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Techniques behind arbitrary response or "drawable" equalizers?

Started by Unknown June 14, 2016
On Sun, 3 Jul 2016 11:50:45 -0500, Les Cargill
<lcargill99@comcast.com> wrote:

> Bob Masta wrote:
<snipped for brevity>
>> Whether for sound exposure or room EQ, there is one caveat >> that many people don't consider. This has nothing to do >> with whether you use manual EQ or an automated FIR approach. >> It is that you can't blithely correct for deep dips without >> drastic overall power reduction. Consider a simple system >> with a single power amp, and a speaker with a 12 dB dip at >> frequency F. To correct for that, the EQ needs to provide a >> 12 dB peak at F. That means the amp has to provide 4x the >> voltage (16x power) at that frequency, which means you need >> to leave that much headroom at other "flat" frequencies. >> > >Just so. The main problem in rooms isn't dips; it's usually >peaks caused by reinforcing modes, and those can't really be >handled in the frequency domain. > >As to your speaker with a 12dB dip - it's not that hard to find one >that doesn't have that these days, what with even small >drivers with low Fs and good compliance.
No doubt true over most of the human audio range. The problem for auditory research is that most of it is done on rodents, which hear well above 20 kHz. It's not unusual to need a flat octave centered at 20 kHz or more, for example. Tweeters tend to have terrible dips at their upper ends. This is compounded by the problem of finding *loud* tweeters, since typical research noise exposures are for studying high-frequency hearing loss or tinnitus origins and/or treatments. To evaluate the efficacy of a treatment, you need to be able to produce uniform groups of animals with the same loss or tinnitus symptoms, then apply your treatment to half and compare. Best regards, Bob Masta DAQARTA v9.20 Data AcQuisition And Real-Time Analysis www.daqarta.com Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, Pitch Track, Pitch-to-MIDI FREE 8-channel Signal Generator, DaqMusiq generator Science with your sound card!
On Sun, 3 Jul 2016 07:28:09 -0700 (PDT), Matti Viljamaa
<viljamaadsp@gmail.com> wrote:

>As I tried to point outamples, > >I'm trying to perform filtering based on data obtained from FFT analysis. A= >nd I desire to have the filtering adjust between FFT analysis frames. Perfo= >rm a sort of dynamic equalization based on the FFT analysis. But obviously = >I need a way to "draw" the equalizer curve and update it even on a per samp= >le basis so that it follow the envelopes that I desire.
Ahh, sounds like you want a "channel vocoder" or just plain "vocoder" (not to be confused with a "phase vocoder"). This has a bank of filters (or in your case a bunch of FFT bins), the output magnitude of each of which sets the gain of a narrow-band filter of the same frequency. These are used in electronic music (or were, "back in the day") to apply human speech formants to other musical instruments, or to anything at all. Check out Wikipedia, and scroll down to the musical section. Vocoders are usually described with "bank of filters" models, but of course you could get the same effect with an FIR approach. I would imagine there are a number of digital implementations out there, though I haven't searched. Best regards, Bob Masta DAQARTA v9.20 Data AcQuisition And Real-Time Analysis www.daqarta.com Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, Pitch Track, Pitch-to-MIDI FREE 8-channel Signal Generator, DaqMusiq generator Science with your sound card!
Bob Masta wrote:
> On Sun, 3 Jul 2016 11:50:45 -0500, Les Cargill > <lcargill99@comcast.com> wrote: > >> Bob Masta wrote: > > > <snipped for brevity> > >>> Whether for sound exposure or room EQ, there is one caveat >>> that many people don't consider. This has nothing to do >>> with whether you use manual EQ or an automated FIR approach. >>> It is that you can't blithely correct for deep dips without >>> drastic overall power reduction. Consider a simple system >>> with a single power amp, and a speaker with a 12 dB dip at >>> frequency F. To correct for that, the EQ needs to provide a >>> 12 dB peak at F. That means the amp has to provide 4x the >>> voltage (16x power) at that frequency, which means you need >>> to leave that much headroom at other "flat" frequencies. >>> >> >> Just so. The main problem in rooms isn't dips; it's usually >> peaks caused by reinforcing modes, and those can't really be >> handled in the frequency domain. >> >> As to your speaker with a 12dB dip - it's not that hard to find one >> that doesn't have that these days, what with even small >> drivers with low Fs and good compliance. > > No doubt true over most of the human audio range. The > problem for auditory research is that most of it is done on > rodents, which hear well above 20 kHz. It's not unusual to > need a flat octave centered at 20 kHz or more, for example.
Yeah, I dunno about all that. LPFing most audio @ 10k doesn't damage it all that much for casual listening.
> Tweeters tend to have terrible dips at their upper ends.
But there's generally little up there. It could matter a lot but it often doesn't - perceptually, the top octave is the least important.
> This is compounded by the problem of finding *loud* > tweeters, since typical research noise exposures are for > studying high-frequency hearing loss or tinnitus origins > and/or treatments. To evaluate the efficacy of a treatment, > you need to be able to produce uniform groups of animals > with the same loss or tinnitus symptoms, then apply your > treatment to half and compare. >
I can't say I understand why anyone would consider animals valid test subjects for evaluations of things related to human hearing.
> Best regards, > > > Bob Masta > > DAQARTA v9.20 > Data AcQuisition And Real-Time Analysis > www.daqarta.com > Scope, Spectrum, Spectrogram, Sound Level Meter > Frequency Counter, Pitch Track, Pitch-to-MIDI > FREE 8-channel Signal Generator, DaqMusiq generator > Science with your sound card! >
-- Les Cargill
On Monday, July 4, 2016 at 8:27:36 AM UTC-7, Les Cargill wrote:
> Bob Masta wrote:
(snip on audio equalization)
> > No doubt true over most of the human audio range. The > > problem for auditory research is that most of it is done on > > rodents, which hear well above 20 kHz. It's not unusual to > > need a flat octave centered at 20 kHz or more, for example.
> Yeah, I dunno about all that. LPFing most audio @ 10k doesn't > damage it all that much for casual listening. > > > Tweeters tend to have terrible dips at their upper ends.
> But there's generally little up there. It could matter a lot but > it often doesn't - perceptually, the top octave is the least > important.
The top key on a piano is 4186 Hz, and you might hear its third harmonic. But then again, it is pretty rarely used in actual music. Seems that years ago, someone figured out that 20Hz to 20kHz was a nice convenient range to specify amplifier response. Babies might hear to 20kHz, but adults probably closer to 15kHz if they didn't go to too many rock concerts when they were teens. But numbers on paper are more important than actual listening, at least when it comes to selling stereo amplifiers. -- glen
Matti Viljamaa  <viljamaadsp@gmail.com> wrote:

>On Sunday, July 3, 2016 at 10:29:42 PM UTC+3, Steve Pope wrote:
>> Why do you include the words "using FFT" in the above sentence?
>Because I don't know of another way of doing spectrum analysis.
Okay, so you're not so much disinterested in solutions that do not include an FFT, as you are of the belief that most or all viable solutions will include an FFT. Fair enough. Steve
herrmannsfeldt@gmail.com wrote:
> On Monday, July 4, 2016 at 8:27:36 AM UTC-7, Les Cargill wrote: >> Bob Masta wrote: > > (snip on audio equalization) > >>> No doubt true over most of the human audio range. The >>> problem for auditory research is that most of it is done on >>> rodents, which hear well above 20 kHz. It's not unusual to >>> need a flat octave centered at 20 kHz or more, for example. > >> Yeah, I dunno about all that. LPFing most audio @ 10k doesn't >> damage it all that much for casual listening. >> >>> Tweeters tend to have terrible dips at their upper ends. > >> But there's generally little up there. It could matter a lot but >> it often doesn't - perceptually, the top octave is the least >> important. > > The top key on a piano is 4186 Hz, and you might hear its third harmonic. >
Maybe.
> But then again, it is pretty rarely used in actual music. >
There ya go.
> Seems that years ago, someone figured out that 20Hz to 20kHz was a nice convenient > range to specify amplifier response. Babies might hear to 20kHz, but adults probably > closer to 15kHz if they didn't go to too many rock concerts when they were teens. >
I'm about 14k these days. That's not going to get any better...
> But numbers on paper are more important than actual listening, at least when > it comes to selling stereo amplifiers. >
Yes.
> -- glen >
-- Les Cargill
Les Cargill  <lcargill99@comcast.com> wrote:

>I'm about 14k these days. That's not going to get any better...
I think there's significant energy about 10 KHz from sources that include perussion instruments including cymbals, high-hats and tambourines, and that even people with significant hearing loss can hear such content, if they have equalized and amplified things enough (say, with headphones). (Not that's I'd recommend this for endurance listening...) S.
On Monday, July 4, 2016 at 4:56:43 PM UTC-5, Steve Pope wrote:

> I think there's significant energy about 10 KHz from sources > that include perussion instruments including cymbals, high-hats > and tambourines
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm http://recordinghacks.com/articles/the-world-beyond-20khz/
Greg Berchin  <gjberchin@charter.net> wrote:

>On Monday, July 4, 2016 at 4:56:43 PM UTC-5, Steve Pope wrote: > >> I think there's significant energy about 10 KHz from sources >> that include perussion instruments including cymbals, high-hats >> and tambourines > >http://www.cco.caltech.edu/~boyk/spectra/spectra.htm > >http://recordinghacks.com/articles/the-world-beyond-20khz/
Greg- thanks. From the first of these links, "a cymbal crash shows no signs of running out of energy at 100 KHz". Steve
Here's an interesting experiment related to this thread. Feed an audio signal into a windowed fft with 50% overlap. Take the magnitude of each bin and apply each bin magnitude value to a peak detector with an adjustable exponential decay rate. 
In parallel, feed white noise into a 2nd fft bank, and apply the peak detector values from the first fft bank as gain factors to each bin of the 2nd fft bank. 
Then inverse fft the 2nd bank, convert back to stream, and add this to the original input. 
What you get is a pretty decent-sounding reverberator where the reverb "tail" is 100% fake. The ear recognizes that the spectrum of the tail is the same as the spectrum of the direct signal, and it fuses them together into a single sound.  You can make the tail as long as you want by changing the decay time constant of the peak detector.
All sorts of improvements can be made including frequency-dependant time constants and adding early reflections. 
This may not make the best reverb but it's interesting how the ear can be fooled. 

Bob