Questions tagged [analog-to-digital]

An analog-to-digital converter (ADC, A/D, or A-to-D) is a system that converts an analog signal into a digital signal.

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If I know the RMS noise/the variance of a DC measurement, can I simply subtract it from the measurement?

Let's say I have an electronic system that's taking a measurement. It provides a simple bipolar excitation current to a resistive load (bipolar square wave so as to cancel out thermal emf), puts it ...
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Time Division Multiplexing [closed]

we design TDM:source1:analog, bandwidth 4 KHz, source2: analog, bandwidth 4 Khz, source3: digital rate16 kbps, source 4:digital, rate 32 kbps. each analog signal sampled at the Nyquist rate and each ...
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Bilinear transform for a system in zero-pole-gain form

MATLAB's bilinear performs the following steps for a system in zero-pole-gain form If fp is present, it prewarps: ...
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FMCW radar and calculating ADC Sampling rate

I am new to FMCW radar and I am using a TI module. I have been reading the TI reference material and they keep mentioning the parameter "ADC Sampling Rate". I am confused by what this means ...
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Interpolating Filter for Quadrature Detector with one ADC

Single ADC is sampling at 16000 Hz the I and Q outputs of quadrature detector. So each I and Q signal is sampled at 8000 Hz, with phase offset 90 degrees between them. Bandwidth of interest is 3400 Hz....
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What does the following structure mean and how to implement this structure in matlab?

Looking forward to explaining what this structure means, and the relationship between the two structures. Paper: An Efficient Spur-Aliasing-Free Spectral Calibration Technique in Time-Interleaved ADCs
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How to apply MUSIC to FMCW radar IF data?

I'm working on target extraction with raw FMCW ADC data. I arranged my received data in an (numADCsamples x numAntennas x numChirpsPerFrame) radar cube. So far I am using range FFT and doppler FFT to ...
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Quadrature Detector with a single ADC

I have a quadrature detector, which provides I and Q baseband signals with 4 kHz bandwidth, set by detector's LPF. I digitiize them on two ADCs with sampling frequency 48 kHz, make Hilbert ...
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What is the value expected on the voltmeter display

I am a complete beginner and i had some problems figuring out what i had to do to solve this. I apologize in advance for not putting the proper tags but i didnt know what to put in there or if even if ...
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What ADC precision and sampling rate for record sound?

What ADC precision and sampling rate would you choose and why if You wish to record sound that has frequency components are 200 to 2000 Hz. And the signal-to-noise ratio of your microphone is 50 dB?
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What could be the possible reason behind curved signal?

I have a MEMS microphone connected to the microcontroller on a PCB. The microphone collects the data and sends the analog data to the microcontroller in real-time. On the microcontroller, this data is ...
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What is the intuition explaining the Shannon-Hartley theorem?

In the digital domain, the signal to noise ratio (SNR) may be interpreted loosely but usefully as the number of discrete or quantized levels one can transmit reliably. For instance, loosely speaking, ...
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mp3 encoding in the frequency domain

Let's start with an arduino signal, which can be periodic over time. When this signal is converted from analog to digital it "turns" into a series of bits. At this point, is this signal in ...
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How does under sampling modify the spectrum of a signal in nyquist zones and BW?

I have two conflicting concepts when looking at the spectrum of a analog to digital signal that has been sampled. The nyquist zones. So any sampling frequency fs will make the spectrum replicate at ...
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How to apply hann window to unsigned integer ADC

I am using Arm-Cortex M3 Microcontroller and CMSIS for FFT, my microcontroller does not support floating point unit therefore I have to use fixed point (Q15 or Q31) for calculation. Before calculate ...
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how to plot analog filter impulse responce and frequency responce?

I an new to signal processing, working on a UWB rada data this data has a shape of frames for whcih every frame (list) has 164 values, I want to pass a band pass filter with the 1hz and 5hz ...
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plot the frequency responce of a sos Analog chebychev filter using python

I am new to signal processing I am using an sos analog chebychev filter, i was wondering what would be different in frequency responce plotting using the sosfreqz function as it is mentioned that it ...
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Performing IQ imbalance correction on sampled quadrature signals

What are the techniques which can be used to perform digital IQ amplitude and phase correction in a wideband receiver system when input signals can be of unknown different frequencies and occupy ...
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Does different frequency signal cause different voltage from antenna?

When antenna received signal,antenna generate voltage,then SDR convert voltage to I=amplitude*cos(t)and I=amplitude*sin(t). When ...
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Confusion implementing Quantization in MATLAB?

I am quantizing a continuous time signal in three different cases(using different number of bits) But Why i am getting same quantized signal in xq2 and xq3,despite using different values of D(...
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Can a digital signal has 3 three values(1,0 and -1)?

Typically digital signals are seen having only two values,but is it possible that a digital signal has three values(1,0 and -1)?as shown highlighted in attached snapshot,which has been extracted from ...
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Simulating quantization of ADC using device SNR

Suppose I have ADC microphone with a given SNR in the sense that is related to its quantization. From this tutorial we have the following connection how bit-depth determines SNR, Eq. (9): $$ \mathrm{...
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Integer v Fractional ADC outputs with FIR Filter Inputs

I have a system I'm working on that inputs an 18 bit ADC output into an FIR filter on a Xilinx FPGA. I'm a little concerned I may be representing the numbers wrong. My ADC outputs a signed 18-bit ...
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ADC sampling of incoming signal and aliasing effect [duplicate]

I know from the sampling theorem, that the signal frequency $f_{sig}$ shouldn't exceed 0.5 of sampling frequency $f_{samp}$. I decided to have a look at it and tried for example $f_{samp} = 50 $ $kHz$ ...
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Convert analog to digital and back to analog

I'm trying to convert a signal from analog to digital and back to analog to observe the working of ADC and DAC, I use AD9248 as ADC and AD9767 as DAC, both of them are available on THDA ADA Altera ...
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How to quantify a deterministic system?

So let me layout my question and the thought process on solving it. Say, we have some system, and we want to see how deterministic the system is. By that I mean, I mean if I put some signal (real ...
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Why is this ring-buffer jumping?

Central problem (Abstract): A simple implementation of a two-buffer ring-buffer jumps after filling up completely. What am I missing? About: ADC to DAC feedthrough STM32L476RG NUCLEO as MCU Intended ...
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Can you identify what's corrupting my signal? I see a lot of "empty spaces" between datapoints

I'm trying to acquire a signal from the brain of a laboratory animal under anesthesia via the implantation of silicon probes. I've done this many times, but upon testing some new recording hardware I ...
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Fourier transform and Zero Order Hold

I was asked to explain why a zero order hold (ZOH) filter is not a good choice for signal reconstruction. To answer this question, I was thinking about studying the Fourier transform of the ...
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How to convert non-standard sample range to standard

I'm working on a project to record sound from a Raspberry Pi. I have ADC values ranging from 0 to 4095. I can convert it to an audio file and hear my voice. 12-bit ADC resolution with a sample rate of ...
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Why does twice the sampling rate (Nyquist Theorem) seem inadequate?

I was told in my electronics course that "to reproduce a wave, we need to sample it at least twice every period." If I take this to be literally true, then a sine wave with only 2-3 samples ...
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Which of the following sampling methods can be used to sample x(t) such that this signal can be uniquely recovered from its samples?

Assuming that a continuous-time $x(t)$ having its frequency content in the frequency band $1612\leq|F|\leq2015(Hz)$ is sampled with the sampling rate $Fs=806$ samples per second. Which of the ...
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4 answers
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Why can't quantization noise be zero?

If I have a 3-bit ADC and a reference voltage of 8 V, any input voltage of exact increment of 1 V (e.g., 1 V, 2 V, 3 V, etc.) should not have any quantization error. With a 2 V DC input, the ADC ...
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When applying analog and digital effects on an audio signal is it better to start with a clean digital signal or use the analog effects before ADC?

If I am digitally capturing an audio signal from an analog source and I know I will want some analog effects applied, is it better to use them on the analog signal before doing the ADC, or should I ...
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How to convert iPhone 12 raw audio to sound pressure level?

I have 'raw' audio files recorded from an iPhone 12 Pro Max. I'd like to convert the raw 16 bit .wav amplitude to SPL. It's been a while since I was doing this kind of stuff hands-on and I was never ...
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Can we have an analogous sampling theorem for quantization?

Sampling theorem tells us how we can sample a continuous signal such that we can reconstruct it without any errors, or without losing information. Quantization, on the other hand always results in ...
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2 answers
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What should be my sampling frequency for a peak detector output where it is extremely important for me to capture the first time the peak occured?

Following is the output of the peak detector: The amplitude is not important to me, however the time when the peak occurs is extremely important. So my microcontroller will have an algorithm like ...
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1 answer
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Signal to noise ratio (SNR) of a CW signal

I understand the formula SNR = 6.02N+1.76 for a N-bit ADC which quantizes a modulated tone sampled at twice the BW. I am trying to understand what the same equation would be for a unmodulated CW tones ...
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Why is z (and not ω) the variable of interest for discrete time systems?

A continuous time domain system is well described by the Laplace transform. It allows to express any continuous signal x(t) as the integral sum of weighted complex and exponentially growing/decaying ...
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Phase locked loop for three phase grid

I have been implementing a control software where one of its core functionalities is the ability to synchronize with the three phase grid. Basically I need to implement some form of the phase locked ...
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Can Oversampling help to mitigate the effect of additive white gaussian noise?

Oversampling the signal at the receiver's ADC is a common technique to improve the ADC's resolution, dynamic range and to improve the signal to noise ratio (SNR) due to ADC quantizaiton noise, thermal ...
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4 votes
3 answers
297 views

How to find peak value of an analog signal efficiently after sampling in the digital domain?

I have a bandlimited analog signal for which I want to find the peak value in real time. The signal is sampled and processed digitally at just enough sampling rate. Since the peak of the analog signal ...
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1 answer
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Filter to eliminate a 60 Hz frequency range

I have a analog signal that is converted into a discrete-time signal with an ideal A/D converter with a sampling frequency $fs$. The bandwidth of the signal of interest is $1 kHz$. The resulting ...
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1 answer
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LPF in the stage of IQ demodulator is it a analgor filter or digital filter?

I recently designed the LPF of the IQ demodulator using the Butterworth LPF refering to https://dspillustrations.com/pages/posts/misc/baseband-up-and-downconversion-and-iq-modulation.html. But I have ...
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Question regarding sampling rate of sine wave

I am trying to determine my sampling rate, or the delay between each sample, but I am a little unsure if my math is correct. I have generated a sine wave f = 1000 Hz...
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2 answers
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Difficulty absorbing Idea of interpolation?

I am trying to develop my understand of interpolation and signal reconstruction and uptill now i have understood that there are 3 commonly studied types of interpolation 1)Zero order hold ...
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2 answers
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Should we use a digital low pass or high pass filter to remove sensor bias?

Consider a continuous signal oversampled at, say $2 \;kHz$, and then system digital low pass filtered to a $100\;Hz$ frequency which is the control loop frequency. It is known that there is some bias ...
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1 vote
1 answer
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Question on PCM Sampling and Quantization order

I had a doubt in my mind regarding the orders of sampling and quantization in PCM. What is the impact if the order is reversed, that is, the continuous time signal is first quantized and then sampled? ...
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piecewise-quadratic vs piecewise-cubic vs higher order polynomial interpolation?

There is a question available on DSP SE that mentions types of interpolation used for signal reconstruction but there isn't any mention about the difference between piecewise-quadratic,piecewise-cubic ...
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2 votes
1 answer
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Zero order hold interpolation and Nearest-neighbor interpolation?

Is there any difference between Zero order hold interpolation and Nearest-neighbor interpolation I want to perform zero order hold interpolation in MATLAB,but there isn't any information about zero ...
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