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

My first home-studio reverberation constructions, both analog and digital are from the 80s, and I've always been impressed with those wonderful high grade musical products with preparation for the speaker and listener room preparation and those great sound scapes coming from all kinds of

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

"I need some possibly quotable real world opinions and experiences on how long stuff can take to design or develop" Sound like that's interesting. But why? Project management, funding, hobby schedule, historic insight, or .. ? There's rather a different angle if you just want to start up

[music-dsp] FIR blog post & interactive demo

Maybe a side remark, interesting nevertheless: the filtering in digital domain, as compared with the analog good ol' electronics filters isn't the same in any of the important interpretations of sampled signals being put on any regular digital to analog converter, by and large regardless of the

[music-dsp] An example video: C to FPGA programming

Thanks for the correction, I hope you guys weren't too disappointed by the music demo video, indeed the corrected links are the ones I intended to give. Hi Scott! The complicated subject of a Silicon Compiler in the sense of C to a working co-routine in FPGA is what interests me, and I'd not do

[music-dsp] An example video: C to FPGA programming

Hi all Maybe it's not everybody's cup of tea, but I recall some here are (like me) interested in music applications of FPGA based signal processing. I made a video showing a real time "Silicon Compile" and test program run on a Zynq board using Xilinx's Vivado HLS to create an FPGA bit file

Re: [music-dsp] Sound Analysis

About the final accuracy of an FFT applied to note recording sample files, it's like if you have a number of waves in you FFT interval, they seldom will wrap around at sample boundaries, and certainly if you want an accurate match in your FFT, the FFT interval should as closely as possible

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

No matter how you go about this, the the Fast Fourier will in almost every case act as some sort of ensemble measurement over it's length, and maybe do some filtering between consecutive transform steps. Maybe you even continuously average in the frequency domain, using per sample sliding FFT

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

It's a complicated subject when you fill in all the boundary conditions properly, isn't it? Lots of frequency considerations look a bit alike but aren't mathematically equivalent. The human (and animal I suppose) hearing is very sensitive to a lot of these issues in all kinds of convoluted

Re: [music-dsp] Antialiased OSC

Content-wise, you also need to consider what the meaning is of the terminology "anti aliased"; technically, it means you prevent the double interpretation of wave partials, which is usually associated with AD or DA conversion, as in that high frequency components that are put on a AD converter

Re: [music-dsp] Antialiased OSC

Kevin Chi wrote: Hi, Is there such a thing as today's standard for softSynth antialiased oscillators? It's easier to know that for Open Source products, which use a variation of methods as far I checked, and nothing out there I've heard (probably some stuff of most known packages) is

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

pa...@synth.net wrote: I think what we can all agree on is; 1) right tool for the right job 2) right level of knowledge to use the tool Disagree, in many cases designing bullets is science, but there are competitions, competing design criteria, and in fact in cases, appropriate "silver

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

The FPGA could do faster version of Midi with accurate time stamping, which CPUs don't always do easily, and the hard-coded FPGA stuff would never be interrupted by kernel activity, etc. Also, FPGAs can be really quite fast for important computations, I've used a $100 Zynq board to demo

[music-dsp] Creating new sound synthesis equipment

Hi DSPers, I would like to reflect a bit about creating (primarily music) synthesis machines, or possibly software, as sort of talking about a dream that has been of some people since let's say the first (mainly analog!) Moogs in the 60s. What is that idea of creating a nice piece of

Re: [music-dsp] sound transcription knowledge

For artistic purposes, there are a number of professional signal recognition tracks, but they are too complicated, and successful "light organ" applications or Rock Show lighting plans are directly correlated with the artistic intentions in the music production. Technology students with some

Re: [music-dsp] Playing a Square Wave

First, let me agree with the notion that through interference of the supersonic elements, audible artifacts through reverberation can come into existence. The early reverb can easily transfer some energy into the audio band and the (partially non-linear) reverb tail of any non-dead listening

[music-dsp] Playing a Square Wave

There's this preoccupation I have since the advent of going "digital", let's say since I heard music being played on CD in the early 80s. I grew up with access to electronics equipment that would generate "square waves" in some sorts of analogue fashion, including originally "digital" chips,

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

Interesting topic since it became a bit of hype when computer audio processing programs started to include "stretching" as an option. Two main thoughts I haven't heard yet: the real time matter, which in the normal digital domain sampling frequency ranges poses the question of the length of

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

gm wrote: What are good frequencies for band splits? (2-5 bands) For standard mastering applications there are norms for binoral and Equal Loudness Curve related reasons. The well known PC software probably doesn't get there but it may be you want to tune those frequencies based on the

Re: [music-dsp] Reading a buffer at variable speed

I cannot understand from the problem definition if it's a matter of learning to understand the frequency at a certain point of re-sampling in mathematical form, supposedly some solution of the constants in a standard e-power formula like a*e^(b*t+c) with two points of the graph of such transform

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

What was that about? I did chip design elements back in the 90s, what does that have to do with making a FPGA design that at high level verifies as interesting because it even seems to out-compute a pentium? I've done things with both DSPs and FPGAs and it strikes me that the way to go for

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

Glad to see there's good interest in the subject, I'm sure it will become more important with time, because of the traditional computer architectures being relatively ineffective for a lot of stuff FPGAs can do, Now, I've put Vivado high level design (free) on the fastest computer I use, even

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

Hi all Well somehow it's alive and kicking, and for good reason, lots of digital signal processing can be done efficiently on FPGA, and it has at least one advantage, which is that possibly the digital signal path design is more sensible (the structure is fixed, maybe like dedicated micro

Re: [music-dsp] Sampling theory "best" explanation

So when is a "system", or better put, it's mathematical model, a linear system ? In a way the scalar high school definition suffices, when the inputs are added, the result must be the same as when the inputs are processed separately, and a mulitplied input gives a multiplied output. Simple

Re: [music-dsp] Sampling theory "best" explanation

Nigel Redmon wrote: Well, it’s quiet here, why not… Right, a good reiteration never hurts! I quickly read through and find your explanation fine, it's not right to expect everybody to be theoretically sound and solid up to the level of mathematical proof, but I'm a strong proponent of

[music-dsp] Linux audio sync and conversion

Hi all, A little venting today about something I've worked on as a sideline while doing an interesting test with phaser/filter banks and different sample rate multicompression for extracting much better reconstructable audio from A grade music. As most will know, I like, to some extend,

Re: [music-dsp] tracking drum partials

Thomas Rehaag wrote: Dear DSP Experts, can anybody tell me how to track drum partials? Is it even possible? What I'd like to have are the frequency & amplitude envelopes of the partials so that I can rebuild the drum sounds with additive synthesis. Hi In theory, just a little like using

[music-dsp] A 1/3 octave multi filter limiting based noise reduction setup

Hi all, Recently I've gotten some work done on a experimental noise reduction setup using the Free and Open Source Jack/Ladspa tools under Linux, at 192khz and with quality enough results that I think some other people might want to try it out. The idea is that when for instance using

[music-dsp] A practical note on using the Ladspa "plugins"

Hi While designing and testing an automatically created filter/multi-compressor bank of 30 "jack-rack" holders of (Linux) Ladspa DSP elements, I was reminded of something I've noticed before. Some audio effects use a lot of CPU resources when idle, or in other words when no signal input is

[music-dsp] Curves from a step function

a little thought to consider in the context of a lot of digital algorithms for audio: if one takes a step function (doesn't matter how exactly, but say a single, unipolar one for the sake of the simple argumentation), like "0" before sample time == 0, and 1 after, in often occurring situations

Re: [music-dsp] ± 45° Hilbert transformer using pair of IIR APFs

Thinking about it, I recall there was some from of transform used for frequency/time analysis for instance for radar problems (maybe books from before WWII, or more recent frequency/time analyzers) and without checking though it was in popular DSP speak something like the Hilbert transform, but

Re: [music-dsp] Recognizing Frequency Components

Always fun to extrapolate into the blue, huh ? Not so much, while it's interesting to look at FFTs and try to form an opinion on using some (maybe readily available) form of it to analyze a frequency, and maybe even interesting parallel/pipe-lined versions are FOS available, it's not just a

Re: [music-dsp] Recognizing Frequency Components

Some serious articles ! From a dry but hopefully somewhat interesting scientific point of view I'd think you need only 4 samples of the original example (accurate quantization, perfectly equal sample distance, noise and distortion-free sine samples), because you need to estimate amplitude,

[music-dsp] Recognizing Frequency Components

Hello DSP list It's one of those subjects that interest me, even though there are lots of solutions around, some partial, some inaccurate, some elaborate, etc etc. Suppose it's given that we have a properly sampled sound file of 1 (one) sinusoidal component of frequency above 20Hz and at

[music-dsp] Using an actual database for storage of DSP settings

Not everybody here will have this problem, but maybe some will recognize the issue, and maybe even some use Linux sound tools. What I propose to do is take the settings of a certain (set of) streaming signal processing graph (based on jackd) with many parameters (like, thousands or at least

Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

robert bristow-johnson wrote: .. From: "Theo Verelst" <theo...@theover.org> .. > > To me it seems the preoccupation of maximizing the mix output isn't wrong, but the digital > domain problems usually has other handles. The choir example of adding say a thousand

Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

To me it seems the preoccupation of maximizing the mix output isn't wrong, but the digital domain problems usually has other handles. The choir example of adding say a thousand voices and needing 10 more bits to capture the highest amplitude of the combined tones at a point where they all

[music-dsp] A game on equal loudness

Hi all, I was working on a (by now) very complicated signal processing graph for studio purposes, about short and mid signal self convolution, encoding sample reconstruction information in digital music streams, acoustic preparation of various frequency bands, as well as safeguarding from

[music-dsp] How much bandwidth ?

Hi DSPers, searching for interesting computer hardware I ended up at one of the important junctions of DSP machine design: apart from available memory size and MIPS/FLOPS: how much bandwidth is available to communicate with the various memories involved in for instance a virtual synthesizer

Re: [music-dsp] Lakh MIDI Dataset v0.1 released

Colin Raffel wrote: Hi all, I'm pleased to announce the release of the Lakh MIDI Dataset v0.1 I haven't gone through the reports/site, but aren't there copyright issues with collecting MIDIs for professional purposes ? T.V. ___ dupswapdrop:

Re: [music-dsp] Choosing the right DSP, what things to look out for?

How important do you reckon FFT hardware acceleration A good question might be to think of what that acceleration is composed of, and that which you want to use it for. I can easily come up with professional use of certain FFT variations, including straight, medium long, and fully

Re: [music-dsp] Anyone think about using FPGA for Midi/audio sync ?

Hi, B. In all honesty, I've looked at some of the supplied materials in the Vivado_hls software, the free web-version has got a number of examples that can work on my very cheap Parallella board. It's the cheapest Zynq board on the marker, which of course might not be the best way to start a

Re: [music-dsp] Anyone think about using FPGA for Midi/audio sync ?

Thanks all for your non-scarce replies to my question and a lot of practical considerations. I suppose mostly it's always nice to know what people are working on. About the reasons for my inquiry: mostly tho whole of the modern day PC as a Complex Instruction Set Computer has become so

[music-dsp] Anyone think about using FPGA for Midi/audio sync ?

Hi all, This is one of those things I've been toying in thought about lately (again): everything on PCs and probably phones and pads as well that does input/output, be it in the family of proper thread management or actual devices like Midi and Audio interfaces (and certainly over networks as

[music-dsp] Intellectual Property management in popular Digital Signal Processing

Hi all, What might be thought to contain some political dimension, IP management, in fact, the way I see it, is a pretty neutral intellectual subject a lot of audio engineers have occupied themselves with in some point of their lives. In the broad field of Electrical Engineering, the

Re: [music-dsp] BW limited peak computation?

Paul Stoffregen wrote: Does anyone have any suggestions or references for an efficient algorithm to find the peak of a bandwidth limited signal? Hi, I think without getting lost in quadratic algebra or endless searches for a holy grail that doesn't exist that I don't take part in, you've

Re: [music-dsp] up to 11

About this usually dreadful subject I know a few easy and some harder suggestions, that might be useful to some. It's not necessarily hard to get loud in this time of technological advancement. Remember those hand held personal alarm units or more contemporary smoke alerms, cooking timers,

[music-dsp] Creating and maintaining digital signal processing graphs

Hi all. Just a little thought about something I regularly encounter and that is interesting me. Working with Linux and Jack/Ladspa (usually at 192kHz/32bit) I make moderately complicated let's call them "patch graphs" of software acting on streaming audio data connected together with a flow

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

HI, Talking about "perfect noise", you may want to consider these theoretics: - what do you do near the Niquist frequency ? Or more practical: noise that gets near the NF will probably cause strange effects in practical DACs and when the digital signal is to be interpreted as "perfectly

[music-dsp] A DAC improvement experiment I did

Hi all, There's been a bit of discussion here at times about Digital To Analog Conversion that inevitably follows a lot of DSP experiments, and the lack of complete accuracy that might make music sound less good than intended. Depending on the POV and varying opinions of course, but it's my

Re: [music-dsp] Delays: sampling rate modulation vs. buffer size modulation

Matthias Puech wrote: Hello, ... I am trying to understand what is the observable relationship between these two scenarios when modulated ... The main thing to remember when thinking of the maybe easier to get example of a tape delay is that the the difference in tape speed at the moment

Re: [music-dsp] Is Beamforming really required in my Speakerphone setup?

Hi, I suppose the question implies the concern for the necessity of the beamforming, or maybe alternatives. It isn't too difficult to do a little estimation about what is going to happen with your setup when signals come from various sides and angles, assuming a sample delay corresponds to

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

Paul Stoffregen wrote: Does anyone have any suggestions or publications or references to best practices for what to do with the state variables of a biquad filter when changing the coefficients? ... I am not directly familiar with the programming of the particular biquad filter variation,

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

Hi I can't offer you a solution to the programming assignment, and in fact there are 2 reasons it might also not become a great general purpose solution. I do however have some affiliation with the interpolation function field and the particular types mentioned in this thread. With interest

Re: [music-dsp] Cheap spectral centroid recipe

Evan Balster wrote: ... To that end: A handy, cheap algorithm for approximating the power-weighted spectral centroid -- a signal's "mean frequency" -- which is a good heuristic for perceived sound brightness . In spite of its

Re: [music-dsp] MIDI Synth Design Advice Please

Ross Bencina wrote: .. http://epickrram.blogspot.co.uk/2015/09/reducing-system-jitter.html ... Interesting read, centered around java and the idea of calling virtual multiprocessing overhead jitter. Probably some form of thread control is on for attention to optimize complex computations,

Re: [music-dsp] MIDI Synth Design Advice Please

Scott Gravenhorst wrote: to: Theo I saw your suggestion regarding JACK and I'm wondering "why?". ... ... Using the example program pcm.c (modified to use threads), I've been able to run the direct write loop on an isolated core and got 45 "voices" of sine production with a period size

Re: [music-dsp] MIDI Synth Design Advice Please

Paul Stoffregen wrote: ... First is the one I've been working on... the Teensy Audio Library. The main advantage of this way is you can integrate with almost all stuff designed for Arduino, such as the MIDI library and USB MIDI, and of course lots of hardware.

Re: [music-dsp] MIDI Synth Design Advice Please

About caching: modern Linuxes, apart from having access to kernel sources and making your own changes, IIRC allow for certain multiprocessing parameters to be influenced, such a locking memory ranges to threads/cores/processes. Not overly discipline thoughts about such semi pro project (I

Re: [music-dsp] MIDI Synth Design Advice Please

Scott Gravenhorst wrote: I'm looking for advice regarding the design of a MIDI synthesizer. ... Advice regarding this endeavor would be appreciated. Questions include which of the transfer methods will come closest to my goal of low latency (from time to MIDI message receipt to .. Hi

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

Hi all, Maybe a bit forward, but hey, there are PhDs here, too, so here it goes: I've played a little with the latest Vivado HLx design tools fro Xilinx FPGAs and the cheap Zynq implementation I use (a Parallella board), and I was looking for interesting examples to put in C-to_chip compiler

[music-dsp] Memory Bandwidth and DSP

Hi list Thinking a bit about my Zynq board having now a working Linux supported AXI connection with it's FPGA that can be used (thus far simple) examples with Vivado 1015.3, I would like to plan using the right niche or corner or "edge" when making some examples that use FPGA acceleration for

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

I don't recall the various chapters of Information Theory and practical digtial electronics or something about these kinds of power estimates, so I'm not putting this thread in some direction, except two small considerations. When the power of a signal is concerned, which in the continuous

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

Everywhere in the exact sciences there's the dualism between statistical analysis and deterministic engineering tools, since the major break through in quantum physics at the beginning of the 20th century. Whether that's some sort of diabolical duality or, as it actually is at the higher levels

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

In the course of these discussions, let's not forget the difference between a convolution with 1/(Pi*t) (a Hilbert transform kernel) and the inversion of the transfer function of a linear system. T. ___ dupswapdrop: music-dsp mailing list

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

Ross Bencina wrote: Hi Everyone, Suppose that I generate a time series x[n] as follows: >>> P is a constant value between 0 and 1 At each time step n (n is an integer): r[n] = uniform_random(0, 1) x[n] = (r[n] <= P) ? uniform_random(-1, 1) : x[n-1] Where "(a) ? b : c" is the C ternary

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

Think of the Fast Fourier Transform as computing the inner product of a piece of signal (the length of the transform) with all kinds of basis functions: the various frequencies that can "fit" in the interval. Without going into engineering basics, you can take a sine and a cosine as a basis

Re: [music-dsp] What would you do with a fl.pt. 1024 FFT transform per 10 microseconds ?

Matthias Meyer wrote: Hey Theo, it is nice to see interest in FPGAs here. ... ..., or the calculation of the power spectrum can be done on the FPGA with only little impact on latency. Best regards, Matthias Maybe this board can be interesting for people wanting to get going a bit, I didn't

Re: [music-dsp] What would you do with a fl.pt. 1024 FFT transform per 10 microseconds ?

Allen Downey wrote: On a related note, how about laying out 2 1024-sample FFTs so you can alternate overlapping windows on a live stream? Hi Allen, I'm not feeling entirely sure about what your suggestion means and why, but I presume you want to take another go at the averaging of bins

Re: [music-dsp] What would you do with a fl.pt. 1024 FFT transform per 10 microseconds ?

Matthias Meyer wrote: Hey Theo, it is nice to see interest in FPGAs here. I have worked with FPGAs for some years now and always wanted to use it for audio processing. Because the FFT is indeed really fast I thought about using it for real-time pitch detection. The Zynq platform is ideal

[music-dsp] What would you do with a fl.pt. 1024 FFT transform per 10 microseconds ?

I've tried to compose this mail from scratch, so there should be no strange (in my mail program) invisible tags left. Anyhow, I was playing around a bit with an FPGA programmer tool called Vivado (from Xilinx), for use with the cheap ($99 and up) Zynq based Parallella board, and looked at he

[music-dsp] DSP connections with basic acoustics

Hi all For those inclined to have an interest for some of the signal processing I have involved myself with (the averaging ideas), and for general interest, I've worked a bit with the (demo version) Comsol 5.1 physics simulation software and in particular one example: the Standing Wave

[music-dsp] How do you, or would you want to, schedule audio data flow and processing in multiple audio stream software

I raise this question out of some form of curiosity about how people deal with the idea of putting their and other people's "plugins" or processing blocks or software pieces together to form a whole. Say you wrote a filter, and you want to add a source and a sink. Once you've chosen your

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

A short note on the Linux sound APIs (or that they should be called) like pulseaudio/alsa/jack. The difference as far as I know is that pusleaudio tried to be a general interface that would work for all applications, intended for general use. Alsa is a relatively low level interface that

Re: [music-dsp] Mails with images?

Good idea to have images and typeset. Bandwidth can soon become a hurdle, and maybe there are still people with older browsers and mail clients. I could see the test image in the first mail of this thread, but I don't mind the method I used myself to just put a link in mail to a stable server

Re: [music-dsp] 20k

Scott Gravenhorst wrote: music-dsp@music.columbia.edu 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, -- ScottG

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

Interesting story about the interpolation noise from very high oversampled signal approximations. I tend to think ïf it doesn't concern an actual sinc function of significant width and accuracy then the up-sampling is wrong unless the signal is prepared for it. I can imagine in sample processing

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

Hi, A suggestion for those working on practical implementations, and lighten up the tone of the discussion with some people I know from worked on all kinds of (semi-) pro implementations when I wasn't even into more than basic DSP yet. The tradeoffs about engineering and implementing on a

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

Thanks to all the participants in this thread, I hope it was at least a little educational, except maybe for some that seem to take everything as a test to their imaginations of themselves being little computers, and not human being with normal associations and lasting affections for serious

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

SOmetimes I feel the personal integrity about these undergrad level scientific quests is nowhere to be found with some people, and that's a shame. Working on a decent subject like these mathematical approximations in the digital signal processing should be accompanied with at least some

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

For people including scientific oriented it always surprises me how little actual science is involved in this talk about tradeoffs. First, what it is you want to achieve by preserving high frequencies (which of course I'm all for)? Second, is it really only at the level of first order

Re: [music-dsp] about entropy encoding

Just to make clear where I stand on this: I've remarked without wasting seas and seas of bandwidth, that it isn't true that a general estimate of the entropy in a bunch of bits on a row can be just defined or given. Some people will like to pretend some sort of half-academic discourse about

Re: [music-dsp] about entropy encoding

Peter S wrote: ... Shannon defines entropy as the minimum average length of a message in bits when you want to transmit messages of a probabilistic ensemble. ... From wikipedia (should be readable for non-academics, non-engineers, etc., too) In information theory, entropy (more specifically,

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

robert bristow-johnson wrote: On 8/9/15 6:23 PM, Sampo Syreeni wrote: 1) a dithered sigma-delta converter is typically better quality than one without dithering Correct. there is and always had been **some** discussion and controversy about that every time i seen it discussed at an AES

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

Again, I don't respond to certain derailers I at some point don't miss content-wise when not reading. I started this thread, like quite a while ago I coined the basics of sampling on this list, because as both practically and theoretically inclined (and talented and educated) I felt a lot of

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

vadim.zavalishin wrote: Hi all! ... http://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_1.1.0.pdf ... Feedback appreciated. I've (pretty quickly) gone over all sections, and I must say I can't find back a lot of important basics that I would have to require to

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

robert bristow-johnson wrote: Peter... ... it's you that are applying concepts of the old or conventional converters here. .. I'm not answering much to treacherous psychopaths (from the use of words and the content of the communication here, anyway) because that doesn't contribute much, and

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

Uli Brueggemann wrote: ... My simple assumption was: if the DAC is a 24 bit DAC it should be possible to get down to e.g. -90 dB with the sum of the signals. But it seems to be quite challenging. It may be wideband (i.e. including sub-sonic and super-sonic) noise that keeps the sum to about

[music-dsp] Non-linearity or filtering

Hi DSPers, For a long time it has bothered me that there's a bit hidden way to deal with all kinds of signal distortion, both in the analog domain as in the digital domain, that isn't necessarily clear. I think this is a real subject, and I'd prefer to have a good angle on it to essentially

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

robert bristow-johnson wrote: ... just for the record, none of them content words were written by me. ... And it's back, in the Prophet-6. I build one of those dual BBD effects with filters and electronics, with interesting options, with a sine wave LFO modulated clock to act as a Leslie

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

Hi all, No theoretical dumbfounding or deep searching incantations from me this Monday, but just something I've through about and that somehow has since long been a part of music and analog and digital productions. I recall when I was doing some computer audio experiments say in the early

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

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 And it's back, in the Prophet-6. I build one of those dual BBD effects with filters and electronics, with interesting

Re: [music-dsp] about entropy encoding

Nonono, you don't get it, but I suppose only academics should try to do a proper universal theory application attempt, I won't respond to this anymore. I do suggest that if you'd take your own impulses and encode them with you own algorithms you would find less interesting and far less poetic

Re: [music-dsp] about entropy encoding

Look, I don't know how far you got in the academic formal training of statistics, and in this case the basics of information theory, but picture it this way: the information game being thought about in the early EE of communication channels and signal processing and analysis had to do with

Re: [music-dsp] Sampling theorem extension

Vadim Zavalishin wrote: ... How about the equation u''=-w*u+g where v is sinc and w is above the sampling frequency? Aw man You're now going to argue your every day signals are the exact outcome of a differential equation, and ON TOP OF THAT are bandwidth limited ? -- dupswapdrop --

Re: [music-dsp] Sampling theorem extension

Charles Z Henry wrote: ... y=conv(u, f_s*sinc(f_s*t) ) Think about it that that is a shifting integral with an sin(x)/x in it, for which there isn't even an easy solution if f_s is really simple. T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source

Re: [music-dsp] Sampling theorem extension

So we're back where I started to make comments on a while ago. Hmm, I knew that. Let's go over the problem shortly again, and let me give one pointer for you guys (and gals ?) who feel lost about the perfection many of us probably would like. It isn't that we cannot create frequency limited

Re: [music-dsp] Sampling theorem extension

HI While it's cute you all followed my lead to think about simple continuous signals that are bandwidth limited, such that they can be used as proper examples for a digitization/synthesis/reconstruction discipline, I don't recommend any of the guys I've read from here to presume they'll make

Re: [music-dsp] Sampling theorem extension

robert bristow-johnson wrote: 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. I agree that there was the possibility of an unstable offense resolution, but I wasn't aware people were being afraid of that

[music-dsp] Did anybody here think about signal integrity

Hi, Playing with analog and digital processing, I came to the conclusion I'd like to contemplate about certain digital signal processing considerations, I'm sure have been in the minds of pioneering people quite a while ago, concerning let's say how accurate theoretically and practically all

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

A comb filter has a certain purpose. Do you know it? Not everybody is good at electronics (where comb filters have a clearer meaning) or physics, but it does pay to take a little time to at least try to understand the difference between free wheel thinking about some glossy subject (DSP)

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