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17 votes
Accepted

What is the audible level for digital audio dB units?

The confusion arises from the fact that the decibel (dB) is a relative unit of measurement, not an absolute one. The dB scale in audio can reference different things depending on the context, which ...
Ahsan Yousaf's user avatar
  • 1,533
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 ...
Marcus Müller's user avatar
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 ...
Dan Boschen's user avatar
  • 52.1k
13 votes
Accepted

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, ...
Hilmar's user avatar
  • 45.4k
13 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 (&...
Hilmar's user avatar
  • 45.4k
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 ...
hobbs's user avatar
  • 227
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 ...
Max's user avatar
  • 2,368
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 ...
Matt L.'s user avatar
  • 90.4k
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 ...
Marcus Müller's user avatar
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 ...
Hilmar's user avatar
  • 45.4k
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; ...
Marcus Müller's user avatar
8 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 ...
Dan Boschen's user avatar
  • 52.1k
8 votes

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 ...
Justme's user avatar
  • 2,303
8 votes

What is the audible level for digital audio dB units?

dB is a logarithmic metric against a well defined reference. There are many different types of "dB" depending on the nature of the signal. For example voltage measurements can be in dBv, dBu,...
Hilmar's user avatar
  • 45.4k
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 ...
Olli Niemitalo's user avatar
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 $$G(z) = \frac{1}{1+z^{-N}}$$ ...
Hilmar's user avatar
  • 45.4k
7 votes
Accepted

Identify abrupt changes in an audio waveform

Synchrosqueezed Wavelet Transform is an option. I have developed a complete algorithm for this task, which scores 100% train accuracy and 86% test accuracy with 0.05 sec tolerance, without machine ...
OverLordGoldDragon's user avatar
7 votes

What is the audible level for digital audio dB units?

There's one broad standard that is only strictly followed in certain conditions. The cinema standard is a 500-2000 Hz band-limited pink noise signal with digital RMS amplitude of -20 dB full scale ...
user71659's user avatar
  • 171
6 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, ...
A_A's user avatar
  • 10.7k
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 ...
rahulb's user avatar
  • 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 ...
MBaz's user avatar
  • 15.3k
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 ...
ScienceGeyser's user avatar
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:
Richard Lyons's user avatar
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 ...
leftaroundabout's user avatar
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 ...
Hilmar's user avatar
  • 45.4k
6 votes
Accepted

Resource recommendations to learn audio processing

Here are a few books that might be helpful: The Scientist and Engineer's Guide to Digital Signal Processing A good resource for you to get familiarized with DSP in general. Understanding Digital ...
Ahsan Yousaf's user avatar
  • 1,533
6 votes

Algorithm to "serialize" impulse responses

You convolve the two impulse responses. $$h_{cascade}[n] = h_1[n]*h_2[n]$$
Hilmar's user avatar
  • 45.4k
5 votes
Accepted

Compute group delay of an audio file from STFT

The problem is that the stft function is splitting the signal up into different windows. That means that the signal from time $n$ to $n+N_{w}-1$ is multiplied by $$ ...
Peter K.'s user avatar
  • 25.8k
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 ...
Cedron Dawg's user avatar
  • 7,570

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