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96 votes
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If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

The sampling rate of a real signal needs to be greater than twice the signal bandwidth. Audio practically starts at 0 Hz, so the highest frequency present in audio recorded at 44.1 kHz is 22.05 kHz (...
endolith's user avatar
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73 votes

If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

44,100 was chosen by Sony because it is the product of the squares of the first four prime numbers. This makes it divisible by many other whole numbers, which is a useful property in digital sampling. ...
dotancohen's user avatar
47 votes
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Why can so little digital information be stored on a cassette tape?

I had heard that tape is still the best medium for storing large amounts of data. well, "best" is always a reduction to a single set of optimization parameters (e.g. cost per bit, ...
Marcus Müller's user avatar
19 votes

Compensating Loudspeaker frequency response in an audio signal

Yes, you can do this with a Wiener Filter which uses the Wiener-Hopf equation to determine the least squared solution to the filter that would compensate for your channel, using the known transmit and ...
Dan Boschen's user avatar
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19 votes

Why isn't silence just zeroes in an audio file, and how can I make it?

Full silence is just zeroes. You likely generated the silence to an audio track which is in some other format than 16-bit PCM, like float or 24-bit or 32-bit PCM. When you exported it as 16-bit wave ...
Justme's user avatar
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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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15 votes

How to deal with low fundamental when using AMDF for pitch extraction?

This is what we call in the pitch-detection biz, the "octave problem". First of all, I would change the AMDF to ASDF. And I would not reduce the window size as the lag increases. (Also, I ...
robert bristow-johnson's user avatar
15 votes

Where do RBJ's analog shelving filters come from?

If you remove (for the time being) that leading factor $A$ as a constant gain factor: $$H(s)=\frac{s^2+\left(\frac{\sqrt{A}}{Q}\right)s + A}{As^2 + \left(\frac{\sqrt{A}}{Q}\right)s + 1}$$ what you get ...
robert bristow-johnson's user avatar
15 votes

Where are these aliasing artifacts coming from in transmitting audio? (Frequency reflection, duplication)

To perhaps help future readers, here is a graphical explanation of the two behaviors seen: harmonic distortion and aliasing / frequency fold-over: Harmonic distortion can result from any kind of non-...
jpa's user avatar
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13 votes
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Band limited wavetables

It looks like you are mixing up two techniques: MinBLEP and band-limited wavetables. Band-limited wavetables Let's assume you have a 256 sample long array with a waveform of interest - for example a ...
pichenettes's user avatar
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13 votes

If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

The Nyquist rate is above twice the bandlimit of a baseband signal that you want to capture without ambiguity (e.g. aliasing). Sample at a lower rate than twice 20kHz, and you won't be able to tell ...
hotpaw2's user avatar
  • 35.3k
13 votes

How do I explain a complex exponential intuitively?

From a graphical point-of view, it is an infinite spring, whose distance between adjacent coils reflects the frequency of the complex exponential: If you have a 1D time x-axis, you may be used to ...
Laurent Duval'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
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13 votes
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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
  • 44.1k
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
  • 44.1k
12 votes

Does it make sense to have complex numbers representing real-world audio signals?

So it seems like real-world (discrete) audio signal might have complex values when being represented digitally, No, you misunderstood that. The discrete audio time signal doesn't have non-real ...
Marcus Müller's user avatar
12 votes

How can I upsample 22 kHz speech audio recording to 44 kHz, maybe using AI?

What you want to do is called "Super Resolution" in the field of imaging. This is an ill posed problem. Hence in order to solve it you need some prior / model about your audio data. For ...
Mark's user avatar
  • 357
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
  • 2,215
12 votes
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Can humans hear Hilbert transform in audio?

Can humans hear Hilbert transform in audio? Generally no. The human auditory system is fairly insensitive to monaural phase shifts. "Monaural" means "same phase shift for both ears&...
Hilmar's user avatar
  • 44.1k
11 votes

Digital Distortion effect algorithm

Thanks to the plot in Olli Niemitalo's answer I got convinced that the formula given in the book has a sign error. The non-linearity used for fuzz or distortion is always some type of smoothed ...
Matt L.'s user avatar
  • 89.5k
11 votes

Can you turn a square wave into a sine wave using a low pass filter?

To be able to analyze what a low pass filter does first you would need to understand what a Fourier transform is, hence some theory first. The Fourier transform essentially represents the time-domain ...
Television's user avatar
11 votes

Show others how I hear myself

The most practical attempt that I am aware of is by Won and Berger (2005). They simultaneously recorded vocalizations at the mouth with a microphone and on the skull with a homemade vibrometer. They ...
StrongBad's user avatar
  • 231
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
  • 89.5k
11 votes

CD quality bitrate

What is missing from your calculations is that it requires a multiplier of two because CD quality audio is two channels for stereo. 2 * 44.1 kHz * 16 = 1411.2 kbps The answer to your second question ...
Justme's user avatar
  • 2,303
11 votes

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

Let's get some basics out of the way first. An audio format is basically defined by three different things: sample rate, bit depth and compression type. The compression type can be either lossless (...
Hilmar's user avatar
  • 44.1k
11 votes
Accepted

Where are these aliasing artifacts coming from in transmitting audio? (Frequency reflection, duplication)

This behavior is likely caused by clipping (as nicely illustrated by @OverLordGoldDragon in the comments). Where exactly clipping is occuring is harder to tell, but a simple remedy is to lower the ...
MBaz's user avatar
  • 15.3k
10 votes

If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

Basically, twice the bandwidth is a common requirement for signal sampling, thus $2\times 20 = 40$ kHz is a minimum. Then, a little more is useful to cope with imperfect filtering and quantization. ...
Laurent Duval's user avatar
10 votes
Accepted

RMS values above 0 dbFS?

0 dBFS refers to a full-scale sine wave or more rarely a full-scale square wave. ITU-T telephony Recs. P.381 and P.382 and reportedly the paywalled digital audio device measurement standards AES17-...
Olli Niemitalo's user avatar
10 votes
Accepted

Show others how I hear myself

It is not impossible but it is not going to be a walk in the park too. What you would be trying to do is to add to the voice signal, those vibrations that are delivered to the ear via the bones and ...
A_A's user avatar
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