These last few days using my program Photosounder ( http://photosounder.com ) I've been experimenting with a new sort of weird reverb-like...

These last few days using my program Photosounder ( http://photosounder.com ) I've been experimenting with a new sort of weird reverb-like effect on sounds (detailed with examples here http://photosounder.com/blog/2009/05/motion-blur-sound-reverberation.html ) and people who heard the result expressed their interest in me making a real-time VST effect out of it. The problem is that since it'...

Hey there everyone! I'm actually right now researching about ofdm based realization to efficiently estimate the doppler shift and I've kinda run...

Hey there everyone! I'm actually right now researching about ofdm based realization to efficiently estimate the doppler shift and I've kinda run into a block In the typical block diagram there's a IFFT/FFT block,I've read on many forums but I'm unable to determine the significance of the said block Also,if its possible could anyone help me understand why actually circular convolution is better ...

Hello Everyone, I am new to the site so bare with me. I am working on a college project involving convolution operations on complex 2D data....

Hello Everyone, I am new to the site so bare with me. I am working on a college project involving convolution operations on complex 2D data. The original code for this program was written in Matlab and the C/C++ code that I am developing has to match the Matlab Implementation. So far I have been able to replicate the same data in Matlab except for the output from the Matlab FFT. When I use the ...

I'm confused Say i have a signal model that looks like r(t) = s(t) * h(t) where * is convolution and h(t) is a filter. I then sample the...

I'm confused Say i have a signal model that looks like r(t) = s(t) * h(t) where * is convolution and h(t) is a filter. I then sample the signal by multiplication with a dirac p(t) = sum_n delta(t-nT), where delta is the dirac delta function. So according to Oppenhiem's book i have rp(t) = r(t)p(t) rp(t) = r(t) sum_n delta(t-nT) so substitute in for r(t) (because i want to expre...

In case of linear time invariant systems we can use convolution to represent the system or to find the system response. However if we have...

In case of linear time invariant systems we can use convolution to represent the system or to find the system response. However if we have a nonlinear system how can we find the system response or the transfer function of the system? any ideas? This message was sent using the Comp.DSP web interface on www.DSPRelated.com

Hi, I was considering using a Number Theoretic Transform to speed up a convolution on fixed point values. It fixes the problem of rounding...

Hi, I was considering using a Number Theoretic Transform to speed up a convolution on fixed point values. It fixes the problem of rounding errors, although overflows have to be carefully handeled. It also reduces complexity since integers multiplications modulus p (p prime) are used instead of complex multiplications. If I understand correctly how NTTs work, you can transform a sequence...

Suppose I drive a LTI system G(z) with white noise unit variance. I can then using various means make an attempt at identifying G(z) (assume the...

Suppose I drive a LTI system G(z) with white noise unit variance. I can then using various means make an attempt at identifying G(z) (assume the white noise cannot be measured but only the output of G(z)). I then drive a similar system G(z) with coloured noise (say in real life speech). I could model this for some short period of time as the cascade (convolution) of G(z) with the colouring...

I am trying to determine the best solution for real time image/video processing in hardware... Does anyone have any thoughts as to which...

I am trying to determine the best solution for real time image/video processing in hardware... Does anyone have any thoughts as to which solution (FPGA or DSP uP) is better. I would like to run several (roughly 10) convolution filters as well as perform image histogram equalization operations on each frame of a gray 640x480 frame at a rate of about 15 fps. Any comments would be most appr...

Following somebody's advice, I'm making a topic to expose my problem without talking about possible solutions. But first of all I'd like to...

Following somebody's advice, I'm making a topic to expose my problem without talking about possible solutions. But first of all I'd like to restrict this topic to solutions on how to implement frequency shifting (that doesn't involve performing a DFT and shifting bins), I don't want to hear about performing a time-domain convolution with a kernel that changes through time, or doing some st...

Hello all DSP gurus, I am struggling with some (seismic) DSP theory and need some help! In seismic the convolutional model states that the...

Hello all DSP gurus, I am struggling with some (seismic) DSP theory and need some help! In seismic the convolutional model states that the observed signal as a function of time, st (seismic trace), equals the convolution between the seismic shot signal, s, and the earth response function, e: st=s*e. Given a known st and s I want to calculate the unknown e. We introduce a pulse shaping fi...

Im trying to get a feel for when Taking the FFT and doing frequency domain processing is more computationally efficient than doing...

Im trying to get a feel for when Taking the FFT and doing frequency domain processing is more computationally efficient than doing discrete convolution. Does anyone have any specific examples? I understand that very long FIR filtering may have a beneifit if filter is done via FFT then IFFT. Is most real time filtering done in the time domain?

Hi, I am trying to simulate the GSM physical layer and have a simple question regarding GMSK modulation: I am trying to implement the GMSK...

Hi, I am trying to simulate the GSM physical layer and have a simple question regarding GMSK modulation: I am trying to implement the GMSK modulator using the quadrature baseband method outlined in: http://www.emc.york.ac.uk/reports/linkpcp/appD.pdf which states : NRZ sequence -> Gaussian filter convolution -> Integration -> I/Q decomposition -> Modulation. However, i have checked severa

Hi, I am simulating a single carrier system in which signals are equalized in frequency domain. The concept is similar to OFDM, except both...

Hi, I am simulating a single carrier system in which signals are equalized in frequency domain. The concept is similar to OFDM, except both FFT and IFFT are reside in receiver side. Prefix is also used to deal with dispersive channel response. I have implemented the rooted raised cosine filter(Alpha=0.2) in both transmitter and receiver. A static dispersive fading channel is used in each ...

Hello there, I'm currently working on a pitch identification program in MATLAB. It uses the Sliding DFT so I need to apply windowing functions...

Hello there, I'm currently working on a pitch identification program in MATLAB. It uses the Sliding DFT so I need to apply windowing functions in the frequency domain. I understand this is done by convolution and have managed to find the kernel for the von Hann window: [-0.25 0.5 -0.25], but am struggling to find a concrete answer for other windows like the Hamming and Blackman window. Anyon...

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x:...

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x: transmitted signal; w: measurement noise). The goal is to find an FIR equalizer g such that xhat=g*y is close to x. Writing x/y as vectors, the relation can be written as y=Hx+w. The equalizer xhat=Gzf y with Gzf = (H^T H)^-1 H^T is called the Z...

Hi everybody, I am working on a C6711 DSP using CCS 2.21. The system has two daughter cards (microline) one with 4 ADs and one with 2 DAs and...

Hi everybody, I am working on a C6711 DSP using CCS 2.21. The system has two daughter cards (microline) one with 4 ADs and one with 2 DAs and 2 ADs. The input from the ADs is FIFO buffered, whereas the output from the DAs is not. That is why the DA output and the whole time domain convolution (FXLMS-algorithm) is interrupt triggered (one sample for each interrupt) on an external timing si...

HI, I want to know about error correcting capability of a convolution code i.e. how many errors can be corrected in a continuous input data...

HI, I want to know about error correcting capability of a convolution code i.e. how many errors can be corrected in a continuous input data pattern (e.g. a 1/2 rate code with constraint length of 7 has free distance of 10 and is said to correct 4 errors, whether these errors cab be burst type also and what will be the relation of this error correction to input data)? Thanks

hi list, Suppose I have some "complicated" time signal that I've performed FFT on to get F1 spectrum. And a second time signal that I've...

hi list, Suppose I have some "complicated" time signal that I've performed FFT on to get F1 spectrum. And a second time signal that I've taken a FFT onto to get F2 spectrum. Now, suppose I want to convolute the two signals. There is clearly a negative frequency(s). F1*F2 Should I convolve in the F direction of the sample rate: -0.5 -> 0 -> +0.5 ( "thru", not "to" the zero point

Hi, I was looking at the upper bound on BER for convolutional encoder. "High Rate Punctured Convolution Coded for Viterbi and...

Hi, I was looking at the upper bound on BER for convolutional encoder. "High Rate Punctured Convolution Coded for Viterbi and Sequential Decoding" by David Haccoun and Guy Begin talks about the BER plots for BPSK and QPSK with and without puncturing. I am not able to find anything which talks about BEr calculation for QAM. Can someone suggest reference? thanks in advance!

Hello, I need to implement very narrow pass band filter and someone told me that "Interpolated FIR" filter can reduce the computational...

Hello, I need to implement very narrow pass band filter and someone told me that "Interpolated FIR" filter can reduce the computational complexity a lot. So, I went online and found some papers on IFIR and it indeed looked promising. However, when I implemented IFIR in Matlab, it did not do much saving in my scenario. This is the spec of the baseband filter which will be used as a prototype o...