EQ: A video about all forms of equalizers
youtube.com
youtube.com
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.
I mentioned the forward/backward bit as this was a way to achieve linear phase in analog times eith more conventional filters (although the cost here was even higher with the additional added noise floor due to tape noise ad the extra work involved).
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):
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
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.
Not sure when I even powered up my stereo system. Probably doesn't even work now.
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 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:
* Threshold (dB)
* Make-up gain (dB)
* Knee width (dB)
* Ratio
* Lookahead (ms)
* Attack (ms)
* Release (ms)
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.
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.
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.
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.
Correcting a speaker requires measuring the *anechoic* response, and only the bass should be tuned to subtract the room's effect.
This video [1] provides a crash course on science of audio reproduction.
I would assume they don't care about the very low bass because they were designed as a unit and physically couldn't produce those notes very well anyway. Hence the need to boost slightly higher frequencies to fake bass.
The more squiggly ones later (eg: 2:45) are what comes out of a speaker. But since he uses a microphone to measure the speakers, so that measurement will also come with room sound. He would need an anechoic chamber to measure "only the speaker".
This is gold. I wish so many people understood and used this. Pro equipment is more robust and more versatile. And if you know some people you can get amazing used stuff at bargain prices from your local musicians or sound engineers. Lot of times, people I know (and I too, though I'm not a pro sound engineer) will help you choose and setup just because you showed interest in the things we are interested in. Beers are assumed.
One drawback is that pro audio stuff may look ugly to non pro people. I made my wife listen to the cheapest studio monitors I could find on amazon, equalized with the same pink noise method in this video, and compare it to her bose and marshall speakers. She liked the sound better but my speakers are "yuck ugly" :(
I was thinking of playing a pink noise on the speaker and recording it with a cheap microphone or displaying it with the Spectroid app, but the microphone probably has it's own frequency response.
Is there an App for that ? With each phone model microphone factory calibrated ?
Is there a way to use known fact about physics like harmonics should have a specific shape (timbres) that's should help equalize frequency with respect to each other ? Or from various microphone positions, calibrate it so that any cheap microphone can do the trick ?
Spot on. When you buy a measurement microphone, you can download it's specific response from the manufacturer.
I've found a lot of enjoyment in equalizing as best you can with hardware, and then doing the finishing touches against your ears.
Take a frequency generator, I use a web one, and play a tone that matches your eq knob, and work your way through making sure all the tones are at the same volume. You'll have to do this a few times.
You might be surprised at how good it will sound. Especially the upper ranges which will usually be far too low if you only calibrated with mics and hardware and not your wetware.
I mean/meant pleasant for average (layman) interest. A compliment, in other words.
No BIPM, no SI units: what the BBC say.
I guess, that's why it says "you can make it look like a straight line".
He then shows a horizontal straight line: that's white noise?
Does this also mean that audio equipment in general will display a spectrum with this kind of logarithmic offset adjustment?
The orders of magnitude in power/loudness is pretty astonishing.
When introducing decibels to new audio engineers, we generally introduce it as a 3db increase is a doubling in power, a 10db increase is a 10x increase in power.
It gets silly when you start talking sound pressure level, because how people perceive a 10x increase in the output power, is about a doubling of the perceived loudness.
The question was whether or not the audio displays have a correction to make pink noise render as a flat horizontal line, whereas general test and measurement spectrum analyzer would show a tilted line.
There are more frequencies in the high-range, so it naturally sounds considerably higher pitched.
i.e. a C6 on a keyboard sits at around 1Khz (1046.502 to be exact) and a C#6 is 1108.731 KHz (a difference of 62.229 Hz)
While a C3 is 130.8128, and C#3 is 138.5913 (a difference of only 7.7785 Hz).
Then using an entire octave as an example, C3 (130.8128) to C4 (261.6256) = 130.8128 (0Hz up to C3)
vs C6 (1046.502) to C7 (2093.005) = 1046.503 (0hz up to c6 +- 0.001hz)
"EQ isn't phase shift, phase shift is EQ.", a video title I saw once, but hits the nail on the head.
"Nothing is for free.", if you eq some frequencies, you affect some other frequencies too. You can use this to your advantage, if you are aware of this.
"EQ graphs are lying." Lack of resolution and not revealing the phase shift makes most stock EQ graphs not telling the truth.
"My hearing aids sound strange/metallic." This is the phase shift, you are hearing.
Have a nice Sunday everybody!
Looks like an incomplete approach if you ask me.