what's the rule of designing 802.11 preamble?
Why is the STS designed to have 12 subcarriers out of 52 subcarriers?
How to consider using 12 subcarriers? why not 16 or other numbers?
On the other hand, how to derive the short training sequence
S(-26,26)=sqrt(13/6)*(0,0,1+j,0,0,0,-1-j,.......)in section 17.3.3 of the
IEEE 802.11a 1999 version?
Maybe the sequence is derived by PN sequence, what's the detail? How to
generate the PN sequence? What's the original state importing to the shift
register?
In fact, i find so many references, but there is no one telling why.
Thanks.
Feifei wang
|
9/9/2010 3:35:46 AM
|
2
|
"feifei.wang" <feifei.w...@n_o_s_p_a_m.nufrontsoft.com>
|
|
|
jya is doing fine.
It may be gratifying to know that I was missed, but it's embarrassing
to learn that I allowed people to worry needlessly. I'm fine, but my
computer is not. Getting to another computer and using Google was more
of a hassle than I wanted to deal with while there was still a chance
to bypass the faulty power-supply switch on the mother board, but I've
given up and bought a new machine that comes with some new problems. I
hope to work them out soon.
Mant thanks to R-B-J for getting in touch by telephone and reminding
me that people care and worry.
Jerry
|
9/8/2010 8:41:07 PM
|
1
|
Jerry Avins <...@ieee.org>
|
Simulating a Very Low IF (VLIF) receiver
Hi,
I'm using Matlab to simulate a complex mixer. It basically performs a
multiplication between the incoming complex signal "y" and the phasor
exp(jw).
It basically performs 2 operations:
1. Move the input signal from IF to 0
2. Remove the interferer at the image frequency
What I would expect more is a 3 dB gain in SNR, like the equivalent analog
image reject mixer.
Since the analog part of the VLIF receiver has a noise figure 3 dB worse
than the direct conversion receiver, I think I should recover the gap in
the digital receiver. Is my assumption correct?
Thanks
Alberto
|
9/8/2010 3:55:03 PM
|
0
|
"alberto.fuggetta" <alberto.fugge...@n_o_s_p_a_m.gmail.com>
|
Weibul Fading in Matlab
I know that Rayleigh fading channel can be generated in Matlab using
H = sqrt(0.5)*( randn( 1,1) + j*randn( 1,1) );
Any idea how to generate Weibull Fading in matlab? There is a command
wblrnd in matlab but it giving me the real number only. Kindly help me!
thanks,
sohaib
|
9/8/2010 2:26:32 PM
|
0
|
"ssohaib" <ssoh...@n_o_s_p_a_m.gmail.com>
|
Cross correlation via Hartley Transform
Hi ...
I am triing to get a cross correlation between two or (later 16) audio
channels with the hartley transform (I am doing that in Labview with FFT
comparison). But the results don't fit:
-> my source: http://rsbweb.nih.gov/nih-image/download/n...docs/thesis.pdf
on page 12.
So I am doing the Hartley transform on both signals H1(f) and H2(f) ... to
get the cross correlation -> iFHT(1/2[ H1(f) * H2(f) + H1(-f) * H2(-f) +
H1(f) * H2(-f) - H1(-f) * H2(f) ]).
I am really appreciated about every tip why this doesnt fit with the result
per FFT ?
Thank you very much
Peter
|
9/8/2010 12:15:38 PM
|
2
|
"pete26" <p2...@n_o_s_p_a_m.sbox.tugraz.at>
|
non-order dependant checksum
Hi
I've searched the internet unsuccessfully trying to find a non-order
dependent error detection algorithm that has low prob of undetected errors
(lower than a 2's complement add checksum). The reason being that my
algorithm processes image data in an order different that what's
transmitted to the FPGA. While I've got portions of the data in my L2, it
is more efficient to compute the checksum in chunks rather than re-read the
whole image before sending it to the FPGA. (wouldn't it be nice of C64x+
DMA's could compute a CRC while they send an image!)
Anyone heard of any non-order
|
9/8/2010 12:15:44 PM
|
7
|
"SailingDreams" <epat...@n_o_s_p_a_m.yahoo.com>
|
Cross Correlation via Hartley Tranform
Hi ...
I am working with a multichannel audio system and I want to calculate the
cross correlation between the channels per hartley transform (I am working
with Labview). I tried to compare the fft with the hartley transform but
got not really the same results.
Whats wrong in this equation please ?
FHT... Fast Hartley Transform
iFHT... Inverse Fast Hartley Tranform
H1(f),H2(f) Hartley Transform of Signal 1,2
H1(-f),H2(-f) flipped Hartley Transform of Signal 1,2
Crosscorr = iFHT(1/2[ H1(f) * H2(f) + H1(-f) * H2(-f) + H1(f) * H2(-f) -
H1(-f) * H2(f)])
Does anyone have
|
9/8/2010 12:15:52 PM
|
0
|
"pete26" <p2...@n_o_s_p_a_m.sbox.tugraz.at>
|
Much Value to DSP RTOS?
I am speaking with a potential customer who wants a DSP board designed
for his needs which I could also market generally. We will consider
the NRE price based on the value of having this as a product, but also
an RTOS he is willing to provide for this product.
We haven't discussed any particulars yet, but he claims his RTOS is
"accurate to" an eighth of a ms. I'm not entirely sure what that
means, but I assume he is saying a process can be kicked off on that
granularity. I don't have a good feel for how much value this would
add to a DSP board product. Do many DSP apps need an RT
|
9/8/2010 4:34:55 AM
|
9
|
rickman <gnu...@gmail.com>
|
[OT] Need input from ~(retirement age) engineers &/or geeks
I'm now > retirement age.
I took early retirement due to "on the job" spinal cord injury.
Though I never received an engineering degree, my in house
designation was occasionally "egineur" - [as in culdn't spell it
now I r 1 ;]
I'm looking for a volunteer opportunity. The local advocacy group
for the disabled couldn't direct me. I now live in *VERY* rural
SW MO. "Wheel chair ramps" are a current issue in *RURAL* SW MO.
One lady I spoke to said a former friend was a member of a group
of "retired engineers". I gathered that it MIGHT be an equivalent
of SCORE (Service Corps of R
|
9/7/2010 6:11:16 PM
|
2
|
Richard Owlett <rowl...@pcnetinc.com>
|
Modem question
Hello,
what a maximum range is possible for 4800bps full duplex wire (a pair of
0.9 mmm wires) modem ?
Regards
Roman Rumian
|
9/7/2010 12:55:16 PM
|
26
|
Roman Rumian <rumian.wytnij.@agh.edu.pl>
|
biquad over butter
Hi all.
I am trying to understand what the benefit would be of having a biquad
instead of a butterworth filter in analogue implementation. For example, if
I assume that -more or less- i need 2 integrating capacitor per pole, then
i need the same for both a 8th order butterworth or a 4th order biquad
(e.g. a OZGF). Therefore, they have the same cost, right? Is there any
application where a biquad is prefereable to a buttenworth??
Thank you very much.
|
9/5/2010 11:28:56 PM
|
1
|
"joeDiHare" <stecos2...@n_o_s_p_a_m.hotmail.it>
|
Could someone tell me what I should do to get rid of the following error.
Could someone tell me what I should do to get rid of the following error.
------------------------------- 14.pjt - Debug
-------------------------------
[Linking...] "C:\CCStudio_v3.1\C6000\cgtools\bin\cl6x" -@"Debug.lkf"
<Linking>
undefined first referenced
symbol in file
--------- ----------------
_comm_intr
C:\CCStudio_v3.1\MyProjects\124\14\Debug\copy.obj
_input_sample
C:\CCStudio_v3.1\MyProjects\124\14\Debug\copy.obj
_output_left_sample
C:\CCSt
|
9/5/2010 11:28:59 PM
|
0
|
"arungnair" <sct.a...@n_o_s_p_a_m.gmail.com>
|
IFIR technique for designing filters
hello
I want to design a lowpass filter using IFIR technique.
I have designed the model filter ( prototype filter) but I have problems
regarding design of interpolator. Plz help me wid that..
Anyone doesnt want to type here .. plz give ur contact no.. I am stuck wid
this problem for the last 3 days..
Plz help
Thanks
|
9/5/2010 6:05:46 PM
|
11
|
"enricophpdsp" <enricophp...@n_o_s_p_a_m.yahoo.com>
|
what is meaning of "WEIGHTS" w.r.t Low pass Filter?
Hello
After designing a lowpass filter ,, is there any term called "weights"
which i can calculate from the obtained result?
My teacher told me to design a simple lowpass filter with different values
of the ORDER. and told me to make a table for the weights? what is meant
by that.. plz reply..
Thanx
|
9/5/2010 5:56:53 PM
|
4
|
"enricophpdsp" <enricophp...@n_o_s_p_a_m.yahoo.com>
|
good books on digital and wireless communications.
Hello All,
I want to know what are the good books in the area of
digital and wireless communications. I already have
a good understanding of DSP and am a begineer to digital
and wireless communications.
Regards
Bharat
|
9/5/2010 6:55:23 AM
|
1
|
"bharat pathak" <bha...@n_o_s_p_a_m.arithos.com>
|
Calculating a cross correlation (to find delay) on irregulary sampled data
Hi there,
I have two sets of data which are irregularly sampled. These two sets are
two measurements of the same product but on different places.
Eg:
Sensor 1: time 1, 3, 7, 8, 10, 15 ...
Sensor 2: time 1, 2, 3, 5, 9, 10, 11, 14 ...
The measurements are irregular because they are done by people, not by an
automatic sampled system.
I would like to calculate the cross correlation between these signals to
obtain an estimate for a time delay.
The question however is how to do this because of this missing data. The
only thing I would come up with is interpolating the missing
|
9/4/2010 12:03:29 PM
|
3
|
"webinn" <da_ju...@n_o_s_p_a_m.telenet.be>
|
anyone heard from Jerry lately?
i'm a little concerned. i sent email. if i don't read anything back,
i'll call tomorrow (saturday).
google groups says that Aug 11 is his current last posting to any
newsgroup (which was comp.dsp).
r b-j
|
9/4/2010 5:32:36 AM
|
11
|
robert bristow-johnson <...@audioimagination.com>
|
need recommendation(s) for FFT algorithm to use
Hi Experts,
My application requires taking FFT and IFFT on a REAL array of points,
where the array size is fairly large (number of points in array ranges
between 2^20 to 2^27, which is 1M to 140M). Once in the frequency domain, I
zero out all bins except for those bins associated with spurs, then IFFT
the array back.
I'm looking for a recommendation for an algorithm suited for this purpose
(I'm not that fluent in understanding which flavor of FFT is better than
another here). I currently use Matlab (based on FFTW), but I've been
looking at what's available on the internet and the
|
9/4/2010 3:02:06 AM
|
11
|
"ggk" <all4...@n_o_s_p_a_m.comcast.net>
|
difference between wideband and narrowband lowpass filter
hello
I came across the word wideband and narrowband while designing a simple
low pass filter. what does these words imply?
when we know that the filter we have to design is lowpass filter, then
what difference does it make whether we mention it as narrowband or
wideband..
plz reply
thanx
|
9/3/2010 6:18:27 PM
|
5
|
"enricophpdsp" <enricophp...@n_o_s_p_a_m.yahoo.com>
|
Question about Continuous Phase FSK
I am trying to understand what is meant by continuous phase FSK.
Right now I am of the opinion that it means that a very quick change
in frequency can take place as long as there is no discontinuity in
the time waveform when the frequency change takes place. Is this a
correct interpretation?
Thanks,
Brent
|
9/3/2010 12:59:44 PM
|
7
|
brent <buleg...@columbus.rr.com>
|
BER performance with non-costant group delay filter
Hi,
I'm trying to set up a Matlab simulation in order to see the effect of real
analog base band filter on my receiver.
If I use a FIR filter, I generally remove some samples equal to the filter
group delay before sending the received vector to the demodulator.
If I model a real filter, the group delay is not constant over frequency (
it can vary over the channel bandwidth too) and moreover it is not an
integer number.
How should I trim the filter output?
Thanks
Alberto
|
9/3/2010 12:02:13 PM
|
2
|
"alberto.fuggetta" <alberto.fugge...@n_o_s_p_a_m.gmail.com>
|
Windowed sinc
When making a filter for a time domain interpolation of a signal using
a windowed sinc kernel, what should be the right way of aligning the
window towards sinc?
I can think of three possibilities:
1) The center of the window could be set at 0.
2) The center of the window could be set to the value of the time
domain shift of the interpolator.
3) The center if the window could be aligned to the "center of mass"
of the set of sinc coefficients.
What is the best option ?
Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com
|
9/3/2010 2:55:21 AM
|
32
|
Vladimir Vassilevsky <nos...@nowhere.com>
|
cygwin and fftw3
Hi all
Tried to link fftw3 in cygwin, but keep on getting error messages such as:
fftw3_prb.o:fftw3_prb.c.text+0x121): undefined reference to `_fftw_malloc'
As an alternative, I just used an example from the web, here
http://people.sc.fsu.edu/~jburkardt/...w3/fftw3_prb.c
compile is using
$ gcc -lfftw3 fft -o fftw3_prb fft-lib.o fftw3_prb.o
... and link it using
$ gcc -lfftw3 fftw3_prb.o -o fftw3_prb
thereby getting these error messages:
fftw3_prb.o:fftw3_prb.c.text+0x1e4): undefined reference to `_fftw_malloc'
Anybody any idea?
fftw3 is installed out of the box
|
9/1/2010 10:42:25 PM
|
3
|
"MacErich" <e.p.zwys...@n_o_s_p_a_m.sms.ed.ac.uk>
|
Dream SAM3716, 24bit (data) x 96 bit (coeff.) info
Hi,
I need a documentation of this DSP chip - could you help me ? :-)
Google couldn't. :-(
Reagards
Roman Rumian
|
8/31/2010 8:12:04 PM
|
2
|
Roman Rumian <rumian.wytnij.@agh.edu.pl>
|
UHPI
Hi,
I am using UHPI for my current ASIC based design. I am facing an issue
for the hstrobe signal generation. hstrobe is generated based on the top
level pin hds1,hds2 and hcs. I am not sure when i have to toggle the hds1
and hds2 signal. I am looking forward some help. Please let me know if some
one has any input on this.
Thanks.
|
8/31/2010 12:41:18 PM
|
0
|
"kramar" <kandavel....@n_o_s_p_a_m.gmail.com>
|
Help needed on F2812
Hi all,
I am very new to ezdsp F2812 .can any one tell me how to use it for my task
to control a sync motor by PWM.
When i connect it to the ccs it gives me the error
Error connecting to the target:
Error 0x80000200/-2082
Fatal Error during: OCS,
Device driver: Emulation Connection Loss Detected on Target CPU.
It is recommended to RESET EMULATOR. This will disconnect each
target from the emulator. The targets should then be power cycled
or hard reset followed by an emureset and reconnect to each target.
Sequence ID: 0
Error Code: -2082
Error Class: 0x80000200
I/O Port
|
8/31/2010 12:41:40 PM
|
1
|
"Atiq" <master.mind....@n_o_s_p_a_m.hotmail.com>
|
dsk_audio4 on tms320c5510 noise cancellation
Hi,
I'm trying to implement lms algorithm to output an 180 degree out of phase
noise signal. I've modify the sample given from dsk_audio4 as shown
below:
void HWI_RINT0()
{ Uint32 antinoise //other variable has been declare as global
Uint32
mySample1 = MCBSP_read32(hMcbsp0);
mySample2 = MCBSP_read32(hMcsp1);
noise_x [1] = noise_x [0];
noise_x [0] = my Sample1; //noise
d = mySample2; // signal
// lms calculation
y = y + w[0]*circular_x[0] + w[1]*circular_x[1];
e = d -y;
w[0] = w[0] + 0.1*e*circular_x[0];
w[1] = w[1] + 0.1*e*circulat_x[1];
//outpu
|
8/31/2010 12:40:42 PM
|
1
|
"jasonyau" <jasonya...@n_o_s_p_a_m.hotmail.com>
|
Direct form II transposed implementation
Hi.
I'd like to implement 2nd or 3rd order IIR low-pass filter on C6747. For
the 3rd order version I get following coefficients from MATLAB:
>> [B,A] = ellip(3,1,60,10/83)
B =
0.005139381603929 0.006187906776077 0.006187906776077 0.005139381603929
A =
1.000000000000000 -2.530901747249838 2.242763702534123 -0.689207378524273
I know how to implement this in direct form I and II. But I'd like to
implement it in direct form II transposed because I will have to do it in
fixed-point later on (16 bit) and on ASIC. I have some troubles getting it
right on paper.
I have somethin
|
8/31/2010 8:23:55 AM
|
5
|
"ombz" <andreas.weisk...@n_o_s_p_a_m.gmail.com>
|
truncation
Hi,
I'm doing some calculation and I'm confused on truncation. For
example, I casted two float number to fixed number, say 16 bit, and
multiply them to get the product, i.e. 32 bit. then I add the product
with another number (16 bit) to get the sum. Finally I need to
truncate the sum to 16 bit. I'm confused here. How can I truncate the
sum, from (MSB to MSB-15) or from (LSB+15 to LSB), or from middle to
get the final number? My colleagues can't agree with each other and
I'm really confused because I have see different truncation way in
other guy's code. Can anybody give me some adv
|
8/31/2010 8:36:53 AM
|
1
|
skyworld <chenyong20...@gmail.com>
|
window functions
1)what does the parameters in the equation of hamming window function
signify.
Eq is
H(n)=0.54+0.46cos((2*pie*n)/N-1)
why the values as 0.54 n 0.46. and why not the higher derivatives of cos
are used.
2)Wen i plotted the frequency response of the above equation i matlab, the
main lobe should have its highest peak at 0db, but i an getting it at above
20db. Why is it so???
|
8/30/2010 5:31:53 PM
|
4
|
"student28" <ritu_phys...@n_o_s_p_a_m.yahoo.co.in>
|
power delay profiles/ofdma
hi,
i am given 2 power delay profiles and i calculate the frequency response on
each subchannel by multiplying the pdp with a random gaussian variable and
then taking the fft. however, the pdp corresponding to the vehicular models
always give higher values of this h^2. Is this normal?
the pdps for vehicular and pedestrian are given below:
PDPdB=[0 -1 -2 -3 -8 -17.2 -20.8]; % pedestrian
PDPdB=[0 -1.5 -1.4 -3.6 -0.6 -9.1 -7 -12 -16.9]; % vehicular
|
8/30/2010 2:38:10 PM
|
0
|
"leonardo232" <leosivri...@n_o_s_p_a_m.gmail.com>
|
IIR filter gain - where to do it?
Hi,
I have the current 3rd order IIR filter:
gain = 1789.111562
b = [1 3 3 1];
a = [1 -1.6009450356 0.9414772490 -0.1971027115];
I'm trying to implement this in fixed-point format, using DF1. I don't
think I need to convert to Second-Order-Section as this is not an
aggressive filter.
The first step was to quantize the coefficients, and I get get by with 12
bits.
Now I don't know what to do with this gain. If I do it at the input, I need
to add ~11 LSB to keep decent SNR. If I do it at the output, well I need to
add ~11 MSB or everything starts clipping within the filter.
|
8/30/2010 2:11:02 PM
|
14
|
"gretzteam" <gretzt...@n_o_s_p_a_m.yahoo.com>
|
General implementation of Butterworth filter questions
Having used up 5 days already, cutting and pasting the internet source code
for Butterworth filters, only to have the audio spectrum analyzer show a
terrible mess, I have to drop back to first principles in order to get
something to work correctly.
1. I have the general Butterworth normalized polynomials for order-n (or
can generate them) See http://en.wikipedia.org/wiki/Butterworth_filter
2. I am told to substitute for s, the value c*(z-1)/(z+1) for the bilateral
transform where c = cotangent(wc*T/2) where T is the sample time. See
http://www.apicsllc.com/apics/Sr_3/Sr_3.htm.
|
8/30/2010 1:07:09 AM
|
4
|
"Randall" <website.read...@n_o_s_p_a_m.gmail.com>
|
fftw3 - plan of different sizes
Hello,
I am a new user of fftw3 and there is something that I do not understand.
I will try to illustrate my question with an example.
First, I start in the physical space with a sinus wave y = sin(2*pi*x/L)
which is discretized over Nx samples. I compute this samples with fftw3 to
generate the spectral representation of y.
Now, I want to inverse the transformation to get y but for a different
number of samples Mx (). L is constant.
This works fine when Mx <= Nx but I cannot recover the sinus wave when Mx >
My. I do not understand why. I think the Nyquist theorem should be
resp
|
8/30/2010 1:07:15 AM
|
0
|
"manuge" <manugerma...@n_o_s_p_a_m.yahoo.ca>
|
should FFT{x-mean(x)} = FFT{x}-FFT{mean(x)} ?
Hi guys,
The Linearity property of Fourier transform says:
z(t)=af1(t)+bf2(t)
the linear combination of the terms is unaffected by the transform.
Z(ω)=aF1(ω)+bF2(ω)
and
Z(ω) = FFT{ z(t) }.
I synthesised a signal, sn as follows. I want to remove the mean values in
the signal. According to the Fourier transform property,
FFT{x-mean(x)} = FFT{x}-FFT{mean(x)}
the left hand side should equal to the right hand side. But my simulation
suggested they are different.
Can anyone sports any mistakes I've made?
Cheers,
---------------------------- Code Starts Here
|
8/28/2010 9:33:39 PM
|
3
|
"thedean515" <jiangwei0...@n_o_s_p_a_m.gmail.com>
|
should FFT{x-mean(x)} = FFT{x}-FFT{mean(x)} ?
Hi guys,
The Linearity property of Fourier transform says:
z(t)=af1(t)+bf2(t)
the linear combination of the terms is unaffected by the transform.
Z(ω)=aF1(ω)+bF2(ω)
and
Z(ω) = FFT{ z(t) }.
I synthesised a signal, sn as follows. I want to remove the mean values in
the signal. According to the Fourier transform property,
FFT{x-mean(x)} = FFT{x}-FFT{mean(x)}
the left hand side should equal to the right hand side. But my simulation
suggested they are different.
Can anyone sports any mistakes I've made?
Cheers,
---------------------------- Code Starts Here
|
8/28/2010 6:03:52 PM
|
5
|
"thedean515" <jiangwei0...@n_o_s_p_a_m.gmail.com>
|
Simple hack to get $5000 to your *Paypal account
Simple hack to get $5000 to your *Paypal account At http://youcanget.co.cc
i have hidden the Paypal Form link in an image. in that website on
Right Side below search box, click on image and enter your name and
Paypal ID.
|
8/28/2010 11:30:31 AM
|
0
|
paypal cash <m.bangara...@gmail.com>
|
Calculate Settling Time of a Transfer Function
I was wondering if anyone could explain the best method is to
calculate the settling time of a continuous transfer function when
given a unit step input?
The settling time, or time to steady-state, can be approximated, I
think (what I've tried to outline below), but I would like to know how
to calculate it exactly for any transfer function (1st order, 2nd
order, 3rd order, etc.).
Below is just some approximation testing, the above question is what
I'm interested in, below I'm just going through a little work to prove
the approximations are not what Matlab is returning for the sett
|
8/27/2010 10:35:01 PM
|
5
|
"John N." <ort...@gmail.com>
|
complex valued impulse response/asymmetric transfer function
hello forum,
given an input signal x(t)=cos(3t), its FT will be
X(w)=.5*[delta(w-3)+delta(w+3)]
If we have a filter that passes the positive part of the spectrum
(therefore the term delta(w-3) and completely blocks everything to the
left, w<0,
the output signal will be the complex valued exp(j*3t).
The filter that produced this result is asymmetric: |H(-w)| is different
from H|(w)|, i.e. the transfer function magnitude is not even and the phase
spectrum is not odd.
That implies that the impulse response is a complex value function....
Now the output y(t)=exp(i*3*t) and th
|
8/27/2010 10:05:32 PM
|
1
|
"fisico32" <marcoscipio...@n_o_s_p_a_m.gmail.com>
|
Damped Sinusoid Parameter extraction
Hello,
I am trying to find out robust a method method to extract pole-residue (or
damping factor + frequency ) of signal that consists of sum of damped sine
wave. So far we use Matrix Pencil Method (MPM). Is there anything else that
can produce superior result? What about ESPRIT method, is that usable and
better than MPM?
|
8/27/2010 9:37:15 PM
|
3
|
"rshahriar" <cshah...@n_o_s_p_a_m.vt.edu>
|
complex debugging
Hi,
I would like to pick the brains of the heavy weights around here on
their best proven debugging practices honed over years of experience.
In particular, how would one go about approaching this when dealing
with adaptive algorithms? The problem here is compounded by the fact
that the output is basically rubbish until convergence is declared.
I am mostly interested in hearing on *practical* methodologies as
opposed to anything too fancy. For instance, to move down from matlab
to c++, one could theoretically write mex functions and phase them in
progressively given the base mat
|
8/27/2010 4:43:33 PM
|
6
|
Manny <mlou...@hotmail.com>
|
Can you download and test my low pass filter?
Hello
I have developed a 50 Hz low pass filter.
Now I would understand if work good....
So please can you download and test it?
This is the link:
http://dl.dropbox.com/u/1992699/50%20Hz%20Low%20Pass%20Filter.zip
Note that you could add more than one waveform.
10 Hz --> Generate base waveform
100 Hz --> Add Noise
For measure Gain change waveform to 1 volt and filter....
Thanks
|
8/27/2010 9:08:54 AM
|
2
|
"gpezzella" <gpezze...@n_o_s_p_a_m.yahoo.com>
|
Off-Topic: Weak network signal. What to do?
I have been asked to help a small school with their wireless network
problem.
In some of the buildings on the school area the network signal is too weak.
I thought a simple repeater would solve the problem, but it didn't.
So do I have to make a network of repeaters?
Or buy a sensitive USB network adapter for each laptop?
Or install a big antenna on the roof of the main building...??
How is this problem typically solved without costs going through the roof?
I am looking for a cheap and simple solution which doesn't require cables.
Thank you.
|
8/27/2010 12:05:21 AM
|
3
|
"Jake" <j...@nospam.invalid.com>
|
displaying graph on CCS V3.3
Hi all,
Can any one please let me know how to use the graph option on CCS V3.3.
I am trying to display a constant value on the graph.
For ex
main()
|
8/26/2010 11:57:06 AM
|
0
|
"ramsurada" <ram.sur...@n_o_s_p_a_m.satcon.com>
|
What is best book for dsp implementation ?
Hi,
I already have a solid fundamental background about dsp topics.
So, What is best book for dsp implementation using TI DSP ?
Note: I am a professional embedded engineer.
BR,
SPHiNX
|
8/26/2010 11:57:18 AM
|
8
|
"SPHiNX" <eng.amrah...@n_o_s_p_a_m.gmail.com>
|
Determining feedback delay for adaptive digital predistortion
Hi,
I am implementing an adaptive digital predisortion technique in my
transmitter. What are the methods to determine the feedback delay from the
PA output to the predistorter input? Is this a constant delay or must it be
adaptively tracked? Please advise. Thanks.
|
8/26/2010 11:57:32 AM
|
0
|
"DerekDSP" <Derek...@n_o_s_p_a_m.gmail.com>
|
Question about a patent of Whitening Matched Filter
Hello All,
I am not sure whether someone here also read the patent about Whitening
Matched Filter (http://www.freepatentsonline.com/6862326.html). You can
just register the webpage and then download the patent in pdf format.
I am trying to understand the technique given in the patent, but so far I
failed in my matlab simulation.
The technique is to get a minimum phase filter by using cepstral transform,
and the filter is running by the Auto Regressive Moving Average (ARMA)
method.
My simulation follows the Figure 4 in the patent, but the result is wrong
in the sense that I am
|
8/26/2010 10:20:39 AM
|
0
|
"hardheart" <hardhear...@n_o_s_p_a_m.yahoo.com.cn>
|
Quickie Poll -- C vs. C++
So, I'm working on a spare-time project that's off the back burner for
at least a day. It's a trainer, and I'd _like_ to be able to set it up
so that advanced students can do their own programming. Hence, the poll.
So, since anyone who responds to a dippy poll on USENET is obviously
100% representative of the embedded software engineering public, I know
that your responses will accurately reflect reality.
If you answer this poll you will _not_ be entered into a contest to win
an iPod, or a free smokeless cigarette, or anything else. So do it for
the glory, and to advance t
|
8/25/2010 9:50:17 PM
|
64
|
Tim Wescott <...@seemywebsite.com>
|
Phase Unwarp - QPSK
Hello..
I want to do some QPSK facility to simple microcontroller. It's seems to
work in real life. I do the (not simplified) Goertzl algorithm and in the
end I just use lookup table to calculate arctan.
So for now I got phase signal in range pi/-pi (in my case -128 to 127 8 bit
output) and as I intend to use difference modulation I got just fine result
if I do BPSK. (ie just subtract last phase from previous one) It work's
well.
But after checking signal's I discover that phase error is so small that I
can do QPSK and double bitrate. Now I hit problem as simple difference of
l
|
8/24/2010 1:01:56 PM
|
2
|
"eslavko" <esla...@n_o_s_p_a_m.gmail.com>
|
how to find order of interpolator in IFIR technique?
hello
I want to calculate the order of interpolator G(z) in IFIR Technique. my L
=2 . I have inserted 1 zero between each impulse response of H(z). I
have
Ws,
Wp
deltaP-ripple value in passband
deltaS- peak ripple value in the stopband
N= -20 log10 (( delP * delS)^1/2) -13 (numerator)
14.6 (Ws-Wp)/2 pi (denominator)
by the above formula i can find order of Hm(z) ..
which formula I have to use for finding the order of G(Z)
please reply...
thanx in advance
|
8/24/2010 1:02:07 PM
|
1
|
"enricophpdsp" <enricophp...@n_o_s_p_a_m.yahoo.com>
|