Questions tagged [dsp]

DSP stands for Digital Signal Processing or Digital Signal Processor.

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36 views

How does stretch processing work for SFCW?

I've been reading about FMCW radar, and I understand that a chirped signal can be de-chirped by mixing it with a copy of itself, which yields a beat frequency which is much easier to process and is ...
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28 views

DSP on STM32 NUCLEO L476RG board [closed]

I am trying to demonstrate how microcontrollers can be used to perform real-time DSP on nerve fibre signals in an implantable device. I have some programming experience but I am fairly new to working ...
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1answer
48 views

Estimating SOC using coulomb counting with DSP

I am trying to estimate the SOC of a battery using the Coulomb Counting technique. I am implementing the following equation on a DSP inside PSIM: $$ SOC[K] = SOC[K-1] - \frac{I_{Bat}}{Q_{Bat}}\cdot{Ts}...
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33 views

formulating specifications for a digital low-pass filter to extract useful DC voltage derivative signal from noise [closed]

System description: sampling frequency at 50 kHZ Noise +- 0.0125 V useful detectable dv/dt signal is from 0.075 V to 0.25 V at 50 kHz (change in DC voltage) linear phase response needed IIR filter ...
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17 views

PROGMEM problem with my project - dynamic allocation error

My project is and RTISR (Real Time Instant Signal Recognizer). I'm sampling a signal from PIN A0 in the Arduino, and I have 6 types of Signals which I compare with (using squared cross correlation to ...
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1answer
61 views

How would I test if my low pass filter works on a remote FPGA?

I am making an example low pass FIR filter design to go on an Intel FPGA. I have an idea about how I'll test it via simulation (by having the input be a sum of sine waves, and the cutoff frequency be ...
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25 views

Filtering Options for an RTL-SDR (DSP and Analog routes)

I have several related questions regarding filtering for an RTL-SDR radio. It's readily swamped by the very, very strong AM and FM radio stations local to me. So much so that your basic passive RC ...
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1answer
27 views

What is sample rate of DSO and what if we have a DSO with low value for sample rate?

I bought a USB DSO and its sample rate is only 48MS/Sec. I never had any problems measuring normal signals from a function generator and other circuits. I have seen many DSOs with high sample rate ...
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2answers
410 views

How is interpolation performed when a sampled sequence is upsampled?

When a sampled sequence is upsampled, zeroes are inserted between the original sample values. But how are the zero values used to effect an increase in the number of sampled values?
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0answers
90 views

How to amplify the output of a digital low pass filter?

I have used the transfer function of a RC low pass filter to design a IIR Digital low pass filter. \begin{equation} H(s)= \frac{1}{sRC+1} ;First \space order \end{equation} \begin{equation} H(s)= \...
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4answers
450 views

How to pick correct filter for a given signal processing task?

I did a study on digital signal processing and I found many theories. But I don't know how to figure out which filter to use when I get a practical signal processing task. Is there a procedure for ...
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21 views

Finding Bluetooth SoC for audio

I'm trying to find a Bluetooth SoC that supports A2DP (at least) and AVRCP. My goal is to connect my phone to this module and have the built-in MCU perform some task when volume commands are received. ...
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1answer
35 views

Input and output array for CMSIS DSP Real FFT Q15 functions

The first question: in the documentations for CMSIS DSP real FFT functions, it is mentioned : The FFT of a real N-point sequence has even symmetry in the frequency domain. The second half of the ...
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0answers
24 views

Converting Clipping stage of guitar pedal to a formula for use in DSP

Been trying to figure out the clipping stage of a tube screamer so i can implement it in a DSP project. I can simulate a clipping effect by using a tanh(input) type function, which gives me an OK ...
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1answer
51 views

Help with Project

[Difference Eqution]: $$LPF_D=\frac{0.1441z^2+0.2881z+0.1441}{z^2-0.6777z+0.254}$$ With a sample of \$T_s = \$ 8e-6 seconds, in scientific E notation. The difference equation: \$Y\$ = Output \$U\$ ...
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1answer
45 views

How do I approximate number of calculations/operations/memory/hardware is required for a 2^18 point FFT on chip?

I am looking for FFT implementation on Chip/FPGA. I need a high-resolution FFT which is a minimum of 2^18 points. However, I need to approximate how much hardware will I require for this process. I ...
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1answer
49 views

How to send different data to three DACs at the same time?

I am building ultrasonic system that needs to generate three different signals for three transducers at the same time.The system is controlled via DSP. Currently I have a DAC with 4 outputs, but as I ...
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60 views

How to synchronize TI high precision ADC and DAC via SPI?

I am working with DAC80504EVM and ADS8920BEVM-PKD from TI. I am trying to design a system which uses TMDSEVM6657EVM processor board to communicate with these converters via SPI. However, I am trying ...
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2answers
52 views

Help with a Closed loop and zero order hold function ZoH * 1/s(s+a) type

i would appreciate some help in solving this problem, i need to find the transient response for a unit step, i have a brief idea of how to solve, but my graphs in matlab doesnt seem to fit with the ...
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1answer
47 views

Is it possible to digitize small signal with large noise?

I have a small signal, e.g. ~100 uV in a band of interest of 0.1 - 30.0 Hz This signal is accompnied with a huge noise of 50 Hz ~100V of amplitude. ADC's input range is 1V. Is it possible to ...
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1answer
40 views

How can this problem be solved with partial fractions? (Zero order holder with g(s) block function)

im trying to figure out how to solve this problems for values a=2; actually im stuck, professor told me how to work it, but i cant figure out what to do exactly, we are using partial fractions inverse ...
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0answers
45 views

STM32F4 DSP noise with inbuilt ADC and DAC

I'm trying to perform signal filtering with STM32F407VGT. I've set up the ADC and DAC to work synchronously, triggered by a timer at 42kHz, and using DMA. The idea is to process half of the buffer ...
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0answers
40 views

Implementing WOLA block for ADC data on FPGA

I have ADC (ads5463) connected to an FPGA (lattice ecp3) using Verilog, and I have built an ADC interface on the fpga which capturing the data sucessfully and stores the data on FIFO. My next step is ...
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0answers
43 views

use of DSP ICs in control systems

I'm doing a project on dc motor control. I heard somewhere DSP ICs are widely used in the aforementioned area. My question is about the advantage of using a DSP IC for implementing a PID loop for dc ...
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1answer
53 views

FPGA dsp-slices highest sampling rate possible

DSP slices (following Xilinx’s terminology) have certain speed grades (extracted from the switching characteristics of the device) that span up to some hundreds of MHz. Those DSP slices are heavily ...
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1answer
104 views

ADC with parallel DDR LVDS interface and also a parallel CMOS interface

The ADS4126 is a 12-Bit, 160-MSPS Analog-to-Digital Converter (ADC). Its data rate is too high for it to be read out using something like SPI or I2C. It does use SPI for control purpose though. The ...
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1answer
116 views

Difference CPU, ASIP, DSP [closed]

During lecture CPU (Central Processing Unit), ASIP (Application-Specific Instruction set Processor) and DSP (Digital Signal Processor) were briefly mentioned. Unfortunately the main differences (and ...
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0answers
56 views

Sampling Frequency of a pulsed radar signal

I'm new to radar. please excuse my question if it doesn't make sense to you. In a pulsed radar signal, why is the required sampling frequency "2 times the inverse of pulse width" instead of "2 times ...
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1answer
49 views

Accurate signal reconstruction

While reading through this page I came across following info: "For accurate reconstruction using sin (x)/x interpolation, your oscilloscope should have a sample rate at least 2.5 times the highest ...
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1answer
240 views

RMS value of fundamental harmonic vs RMS of third harmonic

If we have a periodic signal (square wave) with amplitude of 100V. Then what would the RMS value of the fundamental harmonic be compared to the RMS value of the third harmonic? After some suggestion ...
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2answers
104 views

TMS320F280049C: Unwanted pulse on GPIO during power on/off

I'm using the TI TMS320F280049C microcontroller. While testing its GPIO behavior during power on/off, I found a high to low transition on GPIO pin, on every power on/off. I'm checking the waveform ...
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1answer
189 views

Signal processing in FMCW and SFCW

With regard to signal processing, how is frequency modulated continuous wave (FMCW) radar similar/different from stepped frequency continuous wave (SFCW) radar? Do we actually modulate the RF carrier ...
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0answers
39 views

Filtering analog audio signal using TLV320AIC3111-EVM

I'm working on a project to filter analog microphone data using the audio codec from TI, the TLV320AIC3111. We are using the EVM However, I'm facing problems with implementing the filters on the ...
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2answers
118 views

Guitar to MIDI conversion

I am wanting to measure the pitch of a single guitar string and convert into MIDI within 10ms. FFT takes too much time, I was wondering what other options are worth looking at? The main issue is ...
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1answer
74 views

WM8804 audio routing question

The WM8804 looks like the ideal candidate for a design I'm working on --- this single chip could replace the need for one receiver and one transmitter. The specific routing requirements are: I need ...
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0answers
57 views

USB Audio Interface with DSP [closed]

I am looking to build an audio interface that can also use real-time DSP to process audio coming from (most important) and going into the host. After some research, I have been zeroing in on the ...
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1answer
43 views

Interpretation of I2S data

I am working with a TI PCM1807 ADC (datasheet: https://www.ti.com/lit/ds/symlink/pcm1807.pdf) and try do some signal processing on a FPGA. The PCM1807 output is I2S. I was wondering how to interpret ...
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2answers
310 views

increase the ADC resolution using oversampling [duplicate]

To increase the ADC resolution, oversampling technique can be used which must be followed by decimation (average) method. Oversampling is just take several samples within time period. Unlike the ...
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4answers
812 views

Is there an efficient way for synchronising audio events real-time with LEDs using an MCU?

I'm working on a project in which LED lights synchronise with audio events. I'm not referring to reactive LED projects that read the ADC from audio amplitude or spectrum, but about having some ...
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1answer
75 views

Translating/reverse engineering firmware commands

I'm trying to build my own Hi-Fi system. So, I recently ordered this Japanese DSP that's able to process Dolby audio. I was all excited until the part had arrived and I found something in the ...
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0answers
116 views

Time delay using IIR filters

I am trying to implement a cardioid pattern with two microphones. I was reading this paper titled "Digital Filter Array Optimization for Directivity Pattern". The link to download the paper is this. ...
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1answer
187 views

PLDs, DSPs, MCUs, MPUs and SoCs Typical Application [closed]

PLDs, DSPs, MCUs, MPUs and SoCs are all embedded devices, they can be programmed / configured to achieve a specific application. I've worked with MCUs and FPGAs in the past and realized that they are ...
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1answer
41 views

Discrepancy in my dB SPL calculations

Having read through this post I tried an experiment. I connected a Extech sound level calibrator to a Type 4189 microphone. The output sensitivity of the microphone is 50 mV/Pa,-26 dB ± 1.5 dB re ...
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2answers
77 views

Load vs LDAC pins

Alright, so I'm using a TLV5628 Octal 8 bit DAC with my stm32f4-Nucleo and I'm kind of confused about the difference between LDAC and Load. I am also trying to figure out the clock rate and I might ...
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3answers
289 views

FIR Filter on Cortex M0

Is it possible to implement basic DSP (such as an FIR Filter) on a Cortex M0 or is the architecture too limited?
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2answers
151 views

Hardware audio mixing [closed]

I'm having a hard time trying to understand some basic sound engineering concepts. From https://en.wikipedia.org/wiki/Original_Chip_Set#Paula: Paula has four DMA-driven 8-bit PCM sample sound ...
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1answer
47 views

Continuous, frequency-domain digital equalizer

I'm facing a real case of DSP application. I have a basic knowledge of digital filtering and telecommunication issues. The input of my "black-box" is a digital audio signal (say a common WAV file, ...
1
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1answer
367 views

DSP Library not working in MPLAB X

I have to make a digital filter with a dsPIC30F4011 (I'm completely new to microcontrollers). I was reading the documentation about the DSP Library to figure out how to use it. I created a simple C ...
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3answers
271 views

Inherent DC Bias in ADC sampling

I was just simulating the quantization behavior of ADCs of different bits, on data sampled by a 24-bit ADC. What I noticed that as we go towards ADCs with less bits a DC bias in the signal becomes ...
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2answers
77 views

Signal Strength to ADC quantisation levels

According to the excellent site of AB4OJ, the Icom 7300 in summary has a 14bit analogue to digital converter with a 1.5 volt peak to peak input (LTC2208-14). I make this to be 91.5uV per quantisation ...

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