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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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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33 views

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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25 views

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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46 views

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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42 views

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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22 views

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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5answers
205 views

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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72 views

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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2k views

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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51 views

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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38 views

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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4answers
311 views

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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40 views

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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How to implement this phase rotator?

question 1, I think that we can use si,sq ,siq to control the phase move.In my understanding,in the Fig. 5. Phase rotator ,si, sq correspond to the values on the horizontal and vertical lines.siq ...
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1answer
55 views

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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69 views

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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1answer
98 views

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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1answer
116 views

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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229 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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1answer
80 views

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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1answer
165 views

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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60 views

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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2answers
71 views

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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2answers
196 views

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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1answer
62 views

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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How to apply settings for mu law

I am trying to understand reconstruction of analog signal I am using the files available on mediafire There are two files, M-files through which we get input values from user and then feed those ...
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1answer
56 views

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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1answer
228 views

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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1answer
78 views

Measuring sine amplitude with ADC

My problem is similar to this one: Measure Sine Wave Amplitude from ADC Signal I have a microcontroller that produces a sine wave by means of its DAC. I can choose the frequency anywhere between about ...
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3answers
87 views

Beginner Educational Resources

New member here. Recently got hired into a mid level EE position. Have a BSEE but performed more radar performance analysis at a higher system level previously. In my new group we're doing a lot of ...
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Triangular dither - What does it mean to say that an input signal is uncorrelated with the dither

In the oft-cited ADC dithering appnote: Oversampling Techniques using the TMS320C24x Family http://www.ti.com/lit/pdf/spra461, it notes under triangular dither: However, this approach requires that ...
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How to find the minimum no of bits required to express the filter coefficients in fixed point format from Histogram plot?

I have designed a filter in MATLAB and now I have to convert the filter coefficient in fixed point format from floating point format. For that my professor asked me to find histogram first and then ...
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25 views

Logic analyzer for acquiring EEG signals?

Is it possible? I haven't found anything about it on the internet. I found a logic analyzer here, which is marketed as having the capability to monitor analog signals too: "One useful feature of ...
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1answer
49 views

How to simulate a continuous signal passing through a capacitor (simple coupling capacitor)

the differential equation for the current flowing through a serial capacitor (see for example https://www.allaboutcircuits.com/textbook/direct-current/chpt-13/capacitors-and-calculus/) indicates that ...
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2answers
77 views

ADC sampling rate and signal bandwidth for FHSS

If i am correct; for a radio receiver the ADC sample rate will have to be higher than at least twice the BW of the data (disregarding sparse sampling techniques). So for frequency hopping spread ...
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2answers
64 views

Electric Bass with RJ45 Digital Output

I'm working on a project bass guitar with individual outputs per string using Nu pickups from Cycfi Research: https://www.cycfi.com/ The common sense approach is to simply take their circular ...
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1answer
268 views

Quantization error of SAR-ADCs

My question is related to the quantization error and the functionality of SAR-ADCs. In general, the quantization error of an ADC is defined as +-0.5LSB. If this concept is correct a voltage that ...
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1answer
596 views

I/Q sampling with just one ADC

Usually I/Q sampling is performed on two signals with a 90° phase shift using separate ADCs. Suppose I have only a single (fast) ADC available to perform I/Q sampling, which approaches do exist? Is ...
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1answer
87 views

transfer function of a sampler in the s domain

I would like to modelize my whole system into the S-domain. This is a mixed system, there a numerical part (corrector, ADC, DAC) and an analogic part (plant transfer function, sensors, etc...). I know ...
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1answer
51 views

Increasing ADC output accuracy

Say we have a 12-bit ADC with 30 MHz sampling frequency. I want to perform several DSP operations on output digital signal so that resulting signal would have increased accuracy to 14 bits and output ...
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2answers
163 views

Oversampling in quantization

Q: It is said that "to maintain the same quality in the two cases, we require that the power spectral densities remain the same". Why is this a measure of the same quality? Why is not the ...
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2answers
173 views

Condition for aliasing

Which one of the following is the condition of aliasing? (a) Tails of the replicas enter into the Nyquist interval (b) The tails of the replicas enter into the Nyquist interval and add to the ...
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70 views

Timbre vs sound via oscilloscope

My understanding is that a distinct sound will generate waves of identical amplitude and frequency, but that most things (such as people's voices, music intsruments, etc) produce multiple distinct &...
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2answers
317 views

Using oversampling to increase resolution of a DC-signal as input

Currently I'm working on a project which uses oversampling to increase the resolution of a 12 bit ADC to a maximum of 16 bits. My goal is to fully understand the theory behind oversampling and why it ...
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1answer
48 views

Sampling interval $T$ as a multiplier in digital processing of a continuous-time signals

[from: Discrete-Time Signal Processing, Oppenheim and Schafer, p.224] Q: Why do we have $T$ as multiplier in $TY_a(j\Omega)$ in Eq.155?
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166 views

Is this an error in Oppenheim and Schafer's Discrete-Time Signal Processing?

In Discrete-Time Signal Processing by Alan V. Oppenheim and Ronald W. Schafer (3rd Ed.), in Figure 4.47 the input of D/A converter is $\hat{y}[n]$ but later in Figure 4.64 the input of D/A converter ...
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2answers
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Why do we need DFT when we already have DTFT?

We already have DTFT so why do we need DFT? Please don't write DTFT is used for discrete, aperiodic signal which transfers in the frequency domain to periodic and continuous, but in DFT is discrete ...
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1answer
66 views

Understanding this Hanning Window implemented in microcontroller

I'm trying to calculate RMS of an AC signal which input frequency may vary from 40 to 70 Hz. I would like to use a window function because it accounts for frequency variations and also my ADC sampling ...
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Output signal is being attenuated after the Biquad filter direct Form 1, why?

Thought this would deserve a separate post on its own. What's going on is that I finally got my low pass digital filter working on the STM32H753ZI using the PMODI2S2 peripheral and I noticed the ...