15
votes
Accepted
Multi-channel audio upsampling interpolation
Does cubic interpolation (or any other) have any advantages over linear for the specific case of audio?
You'd use neither for audio. The reason is simple: The signal models you typically assume for ...
- 28k
13
votes
Accepted
Why does the derivative of an audio file act like a high-pass filter?
Yes both the time derivative $d/dt$ and the discrete time difference $x[n]-x[n-1]$ are indeed "high pass filters": consider the extreme case of the lowest frequency, which is DC or a ...
- 43.5k
13
votes
Use of "Audacity"for teaching undergraduate labs of digital signal processing for electrical engineering students?
That depends on a lot on your learning goals for the lessons and the lab.
Audacity is primarily an audio editor designed for music recording and production. It has a lot of signal processing baked in, ...
- 37.9k
12
votes
Why does the derivative of an audio file act like a high-pass filter?
$\frac{d}{dt}\sin(\omega t) = \omega\cos(\omega t)$ as a consequence of the chain rule. $\sin$ and $\cos$ are the same apart from the phase shift, so the derivative is a filter with a response that's ...
- 228
12
votes
What is the maximum possible frequency of human voice/speech(That can be generated through human vocal cords)?
The fundamental speaking frequency of humans can reach up to around 1kHz, although higher values than, say, 500Hz usually appear only while singing.
The harmonics and non-tonal parts of speech can ...
- 2,029
12
votes
What is the maximum possible frequency of human voice/speech(That can be generated through human vocal cords)?
Especially What is the maximum value of frequency that human speech can have?
This depends on how exactly you define it. Fricatives ("s","f","sh" ...) and plosives (&...
- 37.9k
11
votes
Accepted
How do impulse response guitar amp simulators work?
When talking about modeling, there are two things that usually get modeled: 1. the guitar amp, and 2. the speaker cabinet. Only the latter is modeled by an impulse response, which means that the ...
- 84.3k
10
votes
Accepted
Can Principal Component Analysis (PCA) Solve the Cocktail Party Problem?
The Cocktail Party Problem is a Blind Source Separation (BSS) problem.
Given a linear mixture of signals:
$$ \boldsymbol{y} \left[ n \right] = A \boldsymbol{x} \left[ n \right] $$
We're trying to ...
- 47.4k
9
votes
Accepted
Finding Reference Audio Signal in Test Audio Signal and Cropping Accordingly
If you have a reference signal you want to find in a different signal then your model matches almost perfectly (Up to the environment the signal to be found is in) to Matched Filter.
So basically you ...
- 47.4k
9
votes
Accepted
Noise reduction using multiple recordings of the same signal
Is it possible to reconstruct the original pure signal?
No, that is information-theoretical impossible. Also, that signal doesn't exist, probably, to begin with ;)
However, you can definitely ...
- 28k
9
votes
Accepted
Am I using FIR filters correctly for audio filtering?
Do those types of filters usually use FIR filters?
No. Most "standard" filter types are IIR filters. For the design of single biquads Robert Bristow Johnson's EQ Cookup is an excellent ...
- 37.9k
8
votes
How do impulse response guitar amp simulators work?
If you're an EE student, you will have encountered the term LTI System (or you certainly will soon enough!): A system that, no matter the absolute time, outputs, given the same input, the same output; ...
- 28k
8
votes
Accepted
Is adding a mean point between every two points a valid way of upsampling?
That is called linear interpolation.
Technically it is a valid method and just one method among several methods.
It might not be a very good way as it is very simple and will not have very good ...
- 1,866
7
votes
Multi-channel audio upsampling interpolation
Audio signals
An audio special-purpose analog-to-digital converter (ADC) normally has an internal or external analog low-pass filter and samples the analog filtered signal at a multiple of the target ...
- 12.9k
7
votes
Realtime sample rate conversion
The consumption time and transmission time is identical: One second of data is still one second of data regardless of sampling time. However if the transmitter and receiver are not synchronized then ...
- 43.5k
7
votes
Remove single echo from audio signal
That's a difficult problem. This appears to be a single delay of $N=5325$ samples. The transfer function of this filter would be
$$H(z) = 1+z^{-N}$$
The inverse is simply
$$G(z) = \frac{1}{1+z^{-N}}$$
...
- 37.9k
6
votes
Accepted
Modelling Unwanted Signal in a LMS Adaptive Filter
The LMS and many of the variants of Adaptive Filters (In the Linear System context) work in the following settings (Intuitive):
You have access to 2 signals.
One signal is the result of the other one ...
- 47.4k
6
votes
Understanding the frequency domain
In simple terms, Image shown here speaks for itself.
Before speaking about Fourier Transform black magic lets understand idea behind it. Work of the Mathematician Joseph Fourier demonstrated any ...
- 181
6
votes
Accepted
Audio Processing Glossary
My own interpretation:
Sound: a mechanical wave that propagates through the air or water.
Audio: sound in the 20 Hz to 20 kHz range; in other words, sound that is (at least in theory) audible to a ...
- 14.4k
6
votes
What is the principle behind my voice echoing perpetually when 2 separate devices are used for playback and recording on video calls?
On each individual device, the speaker output can get subtracted from the microphone before it gets sent to other locations. This prevents others from hearing themselves through your microphone. When ...
- 261
6
votes
Why does the derivative of an audio file act like a high-pass filter?
@OverLordGoldDragon Your x(n)−x(n−1) expression is called a "first-difference discrete differentiator". Its frequency magnitude response is the blue dashed curve below:
- 5,289
6
votes
Am I using FIR filters correctly for audio filtering?
As far as I can tell it works but much of the higher frequencies seem to get through
This can have multiple reasons. The obvious one is that a FIR with a window length of $t$ can have at most a ...
- 1,239
6
votes
What causes the mismatch between a calculated sine wave and a measured sine wave?
Sorry, can't reproduce. I did the same thing with the dither turned off and found a perfect match between the generated sine wave and the calculated values. Even with a dither turned on things were ...
- 37.9k
5
votes
Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?
The trigonometric functions "do not know" what a Hertz is and they do not care either. The only thing they know is that a full circle is $2 \pi$ radians. Whether this circle concludes in days, hours, ...
- 10.4k
5
votes
Multi-channel audio upsampling interpolation
Sorry MM, I'm agreeing with Havakok on this one: A time domain interpolation solution should do just as well, practically speaking, and be significantly cheaper in terms of computation. (Assuming ...
- 6,982
5
votes
Realtime sample rate conversion
The usual answer is that that you must convert by integer ratios, therefore 44.1 kHz to 48 kHz requires integer up and down conversions. Since this has been repeated in DSP text books for at least the ...
- 761
5
votes
Accepted
Actual Sample Rate slightly off?
If you calculate the error, 44102/44100 is only about 45 parts per million. That is well within operating tolerance of many crystal oscillators used in consumer equipment to generate audio sampling ...
- 1,866
5
votes
Accepted
Analysis of two audio signal by autocorrelation
The "MCU" plot looks like there was a DC component in the signal, that ran through the autocorrelation function. The triangle is typical for this. Try highpass filtering before the analysis.
...
- 2,029
5
votes
Confusion regarding frequency spectrum in MATLAB
The $fs$ you are getting from audioread is the sampling rate and has nothing to do with the actual frequency content of the signal you are analyzing other than the maximum frequency would be $fs/2$.
...
- 2,771
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