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

[rolfsassinger]

Hi Paula,

 

VST's GUIs are diffucult to handle and for me often frustrating to control with mice.

I think, VSTs are much liked, because they are free or easy to steal ( or used to be when they came up in the mid of the 1990tees.)

 

Another question: Were did you hear about MIDI V2?

 

Rolf

 

 

Gesendet: Donnerstag, 26. Juli 2018 um 12:30 Uhr
Von: pa...@synth.net
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] Creating new sound synthesis equipment



 range really come into their own.

 The main advantage over softsynths (like VSTs, etc) is that musicians prefer a "tactile" surface rather than a keyboard/mouse when "playing". Though I know a lot of composers (including film composers) who prefer scoring using VSTs.

 I also agree that MIDI is now at a stage where it's not adequate to meet the demands of modern synths (VST, DSP, FPGA, or otherwise). Yes you can use NRPNs and Yes OSC exists, but noether of these are widely used. There are rumours about a MIDI V2, though I suspect that's a long way away from being ratified and set in stone.





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

[rolfsassinger]

 

 



Am 26.07.2018 um 23:35 schrieb Eric Brombaugh:
> FPGA implementations of DSP algorithms don't have to be significantly
> more troublesome than a CPU/DSP implementation

Right, but where is the benefit of a DSP algorithms on the FPGA?
Having a look at a IIR with some MULs and ADDs,
a CPU needs e.g. 10 steps @ 4GHz to perform in 64Bit
a FPGA operates with 200MHz and often needs more steps,
not to talk about DIV operations.

 

Parallel processing might solve the speed issue, but FPGA silicon is much more expensive

 

Rolf

 

 

 



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Re: [music-dsp] resampling

[rolfsassinger]

Ok, I had a closer look, but will have to get deeper I think, since i did not fully got this.
According to my simulations, using copied samples does sometimes better.
Maybe a question of intelligent filter design.

Regarding Tom's remark:  Using the copied samples also requires no additional multiplcation since the value is already stored and in use (?)

Anyway thanks.

 

Rolf

 

 

Gesendet: Dienstag, 24. Juli 2018 um 18:36 Uhr
Von: "Nigel Redmon" 
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] resampling


(Not sure why I didn’t receive Rolf’s email directly…)
 

Hi Rolf,
 

First, I should have pointed to this newer series of articles (of mine), not old ones, earlier in this thread. You’ll get a detailed explanation of why zeroes (and as Alex points out, the zeros can be handled efficiently so it’s a good thing anyway):

 

http://www.earlevel.com/main/tag/sampling-theory-series/?order=ASC

 

Nigel

 
 

On Jul 24, 2018, at 7:53 AM, Tom O'Hara  wrote:
 



Adding zeros is an advantage as then you don't need to calculate their multiplication, as 0 x coefficient = 0

The filter order will be the same with zeros or repeated samples.

Tom
 

On 7/24/2018 4:37 PM, rolfsassin...@web.de wrote:




Hello Nigel

 

could you please say a word more to what you mean by "2x", "3x"?

Also I am again not sure why in this case, adding zeros is an advantage. I had expected to just copy the samples to have less work to do in filtering. I tested such things in MATLAB and found that feeding zeros needs more filter TAPs to come to the same result.

 

Rolf

 

 

Gesendet: Montag, 23. Juli 2018 um 18:25 Uhr
Von: "Nigel Redmon" 
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] resampling



Some articles on my website: http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/, especially the 2010 articles, but the Amp Sim article might be a helpful overview.

 

48k -> 8k: Filter with a lowpass with cutoff below 4k; keep 1 sample, throw away 5, repeat.

 

8k -> 48k: Use 1 sample, follow it with 5 new samples of 0 value, repeat; filter with a lowpass filter with cutoff below 4k.

 







 

	
		
			
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Re: [music-dsp] WSOLA on Real Time

[rolfsassinger]

Not everbody reads mails every day so it might be an idea to wait for replies before pushing.

 

I am not an expert on this subject but my company is so I would put it this way:

 

1) To my mind, there is no simpler explanation than given in the Wiki. At least this is an entrypoint to start reading. More comprehensive material can be found in several DSP literature but they might not be easy since the subject is not easy.

 

2) There is not only on "WSOLA" implementation. It is all about details and requirements. What in detail do you want to do?

 

3) Even, if people have code ready fpr your app, they cannot give it way that easily because it might be covered by their company. Me i am not allow to post code at all.

 

I suggest you search the github projects for code coming close to your needs?

 

Rolf

 

Gesendet: Mittwoch, 25. Juli 2018 um 20:41 Uhr
Von: "Alex Dashevski" 
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] WSOLA on Real Time



Hi,

Could you help ?



 
2018-07-24 9:18 GMT+03:00 Alex Dashevski :



Hi.

 

I'm looking for a simple explanation and implementation of WSOLA. I tried to google it but I can't understand all meaning of inputs and outputs that WSOLA has.

My goal is to implement it in Real time on Android. Possibly, to use pitch shifting in Real Time.

 

Thanks,

Alex





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

[rolfsassinger]

Hello Theo

 

the word "hip" regarding FPGA seems to be a good hint. In several music groups the new music machines are discussed heavily. In terms analog modelling and recreating these formerly analog machines we know the digital way. At the first sight FPGAs are consequent descision what me and others doing such designs in professional business have a clear view on design speed, cost, amount of work and in many cases FPGAs are not acceptable and totally fail in comparison to DSPs. Yes, FPGAs have become cheaper and more powerfull in the recent decade and so did DSPs and CPUs too. If you look and todays options with multi core CPUs and GPUs, VSTs could take advantage off, i hardly see cases where FPGAs can do well.

I tried FPGA sound synthesis myself and also completed some designs, but found that MIDI treatment is better hosted in the softcore part or in a hard core like in Cyclone V's ARM architecture. A 600MHz ARM design does all required for MIDI rapidly and totally sufficient. The same is with USB. Writing an USB core in FPGA is no fun I can tell you and is also better donw in a CPU / MCU architecture. Things like changes, new requirements, testing and simulation is much easier and can be done in the CPU / PC domain. We have sandboxes, testboxes, trigger cases all available in Python and CC+Libs ready for usage. And they can be accesses by any person for free. FPGA high tech development and simulation requires a profesional license when you want to do it effectively.

 

What should be discussed regarding MIDI and accurate timing is things like channel handling and controllers. Todays synthesis units and VSTs have tons of parameters and MIDI does not really support this. It is already a hazzle to join 2 or more controllers to have 16 channels to control a DAW and add a third one to run the tunes. Synchronisation is an issue too.

 

Rolf

 

 


Gesendet: Sonntag, 22. Juli 2018 um 22:21 Uhr
Von: "Theo Verelst" 
An: "A discussion list for music-related DSP" 
Betreff: [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 electronic equipment to create blips and mieaauuws,
thundering basses, imitation instruments, and as recently has "relived" all kinds
of more or less exciting new sounds that maybe have never been used in music before.
As for some it's like a designer's dream to create exactly the sound they have in
mind for a piece of musique concrète, for others it's maybe to compensate for their
lack of compositional skills or musical instrument training, so that somehow through
the use of one of those cool synthetic sounds they may express something which
otherwise would be doomed to stay hidden, and unknown.

Digital computer programs for sound synthesis in some sense are thought to take
over from the analog devices and the digital sound synthesis machines like "ROMplers"
and analog synthesizer simulations. It's not true this has become the decisive reality
thus far: there's quite a renewed interest in those wonderful analog synthesis sounds,
various manufacturers recreate old ones, and some advanced ones make new ones, too.
Even though it is realistic that most folks at home will undoubtedly listen most of the
time to digital music sources, at the same time there's a lot of effort still in the
analog domain, and obviously a lot of attempts at processing digital sound in order
to achieve a certain target quality or coolness of sound or something else ?

Recently there's been a number of interesting combinations of analog and
digital processing as well as specific digital simulation machines (of analogue
type of sound synthesis) like the Prophets (DSI), The Valkyrie (Waldorf "Kyrie" IIRC)
based on FPGA high sampling frequency digital waveform synthesis and some others.

Myself I've done a Open Source hard- AND software digital synthesizer design based on
a DSP ( http://www.theover.org/Synth ) over a decade ago, before this all was considered
the hip, and I have to say there's still good reason for hardware over software synthesis,
while I of course can understand it is likely computers will get better and
better at producing quality synthesis software. At the time I made my design, I liked to
try out the limits I liked as a musician, such as extremely low, and very stable latency
(one audio sample, with accurate timed Midi message reading in programmable logic)
straight signal path (no "Xruns" ever, no missed samples or re-sampling ever, no multi
processing quirks, etc). My experience is that a lot of people just want to mess around
with audio synthesizers in a box! They like sounds and turning some knobs, and if a
special chip gives better sound, for instance because of higher processing potential
than a standard 

Re: [music-dsp] resampling

[rolfsassinger]

Hello Nigel

 

could you please say a word more to what you mean by "2x", "3x"?

Also I am again not sure why in this case, adding zeros is an advantage. I had expected to just copy the samples to have less work to do in filtering. I tested such things in MATLAB and found that feeding zeros needs more filter TAPs to come to the same result.

 

Rolf

 

 

Gesendet: Montag, 23. Juli 2018 um 18:25 Uhr
Von: "Nigel Redmon" 
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] resampling



Some articles on my website: http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/, especially the 2010 articles, but the Amp Sim article might be a helpful overview.

 

48k -> 8k: Filter with a lowpass with cutoff below 4k; keep 1 sample, throw away 5, repeat.

 

8k -> 48k: Use 1 sample, follow it with 5 new samples of 0 value, repeat; filter with a lowpass filter with cutoff below 4k.

 





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Re: [music-dsp] What is resonance?

[rolfsassinger]

Hello all

 

Is "feedback with delay" really resonance? I recognize many people describe the effects of "room resonanes this way", but to my understanding these are no resonances in the basic meaning but reflections. A resonance is a self standing oscillating system like a guitar string or an air mass in a Helmholtz resonator.

 

 Rolf

 

Gesendet: Samstag, 21. Juli 2018 um 04:33 Uhr
Von: "Andrew Simper" 
An: "A discussion list for music-related DSP" 
Cc: audit...@lists.mcgill.ca, local-us...@ccrma.stanford.edu, surso...@music.vt.edu
Betreff: Re: [music-dsp] What is resonance?





Resonance is just delay with feedback. Resonance occurs when you delay a signal and then feed it back with some gain to the input of the delay "in phase"

 





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Re: [music-dsp] Analysis of an electromagnetic attack

[rolfsassinger]

 

Struggeling if i should takt this for serious. Tissue warming is a common issue with magnetic resonance, therefore the strength of field is limited.

 

But criminals? Not really eh?

 

In former germany of the 1930tes, instruments had been designed to battle with electromagentic waves. This was all withdrawn since totally inefficient.

50 years later I heared a US president and former actor reinvented the idea to battle in space with laser and electromagnetics. Also dropped AFAIR.

 

Rolf


Gesendet: Sonntag, 08. Juli 2018 um 15:09 Uhr
Von: "Tito Latini" 
An: music-dsp@music.columbia.edu
Betreff: [music-dsp] Analysis of an electromagnetic attack

Hi,

I'm writing a document about my experience here:

https://github.com/titola/neuropa

The following video shows a part of the signal directed to my head:

https://github.com/titola/neuropa/blob/master/media/head_945mhz.mp4

Any time a RF pulse is absorbed by the cerebral tissue, the
temperature rises of 5e-6 °C [1]. The rapid thermal expansion produces
a thermoelastic wave that travels to the inner ear. Therefore the
criminals use that thermo-acoustic demodulator to send vocal messages
to me through ultrasounds.

If you hear a voice in the audio file

https://github.com/titola/neuropa/blob/master/media/cuba_attack_decoded.ogg

also the US embassy in Cuba [2] unconsciously suffered a similar offensive
action. The steps to get that result from the recording [3] are:

- FIR filter to select the content between 6 and 9 KHz.

- Pitch 1 octave up (optional).
example: rubberband -f2 in.wav out.wav

- Slope detector.
pseudo-code: env_follower(diff(input))

- BP filter to select the content between 100Hz and 3.5KHz.

The document contains what I can unequivocally explain.


[1] J.A. Elder and C.K. Chou. Auditory Response to Pulsed Radiofrequency Energy.

[2] https://en.wikipedia.org/wiki/Embassy_attack_accusations_in_Cuba

[3] Josh Lederman and Michael Weissenstein. Dangerous sound? What
Americans heard in Cuba attacks. AP News, October 2017.
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Re: [music-dsp] EQ-building with fine adjustable steepness

[rolfsassinger]

Hello Robert

 

thanks, so this means that it will come out with a cascade anyway. Would'nt it then be generally better to put filters in series or use parallel band width limited filters though?

 

Regards Rolf

 

 

Gesendet: Mittwoch, 27. Juni 2018 um 16:49 Uhr
Von: "robert bristow-johnson" 
An: music-dsp@music.columbia.edu
Betreff: 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 somewhere between 0 and -6 dB per octave. If you start with a pole on the left and follow it shortly with a zero, it will be closer to zero.  If you have more space between the pole and zero frequency, then the slope is higher.

 




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