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-1
votes
4answers
79 views

What is meant by Impulse Response

I have read in all books that: Impulse(n) = 1 when n=0 , 0 otherwise So when we state impulse response of signal x(n) I do not understand what is its actual ...
0
votes
0answers
15 views

DSP review: key difference between dsPIC and SHARC or similar DSP (AD/TI) [closed]

I'm reviewing the DSP technology, I'm familiar with dsPIC33 and FFT operation. Can anyone come across article or paper that make critical comparison between various DSP device (most recent market) ...
4
votes
1answer
85 views

Dual-Tone Multi Frequency signalling

Does anybody know why we need 2 different tone signals: one high frequency and one low frequency to create a tone burst in Dual-Tone Multi Frequency signalling? For example, to create a tone burst of ...
1
vote
2answers
55 views

Verifying channel response identity

If $x[n]$ is input signal, $y[m]$ is output signal, $h[m]$ is impulse response of the channel, then the following relation holds $\lvert Y(\omega)\rvert = \lvert H(\omega)\rvert^2 . \lvert ...
0
votes
1answer
74 views

Best way to inverse Hilbert on FPGA?

I have a real signal that is being sampled via ADC and passed to a Xilinx V6 FPGA. I then do a Hilbert transform on it using the FIR compiler core and get the complex signal (I and Q) to tweak the ...
0
votes
1answer
46 views

DSP - Sinewave: what is $2\pi f t $?

I am a beginner in DSP and is familiar with $\sin(\theta)$ declaration (as studied in trigonometry) - where $\theta$ is the angle it makes with X - axis. Please tell me what does $\sin(2\pi f t + ...
3
votes
1answer
115 views

Under what condition is the available bandwidth $f_s$ instead of $\frac{f_s}{2}$?

Can someone help me with this interview question? It is about the sampling frequency rate. Under what condition the available bandwidth is $f_s$ instead of $\frac{f_s}{2}$?
1
vote
2answers
100 views

Recommendations for DSP Hardware

I'm trying to figure out what a good DSP unit would be to try and learn how to do some embedded programming on? I would like something relatively inexpensive but that still has a decent amount of ...
0
votes
1answer
95 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 ...
1
vote
0answers
66 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
63 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
37 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
34 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
26 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
188 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
94 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
63 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
73 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
59 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
46 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
137 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
11 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
100 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 ...
2
votes
0answers
114 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
161 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
70 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
92 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
623 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
820 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
388 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
295 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
160 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
935 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. ...
5
votes
2answers
3k 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
376 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
71 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
253 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
392 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
265 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 arenĀ“t just the same frecuency-lenght in the wavread arguments. i am supposed to use the ...
1
vote
2answers
265 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
74 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
363 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
1k 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
180 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
469 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
811 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
2answers
708 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
319 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 ...