Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

On Dec 6, 2010, at 1:23 PM, Stefan Stenzel wrote: On 06.12.2010 08:59, robert bristow-johnson wrote: This is a continuation of the thread started by Element Green titled: Algorithms for finding seamless loops in audio I suspect it works better to *construct* a seamless loop instead

Re: [music-dsp] Waveform Interpolation

On Dec 8, 2010, at 9:00 AM, cparodi.ug...@libero.it wrote: I am trying to quantify the advantage of more complex forms of interpolation over linear interpolation (in particular when reading samples to populate the audio buffer of a wavetable synthesizer, but not limited to that). just to

Re: [music-dsp] Waveform Interpolation

i just found out that music-dsp list server will not post with my attachment. so i am appending this thing that i've posted occasionally to comp.dsp whenever we have sampling and resampling and interpolation arguments. please contact me if you want this AES paper Duane Wise and i did in

Re: [music-dsp] Waveform Interpolation

On Dec 10, 2010, at 6:47 PM, Nigel Redmon wrote: BTW, not sure if anyone's pointed this out... i don't think anyone has... Another thing to realize, when comparing this like zero-order hold, linear interpolation, and sinc interpolation is that while all are lowpass filters, the "cutoff"

Re: [music-dsp] Waveform Interpolation

On Dec 11, 2010, at 12:26 PM, Nigel Redmon wrote: Naw, actually, right after sending that I started to type up a retraction and my girlfriend showed up, and she hates it when I'm not ready to go to dinner on time. My bad, I started to make the point that linear interpolation response is

Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

thanks, Stefan, for getting back on this. On Dec 13, 2010, at 5:57 AM, Stefan Stenzel wrote: I construct seemless loops in frequency domain in a non-realtime application, and I am quite happy with the results. If you ask for a recipe, this is what I am doing: - detect pitch of (whole) sam

[music-dsp] A wavetable alternative to adjusting the frequencies of harmonics to get seamless loops.

okay, i don't seem to get any time to deal with this except late at night. so this is continuing that thread that was named "A theory of optimal splicing of audio in the time domain." On Dec 15, 2010, at 11:20 AM, Stefan Stenzel wrote: On 14.12.2010 06:15, robert bristo

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 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. And

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 a

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

On 7/22/15 10:16 PM, Peter S wrote: You have your signal S. When you digitize that signal, you add the noise floor of the ADC (among other noises), let's call it N1. When you reconstruct the signal, you add the noise floor of the DAC (among other noises), let's call that N2. So you have S +

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

On 7/23/15 1:12 AM, Peter S wrote: On 23/07/2015, robert bristow-johnson 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 quantiz

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), Peter,

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] 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 maili

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 aud

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

30 Hz bandlimited signal. and that's not counting noise-shaping. On 7/25/15 3:25 AM, Peter S wrote: Okay, a few more thoughts: On 23/07/2015, robert bristow-johnson wrote: okay, since there is no processing, just passing the signal from A/D to D/A converter, there is only one quantization ope

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

d we don't want the modulator loop to fuck that stuff up. It will, unless its specs are down there as well. Not to mention some of those wonky edge case signals, like the one Robert Bristow-Johnson mentioned just a couple of months ago, the "DAC-designer's nightmare": .

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&#x

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

On 8/10/15 10:02 AM, Peter S wrote: On 10/08/2015, robert bristow-johnson 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 "meaningful"

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 rolloff

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 On

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 8/17/15 7:29 PM, Sampo Syreeni wrote: On 2015-08-17, robert bristow-johnson wrote: As I noted in the first reply to this thread, while it’s temping to look at the sinc^2 rolloff of a linear interpolator, for example, and think that compensation would be to boost the highs to undo the

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 i

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 so

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, t

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 po

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 https://lists.co

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

On 8/18/15 11:46 PM, Ethan Duni wrote: > for linear interpolation, if you are a delayed by 3.5 samples and you keep that delay constant, the transfer function is > > H(z) = (1/2)*(1 + z^-1)*z^-3 > >that filter goes to -inf dB as omega gets closer to pi. Note that this holds for symmetric fr

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 Duni 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 Hermite would. Th

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, but

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

On 8/24/15 11:13 AM, Stefan Sullivan wrote: Well that didn't take long roflao :-) On Mon, Aug 24, 2015 at 2:08 PM Peter S > wrote: On 24/08/2015, Theo Verelst mailto:theo...@theover.org>> wrote: > I'm not going to confuse etiquette with thinki

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 FIR

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 th

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 wor

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 exa

Re: [music-dsp] 20k

On 8/30/15 7:06 PM, Sampo Syreeni wrote: On 2015-08-30, Scott Gravenhorst wrote: This amounted to using a microphone to sample the effects of an impulse (starter's pistol or some such) on some audio environment like a church or concert hall, or even a rock face in a forest. The recording was

[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 JUCE

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 platfor

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 sim

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 be

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 mailto:alexandre.pa...@abeem.eu>> 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. -- r b-j r...

Re: [music-dsp] warts in JUCE

lute this mailing-list too much, so that will be my 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 t

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 numb

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

lly told me that in 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 he

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 co

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

On 9/11/15 3:25 PM, Nigel Redmon wrote: Great—glad to hear the articles were helpful, Nuno. (Website back up.) To build the oscillators tables I’m using that multi table technic you describe on your waveforms series posts where maxHarms is: int maxHarms = 44100.f / (3.0 * BASE_FREQ) + 0.5; Is

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

On 9/21/15 1:26 PM, Nigel Redmon wrote: Hi Robert, Yes, my answer was strictly for the special case of one table per octave. In that case, the start of the highest table (which would also be highest harmonic per table), such that transposing up by the maximum amount (in this case, one octave)

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 the

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 window).

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 searc

[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 ht

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 tran

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 con

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 ti

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 th

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

� i have to confess that this is hard and i don't have a concrete solution for you. �it seems to me that, by this description: � r[n] = uniform_random(0, 1) if (r[n] <= P) � �x[n] =�uniform_random(-1, 1); else �x[n] = x[n-1]; � from that, and from the assumption of ergodicity (where all time av

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ross Bencina" Date: Tue, November 3, 2015 11:51 pm To: music-dsp@music.columbia.edu ---

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

lumbia.edu -- > On 4/11/2015 9:39 AM, robert bristow-johnson wrote: >> i have to confess that this is hard and i don't have a concrete solution >> for you. > > Knowing that this isn't well known helps

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

lumbia.edu -- > On 4/11/2015 9:39 AM, robert bristow-johnson wrote: >> i have to confess that this is hard and i don't have a concrete solution >> for you. > > Knowing that this isn't well known helps

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ross Bencina" Date: Wed, November 4, 2015 12:22 am To: r...@audioimagination.com music-dsp@music.columbia.edu

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ross Bencina" Date: Wed, November 4, 2015 12:22 am To: r...@audioimagination.com music-dsp@music.columbia.edu

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

> I think I was slightly off when I said that the units of psd are power per > unit frequency -- since the whole signal has infinite power, � no, i don't think so. � > the units�really need to be power per unit frequency per unit time, which > (confusingly) is the same thing as power. � the

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ethan Duni" Date: Tue, November 10, 2015 8:58 pm To: "A discussion list for music-related DSP" ---

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ethan Duni" Date: Wed, November 11, 2015 5:57 pm To: "robert bristow-johnson" "A discussion l

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Ethan Duni" Date: Wed, November 11, 2015 7:36 pm To: "robert bristow-johnson" "A discussion l

Re: [music-dsp] confirm 2692e89dd013da35bd113d6f644fdcfa865054c3

Original Message Subject: Re: [music-dsp] confirm 2692e89dd013da35bd113d6f644fdcfa865054c3 From: "Douglas Repetto" Date: Thu, November 12, 2015 10:26 am To: "A discussion list for music-related DSP"

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

> Am 16.11.2015 20:00, schrieb Martin Vicanek: >> [..] the autocorrelation is >> >> = (1/3)*(1-P)^|k| >> >> (I checked that with a little MC code before posting.) So the power >> spectrum is (1/3)/(1 + (1-P)z^-1), i.e flat at DC and pink at higher >> frequencies. For reasonably small P

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

Original Message Subject: Re: [music-dsp] how to derive spectrum of random sample-and-hold noise? From: "Martin Vicanek" Date: Mon, November 16, 2015 3:50 pm To: music-dsp@music.columbia.edu -

Re: [music-dsp] confirm 2692e89dd013da35bd113d6f644fdcfa865054c3

� just to let everyone (particularly Douglas) know that our friendly troll is again trying to send people a hint that the troll wants us off the list. � either that or there is something wrong with the list server. Original Message

[music-dsp] automation of parametric EQ .

� � can anyone point me where to find the technical information regarding how automation regarding control settings might be defined, particularly regarding the 3-knob parametric EQ. � first, other than MIDI, with automation data stored as a MIDI files, how else is this information stored? �li

Re: [music-dsp] automation of parametric EQ .

thank you to Nigel, Thomas, Bjorn, and Steffan. essentially you're telling me there is no existing standard of control number assignment or of scaling and offset of that control. regarding MIDI 1.0 (which is what goes into MIDI files), i had noticed that there were some "predefined controls"

Re: [music-dsp] automation of parametric EQ .

� -- From: "Phil Burk" Date: Tue, December 22, 2015 1:33 pm -- > One problem is that controller address space for MIDI is a bit limited. > The

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

� i thought i understood Tchebyshev polynomials well. �including their trig definitions (for |x|<1), but if what you're trying to do is generate a sinusoid from polynomials, i don't understand where the "Tchebyshev" (with or without the "T") comes in. is it min/max error (a.k.a. Tchebyshev no

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

r music-related DSP" -- > On 21/01/2016 2:36 PM, robert bristow-johnson wrote: > > i thought i understood Tchebyshev polynomials well. including their > > trig definitions (for |x|<1), but if what

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

Original Message Subject: Re: [music-dsp] Anyone using Chebyshev polynomials to approximate trigonometric functions in FPGA DSP From: "Ethan Duni" Date: Thu, January 21, 2016 2:34 am To: "A discussion list for music-related DSP"

Re: [music-dsp] Generating pink noise in Python

Original Message Subject: Re: [music-dsp] Generating pink noise in Python From: "Sound of L.A. Music and Audio" Date: Thu, January 21, 2016 3:18 pm To: music-dsp@music.columbia.edu ---

Re: [music-dsp] Cheap spectral centroid recipe

well, i remember a paper from long ago from James Beauchamp where he defines spectral centroid as � � SUM{ |c_n| n } / SUM{ |c_n| } � where c_n is the complex Fourier coefficient for the nth harmonic. �if you wanted to base it on energy � � SUM{ |c_n|^2 n } / SUM{ |c_n|^2 } � it will

Re: [music-dsp] Cheap spectral centroid recipe

> > Evan Balster > creator of imitone so Evan, i took a look at your website. �your product looks very cool. �in 2013 i worked on something similar (Zya), but cloud based. � so you clearly have a pitch detector goin' on there. �are you converting vocal pitch into fully fo

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Ethan Duni" Date: Wed, February 17, 2016 11:21 pm To: "A discussion list for music-related DSP" ---

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Ethan Duni" Date: Wed, February 17, 2016 11:21 pm To: "A discussion list for music-related DSP" ---

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Evan Balster" Date: Thu, February 18, 2016 10:42 am To: music-dsp@music.columbia.edu ---

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Evan Balster" Date: Thu, February 18, 2016 1:55 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] Cheap spectral centroid recipe

From: "Ethan Duni" Date: Thu, February 18, 2016 4:48 pm -- > I've noticed > in my (cursory) searches that some people use amplitude spectra and others > use power spectra, but the only thing I've found in the way

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Risto Holopainen" Date: Fri, February 19, 2016 7:45 am To: music-dsp@music.columbia.edu

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

Original Message Subject: Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation From: "Evan Balster" Date: Mon, February 22, 2016 1:02 pm To: k.s.matheus...@notam02.no music-dsp@music.columbia.edu --

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

Original Message Subject: Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation From: "Evan Balster" Date: Mon, February 22, 2016 6:45 pm To: music-dsp@music.columbia.edu -

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Esteban Maestre" Date: Thu, February 25, 2016 4:59 pm To: music-dsp@music.columbia.edu ---

Re: [music-dsp] Cheap spectral centroid recipe

Original Message Subject: Re: [music-dsp] Cheap spectral centroid recipe From: "Ethan Duni" Date: Thu, February 25, 2016 4:16 pm To: "A discussion list for music-related DSP"

Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

so i read Jean Laroche's paper in the previous decade and i forget what the takeaway was from it besides i thought he had a good model for describing the non-TI in the LTI. had to do with calculation of the states in a way that made an equivalent filter but with possibly unstable conditions.

Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

On 3/3/16 7:23 PM, robert bristow-johnson wrote: 6. another form to consider is the Lattice or, if you're doing it in fixed-point, the Normalized Ladder form. these are all second-order so they all have the same transfer function and you can calculate coefficients as a function of Coo

Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

On 3/3/16 7:46 PM, Stefan Sullivan wrote: I looked into this exact issue a little while ago. I found that my filters sounded better/worse depending on the biquad topology. Basically if your gaining your input going into states, then those states are more likely to be very far off from where the

Re: [music-dsp] Bela low-latency audio platform

From: "Andrew McPherson" > >> That looks really nice - just curious, what?s the boot time of Bela? >> >> Thanks, Tom Erbe > > > Hi Tom -- thanks! I just measured the boot time at around 25 seconds from > power on to audio code running, and 35 seconds from power on to IDE loading >

Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

Original Message Subject: Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state? From: "Ethan Fenn" Date: Mon, March 28, 2016 11:43 am To: music-dsp@music.columbia.edu --

Re: [music-dsp] confirm a2ab2276c83b0f9c59752d823250447ab4b666

nt.) � � Original Message Subject: Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state? From: "vadim.zavalishin" Date: Mon, March 28, 2016 2:20 pm To: r...@audioimagination.com music-dsp@m

Re: [music-dsp] confirm a2ab2276c83b0f9c59752d823250447ab4b666

Original Message Subject: Re: [music-dsp] confirm a2ab2276c83b0f9c59752d823250447ab4b666 From: "Phil Burk" Date: Tue, March 29, 2016 6:09 pm To: "A discussion list for music-related DSP"

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

being that this is a discussion group about music, which is a subset of audio. �and being that our hearing is at best 10 or 11 octaves, why do you need 32 octaves? and then how closely, in dB, does your pink noise need to conform to the 1/f power spectrum? �+/- 0.1 dB? �0.01 dB? all this can

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

>> amplitude of the lowest octave. Your signal will be dominated by subsonic >> audio and the audio range will be below the threshold of hearing. >> >> On Mon, Apr 11, 2016 at 10:46 AM, robert bristow-johnson < >> r...@audioimagination.com> wrote: >>

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