Videos don't do well on Hacker News, but I encourage people to at least watch the first couple minutes of this one. The oscilloscope visual overlay is interesting and the editing is really good.
Also, given the topic (audio equalizers) there's no way it could have been a blog post.
I would hope mostly what doesn't do well is useless titles - like one word, or a pithy joke that makes sense only in retrospect. Unfortunately there is also that guidelines which discourages doing better.
I think the title on HN is misleading.
Sadly I’m not sure what’s the short and clearest title :(
The video is very nicely made and it focus on sound systems / boomboxes frequency response and behavior of included filtering modes.
So when talking about EQs with “all forms” in mind, you should consider:
- EQ is merely combination of one or more filters.
- There are many common filter designs, the video isn’t about that, it also doesn’t mentioned Low-pass/High-pass/band-pass/bell or common structures, but it is only showing them.
- Filters made and behave differently
- Most filters (including the ones in the video) are done on the time domain (vs spectral one)
- Phase, this is the biggest missing piece in the video imho. Naive filtering is “smearing” the signal to achieve the different tone balance. By doing so, they also most likely change the phase (unless using linear phase filters)
You have a slight misunderstanding here about the phase shift bit.
Technically filters don't cause phase shift, phase shift causes filters.
That means in the analog world all realtime filters come with phase shift, if they didn't you wouldn't have a filter.
In the digital world there is a thing called linear-phase mode where clever people found ways of building filters that shift the phase of the whole signal the same way (involves running the signal through forwarwards and backwards through special filters). If you now have an audio program that deals correctly with the latency introduced by each effect, you can shift the rest by the same amount meaning you bought yourself a linear phase response with added latency (a small delay) on everything. Remember: Humans can't really hear phase smears unless they happen in relation to another thing that is in phase. This only happens when you mix two signals. The result of that mix is yet again a filter (e.g. comb filters).
With filters everything is a trade off all the time, but again phase shift is not a just a bad side effect of a filter, it is the mechanism by which it does the work. And sometimes the phase shift introduced cannot be heard.
And also linear phase isn't for free, besides the latency, it also adds pre-ringing, which especially in the lower frequencies can be very audible and annoying.
> Phase, this is the biggest missing piece in the video imho.
It's why I like full-range speakers—no crossover. At most you might throw a cap on a super-tweeter (yes, even so-called full range need help on the top and bottom ends of the spectrum—so super-tweeters + sub-woofers). Complicated crossovers destroy the phase.
Without crossovers/EQ the more stripped-down full-range setup preserves what the Hi-Fi nerds call "soundstage". (A good full-range setup: imagine the "soundstage" you get from wearing headphones—but without wearing headphones.)
Those seeking a full-range setup (especially with a tube amplifier) are kind of adjusting for EQ simply by the type of speakers/drivers and placement. (And to be sure they're not going for a flat, studio-monitor response but what sounds good).
OK, so based on this video, I've turned on EasyEffects, added an Equalizer, set the input source to "Easy Effects Source" in the system settings, started playing pink noise, then tried to adjust the dials until the input looked closer to a horizontal line (it's very chaotic so that is hard to do).
As some sections of the video highlight it ended up needing to dip the mids and boots the lows and highs. This is my end result (so far, I'm probably going to tweak this continuously):
Preamp: -1 db
Filter 1: ON PK Fc 27.782795 Hz Gain -3.36 dB Q 1.7848856
Filter 2: ON PK Fc 49.40557 Hz Gain 1.09 dB Q 1.7848856
Filter 3: ON PK Fc 87.85691 Hz Gain 5.04 dB Q 1.7848856
Filter 4: ON PK Fc 156.23413 Hz Gain 6.43 dB Q 1.7848856
Filter 5: ON PK Fc 277.82794 Hz Gain 3.76 dB Q 1.7848856
Filter 6: ON PK Fc 494.0557 Hz Gain -1.19 dB Q 1.7848856
Filter 7: ON PK Fc 878.5691 Hz Gain -5.54 dB Q 1.7848856
Filter 8: ON PK Fc 1562.3413 Hz Gain -3.96 dB Q 1.7848856
Filter 9: ON PK Fc 2778.2793 Hz Gain -2.97 dB Q 1.7848856
Filter 10: ON PK Fc 4940.557 Hz Gain 0.3 dB Q 1.7848856
Filter 11: ON PK Fc 8785.69 Hz Gain 4.35 dB Q 1.7848856
Filter 12: ON PK Fc 15623.413 Hz Gain 8.7 dB Q 1.7848856
There's also an Output amp of -2.5dB that does not appear in the export for some reason.
TUXEDO InfinityBook Pro 15 - Gen10 - AMD Laptop Speakers.
It does make a positive difference (why don't they teach us this in school? ;) ). Please note I am music illiterate.
EDIT: it looks like changing the input source has no actual effect, just need to make sure the internal microphone's volume is not muted, and have to keep system setting window open while doing this, maybe the microphone is no longer on when the window is closed?. Linux is weird sometimes.
But the reason this is not widely talked about is two fold:
First, because those measurements are heavily room dependent and they sound worse than treating a room. Because the room affects the sound in ways that EQ can’t really fix anyway.
Second, because the goal of mixing and mastering is making music sound good in a big variety of speakers and environments.
In the 70s/80s every home stereo system -- racks of stereo equipment stacked a meter high -- had a dedicated equalizer. It was not just for audiophiles!
There was a company called DAK in the 80's that sold all sorts of interesting stuff and I still use my BSR EQ-3000 which is an equalizer with a spectrum analyzer display fed by a microphone that you walk around the room with to confirm your settings. It even has a pink noise button that injects that into the amp so you have a uniform pattern to equalize. Sort of like an analog Sonos Trueplay I guess. We could do all of this in the 80s :-)
I suspect HiFi culture was slightly different in different parts of the world.
In my part of the world, the original HiFi guys rejected the EQ units. I was influenced by that I think (my father was an original). I still have a metre high stack of HiFi, in almost daily use, and have never felt the need to have an EQ unit.
I built my own 10 band equaliser using instructions from Elektor (I think) back in the 80s. The dream was to acquire a spectrum analyser but real life intervened.
Not sure when I even powered up my stereo system. Probably doesn't even work now.
Oh, that's really cool, the channel seems like it has a lot of other nice videos as well!
I wish there was a video that's so nice about compressors, like in Audacity you have all of these settings and most are also present in OBS and other software:
And I feel like visualizations can really help understanding them better, like: https://codepen.io/animalsnacks/full/VRweeb alongside maybe something that lets you loop an audio sample and see how different it sounds with each change. There obviously already are some videos and discussions and plenty of material out there, but I love a good visualization!
I have watched 10+ hours of lessons on compressors and I still don't hear it. I understand most concepts but don't use much compression besides side-chaining and the built-in Ableton glue compression.
I know what you mean - took me a while too. I understood what it does, how the parameters affect what it does and the mechanics of it, but struggled to "hear" it.
I even bought a cheap $25 Behringer guitar compressor pedal to see what I was missing but it didn't help - later I realized that my guitar playing isn't repeatable enough. So this isn't the way to go.
What made it click was due to an accidental mistake in my normal workflow - I recorded some DAW-less techno jam stuff using GarageBand (normally I just copy the wavs from my little Tascam). While playing back, I noticed that there's a master compressor and I started fiddling with it. With repetitive music like Techno and House, the difference between no compression and full compression suddenly becomes very apparent (although still somewhat subtle compared to other FX commonly used in music production). Also it helped that my recording had no compression on it - comprising just a raw drum machine and mono-synth.
I work with audio professionally and it took me a years to get a feel for dynamics processing. Like, it wasn't until I down sat with a nice compressor (in my case a Neve 453) and just did a lot of experiments, and then took my experiences from those experiments to my live gigs.
IME, you can get a feel for what the threshold and ratio are doing pretty quickly, and that's probably enough to be useful. In broad strokes that's all you need to make them "work". If you have the attack set way fast, you'll start to hear the signal get a bit muddy.
But attack and release (especially on a lot of plugins) are a bit funky, and I still can't tell what the knee is doing unless I move the knob around. And I own a couple clones (76kt or gold comp 2a) and a couple of distressors, and they sound different but I still need play around with them to coax them into what I think they should sound like.
That's normal. For engineers and mixers, compression is one of the more difficult phenomena to hear and build an intuition for.
My advice for learning is to totally overdo the compression on a drum track (snare, kick, hats, etc) and play with the settings. Ideally these drums are uncompressed.
Using a 4:1 ratio, lower your threshold all the way down until you're getting > 10 dB of gain reduction and then start playing with the attack time. What do you hear when the attack time is at 0 ms? What do you hear when you start to slow the attack time? 5 ms? 10 ms? 30 ms? 100 ms?
Then do the same with your release time. Start with it set as fast as it will go and then start to slow it down. What do you hear happening?
Once you have the attack time and release time feeling good then raise your threshold so that the compression is less heavy-handed (unless you like it). Set the threshold where the level of compression feels good to you.
It is pretty normal not to hear compression on material that already is quite compressed. But you will hear compression on very dynamic vocals or a snare drum, I am pretty sure.
My music listening speakers have two built in power amplifiers (one for the tweeter, one for the woofer) and have a DSP feeding right into a DAC, feeding right into those amplifiers.
There's a control box that comes with them, and when you plug a calibrated microphone into that box, and put it in the listening position, you can get it to do some frequency sweeps, one at a time, then they calculate a correction curve for each speaker, based on the actual response of the particular speaker in the particular room, and program that curve into the DSP of the speaker.
It's like night and day toggling the calibration on and off while listening to music.
And yes, as he says, the best hi-fi is just professional audio gear..
My music listening setup is simply a USB->AES converter box that feeds directly into the monitors, the monitors are a pair of Genelec 8050, and then the GLM box and volume knob. Never heard "hi fi" coming even close to it, not at the price, not at five times the price.
Same goes for headphones, you can't get much better than the simple and cheap DT990 (or 770 if you want them closed), sure, you can pay about 10 times as much for some Sennheiser hd800s, and those are pretty good, and I do have a pair of HE1000se, which are not only cheaper, but actually sound better too. But I'd never recommend anyone who's not as stupid as myself to buy anything "above" DT990.. And yeah, I EQ my headphones with a dbx 231x two channel 31 band EQ, and while that's not as scientific as the calibrated monitors, for a headphone listening experience, it gets pretty good.
Readit News
Also, given the topic (audio equalizers) there's no way it could have been a blog post.
> A video about all forms of equalizers. From one-click bass buttons to advanced studio correction.
That can be reduced to:
"Equalizers: From bass buttons to advanced studio correction."
PS: I'd try to keep the original title too, but in this case it doesn't look nice
"EQ: Equalizers - From bass buttons to advanced studio correction."
The video is very nicely made and it focus on sound systems / boomboxes frequency response and behavior of included filtering modes.
So when talking about EQs with “all forms” in mind, you should consider:
- EQ is merely combination of one or more filters.
- There are many common filter designs, the video isn’t about that, it also doesn’t mentioned Low-pass/High-pass/band-pass/bell or common structures, but it is only showing them.
- Filters made and behave differently
- Most filters (including the ones in the video) are done on the time domain (vs spectral one)
- Phase, this is the biggest missing piece in the video imho. Naive filtering is “smearing” the signal to achieve the different tone balance. By doing so, they also most likely change the phase (unless using linear phase filters)
- Filtering might result delay in time.
Technically filters don't cause phase shift, phase shift causes filters.
That means in the analog world all realtime filters come with phase shift, if they didn't you wouldn't have a filter.
In the digital world there is a thing called linear-phase mode where clever people found ways of building filters that shift the phase of the whole signal the same way (involves running the signal through forwarwards and backwards through special filters). If you now have an audio program that deals correctly with the latency introduced by each effect, you can shift the rest by the same amount meaning you bought yourself a linear phase response with added latency (a small delay) on everything. Remember: Humans can't really hear phase smears unless they happen in relation to another thing that is in phase. This only happens when you mix two signals. The result of that mix is yet again a filter (e.g. comb filters).
With filters everything is a trade off all the time, but again phase shift is not a just a bad side effect of a filter, it is the mechanism by which it does the work. And sometimes the phase shift introduced cannot be heard.
It's why I like full-range speakers—no crossover. At most you might throw a cap on a super-tweeter (yes, even so-called full range need help on the top and bottom ends of the spectrum—so super-tweeters + sub-woofers). Complicated crossovers destroy the phase.
Without crossovers/EQ the more stripped-down full-range setup preserves what the Hi-Fi nerds call "soundstage". (A good full-range setup: imagine the "soundstage" you get from wearing headphones—but without wearing headphones.)
Those seeking a full-range setup (especially with a tube amplifier) are kind of adjusting for EQ simply by the type of speakers/drivers and placement. (And to be sure they're not going for a flat, studio-monitor response but what sounds good).
His videos about LCD technology are hypnotizing. Bonus - he makes all the music.
As some sections of the video highlight it ended up needing to dip the mids and boots the lows and highs. This is my end result (so far, I'm probably going to tweak this continuously):
There's also an Output amp of -2.5dB that does not appear in the export for some reason.TUXEDO InfinityBook Pro 15 - Gen10 - AMD Laptop Speakers.
It does make a positive difference (why don't they teach us this in school? ;) ). Please note I am music illiterate.
EDIT: it looks like changing the input source has no actual effect, just need to make sure the internal microphone's volume is not muted, and have to keep system setting window open while doing this, maybe the microphone is no longer on when the window is closed?. Linux is weird sometimes.
But the reason this is not widely talked about is two fold:
First, because those measurements are heavily room dependent and they sound worse than treating a room. Because the room affects the sound in ways that EQ can’t really fix anyway.
Second, because the goal of mixing and mastering is making music sound good in a big variety of speakers and environments.
Deleted Comment
In my part of the world, the original HiFi guys rejected the EQ units. I was influenced by that I think (my father was an original). I still have a metre high stack of HiFi, in almost daily use, and have never felt the need to have an EQ unit.
Not sure when I even powered up my stereo system. Probably doesn't even work now.
I wish there was a video that's so nice about compressors, like in Audacity you have all of these settings and most are also present in OBS and other software:
And I feel like visualizations can really help understanding them better, like: https://codepen.io/animalsnacks/full/VRweeb alongside maybe something that lets you loop an audio sample and see how different it sounds with each change. There obviously already are some videos and discussions and plenty of material out there, but I love a good visualization!https://www.youtube.com/watch?v=K0XGXz6SHco
I even bought a cheap $25 Behringer guitar compressor pedal to see what I was missing but it didn't help - later I realized that my guitar playing isn't repeatable enough. So this isn't the way to go.
What made it click was due to an accidental mistake in my normal workflow - I recorded some DAW-less techno jam stuff using GarageBand (normally I just copy the wavs from my little Tascam). While playing back, I noticed that there's a master compressor and I started fiddling with it. With repetitive music like Techno and House, the difference between no compression and full compression suddenly becomes very apparent (although still somewhat subtle compared to other FX commonly used in music production). Also it helped that my recording had no compression on it - comprising just a raw drum machine and mono-synth.
IME, you can get a feel for what the threshold and ratio are doing pretty quickly, and that's probably enough to be useful. In broad strokes that's all you need to make them "work". If you have the attack set way fast, you'll start to hear the signal get a bit muddy.
But attack and release (especially on a lot of plugins) are a bit funky, and I still can't tell what the knee is doing unless I move the knob around. And I own a couple clones (76kt or gold comp 2a) and a couple of distressors, and they sound different but I still need play around with them to coax them into what I think they should sound like.
My advice for learning is to totally overdo the compression on a drum track (snare, kick, hats, etc) and play with the settings. Ideally these drums are uncompressed.
Using a 4:1 ratio, lower your threshold all the way down until you're getting > 10 dB of gain reduction and then start playing with the attack time. What do you hear when the attack time is at 0 ms? What do you hear when you start to slow the attack time? 5 ms? 10 ms? 30 ms? 100 ms?
Then do the same with your release time. Start with it set as fast as it will go and then start to slow it down. What do you hear happening?
Once you have the attack time and release time feeling good then raise your threshold so that the compression is less heavy-handed (unless you like it). Set the threshold where the level of compression feels good to you.
There's a control box that comes with them, and when you plug a calibrated microphone into that box, and put it in the listening position, you can get it to do some frequency sweeps, one at a time, then they calculate a correction curve for each speaker, based on the actual response of the particular speaker in the particular room, and program that curve into the DSP of the speaker.
It's like night and day toggling the calibration on and off while listening to music.
And yes, as he says, the best hi-fi is just professional audio gear..
My music listening setup is simply a USB->AES converter box that feeds directly into the monitors, the monitors are a pair of Genelec 8050, and then the GLM box and volume knob. Never heard "hi fi" coming even close to it, not at the price, not at five times the price.
Same goes for headphones, you can't get much better than the simple and cheap DT990 (or 770 if you want them closed), sure, you can pay about 10 times as much for some Sennheiser hd800s, and those are pretty good, and I do have a pair of HE1000se, which are not only cheaper, but actually sound better too. But I'd never recommend anyone who's not as stupid as myself to buy anything "above" DT990.. And yeah, I EQ my headphones with a dbx 231x two channel 31 band EQ, and while that's not as scientific as the calibrated monitors, for a headphone listening experience, it gets pretty good.
https://autoeq.app