6 ms·

As an alternative to Fourier Transformations, one can also use classic audio filters applying high pass and low pass filters, or a band pass filter (or similar

As an alternative to Fourier Transformations, one can also use classic audio filters applying high pass and low pass filters, or a band pass filter (or similar EQ settings with the aim of boosting the signal around 2KHz, and silencing the signal below about 1KHz and above 4KHz. Caveat: I love doing this in a musical context for a specific part (e.g. vocals or drums or guitar) amongst many, and so I tend to fine-tune the exact frequencies in the context of the other parts so they evoke the memory and emotion of a telephone line more than the exact specifications of a POTS[0] line. [0] POTS = plain old telephone service or plain ordinary telephone service

TheOtherHobbes 7y ago
Simple audio filters will have a fairly gentle roll-off above/below their cutoff. Zeroing FFT bins will brickwall the frequencies that are being removed. They sound subtly different. You can build brickwall emulated filters but they're never perfect, and you have to make a tradeoff between high latency and phase distortions/ringing. Personally I like FFT filtering because subjectively it can sound more "vintage" than filter emulations. It's interesting that this is being used to emulate a mechanical limitation - the limited mechanical frequency range of carbon granule microphones and small speakers. Modern speakers have an amazing ability to record and generate a much wider frequency range from much smaller transducers.
- unlinked_dll 7y ago
  There's a great plugin called "Engineer's Filter" which can do some more classical filtering (Chebychev, Cauer) to really get that vintage sound. Zero'ing an FFT bin is neat for the phase artifacts/time domain aliasing. https://www.kvraudio.com/product/engineersfilter_by_rs_met https://www.kvraudio.com/product/engineersfilter_by_rs_met
- kragen 7y ago
  Depends on what you mean by “simple”. If you want a circuit you can build with one op-amp and some passives from some coefficients you have memorized, without simulating it first, you're probably going to go with a second order Butterworth or something and get that gentle roll-off you're talking about. But if you're using SciPy or Octave, you can synthesize a tenth-order elliptic IIR filter and apply it in three lines of code (to me that's “simple”), and it'll be pretty brickwallish. It's true that you get more latency as you move to lower-phase-distortion IIR and zero-phase-distortion FIR filters. But that's not a reason to filter in the Fourier domain instead, because in the Fourier domain you get all the latency. If you can pay all the latency that way, you can eliminate the IIR phase distortion with filtfilt. Similarly, ringing in your IIR filter is not a reason to go to the Fourier domain and start bashing bins to zero, because you get all the ringing that way: the Fourier-domain difference between your original signal and the transformed signal is a Fourier-domain impulse, whose inverse Fourier transform is a sinusoid that fills eternity. Also, just to be clear, and you probably already know this, but ringing and phase distortion are two totally different phenomena.
philpem 7y ago
Good shout. That sounds roughly like a pre-emphasis filter, as is commonly used in mobile radio applications. Rolloff for those on the bottom end is usually around 300Hz (the CTCSS block filter), and around 3kHz (channel bandwidth limit) on the top end. Often those filters were implemented as analog designs, so a FIR with -10dB/dec rolloff would be closer to real-life. An FFT filter can be configured to act like a brickwall, which will sound subtly different.
iagooar 7y ago
> Caveat: I love doing this in a musical context for a specific part (e.g. vocals or drums or guitar) amongst many, and so I tend to fine-tune the exact frequencies in the context of the other parts so they evoke the memory and emotion of a telephone line more than the exact specifications of a POTS[0] line. That is super interesting, now I'm intrigued. What's exactly the difference between the memory and emotion and the real sound of a telephone call? Do you have this kind of approach with other sounds as well?
- djaque 7y ago
  Just a guess, but I imagine he's talking about how we've been taught certain context clues from media that aren't exactly accurate. For instance, that bubbly low passed sound when a character is underwater or the crunchy punch sounds that aren't what a real fight sounds like.
- noizejoy 7y ago
  > What's exactly the difference between the memory and emotion and the real sound of a telephone call? When mixing a song, one of the many things to consider is, if a specific part is intended to blend in with other parts, effectively creating a merged sound vs. which parts are supposed to stand out. Since I use the telephone effect to make a part stand out a little extra, it becomes more important for it to be focused in a frequency range not already crowded with other parts. And therefore it’s less important if the highlighted frequency is a precise match to a telephone line. > Do you have this kind of approach with other sounds as well? Yes, very much so. For example in a very full pop or electronica mix with many parts, a grand piano track can overwhelm and muddy too many frequencies used by other instruments - for example the very low end (left side) of a good grand piano occupies the same frequencies as a bass guitar, upright bass or synth bass. So typically I would make the piano track be less prominent in the competing range, by lowering the EQ settings in that range. And I might increase the EQ settings for the frequency range where I want the piano track to me most obvious to the listener. — There are fancier versions of that mixing technique via variable multi-band compression with side chaining, but that might go beyond the interest level of the HN audience and take this thread too far off-topic :-)

TheOtherHobbes 7y ago

Simple audio filters will have a fairly gentle roll-off above/below their cutoff. Zeroing FFT bins will brickwall the frequencies that are being removed. They sound subtly different. You can build brickwall emulated filters but they're never perfect, and you have to make a tradeoff between high latency and phase distortions/ringing. Personally I like FFT filtering because subjectively it can sound more "vintage" than filter emulations. It's interesting that this is being used to emulate a mechanical limitation - the limited mechanical frequency range of carbon granule microphones and small speakers. Modern speakers have an amazing ability to record and generate a much wider frequency range from much smaller transducers.

unlinked_dll 7y ago
There's a great plugin called "Engineer's Filter" which can do some more classical filtering (Chebychev, Cauer) to really get that vintage sound. Zero'ing an FFT bin is neat for the phase artifacts/time domain aliasing. https://www.kvraudio.com/product/engineersfilter_by_rs_met https://www.kvraudio.com/product/engineersfilter_by_rs_met
kragen 7y ago
Depends on what you mean by “simple”. If you want a circuit you can build with one op-amp and some passives from some coefficients you have memorized, without simulating it first, you're probably going to go with a second order Butterworth or something and get that gentle roll-off you're talking about. But if you're using SciPy or Octave, you can synthesize a tenth-order elliptic IIR filter and apply it in three lines of code (to me that's “simple”), and it'll be pretty brickwallish. It's true that you get more latency as you move to lower-phase-distortion IIR and zero-phase-distortion FIR filters. But that's not a reason to filter in the Fourier domain instead, because in the Fourier domain you get all the latency. If you can pay all the latency that way, you can eliminate the IIR phase distortion with filtfilt. Similarly, ringing in your IIR filter is not a reason to go to the Fourier domain and start bashing bins to zero, because you get all the ringing that way: the Fourier-domain difference between your original signal and the transformed signal is a Fourier-domain impulse, whose inverse Fourier transform is a sinusoid that fills eternity. Also, just to be clear, and you probably already know this, but ringing and phase distortion are two totally different phenomena.

philpem 7y ago

Good shout. That sounds roughly like a pre-emphasis filter, as is commonly used in mobile radio applications. Rolloff for those on the bottom end is usually around 300Hz (the CTCSS block filter), and around 3kHz (channel bandwidth limit) on the top end. Often those filters were implemented as analog designs, so a FIR with -10dB/dec rolloff would be closer to real-life. An FFT filter can be configured to act like a brickwall, which will sound subtly different.

iagooar 7y ago

> Caveat: I love doing this in a musical context for a specific part (e.g. vocals or drums or guitar) amongst many, and so I tend to fine-tune the exact frequencies in the context of the other parts so they evoke the memory and emotion of a telephone line more than the exact specifications of a POTS[0] line. That is super interesting, now I'm intrigued. What's exactly the difference between the memory and emotion and the real sound of a telephone call? Do you have this kind of approach with other sounds as well?

djaque 7y ago
Just a guess, but I imagine he's talking about how we've been taught certain context clues from media that aren't exactly accurate. For instance, that bubbly low passed sound when a character is underwater or the crunchy punch sounds that aren't what a real fight sounds like.
noizejoy 7y ago
> What's exactly the difference between the memory and emotion and the real sound of a telephone call? When mixing a song, one of the many things to consider is, if a specific part is intended to blend in with other parts, effectively creating a merged sound vs. which parts are supposed to stand out. Since I use the telephone effect to make a part stand out a little extra, it becomes more important for it to be focused in a frequency range not already crowded with other parts. And therefore it’s less important if the highlighted frequency is a precise match to a telephone line. > Do you have this kind of approach with other sounds as well? Yes, very much so. For example in a very full pop or electronica mix with many parts, a grand piano track can overwhelm and muddy too many frequencies used by other instruments - for example the very low end (left side) of a good grand piano occupies the same frequencies as a bass guitar, upright bass or synth bass. So typically I would make the piano track be less prominent in the competing range, by lowering the EQ settings in that range. And I might increase the EQ settings for the frequency range where I want the piano track to me most obvious to the listener. — There are fancier versions of that mixing technique via variable multi-band compression with side chaining, but that might go beyond the interest level of the HN audience and take this thread too far off-topic :-)