Hi
Here is something with an early 320:
http://www.96khz.org/oldpages/activediffusion.htm
Became a part of a commercial product later
Has been based on this work:
http://www.96khz.org/oldpages/echocancelling2.htm
Later ported to a music bare bone instrument:
Hi all,
does this also cover automated speech recognition? I am working on a
system wich optimizes patterns for recognition that way that it adapts
to different ways of speeking to distinguish dialects an such.
Jürgen
https://www.xing.com/profile/Juergen_Schuhmacher
Am 16.06.2020 um 19:56
I am following this "C to VHDL" thing now for 15 years. Mentor had a
tool to do that already in 2005.
Up to now, the basic issue never changed:
C/C++ has not the options to define those things which are required to
generate an application specific hardware. But exactly this is the point
about
Hello all
Is there a synchronous synchronisation to the given power frequency or
does it have slippage like an asynch engine?
If long term synchronous, there certainly is jitter.
Am 02.01.2019 um 02:50 schrieb Ben Bradley:> Even with that, you're
> still not accounting for variations caused
Hi all
Am 01.10.2018 um 09:21 schrieb Frank Sheeran:
current = previous * multiplier + delta
Am 01.10.2018 um 09:21 schrieb Frank Sheeran:
> current = previous * multiplier + delta
Im a using this multiplication with offset to sequentially generate and
detune the frequencies for music
Hello Theo
Am 08.08.2018 um 20:03 schrieb Theo Verelst:
For instance when a FPGA board, cheaper than the CPU of a PC, beats the PC
in practical sense, there's every reason to prefer that solution,
especially
if the tools are getting more advanced than C compilers on a moderately
functioning
Hi Paula and others
I wrote so many articles about where and when to use FPGAs for wave
synthesis, that I cannot count them anymore. Only some short words in reply:
I agree that FPGAs do offer design techniques that cannot be done with
DPSs. But I hardly see them being made real in music
This code is also dangerous "LGPL" :-)
Seriously, I'm afraid this is also too much for him. Code is not really
good to explain solutions. I prefer the clarification and let people
code themselves.
Let's try it this way:
1. Apply an anti aliasing filter with an edge frequency of about
A HHR performes both simple reflection and self oscillation. At the
beginning of the triggering phase the sound runs into it and is
reflected at the inner walls. A little part of the sound comes out
through the hole while a larger part is reflected creating steady waves
as known in all rooms
Hello Richard and others
Am 10.07.2018 um 15:25 schrieb Richard Dobson:
I am very much into this topic of mapping live musical parameters to
(generative, real-time rendered) visuals, so I'm very interested in what
others have to add to this list.
Music to Video is also a thing which took my
Am 22.06.2018 um 01:11 schrieb robert bristow-johnson:
do you need an accurate fractional-sample-precision delay? that's how
you tune a comb filter to a specific note.
How would you achieve a sub sample delay just in software without a
filter? And consequently when using such a filter, why
This is not surprising since sin*sin + cos*cos = 1 :-)
But the problems, I mentioned remain, whereby people can lower issues by
blending in partitions with low dynamics (if possible).
Am 19.06.2018 um 07:49 schrieb Tom O'Hara:
> On 6/18/2018 6:42 PM, gm wrote:
>>
>> I find that in practice
Am 18.06.2018 um 08:13 schrieb Felix Eichas:
> There's also a paper regarding power complementary crossfade curves.
> Maybe a bit scientific but still worth a read:
>
> http://dafx16.vutbr.cz/dafxpapers/16-DAFx-16_paper_07-PN.pdf
>
> Regards,
> Felix
Interesting paper, I did not expect that
Hi Paula and all
Am 13.06.2018 um 14:35 schrieb pa...@synth.net:
Though, remember these are mass market products, they will use the
appropriate part for a given price point.
Right, wherey according to my exoeriences, the exisiting DACs Chips of
the higher price reagion we have nowadays
Am 13.06.2018 um 15:01 schrieb Niels Dettenbach:
By theory, any square wave could be constructed by a infinite number of
(sinus) signals, while many of that images seems like produced from a finite
number of such "signal parts". this means - if i think correctly - a really
perfect square would
Hello Kevin
I am not convinced that your application totally compares to a
continously changed sampling rate, but anyway:
The maths stays the same, so you will have to respect Nyquist and take
the artifacts of your AA filter as well as your signal processing into
account. This means you
Hello
Apart from the mentioned book of Neumann, I could recommend Sengpiel
Audio. It is still available in the internet and supported by the son of
EBS (AFAIK).
You will find many answers there. Some pages are available in english
too, such as the conversion calculations:
Hello all
Am 22.05.2018 um 14:11 schrieb Theo Verelst:
> fundamentally limited by the length of the sinc (-like) "perfect"
> resample kernel, and the required delay for accurate re-sampling might
> be considerable!
This can be limited by an increasing sampling rate reducing the
coarsness, but
Hello Theo
bandwidth is indeed an interesting point.
I have done a calculation regarding the number or operations requiring
RAM access to get and store the information for pipelined filtering and
sound processing over the time slices. Summarizing all the (up to 4
reads and writes per RAM)
Hello all,
something general about that:
As most auf you know, I developped my first FPGA synth in 2004 (Spartan
2) starting from DSP routines formerly running on DSP 56302 and such,
like available in Sydec's Mixtreme and Soundart's Chameleon and since
then I had been envolved in many
Hello all,
something general about that:
As most auf you know, I developped my first FPGA synth in 2004 (Spartan
2) starting from DSP routines formerly running on DSP 56302 and such,
like available in Sydec's Mixtreme and Soundart's Chameleon and since
then I had been envolved in many
Just for clarification: In theory only one inverter with feedback is
required in order to have an instable, oscillating circuit. Practically
the technology is too quick and the amplitude will not be high enough to
define good levels. More than one of them will lead to steady states of
the
I wonder how a "grey" version of shades might sound like - will have to
think about that :-)
Good paper indeed - seems to be easy to implement with a low number of
resources. I found similar circuits based on LFSR-structures when
designing and experimenting for radar some years ago - but am
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