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3
votes
2answers
95 views

Finding the frequency response of a filter defined as a Z-domain transfer function

I am software engineer studying DSP on my own. I came across this question in a graduate textbook and am not sure how to proceed. ...
4
votes
3answers
173 views

Mathematically inclined Signal and Systems/Signal Processing book?

I'm an electronics engineering student with high inclination to analysis and pure mathematics. I was just wondering if there was any book ( or any resource ) that treats signal and systems and signal ...
2
votes
1answer
60 views

OFDM training symbol format

For OFDM synchronisation, we are using two training symbols according to the Schmidl and Cox method of frequency synchronization and timing synchronization. The paper says: The first OFDM training ...
2
votes
0answers
47 views

Appropriate sampling frequency for bpsk modulated signal

I'm modulating a message with particular baudrate, say 2MHz using BPSK , Carrier frequency(Fc) is 8MHz, What should be appropriate Sampling Frequency(Fs) following Nyquist criteria, Should it be ...
0
votes
1answer
55 views

Need for Oversampling in OFDM and symbol time determination

I am trying to implement OFDM on gnuradio. I have two questions. The sampling frequency which is slightly higher than available bandwidth satisfies Nyqvist criterion for complex baseband signals. ...
1
vote
2answers
72 views

Filter order vs Number of taps vs Number of coefficients

I'm learning DSP slowly and trying to wrap my head around some terminology: Question 1: Suppose I have the following filter difference equation: ...
2
votes
1answer
37 views

Converting floating point number to Q-notation fixed point number in C/Java?

I am trying to convert from a floating point number to a Q-notation signed 16-bit fixed point number (e.g. Q15). How can this be done in C or Java? I searched online but could not find any available ...
2
votes
1answer
57 views

How to show that this filter is a HP filter

i got a system with transfer function given by: $$H(\omega)=1-e^{-j\omega}$$ I already plot it, and that seems to be a periodic function with $H(0)=0$, $H(\pi)=2$, , is that enough to show that this ...
2
votes
2answers
56 views

Q number format vs IEEE 754 format?

I'm reading up on DSP implementations and Q number format for representing decimal and decimal fractional numbers. Why would you use the Q format and not a standard like IEEE 754? When would you use ...
4
votes
0answers
93 views

How does Songify work?

Does anyone know how does the "Songify" app algorithm work? There are 2 inputs: speech (or amateur singing), and underlying music. Somehow the speech is altered and mixed into the music. Which ...
0
votes
1answer
68 views

How to resample in MATLAB one signal that is supposed to be convolve with another one?

i have two audio signals .wav and i want to convolve them, the trouble is that the frequency do not fit, they aren“t just the same frecuency-lenght in the wavread arguments. i am supposed to use the ...
1
vote
2answers
80 views

Decompose video frames into slices

I am working on a video motion feature implementation. I am implementing a work in a research paper. What I want to implement is this: The frames contained in a video clip can be thought of as a ...
2
votes
2answers
65 views

Do I need to worry about synchronization in a digital signal processor?

I understand that a digital signal processor consist of multiple MAD (muliply add units), which execute in parallel to implement a difference equation, like this one: $$ y[n]=10y[n-1] + 45y[n-2] + ...
0
votes
2answers
117 views

Low cost DSP development kit

I need a low cost DSP kit for learning purpose, along with development tools. One main requirement is to process a voice signal (8KHz sampling frequency), with a USB interface. Any suggestions? ...
2
votes
3answers
196 views

How should one select this filter coefficient for a pre-emphasis filter?

My professor asked me why you choose $a=0.97$ in a pre-emphasis filter $H(z)=1-\frac{a}{Z}$ for your pre-processing step. What is the relation between $a$ and the filter? How does this coefficient ...
5
votes
5answers
366 views

Dsp programming in C/C++

I have exposure to MATLAB and have some basic exposure to signal processing and plotting in MATLAB... but I am more comfortable with C/C++ than MATLAB because I have been working in Image processing ...
3
votes
1answer
82 views

Circular Time Delay in signal, effect on phase spectra of DFT

If you circularly shift a signal x[n]= [-3 -2 -1 0 1 2 3 2 1 0 -1 -2 ] to the right by M=1 compared to M=2, the phase spectra has a lot many zeros when M=2 compared to M=1. here, I am talking about ...
4
votes
1answer
224 views

Hexagonal Modulator Comparison with Square Modulator

Hexagonal shaped Modulator is designed to get better BER than Square Modulator. The constellation diagram is like the figure: It is supposed to give better BER measurement than Square as it has ...
0
votes
0answers
263 views

Implementation of goertzel algorithm in C

I am implementing BFSK frequency hopping communication system on a DSP processor. It was suggested by some of the forum members to use Goertzel algorithm for the demodulation of frequency hopping at ...
3
votes
0answers
339 views

Scipy - Audio Processing

I looking for good tool for Audio signal processing. e.g Speech & music analysis, identification. Does scipy provide functions for audio processing? Is this a good tool for Audio Signal ...
4
votes
2answers
185 views

WAV encoding problem

When processing wav files, I encounter this problem: the what I extract PCM from a 8-bit encoded .wav file, I got a sequence of integers. However, when verifying my implementation with ...
2
votes
1answer
432 views

List of all available ARM+DSP processors for surveillance

I have come across TI's C6000/Integra. I want to know is there any other similar or better ARM+DSPs available. Specially for surveillance/vehicle number plate recognition etc. Having camera input at ...
3
votes
2answers
3k views

interview questions for DSP engineer

I am preparing an interview which may require some knowledges on DSP. Is there any questions, fundamental concepts, and theories that I would not want miss? Thanks a lot for the suggestions.
-3
votes
1answer
173 views

How to do scaling in fixed format?

In my project, the ADC is an 16-bit unipolar type, 0 - 3.3 V. This 16-bit data we are making as bipolar ($-1.65$ to $+1.65$ range), converting as Q15 data. The conversion is as follows. ...
-4
votes
5answers
643 views

FIXED POINT conversion Problem (DSP)

Actually after reading the data from ADC is an $Q15$ format (ie. ADC supports 0-3.3v, ADC is an unipolar type). then the $Q15$ data is converted to $q31$ format ($Q31$ value = $q15_{data}$ ...
10
votes
3answers
333 views

How to decide what platform is best to implement real-time audio processing on?

When developing products I do all of my algorithm design in Matlab. They are usually pretty basic, an IIR or FIR filter or two, a couple FFTs, etc. When it comes time to move it to an embedded ...