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# Questions tagged [dsp-core]

A dsp core is a processing unit specialized for digital signal processing tasks.

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### 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 ...
23k views

### Intuitive explanation of cross-correlation in frequency domain

According to the cross-correlation theorem : the cross-correlation between two signals is equal to the product of fourier transform of one signal multiplied by complex conjugate of fourier transform ...
6k 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 ...
1k 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 ...
4k 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 ...
593 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 ...
1k 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 wider ...
4k views

### What does 'canonical' mean?

What does 'canonical' mean? e.g. 'write IIR or FIR in canonical form' I get the gist of it but maybe someone here has a precise definition. In a non-DSP context it would mean something like '...
16k 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.
160 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 $\frac{f_s}{2}$?
302 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 ...
473 views

### SNR of a 16-bit DSP with 12-bit ADC, 40-bit accumulator

According to the 6dB-per-bit rule, a 16-bit DSP would provide 96 dB of dynamic range, approximately. If the ADC and DAC are both 12-bit long, the DSP is still considered to have 96 dB, or the ...
209 views

### Problems with implementation of a band-stop filter on an MCU (dsPIC) using fixed-point arithmetic

I'm trying to implement a band-stop (notch) filter on a microcontroller dsPIC33EP64GS506 to filter out a $100\textrm{ Hz}$ component from an input signal. The ...
1k 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 ...
2k 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 ...
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### Thoretically, what makes multi-pitch detection so difficult?

For multi-pitch detection, I mean for example pressing several piano keys at the same time. For signal pitch detection (pressing one key at a time), after FFT, mostly the frequency with the largest ...
1k 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: (-819,0,-1365,0,-4096,0,4096,0,...
3k 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. ...
247 views

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 X(\omega)\... 1answer 236 views ### DSP interview question: use of the identity in development of a significant transform I'm preparing interview and found this question. But I don't really understand what is the question. Does it ask about Fourier transform or Z transform? How the simple identity $$xy=\frac{1}{2}x^2 + ... 1answer 486 views ### Real-time block processing setup on ARM Newbie question here. I'm working on a system that is running a 32khz sampling rate, audio, 16 bit DAC. Previously, our system had a very hard latency requirement of 1 sample. So no problem, we ... 1answer 359 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 ... 1answer 77 views ### Proof from wave equation of upper limit of active noise cancellation Anybody know of a proof of the hard upper frequency limit of active noise cancellation using the ear canal volume as a boundary condition to solve for the upper frequency limit? I keep on encountering ... 2answers 1k 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 DSP ... 2answers 487 views ### Hilbert transform of a bandpass signal [closed] I have a bandpass signal x. I am taking hilbert transform of it. I want to reconstruct amplitude and phase of the signal.Therefore, i am using Hilbert transform. ... 1answer 611 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 ... 1answer 793 views ### What's the difference between using DFT, IDFT or DCT to calculate cepstrum of a power spectrum? I've seen different equations that calculate cepstrum from power spectrum, but the equations are not consistent. Some people use Fourier transform, some use the inverse Fourier transform, and some use ... 2answers 471 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 ... 3answers 198 views ### How can define an indicator which measures the degree of similarity between two signals? The similarity of two signals is calculated by cross correlation. But, how to define an indicator which quantitatively measures the degree of similarity between two signals? Thanks. 2answers 274 views ### What is so special about this filter coef = [1,2,1]? Can anyone tell what kind of filter is this? I can see it's smoothing/averaging the input signal, but why specifically with [1,2,1] instead of something else? What is so special about this coef=[1,2,1]... 4answers 6k 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 ... 2answers 2k views ### Does sampling in the frequency domain cause time-domain aliasing? Let's say I have an impulse response h[n]. I analyze the power spectrum of that impulse response similar to fourier transformed h[n] corresponding to roughly H[f]. Now I compare H[f] with ... 1answer 581 views ### DSP with 8+ inputs for simple beamforming Does anyone know of a DSP that is cheap and has 8+ inputs (8 I2s mics) and that can perform beam forming calculations? I'm super new to circuit design so preferably one thats beginner friendly. ... 2answers 4k 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 ... 1answer 1k views ### Stereo Widening I recently found this implementation for widening a stereo signal by Michael Gruhn. http://www.musicdsp.org/showArchiveComment.php?ArchiveID=256 I've implemented this myself as follows: ... 1answer 219 views ### FFT of a N-length real sequence via FFT of a N/2-length complex sequence I am doing FFT. The one way I did is to put real part in one array and the imaginary part as zeros and then calculated the FFT. But this take up lots of resources. Now I want to calculate FFT by ... 3answers 2k views ### Get a N-FFT with two N/2-FFT already computed After somme researchs on the web, I don't find the answer of my problem (or I don't understand it) and I hope this post will succeed. I'm working on a real-time FFT convolution algorithm (C++) which ... 2answers 519 views ### Differences between DSP programming vs MPU programming I'm moderately skilled in programming classic microcontrollers. Embedded C, built-in peripherals, interrupts, the daily bread of an Embedded programmer. I'm interested in expanding my skillset onto ... 1answer 87 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}$$I already plot it, and that seems to be a periodic function with H(0)=0, H(\pi)=2, , is that enough to show that this ... 1answer 179 views ### High performance DSP processor with better on-chip memory I am currently using a combination of 4 ADSP 21469 because of the complexity and huge chunk of code that I currently have. BOOT and application will execute from external Flash memory. The code for ... 2answers 557 views ### Raised-Cosine Filter - calculating phase response I want to compute phase response of an raised-cosine FIR filter. I have trouble finding the complex transfer function of the filter in literature (there is a real valued transfer function of frequency)... 1answer 956 views ### How to find phase shift and process I/Q channel I am currently working of 24GHz FMCW radar signal processing. The Module I'm using has 2 receiving antennas and each has 2 outputs (I and Q channel). So overall I've 4 channels. I'm using combination ... 2answers 119 views ### Do I need to worry about synchronization in a digital signal processor? I understand that a digital signal processor consist of multiple MAD (muliply add units), which execute in parallel to implement a difference equation, like this one:$$ y[n]=10y[n-1] + 45y[n-2] + ... 1answer 2k 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 ... 1answer 207 views ### Hardware requirements for ambient active noise cancallation system using adaptive LMS Background I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ... 1answer 62 views ### Compression Sensing for Blind Source Separation I am new to Signal Processing, and am interested in compression sensing for audio files. CS is based on the algorithm that, given some sampling of a signal$f\$ in order to obtain a smaller (compressed)...
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I'm doing a student research project connected with VAD. And I want to find speech corpora with added noise. I found many references to the NIST Speaker Recognition Corpora. But unfortunately, all ...
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### FIR filter design and implementation - sample rate and number of taps

I have been designing and using IIR filters to process audio for a while now but want to design some FIR filters for delay equalization. I have done some preliminary work using the following approach: ...