usenet@apple-o.com (Mad Scientist) wrote in message news: ...
> I'm looking for a utility or to program one that can take a sound file
> (WAV) and take a second file, and isolate what the two files have in
> common, and what is different about them.
>
> For instance if you have two WAV f...
On Mon, 04 Aug 2003 16:48:10 -0500, Dennis@NoSpam.com wrote:
> ricklyon@remove.onemain.com (Rick Lyons) wrote:
>
> > Hi,
> > agreed. Hey guys, did we all agree that the Sliding Goertzel
> > (unlike the standard Goertzel algorithm) only operates at
> > integer values of k?
> >
> > [-...
Hi Guys,
I've been trying to learn about those darned cascaded
integrator-comb (CIC) filters and have reviewed
Hogenaurer's original paper
Hogenauer, E. "An Economical Class of Digital Filters
For Decimation and Interpolation," IEEE Trans. Acoust.
Speech and Signal Proc., Vo...
It's 2 AM and I ought to be sleeping. I'm not. I've been thinking about
cepstra for hours. And here's what came out of it, apart from the hope
that I, if nothing else, can get a half nights sleep after having posted
this...
"Fred Marshall" wrote in message news: ...
> "Rune Allnor" ...
I'm trying to implement "Tempo and Beat Analysis of Acoustic Musical Signals" in C#.
Currently I made LowPass, HighPass, BandPass filter, And Sound IO class.
I filtered the music, and marked All Beats of Lower than 200Hz, Upper than 3200Hz.
And I got the result which has a Lot of Beats. (...
Ronald wrote:
(snip)
> You forget that the simplest true physical component has a close
> to perfect delay. It's called a wire.
> Any non-zero length wire will have a delay constrained by the
> speed-of-light in the medium (usually Cu or Al surrounded by a
> insulating dielectric)....
"Rune Allnor" wrote in message
news:f56893ae.0311231001.3cc8a505@posting.google.com...
> Hi all.
>
> I have just "made" some recordings of birds nearby (which is to say that
> I ripped them off a CD released by the local ornithology society, but
don't
> tell anyone) and want to play...
phunkyman wrote:
> First of all, i thank you for your answer.
>
>
> > Right now, an antenna directly to an ADC is beyond the state of the art.
>
>
> You mean it's not good to do it? I know, i just said that to "go
> fast". I've got a minimum knowledge in electronics! lol
I th...
wrote in message
news:1131134435.441548.34430@g14g2000cwa.googlegroups.com...
> Dear members:
>
> Plz tell me what is the point I am wrong. It is my exam.
>
> I have to plot the magnitude of the freq response a Low Pass filter:
>
> y[n]=1/3*(x[n]+x[n-1]+x[n-2]);
>
>
> I use...
David Joseph Bonnici wrote:
> Hi, I need the name of the filter that has a frequency response of the
> form (x/sin(x))^N.
>
> See a picture of the frequency response at
> http://www.geocities.com/netspiri/xonsinxtothepowerofN.jpg
> N here is set to 3. I know for sure that it is an FIR...
Hi Randy,
Thanks for your input. I did find where I was going wrong all the
while. It was in the interval I was considering my input for and the
length of the FFT being used. That mismatch, made the integrators go
unstable.
Its a simple SDM, I was trying for.
But presently am working to...
That's the whole point - the "hum" can have any number of harmonics,
and the only way to cut them all back is attention to the analog
circuitry. But if it's too late for that, something like an inverse
comb filter will do it.
On Thu, 01 Jul 2004 22:59:17 -0400, Jerry Avins wrote:
> Suoda...
"Jesper Kristensen" wrote in message
news:40ee5183$0$23874$14726298@news.sunsite.dk...
> "Stephan M. Bernsee" wrote:
> > If we're talking about comb/allpass filter (Schroeder/Moorer) type
> > reverbs, there are typically two types of HP/LP in use in such a unit:
> > one is affectin...
Jerry Avins wrote:
> Tim Wescott wrote:
>
> ...
>
>
> > If you need to know harmonic distortion then you need a comb filter. One
> > way to implement this would be as a bunch of bandpass filters in
> > parallel -- one at 60Hz, one at 120, one at 180, etc. If your 35
> > samples p...
On Wed, 22 Sep 2004 08:23:30 -0500, jim
wrote:
>
>
> Jerry Avins wrote:
> >
> > jim wrote:
> >
> > ...
> >
> > > Hi Martin
> > > The frequency response of your filter [1,1] is just cos function.
> >
> > ...
> >
> > That seems very strange to me; I ...
On 27 Sep 2004 10:32:50 -0700, egan_nc@yahoo.com (Eng Gan) wrote:
> Hi,
Hi Eng,
> Can any of you please help me with the question below?
> Some paper said that CIC filter has a high gain and needed to be
> compensated?
Humm, I wonder where you read that(?).
> From Mattew P. Donadio...
Jerry Avins wrote:
(snip)
> First, I want to admit to being flattered that you use the explanation I
> gave at the start of this thread, that the replications are present in
> the analog signal after the (salient or implicit sample-and-hold) before
> any digitizing takes place. Seco...
"Rune Allnor" wrote in message
news:f56893ae.0410052220.4fc1b189@posting.google.com...
> "Fred Marshall" wrote in message
...............................
>
> > I would have said that a FIR filter needs to have enough frequency
> > samples
> > (when you FFT the coefficients ...
Charles Oram wrote:
>
> I forgot to mention an important detail - the instrumentation amplifier has
> a gain of 8 and the ADC has an adjustable gain that is set to 4, so after
> those gains we have a 1.44V signal with 128mV superimposed on it.
> The ADC has a Vref of 2.4V, so 1% of 1...
BobG wrote:
> I listen to the am talk radio during my commute. There are a couple
> places where the 60 and 120 hz hum from power lines drowns out the
> talk. Would a box with a 8.3ms delay filter this out some? (never mind
> the logistics of how to hook it into the darn radio...)
Repor...
in article f56893ae.0412010412.f910ba5@posting.google.com, Rune Allnor at
allnor@tele.ntnu.no wrote on 12/01/2004 07:12:
> Hi all.
>
> I have got involved with this signal detection and characterization
> problem where we have reason to expect that the useful signal is
> hidden in...
On Fri, 07 Jan 2005 19:57:32 -0500, robert bristow-johnson
wrote:
> in article 348maaF464m2bU1@individual.net, Jon Harris at
> goldentully@hotmail.com wrote on 01/07/2005 19:03:
>
> > "glen herrmannsfeldt" wrote in message
> > news:crn4gd$kdm$1@gnus01.u.washington.edu...
> ...
> >...
Cathal Campbell-Shaw wrote:
> Hi,
>
> I'm a student and I'm working on a project where I convert a voltage
> signal to light and then receive this light signal and convert it to
> voltage once again. I use a DAC to create the initial voltage signal
> and a ADC to convert the received s...
robert.w.adams@verizon.net wrote:
.=2E.
> Is there some set of coefficients or
> particular structure that will gaurantee that all poles and zeros lie
> on the unit circle, in a way that is similar to the LC filter mentioned
> above?
You can have N evenly distributed poles on the unit ci...
All,
Looking for any help here from my DSP cronies. I am analyzing a method
of filtering a periodic pulsed sinusoid signal with a bandpass FIR
filter and have hit a stumbling block! As you know, the spectrum of
the pulsed periodic sinusoid results in line spectra based on the pulse
length (...
>
> Exactly what do you do? I believe there is a post filter involved after the
> upsample stage. If you leave that out, you'd get all sorts of artifacts
> in the data.
>
> Rune
>
Hi Rune,
I take a buffer of size bufsize and:
1. Throw away every second sample to obtain a buffer...
in article 070420051129510384%x@x.x, ben at x@x.x wrote on 04/07/2005 06:30:
> In article , robert
> bristow-johnson wrote:
>
> > i wouldn't do that. you still need to deal with the possibility of missing
> > or weak harmonics (inc. fundamental). i agree with Dmitry Terez about n...
Hi Jerry,
I am working on NTSC decoding in software using quadrature sampled NTSC
signals (8 Mhz complex samples with audio at 2.75 Mhz and video at
-1.75 Mhz). I have implemented a comb filter, but it is not working
perfectly because of the drift in video carrier frequency. Can someone
te...
in article 180420052120194542%x@x.x, ben at x@x.x wrote on 04/18/2005 16:22:
> can anyone tell me how or if AMDF & ASDF relates to fourier transforms
> please? -- do AMDF & ASDF make use of fourier transforms? or are part
> of their logic based on fourier transforms in some way? or are AMDF ...
I finally got around to picking up the April 4th EET while eating
supper tonight and who do I see on the inside of the front page?
Rick's got an article on embedded.com on CIC filters. Cool stuff,
man. The competition is tough around here!
There is at least one thing in this article I don't ge...
Ben,
I hate to sound repetative, but a Fourier transform will give you
exactly what you want. It will seperate the spectum into "bins", but
it will be a continuous bin system, because it's no longer in the time
domain. A comb filter is good, if you know what frequencies you're
looking for,...
> Just another remark:
>
> Isn't your hum rather am approximatly squarewave kind off thing?
> So maybe you want to use a comb filter for the hum.
> So I would first try to get rid of the hum and dc offset before
> you split into frames (with a comb for the hum and a dc blocker for the
dc).
> ...
Fred Marshall wrote:
...
> Jerry,
...
> The convolution is maximum when the input frequency is the same as the
> frequency used to define the filter. Except for input pulse length being
> arbitrary in our case here, it's a matched filter to sinusoids at that
> frequency...
in article 6N-dnXYkfZuwpQvfRVn-qg@giganews.com, PaulWiik at paul@wiik.net
wrote on 05/26/2005 16:57:
> I feel I have made som major progress now. I had a lot of errors in my
> code prior to the last post. I believe I've been able to correct most of
> thit now (some of them actually have qui...
robert bristow-johnson wrote:
(snip)
> it wasn't so much the delta's dimensions where i was disagreeing with the
> books. (i really couldn't find any that explicitly addressed that
> particular issue.) it was with the books using the "dirac comb" function
> for ideal sampling, without...
Jerry Avins wrote:
> glen herrmannsfeldt wrote:
> > jagadeeshbp@gmail.com wrote:
> > > I have seen references saying about broadcast quality videos.
(snip)
> > It is about 3.5MHz video bandwidth at 30Hz or 25Hz frame rate.
> > Note that this is considerably better than standar...
"Duane" wrote in message
news:1127408858.623711.316030@f14g2000cwb.googlegroups.com...
> You generally don't do that.. Windowing allows you to better resolve
> some components when you're doing a block based transform. For windows
> that taper to zero near the ends, you really can't go...
On 11 Oct 2005 21:52:41 -0700, jim_nospam_beasley@yahoo.com wrote:
> I would like to know if I can predict what the group delay of a CIC
> filter is.
>
> I have an agressive FIR BPF where the sample rate is 2000 x the filter
> bandwidth. The group delay is longer than I would like it to be. ...
Randy Yates wrote:
...
> Chris (et al.), is this chroma filtering problem what causes the
> moire pattern you see on certain image inputs (vertical lines?).
What image source? With NTSC, it's the beat between luminance and the
3.58 MHz color subcarrier. A comb filter can greatly red...