Questions tagged [digital]

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12 views

$\frac\pi4$-DQPSK receiver query

In the receiver I am using matched filter followed by Gardner timing recovery followed by differential decoding. Is this ok or should I use differential decoding first, then timing recovery, ...
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1answer
37 views

How to quantify phase distortion of a filter specific to a signal?

For example, a EKG input signal is filtered with a known transfer function that does not have linear phase. How do you quantify the phase distortion of the output signal given you have the input? I ...
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2answers
34 views

Length of DFT defines whether I see the harmonic or the dominant frequency

I have a number of signals that are periodic. I use an fft transformation to obtain the dominant frequency of each signal. In order to increase the frequency resolution I zero pad the signal before ...
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1answer
107 views

When calculating SNR, is noise included in signal?

Given a signal $x(t) = s(t) + n(t)$ where $s(t)$ is the desired signal voltage and $n(t)$ is the noise, should the signal to noise ratio of this signal be 20log(xrms/nrms) or 20log(srms/nrms)? i.e., ...
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1answer
40 views

How would I calculate the DC Gain for a Chebyshev Type 2 filter?

I'm currently working on an implementation of a Chebyshev Type 2 filter but the gain is all over the place when playing around with the order and ripple. I haven't found anything online but (as far as ...
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1answer
55 views

Frequency response plot and actual behavior of a digital Butterworth LPF

I have been struggling with a not-too-high (2nd) order digital Butterworth low-pass filter both in SciPy and MatLab (latest of both). The intention is to apply this filter on signals with different ...
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1answer
26 views

What are the definitions of symbols and samples related to PCM?

What are the definitions of symbols and samples related to PCM? For example in binary encoding $\{0,1\}$ with 8 levels of quantization, is one symbol either of $ \ 0 \ or \ 1\ $? or is it the ...
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0answers
53 views

Calculating output of a digital system signal with a filter that introduces delay

We have a step function at the entrance that goes through the following system. The $D(z)$ digital filter introduces a 0.25s delay. Output as $y(kT)$ must be calculated. How can we tackle this problem?...
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15 views

Hardware that Mixes Two Digital Audio Source into One

Is there any off-the-shelf hardware that extracts two digital audio streaming source from two digital audio stream(like USB+SPDIF), mixes, and transmits in single digital stream real-time(if the ...
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2answers
136 views

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. I am interested in applying it in the context of a raw image converter for digital ...
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2answers
71 views

Determine the Given system is linear or nonlinear [closed]

Does the following define a linear or nonlinear system? $$ y(n) -4 y(n)y(2n)=x(n) $$
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1answer
65 views

Design of efficient digital interpolation filter

I came across this paper entitled "Design of Efficient Digital Interpolation Filters and Sigma-Delta Modulator for Audio DAC" where the author oversamples an input frequency, fsig = 1kHz with ratio L =...
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2answers
155 views

Why is the fft magnitude spectra of sine not symmetric?

I thought that the magnitude spectra of real signals was symmetric. When I take the absolute value of the fft of a sine wave the result is not symmetric. There seems to be a peak at the first ...
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1answer
27 views

Understanding Baud Rate

I am new to signal processing and just trying to understand what Baud rate really means. As I understand it, the baud rate speed is really the time that the processor takes when it is ON and when it ...
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1answer
50 views

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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0answers
21 views

Hisotgram equalisation error, help with my code

Iam working on implenting a histogram equalisation of an image without using inbuilt functions specifcally imhist and histeq, I have, for the most part understood what do during a histogram ...
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1answer
40 views

Training Fractionally Spaced Channel Estimator and Equalizer

If you are attempting least squares channel estimation with a fractionally spaced channel estimator, do you want the training sequence to also be fractionally spaced or symbol spaced? It looks like ...
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1answer
265 views

polyBLAMP anti-aliasing in C++

How should 4 point polyBLAMP work. It does not really seem to dampen aliasing. I have an floating point digital audio oscillator that makes a +1.0f jump at the end of every period. My question is, ...
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1answer
110 views

Follow up question regarding: “Complex sampling can break Nyquist?”

I'm having some trouble understanding the sample rate limitations when considering a complex baseband signal. I understand (based on the linked SE questions below), the either (1) physically ...
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1answer
76 views

Using Matlab/Simulink to build Reader based on Software Define Radio

I am a student. I want to learn about Software Defined Radio to build a Reader for Saw tag and simulate them on Matlab/Simulink. These bocks below is what I want to build in Matlab/Simulink I found ...
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1answer
66 views

Modulating ofdm in audio spectrum

Trying to modulate audio with OFDM data. My fs = 44.1khz, i'm using 64 sub-carriers, but i'm only using 5 sub-carriers for data, so my bandwidth is about 3.445khz. I'm not sure how to properly write ...
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1answer
39 views

How do I add more resolution / taps to an analog -> digital filter?

(I am trying to create an IIR audio filter that adds reverb to an initial sample) Say I designed an analog filter to model acoustic attenuation based on the following mathematical model: $$ I = I_0 ...
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1answer
135 views

How does cascading low pass modules affect pass band ripples?

This question 10.21 is taken from GATE IN paper 2015. I don't understand the answer given, I mean why is the frequency response at the pass band frequency considered to be $(1-\delta)$, isn't the ...
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0answers
48 views

Noise Shape Digital Filter

My objective is to build a noise shape filter from a given transfer function (in one case) and from a given PSD (for another case). Checking my precedent questions you can see that this argument is ...
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1answer
587 views

BIBO Stability in Z-domain

I'd really appreciate it if someone could please explain to me the condition for a LTI system to be BIBO stable, in z-domain. I have a background in control, and in linear control for example, if we ...
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1answer
118 views

Digital Derivative

I need to calculate the derivative of a digital signal (sinusoidal). In one of the papers they have mentioned Gilbert transformation is used to calculate the derivative. But I have searched in the net ...
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1answer
477 views

Block Diagram for a difference equation

I have this little doubt regarding how to draw a block diagram representation of a difference equation. Let us implement $ y(n) = ay(n-3) + by(n-2)+cy(n-1) + x(n)$ in block diagram where $a, b, c$ are ...
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1answer
77 views

Why are binary signals drawn continuous in time despite being a digital signal

The conventional definiton of a Digital signal is as follows: A digital signal is a signal that is discrete in time and quantized in amplitude. Almost all resources (textbooks, online sources,etc) ...
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1answer
80 views

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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3answers
87 views

Why must the angle contain $\pi$ for $\cos$ be periodic?

The $\cos(\frac{n}{6})$ is aperiodic, and $\cos(\frac{n\pi}{6})$ is periodic, why? What role does the $\pi$ play? By the way, $n$ must be integer I have seen this question,Why is cos(n/6) aperiodic? ...
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1answer
46 views

Glitches in certain frequencies

I am trying to produce some simple audio waves using WASAPI (Windows Audio Session API). If I skip all the abstractions, my program basically receives a buffer from the API, then writes results of a ...
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2answers
486 views

Amplitude ranges for different bit-depths

In Python, I am using the library Soundfile (which uses the library libsndfile) for digital sound processing. I am working with different sound files coming from different internet databases, and ...
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1answer
122 views

Do integration/differentiation processes work as simple filters?

How these processes are different from simple IIR 1 order filtering, FIR filters in terms of amplitude and phase characteristics?
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1answer
241 views

Second order section digital Bandpass

Does anyone have a ressouce on how I can calculate second order section bandpass coefficients? What I tried allready: I used following formulas from the Book Electronic Circuits (U.Tietze, Ch. ...
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0answers
80 views

Processing digital sound in Audiokit to sound like the 1960's

I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
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1answer
557 views

Bandwidth and Bit rate

I'm kind of confused about digital transmission. Is the bandwidth occupied by some digital signal the Fourier spectrum of the bit (pulse) format,or the one determined by the bit rate?
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1answer
58 views

Subtract a digital filter from another filter

How to subtract a digital filter from another one, if their lengths are different. How to make the length equal.
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0answers
51 views

How does bandwidth affect a modulated digital signal? [duplicate]

I'm currently reading up on how the lower layers of the ISO/OSI model work and I don't quite understand how broadband communication works yet. Suppose I want to send some data from point A to point B ...
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1answer
1k views

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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1answer
334 views

Comparison using cross correlation [closed]

I have 2 stereo audio files to compare. Comparison of files include change in frequency, amplitude, phase shift Code: ...
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3answers
634 views

Applying Nyquist theorem to digital vs. analog audio quality

I had a recent conversation with a friend about analog vs. digital recordings. He shared his opinion that music recorded by analog techniques is of superior sound quality to music recorded by digital ...
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1answer
910 views

How do I calculate the bandwidth from a waveform?

Suppose I have a waveform $x(t)$ where $x(t) = 5 + 30\mathrm{cos}(2000\pi t) + 10\mathrm{cos}(6000\pi t)$ that is to be uniformly sampled for digital transmission, how do I calculate the bandwidth of ...
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1answer
1k views

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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1answer
109 views

Galois Field implementation

Just a quick question. What Galois Field is best if you need a minimum Hamming distance of $n$? I want to design a BCH generator polynomial that is acccording to this Hamming distance.
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2answers
371 views

Why is CDMA not enough for wireless communication?

I have a question about CDMA, FDMA and TDMA. I don't understand why we suffer from limited capacity when we have CDMA. I can understand the limitation in FDMA and TDMA, as time and frequency are ...
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2answers
1k views

Average bit error and symbol energy

How do you draw decision boundaries given this type of 16-qam constellation diagram, I know that there should be no vacant decision boundary but I can't seem to get it.Also, assuming that the correct ...
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2answers
2k views

Analyzing a 16-QAM modulated waveform?

Okay I'm really confused about analyzing the waveform I generated using 16-QAM on MATLAB... I transmitted a $y = [0 \ 1 \ 2 \ 3 \ 4 \ldots \ 15]$ discrete signal (in order) to ...
4
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1answer
26k views

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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1answer
244 views

Why in Overlap Save Method in DSP output length comes out to be same as input length?

I know that when we do convolution of $x(n)$ (Length $L$) and $h(m)$ (Length=$M$), length of output data comes out to be $L+M-1$, but when we break a large sequence of data and find output using ...