply-To: <48220DD2.4000001@music.columbia.edu> References: <481F9535.1000703@music.columbia.edu> <002501c8af50$dc5c7370$0300a8c0@rossmacbook> <48220DD2.4000001@music.columbia.edu> Message-ID: <20080508000636.652e3d53.padawan12@obiwannabe.co.uk> I'm on it. Plus a bit of editing and WC3(HTM4) compliance too - as FAQs grow they take too long to load as a single body document. I'll post it back to you when done Douglas. a. On Wed, 07 May 2008 16:15:14 -0400 douglas repetto wrote: > Does someone want to take a crack at redoing the organization and adding > a numbering scheme? Right now there are three main sections: > > meta FAQs > code/theory FAQs > development FAQs > > > It'd be great to have an index up top with links to the sections and > questions, like most FAQs. -- Use the source From douglas at music.columbia.edu Thu May 8 12:32:00 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Thu May 8 12:32:10 2008 Subject: [music-dsp] FAQ is BAQ! In-Reply-To: <20080508040817.2ca92ff3.padawan12@obiwannabe.co.uk> References: <481F9535.1000703@music.columbia.edu> <002501c8af50$dc5c7370$0300a8c0@rossmacbook> <48220DD2.4000001@music.columbia.edu> <20080508040817.2ca92ff3.padawan12@obiwannabe.co.uk> Message-ID: <48232B00.60302@music.columbia.edu> Great Andy, thanks! It looks so much better. I've also cleaned up the structure a bit so that the dsp FAQs are on the main page and the admin FAQs are just one of the sections. http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html music-dsp members, please feel free to send in update/comments/additions. best, douglas Andy Farnell wrote: > > Hey Douglas, > > See attached: > > Index done > > Some editing to harmonise and streamline. > Could trim down denormals, noise and aliasing a bit more maybe. > > Use of ordered lists > Use of blocks > > all good wishes, > > Andy > > > -- ............................................... 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 From rbj at audioimagination.com Thu May 8 13:10:00 2008 From: rbj at audioimagination.com (robert bristow-johnson) Date: Thu May 8 13:10:26 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. Message-ID: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> > ----- Original Message ----- > From: "douglas repetto" > To: "A discussion list for music-related DSP" > Subject: Re: [music-dsp] FAQ is BAQ! > Date: Thu, 08 May 2008 12:32:00 -0400 > > > > Great Andy, thanks! It looks so much better. > > > I've also cleaned up the structure a bit so that the dsp FAQs are on the main > page and the admin FAQs are just one of the sections. > > http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html > > > music-dsp members, please feel free to send in update/comments/additions. i have to take a little exception to a wee part of the technical content regarding "frequency shifting". in FAQ #19 (the numbering doesn't quite correspond with the table of contents at the top, but that's really small potatoes) "How do I do pitch shifting?", it says regarding "frequency shifting": "Frequency shifting is actually an older technology from the analog days, and is also somewhat simpler than pitch shifting. Nonetheless there are some things to watch out for. When done with ring modulation as part of the process (most methods do this), you must filter carefully to prevent aliasing." so far this is fine. then, "The best (simple efficient clean) current method is Weaver modulation; see this link: http://www.csounds.com/ezine/summer2000/processing/" there is a technical value judgement here ("best, simple[st], [most] efficient, clean[est]") that might not be universally held. another method that is current (even more current because in Weaver's day, they didn't have real-time Hilbert transformers) involves a single "filter" (the Hilbert transformer, i won't count the corresponding delay line as another filter) and two sine/cosine multiply operations. a quick Google search finds this link to illustrate: http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_frequency_shifting.htm i s'pose, if you want, i could code up a simple C-code snippet, but someone else would have to build something around it to test it. just a couple of $0.01 tossed in for your consideration. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge." From jens at optimolch.de Thu May 8 14:36:37 2008 From: jens at optimolch.de (Jens Groh) Date: Thu May 8 14:36:52 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> Message-ID: <956DF8F8-67A3-49A9-84A1-7136ABEFFA17@optimolch.de> One thing about Weaver's trick that could at least be called elegant is that it works with IIR, not only FIR filters. They are pretty efficient. (Well, probably not too too much better compared to FFT-based FIR in terms of "decibels signal-to-alias ratio per multiply-accumulate operation"...) They have low latency compared to FIR filters (unless you want to pay for that low latency convolution patent...). The phase respone is arbitrary in this case, and the passband ripple is uncritical, in contrast to the Hilbert transformer. Apart from that, I would say the Hilbert transformer method is actually the simpler one. It is much easier to understand. Still I'm happy that I seem to be the first one who translated that old tricky ham radio geek circuit to the DSP audio world and that a couple of people seem to like it. In fact I had only remembered how it works and the name Weaver and it was very hard to find the reference at that time. Today it's easy -- here you are: http://www.h4.dion.ne.jp/~ja5fp/weaver.pdf. (Isn't that tube circuit cool? :-) ) Jens Groh (author of that Csound Mag article) Am 08.05.2008 um 19:10 schrieb robert bristow-johnson: > >> ----- Original Message ----- >> From: "douglas repetto" >> To: "A discussion list for music-related DSP" > > >> Subject: Re: [music-dsp] FAQ is BAQ! >> Date: Thu, 08 May 2008 12:32:00 -0400 >> >> >> >> Great Andy, thanks! It looks so much better. >> >> >> I've also cleaned up the structure a bit so that the dsp FAQs are >> on the main >> page and the admin FAQs are just one of the sections. >> >> http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html >> >> >> music-dsp members, please feel free to send in update/comments/ >> additions. > > i have to take a little exception to a wee part of the technical > content regarding "frequency shifting". in FAQ #19 (the numbering > doesn't quite correspond with the table of contents at the top, but > that's really small potatoes) "How do I do pitch shifting?", it says > regarding "frequency shifting": > > "Frequency shifting is actually an older technology from the analog > days, and is also somewhat simpler than pitch shifting. Nonetheless > there are some things to watch out for. When done with ring > modulation as part of the process (most methods do this), you must > filter carefully to prevent aliasing." > > so far this is fine. then, > > "The best (simple efficient clean) current method is Weaver > modulation; see this link: http://www.csounds.com/ezine/summer2000/processing/ > " > > there is a technical value judgement here ("best, simple[st], [most] > efficient, clean[est]") that might not be universally held. another > method that is current (even more current because in Weaver's day, > they didn't have real-time Hilbert transformers) involves a single > "filter" (the Hilbert transformer, i won't count the corresponding > delay line as another filter) and two sine/cosine multiply > operations. a quick Google search finds this link to illustrate: > > http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_frequency_shifting.htm > > i s'pose, if you want, i could code up a simple C-code snippet, but > someone else would have to build something around it to test it. > > just a couple of $0.01 tossed in for your consideration. > > -- > > r b-j rbj@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 From douglas at music.columbia.edu Thu May 8 15:12:35 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Thu May 8 15:12:48 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <956DF8F8-67A3-49A9-84A1-7136ABEFFA17@optimolch.de> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> <956DF8F8-67A3-49A9-84A1-7136ABEFFA17@optimolch.de> Message-ID: <482350A3.9080300@music.columbia.edu> Would one of your like to write this stuff up to be added to the "frequency shifting" part of the FAQ? We don't need to make broad judgments about specific techniques, we just want to give people starting points for their own explorations. douglas Jens Groh wrote: > One thing about Weaver's trick that could at least be called elegant is > that it works with IIR, not only FIR filters. > They are pretty efficient. (Well, probably not too too much better > compared to FFT-based FIR in terms of "decibels signal-to-alias ratio > per multiply-accumulate operation"...) > They have low latency compared to FIR filters (unless you want to pay > for that low latency convolution patent...). > The phase respone is arbitrary in this case, and the passband ripple is > uncritical, in contrast to the Hilbert transformer. > > Apart from that, I would say the Hilbert transformer method is actually > the simpler one. It is much easier to understand. > > Still I'm happy that I seem to be the first one who translated that old > tricky ham radio geek circuit to the DSP audio world and that a couple > of people seem to like it. > In fact I had only remembered how it works and the name Weaver and it > was very hard to find the reference at that time. Today it's easy -- > here you are: http://www.h4.dion.ne.jp/~ja5fp/weaver.pdf. (Isn't that > tube circuit cool? :-) ) > > Jens Groh (author of that Csound Mag article) > > > Am 08.05.2008 um 19:10 schrieb robert bristow-johnson: > >> >>> ----- Original Message ----- >>> From: "douglas repetto" >>> To: "A discussion list for music-related DSP" >>> >>> Subject: Re: [music-dsp] FAQ is BAQ! >>> Date: Thu, 08 May 2008 12:32:00 -0400 >>> >>> >>> >>> Great Andy, thanks! It looks so much better. >>> >>> >>> I've also cleaned up the structure a bit so that the dsp FAQs are on >>> the main >>> page and the admin FAQs are just one of the sections. >>> >>> http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html >>> >>> >>> music-dsp members, please feel free to send in >>> update/comments/additions. >> >> i have to take a little exception to a wee part of the technical >> content regarding "frequency shifting". in FAQ #19 (the numbering >> doesn't quite correspond with the table of contents at the top, but >> that's really small potatoes) "How do I do pitch shifting?", it says >> regarding "frequency shifting": >> >> "Frequency shifting is actually an older technology from the analog >> days, and is also somewhat simpler than pitch shifting. Nonetheless >> there are some things to watch out for. When done with ring modulation >> as part of the process (most methods do this), you must filter >> carefully to prevent aliasing." >> >> so far this is fine. then, >> >> "The best (simple efficient clean) current method is Weaver >> modulation; see this link: >> http://www.csounds.com/ezine/summer2000/processing/" >> >> there is a technical value judgement here ("best, simple[st], [most] >> efficient, clean[est]") that might not be universally held. another >> method that is current (even more current because in Weaver's day, >> they didn't have real-time Hilbert transformers) involves a single >> "filter" (the Hilbert transformer, i won't count the corresponding >> delay line as another filter) and two sine/cosine multiply >> operations. a quick Google search finds this link to illustrate: >> >> http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_frequency_shifting.htm >> >> >> i s'pose, if you want, i could code up a simple C-code snippet, but >> someone else would have to build something around it to test it. >> >> just a couple of $0.01 tossed in for your consideration. >> >> -- >> >> r b-j rbj@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 > linkshttp://music.columbia.edu/cmc/music-dsphttp://music.columbia.edu/mailman/listinfo/music-dsp > -- ............................................... 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 From douglas at music.columbia.edu Thu May 8 15:16:27 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Thu May 8 15:16:35 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> Message-ID: <4823518B.3060401@music.columbia.edu> robert bristow-johnson wrote: in FAQ #19 (the numbering > doesn't quite correspond with the table of contents at the top, but > that's really small potatoes) Ah, thanks. It was a problem with
    tags. douglas -- ............................................... 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 From czhenry at gmail.com Thu May 8 15:36:14 2008 From: czhenry at gmail.com (Charles Henry) Date: Thu May 8 15:36:27 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> Message-ID: <518fe7b20805081236i36b4814fy5a8a6e1359fd2199@mail.gmail.com> On Thu, May 8, 2008 at 12:10 PM, robert bristow-johnson wrote: > there is a technical value judgement here ("best, simple[st], [most] efficient, clean[est]") that might not be universally held. another method that is current (even more current because in Weaver's day, they didn't have real-time Hilbert transformers) involves a single "filter" (the Hilbert transformer, i won't count the corresponding delay line as another filter) and two sine/cosine multiply operations. a quick Google search finds this link to illustrate: > > http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_frequency_shifting.htm > r b-j rbj@audioimagination.com Quick question, here. I have sometimes struggled to create a good RT Hilbert transform (I made a couple of pd patches that while appearing to be correct on the surface with relevant math behind it, but produces horrible phase shifts). How does one make an RT Hilbert transform? (I can also provide details on my method, if desired.) Chuck From rbj at audioimagination.com Thu May 8 17:39:55 2008 From: rbj at audioimagination.com (robert bristow-johnson) Date: Thu May 8 17:40:16 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. Message-ID: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> > ----- Original Message ----- > From: "Charles Henry" > To: "A discussion list for music-related DSP" > Subject: Re: [music-dsp] FAQ is BAQ! - frequency shifting. > Date: Thu, 8 May 2008 14:36:14 -0500 > > > Quick question, here. I have sometimes struggled to create a good RT > Hilbert transform (I made a couple of pd patches that while appearing > to be correct on the surface with relevant math behind it, but > produces horrible phase shifts). How does one make an RT Hilbert > transform? > (I can also provide details on my method, if desired.) it's a filter, but a true Hilbert transformer is not causal. a *delayed* Hilbert transformer (so the constant delay or the linear phase of this has to be accounted for) can be done to any degree of precision (if you're willing to throw taps at it) with an FIR (with a sorta halfband-odd-symmetry) with (non-delayed) impulse response of h[n] = (1-(-1)^n) * 1/(pi*n) * w[n] where w[n] is a window function that you like. you can also use MATLAB's remez() or firls() to design an optimal h[n] for a given length. note that for even n, h[n]=0 (because of halfband-symmetry) which should be incorporated into the FIR code. of course, you need to delay the whole thing by 1/2 of the width of w[n] and your non-hilberted signal needs to be delayed by the same amount so you have an "in-phase" and "quadrature" pair (albeit a delayed pair of signals). -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge." From czhenry at gmail.com Thu May 8 18:04:53 2008 From: czhenry at gmail.com (Charles Henry) Date: Thu May 8 18:05:06 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> References: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> Message-ID: <518fe7b20805081504j623c9419k76ebe591d4d507ac@mail.gmail.com> On Thu, May 8, 2008 at 4:39 PM, robert bristow-johnson wrote: > >> ----- Original Message ----- >> From: "Charles Henry" >> To: "A discussion list for music-related DSP" >> Subject: Re: [music-dsp] FAQ is BAQ! - frequency shifting. >> Date: Thu, 8 May 2008 14:36:14 -0500 >> >> >> Quick question, here. I have sometimes struggled to create a good RT >> Hilbert transform (I made a couple of pd patches that while appearing >> to be correct on the surface with relevant math behind it, but >> produces horrible phase shifts). How does one make an RT Hilbert >> transform? >> (I can also provide details on my method, if desired.) > > it's a filter, but a true Hilbert transformer is not causal. > > a *delayed* Hilbert transformer (so the constant delay or the linear phase of this has to be accounted for) can be done to any degree of precision (if you're willing to throw taps at it) with an FIR (with a sorta halfband-odd-symmetry) with (non-delayed) impulse response of > > h[n] = (1-(-1)^n) * 1/(pi*n) * w[n] Okay, this is the same formula I came up with for the ideal response. It's the Hilbert transform of sinc(n), sampled at the integers (I think... maybe it's the adjoint...) But where I went wrong was to look at the following function as an approximation of it. x(n)= sum ( k=-N/2:N/2, sin(2*pi*n*k/N) ) This is a function having unitary frequency response of the DFT on a block of samples length N, and 90 degrees phase shift. It *looks* like a good approximation, but it only performs correctly at the integer multiples of frequencies. Anything in between turns to garbage... Thanks for the help!! Chuck > > where w[n] is a window function that you like. you can also use MATLAB's remez() or firls() to design an optimal h[n] for a given length. note that for even n, h[n]=0 (because of halfband-symmetry) which should be incorporated into the FIR code. of course, you need to delay the whole thing by 1/2 of the width of w[n] and your non-hilberted signal needs to be delayed by the same amount so you have an "in-phase" and "quadrature" pair (albeit a delayed pair of signals). > > > -- > > r b-j rbj@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 > From jchandjr at bellsouth.net Thu May 8 18:45:43 2008 From: jchandjr at bellsouth.net (James Chandler Jr) Date: Thu May 8 18:45:59 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> References: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> Message-ID: <77A20A88ED674916B579798A9EB50B72@AthlonX2> There was a 'pretty good' IIR approximation Hilbert in CSound source last time I looked (a few years ago). Maybe still in there. The CSound Hilbert was a digital emulation of an analog Hilbert which was published by Bernie Hutchins in his Electronotes magazine long ago. It sends the signal thru two legs of cascaded first-order phase shifters. The 'staggered' FC's of the two phase shifter chains, resulted in the output of one phase shifter cascade, a pretty good approximation of 90 degrees shifted against the other phase shifter cascade output. But neither output had a Hilbert relationship to the original signal. That is how it accomplished the requisite delay so it could run in realtime. One phase shift leg was just delayed more than the other phase shift leg. Pretty ingenious, I thought. jcjr > ----- Original Message ----- > From: "Charles Henry" > To: "A discussion list for music-related DSP" > Subject: Re: [music-dsp] FAQ is BAQ! - frequency shifting. > Date: Thu, 8 May 2008 14:36:14 -0500 > > > Quick question, here. I have sometimes struggled to create a good RT > Hilbert transform (I made a couple of pd patches that while appearing > to be correct on the surface with relevant math behind it, but > produces horrible phase shifts). How does one make an RT Hilbert > transform? From vesa.norilo at saunalahti.fi Fri May 9 03:59:41 2008 From: vesa.norilo at saunalahti.fi (Vesa Norilo) Date: Fri May 9 03:59:48 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <77A20A88ED674916B579798A9EB50B72@AthlonX2> References: <20080508213955.6F03954D875@ws6-9.us4.outblaze.com> <77A20A88ED674916B579798A9EB50B72@AthlonX2> Message-ID: <4824046D.3060307@saunalahti.fi> James Chandler Jr wrote: > There was a 'pretty good' IIR approximation Hilbert in CSound source > last time I looked (a few years ago). Maybe still in there. > > The CSound Hilbert was a digital emulation of an analog Hilbert which > was published by Bernie Hutchins in his Electronotes magazine long ago. > > It sends the signal thru two legs of cascaded first-order phase > shifters. The 'staggered' FC's of the two phase shifter chains, > resulted in the output of one phase shifter cascade, a pretty good > approximation of 90 degrees shifted against the other phase shifter > cascade output. > > But neither output had a Hilbert relationship to the original signal. > > That is how it accomplished the requisite delay so it could run in > realtime. One phase shift leg was just delayed more than the other > phase shift leg. > > Pretty ingenious, I thought. > This sounds like the filters I got from Olli Niemitalo (yes, google confirmed that) http://www.student.oulu.fi/~oniemita/dsp/hilbert/ The explanation is pretty good and interesting. I used them for a "complex ring modulator" with just one sideband, and also fiddled around for envelope detection (in some cases, adding the absolute values 90 deg shifted signals reduces the effect of envelope detector riding the waveform). hth, Vesa From bthrew at gmail.com Fri May 9 20:33:35 2008 From: bthrew at gmail.com (barry threw) Date: Fri May 9 20:33:56 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <518fe7b20805081236i36b4814fy5a8a6e1359fd2199@mail.gmail.com> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> <518fe7b20805081236i36b4814fy5a8a6e1359fd2199@mail.gmail.com> Message-ID: <6FE4EA63-307E-433B-9D44-53E765198A89@gmail.com> A pd Hilbert. bt #N canvas 343 69 601 314 12; #X obj 105 92 biquad~ 0.83774 -0.06338 0.06338 -0.83774 1; #X obj 105 66 biquad~ 1.94632 -0.94657 0.94657 -1.94632 1; #X obj 86 149 biquad~ -0.02569 0.260502 -0.260502 0.02569 1; #X obj 86 175 biquad~ 1.8685 -0.870686 0.870686 -1.8685 1; #X obj 86 39 inlet~; #X obj 105 121 outlet~; #X obj 86 202 outlet~; #X text 34 232 This is a pair of 4th-order all-pass filters whose outputs somehow manage to be about 90 degrees out of phase from each other. Both have different phases from the original. Adapted from a 4X patch by Emmanuel Favreau \, circa 1982; #X obj 502 39 inlet; #X text 461 13 bang to clear; #X text 80 16 signal in; #X msg 502 112 clear; #X connect 0 0 5 0; #X connect 1 0 0 0; #X connect 2 0 3 0; #X connect 3 0 6 0; #X connect 4 0 1 0; #X connect 4 0 2 0; #X connect 8 0 11 0; #X connect 11 0 1 0; #X connect 11 0 0 0; #X connect 11 0 2 0; #X connect 11 0 3 0; On May 8, 2008, at 12:36 PM, Charles Henry wrote: > On Thu, May 8, 2008 at 12:10 PM, robert bristow-johnson > wrote: > >> there is a technical value judgement here ("best, simple[st], >> [most] efficient, clean[est]") that might not be universally held. >> another method that is current (even more current because in >> Weaver's day, they didn't have real-time Hilbert transformers) >> involves a single "filter" (the Hilbert transformer, i won't count >> the corresponding delay line as another filter) and two sine/cosine >> multiply operations. a quick Google search finds this link to >> illustrate: >> >> http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_frequency_shifting.htm > >> r b-j rbj@audioimagination.com > > Quick question, here. I have sometimes struggled to create a good RT > Hilbert transform (I made a couple of pd patches that while appearing > to be correct on the surface with relevant math behind it, but > produces horrible phase shifts). How does one make an RT Hilbert > transform? > (I can also provide details on my method, if desired.) > > Chuck > -- > 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 barry threw Media Art and Technology San Francisco, CA Work: 857-544-3967 Email: bthrew (at) gmail (dot) com Web: www.barrythrew.com From jens at optimolch.de Sat May 10 03:09:58 2008 From: jens at optimolch.de (Jens Groh) Date: Sat May 10 03:10:12 2008 Subject: [music-dsp] FAQ is BAQ! - frequency shifting. In-Reply-To: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> References: <20080508171000.534379EE56@ws6-2.us4.outblaze.com> Message-ID: <8458057A-7683-4470-9D9E-41AE8EE2473A@optimolch.de> I think the principal engineering task is, how good is the alias suppression. A secondary task is, how good is the amplitude fidelity. Yeah, and as always, the question is, how much processing effort do I have to spend. You can trade amplitude fidelity for alias suppression, if you release the strict allpass requirement, right? For the single-filter (Hilbert) phase method and the dual-filter differential phase method, one could design an approximate allpass, resp. allpass pair, with a "minimum distance in the complex plane" design constraint instead of a "minimum distance on the unit circle" design constraint and probably achieve the same alias suppression with less filter order. (About half the filter order??) Such a filter optimization should be possible -- but how??? Probably some modification of some standard IIR design algorithm would do. For Weaver's method, the relaxed constraint is inherent. Any normal lowpass design criteria will do. But Weaver needs two identical filters, so you have only half as many degrees of freedom for the filter parameters compared to the differential phase method. The single Hilbert transformer still has the constraint that the phase be 90 degrees, not arbitrary, and that costs. So, I guess the differential phase method with the relaxed constraint will be the "winner". What do you think? Would be nice if someone could do a filter pair design after this approach for assorted filter orders and give us the coefficient sets. Thank you. ;-) From ben_j_gillett at yahoo.co.uk Wed May 14 07:26:34 2008 From: ben_j_gillett at yahoo.co.uk (Ben Gillett) Date: Wed May 14 07:26:48 2008 Subject: [music-dsp] Compressing phase vocoder data Message-ID: <251789.85558.qm@web27810.mail.ukl.yahoo.com> Hi, I have written a phase vocoder, and would like to store the data on disk in this representation.? I'm using an overlap of 4, and the number of bands can be either 128, 256, 512 or?1024.? I've found that using 10 bits to represent the amplitude and 6 bits to represent the?deviation from the bin frequency gives good quality results.? Currently the phase vocoder data uses up twice as?many bytes?as the equivalent audio file (at 16bit, 44kHz).? I would like to compress it down so that it uses about the same amount of space as the audio file, in such a way that it sounds indistinguishable or very similar to the original file.? I would like the decompression algorithm to be very fast, and would like to implement the compression in a couple of days coding time at most.? I've searched on the web for any relevant information, but have come up with a blank so far.? Can anyone point me to relevant information about this problem, or alternatively give me the benefit of their experience?? The things that I was thinking of trying were storing an amplitude gain per time instant, so that fewer bits could be used to represent the amplitude component.? Also, I was considering assigning different numbers of bits to amp and freq depending on the frequency of the bin, on the assumption that higher frequency bins could be represented less accurately.? I was also wondering whether it might be possible to store amplitude data less frequently in either the time or frequency dimension for higher frequency bins without noticeable degredation in quality.? I was also considering using ADPCM with uLaw compression to compress both the amplitudes and frequencies.? If anyone has any thoughts on how well they think any of the above might work, I'd be very grateful for your thoughts. Thanks Ben From padawan12 at obiwannabe.co.uk Wed May 14 08:33:59 2008 From: padawan12 at obiwannabe.co.uk (Andy Farnell) Date: Wed May 14 08:34:09 2008 Subject: [music-dsp] Compressing phase vocoder data In-Reply-To: <251789.85558.qm@web27810.mail.ukl.yahoo.com> References: <251789.85558.qm@web27810.mail.ukl.yahoo.com> Message-ID: <20080514133359.782bc0da.padawan12@obiwannabe.co.uk> On Wed, 14 May 2008 11:26:34 +0000 (GMT) Ben Gillett wrote: > I was also wondering whether it might be possible to store amplitude > data less frequently in either the time or frequency dimension for higher > frequency bins without noticeable degredation in quality.? I was also > considering using ADPCM with uLaw compression to compress both the > amplitudes and frequencies.? This seems the most useful data reduction approach. The first thing that came to mind is fitting breakpoint line envelopes as is often done with additive resynthesis data. You may be able to find a ready solution and hack it to your data format. -- Use the source From douglas at music.columbia.edu Thu May 15 00:00:00 2008 From: douglas at music.columbia.edu (douglas repetto) Date: Thu May 15 00:00:05 2008 Subject: [music-dsp] [admin] music-dsp FAQ Message-ID: <20080515040000.C4B8F26D58CC6@music.columbia.edu> Hi, Just a reminder that if you are new to the list you should read the music-dsp FAQ. It contains answers to both technical _and_ adminstrative questions that often come up on the list. If your question appears in the FAQ it is safe to assume that it has been discussed on the list many times in the past, and you should probably have a look through the list archives before posting your question to the list. http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html Also of interest to new and not-so-new list members: The music-dsp list archives http://music.columbia.edu/cmc/music-dsp/musicdsparchives.html The music-dsp source code archive http://www.musicdsp.org music-dsp books and reviews http://music.columbia.edu/cmc/music-dsp/dspbooks.html All this and more at: http://music.columbia.edu/cmc/music-dsp Hasta la pasta, douglas (this is an automated message sent out on the 1st and 15th of each month) From Gualtiero.Volpe at unige.it Fri May 23 04:30:57 2008 From: Gualtiero.Volpe at unige.it (Gualtiero Volpe) Date: Fri May 23 04:31:18 2008 Subject: [music-dsp] NIME2008 Program is now available Message-ID: <007b01c8bcaf$548fd470$fdaf7d50$@Volpe@unige.it> Apologies for possible multiple posting. ========================================= NIME'08 is only 2 weeks away! The full program is now available at: http://nime2008.casapaganini.org/ Please have a look at all the excellent papers, performances, installations and events planned. Make sure you have registered if you haven't already. Please pass along this information to colleagues that would be interested. We are looking forward to seeing you at NIME'08 in Genova! The NIME Organizing Committee From d.sbragion at infotecna.it Wed May 28 12:42:15 2008 From: d.sbragion at infotecna.it (Denis Sbragion) Date: Wed May 28 12:42:43 2008 Subject: [music-dsp] [ANN] DRC 2.7.0 Message-ID: <2942.192.168.1.15.1211992935.squirrel@www.infotecna.lcl> Hello, DRC 2.7.0 is out and is available at: http://drc-fir.sourceforge.net/ Here are the release notes: A new method for the computation of the excess phase component inverse, based on a simple time reversal, has been introduced. The sample configuration files have been rewritten to take advantage of the new inversion procedure. Sample configuration files for 48 KHz, 88.2 KHz, 96 KHz sample rates have been added. The homomorphic deconvolution procedure has been improved to avoid any numerical instability. A new Piecewiswe Cubic Hemite Interpolating Polynomial (PCHIP) interpolation method, providing monotonic behaviour, has been introduced in the target response computation. All the interpolation and approximation procedures have been rewritten from scratch to provide better performances and accuracy. Bye, -- Denis Sbragion InfoTecna Tel: +39 0362 805396, Fax: +39 0362 805404 URL: http://www.infotecna.it From stephen.blinkhorn at audiospillage.com Fri May 30 22:48:39 2008 From: stephen.blinkhorn at audiospillage.com (Stephen Blinkhorn) Date: Fri May 30 22:48:56 2008 Subject: [music-dsp] Paper on bandlimited analog waveform synthesis using FM Message-ID: Hi Peter, I just found an old thread on Music-DSP discussing a paper you wrote regarding synthesizing sawtooth wave forms via FM. I'd love to read it. Is it available anywhere? Thanks, Stephen. From Victor.Lazzarini at nuim.ie Sat May 31 03:04:59 2008 From: Victor.Lazzarini at nuim.ie (victor) Date: Sat May 31 03:05:13 2008 Subject: [music-dsp] Paper on bandlimited analog waveform synthesis using FM References: Message-ID: <002201c8c2ec$a3f0b230$0201a8c0@family> I also just checked out that thread, fascinating discussion on the perceptual merits of the method! On that matter, we have developed a method to generate sawtooth & other typical 'analogue' waves which is also (approximately) bandlimited (very low aliasing, -90 dB) using what we call "modified FM". This is a version of FM whose expansion is based on modified Bessels (which is not exactly FM anymore, but damn it, we like the name and it can be traced back to the original FM/PM easily :). The advantage of this is that modified Bessels, unlike their cousins, provide a better behaved spectrum with regard to changes in the index of modulation. Besides, we can get a very good sawtooth, that not only sounds it, but looks like it too in the time-domain. If you are coming to DAFx in Finland, you can check it out there, otherwise, it'll be in the online proceedings. Victor ----- Original Message ----- From: "Stephen Blinkhorn" To: Sent: Saturday, May 31, 2008 3:48 AM Subject: [music-dsp] Paper on bandlimited analog waveform synthesis using FM > Hi Peter, I just found an old thread on Music-DSP discussing a paper you > wrote regarding synthesizing sawtooth wave forms via FM. I'd love to > read it. Is it available anywhere? > > Thanks, Stephen. > -- > 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