Adaptive equalization and synchronization

Started by Tim Wescott in comp.dsp5 years ago 31 replies

How does one maintain synchronization when one is using adaptive equalization? I'm currently working on a highly dispersive, varying channel,...

How does one maintain synchronization when one is using adaptive equalization? I'm currently working on a highly dispersive, varying channel, that needs adaptive equalization to work well. But I've come to realize that one thing that adaptive equalization adapts for is the exact timing of the signal. This means that you can't really used the output of the equalizer as a phase refere...


equalization of 'non-continuous' sample

Started by cl7teckie in comp.dsp12 years ago 2 replies

I am trying to build a 'non-continuous' 12-tap FFE (LMS Blind) equalizer in software, hoping to remove to some extent group delay in a satellite...

I am trying to build a 'non-continuous' 12-tap FFE (LMS Blind) equalizer in software, hoping to remove to some extent group delay in a satellite link. I am doing in software, as I am ONLY keen to know the SNR after equalization. The reading are in chunks of 512 x I/Q samples. Technically, does this even work, due to the fact that it is non-continuous. I do not need a perfect system, just some ...


about IIR Filter implementation...

Started by zeugnim in comp.dsp14 years ago 2 replies

hi, This is my first filter implementation and is made in Scilab, also the design of the 3ord Chebyshev Type I IIR Filter (with the iir...

hi, This is my first filter implementation and is made in Scilab, also the design of the 3ord Chebyshev Type I IIR Filter (with the iir function). I actually have 3 filters, so i can use them as an equalizer for an audio application. I´m using Direct Form I Realization. Since I have to implement this on a 24bit Fixed-Point DSP I had to scale my coefficients by a factor of 256 (same as shifting...


Re: Digital Equalizer Design for a Radio Receiver

Started by Anonymous in comp.dsp16 years ago

I was thinking of something like: http://www.w9gr.com/ "Anonymous" wrote in message news:... > > "Mark" wrote in message > ...

I was thinking of something like: http://www.w9gr.com/ "Anonymous" wrote in message news:... > > "Mark" wrote in message > news:1139417980.646987.93170@g44g2000cwa.googlegroups.com... > > Clark, > > > > The AF (adaptive filter) you are describing automatically adapts the > > overall bandwidth of the signal...OK...good... but does it actuall


The "look ahead" decision algorithm in DFE

Started by tmoshe in comp.dsp15 years ago

Hi, Im currently try to implement a DFE equalizer. When I read on this subject, I see that a special decision algorithm, called "look ahead"...

Hi, Im currently try to implement a DFE equalizer. When I read on this subject, I see that a special decision algorithm, called "look ahead" is performed in order to reduce the effect of error propagation. Unfortunately, I could not find any informative article about this issue. Can you explain it to me the principle of it? Does it a variation of viterby algorithm (a sequence detector)? Thanks...


The "look ahead" decision algorithm in DFE

Started by tmoshe in comp.dsp15 years ago

Hi, Im currently try to implement a DFE equalizer. When I read on this subject, I see that a special decision algorithm, called "look ahead"...

Hi, Im currently try to implement a DFE equalizer. When I read on this subject, I see that a special decision algorithm, called "look ahead" is performed in order to reduce the effect of error propagation. Unfortunately, I could not find any informative article about this issue. Can you explain it to me the principle of it? Does it a variation of viterby algorithm (a sequence detector)? Thanks...


filtering by convolution

Started by RMcPherson in comp.dsp12 years ago 1 reply

Hi all, I have some newbie questions for someone much more experienced than I. I am implementing a 6 band equalizer on the AVR32 processor,...

Hi all, I have some newbie questions for someone much more experienced than I. I am implementing a 6 band equalizer on the AVR32 processor, which has built in functions that do filtering by convolution of the input signal with the impulse response of filter. I have a couple of questions based on the AVR32 implementation of FIR filtering. In matlab I generated 6 bandpass filters and a hig...


Cable Equalizer

Started by Guy Eschemann in comp.dsp12 years ago 14 replies

I'm looking for some inspiration on the following problem: I want to transmit a signal consisting of 8 QPSK-Modulated carriers over a coaxial...

I'm looking for some inspiration on the following problem: I want to transmit a signal consisting of 8 QPSK-Modulated carriers over a coaxial cable, in the frequency range of 1 to 10 MHz. Since the cable has a frequency-dependent attenuation (e.g. 1dB/100m@1MHz, 3dB/ 100m@10 MHz), I need to somehow compensate for this at the receiver before it is forwarded to an optical transmitter. Sinc...


Peaking equalization - low phase delay

Started by FatScouser in comp.dsp11 years ago 17 replies

Hi all, I'm trying to implement a graphic equalizer tool (in c++) for a non-audio system with bandwidth 0 - 40 Hz, and hoped someone in the...

Hi all, I'm trying to implement a graphic equalizer tool (in c++) for a non-audio system with bandwidth 0 - 40 Hz, and hoped someone in the Comp.DSP community would be kind enough to offer helpful advice? We have a quasi-realtime system in which minimal phase delay is of critical importance, but minimal signal distortion is also a requirement (obviously). We want to keep delays below about 7...


SPDIF EQ

Started by Dirk Bruere at NeoPax in comp.dsp12 years ago 7 replies

Does anyone do a s/w 3rd octave equalizer that will work on a stereo (non dolby etc) SPDIF output (from a mobo or soundcard)? --...

Does anyone do a s/w 3rd octave equalizer that will work on a stereo (non dolby etc) SPDIF output (from a mobo or soundcard)? -- Dirk http://www.transcendence.me.uk/ - Transcendence UK http://www.theconsensus.org/ - A UK political party http://www.blogtalkradio.com/onetribe - Occult Talk Show


Filters coefficients in symbol spaced DFE

Started by alberto.fuggetta in comp.dsp11 years ago 9 replies

Hi, I'm trying to equalize a channel with sever multipath using a DFE (12,12) with LMS adaption algorithm. The relative power of the replicas...

Hi, I'm trying to equalize a channel with sever multipath using a DFE (12,12) with LMS adaption algorithm. The relative power of the replicas are quite high w.r.t the main path. (max -4 dB). The equalizer is catastrophic. From the learning curve analysis I can observe that the error is still high after processing the training sequence. Morover, the forward filter coefficients are very small ...


LMMSE estimation

Started by Hany in comp.dsp15 years ago 8 replies

Hello ALL, I'm trying to simulate a LMMSE equalizer for OFDM signal. The general equation requires the calculation of (X.X^H) where X...

Hello ALL, I'm trying to simulate a LMMSE equalizer for OFDM signal. The general equation requires the calculation of (X.X^H) where X is the transmitted signal OR > > -----> X is the diagonal matrix containing the transmitted signal..! And X^H is the hermitian of X (hermitian is the conjugate transpose of the matrix) I'm actually using the DL preamble to estimate the channel (as H


BPSK + OFDM over a very wideband channel

Started by lsw00kor in comp.dsp16 years ago 1 reply

Hello, everyone. I would like to discuss OFDM's robustnesss againt freq-selective channel. 1) I'm using BPSK( for some reasons, even though...

Hello, everyone. I would like to discuss OFDM's robustnesss againt freq-selective channel. 1) I'm using BPSK( for some reasons, even though mostly we use QPSK, or QAM..) on a very very wide channel. 2) I didn't employ a channel equalizer at the frontend before fft. 3) I, as a reference, followed Richard Van Nee's 'OFDM for wireless multimedia communications' to set the parameters: [1] wor...


LDPC code

Started by cpshah99 in comp.dsp12 years ago 10 replies

Hello Everybody Recently I simulated a regular rate 1/2 LDPC code over AGWN and I got the expected BER plot. Then I thought to simulate...

Hello Everybody Recently I simulated a regular rate 1/2 LDPC code over AGWN and I got the expected BER plot. Then I thought to simulate the same code over frequency selective channel. I am using linear MMSE equalizer and then LDPC decoding. So basically it is one time equalization and decoding. However, I am not getting any coding gain over uncoded system even after 100 LDPC iterations....


Channel tracking in an OFDM system

Started by ananth.rs in comp.dsp13 years ago 2 replies

Hello, I am trying to design an adaptive equalizer for an OFDM system which has an extremely low symbol rate. I have a question regarding...

Hello, I am trying to design an adaptive equalizer for an OFDM system which has an extremely low symbol rate. I have a question regarding this. The transmitter and the receiver are moving wrt each other. For what amount of relative motion between them in one OFDM symbol period (as a fraction of the carrier wavelength) can I assume that the channel is reasonably correlated across OFDM symbols...


Cascade All Pass Filters

Started by Dan Brateris in comp.dsp12 years ago 12 replies

Hello All, Im designing and audio equalizer in Simulink. I have 15 FDATool blocks for filters with dbGain blocks after them for the gains. My...

Hello All, Im designing and audio equalizer in Simulink. I have 15 FDATool blocks for filters with dbGain blocks after them for the gains. My filters are IIR filters designed using the butterworth design method. The problem is that in the passband of my filters the phase is not perfectly linear. Could someone help instruct me on how to use matlab to design an all pass filter to linearize...


ISI

Started by aamer in comp.dsp15 years ago 9 replies

Dear friends, As we all know inter symbol interference(ISI) is , by which the transmitted symbol smears or runs into the adjacent symbol....

Dear friends, As we all know inter symbol interference(ISI) is , by which the transmitted symbol smears or runs into the adjacent symbol. But, in one of the IEEE paper I have seen the mathemetical definition as ISI= sum(abs(h).^2)- max(abs(h).^2) / max(abs(h).^2) where h= conv(c,w) c = impulse response of channel w = equalizer taps. what I know is......to negotiate the effects of...


Constant modulue equalization for FSK signals

Started by john in comp.dsp16 years ago 1 reply

Hi group, I have a simulation in which a noisy binary CPFSK signal is filtered with a FIR filter representing the channel. Due to the channel,...

Hi group, I have a simulation in which a noisy binary CPFSK signal is filtered with a FIR filter representing the channel. Due to the channel, the BER curve shifts to the right. I'm trying to construct a blind equalizer for CPFSK using the Constant Modulus (CM) criterion operating on a block of data. Normally the CM criterion is applied in a recursive way to update a gradient descent al...


Gaussian assumption and Calculating the loglikelihood ratio

Started by cpshah99 in comp.dsp11 years ago 4 replies

Hi All On AWGN channel (y=x+n), the LLR, L_y, corresponding to the channel output can be written as L=2/sigma^2*y, where x is BPSK...

Hi All On AWGN channel (y=x+n), the LLR, L_y, corresponding to the channel output can be written as L=2/sigma^2*y, where x is BPSK modulated symbols. However, when the channel is frequency selective, and at the receiver end, a filter based equalizer (LE or DFE) is used to mitigate the ISI, do we still calculate the LLR the same way i.e. L=2/sigma^2*x_hat, where x_hat is the output o...


Convergence of Linear Equalizer

Started by cpshah99 in comp.dsp12 years ago 6 replies

Hi All In Haykin book of adaptive filter, he has explained tracking of LMS in non-stationary environment. In that he says that the tracking is...

Hi All In Haykin book of adaptive filter, he has explained tracking of LMS in non-stationary environment. In that he says that the tracking is problem specific i.e. tracking of system identification is different problem than equalizing. I get what he is saying. And then he derives all the math for system identification problem where he models the time varying environment as first order Marko...