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0
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1answer
44 views

Band Limited Impulse Train Synthesis

I'm currently working on a piece of DSP/Synthesis software and I have run into the issue of generating band limited waveforms. I have been doing some research to try and find a good solution but I ...
0
votes
0answers
34 views

Realtime image-stitching using a CMOS camera array

I am very new to DSP so please excuse any brevity... I have to design a 4x4 CMOS camera array and stitch the images together for further processing... I have tried doing that with opencv but the ...
1
vote
1answer
47 views

How to prove energy is preserved in sampling using Parseval's Relation?

I have a continuous signal,xc(t) which is bandlimited to =pi/T , and x[n] = xc(nT) are samples of this signal. I need to show there is a Parseval's relation between the original signal and the sampled ...
0
votes
1answer
22 views

regarding plots in visual dsp ++ 5.0

For my source code witten in c ,in visual dsp++ 5.0 for a forward error correcting code on adsp 21062 sharc family platform. i want to plot a graph between different data sets. when i tried to use the ...
0
votes
1answer
31 views

regarding visual dsp ++

i've written a source code in visual dp ++ in c language. the platform is sharc processor adsp 21062.after the build is complete. while running the program it is displaying a message ...
1
vote
0answers
18 views

DSP implementation recommended bibliography (good books-material related to fixed/floating point-optimization) [closed]

I am doing my first steps in embedded systems and there is a big deal of fixed point code and of course DSP optimized code.I would like please if possible to recommend any book/material related to ...
0
votes
4answers
163 views

FFT - Effects of Windowing,Improvisation approaches on RealTime FFT Hardware

Fore-word :Kindly Observe this question has multiple updates take the latest update as question ! here below is my 256pt FFT on an 2Khz sinusoid signal,on a fixed point DSP tms320c5515 please ...
0
votes
1answer
72 views

How to evaluate expressions without explicitly computing a discrete fourier transform?

Hello DSP guru nation, I am trying to learn this part but I dont seem to understand. if I have a finite length sequence, and a n-point DFT of it (in an interval)... is it possible to evaluate ...
0
votes
1answer
57 views

A question on filters

I am fairly new to DSP have been reading up on digital filters and have a few questions. I have a signal with very high frequencies whose DC component i want to estimate. The magnitudes of these ...
0
votes
2answers
59 views

Estimating the amplitude of a particular frequency

I am relatively new to DSP and have been reading a lot on the internet. I have a couple of questions. I have a signal in the form of a function $$ f(x) = A_0 + A_1 \cos(\omega_1 x) + A_2 ...
0
votes
1answer
53 views

What plotting / visualization tools are useful in generating good plots? [closed]

It has been said a picture is worth a thousand words! There have been a large number of softwares to write dsp codes and analyze signals and algorithms. It will be good to share on data plotting ...
0
votes
1answer
44 views

What are the tradeoffs for more coefficients for fractional delay filter?

Is there a rule of thumb for how many coefficients to choose for a fractional delay filter? I figure there might be a trade off based on the signal frequency as well as my sample rate (this is ...
4
votes
1answer
91 views

How to determine Hilbert coefficients

I want to take a real signal on my FPGA and create a complex one. I have FIR core implemted for a Hilbert Transform, and Xilinx's datasheet on it shows coefficients of: ...
0
votes
0answers
10 views

understand accelerometer signal [duplicate]

Please i need help, i'm new to signal processing and i'm working with Accelerometer signal to detect abnormal behavior.. i need to know how i can learn signal processing so i can understand signal, ...
0
votes
1answer
63 views

Speech Activity Detection- Adhoc Problem with multi source sounds(birds,trees,heavy air) and how to eliminate them completely

Here i will first try to put my application agenda Typically i am sampling at 8Khz rate and filtering speech components using a BPF(digital) now i have got the 0-4Khz components in which ...
2
votes
2answers
87 views

Comfort Noise Insertion in VAD algorithms

I am in a intention of Porting Voice Activity Detection algorithm in real time hardware,so i was successful in classifying speech and silence/noise but now the problem is replacement of silence frames ...
1
vote
0answers
96 views

Fast DSP with in-chip ADCs

I'm looking for a DSP, around 500MHz, with an in-chip ADC (preferably 2 but 1 will suffice) that samples around 20MSPS. Does any such DSP exist or do I have to use external ADCs?
1
vote
1answer
147 views

Time-reversal and delay commutativity and resulting signals

I'm reading Schaum's DSP book, and in Fig 1-3 they demonstrate why shifting and reversal are order-dependent, showing a couple of simple systems, 1) delay followed by reversal, and 2) reversal ...
0
votes
2answers
68 views

Complexity: Number of MAC operations on a DSP

Suppose a complex $(N \times N)$ matrix $\mathbf{T}$, multiplying a complex vector $\mathbf{x}$ with length $N$. On a given application that I'm studying, which is irrelevant for this discussion, I ...
-2
votes
2answers
84 views

DSP related websites and blog posts [duplicate]

Do you guys know good DSP related websited and blog posts? I know that for example there is this site called Minutify: http://www.minutify.com which is blog posts regarding all areas of electrical ...
2
votes
3answers
484 views

How to pick coefficients for Fractional Delay Filters?

I have a Virtex 6 FPGA running at 200MHz with ADC/DACs on it. I have been converting a WiFi signal (2462MHz) down to a more reasonable IF of 25MHz, sampling, running the signal through taps, tweaking ...
3
votes
2answers
602 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
364 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 ...
3
votes
1answer
225 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 ...
3
votes
0answers
137 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
2answers
725 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. ...
3
votes
2answers
2k 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: ...
4
votes
1answer
293 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
68 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
229 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
2answers
348 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
234 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
224 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
72 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
279 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
995 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 ...
6
votes
5answers
2k 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
168 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
2answers
416 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 ...
1
vote
2answers
638 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 ...
5
votes
1answer
617 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
289 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
678 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
5k 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
220 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
916 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
486 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 ...