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GMSK Simulation

Started by united in comp.dsp19 years ago 1 reply

Hi there, I am currently working on a GMSK Simulation whereby i will need to simulate the raw binary bits -> NRZ -> Convolve NRZ with...

Hi there, I am currently working on a GMSK Simulation whereby i will need to simulate the raw binary bits -> NRZ -> Convolve NRZ with Gaussian Pulse -> (I+Q)*modulator -> GMSK Modulation. However, i am stuck in the integration of the NRZ and convolution of NRZ and Gaussian Pulse. Please kindly advice. Regards, Wee


Image search

Started by KK in comp.dsp15 years ago 1 reply

I'm trying to implement the image search described here http://www.dspguide.com/ch24/6.htm. I have to check for existence of a target image....

I'm trying to implement the image search described here http://www.dspguide.com/ch24/6.htm. I have to check for existence of a target image. Upon convolution we will have some values, each value proportional to the match. There may one or more maximas depending on repetition of the target. so is the case If target does not exist in the source image.Although in the latter case, values will be...


state-space representation

Started by fisico32 in comp.dsp14 years ago 9 replies

hello forum, I know that a linear system can be represented in its state space representation. I am actually more used to the convolution...

hello forum, I know that a linear system can be represented in its state space representation. I am actually more used to the convolution integral, transfer function methods... What is the advantage of describing the dynamics of a system in state space representation? I know that control people used that a lot. Does a state space description always involve a feedback? What is exactly the...


convolving noise with noise will get Gaussian? what does that imply?

Started by kiki in comp.dsp19 years ago 10 replies

Suppose I have a segment of data, which is basically random numbers between 0 and 1. I call it "f". Doing "f" -- the noise -- self...

Suppose I have a segment of data, which is basically random numbers between 0 and 1. I call it "f". Doing "f" -- the noise -- self convolution several times, the resultant data, if plotted, is of Gaussian shape. I know this is somewhat related to CLT... But what does this mean in practice? What does this imply? Does this CLT fact has any implication or application in practice? Any d...


convolution help

Started by Anonymous in comp.dsp19 years ago 1 reply

Hi, I'm using the following loop to convolute the coefficiants of a Kaiser band-pass (200Hz-800Hz passband, 100dB attenuation, 200Hz transition...

Hi, I'm using the following loop to convolute the coefficiants of a Kaiser band-pass (200Hz-800Hz passband, 100dB attenuation, 200Hz transition band). The results are rather good, but it seems that the high frequencies are not getting attenuated as well as the low frequencies. Any ideas as to why the high frequencies are not getting attenuated? Thanks. my coefficiants (0-258): double a0...


how to normalize complex filter?

Started by Andreas Besting in comp.dsp18 years ago 5 replies

Hi! I'm trying to implement a wavelet transform via convolution and my "filter" is a complex wavelet. It works fine but the resulting...

Hi! I'm trying to implement a wavelet transform via convolution and my "filter" is a complex wavelet. It works fine but the resulting amplitude seems to be wrong... I think I have to normalize the wavelet somehow, but i can't figure out how. I understand that i can normalize a real filter (e.g. sinc) by summing up the samples and devide all samples by this number. I tried to do the s...


when should one perform fft based filtering?

Started by bharat pathak in comp.dsp14 years ago 52 replies

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain...

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain filtering to FFT based filtering? bharat


when should one perform fft based filtering?

Started by bharat pathak in comp.dsp14 years ago 2 replies

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain...

what are the problems associated with FFT based filtering? and when should one decide to switchover from convolution based time domain filtering to FFT based filtering? bharat


Newbie: Help with acoustic signature detector

Started by Leonardo Capossio in comp.dsp9 years ago 3 replies

Hello, I am a newbie in DSP, but have received courses in LTI systems, FIR and IIR filters, convolution, and have extensive knowledge on FPGA...

Hello, I am a newbie in DSP, but have received courses in LTI systems, FIR and IIR filters, convolution, and have extensive knowledge on FPGA architectures and development (mostly non-DSP stuff, in DSP only FIR/IIR filters). I would like to make a system that detects audio-frequency signatures, but lets say mostly periodic signals (let's say i would like to detect up to the 2nd harm


Organising the archives of Comp.DSP

Started by stephaneb in comp.dsp19 years ago 6 replies

I am about to create a system to categorize the archives of Comp.DSP on DSPRelated.com and am currently trying to come up with an initial...

I am about to create a system to categorize the archives of Comp.DSP on DSPRelated.com and am currently trying to come up with an initial category structure. Here is my first draft: - Theory -> Adaptive Signal Processing - Theory -> Convolution - Theory -> Correlation - Theory -> Filters - Theory -> Fourier Transforms - Theory -> Image Processing - Theory -> Sampling - Theory -> Signals


FFT windowing and deconvolution

Started by Arrigo Benedetti in comp.dsp20 years ago 7 replies

I would like to get some feedback on this idea. Multiplying a signal in the time domain by a window before computing its FFT is equivalent to...

I would like to get some feedback on this idea. Multiplying a signal in the time domain by a window before computing its FFT is equivalent to the convolution of the transform of the signal with the transform of the window. It seems therefore that one could "undo" the effect of a square window (i.e. FFT with no windowing at all) by applying a complex deconvolution algorithm in the frequency d...


Complex-Valued Signal Processing

Started by smart in comp.dsp15 years ago 13 replies

Hello everybody: Is there any reference (books, surveys) on complex-valued signal processing on 2D domain? Specifically, I want to find...

Hello everybody: Is there any reference (books, surveys) on complex-valued signal processing on 2D domain? Specifically, I want to find something introductory related to complex- valued convolution. I'm digging through google. Most of the reference I found are either dealing with classical signal processing, or rather fragmented to read. Can somebody help? Thanks. W


Frequency resolution of FFT after windowing

Started by spaz in comp.dsp18 years ago 11 replies

When I normally do a FFT, the frequency resolution = sampling frequency/number of samples. When I use windowing (like BH4) to prevent smearing,...

When I normally do a FFT, the frequency resolution = sampling frequency/number of samples. When I use windowing (like BH4) to prevent smearing, this should increase the frequency resolution of the FFT, because of the convolution in the frequency domain. I have seen tables which show the frequency resolution increases by a factor of 1.9 for a Blackman Harris window, so if the frequency resoluti...


convolution with FFT: same resut with or without zero-padding?

Started by Felipe G. Nievinski in comp.dsp15 years ago 6 replies

Hi. Given n = 6 xa = rand(n,1) xb = rand(n,1) If I do ya = fft(xa) yb = fft(xb) z = ya.*yb w = ifft(z) I get...

Hi. Given n = 6 xa = rand(n,1) xb = rand(n,1) If I do ya = fft(xa) yb = fft(xb) z = ya.*yb w = ifft(z) I get the result that I want (w); I'm interested in obtaining exactly the same result zero-padding input vectors to the next power of two, for faster FFT (my n is actually ~ 1e6 and I have several such input vectors); I tried doing n2 = 2^nextpow2(n) ...


vibroseis deconvolution without the target sweep

Started by Nicholas Kinar in comp.dsp18 years ago 2 replies

I have a vibroseis dataset from which I need to find the reflection response of the earth medium r[t]. For a simple model, assume that s[t] is...

I have a vibroseis dataset from which I need to find the reflection response of the earth medium r[t]. For a simple model, assume that s[t] is the original target sweep and s'[t] is the recorded sweep expressible by the convolution s'[t] = s[t] * r[t]. Of course, this does not take into account earth attenuation or noise. What I need to find is the reflection response r[t], but I cann...


Filter Question

Started by Srikanth in comp.dsp18 years ago 15 replies

Hi I have a really basic question regarding digital filters.. mite seem dumb, but I couldn't figure it out. Anyway, here it is: When we...

Hi I have a really basic question regarding digital filters.. mite seem dumb, but I couldn't figure it out. Anyway, here it is: When we make a digital filter, we sample the time domain (for digital) and in frequency domain, we get a convolution with a impulse train. So, suppose I make a lowpass digital filter. If i have my sampling frequency as F, and I take the gain at 0Hz as 0dB, and s...


Fourier transform of switched cos wave

Started by Chris in comp.dsp20 years ago 5 replies

Hi, i am having trouble integrating a switched cosine wave, which is 'on' over -tau/2 to tau/2, and 0 at all other points in time. In a book i...

Hi, i am having trouble integrating a switched cosine wave, which is 'on' over -tau/2 to tau/2, and 0 at all other points in time. In a book i have it explains the process as convolution which is ok. However i believe i should also be able to work out the fourier transform using the following: /tau/2 | | cos(w0.t).exp(-j.w.t) dt | / -tau/2 and using the exponential form of cos...


Are there any DSP benefits of choosing a random seed for an MLS?

Started by Nicholas Kinar in comp.dsp15 years ago 17 replies

To determine the impulse response of a room, a Maximum Length Sequence (MLS) is produced by a loudspeaker. The room is considered to be a LTI...

To determine the impulse response of a room, a Maximum Length Sequence (MLS) is produced by a loudspeaker. The room is considered to be a LTI system. A microphone is used to capture the output of this linear system. A cross-correlation (or in some texts, a "convolution operation") is then used to determine the impulse response of the room, given that we know the original MLS. In...


MB-OFDM-UWB

Started by totohaha in comp.dsp16 years ago 1 reply

I want to simulate a MB-OFDM-UWB system in Matlab. I did pure OFDM system before which consist of convolution code block, modulation block,...

I want to simulate a MB-OFDM-UWB system in Matlab. I did pure OFDM system before which consist of convolution code block, modulation block, FFT block, Cyclic Prefix block, multipath block and corresponding receiver blocks. However, how to modify them to a MB-OFDM-UWB system, what kind of block I need to add into? thank!


[OT]Apple to embrace Intel chips -- Vector Processing??

Started by Himanshu in comp.dsp19 years ago 13 replies

Hello! I read somewhere on the net that Apple is moving towards embracing Intel chips. What about PowerPC's Vector processing power they...

Hello! I read somewhere on the net that Apple is moving towards embracing Intel chips. What about PowerPC's Vector processing power they boasted about earlier? Isn't PowerPC powerful than its contemporary chips? Apple provices benchmarks for its Altivec extensions. The benchmark uses FFT performance and convolution. Weren't they _Serious_ about this? Any comments? Regards --Himanshu...