Suppose I have some sampled signal data in a ROM on a microcontroller or something, and I'm reading it out to a DAC. Let's say that for some...

Suppose I have some sampled signal data in a ROM on a microcontroller or something, and I'm reading it out to a DAC. Let's say that for some reason, I'd like to begin reading this data at some random position, go for a while, and then stop at a random stopping point and jump to another random position in the ROM and continue reading, and so on, to obtain pseudorandom variations on the w...

I asked this before under another subject line, but perhaps it got lost. Assume I want the autocorrelation peak or best least squares fit...

I asked this before under another subject line, but perhaps it got lost. Assume I want the autocorrelation peak or best least squares fit of sampled data for non-integer sample offsets (e.g. the periods of the frequencies of interest are not a multiple of the sample interval). Should I interpolate the data for the autocorrelation/ASDF? Or should I do the autocorrelation for integer sampl...

Hi, I have setup a simulation of an interpolation-based timing recovery loop with Gardner's 2 s/s error detector. The loop seems to work...

Hi, I have setup a simulation of an interpolation-based timing recovery loop with Gardner's 2 s/s error detector. The loop seems to work fine, but I am stuck with this question: Out of the two samples produced from the interpolator, how do we know which one is the correct one ? Basically, the TED, which is given by: e = (y(n) - y(n-2))*y(n-1) would produce zero error on both the sam...

Hello Assume we want to implement an optimal photographic image interpolation algorithm. Using a recommended windowed sinc algorithm (e.g.,...

Hello Assume we want to implement an optimal photographic image interpolation algorithm. Using a recommended windowed sinc algorithm (e.g., Lanczos) would require evaluation of the sinc function (sin(x)/x) at variable arguments. Now I have an alternative algorithm which only requires evaluation of cosine values and multiplication, no variable division. Would this be advantageous on DSP a...

I need to write a mixer of different sample-rate input and need resample function. I implement interpolation filter and decimation filter...

I need to write a mixer of different sample-rate input and need resample function. I implement interpolation filter and decimation filter by polyphase decomposition. However, it takes large computation time when convert 11KHz -> 48KHz, or 44KHz -> 48KHz. I saw using a windowed sinc interpolator can do this. Is anybody explained the algorithm ? Or any fast algorithm to do resample except l

Using Matlab, I need some way to automatically find the point where an arbitrary filter response crosses a threshold. For example, given a...

Using Matlab, I need some way to automatically find the point where an arbitrary filter response crosses a threshold. For example, given a low pass filter, use freqz to find the response, then find the frequency that where the magnitude most closely corresponds to -3dB. If the response never exactly hits -3dB, interpolation might be nice, but not really necessary as I can generate enough po...

Linear estimation theory is normally credited to Wiener around 1949 though he did have a classified report in the war years. Also, Kolmogorov...

Linear estimation theory is normally credited to Wiener around 1949 though he did have a classified report in the war years. Also, Kolmogorov published the discrete-time version.... Andrei N. Kolmogorov. Interpolation and extrapolation of stationary random sequences (in Russian). Izvestiya AN SSSR. Mathematics series, 5:314, 1941 in 1941. Wiener did continuous time. So why talk of Wiener f...

Hi all I need to use six-points cubic spline interpolator on non-uniformly sampled input data. As matlab's spline function works with 4 points,...

Hi all I need to use six-points cubic spline interpolator on non-uniformly sampled input data. As matlab's spline function works with 4 points, I have to code and implement it myself. I shall be grateful if someone could point me out some references which detail the theory of the said thing. Once I know it, I hope I would be able to code without much hassle. Though I wouldn't mind using if s...

Hi, When looking at many of the video scalers and DSPs for consumer video, the datasheets usually list several cascaded polyphase FIR filters. ...

Hi, When looking at many of the video scalers and DSPs for consumer video, the datasheets usually list several cascaded polyphase FIR filters. They usually supply software libraries that generate the filter coefficients given a pair of numbers describing how much interpolation and decimation. In attempting to get a much better understanding of basic DSP skills, I started to build my own mod...

Hi, I'm researching ways to interpolate the sampled spectrum of a signal as given by a 512 point FFT. I typically only use the output of...

Hi, I'm researching ways to interpolate the sampled spectrum of a signal as given by a 512 point FFT. I typically only use the output of the FFT as is, but I occasionally need to resample the continuous spectrum in a non-uniform manner. I can't take a longer FFT to interpolate by zero-padding in the time domain, and usually only a few new samples of the spectrum are needed anyway. I fou...

There are a number of techniques available for estimating the frequency of a off-center sinusoid from the FFT bins around the one with peak...

There are a number of techniques available for estimating the frequency of a off-center sinusoid from the FFT bins around the one with peak magnitude. Can we also estimate the phase of the off-center sinusoid using similar methods? Would a legal approach be to find the frequency using the well-known methods and then interpolate the complex FFT data to get a complex response at the estimated f...

Hey, I'm implementing a flanger effect as part of a guitar effects processor on an ADSP-21364 development board. A Flanger effect is produced...

Hey, I'm implementing a flanger effect as part of a guitar effects processor on an ADSP-21364 development board. A Flanger effect is produced by adding a single delay to the current input sample, the length of the delay is dictated by a sin wave which i have in aa lookup table. This implements a comb filter. My problem is that on top of the desired output I am also getting a 'police si...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a sine wave (+1 to -1) of 24 bit and convolving with filter taps of 1.31 format. 2. when i save my result after filtering to 24 bit, i need to truncate the result to 24 bit and save it 3. in the above process i found the output sine wave having a ...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a sine wave (+1 to -1) of 24 bit and convolving with filter taps of 1.31 format. 2. when i save my result after filtering to 24 bit, i need to truncate the result to 24 bit and save it 3. in the above process i found the output sine wave having a ...

Hello Group! I read that while interpolaton for upsampling we pad zeros (their number decided by the factor of upsampling) and then the...

Hello Group! I read that while interpolaton for upsampling we pad zeros (their number decided by the factor of upsampling) and then the resultant is low-passed. Its like that the data is being compressed (in time) in the time domain and the corresponding frequency domain is expanding, "pushing" the frequencies into the adjacent periods. But when we pad a period with zeros (all zeros in t...

I want to simulate a circuit of timing recovery in a QPSK digital demodulation system . In the receiver,the upsampled signals are...

I want to simulate a circuit of timing recovery in a QPSK digital demodulation system . In the receiver,the upsampled signals are match filtered and tranmitted to a interpolation filter ,then followed a TED and loop filter. In order to determine the parameters of the loop filter,the error detector sensitivity Kd should be known first,Kd is the slope of the detector around zero error.Kd i...

Hello. Who knows how matched filter can be realized if sample rate is not integer multiple of symbol rate? In book of Meyr "Digital...

Hello. Who knows how matched filter can be realized if sample rate is not integer multiple of symbol rate? In book of Meyr "Digital Communications Recievers" matched filter is placed before controlled interpolation. What is the impulse response of fiter matched for example with rectangular transmit filter or raised cosine filter in that case? In Matlab I can calculate filter impulse respon...

Hi, I have a motion control system comprising a realtime component and a non-realtime component. Simple 2D data (position/time) is being...

Hi, I have a motion control system comprising a realtime component and a non-realtime component. Simple 2D data (position/time) is being streamed from the non-realtime area into a FIFO queue in the realtime area. I have a few things that I'm hoping someone could help me with: The realtime component needs to keep running at all cost. Thus, when data is delayed, my buffer is in danger of runni...

So I want to use a Time to Digital Converter(TDC) which is used in Time of Flight Measurements. It normally uses delayed clock samples to achieve...

So I want to use a Time to Digital Converter(TDC) which is used in Time of Flight Measurements. It normally uses delayed clock samples to achieve a resolution down to pico second range. I'm not really clear about the process. So is it possible to implement a TDC in discrete time domain? I was thinking of interpolation and low-passing to increase sampling rate. Simply stated, the problem is t...

Guys (Gals:), I've changed the domain of my website containing some article on DSP: http://www.strauss-acoustics.ch/ (previously...

Guys (Gals:), I've changed the domain of my website containing some article on DSP: http://www.strauss-acoustics.ch/ (previously www.philou.ch) contains : Pure javascript HTML5 canvas bilinear image interpolation Easy to read C Fast Fourier Transform (FFT) signal processing code FIR filter design experiment using the simplex method Zölzer-Boltze (ZB) peak/notch parametric EQ in