Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the...

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the Nyquist rate (2 times the frequency of the baseband signal) but I believe that it is necessary to oversample in order to perform the Root Raised Cosine Filtering. If I am right, what is the method to oversample : - Interpolation (using an IDFT size 2 times ...

I need to simulate symbol recovery of a 16QAM or a higher constellation QAM. I am looking for an algorithm which I can simulate using c code. I...

I need to simulate symbol recovery of a 16QAM or a higher constellation QAM. I am looking for an algorithm which I can simulate using c code. I am using the baseband signal modulated using 16QAM. I have goe through few papers like Interpolation in digital Modem Part I/II. It doesnt seem to work well with QAM. I would also like to tell that I plan to use this recovery mechanism for DVB environmen...

Hello, Back in May, Steven G. Johnson posted an algorithm for performing a non-integer time shift in the frequency domain. The method he...

Hello, Back in May, Steven G. Johnson posted an algorithm for performing a non-integer time shift in the frequency domain. The method he posted was > The minimal-slope interpolation, which has the nice property of > producing a real signal from the transform of real inputs, is to > multiply the k-th Fourier coefficient by > exp(2*pi*i * s*k/N) for k < N - k, > by > exp(2*pi*i * s*(k-N)

Hi, I've read the Reference "IEEE Transaction on acoustics, and signal processing" about Cic filters for decimation and interpolation and I...

Hi, I've read the Reference "IEEE Transaction on acoustics, and signal processing" about Cic filters for decimation and interpolation and I found out something not clear to me. Reading the paragraph about the decimation filter, the reference explain that the number of bits to discard at each stage, in order to make the variance from the first 2N sources less than or equal to the variance for th...

Hi Folks Hope this isn't OT for this forum. I'm using zero-crossing analysis to extract instantaneous frequency from a signal recorded...

Hi Folks Hope this isn't OT for this forum. I'm using zero-crossing analysis to extract instantaneous frequency from a signal recorded from a frequency-division bat detector. The detector itself uses zero-crossing decimation to divide the frequency, so dominant frequency is the only information available in the signal. (The detector filters the output and re-applies the original amplit...

Hey guys, I am designing cic interpolation filter using system genartor.I have gone through xilinx website & try to understand about...

Hey guys, I am designing cic interpolation filter using system genartor.I have gone through xilinx website & try to understand about designing. But still i am confused that how can i implement QPSK modulator like an application of cic filter?can anyone suggest me? or how can i combined qpsk modulator & cic filter? Thanks

I need to analyze the magnitude spectra of some data sets. Every data vector will have 128 or fewer complex samples. I window all the...

I need to analyze the magnitude spectra of some data sets. Every data vector will have 128 or fewer complex samples. I window all the data using a Chebyshev window with a certain level of relative sidelobe attenuation. Normally, I zero pad the windowed data and take a 128-pt FFT. Now I'm experimenting with longer FFTs (256 and 512) in an attempt to get a higher resolution look at the spec...

Good evening everyone, unfortunaatly I didn't find any answer in google. Designing a digital audio synthesizer, how do I handle the fast change...

Good evening everyone, unfortunaatly I didn't find any answer in google. Designing a digital audio synthesizer, how do I handle the fast change of filter coefficients? For now I implemented say 4 coefficients sets for a lowpass filter, with (say) cutoff freqs of 5000, 3000, 1000, 500 hz. I can't generate and keep in memory 128 (MIDI.... ) different coefficient sets, but even if my sets are all ...

Hi John, I think you are on the right track, but may be able to improve the algorithm. You comment about the last odd point being an...

Hi John, I think you are on the right track, but may be able to improve the algorithm. You comment about the last odd point being an extrapolation is the key. It is not an extrapolation-- it is an interpolation. The DFT views the time domain signal as being circular, that is, point N-1 is positioned next to point 0, just as point 105 is positioned next to point 106. So this last odd point is...

Hi, I had to open another thread because this keeps bugging me. I have a confusion in the process of DAC. (Method-1) Couple of books...

Hi, I had to open another thread because this keeps bugging me. I have a confusion in the process of DAC. (Method-1) Couple of books (Bernard Widro/ Proakis) states that DAC process consists of sample-and-hold filter followed by a low-pass filter. The LPF smooths the sharp edges of S/H filter. (Method-2) But, the 'Whittaker?Shannon interpolation formula' (http://en.wikipedia.org/wi

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with bunch of interpolation and decimation filters and Sigma-Delta ADDA, and we have already adjusted all the filters so now it works just fine. However, we are still not 100% satisfied with the audio quality, we want to make the sound more clear so we are thin...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with bunch of interpolation and decimation filters and Sigma-Delta ADDA, and we have already adjusted all the filters so now it works just fine. However, we are still not 100% satisfied with the audio quality, we want to make the sound more clear so we are thin...

Hi Guys, I have a question for any of you who have studied Hogenauer's original paper on cascaded integrator-comb (CIC) digital filters. ...

Hi Guys, I have a question for any of you who have studied Hogenauer's original paper on cascaded integrator-comb (CIC) digital filters. E. Hogenauer, E. "An Economical Class of Digital Filters For Decimation and Interpolation," IEEE Trans. Acoust. Speech and Signal Proc., Vol. ASSP-29, April 1981, pp. 155-162. I'm having a disagreement with the a signal proce...

On Tue, 01 Apr 2008 07:46:50 -0600, jim wrote: > > > Eric Jacobsen wrote: > > > > On Mon, 31 Mar 2008 16:03:42 -0600, jim > > ...

On Tue, 01 Apr 2008 07:46:50 -0600, jim wrote: > > > Eric Jacobsen wrote: > > > > On Mon, 31 Mar 2008 16:03:42 -0600, jim > > wrote: > > > > > Jerry Avins wrote: > > > > > > > > It's all semantics, then. The alteration comes about not from the > > > > downsampling, but from the filtering that precedes it. > > > > > > > > > > When there is no

Hi all, I would like to model a channel when the symbol rate changes. Assume I'm using 3 tap channel h = [0.407 0.815 0.407], how can I get the...

Hi all, I would like to model a channel when the symbol rate changes. Assume I'm using 3 tap channel h = [0.407 0.815 0.407], how can I get the coefficients (6 taps) when I double the symbol rate, should interpolation give the channel coefficients? How can I get it from Matlab ? What happens if the symbol rate is increased by 1.33, how can I get the same channel response with 4

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a...

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a 5 times interpolation. Then, a low pass filtering to eliminate the aliasing frequency. Now, I have a low pass filter from 0 to 10 MHz pass band with a sampling rate of 40 MSPS. I want to get the same 0 to 10 MHz response (it is not a flat pass band...

Hi Guys, I've been trying to learn about those darned cascaded integrator-comb (CIC) filters and have reviewed Hogenaurer's original...

Hi Guys, I've been trying to learn about those darned cascaded integrator-comb (CIC) filters and have reviewed Hogenaurer's original paper Hogenauer, E. "An Economical Class of Digital Filters For Decimation and Interpolation," IEEE Trans. Acoust. Speech and Signal Proc., Vol. ASSP?29, pp. 155-162, April 1981. and Matt Donadio's CIC write-up (on www.dspguru.com). ...

Hi - I am trying to use Apple's AltiVec FFT routine fft2d_zrip in a scheme for doing 2D sinc interpolation by zero-padding the 2D DFT of an...

Hi - I am trying to use Apple's AltiVec FFT routine fft2d_zrip in a scheme for doing 2D sinc interpolation by zero-padding the 2D DFT of an image. I am able to do forward and reverse 2D FFT's and recover the original image, but when I zero pad I get clearly wrong results. The FFT routine in question makes use of what Apple calls a "split complex vector" format, which is a reordering...

I am trying to simulate a rational sample rate conversion architetcure using CIC filters (decimation) and farrow structure (interpolation)....

I am trying to simulate a rational sample rate conversion architetcure using CIC filters (decimation) and farrow structure (interpolation). Input data is QPSK modulated and upsampled by a large factor (325 to be exact). I try to decimate the signal by 125 in three stages. each stage is in turn a CIC (3-stage) decimation (by a factor of 5) filter. However the constellation i get after CIC filter ...

I've been studying an approach to implementing a lookup table (LUT) to implement a sine function. The two msbs of the phase define the...

I've been studying an approach to implementing a lookup table (LUT) to implement a sine function. The two msbs of the phase define the quadrant. I have decided that an 8 bit address for a single quadrant is sufficient with an 18 bit output. Another 11 bits of phase will give me sufficient resolution to interpolate the sin() to 18 bits. If you assume a straight line between the two en...