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Questions tagged [dsp]

DSP stands for Digital Signal Processing or Digital Signal Processor.

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DAC and ADC Clipping, Headroom before Clipping, Peak and RMS confusion

If I wanted to calculate the back off or headroom of a signal before clipping in an ADC, should this be the difference between the ADC peak full scale and the RMS of my signal or should it be the ...
cesar's user avatar
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What's the maximum sample rate for lattice wave filter?

From Wanhammar, Lars DSP Integrated Circuits: EXAMPLE 6.5 The filter shown in Figure 6.30 has been implemented using so-called redundant, bit-parallel arithmetic in which long carry propagations are ...
kile's user avatar
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What's the essentially equivalent network in digital circuit?

The following is a snippet from DSP Integrated Circuits (Academic Press Series in Engineering) by Lars Wanhammar. Essentially equivalent network in digital circuit. Theorem 6.3 states: If an ...
kile's user avatar
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Problem with I2S and IIR-filter

I am working on a system that receives audio data over Bluetooth using a BM83. This is transmitted to an STM32F7 over ADC, an IIR-filter is applied and the data is transmitted out over I2S. The ADC ...
mgr's user avatar
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DSP VS FPGA for Power electronics Solar inverters

I need some help regarding the use of an FPGA or DSP for the control loop of the three-phase solar inverter. I want to implement a digital feedback system (as shown below) instead of analog and ...
kam1212's user avatar
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217 views

Digital Beamforming performance issues - ADC and DAC clipping

Digital Beamforming when compared with analog beamforming provides many benefits in terms of performance and flexibility. It is however more complex and each antenna will have its own frontend and own ...
cesar's user avatar
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Doubts on implementation of a beamforming algorithm to produce narrower beams

I am trying to understand and implement in MATLAB, the algorithm described in the paper A Beamforming Method Based on Polarization Matching but my math background is not strong nor am I used to ...
RajaKrishnappa's user avatar
1 vote
1 answer
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Rx digital beamforming

I am using Rx digital beamforming with independent ADCs . These ADCs introduce quantization noise which I am trying to calculate and model. I am not sure though if that quantization noise adds in ...
cesar's user avatar
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How to Derive Digital Filter Coefficients From Given Specifications

I am looking to further understand how to derive the coefficients of a digital filter from design specifications. For example, if I wanted a FIR Butterworth low pass filter with a pass band till 0.2 (...
GMoney's user avatar
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2 votes
2 answers
67 views

Switching from time domain to frequency domain and vice versa in signal processing using FFT

I have a problem dealing with a real signal in time domain or in frequency domain when I need to do some processing on it in the dual or other domain via FFT or IFFT. I find myself dealing with a ...
cesar's user avatar
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EPWM1 and EPWM2 are not in sync TMS320F28379D

I am using LAUNCHXL-F2379D, I am creating sine wave of 50Hz using math function. positive half of that sinewave I am giving it to EPWM2 (working on 10kHz) whereas during negative half of the sine wave ...
GANESH JADHAV's user avatar
2 votes
1 answer
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Amplification for digital caliper

I am making a DIY digital caliper, more on the project you can find here. From various places I found how to design the patterned for the PCBs and with a scope I measured existing calipers to ...
DimDqkov's user avatar
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Understanding the nature and the impact of gain flatness on EVM of a transmit signal

I have a signal which has a quite a steep slope in frequency domain (or gain non-flatness), in fact 3 dB. This signal will go through an amplifier and it is going to be operated in the saturation ...
cesar's user avatar
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1 answer
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What exactly does the limiter do in FSK demodulators?

I wonder what exactly the limiter does in FSK demodulators. What are the advantages? Could it be just a saturation in ADC or fixed-pointing stages?
Ali's user avatar
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PCM to decibel conversion in ESP-IDF using PDM microphone [closed]

I am using PDM microphone with ESP32-Wroom-32. By using ESP-IDF i2s_pdm_rx example. I was able to get PCM data steam in an array buffer from these int16_t values. ...
Moaz Azam's user avatar
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Understanding reflection calculation in Wave Digital Filters

I'm trying to grasp this implementation. It involves calculating the reflected wave (b1) using the reflection coefficient and the incident plus reflected waves at two ports in a series adaptor. The ...
thc's user avatar
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What to choose for an RF spectrum analyzer A/D conversion

I assigned myself a (long term) project to build an RF spectrum analyzer. In the basis it will be a super or triple (depending on frequency band) heterodyne frontend with digitized IF and some final ...
Waldorf's user avatar
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FFT of a real sequence

I know there are many questions (almost the same as this one) on this forum and I am sure I have read most of them, however, I still cannot answer this question: I use the RFFT function of ARM CMSIS (...
KaleM's user avatar
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1 answer
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DSP system step response

I am preparing for my DSP exam and have this question about block diagram of DSP system: "Fsampling = 16 kHz, Delay = 4000 samples, Gain = 0.75. Provide impulse response of this system:" I ...
Dominykas's user avatar
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Gain of a signal passed through a N element phased array?

Assuming a isotropic phased array of N elements, the gain would be 10*log10(N) dB which is provided in theory. However, when same signal is passed through all the N elements, the gain is 20 * log10(N) ...
Pacific's user avatar
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How to find unit-sample response

I have found: $$Y = (X + Y)R^3$$ then $$\frac{Y}{X} = \frac{R^3}{1-R^3}$$ then $$\frac{Y}{X} = \frac{1}{(z-1)(z-e^{j2\pi/3})(z-e^{-j2\pi/3})}$$ then I have three poles: $$1, e^{j2\pi/3}, e^{-j2\pi/3}$$...
South goodman's user avatar
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1 answer
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PWM signal gets disrupted when writing to a file (TMS320F28379D)

I am trying to establish a form of communication between my DSP (TMS320F28379D launchpad) and a .py script on my laptop. I have a .csv file with 7 duty cycle values from 0.2 to 0.8. My objective is to ...
Jonathan_the_seagull's user avatar
5 votes
4 answers
2k views

Can an entire signal with transients be reconstructed from an inverse Fourier transform?

If we have a long signal (say some music) that contains a lot of transients at different frequencies (i.e. the relative "volume" of each different frequency changes over time) and we take a ...
thepman's user avatar
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1 vote
1 answer
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What is the autocorrelation function of this energy signal? [closed]

I would like to know how I would go about finding the autocorrelation function of this energy signal. I tried to integrate the function, but I don't know where to start. The autocorrelation function ...
Arcadius's user avatar
3 votes
1 answer
155 views

How to demodulate a signal that was frequency-modulated by an offset sine?

I have a strain-gauge (SG) in a Wheatstone-bridge (WB) configuration that I would like to measure on the 0-250Hz band on 16 bit ...
davidanderle's user avatar
1 vote
1 answer
162 views

Digital Low Pass Filter and PWM Signal

I have a PWM signal going into an ADC. After the ADC the signal is being filtered using an IIR single pole filter. Performance is exactly what I want with one exception. The purpose of the filter is ...
Tony's user avatar
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2 votes
0 answers
102 views

How to Switch DSP to Passive Speaker Driver?

I am working on an Active Noise Reduction project for a headset using Codec ADAU1772 from AD. There are two inputs, one of which is feedback mic. connected to AIN0, second is INT+ (Intercom or music) ...
Firat Dagkiran's user avatar
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0 answers
53 views

FPGA Ressource Estimation

Currently I'm planning my 'first' FPGA project. I did some VHDL during university, but have no 'practical' experience yet. Therefore, I'm having a hard time estimating the resources required before ...
ElectronicsStudent's user avatar
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123 views

Eliminating noise from ADC readings using digital filters

I need to take some measurements using MCU internal ADC from the following circuit, to be able to calculate the amount of current produced from the programmable current source which discharging the C2 ...
TomAndreson's user avatar
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Magnitude of transfer function

I was able to find the transfer function of this LTI system block diagram, but am unsure of where to start on how to find the magnitude. The transfer function I have from this is: $$H(s) = \frac{Y(s)}{...
Kuronome's user avatar
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1 answer
241 views

DC offset to an ADC reduces dynamic range which reduces accuracy. How to quantify accuracy due to loss of dynamic range?

It’s understood that the dynamic range of the ADC should match the max amplitude of the signal to achieve maximum accuracy. If it doesn’t, you lose digital codes which reduces accuracy. How can one ...
Efanatic's user avatar
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3 votes
3 answers
405 views

SPI vs parallel data/address bus - speed and data troughput

I have a digital board (developed more than 10 years ago) which has a DSP and an FPGA that communicate over a parallel 16-bit data + 16-bit address interface. In the FPGA, there is a sort of dual-port ...
Alessio Caligiuri's user avatar
1 vote
1 answer
96 views

Understanding how a digital microphone's output gets stored in DSP memory

I am working with a DSP that takes audio input from a digital microphone. For the sake of this discussion all input sounds are pure sinewaves. The DSP receives microphone data as 24-bit numbers. Let's ...
user13267's user avatar
  • 621
0 votes
2 answers
172 views

Splitting a mono line, instrument, or microphone audio signal for two consumers

I am building an audio device that receives a mono input signal from a microphone or instrument via a regular 6.3mm jack. From there it goes into a digital signal processor (a USB sound card, actually)...
Simon Fischer's user avatar
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0 answers
54 views

Why is the filter output not in sin form?

The lowpass filters I created with the Matlab fda tool are different from the sinus form in the simulation. What could be the reason for this?
sezi's user avatar
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0 votes
1 answer
325 views

Autocorrelation digital logic design

I'm trying to implement a autocorrelation module into my FPGA. I have been doing some research on the internet and there are some complicated and advanced methods that use a Fast Fourier Transform (...
Tim_h11's user avatar
1 vote
0 answers
80 views

How would I draw a signal flow diagram for the transfer function of a resonant filter?

The transfer function is: $$H(z)=\frac{(1-r)(1-rz^{-2})}{1-2r\cos(\omega T)z^{-1} + r^2z^{-2}}$$ I have started by finding the difference equation, but I don't know how to draw a signal flow diagram ...
Ben120's user avatar
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0 votes
4 answers
1k views

Repeated poles at origin: does this make system unstable, and why?

Let's say we have a control system that ends up having two poles at the origin. I've simulated this and it appears as though this takes the system into instability. I just wanted to confirm, does ...
your_best_friend's user avatar
5 votes
3 answers
1k views

Noise power v.s noise amplitude

I have started reading the following book: The scientist and Engineer guide to digital processing. At the beginning of chapter two, the following is said when talking about mean deviation v.s. ...
InsaneMonkey's user avatar
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1 answer
105 views

Retrieving Transfer function from Block Diagram (Direct Form I or II) [closed]

I've done a few examples for the opposite of the asked question that is to make Direct Form I or II diagrams from the given Transfer Function H(Z). But when I try to backtrack and apply the same logic ...
dcypher's user avatar
  • 13
0 votes
1 answer
55 views

How is noise analysis done in practice

If we identify a signal as a Random Process how do we study it in practice, how to determine its characteristics such as stationarity(whether it is Stationary, Wide Sense Stationary or ergodic),how to ...
jomon's user avatar
  • 67
0 votes
3 answers
254 views

Problem with SN7407 output

I'm carrying out an experiment using an SN7407, which is an open-collector output buffer. The use of this IC is to make the pulses (PWM) coming from a DSP (~3.3 V) be sent to a gate-driver circuit (~...
Erick's user avatar
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0 votes
2 answers
121 views

Way to transfer large amount of data between NIOS and FPGA

I'm searching a method to transfer large amount of data (350 bytes) from NIOS processor to FPGA via Avalon Interface. The board I'm using is DE0-Nano from Altera, therefore, I'm trying to implement ...
LowK's user avatar
  • 27
0 votes
1 answer
245 views

GNU radio FRS transceiver

I am trying to implement the transmit section of this GNU Radio transceiver design, documented here. This design is intended for a bladeRF SDR, but the hardware available to me is the USRP B210. The ...
choga12's user avatar
5 votes
2 answers
383 views

Performing block reduction with many takeoff/summing points in sequence

I'm an audio programmer, and I've recently gotten into using transfer functions to model the response of my DSP. However, I've hit a roadblock in trying to reduce a block diagram of one of my DSP ...
Jackson Kaplan's user avatar
1 vote
1 answer
100 views

Noise reduction algorithm for an AC mains voltage measurement

I am measuring and displaying the mains voltage at six inputs with a 12-bit ADC. To make the calculation simple and fast I am only using 11 bits. I am sampling the 50 Hz sine wave at 20 points using a ...
Mumbai Verve's user avatar
0 votes
2 answers
122 views

Waveform acquired from scope, what is the sample rate for post processing?

I want to make a noise measurement with my scope, but I need it to be band-limited. I am using a Tektronix TBS1102, which allows me to save to a USB flash drive an Excel/CSV file with the X and Y ...
S.s.'s user avatar
  • 2,780
-2 votes
1 answer
112 views

Matlab: using a small sample time for sine wave, makes the wave's frequency "accelerate"

I noticed that if I use a small sample time of DSP sine wave in matlab, the signal's frequency "accelerates": This is the signal with a proper sample time: But if I reduce the sample time ...
Christianidis Vasilis's user avatar
0 votes
3 answers
570 views

How to modulate the amplitude of a square wave at high speed (MHz)?

I want to build a modulator for driving a laser. I want the laser to produce square pulses with varying amplitudes. Hence I built a 1 MHz square wave oscillator (relaxation oscillator with an opamp.) ...
Ravi Pradip's user avatar
4 votes
5 answers
799 views

Digital signal processing vs. analog signal processing for a 100kHz DAQ project

I have a dumb question to ask (for a lot of you) but it has been bugging me for a while and I need someone who can give me a clear answer. (I'm a second-year in electrical engineering.) I am working ...
M.K. Saunders's user avatar

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