Questions tagged [digital]
The digital tag has no usage guidance.
156
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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 ...
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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 ...
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57
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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 ...
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1
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92
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Carrier and timing offset
Assume my reference is 1 ppm so if my transmit frequency is 400 MHz and
symbolrate is 100 kHz.
Then, my carrier will have +/- 400 Hz offset and then my symbol will have +/-0.001 Hz offset. Similarly,...
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34
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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:
...
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1
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51
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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 ...
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1
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182
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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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2
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72
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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. ...
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What is IQ Data?
I have no background in telecommunication. Could anyone please explain to me what is the IQ data in telecommunication? It will be best to explain it with some basic analogies. Many thanks!!
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138
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Digital beamforming and phase shift stage insertion
I have read some literatures about the different topologies of beamforming and it came to my mind a question : where is applied the phase shift for each channel depending on the type of topology/...
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53
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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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3
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95
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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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92
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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
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2
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325
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What Is the Correct Way to Apply Richarson Lucy Deconvolution to Luminance Data?
My question concerns the Richardson–Lucy deconvolution algorithm, which is described in Richardson's original paper - William Hadley Richardson - Bayesian Based Iterative Method of Image Restoration. ...
19
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3
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3k
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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 ...
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37
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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 ...
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67
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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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1
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43k
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what's the pass band ripple and stop band attenuation of a digital filter?
Hi i'm a beginner in signal processing i want to know what'sthe pass band ripple and stop band attenuation of a digital filter ?
Thanks.
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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 ...
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42
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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 ...
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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?
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135
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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, ...
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4
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483
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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 ...
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2
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46
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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 ...
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32
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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)?
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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 ...
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119
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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-...
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1
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101
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phase response of an analog filter
I am trying to understand the relation between phase response of an analog filter and its influence on the quality of a signal with PSK modulation.
For example I want to design a BPF for HF ...
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1
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4k
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Differential BPSK vs BPSK what are the advantages and disadvantages?
Recently I have come across different variation of the sane modulation techniques. I am quite confused about the advantages of different constellations. In particular, what are the advantages of using ...
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difference between GMSK (Gaussian Minimum shift keying) and GFSK (Gaussain frequency shift keying) [closed]
I am unable to differentiate GMSK and GFSK.
How do the modulation techniques differ in Tx?
How do their receiving techniques differ..?
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51
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Does Scrambling increase speed of signal transmission overall? And is Scrambling a redundant process? [closed]
Title is a badly worded question, on which I do not have enough knowledge on the subject to answer on my own.
Anyone to help out?
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117
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digital phase lead algorithm
I have two sine signals A and B which are simultaneously sampled at 2500 Hz using ADC. Through oscilloscope, I find that signal B lags A by about 5 degrees. I would like to digitally lead signal B to ...
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65
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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 ...
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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 ...
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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 ...
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What is the difference between continuous, discrete, analog and digital signal?
It's my first time studying DSP and I've faced a problem finding a convenient definition.
Are the following definitions correct? And if so why there are some resources defining it in other terms such ...
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46
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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?
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3
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Digital Distortion effect algorithm
I read the DAFX book by Udo Zölzer about the distortion effect at pages 124-125
and it says that suitable simulation of the distortion is given by the function:
$$f(x)=\frac{x}{|x|}\left(1-e^{x^2/|x|}...
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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 ...
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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
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What does one need to know in order to build a software synthesizer?
For someone who has some experience in software development, loves his synths and knows the [very] basics of digital audio representation, what are the first steps he should follow in order to program ...
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389
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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 ...
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2
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3k
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Compare two wave files
I have two .wav audio(stereo) files to be compared. The files may vary in frequency or amplitude or phase shift. How to compute the difference between the wave files in Scilab? The expected output is ...
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0
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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 $\...
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3
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Linearizing digital-analog converters
I'm looking at ways to improve the effective resolution of DACs. It seems that element mismatch (i.e. integral non-linearity) is the main contributor to distortion in modern DACs. Apart from ...
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2
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Exact definition of soft bits in digital receiver
At which stage of digital receiver, the bits are considered as soft bits?
I am unable to find the exact definition. I have rough idea how soft bits are different from hard bits but in terms of signal ...
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53
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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 ...
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2
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864
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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.
...
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2
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304
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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 ...
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1
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61
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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 ...