So, I’m trying something a little different. Now that I have the HD58X, I actually have something that’s naturally pretty close to the Harman target (except with a bit less bass and a bit more treble, both of which suit my tastes well). Going back to the LCD2C, I’ve become interested in what makes the LCD series special and just enhancing that a bit. In that series, the most special is the LCD4. Now, I’m sure the LCD2C can’t replicate all the technicalities (in particular the speed of decay), but its distortion is actually competitive and I wondered if bringing the frequency response closer to the LCD4 would yield any dividends. As usual, diyaudioheaven is a valuable resource. Solderdude compares the frequency response of the LCD2C to the various other LCD series headphones. What seems to make the LCD4 different from the LCD2C are two broad differences:
No peak at around 800 Hz
Slightly higher level of treble starting at 2 kHz and up
So, I made an EQ profile to roughly mimic the LCD4:
Peak/Dip -1.5 dB at 825 Hz Q 2.00
High Shelf +6.0 dB at 4 kHz Q 0.47
I have to say, so far I like it. As noted before, the dip at 825 Hz makes vocals sound just a tad more natural. The slight treble boost across a broad range opens up the sound somewhat, seems to improve imaging a bit and helps percussion in particular sound a little more punchy and sparkly. It still preserves the general character of the LCD2C but gives it a little shot of adrenaline. So far I like it.
When I was using my HiFi Man HE-400S headphones as every work day listeners I used equalization to fix their low bass weakness. I first added the HiFi Man Focal pads, which has been well documented, but the equalization was the trick that made them much better.
I have been using Foobar 2000 as a high rez player and it has several equalizers available as plug-ins. I tried those and they worked well but I wanted to try a traditional approach.
Since I had luck with Schiit Audio gear, I bought the Loki tone control box when it was released. I added it to my headphone stack (all Schiit Audio…Wyrd, Modi Multibit, Lyr 2) and removed the plug-in from Foobar. That did the trick and I have a standard four gang tone control handy.
The Loki turned out to be so good I added one to my main system downstairs. That’s a pretty good setup and the Loki took nothing away. I have tone controls once again. I use it to cut the low end fed to my power amp.
My main speakers are Magnepan MC2s and they aren’t designed to be reproducing frequencies below 100Hz. I have a REL subwoofer for that purpose. The Loki is inline between my preamp output and my power amp input. I have the low end tone control set to 9 o’clock to cut the low end and it works beautifully.
The Loki takes nothing away from the rest of the sonic picture and the dammed thing only costs $150 USD. Its so inexpensive most audiophiles would immediately dismiss them but that’s a mistake. If you need a simple four gang tone control for any purpose try these out. The four bands are centered at 20Hz, 400Hz, 2kHz and 8kHz.
It works very well, doesn’t add noise and the tone controls seem to work as advertised.
I just tried Toneboosters Morphit. It doesn’t have a profile for the LCD2C, so I’m using the profile for the LCD2.2. With a target profile of “Generic HIFI”, it sounds pretty darned great, probably better than any of my own efforts at EQ.
I’m working on emulating this myself with some EQ but filling in the presence region some more. Something I like about what Morphit is doing is to put the bass hump at a low 30 Hz. Even with this substantial boost it doesn’t sound muddy, though I suspect the LCD2C’s ultra low distortion helps too.
I find your posts about EQ very interesting. Real contributions to our community. As a young man around 55 years ago. The only EQ I remember were the little levers that I’d lower or lift to change the sound. As I recall as I progressed in my musical journey, buying equipment that could sound good on its own was the way to go. I could be wrong since it was so long ago, but I think that’s how it was. I think I bso used to it that I still believe it.
Seeing EQ is so popular and talked about, I am wanting to learn more about it. I downloaded the Audeze Reveal plug in a couple of days ago. It asked me to choose my headphones, so I did. I chose my Lcd 4 headphones I see the plugin is install in preferences in my Audirvana software.
I’ll go online to see what I do next. I’m maybe concept is Auzeze has decided what’s best for the Lcd4s.
My question is for anyone else remember the levers and the belief that using EQ was frowned about. I might have gotten that all wrong and still believed it all these years.
I also really enjoy @pwjazz EQ posts, I’m completely new to EQ, but with the ADI-2 DAC I want to start messing with it more… Maybe create a specific profile for different headphones.
IIUC old analog EQ circuits could introduce quite a bit of distortion, which might be how EQ got a bad rap. To my knowledge and in my experience, modern digital EQ does not suffer from such problems. Most digital EQ is minimum phase, so it does introduce some frequency dependent phase shift, but I’ve personally never heard any ill effects from it.
I personally have fallen out of love with the stock LCD2C sound signature. It’s missing too much in the presence region, is a bit too forward around 900 Hz, and it lacks both punch and warmth because of the generally flat bass and midrange response. However, it has exceptionally low distortion throughout most frequencies (and especially in the bass, where dynamic drivers struggle) and pretty good transient response. As such, it makes a really nice platform for tuning via EQ.
I think the amalgamation of digital and mechanical is the future (and for that matter the past). Whether it’s something as sophisticated as using fly-by-wire to overcome the behavior of relaxed stability wings like on an F-16 or as simple as using digital controls on my instant pot pressure cooker, digital allows us to push the boundaries of the mechanical. With stuff like their iSine + Cipher cable and their Mobius, Audeze seems to be one of the leaders in the field of integrating digital enhancement into their headphones. I’ve never heard it, but from what I’ve read, the iSine without Cipher (or Reveal plugin) is pretty horrible.
With the proliferation of bluetooth and the slow elimination of headphone jacks, not to mention people’s varying sound preferences, I suspect that carefully physically tuned headphones will become a thing of the past. It just seems cheaper, more effective and more customizable to build a low distortion and low resonance driver and enclosure, and then use DSP to achieve the desired tunings.
Inspired by Morphit, I came up with my own EQ. It keeps some bass boost, cuts the peak at 800 Hz, adds more clarity in the 1-4 kHz region, and boosts 10 kHz and above but not as aggressively.
My HD58X has become my benchmark for “natural” sound and this EQ allows the LCD2C to come close on that score, while removing the lingering graininess (veil?) that I hear in the HD58X, delivering a bigger and more open sound, more clarity across the board and more articulate bass. It still gives up a little to the HD58X in punch and sounds generally more “ethereal”, but it’s still plenty dynamic sounding. I feel like with EQ the on-center imaging has improved a little as well, but the I think the HD58X still has it beat in that area.
The one thing I don’t like about this EQ and the LCD2C is that they encourage louder listening. It doesn’t seem to matter how much I turn up the volume, but it sounds better and better the louder it gets!
Applying parametric EQ does help to cater the headphone to your preferred tonality but it will not really fix any timbre issues. Still though interesting post.
That does not jive with my experience. Tonality, in particular the relationship between fundamental and overtones, is one of the main parameters that affects timbre. It’s not the only one, but it makes a difference.
What led you to the conclusion that changing frequency response can’t improve timbre?
The problem with equalization is that all tone controls have effect on frequencies long removed from the fundamental frequency of the filter.
Whether it is a high or low pass filter the equalization will effect frequencies many times the fundamental frequency. Its why tone controls fell out of favor with audiophile components and those components that still have tone controls include a bypass switch to pull the tone controls out of the signal path.
A 2kHz equalizer will effect frequencies at 1kHz and even at 200Hz. The filters introduce phase shift at those frequencies that are many times lower or higher than the effective frequency of the filter.
As such, equalization has a price and will do what it designed to do but it will also introduce effects long removed from the intended frequency effect.
Like we have all heard, there are no perfect answers.
Personally, I like the introduction of tone controls as long as they are well designed filters, with as little effect as possible to other parts of the frequency domain. Too many equalizers are available that introduce too much noise or added distortion. Nothing can be done about the aforementioned phase shift issues, unless you do the equalization in the digital domain, and then you introduce ADC and DAC related issues.
Unless you were trying to apply digital/software EQ to an existing analog output (e.g. a turntable, SACD player from it’s analog outputs, etc.), then there’s no reason why there’d be an ADC in the equalization chain at all. And the DAC would be the same DAC, at the same point in the chain, with or without EQ, so no additional issues there either.
Analog “tone” controls suffer from many issues, as you state, but coming from a digital source there’s no reason for there to be anything analog in the chain until you reach the output stage of the DAC.
Took the words out of my mouth! The main thing I worry about when boosting frequencies in the digital domain is clipping, but applying pregain takes care of that. There are both minimum phase and linear phase DSP implementations, with the former introducing some frequency-dependent phase shift and the latter introducing some latency. IIUC, most software EQ is implemented with IIR filters which are minimum-phase but computationally cheaper than the alternative (on general purpose CPUs). I haven’t personally pinpointed any negative effects from applying IIR filters to music.
One of the interesting things about the high-end software EQ solutions is the raw degree of control they allow you to exercise over not just the EQ curve, but how the actual processing is done. Most of my EQ work is done with the DMG “EQ” line (partly because I run it for studio use, also), and this is heavily in evidence there.
At the lower end of their product line (e.g. with “EQuick”, it’s JUST the EQ curve you get to play with - with the usual complement of shelf/cut/boost type options and the normal Q, amplitude, center settings you find with any good parametric EQ tool. HOW those parameters are applied is all handled behind the scenes, with the specific parameters of the filters (IIR vs. FIR, latency/depth and myriad others) behind pre-set by the software.
By the time you step up in their line to “EQuilibrium” you can make your own trade-offs in how the filters are applied, how much latency is acceptable (effectively changing the tap-length of the filter), IIR vs. FIR, and control over phase shifts, so an almost pathological degree. The defaults are, fortunately, sane and work in most cases, but you can really go to town to get as close to the ideal filter response for your purposes as your hardware (and latency allowances) will permit.
When just using EQuick on a little MacBook when traveling, I really don’t notice any unintended audible artifacts from applying EQ. Compare those settings in EQuick to the fully-optimized configuration I run for EQuilibrium on my Mac Pros or MacBook Pro, and you can tell a difference - if you listen for it. I’ll leave it to the individual listener to figure out which settings they personally prefer.
This reminded me of something that’s been on my mind lately, which is the distinction between a transducer and an instrument. To me, headphones and the related source chain are primarily a transducer, in that they’re meant to take a complex electrical signal that represents sounds and reproduce it as accurately as possible, with as little distortion (i.e. extra sounds) as possible. This is in contrast to an instrument, which takes a relatively simple signal like the pluck of a string or the press of a key and transforms that into a complex sound with resonances, overtones, attack and decay, envelope and all that good stuff. I would include all the work that goes into recording those sounds as part of the same “instrumental” domain whose goal is to produce sounds by combining an idea with tools.
Now, when us headphone hobbyists talk about headphones and source chains, we sometimes get into concepts like adding warmth, sharpening attack, introducing euphonic distortion with tubes, etc. It seems to me that this starts to blur the line between transducer and instrument. Are we purely consumers of music or do we in a sense become producers of music as well? And insofar as we’re producers of music, do we miss out by being the only ones who consume that product on our own headphones? Is that why we find ourselves drawn to forums such as this one to share our experiences, ideas, recommendations and techniques, as a sort of substitute for being able to more directly share our product with others?
You have a rabbit hole within a rabbit hole within a rabbit hole here … first there are the issues pertaining to applying EQ in the first place.
Then you can go further and start spending more time considering the strengths of different approaches and specific tools.
And then even within a single tool, especially the more sophisticated ones, there are myriad little controls and settings that effect exactly how your chosen EQ points/values are actually applied/filtered behind the scenes.
Using a non-specifically EQ example, I tend to find that with any DAC that has selectable filters I will often start on a minimum-phase filter and in fairly short order wind up settling on a linear-phase filter as my preference. The longer I listen, the more I find linear-phase to work better - especially in terms of overall listening fatigue, which occurs much more often for me with minimum-phase approaches (though for studio work other factors, principally latency, can trump that).
I’m lucky in that I tend to find things I like, get them setup the way I enjoy them, and after an initial period of fiddling around to get to those points, am happy to leave them alone and enjoy the music.
Honestly this is how I feel about most things! I tinker initially until I’m happy then leave it alone…unless I learn something new that is worth looking into then the cycle starts again lol
