5 ms·

Making high-fidelity audio sound like it came through the phone (2018)

62 points by jonluca 7y ago

porjo 7y ago
Only the first sound file plays for me!? Using Firefox on desktop and mobile.
- scalableUnicon 7y ago
  From the console: > Media resource https://blog.jonlu.ca/assets/eightkhz_resampled.wav https://blog.jonlu.ca/assets/eightkhz_resampled.wav could not be decoded, error: Error Code: NS_ERROR_DOM_MEDIA_METADATA_ERR (0x806e0006) You can download and play those audios directly: https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencied.wav https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencie... and https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencied.wav https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencie...
- jonluca 7y ago
  Looks like Firefox has issues with playing wav files in HTML5 audio blocks? Not sure actually. Added an mp3 version, should work across browsers now.
lurker2823 7y ago
works here, firefox mac, version 74.
bravoetch 7y ago
Pots chops the frequency off below 300hz and above 3300hz. The simulation should be almost perfect - but it doesn't play.
- luismarques 7y ago
  It also happened to me. It was a Firefox compatibility issue, it played with Chrome.
noizejoy 7y ago
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.
AstralStorm 7y ago
The "too good" part is due to missing compander - very similar or identical to ADPCM compander for better analog lines. ADPCM itself in form of G.711 is still used in older digital phone exchanges.
- philpem 7y ago
  muLAW and aLAW are also worth considering, and give a slightly different effect to ADPCM. The LPC and CVSD series vocoders also have a very unique(ly lossy) sound profile.
  - kragen 7y ago
    A fun thing with LPC is the lpc10 codec built into Sox, making it trivially easy to play with and get a very pleasing distortion.
ThePowerOfFuet 7y ago
> EIGHT_KHZ = 8096 Was this deliberate (vs 8192)?
- Geee 7y ago
  Sampling rates don't need to be powers of two. Usually used sampling rates are e.g. 22050 Hz or 44100 Hz. Maximum signal frequency is half of the sampling rate, and sampling higher frequencies cause aliasing if they're not cut off first. So, 8096 Hz can sample up to to 4048 Hz without aliasing. If their cut-off filter is at 4000Hz, they're on the safe side by sampling a bit more, because the filter isn't perfect.
pjc50 7y ago
AMR is a very "cheap" codec to run, if you want something to sound like it's been over the GSM phone system you could just run it through AMR and back again. Possible patent encumberance but implementations are available in ffmpeg.
gwbas1c 7y ago
The phone system has been mostly digital "forever." Pretty much by the 1990s, even if you had an analog copper phone line, it was digital by the time it made it to the switchboard.
grabbalacious 7y ago
Coming soon, AI prediction of what untelephonic voices would have sounded like! Starting with a detelephonised One Night in Bangkok. https://www.youtube.com/watch?v=rgc_LRjlbTU https://www.youtube.com/watch?v=rgc_LRjlbTU
- paule89 7y ago
  Now that i hear it, i have to admit, he sang through a telephone. Would really love to here an ai prediction version.
- degenerate 7y ago
  Hmm... you may be on to something. In the recommended feed was Toto's "Africa" remastered with AI: https://www.youtube.com/watch?v=AqtBdKP_FPs https://www.youtube.com/watch?v=AqtBdKP_FPs The lyrics do indeed sound crisp as if they were re-recorded, which is impossible. Anyone know if this was really done with AI, and what this field is called (music restoration?) The video is uploaded on some random spanish channel and I can't find any information about it by searching.
dropoutcoder 7y ago
Eventide’s classic DSP4000b has algorithms for phone call sonic simulation. The patches are highly modular (and thus can be inspected for analysis - at least down to the blocks used to construct the patch) and it would be interesting to see how that old school hardware DSP approach compares to the author’s design and implementation.
8bitsrule 7y ago
Every now and then I hear someone call a radio station using a landline phone -in good repair-. The difference in quality (despite the LL bandwidth filtering) compared to mobile-phone callers is often unmistakable. For one thing, the mobile artifactual garbling and missing audio segments stand out. So it's helpful to stipulate what kind of 'phone audio' you're trying to mangle into 'fidelity'.
aiphex 7y ago
I remember landlines sounding quite different from the author. To me they sound much better than cell phones. It sounds as if the other person is actually on the other end of the line. Cell phones sound very artificial with compression artifacts and such.
bcrl 7y ago
He forgot to add in some 60 Hz (or 50 Hz) buzz. Maybe add in a bit of crackling as well since it's now spring and wet around here...

62 points by jonluca 7y ago

porjo 7y ago

Only the first sound file plays for me!? Using Firefox on desktop and mobile.

scalableUnicon 7y ago
From the console: > Media resource https://blog.jonlu.ca/assets/eightkhz_resampled.wav https://blog.jonlu.ca/assets/eightkhz_resampled.wav could not be decoded, error: Error Code: NS_ERROR_DOM_MEDIA_METADATA_ERR (0x806e0006) You can download and play those audios directly: https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencied.wav https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencie... and https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencied.wav https://blog.jonlu.ca/assets/eightkhz_resampled_unfrequencie...
jonluca 7y ago
Looks like Firefox has issues with playing wav files in HTML5 audio blocks? Not sure actually. Added an mp3 version, should work across browsers now.

lurker2823 7y ago

works here, firefox mac, version 74.

bravoetch 7y ago

Pots chops the frequency off below 300hz and above 3300hz. The simulation should be almost perfect - but it doesn't play.

luismarques 7y ago
It also happened to me. It was a Firefox compatibility issue, it played with Chrome.

noizejoy 7y ago

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.

AstralStorm 7y ago

The "too good" part is due to missing compander - very similar or identical to ADPCM compander for better analog lines. ADPCM itself in form of G.711 is still used in older digital phone exchanges.

philpem 7y ago
muLAW and aLAW are also worth considering, and give a slightly different effect to ADPCM. The LPC and CVSD series vocoders also have a very unique(ly lossy) sound profile.
- kragen 7y ago
  A fun thing with LPC is the lpc10 codec built into Sox, making it trivially easy to play with and get a very pleasing distortion.

ThePowerOfFuet 7y ago

> EIGHT_KHZ = 8096 Was this deliberate (vs 8192)?

Geee 7y ago
Sampling rates don't need to be powers of two. Usually used sampling rates are e.g. 22050 Hz or 44100 Hz. Maximum signal frequency is half of the sampling rate, and sampling higher frequencies cause aliasing if they're not cut off first. So, 8096 Hz can sample up to to 4048 Hz without aliasing. If their cut-off filter is at 4000Hz, they're on the safe side by sampling a bit more, because the filter isn't perfect.

pjc50 7y ago

AMR is a very "cheap" codec to run, if you want something to sound like it's been over the GSM phone system you could just run it through AMR and back again. Possible patent encumberance but implementations are available in ffmpeg.

gwbas1c 7y ago

The phone system has been mostly digital "forever." Pretty much by the 1990s, even if you had an analog copper phone line, it was digital by the time it made it to the switchboard.

grabbalacious 7y ago

Coming soon, AI prediction of what untelephonic voices would have sounded like! Starting with a detelephonised One Night in Bangkok. https://www.youtube.com/watch?v=rgc_LRjlbTU https://www.youtube.com/watch?v=rgc_LRjlbTU

paule89 7y ago
Now that i hear it, i have to admit, he sang through a telephone. Would really love to here an ai prediction version.
degenerate 7y ago
Hmm... you may be on to something. In the recommended feed was Toto's "Africa" remastered with AI: https://www.youtube.com/watch?v=AqtBdKP_FPs https://www.youtube.com/watch?v=AqtBdKP_FPs The lyrics do indeed sound crisp as if they were re-recorded, which is impossible. Anyone know if this was really done with AI, and what this field is called (music restoration?) The video is uploaded on some random spanish channel and I can't find any information about it by searching.

dropoutcoder 7y ago

Eventide’s classic DSP4000b has algorithms for phone call sonic simulation. The patches are highly modular (and thus can be inspected for analysis - at least down to the blocks used to construct the patch) and it would be interesting to see how that old school hardware DSP approach compares to the author’s design and implementation.

8bitsrule 7y ago

Every now and then I hear someone call a radio station using a landline phone -in good repair-. The difference in quality (despite the LL bandwidth filtering) compared to mobile-phone callers is often unmistakable. For one thing, the mobile artifactual garbling and missing audio segments stand out. So it's helpful to stipulate what kind of 'phone audio' you're trying to mangle into 'fidelity'.

aiphex 7y ago

I remember landlines sounding quite different from the author. To me they sound much better than cell phones. It sounds as if the other person is actually on the other end of the line. Cell phones sound very artificial with compression artifacts and such.

bcrl 7y ago

He forgot to add in some 60 Hz (or 50 Hz) buzz. Maybe add in a bit of crackling as well since it's now spring and wet around here...