Questions tagged [digital-to-analog]

A digital-to-analog converter (DAC, D/A, D2A, or D-to-A) is a system that converts a digital signal into an analog signal.

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Is the ideal reconstruction process BIBO stable?

It is well known that ideal lowpass filter, i.e. the lowpass filter whose impulse response is $h(t) = \text{sinc}(t)$, is not BIBO-stable because $h(t)$ is not absolutely integrable. However, think ...
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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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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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Efficient Implementation of Digital Upconversion

I have an application where I have a pulse shaped QAM signal, at 2 samples per symbol. I need to take the I/Q samples and perform quadrature upconversion so that they can be output on a DAC that's ...
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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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Attenuation of reconstruction filter for Digital-to-Analogue (DAC) converter

I am using a 12-bit DAC to produce signals between 10 kHz and 400 kHz, produced at a rate of 3.33 MSamples/s (Nyquist frequency of 1.665 MHz). I would like to create a "reconstruction filter"...
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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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Does Sampling frequency come into picture in Samples generation for DAC to generate sinewave (analog signal with particular frequency & amplitude)?

I'm generating samples for dac in STM32. sin(2pii /N)*4096/2. This is the formula I'm using. How to I come to know sampling frequency? Whether sampling frequency comes into picture?
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How to reconstruct original signal from sampled signal?

My original signal as f1=2; f2=5; fs=100; Ts=1/fs; t=0:Ts:1; xt=cos(2*pi*f1*t)+cos(2*pi*f2*t); figure plot(t,xt) as shown below figure. and my sampled signal ...
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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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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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Analysing DAC Spectra: Transient Noise Analysis

I am working with a new Digital-to-Analog Converter (DAC) design in simulation and I'm trying to analyse the output. The device takes in an ideal 14-bit digital representation of a sine wave and ...
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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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What is the gain (advantage) of oversampling and noise shaping in D/A conversion?

It is clear that oversampling and noise shaping in A/D conversion can help in shaping quantization noise. But for D/A conversion, normally there is no quantization and hence I did not understand its ...
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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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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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First order hold from zero order hold filter

I'm stuck in this question for more than 4 hours. (h is the sampling period) I simplified the block diagram but it just seems impossible to get the FOH transfer function from the reduced block ...
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Modulate Bipolar(?) signal

I want to modulate this signal if possible with an Arduino, but have no idea what circuit can help me here. Let's ignore the timing and the measurements from the image for a moment, my problem is more ...
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DSP low pass filter (IIR) no longer works when changed to a new MCU

Having troubles understanding why a DSP low pass filter was that working on the M4 is no longer working on an M7. I recently switched over to a STM32H753ZI from a STM32L432KC. In addition to switching ...
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FSK technique to superpose the digital communication on top of 4–20 mA signals

I start a working with HART devices and I really try to understand how digital data is transferred trough 4-20ma current loop. I start read this article: https://instrumentationtools.com/hart-...
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SNR of μ-law companding

I know the snr after μ-law compression is $\text{SNR}\approx\frac{3L^2}{[\ln(1+ \mu)]^2}$ when $\mu \gg \frac{\max_t m(t)}{\text{rms}(m(t))}$, where $m(t)$ is the message signal, and $L$ the number ...
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D/A converter with

I am trying to figure out how to consider digital-to-analog conversion and how to consider oversampling based on a course I am taking online and figuring out the logic behind this. Unfortunately there ...
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Decoder implementation

Im struggling to implement / build convolutional decoder . I've done implementation of FEC encoder which it's as shown in the photo below: So this is the encoder that I've implemented, and I want to ...
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What is the pipeline DAC tollerable offset from plot

I have analyzed the circuit in full extent as shown bellow. I know that I showed imagine two-stage back to back and see "which input could handle both". Could you please show how exactly I ...
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How does the world's simplest sigma-delta DAC work?

I have run into a simple delta-sigma (supposedly, where is the delta part?) DAC implementation utilizing an accumulator and then using the overflow bit as a modulated 1-bit output. From https://www....
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Resources on multiplexer and demultiplexer?

If I search multiplexer in Google and start reading, I find sources that describe a multiplexer as a device that selects between several analog or digital signals, then outputs a single signal. For ...
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What causes this type of distortion in a constellation I/Q plot?

I'm transmitting QPSK through a digital pulse shaping filter (RRC, 35% excess BW, 2 samples per symbol, 64 taps) at a rate of 56Msym/sec. Receiving it on standard lab equipment (Vector signal ...
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DDS generated chirp and antialiasing filter : what is the best suited filter for a chirp?

I am defining the required antialiasing (analog) filter for a chirp generated by a DDS. The chirp is produced in a FPGA (NCO generation) and forwarded to the DAC for the samples generation (with ...
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How to filter OFDM Signal with an analog filter

I am using OFDM system in Matlab, I want to use an analog filter after the DAC. Because we don't a simulation for the DAC, the generated OFDM analog signal does not have a specific frequency (they ...
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Sample and Hold Circuit at Digital to Analog Converters

Does DACs contain sample-and-hold circuits? I am a bit confused thanks
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Aperture Jitter in Digital to Analog Converters

I have noticed in the literature that there is an internally induced timing jitter inside Analog to digital converters called (aperture jitter). It is caused by the sample-and-hold circuit of the ADC. ...
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Reconstruction of Contiuous - Time Signals

In terms of analog signals, we can represent digital signal as : $$ x[n] \triangleq x_{a}(nT) = \int_{-\infty}^{\infty}X_{a}(f) \, e^{j2\pi f nT} \ \mathrm{d}f $$ While if we focused on the integral ...
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2 answers
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DAC and ADC architecture in SDRs

What are the DAC and ADC architectures in SDRs for inphase and quadrature phase channels? Do we have separate ADC and DAC for each channel or there is some clever way that they are multiplexed to ...
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Types of interpolation used for reconstruction in DSP?

What are the different types of interpolation used in DSP for reconstruction of analog signal from discrete/digital signal I am able to somehow learn two types of interpolation 1st is "zero order ...
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DDS - Supersampling above the FPGA clock frequency

I'm using a Xilinx Virtex 6 FPGA with 200 MHz clock connected to a 1.6 GSa/s DAC to generate sine waves of up to 800 MHz using 8 parallel Xilinx DDS Compiler IP cores. Initially, I calculate the 32 ...
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What is the difference between (Sample Timing Offset), (Carrier Frequency offset), and (Sampling Frequency Offset)?

Briefly, can anyone explain to me what is the difference between (Sample Timing Offset), (Carrier Frequency offset), and (Sampling Frequency Offset)? and is Sampling Frequency Offset (SFO) like STO ...
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4 votes
3 answers
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Digital to analog aliasing or mirror query... DAC can output negative frequency?

Using a clock Fs = 1 MHz. The Digital to Analog (DAC) can make frequencies upto 500 KHz. 500 kHz to 1 MHz is an alias? Or is it called a mirror?? Is aliasing a concept only on the ADC and sampling ...
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2 votes
2 answers
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Direct Digital Synthesis - Super Sampling - Unknown Harmonics

I'm using a Xilinx FPGA (Virtex) with 4 DDS cores (each supplied a 250MHz clock) used in parallel to provide a samples to a DAC38J82IAAV from TI, 16 bit DAC running at 1 Gsps. The four cores super ...
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reconstruction filter - How does it actually work?

I'm trying to form my own understanding on the religious war around using 192kHz as a sampling rate for playback (the Internet seems to have a wealth of material on both sides). I'm struggling to ...
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1 answer
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explanation of Hybrid systems?

I want to study Detailed explanation of hybrid systems?Which incorporate both continuous and discrete time signals & systems? for example In which a continuous-time input signal is transformed ...
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Audio processing. Is it possible to directly access the decoded audio data going into the analog input of a computer

When a computer transcodes an audio file from one file format to another, does the computer first decode it into the raw digital stream (exactly what is fed into the DAC for the audio output) , or ...
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Why do IQ Impairements, Quadrature Skew and DC offset have these geometric effects on the constellation?

I'm studying telecommunications, and would like to be sure to understand the maths behind these effects. For IQ Imbalance and DC offset, it seems pretty clear why we observe a rectangle shape or a ...
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How can we do finite-step signal sampling in practical?

The signal has infinite length. We cannot sample it with a whole length. How can we truncate a infinite signal and sample it in finite step? And then how can we analyze it.
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What is the difference between undersampling and oversampling in analog to digital conversion ?

When converting a signal from analog to digital, we observe undersampling and oversampling. Does oversampling means that the sampling frequency is greater that the signal's frequency and does ...
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About Data spreading in CDMA

If we have data we need to transmit which is $i$, so we should multiply is using XOR function with spreading code $y$. How many bits (Maximum and Minimum) should $i$ and $y$ can have ?
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confusion between frequency and amplitude in digital to analog and vice versa conversion

bandwidth is defined as the difference between the highest frequency and lowest frequency. amplitude is the distance between the, let's call it a crest and a trough, in a wave. cycle is the time per ...
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Why do we use conjugate transpose IFFT in OFDM

I have a question regarding the OFDM and using the conjugate transpose of vector (matrix). As you know, if we have a block of data,let's call it $X_n$, consisting of $N$ data symbols, we should ...
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Multirate Signal processing for matching DAC of the SDR

I have been breaking my head for a long time. Please help with this..any input would be helpful! I am trying to implement a OFDM system using SDR (USRPN210 for Transmitter and USRP2 for receiver) ...
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-2 votes
1 answer
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Quantization of analog sound signals

I am using Librosa to detect pitch in a guitar audio signal. I have very little background in signal processing, and I was quite confused with the distinction between analog - digital signals and ...
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Bandwidth vs. sampling rate of DAC

What is the difference between the sampling rate and the bandwidth of a DAC? I have found in one paper a DAC speced as 15GHz 3dB bandwidth, and 64GS/s sampling rate, and I haven't been able to relate ...
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