Techniques behind arbitrary response or "drawable" equalizers?

Started by Anonymous in comp.dsp5 years ago 150 replies

I need to design a filter that has arbitrary response. Basically, I want to be able to almost "draw" the equalizer curve. What are the...

I need to design a filter that has arbitrary response. Basically, I want to be able to almost "draw" the equalizer curve. What are the techniques behind (FFT-based) arbitrary response or "drawable" equalizers such as: http://wiki.cockos.com/wiki/index.php/ReaFIR or http://www.elevayta.net/product4.htm


Variable step size LMS

Started by Richard_K in comp.dsp15 years ago 2 replies

Currently I am trying to design an equalizer based on variable step size (VSS) NLMS algorithm. The adaptation of the step size is based on...

Currently I am trying to design an equalizer based on variable step size (VSS) NLMS algorithm. The adaptation of the step size is based on the instataneous squared error. The number of taps is 55. My initial step size (which is also equivalent to max. step size allowable) is 0.6 and the minimum step size allowable is 0.35, with the "Alpha" = 0.97, "Gamma" = 0.0005. I compare it to the conventional...


Seeking contractor to design some FIR or IIR filters

Started by NugenAudio in comp.dsp10 years ago 22 replies

Hi, We are seeking a contractor to provide the coefficients for a set of EQ filters with specific frequency responses, at given sample...

Hi, We are seeking a contractor to provide the coefficients for a set of EQ filters with specific frequency responses, at given sample rates. The filters we need the coefficients for are the frequency weighting equalizer for LEQ(m) as given by http://www.tasatrailers.org/TASAStandard.pdf section 1.4.2 at various standard audio sample rates (eg- 44.1kHz, 48kHz, 96kHz, 192kHz), and preferably ...


problem understanding LMS noise whitening alg.

Started by samwo123 in comp.dsp14 years ago

Hi. I am self-studying adaptive signal processing. I've read C.R. Johnson's Adaptive Signal Processing, and some parts of Haykin's. I am...

Hi. I am self-studying adaptive signal processing. I've read C.R. Johnson's Adaptive Signal Processing, and some parts of Haykin's. I am reading this paper "Self adaptive decision feedback equalization: Application to high order QAM signal" In the paper, adaptive whitening filter is applied to the received signal. The whitened signal is then fed to another adaptive FFE equalizer to recov...


Question about broadband equalizer in frequency selective fading channels

Started by fl in comp.dsp16 years ago 5 replies

Hello, I am doing some simulation about SC-FDE in a typical Hiperlan2 channel B environment. For each burst, the channel can be thought of as...

Hello, I am doing some simulation about SC-FDE in a typical Hiperlan2 channel B environment. For each burst, the channel can be thought of as static. I use channel coding similar to 802.11 standard and interested most in QAM64 modulation. I have a question about the simulation result. When the channel is a mild frequency selective fading one, channel coding is really to improve performanc...


GSM uplink burst freq recovery

Started by Pavel in comp.dsp12 years ago

Hello. I have a problem with uplink GSM burst processing because of unknown freq offset. With all other GSM receiving steps implemented(time...

Hello. I have a problem with uplink GSM burst processing because of unknown freq offset. With all other GSM receiving steps implemented(time sync, channel estimation, mlse equalizer) there is strong BER increasing due to frequency offset. Realization is in fixed-point machine so I try to avoid division and atan operation for carrier estimation. Who can give me some info about feedback methods or ...


MPEG Audio layer 2 for SHARC processors

Started by Anonymous in comp.dsp12 years ago 7 replies

Hello, http://www.autex.ru/dspa/dspa2008/09.pdf At the page 27 of this presentation have found: --- Available algorithms (factory...

Hello, http://www.autex.ru/dspa/dspa2008/09.pdf At the page 27 of this presentation have found: --- Available algorithms (factory mask-programmed) ‹ PCM ‹ Dolby Digital*, Dolby Digital EX2*, Dolby Pro Logic IIx* ‹ DTS 5.1*, DTS ES*, DTS Neo:6*, DTS 96/24* ‹ SRS Focus* ‹ MPEG2 AAC LC ‹ MP3 ‹ Graphic Equalizer, Balance/Fader, Bass Management, Delay Management Requirements: ‹...


How to decide the number of ADC bits needed for baseband receiver?

Started by Anonymous in comp.dsp15 years ago 6 replies

Hi, I am planning to build a BPSK/QPSK/8PSK/16QAM/32QAM/64QAM baseband simulator in matlab and will eventually add in equalizer and decoder for...

Hi, I am planning to build a BPSK/QPSK/8PSK/16QAM/32QAM/64QAM baseband simulator in matlab and will eventually add in equalizer and decoder for AWGN and fading channels. I have 2 question regarding the ADC requirements: 1)What is the minimum # of ADC bits needed to represent the IQ baseband signals to demodulate correctly? 2)What is the optimum AGC setting under AWGN /Fading environment? Wh...


LMS for co-channel Interference supression

Started by Mishra in comp.dsp11 years ago 7 replies

Hi There is a old topic called "LMS implementatiom for echo cancelation ad co-channel interference supression" which says we can use LMS to...

Hi There is a old topic called "LMS implementatiom for echo cancelation ad co-channel interference supression" which says we can use LMS to suppress co-channel interference. I agree adaptive Equalizer helps against co-channel interference. But I don't agree regarding NTSC interference in ATSC. I had done this experiment and couldn't meet the performance required. ATSC standard suggests to...


Symmetric boost and cut Equlizer using bandpass filters

Started by Anonymous in comp.dsp14 years ago 14 replies

Hi, I am trying to implement Symmetric boost and cut digital Equlizer using bandpass filters. I found this link which has some...

Hi, I am trying to implement Symmetric boost and cut digital Equlizer using bandpass filters. I found this link which has some help: http://72.14.235.104/search?q=cache:UIvZnXeDalgJ:www.freepatentsonline.com/5524022.html+symmetri c+response+bandpass+filters+for+graphic+equalizer&hl=en&ct=clnk&cd=1&gl=in In fact I did implement in Matlab. But for -ve gains, output is incorrect. (gain is


channel estimation in ofdm using preamble

Started by adi_sharma1984 in comp.dsp13 years ago 1 reply

Hi all, In ofdma the preamble is of 2 symbol long first one consist of 4 shorth and then long of some sequence. how do we use second symbol...

Hi all, In ofdma the preamble is of 2 symbol long first one consist of 4 shorth and then long of some sequence. how do we use second symbol preamble (or long preamble) to find the freq response of channel to train equalizer? currently when i do a cross correlation of a preamble received after channel with known sequence of preamble i don't see the frequency response of the channel am i doing...


Frequency /time domain equalization

Started by sair...@gmail.com in comp.dsp14 years ago 2 replies

Hello, I am doing simulation studies for frequency and time domain equalization. From the complexit point of view I completely agreed...

Hello, I am doing simulation studies for frequency and time domain equalization. From the complexit point of view I completely agreed to the fact that frequency domain is better in the number of multiplications/additions. Regarding the performance (BER ) point of view I can see there is a large 3-4 dB gap between time and domain equalization.i.e. time domain equalizer is perform...


Simulate shelf/peak biquad filters using low/high/bandpass filters?

Started by jungledmnc in comp.dsp10 years ago 8 replies

Hi, I'm using biquads from rbj's cookbook and trying to create a low-shelf filter by combining dry signal with lowpassed signal for a purpose...

Hi, I'm using biquads from rbj's cookbook and trying to create a low-shelf filter by combining dry signal with lowpassed signal for a purpose of a dynamic equalizer. Similar thing with the high-shelf via high-pass and peak via band-pass. It works quite well for amplification with peak filters: H_PEAK(z) = 1 + H_BANDPASS(z) * (g - 1) where g = 10^(gain_in_db/20) but I need to use 4x higher...


DMFX-1: Open Source HW/SW Audio Digital Multi Effects

Started by rezzonics in comp.dsp6 years ago 4 replies

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will...

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will include multitude of effects: Distorsion, Overdrive, Fuzz, Compressor, Equalizer, Phaser, Chorus, Delay, Reverberation, Flanger, Tremolo, Pitch Shift, Wah-wah ... HW is based on two TI TMS320C5535 DSP, one is used for digital audio and the other one as user ...


Question about the impulse response of a lead-lag compensator

Started by fl in comp.dsp8 years ago 1 reply

Hi, I am studying an equalizer which uses a lead-lag compensator after a high frequency attenuated distortion. At first, it seems quite simple:...

Hi, I am studying an equalizer which uses a lead-lag compensator after a high frequency attenuated distortion. At first, it seems quite simple: The equalizer emphasizes high frequency gain to make the total response flat at the high frequency part. When I try to get the total impulse response, I find that the impulse response of the lead-lag filter (nominator:[2.5e10, 1], denominat


SOVA soft-output for the coded bits

Started by psidemus in comp.dsp7 years ago 3 replies

Hi All, I have a question regarding convolutional code SOVA soft output. I'm implementing a SOVA decoder as part of a Turbo-Equalization...

Hi All, I have a question regarding convolutional code SOVA soft output. I'm implementing a SOVA decoder as part of a Turbo-Equalization effort. The standard SOVA algorithm provides soft-output per uncoded bit. However, In conjunction with a turbo-equalizer I have to produce a modified soft-output version, this time per coded bit (as opposed to the standard soft output per uncoded bit), ...