Questions tagged [dsp]
DSP stands for Digital Signal Processing or Digital Signal Processor.
415
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How to take z-transform of a function with an absolute exponent value?
Can you help me find the z-transform of the impulse response below:
$$h(n) = (\frac{1}{2})^{|n-1|} + (\frac{1}{2})^{|n|} $$
I know the z-transform of
$$(\frac{1}{2})^{n}$$
is equal to:
$$(\frac{...
5
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1
answer
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FFT Beat Detection Circuit
I have for over a year now tried to figure out both the electronic aspect and the software aspect of my project.
I have somewhat successfully managed to make the overall idea to work,
the bumps I have ...
0
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1
answer
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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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1
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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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1
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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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1
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Best Approach Methods for Signal Processing
I'm looking for some advice, I am hoping one of the many forum experts could help me with some advice on the sampling and processing of signals. I am using an STM32F4 processor with 12 bit ADC and DSP ...
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4
answers
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Different ways of using DSP slices in Spartan 6 FPGA
I am reading the Spartan 6 DSP slice user guide, and I need to use the DSP slice in a project of mine.
I stumbled upon this question, which basically suggests 3 ways of using the DSP slices
...
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What actually is a shadow register?
I noticed the term Shadow Register while going through a datasheet of a TMS320F28335 DSP. What does it actually mean? Does it have a physical location in the CPU as ...
0
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1
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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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3
answers
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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 ...
2
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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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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 ...
0
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2
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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 ...
0
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1
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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 ...
2
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0
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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 ...
1
vote
1
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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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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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How to improve my digital filter to extract DC from noises?
Digital filter description:
The filter's input are sensor data and output will be provided to a PID controller.
Sampling rate: 64kHz
Noises spectrum: from DC to 10kHz
Filter target: to get DC ...
5
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4
answers
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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 ...
0
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1
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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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0
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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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2
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How has this Daikin control signal been encoded?
I just installed a Daikin split system (AC/Heat Pump) and would like to control it remotely. There are similar units available which can be controlled with an IR remote, but mine does not support this ...
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Is it possible to create an IIR filter in an FPGA that's clocked at the sample frequency?
This question is about implementing a IIR filter in a FPGA with DSP slices, with very specific criteria.
Lets say you're making a filter with no forward taps and only 1 reverse tap, with this ...
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1
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878
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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 ...
2
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1
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Are digital signal processors categorised in ASIPs?
Reading Wikipedia, it seems like DSPs use short instruction sets specialized for digital signal processing with hardware designed to specifically accelerate these instructions (e.g. multiplying, ...
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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 ...
2
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3
answers
303
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Sharp and Simple Analog Filter suggestions
I'm looking for schematics of some simple opamp-based analog filter, which has a sharp band-pass functionality. I want it to control small robot with sounds of different pitch.
The matter is like ...
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2
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problem with fixed point butterworth implementation
I have designed a 2nd order Butterworth filter with MATLAB fdatool for VHDL implementation. the cuttoff frequency is less than Fs/10, which should be okay since fdatool uses prewarping.
Indeed, the ...
2
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3
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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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2
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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 ...
1
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1
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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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1
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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, ...
2
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1
answer
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How can I Access Tiger Sharc DSP processor through host pc?
I am working on designing of a test board which involves both FPGA(Xilinx- Kintex 7) and DSP(Tiger Sharc) based signal processing.
Now the end user wants to access the DSP processor programming ...
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2
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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 ...
12
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3
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What's so great about PPC? There's gotta be something
With this question in mind (the first one that stackexchange is offering me right now among the "Questions with similar titles"), I'd like to steal the idea of the question and ask it about PPC.
What'...
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2
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dsPIC33E ADC to FFT help
I am sampling an audio range signal with a bandwidth of 3100Hz and then applying a FFT using the DSP library example from Microchip to determine the most dominant frequency of the signal.
On the last ...
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2
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What is the easiest way to add noise to a circuit?
I'd like to have a sinousid wave with some noise. Then use that signal as input for an arduino, visualize that signal in Matlab and use a simple filter (moving average, for example) to remove the ...
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2
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When is proper to use <= or = assigments in Verilog
I have not clear at all this part of Verilog when using <= or when =
I have some always blocks that make some adds, subtracts and multiplies an example is like this:
...
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1
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Voice playback IC
I was looking for cost effective voice playback IC and came up to ISD 1700 series. Another brief datasheet is here. I don't believe with the same footprint and only for 3-4 bucks I can get up to 16 ...
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What is the "Nyquist" rate for sampling the derivative of a signal?
Background: I'm sampling the current through a capacitor. The signal of interest is the voltage across the capacitor. I will digitally integrate the current measurement to obtain the voltage.
...
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answer
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What are advantages of 1's complement over 2's complement in DSP? [closed]
I wanted to know about the advantages of 1's complement over 2's complement in DSP applications.
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Designing circuit with a FIR filter with feedback
I am trying to implement a VHDL circuit similar to the one in this image:
This Simulink simulation works as expected, but trying to implement the same in VHDL is another issue. The problem is that ...
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1
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Running DSPIC33FJ64GP802 at high temperature
Has anyone run this part to its rated temperature of 150 Celsius? We are able to run up to and beyond this temperature, but if power is cycled the part will not start reliably. We have tried running ...
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Can I2S tdm interface send more channels with less sampling frequency?
We have a arm SoC(system on chip) connected with I2S to a DSP (digital signal processor). SoC and DSP both support 48khz, 8 channel and 16bits in TDM mode.
Is it possible to use this interface to ...
2
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2
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Digital Audio Effects Design [closed]
I've recently graduated university with an MEng degree in Electrical and Electronic Engineering. As a third year project, I've used an ST-Microelectronics ARM Cortex-M4 based Nucleo microcontroller ...
0
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1
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To convert Volts in dB SPL confusion
I wanted to convert volts into dB SPL. Thankfully, I found this article
convert Volts into dB SPL
While trying to understand why he was adding the mic sensitivity along with 94 dB SPL, I found ...
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1
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Signal processing of an IVQ-3005 Radar Module
I'm going to use that module for detecting range, velocity and angle, for that radar.
https://www.innosent.de/fileadmin/media/dokumente/datasheets/180222_Datenblatt_IVQ-3005.pdf
So I have read a lot ...
3
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1
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Understanding digital filters, specifically their effects on time domain information, how to preserve the waveform of a signal being filtered?
According to Steven W. Smith, in his book: The Scientist and Engineer's Guide to Digital Signal Processing, when designing a digital filter, "Good performance in the time domain results in poor ...
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Wrong FFT result when using rfft function on STM32F4
I'm trying to use rfft function in STM32F407G and I am getting wrong results when I compare it with an online fft calculator . I give a sample input as {1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,...
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3
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How does playback rate of an arbitrary waveform generator determine frequency content of that signal?
I'm trying to clear up some conceptual confusion I have about digital signal processing.
Say I have a 1000 point digital signal (stored in the memory of my arbitrary waveform generator "AWG"), and I ...