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16 votes
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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
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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
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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
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7 votes
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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
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6 votes
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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
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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
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5 votes
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What additional inputs are required to convert dBFS to dB SPL?

Intro As already stated in another answer, this is not an easy task to perform when your setup is complex (comprised of many discrete apparatus pieces), as each piece must be included in the process, ...
ZaellixA's user avatar
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5 votes

How can the quality of a resampling algorithm be measured?

which resampling algorithm should I look at? Polyphase FIR is the way to go. It's actually pretty east to implement from scratch. Much easier than anything FFT related. And you get a lot of knobs to ...
Hilmar's user avatar
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4 votes

What additional inputs are required to convert dBFS to dB SPL?

What is missing needed for the conversion? You need to identify the entire signal chain of your specific setup. This typically includes at least the microphone, pre-amplifier, ADC input stage, and ...
Hilmar's user avatar
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4 votes

Identify abrupt changes in an audio waveform

One domain where continuous (complex) wavelets (CWT) are especially efficient is when you expect the 1D signals to be piecewise regular. When a signal is $C^\alpha$ by pieces, then the $\alpha$ ...
Laurent Duval's user avatar
4 votes

AES encryption of an audio file

Processing any file with an algorithm that is reversible does not change the file in any way. This includes AES encryption and decryption, compressing and decompressing with a generic algorithm like ...
Justme's user avatar
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4 votes

Is adding a mean point between every two points a valid way of upsampling?

Consider this answer as a supplement material to the other answers provided. The upsampling method (mean point between every two points) we are talking about can be represented in the form of ...
SakSath's user avatar
  • 508
4 votes

Identify abrupt changes in an audio waveform

This is more meat-and-potatoes. An oft-used algorithm in audio is the pitch detector, which of course is attempting to accurately determine the instantaneous pitch of a note. The note is a waveform ...
robert bristow-johnson's user avatar
4 votes

How would you compute Fourier transform of a real world signal where the signal keeps getting updated (not a static one)?

The most common method for time frequency analysis is the Short Term Fourier Transform (or STFT for short). Basically the input signal is chopped up into small chunks and then you perform a Fourier ...
Hilmar's user avatar
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4 votes

Resource recommendations to learn audio processing

Julius O. Smith's original book is here: Julius O. Smith III,'s Spectral Audio Signal Processing, which is available online. And for a music background, I'd recommend: Bill Sethares' Tuning, Timbre,...
Peter K.'s user avatar
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4 votes
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Approximating average power with RMS vs RMS^2: Why do we even take the root?

Technically answer 2 is wrong. There is two things going on here. Power vs Energy Roughly speaking: Power is what happens right now and Energy is the time integral over power. Let's say you play a ...
Hilmar's user avatar
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4 votes
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Why do we have negative sign dB in spectrum in audacity?

The audio amplitude of 100% is full scale and that will be 0 dB in comparison to the 100% full scale. That's the amplitude of 1.0 in Audacity. As your audio amplitude is not full scale 100% but less, ...
Justme's user avatar
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4 votes

How to reduce clicking in an audio signal with seek jumps?

Clicking noise on seek is a result of discontinuity in the audio signal. One way of reducing this noise is applying a fade-in effect for few frames post seek.
SakSath's user avatar
  • 508
4 votes

How are the results of a fourier-transform interpolated to cover the entire frequency range?

This is coming from an audio developer (me): Most FFT analyzers in audio plugins are actually very simple. From the 1024-4096 frequency bins that are usually output by the FFT a 2D Path is created. ...
Tobias Hienzsch's user avatar
3 votes

Identify abrupt changes in an audio waveform

I would point on 2 approaches: Parametric Model You build a parametric model for the data and then apply simple change detection methods for the paraments. For instance, you may build a Kalman Filter ...
Royi's user avatar
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3 votes

Is adding a mean point between every two points a valid way of upsampling?

Adding to Dan Boschens excellent answer. Linear interpolation can be a decent solution especially for relatively «smooth» input vectors and small upsampling ratios. It is particularly attractive for ...
Knut Inge's user avatar
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3 votes
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Is adding a mean point between every two points a valid way of upsampling?

I will first summarize the frequency domain view of the interpolation process in general. This will then be used to provide a quantitative result of the distortion when interpolating using linear ...
Dan Boschen's user avatar
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3 votes

Why do we choose 44.1 kHz as recording sampling rate?

The detailed historical reason for the choice of 44.1 khz vs basic Nyquist 40K, etc is given in The Art of Digital Audio by John Watkinson. The excerpt is shown at: http://www1.cs.columbia.edu/~hgs/...
JDiddy's user avatar
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3 votes
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Brrrrrrrrr clanky clank VHS noise

The noise appears to be primarily rectangular pulses that have been bandpass filtered and that a fairly consistent. Here is an outline on how this could be done: Build a model of the noise. I tried a ...
Hilmar's user avatar
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3 votes
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Why does extending the time domain of a signal with a fft introduce a fade out and fade in?

There's two FFT properties at work here. Just for convenience, let $x_n$ be your original signal, and $X_k$ be its DFT (remember that an FFT is just a fast way to implement the discrete Fourier ...
TimWescott's user avatar
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3 votes

IIR/FIR equations for custom frequency response

Here is one way to do this: You can implement a more or less constant slope by cascading high-shelf filters as described in the Audio EQ Cookbook. Create a list of frequencies. Start with a frequency ...
Hilmar's user avatar
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3 votes

Dominant Speaker Selection in a Proximity Cluster of Smartphones with Arbitrary Positions (Not static)

Tricky. Unfortunately phones microphone pickup systems have a LOT of variability. Are there any alternative or indirect ways to estimate the microphone sensitivity of smartphones, or is there any ...
Hilmar's user avatar
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3 votes

Adding mono sources to Ambisonic tracks

The best approach to add a mono source to an ambisonic project really depends on how you want the mono source to behave spatially so you will have to elaborate more on that end. If you want the mono ...
Ahsan Yousaf's user avatar
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3 votes
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What does the frequency of oscillating microphone feedback depend on?

Let's look at a simple block diagram The microphone receives the input sound but also reproduced sound from the loudspeaker. There are two transfer functions in place. One, $A(\omega)$ from the ...
Hilmar's user avatar
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