Questions tagged [dsp]
DSP stands for Digital Signal Processing or Digital Signal Processor.
433 questions
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Confused about the right sequence or order of receiver DSP functions
I am looking for a simple answer for a question (or questions) which may be complex ( I tried to answer this question by reading Michael's Rice book on digital communications , and I still couldn't ...
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FPGA Fixed Point Arithmetic
I was currently reading this blog (highly recommended) about FPGA doing math (Doing fixed-point Math in FPGA).
But I am having some doubts while reading the second example (named ...
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spreading factor in LoRa PHY (CSS)
I wonder how the spreading factor (SF) is defined in LoRa PHY.
Specifically, in other spread spectrum techniques (e.g DS, FH), we have:
W = R * SF
W is the (spread) bandwidth, and R is the symbol rate....
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Confusion in power delivered into the grid
I have been working on grid tied inverters but am confused about the working of the current injected into the grid.
I have implemented the phase locked loop using the notch filter according to the ...
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Boost converter ground to ground noise issue
I’ve been working on a boost converter with all the control circuitries being controlled by a microcontroller. I am facing a huge problem of noise on the ground. I have all the grounds common and when ...
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Is Rx Automatic Gain Control useful in such a situation?
This is referring to AGC before hitting the ADC, i.e. before any DSP
We have a situation where the signal of interest is always buried in noise by at least 10 dB nomatter how strong it is.
Is there ...
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Understanding Pulse Doppler Radar pulse compression [closed]
I am reading about pulse doppler radar book where it states that to increase the SNR we use wider pulse width and to increase the range resolution we use pulse compression i.e., match filtering. After ...
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Does sub-sampling apply to digital to analog converters as well?
Sub-sampling is a known technique used in receiver's analog to digital converters (ADC). It allows the minimum sampling rate to be equal to twice the analog signal bandwidth instead of twice the ...
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Confusing FM with CDMA
Frequency modulation (FM) causes expansion in the frequency domain, an expansion which is beneficial in terms of signal SNR.
Direct Sequence Spread Spectrum or CDMA causes expansion in frequency ...
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Phase noise compensation in carrier tracking loops at 38 GHz
I am trying to reach an estimate for a number in dBs of phase noise improvement in a Costas tracking loop. I know that there is a trade off there as increasing the loop BW helps reducing the input ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 (...
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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 ...
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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 ...
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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 ...
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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 ...
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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?
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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. ...
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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 ...
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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 ...
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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 (...
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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 ...
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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) ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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) ...
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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 ...
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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 ...
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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)}{...
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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 ...
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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 ...
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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 ...
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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)...
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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?
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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 (...
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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 ...
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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 ...
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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. ...
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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 ...
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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 ...
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