Re: [music-dsp] TMS320 commercial synth/effects?

s for the info. -- r b-j r...@audioimagination.com "Imagination is more important than knowledge." > On Mon, Jul 20, 2020, 22:18 robert bristow-johnson > wrote: > > > > The H3000 is a legendary piece of gear. I've worked with the two main > > designers o

Re: [music-dsp] TMS320 commercial synth/effects?

The H3000 is a legendary piece of gear. I've worked with the two main designers of it and they both live in the same town in Vermont that I do. I did not get to work on that product line when I joined Eventide in late 1991. >From a simple and effective user-interface POV, it's also quite

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

> On June 24, 2020 4:53 PM Zhiguang Zhang wrote: > > > I don't think there's any issue - I just posted about the TrackSpacer plugin > and the thread started up again. Actually what I've been trying to get across > is that the Gibbs "nastiness' is ever present in both hardware and software

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

is this the same thing we were discussing in March? wasn't that three months ago? what, exactly, is the issue? there *are* some things in common between OLA phase vocoder and OLA fast convolution. in fact, if you're willing to make your fast convolution less fast than optimal, you can use

Re: [music-dsp] the time it takes to design a reverberator and related

> On May 21, 2020 5:04 PM gm wrote: > > > I need some possibly quotable real world opinions and experiences on how > long stuff > can take to design or develop, especially takeing Hofstadter's Law into > account > > For instance reverberators, hard to estimate, and I dont recall all the

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

> On March 20, 2020 4:58 PM Andreas Gustafsson wrote: > > > robert bristow-johnson wrote: > > anyway, while i have done this sliding Hann window before, i haven't > > done it for a sliding DFT. but i would be excited to see a good > > implementation of cons

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

> On March 20, 2020 2:45 PM STEFFAN DIEDRICHSEN wrote: > > > Actually, you can do a window size per bin and an arbitrary spacing of the > frequencies and create a “true” constant Q SDFT. Somehow, it reminds me on > the modal synthesis stuff, which can be used to create weird processing.

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

so the "implicit" sliding rectangular window has just as much mathematical meaning as if it were explicit. as Steffan points out, this implicit sliding rectangular window used in the sliding DFT is essentially the same implicit sliding rectangular window used in the efficient method of

Re: [music-dsp] FIR blog post & interactive demo

> On March 12, 2020 5:35 PM Ethan Duni wrote: > > > Hi Robert > > > On Wed, Mar 11, 2020 at 4:19 PM robert bristow-johnson > wrote: > > > > i don't think it's too generic for "STFT processing". step #4 is pretty > > generic. &g

Re: [music-dsp] FIR blog post & interactive demo

> On March 11, 2020 6:53 PM Ethan Duni wrote: > > > On Tue, Mar 10, 2020 at 8:36 AM Spencer Russell wrote: > > > > The point I'm making here is that overlap-add fast FIR is a special case > > of STFT-domain multiplication and resynthesis. I'm defining the standard > > STFT pipeline here

Re: [music-dsp] FIR blog post & interactive demo

> On March 10, 2020 11:34 AM Spencer Russell wrote: > > > Thanks for your expanded notes, RBJ. I haven't found anything that I disagree > with or that contradicts what I was saying earlier - I'm not sure if they > were intended as expanded context or if there was something you were >

Re: [music-dsp] FIR blog post & interactive demo

> On March 8, 2020 7:55 PM Ethan Duni wrote: > > Fast FIR is a different thing than an FFT filter bank. > > You can combine the two approaches but I don’t think that’s what is being > done here? > On March 9, 2020 10:15 AM Spencer Russell wrote: > > > I think we're mostly on the same

Re: [music-dsp] FIR blog post & interactive demo

agree on the terminology. -- robert r b-j r...@audioimagination.com "Imagination is more important than knowledge." > > On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson > > wrote: > > > > > > > > > > On March 8, 2020 1

Re: [music-dsp] FIR blog post & interactive demo

e impulse response > > but what you might get are ripple artifacts from the FFT windowing > > function. Otherwise the algorithm is inherently zero-phase. > > > > > > On Sat, Mar 7, 2020, 7:11 PM robert bristow-johnson > > wrote: > > > > > >

Re: [music-dsp] FIR blog post & interactive demo

onse but > what you might get are ripple artifacts from the FFT windowing function. > Otherwise the algorithm is inherently zero-phase. > > > On Sat, Mar 7, 2020, 7:11 PM robert bristow-johnson > wrote: > > > > > > > On March 7, 2020 6:43 PM zhiguang zhang

Re: [music-dsp] FIR blog post & interactive demo

> On March 7, 2020 6:43 PM zhiguang zhang wrote: > > > Yes, essentially you do have the inherent delay involving a window of samples > in addition to the compute time. yes. but the compute time is really something to consider as a binary decision of whether or not the process can be real

Re: [music-dsp] FIR blog post & interactive demo

Like a lotta things, sometimes people use the same term to mean something different.   A "phase vocoder" (an STFT thing a la Portnoff) is not the same as a "channel vocoder" (which is a filter bank thing).--r b-j                      r...@audioimagination.com"Imagination is more important

Re: [music-dsp] high & low pass correlated dither noise question

  Original Message Subject: [music-dsp] high & low pass correlated dither noise question From: "Alan Wolfe" Date: Thu, June 27, 2019 7:42 am To: "A discussion list for music-related DSP"

Re: [music-dsp] Audio Plugin Listening Test

� listen, i am an old fart.� a decade ago i discovered that i lost about 30 dB around 4 kHz.� but i have tried to adapt and for the most part enjoy full bandwidth music. in **none** of the 4 snippets could i hear any real difference between the 5 files presented in the snippet. sorry, dunno

[music-dsp] Google's app is wonderful today!

in honor of JS Batch. algorithmic composer.  a little like band in a box. -- r b-j                         r...@audioimagination.com "Imagination is more important than knowledge."         ___ dupswapdrop: music-dsp mailing list

Re: [music-dsp] Looking for DSP/Plugin/... developers (3D audio related)

  Bram, if the listing is for work that involves some combination of music (or audio) and DSP, none of us consider it to be abuse to post it here. and some folks might be living around Berlin or Germany or the EU somewhere (and not in the soon-to-be-outside-the-EU UK) and may very well find

Re: [music-dsp] pitch shifting vs sample rate

� Original Message Subject: [music-dsp] pitch shifting vs sample rate From: "Alex Dashevski" Date: Thu, March 14, 2019 2:55 pm To: music-dsp@music.columbia.edu

[music-dsp] Who uses YIN or pYIN for pitch detection?

  Hay, any peeps around here that use YIN?  or pYIN? Some of you who hang around the DSP Stack Exchange might know that I am unimpressed with YIN, namely that I don't think there is anything novel about it (w.r.t. Average Squared Difference Function, ASDF) other than this so-called

Re: [music-dsp] SOLA, PSOLA and WSOLA.

� i often ask the same question and had thought at one time i knew the answer.� and then another paper comes out and muddies the water. i have most often thought that PSOLA means what i like to call "Lent's Algorithm" (that can also be credited to Hamon).� it's a pitch shifter that does not

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

Fri, Feb 22, 2019 at 9:08 AM robert bristow-johnson < > r...@audioimagination.com> wrote: > >> i just got in touch with Olli, and this "triangle wave to sine wave" >> shaper polynomial is discussed at this Stack Exchange: >> >> >> >> h

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

al Message Subject: Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator From: "robert bristow-johnson" Date: Thu, February 21, 2019 1:33 pm To: "A discussion l

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

Date: Thu, February 21, 2019 9:25 am To: "robert bristow-johnson" "A discussion list for music-related DSP" -- > Another approach is to use a Taylor Expansion. It's pretty accurate in the >

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

Original Message Subject: [music-dsp] Time-variant 2nd-order sinusoidal resonator From: "Martin Vicanek" Date: Thu, February 21, 2019 10:33 am To: music-dsp@music.columbia.edu

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

Original Message Subject: Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator From: "Andrew Simper" Date: Wed, February 20, 2019 9:20 pm To: "Robert Bristow-Johnson" "A discussion l

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

i did that wrong.  i meant to say:      x[n] = a[n] + j*b[n] = g[n-1]*exp(j*w[n]) * x[n-1]   this is the same as     a[n] = g[n-1]*cos(w[n])*a[n-1] - g[n-1]*sin(w[n])*b[n-1]  

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

On Wed, February 20, 2019 9:10 pm, "Ethan Fenn" wrote: > > A very simple oscillator recipe is: > > a(t+1) = C*a(t) - S*b(t) > b(t+1) = S*a(t) + C*b(t) > > Where C=cos(w), S=sin(w), w being the angular frequency. a and b are your > two state variables that are updated every sample

Re: [music-dsp] Auto-tune sounds like vocoder

Original Message Subject: Re: [music-dsp] Auto-tune sounds like vocoder From: "Eder Souza" Date: Thu, January 17, 2019 6:46 am To: "A discussion list for music-related DSP"

Re: [music-dsp] Auto-tune sounds like vocoder

� David and Neil, you should look at Keith Lent's 1989 article: https://www.jstor.org/stable/3679554 and my 1995 article: https://secure.aes.org/forum/pubs/journal/?elib=7947 it's really crude, but with a really good pitch detection alg (in this pitch shifter, octave errors in the pitch

Re: [music-dsp] Auto-tune sounds like vocoder

Do you mean that the use of autotune to quantization pitch to hard, non-vibrato notes? --r b-j                     r...@audioimagination.com "Imagination is more important than knowledge." Original message From: David Reaves Date: 1/15/2019 11:05 AM (GMT-08:00)

Re: [music-dsp] Sound Analysis

> Thank you Nigel, RB-J, Steffan, and Neil. > yer welcome from me.� armchair quarterbacking is pretty easy. > > > i suspect that those tone wheel waveforms are close to sinusoidal. >> > > Early models were. Starting I think around '53 with the B-3, C-3 and A1xx > series (A100

Re: [music-dsp] Sound Analysis

> I have a file which contains a second's worth of sound of each of the 91 > tonewheels of a Hammond B-3 organ in order. (Hammonds have spinning disks > whose edge is fluted in a shape of a desired output sound wave. This spins > in front of a mechanical pickup, which converts that

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

� i don't wanna lead you astray.� i would recommend staying with the phase vocoder as a framework for doing time-frequency manipulation.� it **can** be used real-time for pitch shift, but when i have used the phase vocoder, it was for time-scaling and then we would simply resample the

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

what you're discussing here appears to me to be about perfect reconstruction in the context of Wavelets and Filter Banks. there is a theorem that's pretty easy to prove that if you have complementary high and low filterbanks with a common cutoff at 1/2 Nyquist, you can downsample both

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

different 8 point FFTs that they illustrate.--r b-j                      r...@audioimagination.com"Imagination is more important than knowledge." Original message From: Ethan Fenn Date: 11/5/2018 11:34 AM (GMT-08:00) To: robert bristow-johnson , music-dsp@music.co

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

� Ethan, that's just the difference between Decimation-in-Frequency FFT and Decimation-in-Time FFT. i guess i am not entirely certainly of the history, but i credited both the DIT and DIF FFT to Cooley and Tukey.� that might be an incorrect historical impression.

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

and the other thing you're describing is what they usually call "sinusoidal modeling.".--r b-j                     r...@audioimagination.com"Imagination is more important than knowledge." Original message From: gm Date: 11/4/2018 4:14 PM (GMT-08:00) To:

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

mr. g,I think what you're describing is the Cooley-Tukey Radix-2 FFT algorithm.--r b-j                     r...@audioimagination.com"Imagination is more important than knowledge." Original message From: gm Date: 11/4/2018 4:14 PM (GMT-08:00) To:

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

personally, i believe this issue is resolved with the choice of the window size (or "frame length", not to be conflated with "frame hop") that goes into the FFT. for a sufficiently large window, the two sinusoids will appear as two separate peaks in the FFT result. for a sufficiently

[music-dsp] Roland AX-Synth

� do you folks know if the Roland AX-Synth has a real-time vocal pitch shift function like the Digitech Vocalist or the TC-Helicon VoiceTone?� i know that Korg and Roland and Novation had/have other synths with a vocoder of some sort in it.� i know that, back in the olden daze, Roland had a

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Sampo Syreeni" Date: Wed, October 31, 2018 9:35 pm To: philb...@mobileer.com "A discussion list for music-related DSP" Cc: &qu

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

Original Message Subject: [music-dsp] two fundamental questions Re: FFT for realtime synthesis? From: "gm" Date: Tue, October 30, 2018 8:17 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Original Message Subject: Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis? From: "gm" Date: Mon, October 29, 2018 7:57 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

my comments below are about time-scaling without pitch shifting.� a time-scaler coupled with resampling (like sample-rate conversion) will make you a pitch shifter. Original Message Subject: Re: [music-dsp] pitch shifting in

Re: [music-dsp] FFT for realtime synthesis?

Original Message Subject: [music-dsp] FFT for realtime synthesis? From: "gm" Date: Tue, October 23, 2018 5:51 pm To: music-dsp@music.columbia.edu -- >

Re: [music-dsp] phase vocoder

i have some old MATLAB code that does the phase vocoder implementing time-scaling with a little extra thing that scales the sweep rate of sinusoids inside the frame.� it is a consequence of this very old paper i did for the IEEE WASPAA (Mohonk) in

Re: [music-dsp] WSOLA on RealTime

Original Message Subject: Re: [music-dsp] WSOLA on RealTime From: "Alex Dashevski" Date: Thu, September 27, 2018 2:15 am To: music-dsp@music.columbia.edu Cc: "robert

Re: [music-dsp] WSOLA on RealTime

Original Message Subject: Re: [music-dsp] WSOLA on RealTime From: "Jacob Penn" Date: Wed, September 26, 2018 5:00 pm To: r...@audioimagination.com music-dsp@music.columbia.edu

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Phil Burk" Date: Tue, August 7, 2018 12:59 am To: "robert bristow-johnson" "A discussion l

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Scott Gravenhorst" Date: Mon, August 6, 2018 8:06 pm To: music-dsp@music.columbia.edu --

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Ross Bencina" Date: Sat, August 4, 2018 2:12 am To: "A discussion list for music-related DSP"

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Nigel Redmon" Date: Sun, August 5, 2018 1:30 pm To: music-dsp@music.columbia.edu -- >

Re: [music-dsp] Antialiased OSC

lumbia.edu -- > Hi Robert, > > On 5/08/2018 8:17 AM, robert bristow-johnson wrote: >> In a software >> synthetic that runs on a modern computer, the waste of memory does not >> seem to be salient.� 4096 � 4 � 64 = 1 meg.� Thats 64 wavetables for >> some inst

Re: [music-dsp] Antialiased OSC

I am not sure what a "pure sawtooth phasor" is.  Do you mean a "naive sawtooth" a.k.a. a ramp function? The technique that I have suggested is, say, 4096 samples for all active wavetables so that alignment and crossfading are simpler.  In a software synthetic that runs on a modern

Re: [music-dsp] Antialiased OSC

Original Message Subject: Re: [music-dsp] Antialiased OSC From: "Kevin Chi" Date: Sat, August 4, 2018 2:44 am To: music-dsp@music.columbia.edu -- >

Re: [music-dsp] Antialiased OSC

-- > Can you provide the code with something like pastebin/ Dropbox / gdrive? > I'm also very interested in seeing this implementation. > Thanks, > napent > > sob., 4 sie 2018, 00:57 użytkownik robert bristow-johnson < > r...@audio

Re: [music-dsp] Antialiased OSC

� Original Message Subject: [music-dsp] Antialiased OSC From: "Kevin Chi" Date: Fri, August 3, 2018 2:23 pm To: music-dsp@music.columbia.edu -- > > Is there

Re: [music-dsp] WSOLA on Real Time

t;)? r b-j � Original Message Subject: Re: [music-dsp] WSOLA on Real Time From: "Alex Dashevski" Date: Fri, July 27, 2018 5:02 pm To: "robert bristow-johnson" music-dsp@music.columbia.edu

Re: [music-dsp] WSOLA on Real Time

Alex, may i ask you where you are regarding development on the android?� you're coding in C++ or C (or is it C# or something)? are you at a place where you can make a simple "passthru" app where you can input samples from the ADC and output them to the DAC?� if you are there, can you make a

Re: [music-dsp] Creating new sound synthesis equipment

� dunno how, but a block of some text got moved spuriously. Original Message Subject: Re: [music-dsp] Creating new sound synthesis equipment From: "robert bristow-johnson" Date: Thu, July 26, 2018 4:11 pm To:

Re: [music-dsp] Creating new sound synthesis equipment

Original Message Subject: Re: [music-dsp] Creating new sound synthesis equipment From: "Sound of L.A. Music and Audio" Date: Thu, July 26, 2018 3:16 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] What is resonance?

. Stefan On Sun, Jul 22, 2018, 18:11 robert bristow-johnson wrote: I've been wondering about the connection that resonance and filter orders at least 2.  That's 2 delays (and feedback). But if you're limiting the resonant frequencies to DC and Nyquist, then with a one-sample dela

Re: [music-dsp] What is resonance?

I've been wondering about the connection that resonance and filter orders at least 2.  That's 2 delays (and feedback). But if you're limiting the resonant frequencies to DC and Nyquist, then with a one-sample delay digital filter, you can have something like "resonance".  Even if the

Re: [music-dsp] wavetable filtering

On 7/3/18 7:23 AM, alexandre niger wrote: Thank you for all the help. Gain loss was finally fixed after normalizing. In an other hand, using fft and inverse effectively gave better results than FIR or IIR. With very rich signals, I can still hear an harmonic difference between WTs. I guess I

Re: [music-dsp] wavetable filtering

� you don't really need symmetric to prevent phase cancellations.� you just need to make the phases of each harmonic (and in wavetable synthesis, **each** partial or overtone is harmonic) of the two wavetables that you're crossfading or blending to be aligned. what even symmetry does is line

Re: [music-dsp] EQ-building with fine adjustable steepness

Original Message Subject: Re: [music-dsp] EQ-building with fine adjustable steepness From: rolfsassin...@web.de Date: Fri, June 29, 2018 12:06 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] EQ-building with fine adjustable steepness

So with a one-pole LPF with its corner frequency set very low, you wI'll get a -6 sB slope, which is twice the slope that you desire for pink noise.if you follow that with a zero, the slope will bend back to zero slope. So repeating and alternating poles and zeros, will get you a slope

Re: [music-dsp] Blend two audio

readable. Original Message Subject: Re: [music-dsp] Blend two audio From: "Magnus Jonsson" Date: Wed, June 20, 2018 6:55 pm To: "robert bristow-johnson" music-ds

Re: [music-dsp] Blend two audio

� okay, Benny, i am changing your "a(t)" to "x(t)", because i have been using "a(t)" for the crossfade gain function. now if you want to splice from� x(t) to x(t+T) when T is "estimated", does that mean you can add or subtract a couple of milliseconds to T for the purpose of minimizing the

Re: [music-dsp] Blend two audio

From: "gm" Date: Mon, June 18, 2018 9:10 pm To: music-dsp@music.columbia.edu -- > > > Am 19.06.2018 um 02:52 schrieb robert bristow-johnson: >> �Olli Niemitalo had some ide

Re: [music-dsp] Blend two audio

Original Message Subject: Re: [music-dsp] Blend two audio From: "Nigel Redmon" Date: Mon, June 18, 2018 7:14 pm To: music-dsp@music.columbia.edu -- >

Re: [music-dsp] Playing a Square Wave

Original Message Subject: Re: [music-dsp] Playing a Square Wave From: "Uli Brueggemann" Date: Wed, June 13, 2018 4:57 pm To: "robert bristow-johnson" "A discussion l

Re: [music-dsp] Playing a Square Wave

Original Message Subject: Re: [music-dsp] Playing a Square Wave From: "Neil Goldman" Date: Wed, June 13, 2018 11:16 am To: ra...@raito.com music-dsp@music.columbia.edu

Re: [music-dsp] Antialias question

Original Message Subject: Re: [music-dsp] Antialias question From: "Sound of L.A. Music and Audio" Date: Fri, June 1, 2018 4:48 am To: music-dsp@music.columbia.edu

Re: [music-dsp] Antialias question (Kevin Chi)

Original Message Subject: Re: [music-dsp] Antialias question (Kevin Chi) From: "Kevin Chi" Date: Fri, June 1, 2018 2:50 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] WSOLA

for what it's worth... � -- r b-j� � � � � � � � � � � � �r...@audioimagination.com "Imagination is more important than knowledge." > > > 2018-05-29 12:04 GMT+03:00 robert bristow-johnson > : > >> >> Do you mean as a time-scaler or as a pitch-shifter? >> >

Re: [music-dsp] WSOLA

mean to do a processing with 8Khz(subsample) ? I also want to achieve the high performance and minimum latency. How can I proof to my instructor that correct way to implement is pitch shifting and not WSOLA  on RealTime? Thanks,Alex    2018-05-29 4:19 GMT+03:00 robert brist

Re: [music-dsp] WSOLA

Original Message Subject: Re: [music-dsp] WSOLA From: "Alex Dashevski" Date: Sun, May 27, 2018 2:56 pm To: philb...@mobileer.com music-dsp@music.columbia.edu

Re: [music-dsp] WSOLA

On 5/25/18 2:06 PM, Alex Dashevski wrote: I want to implement WSOLA on Real Time. The pitch is between 5ms and 20ms. do you mean the *period* is between 5 ms and 20 ms? or that the fundamental frequency is between 50 Hz and 200 Hz? this appears to be a bass instrument Frequency samples

Re: [music-dsp] Real-time pitch shifting?

Original Message Subject: Re: [music-dsp] Real-time pitch shifting? From: "Chris Cannam" Date: Mon, May 21, 2018 3:35 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] Real-time pitch shifting?

Original Message Subject: Re: [music-dsp] Real-time pitch shifting? From: "RJ Skerry-Ryan" Date: Sat, May 19, 2018 4:34 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] Real-time pitch shifting?

� the current state of the art for pitch manipulation is, of course, Melodyne. dunno how Peter Neubacker does it. somewhere, i have some old MATLAB code that does a time-scaling via phase-vocoder.� you can combine that with resampling to get a pitch shifter. if it's pitch shifting a monophonic

Re: [music-dsp] Build waveform sample array from array of harmonic strengths?

Original Message Subject: [music-dsp] Build waveform sample array from array of harmonic strengths? From: "Frank Sheeran" Date: Sun, April 15, 2018 2:55 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] bandsplitting strategies (frequencies) ?

Original Message Subject: Re: [music-dsp] bandsplitting strategies (frequencies) ? From: "gm" Date: Tue, March 27, 2018 6:10 am To: music-dsp@music.columbia.edu

Re: [music-dsp] bandsplitting strategies (frequencies) ?

On 3/23/18 12:01 AM, gm wrote: What are good frequencies for band splits? (2-5 bands) What I am doing is divide the range between 100 Hz 5-10 kHz into equal bands on a log scale (log2 or pitch). Are there better strategies? Or better min/max frequencies? How is it usually done?

Re: [music-dsp] Wavetable File Formats?

Original Message Subject: Re: [music-dsp] Wavetable File Formats? From: "gm" Date: Wed, March 14, 2018 6:46 am To: music-dsp@music.columbia.edu

Re: [music-dsp] Wavetable File Formats?

Original Message Subject: Re: [music-dsp] Wavetable File Formats? From: "gm" Date: Wed, March 14, 2018 6:39 am To: music-dsp@music.columbia.edu

Re: [music-dsp] Wavetable File Formats?

Original Message Subject: Re: [music-dsp] Wavetable File Formats? From: "Risto Holopainen" Date: Mon, March 12, 2018 1:19 pm To: music-dsp@music.columbia.edu

Re: [music-dsp] Wavetable File Formats?

Original Message Subject: Re: [music-dsp] Wavetable File Formats? From: "Risto Holopainen" Date: Sat, March 10, 2018 11:58 am To: music-dsp@music.columbia.edu

Re: [music-dsp] granular synth write up / samples / c++

t; I use cubic hermite interpolation to get fractional samples, my grain size >> is 20 milliseconds and the cross fade time is 2 milliseconds. >> >> Would you consider this enough in the family of granular synthesis to call >> it GS for a layman / introduction? >

Re: [music-dsp] granular synth write up / samples / c++

� this is very cool.� i had not read through everything, but i listened to all of the sound examples. so there are two things i want to ask about.� the first is about this "granular" semantic: Thing #1:� so the pitch shifting is apparently *not* "formant-corrected" or

Re: [music-dsp] Clock drift and compensation

Ben, can you confirm that what you want to do is Asynchronous Sample Rate Conversion (ASRC)? this is what Steffan is talking about and what it looked like you were looking for in your first post. If ASRC is what you wanna do, that is a combination of the SRC task (like what is done to a

Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?

through an impedance-matching (or isolation) transformer. � r b-j Original Message Subject: Re: [music-dsp] PCM audio amplitudes represent pressure or displacement? From: "robert bristow-johnson" <r...@audioimagination.co

Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?

things, because they >> can mix a signal together with its integrals and derivatives. It's simple >> to transform that velocity signal into a displacement signal or vice versa, >> but in practice any filtering effects will result in a frequency-dependent >> mixture of the two

Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?

� i know relatively little about transducers.� but these are getting to be pretty fundamental questions. Renato, we don't know exactly how much the digital value will translate to a precise loudspeaker piston displacement because there is an amplifier with a volume control in between the D/A

Re: [music-dsp] What happened to that FPGA for DSP idea

� well, i know of at least one company that uses an FPGA to replace the ASIC they used to use. �they use it to fetch samples, interpolate and loop. �also to do some post-processing DSP to the note. -- r b-j � � � � � � � � �r...@audioimagination.com "Imagination is more important than

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