Knut Inge
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What do Signal Processing experts do, in essence?
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6 votes

Signal processing is often at a junction between applied statistics, numerical math, analytic math, physics, electronics and programming. Big data, visualization and machine learning may be ...

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Issue understanding implementing digital filters in practice
4 votes

A real operation on a complex input can be implemented as two real operations in parallelle, rather than a truely complex operation. An example would be multiplying a complex number with a real number....

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How does Matlab's filter() return a signal of the same size?
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4 votes

A full convolution is Lx + Lh - 1 samples long. This can be truncated to two meaningful subsets, as documented in conv(). filter() is generally an iir filter, and will have an infinite output. Thus it ...

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Identify which of the three signals is closest to a sinusoidal curve
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Is it sufficient to identify the «most sinuoidal» of those 3, or would you also want linear projections of those (consistent with a IMU sensor tilted vs the plane of motion)? A simple solution might ...

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How to tell if a high-res flac file has been upsampled from a CD-quality file?
4 votes

A file that where the signal path has been kept 96kHz from microphone to loudspeaker has the potential for signal components up to 48kHz. Most physical sound sources, rooms and microphones will roll ...

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draw sines at higher freq
4 votes

For a given «visual accuracy», you need to sample the sine at a sufficient number of time-steps per period. At some point the display pixel density will be to low to render a sine accurately. For ...

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Trouble understanding how image digitization works
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3 votes

The image in a digital camera/sensor is represented as a matrix of discrete numbers, as you say. The (spatial) digitization consists of an array of sensels each with some non-zero area that are ...

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What are typical audio waveform synchronization methods?
3 votes

Crosscorrelation, find the maximum, and use that to offset one file vs the other.

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Function implementation in Matlab
3 votes

I believe this will do: x=a.^((0:40)+(0:40)’)

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Why not use 'higher power of 2' radix FFT?
2 votes

I believe that there are diminishing returns beyond radix 4. While arithmetic complexity probably continues to decrease (?) that decrease is too slow to compensate for «other factors». Finding new ...

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How to interpret values of the autocorrelation sequence?
2 votes

I think of auto correlation as «self similarity». In practice: line the signal up with a copy of itself. Step the copy one sample at a time. Multiply corresponding pair of samples from the two and ...

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Where is truncated-SVD image compression actually used?
2 votes

It has been said that the DCT reasonably closely matches the KLT for a representative set of images. KLT is essentially the same as PCA, I believe and SVD is only a different way to compute the same? ...

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Why is Complex Baseband better than Real Representation?
2 votes

The bandwidth part is somewhat trivial. If you do eg 1000 complex samples/symbols/... per second, that represents twice as much information as 1000 real samples/symbols/... per second. The convenience ...

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Why we sometime use some values of fixed point and floating point both in dsp?
2 votes

The transition from text-book math to product code can be a lot simpler with floating point than with fixed point. The latter requires some specialized knowledge that, while not rocket science, may be ...

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Is multiplying by Ts faster than dividing by fs?
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2 votes

Generally, it makes sense to ensure that your code is logically correct, that it is numerically well-behaved, intuitive to read and tested. That is hard enough. Only when you observe that some ...

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Downsampling vs. ADC with lower Sample-Rate
2 votes

With downsampling you have complete control over the process and it comes down to what compromise of processing complexity, delay, aliasing and loss of passband you can accept. With a lower rate A/D ...

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An intuitive explanation as to why a matched filter is time reversed?
2 votes

Picture the transmitted signal as a «signature». You want to find some process that maximize the probability of detecting that signature even when there is noise. What do you do to find some signature ...

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changing angular data to remove the sharp changes
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2 votes

Could you do something like: (1/2)*unwrap(2*theta)? edit: More concretely, I think about this. I would think that the numerical example and asserts are sufficient to see if it can be adapted to your ...

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Sampling: question regarding sampling period
2 votes

You are sampling a sine function that should return zero, but because of finite precision argument into that sine function, the error tends to increase towards the right?

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Loosing information in decimation - without aliasing
2 votes

The whole point of aliasing and the sampling theoreme is that you cannot (generally) know what an aliased signal was, as you cannot represent infinite bandwidth using finite infornation. If you have ...

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Formula to calculate/generate gaussian noise?
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2 votes

In MATLAB, you can do it that way: >> nsamples = 1e9; mu = 11; sigma_squared = 18; x = mu + sqrt(sigma_squared)*randn(nsamples,1); mean(x) ans = 10.9998 var(x) ans = 17.9994 Edit: ...

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why only impulse signal for convolution?
2 votes

Unit impulse is a nice pedagogic construct in order to understand how we can measure and execute a LTI system. The stimuli of choice when characterizing real-world near-LTI systems is often not a ...

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Volume increase of a magnitude spectrogram
2 votes

Yes. The DFT/FFT is a linear function of the audio waveform, and taking the absolute value does not break the proportional relationship. -k

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Computationally Efficient Ways of Determining If Pixels Are Clumped
2 votes

Median filter? Use it to «thin out» bad pixels that appear locally sparse (for some definition of «local»). Then count the remaining bad pixels. Or convolve with a large-ish 2d kernel (eg flat ...

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Help needed for Matlab Spectrogram plot
2 votes

spectrogram(x, N, N/2, N, fs, ‘yaxis’) is my default one liner. To get rid of the non-SI time units, I think you need to do [S, F, T] = spectrogram(x, N, N/2, N, fs); pcolor(S, F, 20*log10(abs(S))); ...

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Is there any science to explain the "weight" or "3D depth" of hardware audio vs software that some people claim?
2 votes

Most Audiophile stuff*) has a credibility on par with homeopathy. It might work - provided that some sales-oriented cable manufacturer single handedly revolutionized science. But in all likelihood it ...

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Convert a list of values and times into an audio file
1 votes

An asynchronous resampler should do the trick. Basically positioning a continuous time windowed sinc at the desired (uniform) output time instants, sampling it by a neighbourhood of input time ...

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Difference H264 and H265 algorithms
1 votes

The high-level picture is that h.265 allows lower bandwidth at the same visual quality (or more visual quality at the same bandwidth). It achieves this by using more advanced techniques that require ...

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How plausible is undetectable (covert) radio communication?
1 votes

I guess it depends on the channel. If you can occupy a large relative bandwidth and use really long spread sequences without fear of fading or doppler making the signal unreadable, then you can go ...

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Why does convolution reverb work?
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1 votes

A room consists of many hard surfaces. When you generate a wideband click sound in that room («perturbations about the mean pressure»), those waves will travel into the room, be reflected by surfaces, ...

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