Skip to main content

Questions tagged [digital]

The tag has no usage guidance.

Filter by
Sorted by
Tagged with
1 vote
1 answer
87 views

Removing DC bias

I have a signal that I have recorded from a AirSpyHF and want to remove a unwanted narrow signal with a notch filter (see figure). The spike in the image below appears to be a center spike, however it ...
Al Grant's user avatar
  • 131
2 votes
1 answer
89 views

How does a time delay affect the difference equation of a LTI-system?

I'm right now working on my digital signal processing homework and among the exercises is a tough LTI-system in a canonical-like form, however right before the output is a time delay. Suppose the ...
Luxdragon's user avatar
2 votes
1 answer
55 views

Digital Audio Resampling

I've been working from this reference document, essentially trying to recreate Figure 7 with the understanding that the two functions place_signal and ...
Michael Blazej's user avatar
4 votes
1 answer
53 views

Show that the similarity between a signal at an analysis frequency with a phase offset with a reference signal is $\frac{N}{2}\text{cos}(\phi)$

I'm reading Brian McFee's excellent book Digital Signals Theory. He presents the DFT as a measure of similarity $S$ between a digital signal $x[n]$ with $N$ samples and a set of reference signals $y_m[...
NickleDave's user avatar
1 vote
3 answers
153 views

Energy of Transmit Pulses in Digital Communication

Consider the problem of transmitting just two SRRC pulses through an noiseless Channel. The two transmit bits are {+/-1}. In plots below, the individual transmit pulses are shown in black, while the ...
lakshminarayanan raghavendran's user avatar
2 votes
3 answers
2k views

Trying to understand the general scheme of how real-time audio signal processing works on embedded system

I played around with many microcontrollers before but never had a chance to do any kind of signal processing. I'm trying to learn how to implement digital signal processing algorithms on the ...
rairairaira's user avatar
0 votes
2 answers
35 views

What does the frequency domain of bipolar Alternate Mark Inversion (AMI) look like?

Someone said to me that Bipolar AMI is not low-pass; it does not include the 0Hz in its frequency domain. I doubt that that is the case. However, at present I am unable to do Fourier transformations. ...
tryingtobeastoic's user avatar
1 vote
1 answer
61 views

What method is used in DSP to determine which discrete time frequency another frequency aliases to?

I am reading a DSP textbook and I am on the sampling portion. I understand that a discrete time frequency greater than 1/2 will alias to frequencies in the interval -0.5 to 0.5 and other frequencies ...
Michael's user avatar
  • 23
0 votes
1 answer
119 views

How to find sampling frequency in aperiodic signal?

I have a signal that is $x(t)=e^{-3/t^2}+2\cos(3\pi t^3)$ and I'm trying to find its proper value for fs. But I couldn't find a way on how to do it. I'm working on an interval $0<=t<=2$ and $x(t)...
EEStudent14's user avatar
0 votes
2 answers
195 views

Band pass filter = Band stop filter if you switch the cutoff frequencies?

The normalized frequency response of a N points bandpass digital filter is: $\frac{sin(\omega_{c2}(n-M))}{\pi(n-M)}-\frac{sin(\omega_{c1}(n-M))}{\pi(n-M)}$ where $\omega_{c1}$ is the normalized lower ...
Volpina's user avatar
  • 295
0 votes
1 answer
91 views

Learning transceiver algorithms

I want to learn basic transceiver algorithms. I already have a good grasp of the mathematics and statistics involved, so I am primarily interested in algorithms. Any books or site recommendations, so ...
QAM's user avatar
  • 1
-2 votes
2 answers
735 views

I have a doubt regarding FFT size in 5G NR 60 MHz Bandwidth

maximum RBs for 60MHz with SCS = 30 kHz is defined to be 162 RBs, so NSubcarriers is 162 x 12 = 1944, 1944 < 2048. Then NFFT size for 60 MHz bandwidth with SCS = 30 kHz should be 2048 instead of ...
Eliza's user avatar
  • 1
0 votes
0 answers
40 views

Phase Unwrapping [duplicate]

I'm also quite new to world of radar signal processing. I don't understand why we have to make phase unwrapping process. Could you light in me about it? It would be nice conversation speak about it. I ...
Paroz's user avatar
  • 1
1 vote
1 answer
175 views

Is the Final value theorem applicable for some non-causal signals?

A digital signal $x[n]$ is zero at odd sample numbers and one at even sample numbers. What will be the value of this Signal at $n = \infty$? Not exactly sure, but I thought that infinity (as a integer)...
shaycormac1970's user avatar
2 votes
0 answers
18 views

How to emulate lower resolution digitized data (higher step-size)

I am a statistician by training, and I am very new to signal processing, but I would appreciate some help and patience! Suppose I have a digitized signal measured at a given frequency, range, ...
Lacey Etzkorn's user avatar
1 vote
1 answer
161 views

Sampling Precision, Jitter, and PID

I am having difficulty understanding the effect of quantization noise and ADC resolution in a digital PID controller. I am also trying to understand the effects of sampling jitter on a PID controller. ...
FooAnon's user avatar
  • 296
0 votes
1 answer
481 views

USRP N210 Main clock rate and The transmission signal Sampling Rate

I am testing the transmission and receiving signal using two USRP N210. The communication setting as follows: ...
Ze Zh's user avatar
  • 13
2 votes
2 answers
1k views

Do the digital filter coefficients need to be forced to sum to 1.0?

I coded a digital filer in Matlab based on equations I found for a Martin-Graham (low pass type). It looks like it is working. So I want to smooth some data and then calculate the first derivative. ...
Jim F.'s user avatar
  • 21
2 votes
1 answer
130 views

Standardize fft amplitude with different encodings

I'm trying to "standardize" the results of the fast Fourier transformation. I've one wav with two peaks (3000 and 3300 Hz) saved with three different encodings(int16, float32, int32). And ...
Cholas's user avatar
  • 23
0 votes
1 answer
133 views

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(...
DSP_CS's user avatar
  • 1,884
0 votes
3 answers
480 views

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 ...
DSP_CS's user avatar
  • 1,884
1 vote
1 answer
153 views

Why is Line coding called "Line coding"?

Guys do you now why line coding also called digital base band modulation named "line coding" where does it come? and how is the name related to modulation
yohans hailu's user avatar
20 votes
3 answers
4k views

What sampling frequency should I use if Nyquist is not available?

I have the following homework question that confuses me: We have an audio emitter that can emit two signals: It either emits a sine wave at 23 kHz or it emits a sine wave at 25 kHz. The receiver has ...
NN amateur's user avatar
1 vote
0 answers
77 views

How to recover modulation symbols?

I am using GNURadio to send and receive a limited number of modulation symbols. I have constructed the following flow chart:random source(no repeat)->chuncks to symbols(file sink1)->polyphase ...
lin's user avatar
  • 11
0 votes
1 answer
48 views

Dual functional integrated Digital Cathode Ray Oscilloscope and Digital Multimeter product

Is it feasible to design & construct a Dual functional integrated Digital Cathode Ray Oscilloscope and Digital Multimeter products in a single product? Idea is to integrate the Digital Multimeter ...
Prashant Akerkar's user avatar
0 votes
1 answer
62 views

Phase Error Growth in Digital Mixer Signal

I am working on a code where digital mixing is to be performed for down-conversion of incoming signal. The mixer signal is created using the phase accumulation approach. However it was observed that ...
malik12's user avatar
  • 488
0 votes
0 answers
25 views

Speaker Walkaround

Does anyone know what Speaker Walkaround means in context of a digital amplifier? Also how is this test performed? Lastly, how is it different from Load Diagnostics in Amplifiers?
Zidane's user avatar
  • 1
2 votes
1 answer
360 views

Why symbols have to be upsampling, if the symbol rate and sample rate is same what happen?

In digital modulation process, after constellation mapping and before pulse shaping, the symbols are upsampled with the number L(samples per symbol). At this moment, why symbol have to be upsampled, ...
agile's user avatar
  • 109
1 vote
1 answer
236 views

Balance Fader in Digital Amplifiers

What is the purpose of Balance / Fader in a digital Audio amplifier? In a signal flow what is the most apt position to place balance and fader(for example after before a compressor unit or at the ...
Clauvunate's user avatar
0 votes
2 answers
128 views

Centre channel derivation using correlation

I would like to know the algorithm for how centre channel is derived from rejecting left and right channel in audio upmixing process. Can someone explain intuitively rather than hard hitting ...
Clauvunate's user avatar
0 votes
1 answer
36 views

Is there a separate word for the time domain when working with digital signals?

Typically the phrase "time domain" implies an analog signal x(t), but I've also seen it referring to a digital signal x[n]. Is there any phrase that refers to x[n], but not x(t)?
user8577930's user avatar
4 votes
4 answers
1k views

Why is phase range between $-\pi$ and $+\pi$ (instead of $0$ and $2\pi$)?

I wonder why the phase range in DSP applications is between $-\pi$ and $+\pi$. For example, why is that not between $0$ and $2\pi$? This question popped when I was reading about synchronization in ...
Ali's user avatar
  • 95
3 votes
1 answer
197 views

Wave Digital Filter Bridge-T Resonator implementation, gives expected cutoff frequency but incorrect gain and roll-off

Trying to implement the WDF in Fig 5(a) of this publication. The response of the ideal op-amp implementation is given by the black curve in Fig 7: Here is the plot I get when trying to implement the ...
Nick Nagy's user avatar
0 votes
1 answer
476 views

how to convert SPL in dBA to dB

I wonder how I can convert dBA (A weighted) to dB and vice versa. can anybody provide any kind of table or online conversion app? thanks following formula comes from https://en.wikipedia.org/wiki/A-...
SeAlGhz's user avatar
  • 91
0 votes
1 answer
215 views

Band limited discrete signals division

If I take two analog, band limited signals $ x(t) $ and $ y(t) $ with non-zero $y(t)$ I can define the division: $$ z(t) = \frac{x(t)}{y(t)} $$ In general $z(t)$ may have a very wide bandwidth, also ...
Andrea's user avatar
  • 539
0 votes
1 answer
137 views

Find which signal was transferred in 32QAM

I've been assigned to solve a problem for 32QAM signals. The thing is, there are little to none sources in my native language to get any decent information on the topic. The problem is (in its ...
MaxelRus's user avatar
  • 103
2 votes
0 answers
188 views

How can I automate the calculation for P and I coefficients of an FPGA-based digital PLL?

I am working on designing an FPGA-based PLL through an NI myRI0-1900 FPGA and LabVIEW software. I have been able to create a functioning PLL with a PI loop filter, and but I found functioning P and I ...
ExactPlace441's user avatar
0 votes
0 answers
41 views

MAP Estimation Problem for Frequency Band Occupancy Calculation

Consider designing a wireless communication system. Your communication link is not iso- lated in the environment, there are other similar devices trying to communicate in the same general frequency ...
ranjana sengupta's user avatar
0 votes
1 answer
87 views

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 ...
DavidJohnsons's user avatar
0 votes
1 answer
64 views

Hw does a digital piano work?

I know someohow it combines different harmonics to synthsis note, but I don't exactly know how this happens. Does digital piano use pwm to synthesis a note?
Ardawan's user avatar
0 votes
0 answers
154 views

For an LTI system, the zeos and poles are defined in the z plane. How to add additional poles and zeroes so LTI system is real valued?

The question already has a zero and a pole in the complex plane. It is asking to add additional zero and pole so LTI becomes zero valued. zeros=-1/2 + 1/2 i , poles = -1/3 -1/3 i . What I understand ...
Abdul's user avatar
  • 1
1 vote
1 answer
176 views

Are there analog re-configurable filters?

I know that we can build a chip (or circuit) or use FPGA to have re-configurable digital filters. But how about analog filters, can we have re-configurable analog filters? thanks
Amro Goneim's user avatar
1 vote
1 answer
992 views

Discrete signal vs digital signal?

Is there any difference between the two? When we say discrete signal, does it only imply that the x-axis or time is discrete? Or it can also mean discrete amplitudes on the y-axis? Or is the signal ...
DSP_CS's user avatar
  • 1,884
1 vote
0 answers
33 views

What are the applications of Non Unique Factorization Domain (Non-UFD) in signal processing? [closed]

Let $m$ be a composite, and $\mathbb{Z}_m$ is the residual ring of modular $m$. The $\mathbb{Z}_m$ is clearly a Non-UFD. What are the applications of the polynomial with coefficients defined on $\...
worm's user avatar
  • 11
0 votes
1 answer
76 views

Combining CDMA with differential Manchester-like encoding?

I wonder to which extend it is possible to combine a self-clocking code (like Differential Manchester encoding) with synchronous CDMA, and how it would be done? Suppose I want to receive some bits ...
divB's user avatar
  • 473
1 vote
2 answers
2k views

Sampling and Reconstruction of digital signal in Matlab

I'm trying to write a program in Matlab that samples (using Nyquist theorem) and recovers signal. However, I cannot write sampling part for sum of 2 signal. ...
Oğuzhan Kırlar's user avatar
1 vote
2 answers
494 views

Z-Transform and DFT

I have $$H(z) = \frac{\left[b_0 + b_1 \cdot z^{-1}\right]}{\left[1 + a_1z^{-1} + a_2 \cdot z^{-2}\right]}$$. Why is it that I can calculate the spectrum between $0$ and $0.5 \cdot f_s$ by calculating ...
dimwit_recluse's user avatar
0 votes
1 answer
1k views

How the Grayscale images are converted back to RGB?

In case of RGB to grayscale conversion we take the average of R,G,B channels and convert them to Grayscale which has songle channel. How to convert back the Grayscale to RGB ??
vignesh babu's user avatar
0 votes
1 answer
82 views

Obtaining a high sample rate digital approximation to an analog filter from lower frequency measurements

Consider a set of dense, but generally irregularly-spaced frequency response measurements of some real low-pass analog filter. Denote the maximum frequency for which a frequency response measurement ...
rhz's user avatar
  • 457
0 votes
1 answer
92 views

Two-GMSK channels demodulation: 3dB difference in BER

I am trying to demodulate individually two GMSK channels separated by 50kHz (one channel = 25kHz). I am working on a limited resources hardware so I decided to make some optimization on the signal ...
phaseer's user avatar