Re: [music-dsp] Fourier and its negative exponent

On 10/7/15 3:02 PM, Theo Verelst wrote: Stijn Frishert wrote: Hey all, In trying to get to grips with the discrete Fourier transform, Depending on how deep you want to study/understand the subject, get a good textbook on the subject, like "The Fourier Transform and its Applications" from

Re: [music-dsp] Fourier and its negative exponent

On 10/5/15 9:28 AM, Stijn Frishert wrote: In trying to get to grips with the discrete Fourier transform, I have a question about the minus sign in the exponent of the complex sinusoids you correlate with doing the transform. The inverse transform doesn’t contain this negation and a quick

Re: [music-dsp] Fourier and its negative exponent

On 10/5/15 5:40 PM, robert bristow-johnson wrote: about an hour ago i posted to this list and it hasn't shown up on my end. okay, something got lost in the aether. i am reposting this: On 10/5/15 9:28 AM, Stijn Frishert wrote: In trying to get to grips with the discrete Fourier transform

[music-dsp] test (sorry)

about an hour ago i posted to this list and it hasn't shown up on my end. -- r b-j r...@audioimagination.com "Imagination is more important than knowledge." ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] test (sorry)

On 10/5/15 5:58 PM, Stijn Frishert wrote: > Your mail (the first copy) was well received and is still ringing through my > mind. Especially the part about -j and +j having equal claim to square to -1 > is an eye opener. check out "Imaginary unit" at Wikipedia. > I'm still thinking about the

Re: [music-dsp] Fast convolution with synthesis window

On 10/4/15 12:36 PM, Earl Vickers wrote: rbj wrote: why would you *want* to use a synthesis window if you're doing OLA fast-convolution? Good question. [ ] it might be a very nice way to have a changing filter kernel and have it sound nice in the transitions. Good answer! Yes, I’m doing

Re: [music-dsp] Fast convolution with synthesis window

On 10/3/15 1:06 PM, Earl Vickers wrote: Is there any way to do STFT-based fast convolution using a (non-rectangular) synthesis window? do you mean analysis window instead of synthesis window? No, I mean using a synthesis window, as with WOLA or Griffin/Lim (in addition to an analysis

Re: [music-dsp] Fast convolution with synthesis window

On 10/2/15 11:29 AM, Earl Vickers wrote: Is there any way to do STFT-based fast convolution using a (non-rectangular) synthesis window? do you mean analysis window instead of synthesis window? sure, but the rules of overlap-add fast convolution still have to be followed. the length of

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

if, by a 1024 window that means you are looking at your original samples from x[n-512] up to x[n+512], i would say that it's overkill. sometimes, when doing interpolation there is an upsampling factor implied. if there is an upsampling factor of, say, 32 in there, and you're really only

Re: [music-dsp] warts in JUCE

one of the emails. i disagree. it is what it almost is. On 9/6/15 9:03 AM, Chris Cannam wrote: On Sun, Sep 6, 2015, at 01:50 PM, robert bristow-johnson wrote: otherwise, you're always going to be passing sampleRate along with every AudioSampleBuffer. This bit here -- this is the part th

Re: [music-dsp] warts in JUCE

On 9/6/15 12:59 AM, Ross Bencina wrote: On 6/09/2015 8:37 AM, Daniel Varela wrote: sample rate is part of the audio information so any related message ( AudioSampleBuffer ) should provide it, no need to extend the discursion. There's more than one concept at play here: i very much agree.

Re: [music-dsp] warts in JUCE

On 9/4/15 8:58 AM, mdsp wrote: On 04/09/15 02:44, robert bristow-johnson wrote: In both cases the sampling rate is already available before the processing starts using prepareToPlay(int samplesPerBlockExpected, double sampleRate). Having it stored on AudioSampleBuffer while handy would

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

On 9/4/15 1:31 PM, Andrew Kelley wrote: On Fri, Sep 4, 2015 at 10:30 AM Alexandre Pages > wrote: Yes, why re-invent the wheel over and over again? I prefer round wheels :-) woot! i guess i'm gonna have to check this out. --

Re: [music-dsp] warts in JUCE

last post on this subject. understood. On 04/09/15 17:27, robert bristow-johnson wrote: oh, that's interesting. so Jules also prefers non-backward-compatible changes in JUCE. i didn't know that. doesn't make that much sense to me. iif the change is a disruptive one that could have important si

Re: [music-dsp] warts in JUCE

On 9/4/15 4:35 PM, Chris Cannam wrote: (I have never used JUCE, don't know its interfaces, and have no side to take) On Fri, Sep 4, 2015, at 08:38 PM, robert bristow-johnson wrote: i find it odd that it seems to all inherent to a parcel of sound represented in a computer are the number

Re: [music-dsp] warts in JUCE

On 9/3/15 5:57 AM, mdsp wrote: As a long-time JUCE user and observer let me give you my opinion regarding AudioSampleBuffer. thank you. i hope it's okay if i respond (and disagree, respectfully). now, i want us to be clear about the definition of "backward compatible". Google defines it

[music-dsp] warts in JUCE (was Re: Implementing GMM for voice diarization on iOS and Android)

On 9/2/15 7:13 AM, Jean-Baptiste Thiebaut wrote: I'd recommend JUCE (juce.com ). There's already support for iOS and Android, and it's about to get better. And it's all cross platform C++, so you can do real-time applications. (full disclosure: I'm the product manager for

Re: [music-dsp] warts in JUCE

On 9/2/15 5:14 PM, Tom Duffy wrote: On 9/2/2015 1:48 PM, robert bristow-johnson wrote: On 9/2/15 7:13 AM, Jean-Baptiste Thiebaut wrote: I'd recommend JUCE (juce.com). There's already support for iOS and Android, and it's about to get better. And it's all cross platform C++, so you can do

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/26/15 9:47 PM, Ethan Duni wrote: 15.6 dB + (12.04 dB) * log2( Fs/(2B) ) Oh I see, you're actually taking the details of the sinc^2 into account. really, just the fact that the sinc^2 has nice deep zeros at every integer multiple of Fs (except 0). What I had in mind was more of a

Re: [music-dsp] [admin] list etiquette

On 8/28/15 9:23 AM, Peter S wrote: You're speaking about an event that happened in the past. it appears to be ongoing. Which has nothing to do with the present, or the future, or the accessibility of this mailing list. You can learn from the mistakes or faults of others, can't you? For

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/25/15 7:08 PM, Ethan Duni wrote: if you can, with optimal coefficients designed with the tool of your choice, so i am ignoring any images between B and Nyquist-B, upsample by 512x and then do linear interpolation between adjacent samples for continuous-time interpolation, you can show

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/24/15 11:18 AM, Sampo Syreeni wrote: On 2015-08-19, Ethan Duni wrote: and it doesn't require a table of coefficients, like doing higher-order Lagrange or Hermite would. Robert I think this is where you lost me. Wasn't the premise that memory was cheap, so we can store a big prototype

Re: [music-dsp] [admin] list etiquette

On 8/23/15 7:32 AM, Peter S wrote: Well, no thanks. I don't think that would work out well. I'll rather keep my thoughts to myself. hey Peter, why don't you come over to the USENET newsgroup comp.dsp and see how nice we are there. one interesting Russian-American, Vlad, might engage you,

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/19/15 1:43 PM, Peter S wrote: On 19/08/2015, Ethan Duniethan.d...@gmail.com wrote: But why would you constrain yourself to use first-order linear interpolation? Because it's computationally very cheap? and it doesn't require a table of coefficients, like doing higher-order Lagrange or

Re: [music-dsp] Mails with images?

On 8/18/15 6:15 AM, STEFFAN DIEDRICHSEN wrote: As it seems, it’s not a technical hurdle. yay!!! now let's see if we can get some LaTeX math markup going here and we'll have somethin... (other than Stack Exchange). -- r b-j r...@audioimagination.com Imagination is more

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/18/15 4:28 PM, Peter S wrote: 1, -1, 1, -1, 1, -1 ... is a proper bandlimited signal, and contains no aliasing. That's the maximal allowed frequency without any aliasing. well Peter, here again is where you overreach. assuming, without loss of generality that the sampling period is 1,

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/18/15 4:50 PM, Nigel Redmon wrote: I’m sorry, I’m missing your point here, Peter (and perhaps I missed Roberts, hence the “No?” in my reply to him). The frequency response of linear interpolation is (sin(pi*x)/(pi*x))^2, -7.8 dB at 0.5 of the sample rate... i will try to spell out my

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/18/15 5:01 PM, Emily Litella wrote: ... Never mind. too late. :-) -- r b-j r...@audioimagination.com Imagination is more important than knowledge. ___ music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/18/15 3:44 PM, Ethan Duni wrote: Assume you have a Nyquist frequency square wave: 1, -1, 1, -1, 1, -1, 1, -1... The sampling theorem requires that all frequencies be *below* the Nyquist frequency. Sampling signals at exactly the Nyquist frequency is an edge case that sort-of works in

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/17/15 12:07 PM, STEFFAN DIEDRICHSEN wrote: I could write a few lines over the topic as well, since I made such a compensation filter about 17 years ago. So, there are people, that do care about that topic, but there are only some, that do find time to write up something. ;-) Steffan

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/17/15 2:39 PM, Nigel Redmon wrote: Since compensation filtering has been mentioned by a few, can I ask if someone could get specific on an implementation (including a description of constraints under which it operates)? I’d prefer keeping it simple by restricting to linear interpolation,

Re: [music-dsp] Compensate for interpolation high frequency signal loss

On 8/16/15 4:09 AM, Sham Beam wrote: Hi, Is it possible to use a filter to compensate for high frequency signal loss due to interpolation? For example linear or hermite interpolation. Are there any papers that detail what such a filter might look like? besides the well-known sinc^2

Re: [music-dsp] Non-linearity or filtering

On 8/10/15 10:02 AM, Peter S wrote: On 10/08/2015, robert bristow-johnsonr...@audioimagination.com wrote: the thing that i *think* Peter is missing in this is the same as some of the early manufacturers when they truncated the 30-bit words (or whatever they had in the decimation filters) to 18

Re: [music-dsp] [ot] about entropy encoding

On 8/9/15 5:07 PM, Sampo Syreeni wrote: On 2015-07-18, robert bristow-johnson wrote: even so, Shannon information theory is sorta static. it does not deal with the kind of redundancy of a repeated symbol or string. In fact it does so fully, really? like run-length encoding? and i've

Re: [music-dsp] Non-linearity or filtering

signals, like the one Robert Bristow-Johnson mentioned just a couple of months ago, the DAC-designer's nightmare: ..., +1, -1, +1, +1, -1, +1, ... Even if you bury something like that well within noise, it's going to pop up upon proper reconstruction, and not pop up if you fucked up your modulator

Re: [music-dsp] Non-linearity or filtering

On 7/25/15 10:57 AM, Tom Duffy wrote: You didn't change the bandwidth. If the target signal is max 30Hz and you have a 192kHz sampler, you low pass at 2x your max frequency (60Hz, but lets say 100Hz for convenience) using a brick wall digital filter (processed at 192kHz). Then you do a

Re: [music-dsp] The Art of VA Filter Design book revision 1.1.0

hey Vadim, i love the rigor in your paper. i'm still looking through it. in the 2nd-order analog filters, i might suggest replacing 2R with 1/Q in all of your equations, text, and figures because Q is a notation and parameter much more commonly used and referred to in either the EE or

Re: [music-dsp] Non-linearity or filtering

On 7/23/15 4:38 PM, Peter S wrote: ... https://en.wikipedia.org/wiki/The_Paradox_of_Choice You're welcome. http://www.imdb.com/title/tt1386011/ -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp

Re: [music-dsp] Non-linearity or filtering

On 7/23/15 1:12 AM, Peter S wrote: On 23/07/2015, robert bristow-johnsonr...@audioimagination.com wrote: okay, since there is no processing, just passing the signal from A/D to D/A converter, there is only one quantization operation, at the A/D. if it's an old-fashioned conventional A/D, the

Re: [music-dsp] Non-linearity or filtering

On 7/23/15 3:36 AM, Peter S wrote: Also if you fail to notice that the current year is 2015, and the rules you learned 20 years ago for 8-bit and 16-bit converters do not necessarily apply for today's typical 24-bit converters (that usually have several bits of noise in the lowest bits),

Re: [music-dsp] Non-linearity or filtering

i wrote: the *major* component of audible noise is coming from the numerical processes inside the codec On 7/23/15 12:43 PM, Peter S wrote: Seriously, where do you get that from? well, i take it that the answer to the question i asked is no. so there are a few docs on the web like at

Re: [music-dsp] A little frivolous diversion on the effect of using a delay

On 7/20/15 7:49 PM, Nigel Redmon wrote: To add to Robert’s comment on discrete-time analog… The only thing special about digital sampling is that it’s stable (those digital numbers can be pretty durable—the analog samples don’t hold up so well) and convenient for computation. But the digital

Re: [music-dsp] A little frivolous diversion on the effect of using a delay

On 7/20/15 3:00 PM, jpff wrote: The first delay of which I was aware was in the piece Echo III played on the viola by Tim Souster in Cambridge in the early 1970s. Not an echo or reverb but a cannon. Delay was via two reel-to-reel tape machines, with a carefully measured distance between them.

Re: [music-dsp] A little frivolous diversion on the effect of using a delay

On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD

Re: [music-dsp] A little frivolous diversion on the effect of using a delay

On 7/20/15 4:52 PM, Theo Verelst wrote: robert bristow-johnson wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines just for the record, none of them content words were written by me

Re: [music-dsp] about entropy encoding

On 7/18/15 10:10 PM, Peter S wrote: It follows from the above, that Shannon's entropy model is a simplified, idealized model of information, that pretends that algorithms have zero length and thus no entropy, and you can magically share a codebook telepathically without actually transmitting it.

Re: [music-dsp] about entropy encoding

On 7/17/15 1:26 AM, Peter S wrote: On 17/07/2015, robert bristow-johnsonr...@audioimagination.com wrote: in your model, is one sample (from the DSP semantic) the same as a message (from the Information Theory semantic)? A message can be anything - it can be a sample, a bit, a combination of

Re: [music-dsp] about entropy encoding

On 7/17/15 2:28 AM, Peter S wrote: Dear Ethan, You suggested me to be short and concise. My kind recommendation to you: 1) Read A Mathematical Theory of Communication. 2) Try to understand Theorem 2. 3) Try to see, when p_i != 1, then H != 0. I hope this excercise will help you grasp this

Re: [music-dsp] about entropy encoding

On 7/17/15 12:08 AM, Peter S wrote: On 17/07/2015, Peter Speter.schoffhau...@gmail.com wrote: Think of it as this - if your receiver can distinguish only two different sets of parameters, then you need to send at least *one* bit to distinguish between them - '0' meaning square wave A, and '1'

Re: [music-dsp] about entropy encoding

On 7/15/15 11:39 AM, Peter S wrote: So far my best entropy estimator algorithm using sophisticated correlation analysis, i've only been following this coarsely, but did you post, either in code or pseudo-code the entropy estimator algorithm? i'd be curious. gave entropy rate of 1 for

Re: [music-dsp] Sampling theorem extension

On 6/29/15 6:43 PM, Sampo Syreeni wrote: On 2015-06-29, Emanuel Landeholm wrote: But all waveforms can be antialiased by brick wall filtering, ie. sine cardinal interpolation. The point is that you can't represent the continuous time waveforms in the usual sampled form, and then apply a

Re: [music-dsp] Sampling theorem extension

On 6/19/15 5:03 PM, Sampo Syreeni wrote: On 2015-06-19, Ethan Duni wrote: I guess what we lose is the model of sampling as multiplication by a stream of delta functions, but that is more of a pedagogical convenience than a basic requirement to begin with. pedagogical convenience,

Re: [music-dsp] [ot] other than sampling theorem, Theo

On 6/11/15 1:20 PM, Sampo Syreeni wrote: On 2015-06-11, Theo Verelst wrote: [...] I don't recommend any of the guys I've read from here to presume they'll make it high up the mathematical pecking order by assuming all kinds of previous century generalities, while being even more imprecise

Re: [music-dsp] Sampling theorem extension

On 6/11/15 5:39 PM, Sampo Syreeni wrote: On 2015-06-09, robert bristow-johnson wrote: BTW, i am no longer much enamoured with BLIT and the descendents of BLIT. eventually it gets to an integrated (or twice or 3 times integrated) wavetable synthesis, and at that point, i'll just do

Re: [music-dsp] Sampling theorem extension

On 6/9/15 4:32 AM, Vadim Zavalishin wrote: Creating a new thread, to avoid completely hijacking Theo's thread. it's a good idea. Previous message here: http://music.columbia.edu/pipermail/music-dsp/2015-June/073769.html On 08-Jun-15 18:29, Sampo Syreeni wrote: On

Re: [music-dsp] R: Comb filter decay wrt. feedback

On 5/11/15 3:25 AM, Marco Lo Monaco wrote: we should have a drink together (do you come to U.S. AES shows?) and all you do is mention one of several political topics and the hyperbole coming outa me will be worse than this. Hey, I also wanna have a drink with RBJ and talk about life,

Re: [music-dsp] Comb filter decay wrt. feedback

On 5/10/15 3:42 PM, Matthias Puech wrote: I have a recursive comb filter, implemented with a simple delay line of size N and feedback F in [0..1]. If feedback is high and I ping it, it decays exponentially as it should, to give the typical ringing effect. The decay time D is also proportional

Re: [music-dsp] Comb filter decay wrt. feedback

On 5/10/15 3:42 PM, Matthias Puech wrote: This is my first post on this list, although I have been reading it for quite some time, with great interest. welcum I am a CS researcher in an unrelated field, but fascinated for as long as I can remember by DSP and sound synthesis.

Re: [music-dsp] Comb filter decay wrt. feedback

On 5/10/15 8:00 PM, Peter S wrote: On 10/05/2015, robert bristow-johnsonr...@audioimagination.com wrote: it's not trivial and it's not in the basic lit. It's trivial, and it's in the 1961 Schroeder paper linked in the previous post. apologies for the hyperbole. if you consider both

[music-dsp] Wow!!! Andy Moorer on All Things Considered!!!!

not to late for you guys living on the left coast to hear it. just ended two minutes ago (21:44 GMT). -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ,

Re: [music-dsp] Wow!!! Andy Moorer on All Things Considered!!!!

On 4/11/15 6:46 PM, Vicki Melchior wrote: It should appear in their listings and archives but I don't see anything very obvious on the NPR-ATC website for Apr 11 : http://www.npr.org/programs/all-things-considered/ What was it about? sorry, just now clicked your link, Vickie. it's this:

Re: [music-dsp] Wow!!! Andy Moorer on All Things Considered!!!!

On 4/11/15 8:20 PM, Chad Wagner wrote: Haha - see also: http://createdigitalmusic.com/2015/04/thx-deep-note-creator-remade-iconic-sound/ The original 30-year-old C program is 325 lines, and the “patch” file for the synthesizer was 298 more lines. I guess it just felt like 20,000 lines when I

Re: [music-dsp] Uses of Fourier Synthesis?

On 4/5/15 5:21 PM, Theo Verelst wrote: In the context of synthesis, or intelligent multi sampling with complicated signal issues, you could try to make the FFT analysis and filtering a targeted part of the synthesis path, so that the playing back of samples contain variations and sample

Re: [music-dsp] Uses of Fourier Synthesis?

On 4/5/15 10:15 PM, Didier Dambrin wrote: That's not at all what I was saying. I only wrote that additive synthesis could be used for an identical result to a KS, with better control on the filtering. please accept my apology. i did not correctly grok what you were saying Didier. of

Re: [music-dsp] Uses of Fourier Synthesis?

: robert bristow-johnson Sent: Sunday, April 05, 2015 9:23 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Uses of Fourier Synthesis? On 4/5/15 3:11 PM, Didier Dambrin wrote: I've created plucked strings using additive (not FFT), and it sounds the same as a Karplus Strong

Re: [music-dsp] Uses of Fourier Synthesis?

On 4/5/15 2:45 PM, Alan Wolfe wrote: Interesting, thanks for the info! What is the usual technique for simulating plucked strings? (: i dunno what is the most common (probably sample playback), but lately Physical Modeling seems to be the craze. Karplus-Strong is another method that

Re: [music-dsp] Uses of Fourier Synthesis?

On 4/5/15 2:51 PM, Zhiguang Zhang wrote: Plucked strings can be done using the Karplus Strong algorithm, which uses a noise excitation put through a comb filter, or something similar. it's the kind of comb filter with feedback. and you need fractional-sample precision delay before feedback.

Re: [music-dsp] Uses of Fourier Synthesis?

On 4/5/15 3:11 PM, Didier Dambrin wrote: I've created plucked strings using additive (not FFT), and it sounds the same as a Karplus Strong, but with more control. So it's definitely doable. The key is: all oscillators have to be phase-unrelated, or it won't sound metallic. are the

Re: [music-dsp] Real-time Polyphonic Pitch Detection

On 4/3/15 5:12 PM, Alex Cannon wrote: Hi all, I was wondering what the state-of-the-art was with regards to real-time polyphonic pitch-detection. The best solution for the MIREX task

Re: [music-dsp] Real-time Polyphonic Pitch Detection

On Fri, April 3, 2015 6:32 pm Matt Jackson matt.jack...@ableton.com wrote: I wasn't aware ~fiddle works polyphonically. I'm not an expert on the subject, but according to a few AES papers I glazed over, I think you find one pitch, remove it with notches and iterate. polyphonic is

Re: [music-dsp] Fwd: Array indexing in Matlab finally corrected after 30 years!

On 4/2/15 12:20 PM, Phil Burk wrote: Speaking of zero based indexing, my neighbor's street address is 0 Meadowood Drive. There was a 4 Meadowood Drive already existing. They left room to build one more house at the end of the street. But instead of building a house they built two cottages. So

Re: [music-dsp] Array indexing in Matlab finally corrected after 30 years!

On 4/1/15 12:56 PM, Nigel Redmon wrote: On Apr 1, 2015, at 7:19 AM, robert bristow-johnsonr...@audioimagination.com wrote: On 4/0/15 6:24 AM, Max wrote: Well Played. credit Dilip Sarwate at comp.dsp (who also hangs out at the dsp.stackexchange forum). -- r b-j

Re: [music-dsp] oversampled Fourier Transform

On 3/31/15 6:53 PM, Justin Salamon wrote: To expand on what Ethan wrote, it sounds like what you're trying to do is zero-pad the signal: http://www.dsprelated.com/dspbooks/mdft/Zero_Padding.html That said, whilst zero padding will give you an interpolated spectrum in the frequency domain, you

Re: [music-dsp] R: Glitch/Alias free modulated delay

On 3/20/15 2:58 PM, Alan Wolfe wrote: One thing to watch out for is to make sure you are not looking backwards AND forwards in time, but only looking BACK in time. right!! pretty hard to cross into the future with a real-time alg (because there ain't no future, it's really wrapping around

Re: [music-dsp] Glitch/Alias free modulated delay

On 3/20/15 10:20 AM, Bjorn Roche wrote: Interpolating the sample value is not sufficient to eliminate artifacts. You also need to eliminate glitches that occur when jumping from one time value to another. what do you mean? do you mean the normal speed-up-the-tape effect when smoothly

Re: [music-dsp] Glitch/Alias free modulated delay

On 3/19/15 2:28 PM, Nuno Santos wrote: Hi, Thanks for your replies. What I hear is definitely related with the modulation. The artefacts are audible every time the modulation is applied: manually or automatically (please not that I have an interpolator for manual parameter changes to avoid

Re: [music-dsp] Approximating convolution reverb with multitap?

On 3/18/15 1:39 PM, STEFFAN DIEDRICHSEN wrote: With most IRs, you don’t see discrete peaks, it’s a continuous signal. This is due to the response of the speaker and microphone being used. This causes some smear. sure, so let's say we deconvolve that smear because we think it's a lot

Re: [music-dsp] Audio Latency Test App for iOS and Android

On 3/2/15 4:29 PM, Patrick Vlaskovits wrote: Hiya! We've released a free app for Android and iOS developers that measures roundtrip audio latency. http://superpowered.com/latency/ and you're doing all of your bookkeeping accurately so that the app itself isn't adding delay that it includes

Re: [music-dsp] Linearity of compression algorithms on more than one sound component

On 2/12/15 3:02 PM, Theo Verelst wrote: Hi all, Just a thought I share, because of associations I won't bother you with, suppose you take some form of audio compression, say Fmp3(wav) which transforms wav to an mp3 form, with some encoding parameters. Now we consider the linearity of the

Re: [music-dsp] Dither video and articles

On 2/10/15 8:49 AM, Didier Dambrin wrote: What are you talking about - why would phase not matter? It's extremely important (well, phase relationship between neighboring partials). well, it's unlikely you'll be able to hear the difference between this: x(t) = cos(wt) - 1/3*cos(3wt) +

Re: [music-dsp] Dither video and articles

On 2/9/15 10:19 PM, Nigel Redmon wrote: But it matters, because the whole point of dithering to 16bit depends on how common that ability is. Depends on how common? I’m not sure what qualifies for common, but if it’s 1 in 100, or 5 in 100, it’s still a no-brainer because it costs nothing,

Re: [music-dsp] Dither video and articles

On 2/10/15 1:30 PM, Didier Dambrin wrote: Of course 24bit isn't a bad idea for intermediate files, but 32bit float is a better idea, even just because you don't have to normalize store gain information that pretty much no app will read from the file. And since the price of storage is

Re: [music-dsp] Dither video and articles

On 2/10/15 1:51 PM, Ethan Duni wrote: So to you, that Pono player isn't snake oil? It's more the 192kHz sampling rate that renders the Pono player into snake oil territory. The extra bits probably aren't getting you much, but the ridiculous sampling rate can only *hurt* audio quality, while

Re: [music-dsp] Dither video and articles

On 2/7/15 8:54 AM, Vicki Melchior wrote: Well, the point of dither is to reduce correlation between the signal and quantization noise. Its effectiveness requires that the error signal has given properties; the mean error should be zero and the RMS error should be independent of the signal. The

Re: [music-dsp] Dither video and articles

Original Message Subject: Re: [music-dsp] Dither video and articles From: Vicki Melchior vmelch...@earthlink.net Date: Fri, February 6, 2015 2:23 pm To: A discussion list for music-related DSP music-dsp@music.columbia.edu

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

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

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

Re: [music-dsp] R: Thoughts on DSP books and neural networks

On 2/5/15 6:48 PM, Marco Lo Monaco wrote: I dont know very much about US IP laws but in Italy (EU), reverse engineering in illegal unless for interoperability issues. You can understand an algorithm only by measuring in-out relationships (very difficult to understand the details of an algo by

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

[music-dsp] 14-bit MIDI controls, how should we do Course and Fine?

this has probably been settled long ago, but i cannot from the standard, make perfect sense of it. so the MIDI 1.0 Spec says: ___ ... STATUSDATA BYTESDESCRIPTION ... ... ... 1011

[music-dsp] 14-bit MIDI controls, how should we do Coarse and Fine?

i'm resending this because of unexpected word wrapping and, as much as i intended to avoid it, a persistent misspelling. so this may have been settled long ago, but i cannot from the standard, make perfect sense of it. so the MIDI 1.0 Spec says:

Re: [music-dsp] 14-bit MIDI controls, how should we do Coarse and Fine?

On 2/4/15 4:50 PM, STEFFAN DIEDRICHSEN wrote: There's an easy trick to overcome this. Put the MSB into the upper and lower 7 bits. ??? If an LSB comes along, put it into the lower bits. This way, you reach with the MSB alone the maximum value of 0x3FFF. If a LSB comes along, it's welcome

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

i am mostly just listening here. i'll get back to you Ross. when filters blow up because of varying coefficients, don't they settle down some finite time after the coefs stop varying? On 2/2/15 6:21 PM, Stefan Sullivan wrote: I actually found by playing around with a particular biquad

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

On 2/3/15 7:07 PM, Ethan Duni wrote: well, the output states, y[n-1] and y[n-2], will change if coefs change. No, those have already been computed and (presumably) output at that point. It's true that these states won't match what they would have been if you'd been running the new coefficients

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

On 2/1/15 6:32 AM, Ross Bencina wrote: Hello Alan, On 1/02/2015 4:51 AM, Alan O Cinneide wrote: Dear List, While filtering an audio stream, I'd like to change the filter's characteristics. You didn't say what kind of filter, so I'll assume a bi-quad section. In order to do this without

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

On 12/24/14 4:32 AM, Nigel Redmon wrote: Naw, mhos is a one-off. It's fun, pronounceable, and in common use (since 1883!). Don't get carried away. Besides, it makes me think of The Three Stooges, and smile. Siemens makes me think of...er, um—oh Q: so what's long and hard and full of Siemen?

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

On 12/23/14 11:36 PM, Andrew Simper wrote: On 24 December 2014 at 08:55, robert bristow-johnson r...@audioimagination.com wrote: the OTAs are there for voltage-controlled gain or, really, a voltage-controlled resistor to change the tuning of the VCF. from the datasheet They are an idealised

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

On 12/23/14 4:43 AM, Andrew Simper wrote: Everyone on the synth diy list didn't even bat an eyelid at that diagram, they just said stuff like Thanks and That's very interesting, I have not seen it done that way before. i'm not ashamed to point out what i don't know. you can't spell analysis

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

On 12/22/14 12:27 AM, Andrew Simper wrote: I've seen in many Sallen Key circuits people stick the input signal into various points to generate some different responses, but always the high pass is only 1 pole. i haven't seen that with the SK. for HPF, i've only seen it with the the R's and C's

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