zero's do not need to be evaluated, meaning the problem is
changed to one of offsetting your convolution algorithm (which may or may not
be practical in your situation), but does allow you to use half the number of
coefficients.
Thomas Young
Core Technology Programmer
Rebellion Developments LTD
In principle a patent protects the investment that a company makes in order to
develop a new technology; companies are unlikely to invest large amounts on
research if their ideas are going to be copied and their product beaten to
market by an opportunistic competitor.
In practice, as we all
What resources would you recommend Theo?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo Verelst
Sent: 24 August 2011 17:01
To: music-dsp@music.columbia.edu
Subject: [music-dsp] Electrical Engineering
If the goal here is education then I would suggest formatting your work as a
syllabus or structured as lessons with exercises. The raw information is
already available through books and sites like Wikipedia (where the wording and
presentation is already carefully reviewed), so I would have
Is there something wrong with just summing the buffers for each channel?
It's not really clear what you are trying to achieve, do you want to downmix
your channels?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of
Regarding:
buffer[i] = Math.sin( ( phase * ratio * frequency + buffer[ i ] ) * PI_TWO );
// FM (PM whatever)
phase += 1.0 / 44100.0; // advance phase in normalized space
Your oscillator will be prone to drifing out of phase due to the way you are
adding the reciprocal of the sample rate (a
I don't think musicians mind tweaking and poking presets (what's the worst that
can happen?), so it's really up to the makers to provide plenty of decent
preset sounds. NI FM8 for example has a pretty good selection of presets and is
a popular FM synth these days, I rate it pretty highly
Interesting. I was referring to 'propper' synths really, which I wouldn't
really group with romplers and black boxes personally, but I guess end users
don't necessarily make the same distinction. I can definitely
Personally I'm not making black boxes, because to do this you can't just be a
plan on implementing a multiband compressor shortly myself. Multiple
bandpass filters would seem logical, but there must be a bit of an issue with
colouring the signal.
Thomas Young
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun
'Biquad' is a filter topology meaning the transfer function is expressed as a
quadratic divided by another quadratic (biquadratic). They are pretty common in
the world of digital filters because there are very simple to convert into an
efficient algorithm (Direct form 1 etc...). See
I'd like to say well done to everyone who has edited this so far, it looks
massively better :)
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
Sent: 16 January 2012 16:16
To:
Most people seem to be overthinking this, Uli has posted the important equation
here:
Y [dBFS] = 20*Log10(X/FS)
That is all you need for converting normalised peak values to dbFS.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
conversion to dB.
On Jan 18, 2012, at 2:34 AM, Thomas Young wrote:
Most people seem to be overthinking this, Uli has posted the important
equation here:
Y [dBFS] = 20*Log10(X/FS)
That is all you need for converting normalised peak values to dbFS.
--
dupswapdrop -- the music-dsp mailing list
I've tried that before, basically writing a soft synth and a sequencer. I won't
pretend I made anything that sounded very good but it was fun. In the demoscene
there is a whole field of 'programmed music' which is just this, although
generally they use a sequencer for the composition and just
Ringing in your ears due to exposure to loud noise is the stereocilia (small
hair cells) being damaged and falsely reporting to your brain that there is
still sound vibration present. The frequency of the ringing is not a function
of the sound that damaged your ears (a super loud bassy sound
float pan = sin(2 * PI * frequency * time++ / 44100);
As 'time' increases, changes to 'frequency' will result in larger and larger
discontinuities. You should offset (add) the change in time rather than
multiplying by it.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
Maybe submissions should be added to a moderation queue rather than added
directly (i.e. they need to be manually whitelisted). I don't think a super
quick turnaround on new algorithm submissions is really important for something
like musicdsp.org.
-Original Message-
From:
:11 PM, Thomas Young thomas.yo...@rebellion.co.uk
wrote:
Maybe submissions should be added to a moderation queue rather than added
directly (i.e. they need to be manually whitelisted). I don't think a super
quick turnaround on new algorithm submissions is really important for
something like
You haven't really explained which aspect of the timbre you want to use to
organise the sounds, and timbre is such a catch-all word that you need to
specify the characteristics you are looking for in more detail to have any
chance of producing something useful.
Categorising by amplitude
Not getting a response on a mailing list doesn't mean people are disrespecting
you, we can do without the self righteousness thank you.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Bastian Schnuerle
Sent: 10
1) The low frequencies are audible
2) It's not speaker distortion, the low frequencies are present in the signal
I think the spectrum of the first signal can be a bit misleading, if you are a
bit more selective about where you take the spectrum (i.e. between the
asymptotic sections) the low
to be able
My thought was to interpolate the coefficients towards a flat response, i.e.
b0=1 b1=0 b2=0
a0=1 a1=0 a2=0
I tried some plots and this does basically work except that the characteristics
of the filter are affected (namely the peak gain exceeds 0dB).
Thomas Young
--
dupswapdrop
to the confusion, my peaking implementation is different for gain and
boost, so that the EQ remains symmetrical, a la Zolzer).
On Jan 3, 2013, at 9:34 AM, Thomas Young thomas.yo...@rebellion.co.uk wrote:
I'm pretty sure that the BLT bandpass ends up with zeros at DC and
nyquist
Yes I
: 04 January 2013 09:26
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response
On 4/01/2013 4:34 AM, Thomas Young wrote:
However I was hoping to avoid scaling the output since if I have to
do that then I might as well just change
Aha, success! Multiplying denominator coefficients of the peaking filter by A^2
does indeed have the desired effect.
Thank you very much for the help
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thomas Young
! Multiplying denominator coefficients of the peaking filter by
A^2 does indeed have the desired effect.
Thank you very much for the help
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Thomas
Young
Sent: 04 January
-johnson
Sent: 04 January 2013 18:25
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Lerping Biquad coefficients to a flat response
On 1/4/13 1:11 PM, Thomas Young wrote:
someone tell me what it was about
In a nutshell...
Q: What is the equation for the coefficients of a peaking
coefficients to a flat response
On 1/4/13 1:29 PM, Thomas Young wrote:
Er.. yes sorry I transcribed it wrong, well spotted
a1: ( - 2 * cos(w0) ) * A^2
ooops! i'm embarrassed! i was thinking it was -2 + cos(), sorry!
can't believe it. chalk it up to being a year older and losing one year's
Your mean square error procedure is slightly incorrect. You should take the
final signals from both processes, say A[1..n] and B[1..n], subtract them to
get your error signal E[1..n], then the mean square error is the sum of the
squared error over n.
Sum( E[1..n]^2 ) / n
This (MSE) is a
There is quite an audible loss of high end, it's especially noticeable on the
'double processed' example which sounds very low passed. Is that intentional?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Theo
As someone pointed out what you really want here is c++ templates, those
compile out completely and can be specialised for different types - which you
need for speed.
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf
apparently that is more emotional and personal for some people to be able to
neutrally communicate about
This from the man who wants to recapture the sound of records from his lost
youth :P
I think Sampo had a lot of good points personally. I would not dismiss the
value of simplification,
Interesting, thanks
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Ove Karlsen
Sent: 28 October 2013 11:57
To: music-dsp@music.columbia.edu
Subject: [music-dsp] Analog 24dB lowpass filter emulation perfected.
This reminds me of experimenting with polynomials as an amplitude enveloping
function for a soft synthesiser. There was something rather alluring about the
idea of a one-line-of-code amplitude envelope - unfortunately it made creating
the envelopes pretty tiresome, and when I thought about the
Thanks Doug :)
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of douglas repetto
Sent: 11 November 2013 17:19
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] [admin] another HTML test
Okay,
Thank you Douglas. Some of the toxic replies here have really harmed the
friendly spirit and approachability of this mailing list.
-Original Message-
From: music-dsp [mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of
Douglas Repetto
Sent: 22 August 2015 16:22
To: A discussion
http://musicdsp.org/ seems to be down, does anyone know if the webmaster can be
contacted to fix it?
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