design of adptive DFE

On Jun 10, 8:56&#4294967295;pm, dbd <d...@ieee.org> wrote:
> On Jun 10, 10:48 am, robert bristow-johnson
>
> <r...@audioimagination.com> wrote:
> > On Jun 10, 11:45 am, Richard Owlett <rowl...@atlascomm.net> wrote:
> > > rickman wrote:
> > ...
>
> .> > > I think your post would have been even more annoying to V if
> you had
> .> > > just left out the reference to his insult altogether. &#4294967295;Most
> trolls are
>
> > > > infuriated if, while he is standing right next to you, you continue
> > > > your conversation as if he wasn't even in the room.
> > > It may be even more effective to politely acknowlege V's existence and
> > > then carry on a polite conversation. IT might catch on - though I doubt
> > > he'll notice.
>
> ,> i don't even know what it is? &#4294967295;student with a "pi" stuck in it?
> ,>
> ,> curiopius,
>
>
>
> > r b-j
>
> Gee guys, are you sure that V-dude is trying to be insulting?
>
> I always thought "STUPIDENT" was simply an involuntary utterance. Sort
> of a self-deprecating keyboard based version of Tourette's syndrome
> that produces a response whenever his subconscious detects a posting
> that he lacks the knowledge, competence, communication skills, common
> decency and cool to reply adequately to, even though this doesn't
> always deter him from responding. It seemed less likely at first, but
> by now the evidence seems overwhelming.
>
> Dale B. Dalrymple

It also could be a vicious cycle of insolence he's caught up with.
Freud would trace this back to an early childhood abuse. It's like
mother_1 who's having a hard time disciplining children_1. Mother_1
not feeling in the mood, Boss is not having _s e x_ lately. Growing
ever grumpy, boss starts reprimanding father excessively at work.
Becoming polarized, father can't help but passing on his negative
energy to mother_2 who, in turn, passes it on to children_2.
Children_2 go about bullying other children on the block and as a
result children_1 end up getting some of their own medicine. On and on
the cycle goes. What remains is for us to know where in the chain he
truly lies.

-Manny

On Jun 10, 6:00&#4294967295;pm, Manny <mlou...@hotmail.com> wrote:
> What remains is for us to know where in the chain he
> truly lies.
>
> -Manny

Let's see....  Independent consulting company, lots of time on his
hands to constantly make snide remarks on Usenet.....  I think I can
connect the dots..... Not very many customers.

robert bristow-johnson wrote:


> 
> i don't even know what it is?  student with a "pi" stuck in it?
> 
> curiopius,

STUPIDENT is the same thing as the STUDIOT

//---------------------
Hi
I saw a lot of posting of yours on dsprelated.com,
I saw on your website a name of a software as MathLab. I want to know 
more about this.
I have seen only MATLAB and MathCad, what is this mathLab.
Please please Please reply sure
//-----------------
Hello
I need Matlab code for BPSK/PM modulation. Can you send me this code? 
Please i will be very grateful.
Thank you
//--------------------------
In my final year project , I need a convolutional encoder (rate = 1/2 
with constraints length = 5) and the viterbi decoder implemented in C 
for blackfin DSP(Analog devices)
Can you help me?
//-----------------
hello
i had few queries about..the concepts of non gaussian noises.
In communication channel..what are the models for impulsive noises and 
other noises.. so that we can arrive at a more practical simulation 
while making a communication channel on matlab
i read some work of Middleton which was relevant.
pls help me out with the appropriate model or the links..
thanking you
geetika
btech final yr student..working on simulating and comparison of 
modulation tech
//---------------------
I hope you don't mind me contacting you. I saw your name in a 
mathematics forum discussing DSP on sine waves. And so hope you may be 
able to help with a perhaps simple problem, but which escapes me.
I have a clean sine wave (A) with know frequency and amplitude.
I sampled a 2nd sine wave (B) at e.g. 4 random points. I know its 
frequency (same as A), however I wish to calculate the phase difference 
between the two sine waves(A)-(B).
//---------------------
I'd like to receice some advice about implementing an FM demodulator in 
a DSP proccessor (TMS320C6416T). My baseband frequency is 200Khz max on 
the input ,after downconverting from 88-108Mhz , and here are my issues.
//---------------------
I am doing a project on secure speech communication where i m trying
to find out a suitable encrytion algorithm which has got low
complexity as well high efficiency but before that i m trying to 
implement GSM 06.10 speech coder in MATLAB i ve implemented the coder 
but skipped a large no. of block this is where i got stuck up.
//---------------------
Hello Mr.s,
I'm a student of eletronics and I woul like implement a voice scrambler 
using the algorithm RSA on plataform DSP tms320c6711, I have watched 
that you done some algorithms to work in a DSP plataform, maybe you can 
help me with this,If dou you have any  RSA algorithms for DSP, I'll be 
very gratefully. Thanks sincerely
//-----------------------
I would like to ask you something. I'm implementing a GMSK demodulator 
based on Laurent decomposition. After using the matched filter, the 
baseband signal has to be sampled. Could you recommend me a suitable 
synchronization algorithm? I read about Gardner's method but I'm not 
sure about how it works. Is the interpolation needed if the signal has 
already 5 samples per bit?
//------------------------
I am a student at the Ohio State Univ, USA and presently for my project 
I am trying to implement the frequency domain LMS algo.
I have a few questions for which I seek help. They are as follows:
1) Do we have to start with a FD (frequency domain) weight vector that 
is all zeros (wont this give a zero output).
2) How do you convert the FD weight vector to time domain (My FD weight
vector makes sense (in matlab simuation), but in time domain
(IFFT(W(k+1)/N)it is a periodic vector (vector with periodic data elements).
3) How do you decide the convering factor 'mu' and or the converging
matrix. what are its typical values.
4) Can the algo. be applied to complex signals as it is?
//---------------------------
    I am [...] an M tech student in embedded systems from Kerala, India.
Right now I have completed my first year and now I have a full one year 
Project work. And I selected Active noise control as my topic. I have a 
good basic with digital signal processing. I would like to have ur 
suggestions on this topics.I hope u can give valuable suggestion that 
can help me to improve my work.
With prayers
//----------------------------

VLV

>Hi VLV
>
>Ok so this is what I did.
>
>I simulated a frequency selective channel with delay spread of 200
>symbols. And then for the adaptive DFE I used LMS algorithm with gear
>sshifting to update the taps and I selected the number of DFE taps (50
for
>FF and 50 for FB).
>
>So after conventional equalization, I used Turbo code and it gives good
>BER performance.
>
>So basically the point I was making is that how much load a Turbo
decoder
>can take. Because if the received signal was equalised properly then
>decoder would nt have to do much.

Interesting (i.e. weird) conclusion.

Vladimir is right. You should ignore channel coding when looking at
equalization issues. To maximise the effectiveness of channel coding you
want to leave the channel decoder with only one job to do - defeat noise.

If you don't see the Turbo decoder having much to do I assume you've only
simulated channel distortion, and haven't tried cranking up the noise
level. Channel coding is there to deal with noise, not channel distortion.
Channel decoders will, to some extent, do a similar job to an equalizer, if
you let them. However, this reduces their ability to deal with noise, which
is the whole reason for engaging in such complexity. You want the equalizer
to do the best it can in advance of the channel decoder, so you can
maximise the effectiveness of the channel coding.

Steve

>Interesting (i.e. weird) conclusion.
>
>Vladimir is right. You should ignore channel coding when looking at
>equalization issues. To maximise the effectiveness of channel coding you
>want to leave the channel decoder with only one job to do - defeat
noise.
>
>If you don't see the Turbo decoder having much to do I assume you've
only
>simulated channel distortion, and haven't tried cranking up the noise
>level. Channel coding is there to deal with noise, not channel
distortion.
>Channel decoders will, to some extent, do a similar job to an equalizer,
if
>you let them. However, this reduces their ability to deal with noise,
which
>is the whole reason for engaging in such complexity. You want the
equalizer
>to do the best it can in advance of the channel decoder, so you can
>maximise the effectiveness of the channel coding.
>
>Steve
>
>

%%%%

Hi Steve

The thing is that I thought in real time you will just design your
equalizer and leave it. And imagine situation comes where multipath is
longer than DFE so what will happen? I did simulated channel distortion
plus noise with whole range of SNR.

I dont knw where I am gng with this.

Chintan

On 11 Jun, 11:28, "cpshah99" <cpsha...@rediffmail.com> wrote:
> >Interesting (i.e. weird) conclusion.
>
> >Vladimir is right. You should ignore channel coding when looking at
> >equalization issues. To maximise the effectiveness of channel coding you
> >want to leave the channel decoder with only one job to do - defeat
> noise.

> I did simulated channel distortion
> plus noise with whole range of SNR.

The problem is that there is only so much you can simulate.
Easily or at all. What is easy to simulate has 'nice', 'simple'
mathemathical formulations. By coincidence, filters and
equalizers are also based on 'nice' and 'simple' mathemathics.
More often than not, these systems are particularly well
designed to handle the effects that are easily simulated.

So to really test your methods, you need to mess up the
simulations as much as possible. And then some. No matter
how bad your simulations are, the Real World will throw
something worse at the operational systems.

On Jun 11, 5:28&#4294967295;am, "cpshah99" <cpsha...@rediffmail.com> wrote:
>
> Hi Steve
>
> The thing is that I thought in real time you will just design your
> equalizer and leave it. And imagine situation comes where multipath is
> longer than DFE so what will happen? I did simulated channel distortion
> plus noise with whole range of SNR.
>
> I dont knw where I am gng with this.
>
> Chintan

Chintan, of course one can find cases for which one's system no longer
works.  That's why you need a good specification, hopefully based on
realistic models derived from good measurements and experiments.
You have alluded to "20 symbols long" etc etc.  What is it really?
Your
posts tend to be incomplete or in pieces, and I agree with Vladimir
and Steveu that designing for a very generic problem statement doesn't
make sense.

Even for standards development, typically there are a few channel
examples that the design is based off of.  Of course, there is an art
in
"choosing" which channel examples can be taken as the typical case,
worst case, and best case.

Is this for a course or academic project, or for an actual
implementation
target?  If it's the former, then I guess you can pick a statistical
and synthetic example.  For the latter, you need to look at the
problem
statement more carefully.

Julius

Hi Julius

I knew I will get hammered :-) but anyways.

This is for the actual implementation. It was my idea to simulate worse
channel, time varying and frequency selective. Perform one time
equalization and then see how much coding gain you can achieve.

Because there are many things to be considered. From the past experiments,
we had seen that suddenly one reflection will appear for a moment and then
disappear. The basic idea was to see if Turbo code can deal with this
residual ISI or not.

Thanks

Chintan

On Jun 11, 6:52&#4294967295;am, "cpshah99" <cpsha...@rediffmail.com> wrote:
> Hi Julius
>
> I knew I will get hammered :-) but anyways.
>
> This is for the actual implementation. It was my idea to simulate worse
> channel, time varying and frequency selective. Perform one time
> equalization and then see how much coding gain you can achieve.
>
> Because there are many things to be considered. From the past experiments,
> we had seen that suddenly one reflection will appear for a moment and then
> disappear. The basic idea was to see if Turbo code can deal with this
> residual ISI or not.
>
> Thanks
>
> Chintan

I think you asked the wrong question.  You should instead post about
observations from your experiments.  Your post comes across as
a homework problem that's either poorly written or poorly understood.
It assumes the correctness of many choices, and then asks how to
tweak them.  Of course you're going to get criticized for it.

So what were the observations from your experiments?  How did you
do the measurements?  What you should ask is how to model and/or
represent your channel, not whether solution A or B can be tweaked.

Julius

Hi Julius

So the observations are as follows:

System Parameters:

Delay spread of channel is 200 symbols, QPSK modultion, 511 long training
sequence, rate 1/2 turbo code, adaptive DFE with LMS algorithm. The FF taps
=50 and FB taps =50. The channel has 5 taps and taps are time varying.

So I fixed the SNR at 14 dB and after one time equalization and 3 turbo
iterations I am getting BER of 10^-5. Which I think is quite good.

Next I will be more specific when I ask any que.

Thanks as always

Chintan