On Dec 21, 2010, at 11:22 AM, Nigel Redmon wrote:
As for things like distortion modeling of guitars, I can tell you
that windowed sinc is involved, at least on the upsampling leg where
you likely want to preserve phase.
...
As long as you lowpass filter the signal first, then you're only
On Dec 21, 2010, at 10:45 PM, Ross Bencina wrote:
robert bristow-johnson wrote:
one thing i might point out is that, when comparing apples-to-
apples, an optimal design program like Parks-McClellan (firpm() in
MATLAB) or Least-Squares (firls()) might do better than a windowed
(i presume
On Dec 23, 2010, at 8:56 AM, Charles Turner wrote:
Here's my limit case: let's assume some typical laptop with CD-
quality sound generation capability with a sample rate of 44.1khz
and sample size of 16 bits. I create a sinusoidal waveform on the
computer with a period of 4,410hz. I choose
On Dec 23, 2010, at 8:56 AM, Charles Turner wrote:
Here's my limit case: let's assume some typical laptop with CD-
quality sound generation capability with a sample rate of 44.1khz
and sample size of 16 bits. I create a sinusoidal waveform on the
computer with a period of 4,410hz. I choose
On Dec 23, 2010, at 12:46 PM, Nigel Redmon wrote:
On Dec 21, 2010, at 8:36 PM, robert bristow-johnson wrote:
and trying to point to an obvious advantage to any windowed sinc
(that you don't have to compute the FIR when the output same lands
squarely on top of an input sample when al
On Dec 23, 2010, at 2:09 PM, Nigel Redmon wrote:
Somehow, we are talking about different things maybe?
possibly. i'll admit that i'm trying to be a little-bit anal (or OCD)
with the language.
In what I'm talking about, the key is that "n" is not integer,
not even when the output sampl
nput over. this is because of
the nature of the sinc() function and is not directly because of half-
band filter, but it *happens* to be the case that for 2x upsampling,
this windowed-sinc is *also* a half-band filter.
On Dec 24, 2010, at 5:16 AM, Nigel Redmon wrote:
On Dec 23, 2010,
On Dec 27, 2010, at 2:05 AM, Nigel Redmon wrote:
Hi Robert,
hi Nigel,
No need for me to address point by point, because I agree with
everything you say, except for one major point (which affects a few
things you said)...
You seem to imply that a windowed-sinc created for 2x oversampl
*only* advantage you
get with windowed-sinc and you may as well lay it by the wayside and
move on to an optimal design. i had never suggested any other
windowed-sinc design and i wouldn't really consider doing such. i
will confess that when i said early on:
On Dec 23, 2010, at 1:18 PM,
red no other
use of the sinc() design until this exchange:
On Dec 27, 2010, at 11:03 AM, robert bristow-johnson wrote:
On Dec 27, 2010, at 2:05 AM, Nigel Redmon wrote:
...
Ideally, you would want everything from 0.50 to 1.00 to be "clear"
to a reasonable degree. It's not. It'
well, okay, one more round...
On Dec 28, 2010, at 12:59 PM, Nigel Redmon wrote:
On Dec 21, 2010, at 11:22 AM, Nigel Redmon wrote:
As for things like distortion modeling of guitars, I can tell you
that windowed sinc is involved, at least on the upsampling leg
where you likely want to prese
On Dec 28, 2010, at 5:43 AM, Stefan Westerfeld wrote:
On Mon, Dec 27, 2010 at 02:53:30PM -0500, Bogac Topaktas wrote:
I'd like to compute the continuous wavelet transform (CWT) of my
input
signal (audio file) with the morlet wavelet, to get a time
frequency plane
which corresponds to the ti
so was (or is) it Line6 or was it someone else? (this is my excuse
for responding when i said you could have the last word.)
i sayed:
"was" is not the same as "is".
On Dec 28, 2010, at 6:54 PM, Nigel Redmon wrote:
It "was" designed in. It "is" in products that I could go buy at
guita
On Dec 28, 2010, at 11:51 PM, Nigel Redmon wrote:
Would it have been better if I said, "I can tell you that windowed
sinc is used"? Hmm, I have a feeling that you might read that as
"... is exclusively used", not sure...
depends on what the meaning of "is" is. when Bill Clinton, when firs
On Dec 29, 2010, at 9:10 PM, Nigel Redmon wrote:
i think we skeered 'em, Robert
;-)
my driver's license photo looks pretty scary (but my facebook, linked-
in, whatever isn't so scary). i got accosted once by plain-clothes
NYC cops about a year ago. they said they stopped me because i
On Jan 4, 2011, at 11:03 PM, Didier Dambrin wrote:
My new additive synth features full control on the filter, and I
learnt a lot about "good sounding" resonance. Since I can control
pretty much anything, I can shift the resonance point around the
cutoff point, it's very useful musically. T
to mean the thing we don't like,
not the "images", which just exist.
On Dec 23, 2010, at 8:41 AM, robert bristow-johnson wrote:
On Dec 23, 2010, at 11:31 AM, Nigel Redmon wrote:
Technically, there's always aliasing--it's a matter of whether
it's in the audio band,
On Jan 5, 2011, at 12:02 AM, Didier Dambrin wrote:
I said "additive" :)
I was talking fully in the freq domain, & it's really nice to be
free of the restrictions of IIRs (which I never really understood).
i get it. no post-filtering.
so are you applying to the additive components some s
On Jan 11, 2011, at 1:23 PM, Thomas Young wrote:
I need to develop a real-time multiple band EQ DSP effect, but I am
unsure about how to approach it.
do you mean a graphic EQ?
My preferred approach would be to FFT-> Modify Spectrum-> IFFT,
if you do that, better look up the concepts call
On Jan 11, 2011, at 7:01 PM, Tom Wiltshire wrote:
I'd approach this from a analogue-thinking angle and design a
tunable parametric EQ stage and then parallel a load of them up,
like Robert suggested.
that's not exactly what i meant to suggest. what goes in parallel are
not simply these
On Jan 25, 2011, at 7:34 PM, Jan Baumgart wrote:
When the two signal portions are alike, they are strongly correlated
- so you get a maximum value for the correlation.
If they have "nothing in common" you get a correlation value near
zero.\
he said he was using periodic function generation
On Jan 28, 2011, at 4:47 PM, Nigel Redmon wrote:
I've been on a number of patent cases (as software expert, sometimes
electronics), big players, on both sides...
First, patents are important, and help progress. Non-obvious
advances often come from expensive and lengthy research. Imagine a
On Jan 31, 2011, at 12:02 PM, Andy Farnell wrote:
Hi Ross,
Are you suggesting by stating the above axiom that algorithms are
_simply_
ideas and that for this reason alone they shouldn't be patentable?
Yes I am, you've got it.
An algorithm is unsufficiently concrete to deserve a patent,
On Feb 7, 2011, at 6:54 PM, Tom Wiltshire wrote:
On 7 Feb 2011, at 20:54, Andy Farnell wrote:
Do a search on "Yamaha Patent FM". Does that look like a
widespread interpretation that is clear and unambiguous to you?
My argument is simple at this point. Development was stifled.
This is an i
On Feb 20, 2011, at 5:37 PM, Thomas Rehaag wrote:
> 1. oversample 2 times
> 2. multiply
> 3. downsample 2 times
Wow, why didn't I think of this myselfe? The convolution would be
much easier / better for me but you already saved me with this
suggestion.
> I thought it goes like: "convolut
On Mar 13, 2011, at 12:09 AM, Ross Bencina wrote:
Andy Farnell wrote:
How do you know these filters don't have a resonance?
That could explain your results.
I doubt those filters would have explicit resonance/peaking at the
cutoff (it is a lowpass EQ after all).
But assuming they are us
another way to think about it is to pretend that your filter, whatever
it is, is a "matched filter". "matched to what?" you say. it's
matched to a signal that looks just like a time-reversed copy of the
filter's impulse response. so whatever the impulse response of the
filter is, if th
On Mar 17, 2011, at 9:21 AM, Wen X wrote:
As far as causality is concerned it's the *group* delay that should be
non-negative.
well, even group delay is negative with the peaking filters, for
*some* frequencies.
with group delay, there is no issue of phase unwrapping since the
phase del
On Mar 17, 2011, at 12:00 PM, Wen X wrote:
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
well, even group delay is negative with the peaking filters, for
*some* frequencies.
Yes, but only if the filter
On Mar 17, 2011, at 2:27 PM, xue wen wrote:
Yes, but only if the filter has high (negative?) dispersion
at that
frequency.
i'm not sure what that means. my understanding of dispersion would
be
a rapid change of phase or delay vs. frequency.
my understanding is if different freq
On Mar 17, 2011, at 8:05 PM, Andreas Beisler wrote:
Hi. Sorry, I messed up the subject of the thread.
that's whacha get fer using the digest form. that'll teach ya!
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
--
dupswapdrop -- th
On Mar 18, 2011, at 1:55 PM, Wen X wrote:
- when considering finite duration there is the uncertainty
principle, so
you always deal with a pack of frequencies rather than one
frequency, which
makes "latency" dependent on the content of that pack.
- however, using FT[th(t)]=j(FT[h(t)])', on
On Mar 26, 2011, at 8:30 PM, kalle@helsinki.fi wrote:
this might be of interest:
http://www.csounds.com/node/1475
hi Kalle,
i haven't dug into the detail, but i have some idea of the method. i
thought that there was some kind BLIT technique that was similar. i
remember it integra
On Apr 6, 2011, at 10:10 AM, Diemo Schwarz wrote:
"Common digital specifications are 24 bit/96 kHz. 24 bits provide
enough dynamic resolution, but 96 kHz is far from being sufficient
when it comes to time resolution: our hearing capabilities would
require sample rates of around 500 kHz.
On Apr 7, 2011, at 3:33 AM, Victor Lazzarini wrote:
and do you have hunch what the result might be?
i know i might just be speaking for myself (i'm 55 and, probably due
to both genetics and that i like to listen to loud rock music, live or
not, and had done that since my teens, and i'm 30
On Apr 27, 2011, at 1:38 AM, Ross Bencina wrote:
eu...@lavabit.com wrote:
*out++ = data->amplitude[0] * sinf( (2.0f * M_PI) * data->phase[0] );
*out++ = data->amplitude[1] * sinf( (2.0f * M_PI) * data->phase[1] );
/* Update phase, rollover at 1.0 */
data->phase[0] += (data->frequency[0] / SAM
On May 17, 2011, at 5:09 AM, Vadim Zavalishin wrote:
You mean this one?
Analyzing the Moog VCF with Considerations for Digital Implementation
by Tim Stilson, Julius Smith
http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.50.3093
No, there was another one dealing with zero delay feedbac
On May 17, 2011, at 6:27 PM, Ross Bencina wrote:
robert bristow-johnson wrote:
even though the cookbook yields coefficients for Direct 1 or
Direct 2 forms, it's pretty easy to translate that to the state-
variable design if that is the form you wanna use.
I've often wondered
On May 18, 2011, at 5:28 AM, Vadim Zavalishin wrote:
As far as I can tell there are only two state variables, v1 and v2,
and also their previous values v1z and v2z. I'm not sure that the
input v0 and its previous value count as state in this sense, but I'm
not really up with the lingo, so pleas
On May 20, 2011, at 7:43 AM, Ross Bencina wrote:
robert bristow-johnson wrote:
i don't have time now to complete the analysis, but here is my
first pass at getting the z-plane transfer function (something to
compare to the DF1 or DF2).
Thanks very much Robert,
yer welcome. i th
On May 21, 2011, at 11:27 PM, robert bristow-johnson wrote:
t-1t
y(t) = integral{ x(u) du} + integral{ x(u) du}
-inf t-1
~= t(t-1) + x(t)
this should be
On 5/22/2011 5:27 AM, robert bristow-johnson wrote:
[...]
which might be what Hal gets, i think. it's the only way to make
the claim that the Qc coefficient is independent of w0 and depends
only on Q. but if the resonant frequency is closer to Nyquist, you
need to scale Q with a
On Jun 26, 2011, at 7:52 PM, Didier Dambrin wrote:
pretty sure that on a piece of hardware 20 years ago, it couldn't be
anything else / anything in the freq domain
As an introduction to time stretching I thought I'd try and emulate
how some of the older hardware samplers used to do it. The
On Jun 26, 2011, at 8:41 PM, Stephen Blinkhorn wrote:
I'm looking at the time stretch for now preferably real-time.
okay, so now we gotta get something clear: this time-stretcher has
more samples coming out than going in, right? now, how is that done
real time? you could have it empty a
On Jun 27, 2011, at 12:38 PM, Stephen Blinkhorn wrote:
On 26 Jun 2011, at 20:32, robert bristow-johnson wrote:
On Jun 26, 2011, at 8:41 PM, Stephen Blinkhorn wrote:
I'm looking at the time stretch for now preferably real-time.
okay, so now we gotta get something clear: this
hi Olli (and others)...
i was reviewing this thread because i wanted to read what Stefan
Stenzel had said and realized that you had posted this response, and i
don't think i or anyone had responded to it. i don't remember reading
it (it must be the cannabis). i hope you're listening Olli
On Jul 13, 2011, at 9:29 AM, Olli Niemitalo wrote:
On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson
wrote:
On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote:
[I] chose that the ratio a(t)/a(-t) [...] should be preserved
by "preserved", do you mean constant over all t?
On Jul 14, 2011, at 5:36 PM, Olli Niemitalo wrote:
On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson
wrote:
g(t) = 1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )
might this result match what you have?
Yes! I only derived the formula for the linear ramp, p(t) = t/2,
because one can get
On Jul 15, 2011, at 12:46 AM, Sampo Syreeni wrote:
What are you trying to accomplish here, really? Optimum splicing,
sure, but against which precise criterion?
the precise criterion is how well the two signals being spliced
correlate to one another. i tried to set that up with the inner
On Jul 29, 2011, at 5:00 AM, Alexandros Tsilfidis wrote:
In the dereverberation context, room reverberation is regarded as
the combination of early reflections and late reverberation. It is
well known that early reflections produce a spectral degradation
which is perceived as coloration, wh
.burl.east.myfairpoint.net
[70.109.187.95]) (authenticated bits=0)
by mail10c26.carrierzone.com (8.13.6/8.13.1) with ESMTP id
p6TMjFnh006345
for ; Fri, 29 Jul 2011 22:45:16 GMT
Message-Id: <47043a22-b58b-4d04-81f5-369a4bd0f...@audioimagination.com>
From: robert b
On 8/2/11 9:32 AM, Igor Brkic wrote:
On Tue, Aug 2, 2011 at 2:28 PM, Conley, Dylan
wrote:
Is anyone aware of an open source pitch-shift algorithm implementation that is
quick (< 2ms) precise (to within 0.5 cents) and leaves the formant intact?
... you can do that in two steps: first do
pitc
On 8/2/11 12:01 PM, Wen Xue wrote:
This might be purely theoretical -
but can you pitch-shift something below 500Hz with<2ms delay at reasonable
precision?
no, not in a meaningful way. i didn't realize in my earlier response
that the OP spec'd that. it's an unreasonable spec.
There doesn't
On 8/2/11 2:04 PM, Steffan Diedrichsen wrote:
Since you implement for a synthesizer, you may look into the option for an
off-line pitch detection and real-time grain-synthesis. Grain synthesis has a
nice formant control and is fairly easy to implement.
i think that this "grain synthesis" is ess
On 8/4/11 8:32 AM, Conley, Dylan wrote:
I did have a couple follow up questions that I hope aren't too irrelevant.
Because we are working with the VST spec (and temporarily within an
implementation of the Java MIDI Interface) we will have access to all MIDI
information. Assuming the instrument
well, the math for the sampling and reconstruction theorem (from where
we understand the zero-order-hold effect on frequency response from a
conventional D/A converter and from where we understand the basis of
bandlimited interpolation, resampling or sample-rate conversion) is
pretty straight
On 8/24/11 1:51 PM, Andy Farnell wrote:
...
So my question for you Theo ... put on the profs hat...
How would you make these very powerful and (to me) wonderful
and mind boggling things in signals theory interesting and relevant
in an age where we have to compete with autotune and facebook?
i am not a Java programmer, but i think i can read this code.
where does the symbol "buffer[]" get declared? i resume you're getting
opBuffer[] operator.buffer.
private void modulate( final int numFrames )
{
clear( numFrames ); // zero buffer
for( @NotNull final Lin
what Brad Smith points out (that at least 1 sample delay is necessary
for feedback) is true for any discrete-time processing alg. and we know
that if "block processing" or "chunk processing" (whatever you wanna
call the technique) would require a minimum delay of BLOCK_SIZE samples
for any sig
if you don't want intermodulation distortion, then you mix audio signals by
scaling and adding. can be done in the frequency domain, but it's still
rectangular form addition. the mix scaling can be done with the polar-> rect
conversion. you can do that conversion reasonably efficiently if yo
Thomas Young [thomas.yo...@rebellion.co.uk] writes:
> Refactoring the filter to be with respect to n outputs behind (when
> using vectors of length n) is an excellent idea. I was a bit skeptical
> that the maths was correct there, but having read it over and
> stuffed some numbers into excel in
On 11/2/11 2:37 PM, David Reaves wrote:
When you use two-pole (second-order) filters, not only is the design more
complex, you also risk phase anomalies around the crossover point, usually
requiring you to invert the polarity of one of the bands.
this might be when it's useful to look up Link
On 11/27/11 12:23 PM, Dominique Würtz wrote:
Hi all,
I recently got interested in the approach from [1] to design of digital
EQs. The main idea here is to introduce a new degree of freedom G1 in
the prewarped analog prototype Hp(s) where G1 is the filter transfer gain
at Nyquist frequency which
On 11/27/11 3:17 PM, Dominique Würtz wrote:
Any ideas?
Knud Christensen "A Generalization of the Biquadratic Parametric"
http://www.aes.org/e-lib/browse.cfm?elib=12429
Hmm, reading the abstract I'm not 100% sure if it really addresses what
I'm aiming at. Sorry for being sceptical, b
On 12/8/11 4:36 PM, Theo Verelst wrote:
robert bristow-johnson Sun Nov 27 17:29:14 EST 2011
wrote:
On 11/27/11 3:17 PM, Dominique Würtz wrote:
>
>>>Any ideas?
>>Knud Christensen "A Generalization of the Biquadratic Parametric"
>>http://www.a
On 12/9/11 12:55 AM, Michael Olsen wrote:
Robert,
well, since, i have received a pdf copy of the Christensen paper. i
am willing to send it along to any small quantity of people who ask.
i realize the AES would rather that people get the paper from them
and pay for it, but if the cost is $
there's a guy there with handle "Clusternote" (who might be lurking here
for all's i know) who is slugging it out with an IP (can't imagine who
that is) about the math that goes into additive synthesis. if you ever
bother to edit the en WP, it might be a good time to examine the article
and
On 1/9/12 11:00 AM, Victor Lazzarini wrote:
Wouldn't it be nice if all of the knowledge embodied in this list could find
its way into Wikipedia, fixing the howlers and myths that exist in some of the
audio, synthesis, effects, computer music, etc pages? I know that some of us
have at time cont
On 1/9/12 11:58 AM, Scott Nordlund wrote:
I looked at it a bit, and it's a lot to juggle, looking at diffs and the back and forth. Maybe it's
just getting late, and I played a lot of basketball earlier, but the final thing that told me
"it's bed time" was, in skimming the article, "Its [RMI] wa
On 1/10/12 9:31 PM, Alen Koebel wrote:
I get paid to write, so I'm no stranger to research. I have edited the work of
others and had my work edited. Many here can say the same, I'm sure. With that
background I have tried to edit articles on Wikipedia. IMO, Wikipedia is
fundamentally a bad idea
On 1/10/12 11:29 PM, Scott Nordlund wrote:
On January 9, 2012 at 3:02:04 PM Veronica Merryfield
veronica.merryfield@shaw.cawrote:>
My feel is that to make it right, it probably needs more than a bit of
adjustment.
If this is to be fixed, I think it needs to be an organized effort. I scan down
On 1/11/12 10:50 AM, Thomas Young wrote:
Man I wish I hadn't gone to that wiki page now, it really is a mess and there
are some pretty glaring errors (missing brackets on the summation in the
Fourier series equation, and citation needed... wtf?)
-Original Message-
From: music-dsp-boun
On 1/12/12 2:41 AM, Ross Bencina wrote:
On 12/01/2012 4:01 AM, robert bristow-johnson wrote:
well, i cannot tell that the WP admins are going to do anything about
this other than wait for the page protection to expire (about 26 hours)
and then see what happens. if enough of us converge upon
hey, i appreciate the help from folks here (namely Olli and Ross)
dropping in on that Wikipedia article, now that it has been released
from protection.
please don't go away, there is lotsa stuff to do and we have time to do
it. it appears that this editor who wanted to rewrite everything
a
On 1/16/12 1:16 AM, Nigel Redmon wrote:
Nice improvements.
This may seem like nitpicking, but the "Timeline of additive synthesizers"
section seems to choose keeping the instrument name as the start of the sentence over
proper grammar. For instance:
Hammond organ, invented in 1934[26], is
On 2/6/12 3:28 PM, Nils Pipenbrinck wrote:
A quick question:
I am writing a little 31 band graphical equalizer (three bands per
octave), and I want to use the peaking-eq biquads from Roberts excellent
filter cookbook.
Everything is working fine so far, but I wonder what Q should I choose for th
On 2/7/12 1:45 PM, Nils Pipenbrinck wrote:
On 02/07/2012 06:04 AM, robert bristow-johnson wrote:
so it looks like you have 31 biquads in cascade, right? and they are
all peaking-EQ filters from the cookbook, right? (perhaps the bottom
band and the top band are shelving EQs.)
i would suggest
test.
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
--
dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp
links
http://music.columbia.edu/cmc/music-dsp
htt
you're not related to Miller Puckett, are you?
just curious.
and you're still welcome to the group no matter the answer.
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
--
dupswapdrop -- the music-dsp mailing list and website:
subscript
On 2/21/12 9:20 AM, Adam Puckett wrote:
No, I'm not related to Miller Puckette.
that's okay. yer still welcome.
:-)
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
--
dupswapdrop -- the music-dsp mailing list and website:
subscription
On 2/22/12 9:20 AM, douglas repetto wrote:
This is driving me nutz:
http://www.google.com
And now an image search for Hertz features lots and lots of pictures
of a non-sinewave!
Arrg!
i was wondering if it was the same Hertz. i guess it is.
sometimes Google's authority is dubious.
--
On 2/20/12 10:28 AM, douglas repetto wrote:
Hi Adam,
Welcome to the list. It's slow right now, but no doubt it'll flare up
again soon!
no shit
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
--
dupswapdrop -- the music-dsp mailing li
changed the subject line to something more accurate...
On 2/26/12 9:25 AM, Ross Bencina wrote:
On 27/02/2012 1:11 AM, Brad Garton wrote:
We're fooling around with the new Max/MSP gen~ stuff in class, it
seems an interesting alternative model for low-level DSP coding.
Once they figure out how t
On 3/24/12 4:45 PM, Linda Seltzer wrote:
Kindly allow me to provide further information on the job ad. The
experience requires advanced degrees in engineering or physics (this is
not a position for a music major unless the music major double majored in
engineering or physics). The areas of expe
i hadn't heard of this dev board before. at
http://www.st.com/internet/evalboard/product/252419.jsp it says that the
single unit prices is US$14.9 . is that right? that's nearly free.
where do the software tools (the compiler/linker/loader/etc) come from?
regarding wavetable indexing, som
On 4/9/12 1:29 PM, Julian Schmidt wrote:
Am 09.04.2012 19:16, schrieb robert bristow-johnson:
i hadn't heard of this dev board before. at
http://www.st.com/internet/evalboard/product/252419.jsp it says that
the single unit prices is US$14.9 . is that right? that's nearly f
i dunno why, but i can no longer reply to the thread that Julian
started. if this post gets to the list, then i think there is some
damaged header or something. this has happened to me before and it only
happens with this mailing list.
after hitting "Send", Thunderbird tries sending it and
[this is a fresh message, not a reply to any other, since none of those
seem to get past my SMTP server.]
On 4/9/12 5:25 PM, Julian Schmidt wrote:
> Am 09.04.2012 23:22, schrieb Olli Niemitalo:
>> On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt
>> wrote:
>>> I really think it is an aliasing
testing 1,2,3...
this is identical to a previous message (that would not get past my
SMTP) with this sentence added and the subject header changed..
On 4/9/12 5:25 PM, Julian Schmidt wrote:
Am 09.04.2012 23:22, schrieb Olli Niemitalo:
On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt
wrote:
another day, another restarted computer, let's see if i can post to this
thread (that was the weirdest of problems). here's hoping my SMTP
server doesn't reject this...
it *did* reject it. Thunderbird says: "Alert. An error occurred while
sending mail. The mail server responded: 5.7.1 q3A
now that i think of it, if you're doing linear interpolation and you
forget to add that extra repeated point at the end of the wavetable:
float wavetable[257]; // one extra point for doing linear interpolation
// make sure that wavetable[256] = wavetable[0]
you will
On 4/10/12 1:18 PM, Nigel Redmon wrote:
Excellent point, Robert. (Another way to avoid it is to mask the index with
0xff, if you want to keep the table 256, but not check for wrap in the
mid-interpolation.)
sure, but that masking is otherwise unnecessary (the initial index will
always be 0 <
it was pretty spare in the mail. essentially just the board and a cute
little card in a bubble box.
the card has some "Getting started" instructions and number 5. says to
got to http://www.st.com/stm32f4-discovery tutorial, and i'll do that
soon. it also mentions dev toolchains: Altium Ato
On 4/12/12 10:06 PM, Eric Brombaugh wrote:
On 04/12/2012 05:53 PM, robert bristow-johnson wrote:
it was pretty spare in the mail. essentially just the board and a cute
little card in a bubble box.
Yes, that's pretty much all you get. Bring your own mini-USB cable.
the card has
i wish i could program my Thunderbird client to *not* copy HTML formatting
(like this font change) from the quoted post in replying. and i wish that
music-dsp's Majordomo or whoever would not copy it either. that said...
everythingXue says is congruent to my experience (which is no longer ve
On 4/20/12 8:51 PM, douglas repetto wrote:
On 4/20/12 1:10 PM, robert bristow-johnson wrote:
i wish i could program my Thunderbird client to *not* copy HTML
formatting (like this font change) from the quoted post in replying.
and i wish that music-dsp's Majordomo or whoever would not co
engineer. You also learn engineering by solving real problems
and maybe breaking things. Chances are pretty good that your early
attempts are/were crap.
Many of you know Robert Bristow-Johnson.
oh jeepers.
He is a bit famous in this group because in part, he did the rb-j
cookbook.
one
On 5/7/12 5:45 PM, ChordWizard Software wrote:
I am working on a new project using PortAudio and testing it with a waveform
stored in a buffer. This could be generated myself (sine, square, sawtooth,
etc) or a more complex waveform loaded from a file.
I want to be able to render the waveform a
On 6/8/12 1:36 PM, Charles Turner wrote:
I was initially hesitant to post to the list as I haven't explored
this topic very deeply, but after a second thought I said "what the
hell," so please forgive if my Friday mood is more lazy than
inquisitive.
nothing wrong with posting this. nothing at
Douglas, i am getting that weird refusal to accept the email (saying
that it is not properly formatted) unless i change the subject line. i
just don't get it.
On 6/11/12 1:58 PM, Thomas Young wrote:
GA isn't really supposed to mimic the real world as closely as you are
suggesting, in the r
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