# The most reliable/easiest way to EQ headphones properly to achieve the most ideal sound (for non-professionals)



## Lunatique (Apr 16, 2020)

(UPDATED on April 16th, 2020, with new links to missing test tones and YouTube videos.)

I've been wanting to write a comprehensive guide on how to do surgical EQ corrections to achieve the best sound possible, in a way that is the easiest and most reliable, without any guesswork and constantly wondering if you got it right.

There are other tutorials on how to EQ headphones, but I haven't seen one that teaches people how to do it in the most reliable and easiest way possible. The other tutorials usually tell people to use pink noise and sinewave generator, and while those are useful tools, they are not the easiest or most reliable for people who are inexperience in critical audio assessment at the professional level.

I want to state at the beginning that experienced, informed, educated professional audio and advanced audiophiles understand that the most ideal sound is within that small range of acceptable accuracy/neutrality in sonic signature. This is the golden standard used by the professional audio industry and everything you've ever listened to that's produced by professional musicians, producers, mixers, mastering engineers, etc., adhere to that golden standard. Whatever slight subjective coloration among them all fall within that acceptable range of accuracy/neutrality. Before we begin, let me try to settle once and for all why achieving the most accurate and neutral sound should be the golden standard that all music lovers, audiophiles, and audio professionals, and why intentional coloration is not a good thing for most people.


*Why would you want neutral/accurate sound? Isn't that a bad thing for musical enjoyment? Doesn't neutral mean boring, cold, thin, sterile, clinical, etc.?*

One of the most popular but grossly mistaken misconceptions in the hobbyist headphone/audiophile world, is that neutral/accurate sound means "clinical," "sterile, "cold," "bright," "sharp," "boring," etc. I have tried to to do my part to educate others about this for years in forums but it's such a pervasive misinformation that it gets passed around and regurgitated at frightening frequency--I just can't seem do anything to turn the tide (neither can other informed people who are also trying to combat this type of pervasive ignorance).

True neutral/accurate sound has no coloration, just like if you look out a very clean and clear glass pane window and all you see is reality, as if the glass panes don't even exist as you accidentally walk into it and bump your head. If you have a television or computer display that is high fidelity with high dynamic range, accurate and wide color gamut, optimal brightness and contrast, refined detail, high refresh rate, etc., you would never describe that display as being "clinical" or "sterile" or "cold" or "boring," right? It simply looks very natural and life-like. You would also never intentionally push the color balance of your display so that the flesh tones looks too green and with exaggerated saturation that looks neon, or the shadows are completely crushed into flat blacks with no details left, or the highlights are burned out so it's just solid patches of white and no micro details, etc. What you want is a display that is perfectly accurate/neutral, so everything appears natural and life-like. So if you wouldn't want significant deviation from accuracy in your display, why would you want it in your audio gear?

A truly neutral/accurate sound system will have authoritative and deeply extended powerful bass that is very well controlled, articulate, detailed, punchy, and full. It will have mids that are natural, smooth, detailed, and rich. Its treble will be airy, refined, and clear. There should be no hint of muddy, bloated or anemic bass, no recessed, sibilant, or nasal mids, and no hard-edged, peaky, splashy, or dull treble. Your audio system should simply disappear and what you hear is very natural life-like, as if the audio gear has disappeared.

The fact is, outside of the small minority of higher-end or well-informed/experienced audio professionals and audiophiles, the vast majority of audio hobbyists and musicians have never even heard truly neutral/accurate sound before (and this is taking into consideration the amount of acceptable variations within the threshold of what we can objectively call neutral/accurate that's measurable), and this is exactly why this misconception about neutral/accurate sound is so popular. It really takes having heard it at least once in your life to understand why it is and should be the golden standard to judge everything else by.

Some of you might want to jump in and say that there's no true neutrality/accuracy for speakers and headphones since our individual physiological differences (such as HRTF, ear canal shapes) means we all hear differently from each other, regardless if the manufacturers produce audio devices that measure flat in anechoic chambers or a testing dummy head. There's also the matter of room acoustics that will significantly alter the sound of speakers in a room. I'm going to address that right now.

I can appreciate that individual HRTF can alter what each of us hear, and a carefully tuned product that measures flat might not be flat sounding to some people due to that fact. However, there is still individual objectivity based on what you actually hear in relation to a measurement that is neutral. I can use test tones as an invaluable tool to perform surgical EQ with parametric EQ, adjusting frequency bands that are both very narrow and very wide and in-between, and I can patiently adjust until the entire range from 20Hz to 20KHz sounds very smooth when playing back a logarithmic sweep, or very even amplitude when playing sequential sinewave test tones at regular intervals. That is what you can do for yourself individually, adjusting for your own idiosyncratic physiology so that you are getting a neutral frequency response. And if you are an audio professional, you can compare your own physiological biases against a measured neutral response to understand where you differ from it, so you can make critical mixing/mastering decisions more accurately.

As for the argument that the consumers are listening on all types of devices anyway, so what's the point for mixing/mastering engineers to care about neutrality/accuracy--it's very simple. If you don't start with the most neutral/accurate point possible, you will make unwanted deviations far worse because you're already starting with something that's inaccurate and the consumer audio devices will only exacerbate it. If you start with neutral/accurate, then at least the starting point is ideal, and whatever coloration the individuals add to it with their audio devices will be starting with a source that's accurate/neutral, so the deviation won't be as severe.

If I can't convince some of you that neutrality/accuracy should be the golden standard we judge audio by, then that's fine--you can believe whatever you want and enjoy audio however you want. I'm writing this for people who do care and want to learn how to achieve neutral/accurate sound, so let's just live and let live and not get into a debate about this. If you really want to debate this, I already have a long thread about this I posted years ago with many pages of raging debates on this topic--you can simply add your 2 cents to that thread, which is located here:

http://www.head-fi.org/t/564465/misconception-of-neutral-accurate

I'll say this though--if you suspect that you probably have never heard truly neutral/accurate audio reproduction before at the level of a professional mastering sound system in a professional level audio production studio with proper studio design schematics and acoustic treatment (or the headphone equivalent of it--which is extremely rare and likely does not exist as of yet--especially if you don't know how to do proper surgical EQ corrections), then you should at least accept the fact that maybe you have no authoritative opinion on this matter and you need to experience it at least once in your life before you start arguing against neutral/accurate audio reproduction. For many people, finally hearing what it sounds like is a profound revelation that forever changes how they feel and think about audio and music reproduction.

Also, I want to make it clear that some levels of intentional coloration (such as using tube amps to achieve a warmer sound) still falls within the general acceptable range of neutral/accurate sound, because the deviation is much more broad and subtle and not in isolated frequency bands that causes really uneven frequency response with erratic dips and spikes, or silly bombastic bloated bass that completely overwhelms other frequency ranges and sounds uncontrolled and muddy and lacking in articulation and detail. So those of you who like some subtle level of intentional coloration--you are safely within the general range of acceptable neutrality/accuracy.


*Okay, I now know I want neutral/accurate sound. Show me how to achieve it in the easiest and most reliable way possible.*

Now that I've established that for some people, truly accurate/neutral sound means aural bliss and is the one and only standard we should have/need, let me show you how to actually achieve it in the easiest, most reliable way possible.

I mentioned that using pink noise and sinewave generators are not the easiest or most reliable way, and here's the reason why: It's actually harder to judge frequency balance when the entire audible frequency range is playing back at the same time (pink noise), because unless you are an experienced audio professional (or equivalent of one), it will be very difficult for you to assess just which frequency bands are actually out of balance, and by exactly how much. We have a natural imbalance in our human hearing known as the Fletcher-Munson curve, where extreme high and and low frequencies sound quieter to us when playing at quiet levels, despite being at the same amplitude as other frequencies, and that imbalance changes as the amplitude gets louder and louder. We also get confused and overwhelmed easily when too much sensory information if presented to us all at once.

Sinewave generators are also not the easiet to use to do critical audio assessment, because you need to input the number for the frequency and then generate one frequency at a time, and there's no way to instantly switch between various different frequency bands in split seconds to judge the differences in their amplitude.

So while pink noise and sinewave generators are helpful tools, they shouldn't be your only tools. Instead, what you want to use as a more reliable and easier way to assess frequency response is to use log (logarithmic) sweep and pre-rendered sinewave test tones at regular intervals from 20Hz to 20KHz.


*Here are the tools you'll need (all free or have trial versions, or very cheap):*

A music player that has a good parametric EQ (don't use graphic EQs--they are not suitable for surgical EQ corrections). My recommendations are:

*J River Media Center (desktop/laptop computers)* - This media librarian/player has the best implementation of DSP plugins management I've tried (and I tried many), and is vastly more ergonomic and intuitive than something like foobar2000. It is also one of the best in terms of sound quality, and very speedy in response even with extremely large music libraries. You can also easily host VST plugins in Media Center without having to jump through hoops.

In Media Center, I use a freeware parametric EQ called *EasyQ*, from RS-MET:

http://www.rs-met.com/freebies.html

It's extremely easy to use, has no additional coloration (some EQs apply intentional coloration such as analog emulations), and I get great results from it. If you are concerned about phase distortions, then use a linear phase EQ (just search for one online). It's really not necessary most of the time, unless you are actually hearing weird distortions (this has never happened to me in all the years I've been doing this, and when there were distortions, it had to do with gain staging, not phase issues). There are lots of free and commercial parametric EQs out there, and you'll find them easily with a quick search, particularly with KVR Audio's plugin database:

http://www.kvraudio.com/plugins/newest

To install a VST plugin in Media Center, just open up the DSP Studio in Media Center and click on "Manage Plug-ins," then "Add JRiver, VST, or Winamp plugin," then navigate to the dll file and open it.

*Neutron (Android) *- Neutron is the best audio player on Android, period. It is the only one that has extensive professional features such as advanced processing algorithms, a useful parametric EQ, a proper crossfeed (some other player apps claim they have crossfeed but they have no idea what a proper crossfeed is and theirs sound horrible and will harm the sound quality far too much), ReplayGain, etc.

*Equalizer (iOS) *- I switched to Android around 2013, so I haven't kept up with iOS apps, but this was the parametic EQ/audio player I used. There are probably other alternatives that are just as good or better by now, but as long as it's got a good parametric EQ, it should do the job.

When you are doing comparisons of how the headphone sounds with and without the EQ, you can use the bypass button that most EQs have, or there might be a activate/deactive button or checkbox in your player of choice.

*Audio Frequency Sprectrum Analyzer* - This is a very important tool for assessing audio. It shows you visually in real-time the frequency of the audio content being played. You need this to identify frequencies during a log sweep or in musical materials where the prominent audible energies are.

J River Media Center already has a built-in analyzer, so you don't need to go find one. If you need an analyzer, just google for "Audio Frequency Sprectrum Analyzer " and you'll find many. If you want to search only for freeware, you can use KVR Audio's plugin search database to specify you want freeware only.

*Various test tones* - I've uploaded all the test tones you really need to assess frequency response balance. It includes a log (logarithmic) sweep, pre-rendered sinewave test tones at regular intervals from 20Hz to 20KHz, and pink noise. You can download them here:

https://1drv.ms/u/s!AuJvJJ14eSzGge14GMaT9CPbob8z-w?e=mqLIA0

You can also search YouTube for "sine wave test tones" and "sine wave sweep."

Choose a sweep that doesn't take too long. Keep it around 30 seconds for the entire sweep. Here's an example of a good one:


Here's one with sine wave tones at regular intervals. You can just drag the playback seek bar to jump around to various frequencies (or just keyboard shortcuts J and L for big jumps, or left and right arrow keys for smaller jumps):


*A quick side-note about speaker/room correction:* For those of you who are interested in correcting your speakers and the room modes caused by acoustic issues in your listening area, I highly recommend IK Multimedia's ARC System 2:

http://www.ikmultimedia.com/products/arc/

I use it and swear by it. I get perfectly neutral/accurate frequency response at my listening position. My Klein+Hummel O 300Ds and Neumann KH805 sounds amazing with the ARC System. There are other similar products on the market (both hardware and software), and you can find them by searching for speaker/room correction software/hardware. ARC System is the best IMO because of how much it improved from the first version (the first version wasn't nearly as good, which is why some people used to poopoo it), how easy it is to use, and the result speaks for itself.


*Here's how to use the tools:

Audio Frequency Sprectrum Analyzer*  - You need this to see exactly what frequency is playing during the log sweep. You need to put this first in your DSP signal chain, before the EQ, otherwise the EQ (or any othe DSP plugin) will skew its reading. So for example if you are using J River Media Center, you want to open up the DSP Studio and CTRL+Drag the Analyzer plugin all the way to the top of the plugins list, as far as it'll be allowed to go. Make sure the EQ plugin is lower than the Analyzer plugin in the signal chain.

Some analyzer plugins will by default use an algorithm  that's not ideal for easy visual assessment, since it does not take into consideration human hearing sensitivity shift in high frequency range. This means if you look at the analyer during a log sweep, it'll seem like the amplitude gets lower and lower the higher the frequency gets. But don't worry about it, because we're only using the analyer to identify the frequency, not its amplitude. If it bothers you, you can switch to a different algorithm that compensates for it and will appear flat across the frequency range when playing back a log sweep.

This is how it should look like when playing a log sweep:


*Log sweep* - When you play a logarithmic sweep, what you'll hear is a sinewave tone that sweeps the entire audible frequency range from 20Hz to 20KHz. This is the fastest and most reliable way to hear any imbalances in your headphone, with the exception of the lowest sub-bass or highest treble frequency ranges (and we'll use pre-rendered sinewave test tones for that). When you listen to the sweep, if there are dips and spikes, you will hear them very clearly. A perfectly neutral/accurate headphone will sound very smooth during the sweep, without any significant dips or peaks. Most headphones will have significant peaks and dips somewhere in the upper-mids / sibilance range (3KHz to 8KHz) and the lower treble range (9KHz to 12KHz). You might also hear bloated mid to upper bass around the 125Hz to 250Hz range. Some headphones also have a dip right before the upper-mids in the range of 2KHz and 3KHz.

Note that it's natural to hear a gradual drop in amplitude when going from 6KHz and up. This is because highest frequencies will have far less energy to our human hearing. It's also natural  to hear that from 50Hz to 40Hz, the bass energy feels stronger than the other bass frequencies even when they are in fact not louder. It'll sound like your skull is vibrating, and this is normal. In fact, if you don't hear them with slightly strong energy, that means there's a roll-off/dip. From 20Hz to 30Hz, it'll seem like the energy gets lower, and this is normal too. At such low frequency, they are more felt than heard. Also, many headphones cannot handle frequencies that low and are completely rolled off at such low frequencies. Just because the manufacturer lists 20Hz-20KHz in the specifications does not mean much because without specifying how much deviation, that headphone could very well be playing 30Hz at -12 dB, which is essentially useless. This is why the standard for specifying acceptable deviation in the professional audio industry is to use ± 3 dB (plus or minus 3 dB of threshold in acceptable accuracy). Consumer audio industry is far less strict with standards, which is why there is so much bull$hit and snake oil and outright lies and false advertising in consumer audio industry.

When you have identified the dips and peaks, write them down. Don't worry about how many dBs of difference to write down--just write down at what exact frequencies you're hearing audible differences.

*Pre-rendered sinewave test tones* - These are much better than sinewave generators because you can instantly switch from one exact frequency to another without having to waste time entering any numbers, and can jump between any frequencies instantly and repeatedly.

When you playback the sinewave tones, make sure you do not have any kind of normalization /smart volume/RepayGain turned on. This means any settings that will automatically make different songs playback at the same volume. Although the sineware test tones should be already normalized to the same volume, it's good to take this extra precaution just in case.

What I usually do when using pre-rendered sinewaves, is to first assess the bass region. start at 200Hz and sequentially play each interval all the way down to 20Hz. If the low sub-bass is rolled off or bloated, you will hear it.

Next, I would start from about 80Hz and start going up sequentially all the way to about 1KHz. If there's bloated upper to mid-bass I'll hear it. Headphones that sound overtly warm tend to have a thickness in the 200Hz range, and you should hear it.

Then from 1KHz, you go up sequentially all the way to 20KHz. Listen for noticeable dips and spikes. Some headphones will dip around 2~3KHz, and many tend to have significant dips or peaks between 4KHz and 8KHz. For IEMs, the resonance peak at around 7KHz is a big problem for some people due to the ear canal shape, and can spike as much as 12 dB or more.

Many headphone manufacturers tune their headphones to have a peak at around 10KHz to 12KHz (either narrow or wide bandwidth), to create the illusion of detail. This is not a good thing because it's not natural detail--it sounds artificial. Frequencies above 10KHz tend to sound rolled off, and the headphone itself will often in fact be rolled off in the highest frequencies. For those of you who are past teenage years, the older you are, the more you will lose hearing in the highest frequencies. Most older adults might only be able to barely hear 16KHz.


*Understanding headphone measurements and ideal target response before we do corrections:*

Now that you've listend and noted all the areas of imbalance in your headphone, we'll start correcting those problems.

Before we start with the EQing, it's important to learn about the Harman Target Response Curve. Do not skip this step--it is crucial to your understanding of how headphones are supposed to sound different from speakers, and why a neutral pair of headphone shouldn't measure totally flat. Read this two-part explanation of headphone frequency response measurements before moving on:

http://www.innerfidelity.com/conten...equency-response-part-one#PxVkSf8KrXBSEM6I.97

http://www.innerfidelity.com/conten...equency-response-part-two#W6HDd66v1yfLMk0R.97

You might also want look at the photo I posted of the Audez'e frequency response graph and superimposed correction in one of my old threads about EQing heasphones here (and read what I wrote about it just below the photo):

http://www.head-fi.org/t/551426/my-eq-curves-for-lcd-2-hd650-m50-and-007mk2

See that gently sloping white line of -10 dB I drew from 1KHz to 20KHz? That is what Tyll (of InnerFidelity) used to recommend as the ideal frequency response curve. As you probably already noticed, the Harman Target Reponse Curve is similar, but with a few differences. I currently prefer the Harman Target Response Curve (or Tyll's slightly modified version of it) since it's more updated and closer to what a full-range speaker system sounds like.

*Now, lets start correcting the imbalance:*

You start with any frequency ranges--it doesn't really matter. The key is to constantly assess your correction with the pre-rendered sinewave test tones and the log sweep (and you don't have to playback the entire log sweep when only listening for a specific requency range. For example, if you are working on the 3 to 8KHz range, you can jump ahead to around halfway point to start the sweep at 1KHz).

I usually start with the narrowest bandwidth (using peak/dip mode, NOT low/high pass/shelf mode) when applying EQ at a specific frequency. I will adjust the gain and then use the pre-rendered sinewave test tones in that range to listen for balance, until I get perfect balance. Once I have achieved that, I'll use the log sweep and listen of any dips or peaks right before and after the exact frequency I'm correcting. If the sweep is not perfectly smooth, I'll adjust the bandwidth until it smooths out, and if necessary continue to adjust the gain too. This takes patience because you need to get the bandwidth just right so the correction is countering the imbalance precisely. You do this for every single frequency imbalane that is audible.

Sometimes, a headphone might have the entire treble or bass region recessed or bloated, and you can switch to high or low pass/shelf mode instead of peak mode. Adjust so that visually, you can see the correction applied to the entire affected region. Then within the shelved area, if there are frequencies that needs to be corrected, you can do that with more dip/peak bands.

Keep in mind that sometimes you need to have EQ bands very close to each other (for example, one at 6KHz and one at 7KHz). when corrections happen in frequencies that are in close proximity to each other, they will also counter or amplify each other, so this is why you cannot do corrections by numbers alone and must visually look at the curve itself to make sure you are getting the exact correction you need.

Ideally, once you have finished all corrections, the log sweep will sound far smoother and balanced compared to no correction. And when you play the pre-rendered sinewave test tones sequentially through the entire frequency range, you'll hear no significant imbalances in amplitude (other than the natural Fletcher-Munson curve).

EDIT: It appears some people were confused by exactly what my method involves, especially regarding whether the Fletcher-Munson equal loudness curve and Harman Target Response Curve are involved in the EQ process. I'll summarize my process here again:



> It's not just the headphone measurements and audible dips and peaks. The Harman Target Response Curve is also part of the EQ assessment. So basically, it's like this:
> 
> 1) Look at the frequency response measurement graph and note where the dips and peaks are and how far they deviate from the Harman Target Response Curve.
> 
> ...




And here's an article/video showing Bob Katz and Tyll of InnerFidelity talking about EQing headphones to match the Harman Target Response Curve as a great starting point: http://www.innerfidelity.com/conten...es-harman-target-response#lgR1tm6s3SeYdDEO.97

*Testing with musical material:*

While it's important to also use musical tracks to test the results of your EQ correction, I have noticed that many people don't know how to select musical material that makes it very easy to hear problems in their audio gear.

The trick with easy audio assessment isn't simply to use music that's mastered perfectly--it's often the opposite (though perfect mastering is useful too, but in a different way). Music that is mastered too bright is great for revealing problems in the sibilance range, such as assessing headphones that are too bright and fatiguing. These masterings that are the bright side should still be within tolerable threshold when playing back on audio gear that is neutral/accurate, but if the gear is too bright, these bright masterings will become unbearable, and that's a very easy way to test/correct overly bright headphones.

You also want music that contains ample low frequency information, so that you can easily hear when the headphone's sub is rolled off and sounds too anemic, or when it's excessively bloated and muddy, or if it lacks proper amount of punch, articulation, and speed.

For the mids, you want music that has enough focus on the mids, such as vocals and instruments that are in similar frequency range as vocals (such as string instruments, woodwind instruments, guitars, etc.) If the headphone lacks presence or bite in the mids, you will hear it, and if it's too aggressive and nasal, you'll hear it too.

And of course, music that has a great balance in its arrangement and mastering is good for assessing overall balance, imaging, depth, etc.

Over the years, I've put together a playlist that I use specifically for testing audio gear, and I'm now going to share them with you and show you how to use them. If you really like any of the tracks and want more information to hunt down the composer/artist so you can purchase their music, don't hesitate to ask to find out more. Here's the link to download the tracks:

*edited:sorry no link to copyrighted musics(signed evil modo)

(With the link to the MP3s removed, I've tried to find the songs on Youtube to link to instead. I couldn't find all of them, but found enough of them to be very helpful.)*

I'll group these tracks into categories based on how I use them. (Don't think about the actual genre/language of these tracks because you're using these tracks for testing audio gear, not for leisurely listening. If you happen to enjoy the music, that's just icing on the cake.)

And don't get hung up on the fact that these are MP3/YouTube audio. They are perfectly fine for the kind of tests you will be doing with them, and using uncompressed versions will not change anything. You'd have to focus intensely and strain really hard to hear any differences compared to the original CD quality (and you often won't hear any differences reliably if the YouTube video is HD) and at that point the differences are too subtle to be relevant in this specific context.

When you do comparisons of musical material, it's important to not switch between the EQ curve and the bypassed/unEQd sound very quickly without allowing your brain to first adapt to the sonic signature. That is not the correct way to do audio assessment. What you need to do is to listen for at least 30 seconds so your brain can adjust itself to that sonic signature. When you do switch, stay with it for at least 30 seconds before you switch again.

*Testing for bass:

04 - Love For Sale *- The upright bass in this track will sound too muddy and bloated with headphones that have too much bass. The bassline will overwhelm the rest of the frequencies too much and you won't hear the articulation of each bass note clearly. On neutral sounding gear, the bass notes will sound balanced with the rest of the arrangement.


*12.sub-ID%20Mi%20Woof* - This track can test just how deep and powerful the bass frequency response is. If your headphone's bass it too rolled-off, you will not hear/feel that really powerful and deeply resonant monster Roland 808 drum machine kick drum. It is one of the most famous kick drum sounds in the world (at least in hip-hop and electronic music), and you definitely want to be able to play it with enough authority and power. If the gear is too exaggerated in the bass region, the 808 kick drum will sound ridiculously monstrous and way out of balance. Also, if your headphones cannot handle the powerful bass of the 808 kick drum, it might distort.

*02 - klendathu drop *- At 0:49, you should hear a powerful sustained bass note centered at around 45Hz. If it doesn't sound powerful and majestic, then your headphone's sub-bass is too rolled off. If it sounds too overwhelming, then your headphone's sub-bass is too exaggerated.


*lucite-dry* - This short clip has bass kick that should sound punchy at with deep and powerful resonance (though not as low in frequency as the track with the 808 kick drum). If the bass kick sounds limp and weak, then your headphone's bass response is too anemic. If the bass kick sounds too overwhelming, then your headphone's bass is too exaggerated.

*03 - Mahalle* - Starting at 0:16, you should hear that deep sub-bass thud that accompanies the  strumming instrument each time it repeats. It should be very obvious and accents the start of each repeated strumming pattern. If it's not strong enough, the track will sound too thin.

*02 - L.F.O.* - At 1:38, you will hear just the sub-bass notes by itself. Like the 808 kick drum, if your headphones cannot handle the sub-bass energy, it could distort. On neutral gear those bass notes should sound clean and deep.



*Halo Theme* - The intro section of the Halo theme is great for testing sub-bas because it has prominent low sub-bass energy going all the way down to 20Hz. If your speakers or headphones cannot handle that much low sub-bass frequency, it will likely distort, or sound too anemic/wimply instead of majestic and epic.



*11 - Dogfighter* - Although this track is a hybrid of jazz fusion and orchestral, the bass kick should have a solid and beefy thud that's fairly strong, but not overwhelming.



*02 - 戦いの達命 (Battle of our lives) *- The big bass drum hits between 0:17-0:18 and 0:19-0:20 needs to sound strong and deep and really punctuate the orchestration with a sense power.

*bigbeat01* - Here's another deep sub-bass track. When the bassline starts at 0:24, it should be very full and deep and powerful, as if your entire head is rumbling. If you are not hearing that, then your headphone's sub-bass is not deep/powerful enough. If it's way too overwhelming and sounds like a muddy mess, then the sub-bass of your headphone is too exaggerated.

*track 06 - cats on mars (dmx krew remix)* - A fun track for sub-bass. The bass kick should should sound full and deep and with ample body. If it sounds weak then the sub-bass is too rolled off. If it's too overwhelming, the sub-bass is too exaggerated.



*Testing for the mids:

02 - Rivers Of Love *- This track's vocals need to sound clear, warm, and rich. At 0:54 when she sings "through a scene," the word "through" will sound too wooly if the headphone has problems with the mids such as being too thick sounding. Also, there are some spots in the vocals that can have a bit of sibilance if the headphone's too bright.



*156608_rp350_11* - This electric guitar track needs to have enough bite in the upper-mids in order to retain its sense of raw power. I distinctly remember when I first got the Aucez'e LCD-2 and tested it, I was disappointed by how recessed it's upper-mids were and how wimpy those guitars sounded.

*06 Rain (I Want A Divorce)* - This string arrangement needs to sound rich and full, and not thin and brittle. The bass notes that punctuates throughout the track needs to be deep and rich and really fill out the bass frequencies and provide an authoritative anchor for the arrangement.


*02 - 戦いの達命 (Battle of our lives)* - This track was already mentioned for the big bass drum hits, but the brass-heavy orchestration is also very good for testing the mids. Brass intruments, like distorted guitars, needs to have enough bite and presence to convey that sense of raw power.

*[BARBEE BOYS]BARBEE BOYS 2(07)‚Í‚¿‚ ‚í‚¹‚ÌƒƒbƒJ* - A J-Rock song from the 80's that can sound too aggressively bright if the headphone has too much emphasis in the upper-mids.


*Testing for upper-mids/brightness problems:

15 - Fade* - The bright sounding percussive chords that starts at 0:38 will sound fine on neutral sounding audio gear. It's bright but not grating. On gear that's too bright, they can sound piercing and painful.


*09 - 遠雷 (Distant Thunder) / 06- MORNING CALL *- Both of these tracks have intros that have bright shimmering sounds that repeat in patterns, and on headphones that are too bright, they can sound too sharp and fatiguing.



*05 - 少年は天使を殺す (The Boy Killed the Angel)* - This track is mastered quite bright and on neutral audio gear, the backing vocal chant that comes right after the opening drum beat will have sibilance that's very prominent, but it's still within tolerance and should not sound painful. But on audio gear that is too bright, it will become very annoying and even painful. This continues all the way through the song with both the main vocals and the backing vocals. Also, that snare drum can sound fatiguing on audio gear that's too bright.


*02 - Here's Where The Story Ends *- This track isn't overtly bright, but when the chorus starts at 1:10, the sibilance can become too hot and annoying on headphones that are too bright.


*01 - Already Yours* - This wall-of-distorted guitars track can get fatiguing fast if the headphone is too bright. On a neutral sounding audio gear, it's tolerable to listen all the way through, but any skew towards excessive brightness it'll become unbearable.


*03 PARTY* - This track's chorus starting at around 0:48 has some sibilance issues, and on neutral audio gear, it'll sound a bit hot in the sibilance but still tolerable. On gear that's too bright, those distinct sibilances will become painful.


*05 - 淋しいから言えないから (Because I Was Lonely, I couldn't Say)* - Morikawa Miho's voice during the chorus starting at 1:07 can sound overly sharp on audio gear that's too bright, but if the gear is neutral, she'll sound bright but not painful or grating.


*Testing for treble:

09 第七感（セッティエーム サンス) (Seventh Sense)* - This track has really tizzy and splashy treble. It's still within tolerance on neutral audio gear, but on gear that's too bright in the treble, it'll become really tizzy and annoying, and maybe even painful.


*10 William, It Was Really Nothing *- This track also has a lot of treble energy, but it's not quite as drastic as Seventh Sense. On headphones with exaggerated treble, it will sound like a splashy mess instead of shimmering.


*For overall balance, clarity, and richness in complex arrangements:

Mojo_Madness /  ST_Theme_v1 / 03 Come in 007, Your time is up / 07 BROTHER *- These action/adventure/drama oriented score cues have rich orchestrations and I like to use them to listen for overall richness, balance, clarity, dynamic range, imaging, depth, etc.



The forum only allows 30 media links per post, so I'm moving the ones past #30 to another post. Here's the link to that post:
https://www.head-fi.org/threads/the...nd-for-non-professionals.796791/post-15562131

*Wrapping up:*

Okay, I hope this has been helpful to those of you who always wanted to try EQing your headphones but don't know how, or felt confused/lost when using pink noise and sinewave generators. My method of using logarithmic sweep tone and pre-rendered sinewave test tones is much more straightforward and intuitive, and anyone should be able to do very precise EQ curves using my method and achieve the most neutral/accurate frequency response for any headphone (unless that headphone's frequency response and physical drivers are so bad that it just can't be corrected properly).


----------



## Sound Eq

you are a legend now time to dig deep into all you wrote


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## hqssui

Very interesting and helpful. Thanks


----------



## AudioBear

Really useful and clearly written.  Thanks.  You have given us much to think about.
  
 I use a Mac so I assume EasyQ isn't an option.  For other reasons I have a plug-in called DMG QuickEQ available but it is not free (http://dmgaudio.com/products_equality.php).  I like it and found that although it's intended for Pro Tools it works with Audirvana among others (also Fidelia).  Do you have any other Mac OS X software recommendations?


----------



## spruce music

A perfectly flat and accurate transducer playing a log sweep would have perceived up and down areas due to Fletcher-Munson.  So would it not make sense to pre-EQ the log sweep with the Fletcher-Munson curves?  That way a correct sweep will have nearly equal volume bottom to top and your EQ should leave you much closer to flat.  Then once you have flat you could make adjustments toward the Harman suggested response.


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## AudioBear

From my understanding the F-M curves are a volume dependent family of curves, not just a single curve.  So if you have a way to play at constant volume (whatever that means depends on your definition) that might work.  My guess is that you could approximate one solution for average listening volume but no one corrected curve will fit all listening volumes.


----------



## spruce music

audiobear said:


> From my understanding the F-M curves are a volume dependent family of curves, not just a single curve.  So if you have a way to play at constant volume (whatever that means depends on your definition) that might work.  My guess is that you could approximate one solution for average listening volume but no one corrected curve will fit all listening volumes.


 

 Yes, the curves vary a bit by volume.  One could of course choose the right curve for the volume used.  Even without that an average of the curves is much closer than doing nothing.


----------



## Vkamicht

lunatique said:


> *Why would you want neutral/accurate sound? Isn't that a bad thing for musical enjoyment? Doesn't neutral mean boring, cold, thin, sterile, clinical, etc.?*
> 
> One of the most popular but grossly mistaken misconceptions in the hobbyist headphone/audiophile world, is that neutral/accurate sound means "clinical," "sterile, "cold," "bright," "sharp," "boring," etc. I have tried to to do my part to educate others about this for years in forums but it's such a pervasive misinformation that it gets passed around and regurgitated at frightening frequency--I just can't seem do anything to turn the tide (neither can other informed people who are also trying to combat this type of pervasive ignorance).
> 
> ...


 
  
 Thank you for writing this. "Neutral/accurate" sound (as you put it) is probably the only thing I've heard that isn't boring, to my ears. Nothing is under or over-represented. Music "just is." I'm consistently every day, wowed by my system, and disabling my EQ makes my HD-650s sound like a tin can. That alone should clear up any misconceptions about accurate sound being cold or sterile.


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## Music Alchemist

Awesome. I have added your thread to my list of EQ threads to share with those in need. The others are:
  
 http://www.head-fi.org/t/413900/how-to-equalize-your-headphones-a-tutorial
 http://www.head-fi.org/t/587703/how-to-equalize-your-headphones-a-tutorial-part-2
 http://www.head-fi.org/t/615417/how-to-equalize-your-headphones-advanced-tutorial-in-progress
 http://www.head-fi.org/t/794467/how-to-equalize-your-headphones-2016-update
  
 Edit: They're now in my signature.


----------



## Lunatique

audiobear said:


> Really useful and clearly written.  Thanks.  You have given us much to think about.
> 
> I use a Mac so I assume EasyQ isn't an option.  For other reasons I have a plug-in called DMG QuickEQ available but it is not free (http://dmgaudio.com/products_equality.php).  I like it and found that although it's intended for Pro Tools it works with Audirvana among others (also Fidelia).  Do you have any other Mac OS X software recommendations?


 
 I'm not a Mac person, so I can't help you there. Your best bet is to search the KVR Audio database for Mac-compatible plugins. You'll definitely find one you'll like, since Mac is very popular with professional audio/music folks.
  


spruce music said:


> A perfectly flat and accurate transducer playing a log sweep would have perceived up and down areas due to Fletcher-Munson.  So would it not make sense to pre-EQ the log sweep with the Fletcher-Munson curves?  That way a correct sweep will have nearly equal volume bottom to top and your EQ should leave you much closer to flat.  Then once you have flat you could make adjustments toward the Harman suggested response.


 
 There actually is one readily available, but the loudness level problem already mentioned by AudioBear makes it less useful. The people to created the "perceptual sweep tone" had to make it very low in volume in order to combat that problem. You can find it here:
 http://www.audiocheck.net/testtones_perceptualsinesweep.php
  


vkamicht said:


> Thank you for writing this. "Neutral/accurate" sound (as you put it) is probably the only thing I've heard that isn't boring, to my ears. Nothing is under or over-represented. Music "just is." I'm consistently every day, wowed by my system, and disabling my EQ makes my HD-650s sound like a tin can. That alone should clear up any misconceptions about accurate sound being cold or sterile.


 
 I just recently posted my updated EQ curve for the HD650, to get it closer to the Harman Target Response Curve:
  
 http://www.head-fi.org/t/785154/need-recommendations-for-most-neutral-accurate-yet-musical-and-enjoyable-iem-in-sub-1-000-range-going-over-is-okay-if-its-really-worth-it/45#post_12149531


music alchemist said:


> Awesome. I have added your thread to my list of EQ threads to share with those in need. The others are:
> 
> http://www.head-fi.org/t/413900/how-to-equalize-your-headphones-a-tutorial
> http://www.head-fi.org/t/587703/how-to-equalize-your-headphones-a-tutorial-part-2
> ...


 
 Thanks! Hopefully we can help educate the community on how to attain aural bliss so they can get so much more out of their headphones without needing to spend much or any money.


----------



## Music Alchemist

lunatique said:


> I just recently posted my updated EQ curve for the HD650, to get it closer to the Harman Target Response Curve:


 
  
 Here's an interesting question: With planar magnetic headphones, do you feel the need to boost the bass? Because despite being widely regarded as having neutral bass, they follow the diffuse-field curve in the bass instead of the Harman curve.
  
 In my experience, the diffuse-field curve sounds more accurate to me. I say this because I owned a STAX SR-207 that follows that curve very closely, and it just sounded so much more neutral to me than other headphones, including very high-end ones.
  
 But you can equalize headphones to follow either (or any other) curve, so it's no biggie.


----------



## castleofargh

I removed the link to download your songs, maybe if you were to cut them into really short passages for testing purpose it would be ok, but I can't just let you post full songs online like that.
 I'm sorry as it certainly screw up the last part of your post, but that's not even headfi's rule, that's legal real world stuff.
  
  
 now about the post itself, it's full of very good advices, and I believe people can learn a few things from it. but IMO the title is kind of BS
	

	
	
		
		

		
			





. I don't see how this all process leads us to neutral sound. I'd call it "remove the big spikes and dips from your headphone", that would be much closer to what can be achieved here. and that's already an amazing progress for many headphones.
 also the guy doing what you suggest to the letter will end up with his equal loudness contour. not with a neutral signature. or did I miss something? you warn about both ends of the audible range, but even in the mids it's not a straight line. given how you EQ your stuff, I suspect that you're used to hearing sweeps on relatively neutral gears, and you work with your experience of how a sweep should sound(which is fine and works pretty well). but the guy with no experience or neutral reference will simply go with how loud he feels the sound is, and that's an equal loudness contour, not a flat frequency response.
  
  
 I'm going to try your EQ for the hd650, I'm surprise to see nothing in the 5/6khz, as that's pretty much what's missing to your EQ to look strongly like mine. but I'm always interested in trying EQs. would you be so kind to give me your exact values for the EQ (you can just edit the EQ file made by easyQ with a notepad and copy the values).


----------



## Lunatique

music alchemist said:


> Here's an interesting question: With planar magnetic headphones, do you feel the need to boost the bass? Because despite being widely regarded as having neutral bass, they follow the diffuse-field curve in the bass instead of the Harman curve.
> 
> In my experience, the diffuse-field curve sounds more accurate to me. I say this because I owned a STAX SR-207 that follows that curve very closely, and it just sounded so much more neutral to me than other headphones, including very high-end ones.
> 
> But you can equalize headphones to follow either (or any other) curve, so it's no biggie.


 
 I think it depends on the actual frequency response of the individual headphone model. For example, the LCD-2 is almost perfectly flat down to 20Hz, but it's not "exactly" perfectly flat--it's short by a small few dB in the sub-bass. At the same time, even if it's EQed to be perfectly flat, you still have to deal with the fact that headphones don't transmit physical vibrations the way a full-range speaker system could, so it'll "feel" like the sub-bass is a bit lacking in visceral impact. This is why the Harman Target Response Curve adds a few more dBs to the sub-bass region to compensate for that, and I do that for my EQ curves as well.


----------



## Lunatique

castleofargh said:


> I removed the link to download your songs, maybe if you were to cut them into really short passages for testing purpose it would be ok, but I can't just let you post full songs online like that.
> I'm sorry as it certainly screw up the last part of your post, but that's not even headfi's rule, that's legal real world stuff.
> 
> 
> ...


 
 Yeah, I debated with myself whether it was a good idea to include those tracks in full, or spent the extra time to edit them into just usable snippets. 
  
 What if I included Youtube links that already exists instead? I'm assuming that's okay since Youtube videos are linked everywhere in the forums here. 
  
 Here's the EQ curve preset for the HD650. I just updated it to bring 8KHz down since it's too prominent compared to the relative amplitude of 6.3KHz and 10KHz. 
  
 <?xml version="1.0" encoding="UTF-8"?>
 <Equalizer PatchFormat="2">
   <Band Mode="Low Shelving" Frequency="44.7744228" Gain="6" Bandwidth="2.44"/>
   <Band Mode="High Shelving" Frequency="14000" Gain="5" Bandwidth="1.92"/>
   <Band Mode="Low Shelving" Frequency="119.132429" Gain="5" Bandwidth="1.82"/>
   <Band Mode="Peak/Dip" Frequency="1200" Gain="3" Bandwidth="2.5"/>
   <Band Mode="Peak/Dip" Frequency="8000" Gain="-6" Bandwidth="0.25"/>
 </Equalizer>
  
 As for equal loudness contour (Fletcher-Munson curve), I think all of us have an ideal listening level where we feel the music is the most dynamic, immersive, and pleasurable, without being so loud as to cause us discomfort. I would recommend everyone to do their EQ curve that their favorite listening volume. For me, it's around 78~80 dB average, with the rare sudden peaks within the 85 dB limit. I know this because I measured my ideal listening volume with a SPL meter (sound pressure meter) and then volume-matched the headphones with my speaker system using pink noise. If the person has to listen to the headphones at lower or higher than their favorite/most used listening volume, I supposed they can create variant EQ curver that compensates for it.


----------



## castleofargh

yeah youtube is very fine. I'm sorry that it makes more work for you but the law is the law.
  
 thank you for the EQ, I'll try that tomorrow seriously(it's 1.30AM for me right now ^_^)
  
 the loudness has an impact for sure. I imagine it would work better at louder volumes as the FM loudness curve flattens out a little, but loud test tones, that's not a lot of fun.
	

	
	
		
		

		
		
	


	




 
 still I do not feel like equal loudness is the same as perceived neutral. and you're making people do equal loudness(and then correct it with actual music). that's why I said it's good for spikes, but not to get overall neutral sound IMO.  I doubt anything but a neutral reference from good speakers can let people achieve neutral on headphones. but again I'm good with your method, it will improve some things on most headphones, and let people not too far off of neutral, so they can fine tune by a few dbs later on with music. or maybe apply a default fletcher munson compensation? we all have our own + the loudness stuff, but the general direction would still be good I guess.
  
 when I EQ my music, the ultimate judge is.... November rain 
	

	
	
		
		

		
		
	


	




 on this video starting at around 7mn, when my EQ is right, 1/ the guitars on both sides feel balanced, and 2/ the third one in center must dominate clearly the other 2, and 3/not be overwhelmed by the percussion. then if the voices are good it's even better, but if I get it right with the guitars+percussion, I know music will sound nice. ^_^
 I wish my reference song was the Al Jarreau version of blue rondo a la turk, or some fancy Pink Floyd stuff, or the usual Bach organ stuff. but nope, the one song that does it for me is November rain


----------



## Music Alchemist

"Neutral" is kind of a loaded word, especially due to the debate over which compensation curve is the most neutral one.
  
 I think we can all agree that "more accurate" is a more accurate (*snicker*) term to use.


----------



## castleofargh

neutral exists, it's whatever we get flat on some graph
	

	
	
		
		

		
			





.
  
 with a DAC electrically flat is neutral. with speakers it's the same, even if we don't get perfect flat, having flat speakers will sound "neutral" to us, as in real life usual sound.
 so I believe there is the exact same neutral for our ears with a headphone. it's probably an individual quest at some point, but if my headphones could sound to me like they have the same signature as good speakers, I'm very confident that would be my neutral.
 accurate may involve more than just frequency response no?


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## Music Alchemist

castleofargh said:


> neutral exists, it's whatever we get flat on some graph
> 
> 
> 
> ...


 
  




  
 Where did you get the idea that a flat response with loudspeakers or headphones looks like a flat line?
  
 Because any of the curves (diffuse-field, free-field, Harman, etc.) that represent what our ears hear are not a flat line.
  
 And neutral-ish headphones (or any headphones for that matter) most certainly do not measure as a flat line.
  
 You have to compensate for human hearing. Transducers (headphones and speakers in this case) are not like electronics in this respect. I'm surprised someone of your stature was not aware of this. The only time it would look like a flat line is if you were comparing post-equalization measurements to a compensation curve and managed to get them to align perfectly.
  
 Some reference links:
 http://www.innerfidelity.com/headphone-data-sheet-downloads
 http://www.innerfidelity.com/headphone-measurements-explained
 http://www.innerfidelity.com/content/harman-researchers-make-important-headway-understanding-headphone-response
 http://www.innerfidelity.com/content/headphone-target-response-curve-research-update
  
 But I may have misinterpreted you. I thought you meant a literal flat line on a graph. If not, disregard my rant. ^_^
  
 In any case, I'm curious which compensation curve you feel is the neutral one for headphones, and why.


----------



## RRod

music alchemist said:


> Where did you get the idea that a flat response with loudspeakers or headphones looks like a flat line?
> 
> Because any of the curves (diffuse-field, free-field, Harman, etc.) that represent what our ears hear are not a flat line.
> 
> ...


 
  
 It's all about where you're talking about the measurement. When we say "flat" speakers we mean "measured at a given distance and orientation in an anechoic environment." Once you put a reflective room into the mix then you compensations to worry about, but the paradigm of "flat from dac to amp to speakers" is a good one, because it removes variables. Similarly we can define a flat headphone as "matches the at-ear-canal response of flat speakers in a subjectively 'nice' room." A bit more fuzzy, but still something measurable. I don't think argh was trying to say we want cans to measure ruler flat in and of themselves (if I can put English words into his mouth).


----------



## Music Alchemist

rrod said:


> It's all about where you're talking about the measurement. When we say "flat" speakers we mean "measured at a given distance and orientation in an anechoic environment." Once you put a reflective room into the mix then you compensations to worry about, but the paradigm of "flat from dac to amp to speakers" is a good one, because it removes variables. Similarly we can define a flat headphone as "matches the at-ear-canal response of flat speakers in a subjectively 'nice' room." A bit more fuzzy, but still something measurable. I don't think argh was trying to say we want cans to measure ruler flat in and of themselves (if I can put English words into his mouth).


 
  
 Okay, as long as he didn't mean a literal flat line.


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## Lunatique

Okay, I just edited the original post with as many Youtube links to the songs as I can find. I tried to find the highest resolution versions on Youtube too, whenever I could.


----------



## castleofargh

music alchemist said:


> rrod said:
> 
> 
> > It's all about where you're talking about the measurement. When we say "flat" speakers we mean "measured at a given distance and orientation in an anechoic environment." Once you put a reflective room into the mix then you compensations to worry about, but the paradigm of "flat from dac to amp to speakers" is a good one, because it removes variables. Similarly we can define a flat headphone as "matches the at-ear-canal response of flat speakers in a subjectively 'nice' room." A bit more fuzzy, but still something measurable. I don't think argh was trying to say we want cans to measure ruler flat in and of themselves (if I can put English words into his mouth).
> ...


 

 yeah what RRod said.  electrically flat headphones would sound like crap of course ^_^. I meant to say that neutral could be seen as the notion of having a flat line somewhere on some graph. you just have to stop and say "this is the new flat". we can just take whatever signature, calibrate with that and now that signature is "neutral" 
	

	
	
		
		

		
		
	


	




 in that given compensation, and would give a flat line when measured that way. 
 it would have little to do with my perceived balance of frequencies which would be my "neutral", but a totally different thing. that was a little ironical. sorry for the lost in translation.


----------



## spruce music

castleofargh said:


> yeah youtube is very fine. I'm sorry that it makes more work for you but the law is the law.
> 
> thank you for the EQ, I'll try that tomorrow seriously(it's 1.30AM for me right now ^_^)
> 
> ...




 All of that is why I think the log sweep would benefit being pre-EQ'd for Fletcher Munson.  Pick the 70 or 80 phon version.  It will not be perfect, but will be very much in the right direction.  Otherwise, as you have been saying castleofargh inexperienced people will end up with their personal loudness contour.  Taking most of this out by EQ of the sweep will put them much closer from the beginning to flat response which most definitely is not the loudness contour curve.


----------



## Music Alchemist

castleofargh said:


> yeah what RRod said.  electrically flat headphones would sound like crap of course ^_^. I meant to say that neutral could be seen as the notion of having a flat line somewhere on some graph. you just have to stop and say "this is the new flat". we can just take whatever signature, calibrate with that and now that signature is "neutral"  in that given compensation, and would give a flat line when measured that way.
> it would have little to do with my perceived balance of frequencies which would be my "neutral", but a totally different thing. that was a little ironical. sorry for the lost in translation.


 
  
 I see. So we're on the same page after all! The only problem is which compensation curve to use. That is a heated debate in itself.


----------



## Lunatique

castleofargh said:


> yeah what RRod said.  electrically flat headphones would sound like crap of course ^_^. I meant to say that neutral could be seen as the notion of having a flat line somewhere on some graph. you just have to stop and say "this is the new flat". we can just take whatever signature, calibrate with that and now that signature is "neutral"
> 
> 
> 
> ...


 
 I'll clarify what I meant exactly by neutral/accurate.
  
 In my studio, at my listening position, the measuring mic of the ARC System 2 takes measurements of my speaker system (full-range mastering grade 2.1 system that includes the Klein+Hummel O 300Ds and Neumann KH805) and the effects of room mode due to any acoustic issues that's not fully taken care of by the acoustic treatments in my studio. Also, the exact placement of my speaker system was the result of an on-going consultation with Neumann's product portfolio manager of studio monitor systems (as some of you might know, Sennheiser is the parent company of Neumann). I'd take measurements and then show the screenshot to him, as well as take photos of the studio. He'd give me a set of tips and instructions, and I'd carry them out (tweaking the hardware settings on the speakers themselves, testing various positions with measurements, do the famous "subwoofer crawl," record a $hitload of data and calculating them, etc., then take measurements and photos again, and repeat the process over and over. The entire process took over two months of refining, and finally I was able to achieve the most neutral/accurate response I have ever gotten without any corrections. Then to finish it off, I take one final measurement at the listening position with the ARC System 2, and it corrects the frequency response and time-domain issues and gives me perfect 20Hz-20KHz frequency response and precise imaging with no time-domain issues. 
  
So that is my golden standard for neutrality/accuracy--a full-range mastering grade 2.1 sound system that does 18Hz-20KHz and is measured perfectly flat from 20Hz-20KHz at the listening position, in an acoustically treated studio. 
  
 With headphones, my goal is always to try to get as close to that as possible. So, no, I would not want a perfectly flat line for a headphone's frequency response, and I mentioned this in the original post when I posted those links to InnerFidelity's articles on understanding headphone measurements and the Harman Target Response Curve.


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## Music Alchemist

lunatique said:


> With headphones, my goal is always to try to get as close to that as possible. So, no, I would not want a perfectly flat line for a headphone's frequency response, and I mentioned this in the original post when I posted those links to InnerFidelity's articles on understanding headphone measurements and the Harman Target Response Curve.


 
  
 Yes, but he did confirm that he wouldn't want a flat line for the actual measurements either; he was just referring to aligning the equalized response of the headphone with a compensation curve, then using that compensation curve as the new flat line, so to speak, similar to how Golden Ears and other projects do it.
  
 So since you got such an accurate response for the speakers...would you say that headphones sound like that when equalized to the Harman curve?
  
 Various other companies and individuals use their own compensation curves. I'm very interested in learning which ones are the most accurate.


----------



## Lunatique

music alchemist said:


> Yes, but he did confirm that he wouldn't want a flat line for the actual measurements either; he was just referring to aligning the equalized response of the headphone with a compensation curve, then using that compensation curve as the new flat line, so to speak, similar to how Golden Ears and other projects do it.
> 
> So since you got such an accurate response for the speakers...would you say that headphones sound like that when equalized to the Harman curve?
> 
> Various other companies and individuals use their own compensation curves. I'm very interested in learning which ones are the most accurate.


 
 I was responding to them wondering if I meant I wanted totally flat response for headphones. 
  
 Speakers will always sound better than headphones. The reason is because soundwaves from speakers must travel in the air and interact with the molecules in the air and excite them before entering our ears, as well as interact with the acoustic space of your listening space, and also the actual vibrations that our body can sense. That simple fact makes speakers far more visceral, dynamic, alive, dimensional, and have wider and deeper soundstage. It's possible to get headphones to sound like speakers in terms of frequency response, but in other aspects it's extremely hard or maybe even impossible to get them to match speakers. I personally don't think it's possible without some really high-quality and complex DSP magic, such as using real recorded impulse response from actual speakers in a listening space, and even then it probably won't be 100%.
  
 I would pick my speaker system over any headphone, any day, no contest, especially after I've spent the past 8 years tweaking and perfecting that system (including the actual studio space) to the absolute pinnacle of performance. I don't know if you guys realize this, but it's extremely rare for those in the head-fi community to achieve perfect neutral/accurate sound with a speaker system that goes from 18Hz to 20KHz. Those who can afford it often don't know about or use room correction, and those who know about room correction often can't afford it. Those who do know about it and can afford it often don't know enough about acoustics and room modes to know where exactly to place the speakers/subwoofer in their listening space to avoid significant nulls and peaks caused by room modes. Even if they did research about it online, there is a lot information missing that you really have to consult the experts extensively in order to achieve this goal. For me, it took prolonged correspondences with an engineer at Klein+Hummel and a product manager at Neumann to finally get my speaker system to sound this amazing. 
  
 Essentially, I would never put on a pair of headphones unless I absolutely need to. And when I do, I want the headphone to sound as close to my speaker system as possible. Even if I know I'll never get it to sound close enough, it's still going to sound much better than if I don't try to correct the headphone at all.


----------



## spruce music

The problem with EQ of headphones is what is the reference?  With speakers you can measure it with calibrated gear.  With headphones they interact with each individuals ear and fit so much that won't work (at least for DIY efforts).
  
 Now just as a thought experiment suppose we had what we already knew was a truly flat phone for the person wearing it. If we play a log sweep it will not sound of even loudness.  If the user then adjusts for even loudness every adjustment is moving away from flat accurate response.  The user ends up with his own personal equal loudness curve and not accuracy. 
  
 Now let us flip it around suppose we also know not only our phone is truly accurate, but the equal loudness contour for the user.  If we EQ the log sweep for the users equal loudness curve and play it back over the flat phone the user will hear even loudness across the band.  Now if the user swaps out his headphones for another model not flat in response, he could play the EQ'd for equal loudness log sweep and adjust the second headphone until equal perceived loudness across the band was achieved.  This second phone would now be EQ'd in a way to provide flat response. 
  
 Now I would suggest Fletcher-Munson curves are known to be generally close on average for the majority of people.  They aren't perfect for every person especially at high frequencies with the effects of aging. But using the curve for something like the 80 phon level should be at least in the general ballpark.  So doing EQ to your log sweep based upon that is much more than likely to be closer to flat that not doing it. Then once you have EQ for your phones as close to flat, you can use the suggested Harman EQ or others.  I have in the past used 3 db/decade slopes starting at 200 hz.  So down -3db for 2khz and -6 db for 20 khz.  Actually you should use the latest ISO standard for this.
  
 http://www.sengpielaudio.com/Acoustics226-2003.pdf
  
 Another possibility is using your speakers as a reference if they are better than your phones in flatness.  Or using your speakers to develop your own equal loudness curve which then can be applied to pre EQ the log sweep to adjust headphones for equal perceived loudness.  It isn't perfect, but it is a reference to adjust off of which is far better than nebulous adjusting to taste with no reference touchstone. 
  
 It is possible using the Harman Headphone target response curve is actually better to EQ the log sweep.  I have not tried that.  If the right curve it should work more or less. EQ the log sweep with the Harmon curve and adjust for equal loudness.  I suspect this might end up too bright if adjusted to equal loudness.  Apply some tasteful tilt and it should be good.


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## castleofargh

yup, the ideal situation is to have a nice flat speaker system calibrated in a nice room. then we have one reference and there are ways to try and get something similar (like following @Joe Bloggs tuto that does involve equal loudness contour for 2 devices as a way to know the FR response difference between the 2).
  
 but without a reference headphone/speaker, it's pretty hard to know where we're going just with loudness. it's a lot of trial and error before getting something that does sound clear everywhere. now there seem to be some common grounds between hearing, measurements, taste, etc. at least for maybe 200hz to 7khz ou 8khz when I get something I like, I can just take another headphone, EQ it from measurements and get something not too far off in that range.
 where I get totally lost is when I try to EQ an open back and a closed back, or for IEMS, a vented dynamic driver and a sealed balanced armature driver. measurements seem to be very different to what I'm hearing from each techs(and that's the only reason why I keep EQing by ear for some IEMs).
 like right now I'm in love with the fiio EX1(AKA titan1), I find that it's a very good EQ material, the low end doesn't drop much if at all, and the trebles are too strong, so easy to just turn down without having to fear distortions. but it's not isolating so it's use is limited sadly.  I measured it with the vibro veritas coupler and got a flat mid and low end response, something rather close to what I got from measuring my hf5.  but here is the thing, I always feel like I need more low end on the hf5, when I find the EX1's low end to be amazing...
	

	
	
		
		

		
		
	


	




 I'm guessing the coupler doesn't react the same way if it's sealed of not, or other acoustic reasons I don't understand. but I clearly reach some situations like that where measurements just don't make for a proper comparison at all.
  
 and I don't have a good pair of speakers I can calibrate and rely upon as reference. until I move to some other place, speakers are a no go for me.


----------



## Music Alchemist

lunatique said:


> I was responding to them wondering if I meant I wanted totally flat response for headphones.
> 
> Speakers will always sound better than headphones. The reason is because soundwaves from speakers must travel in the air and interact with the molecules in the air and excite them before entering our ears, as well as interact with the acoustic space of your listening space, and also the actual vibrations that our body can sense. That simple fact makes speakers far more visceral, dynamic, alive, dimensional, and have wider and deeper soundstage. It's possible to get headphones to sound like speakers in terms of frequency response, but in other aspects it's extremely hard or maybe even impossible to get them to match speakers. I personally don't think it's possible without some really high-quality and complex DSP magic, such as using real recorded impulse response from actual speakers in a listening space, and even then it probably won't be 100%.
> 
> ...


 
  
 Loudspeakers may sound better than headphones for some people, but not me. Sure, loudspeakers can do plenty of things headphones can't, and they do sound big and impressive...but also distant and impersonal. I don't connect with the music at all with them. I prefer headphones nearly 100% of the time, partially due to the more intimate presentation that makes me feel one with the music.
  
 Here's one thing that can make headphones sound more like speakers, but it's a little pricey: http://www.smyth-research.com
  
 If you don't mind divulging, how much did it cost to do all that?
  
 Also, I'm still curious about my question relating to whether Harman-equalized headphones sound like your speaker system in terms of frequency response.


----------



## watchnerd

music alchemist said:


> Loudspeakers may sound better than headphones for some people, but not me. Sure, loudspeakers can do plenty of things headphones can't, and they do sound big and impressive...but also distant and impersonal. I don't connect with the music at all with them. I prefer headphones nearly 100% of the time, partially due to the more intimate presentation that makes me feel one with the music.


 
  
 So when you listen to live music does it sound distant and impersonal to you?


----------



## castleofargh

I'm also one to find that speakers are better than any headphone. simply because they're so close to the experience of real sound. when headphones, as long as the music is mastered with speakers, it will never be completely right (or we'll have to go with DSPs and HRTF like the realiser device). also bass vibrating into the body, that's something special to me.
 now all isn't dark for headphones, they actually tend to measure way better than speakers on many aspects. so the potential is certainly there. but we need stuff mastered for headphones or some other trick.


----------



## Music Alchemist

watchnerd said:


> So when you listen to live music does it sound distant and impersonal to you?


 
  
 Sometimes. lol. I see your point, but for me it's different. Headphones are just magical for me despite their limitations.
  
 I'm a fairly experienced musician too. From my profile:
  
 "I have been in everything from orchestras to jazz bands to metal bands, and have done vocals practically my entire life, piano since ~1993, and trombone and guitar since ~1996."
  


castleofargh said:


> I'm also one to find that speakers are better than any headphone. simply because they're so close to the experience of real sound. when headphones, as long as the music is mastered with speakers, it will never be completely right (or we'll have to go with DSPs and HRTF like the realiser device). also bass vibrating into the body, that's something special to me.
> now all isn't dark for headphones, they actually tend to measure way better than speakers on many aspects. so the potential is certainly there. but we need stuff mastered for headphones or some other trick.


 
  
 Since you've seen even more anime than I have, do you enjoy watching anime more on a TV with speakers across the room? I enjoy it more watching it on my laptop with headphones, personally.
  
 But for gaming, I do prefer a TV and speakers most of the time, unless it's a portable console.


----------



## briskly

music alchemist said:


> "Neutral" is kind of a loaded word, especially due to the debate over which compensation curve is the most neutral one.


 
 Neutral curve has a simple answer: headphone insertion gain matched to or approximately close to the subject's diffuse field transfer function. You would get that using reference noise and sweeps.
  
 The problem is that people tend not to like that, which is what inspires all this research. Harman takes the position that a downward sloping response, from room absorption and LF bumps, in speaker listening seem more natural to listeners, and thus should be reflected in headphones.


----------



## Music Alchemist

briskly said:


> Neutral curve has a simple answer: headphone insertion gain matched to or approximately close to the subject's diffuse field transfer function. You would get that using reference noise and sweeps.
> 
> The problem is that people tend not to like that, which is what inspires all this research. Harman takes the position that a downward sloping response, from room absorption and LF bumps, in speaker listening seem more natural to listeners, and thus should be reflected in headphones.


 
  
 There are a few things your post made me think about. The first post of this thread suggests using reference noise and sweeps. But then, he also suggests following the Harman curve instead of diffuse-field. I'm not sure whether the OP implied that simply using the reference noise and sweeps would get you a response that follows either curve in itself. And some have said that the equal loudness contour thing would result from that instead of a response that follows either curve. Thoughts?


----------



## briskly

music alchemist said:


> There are a few things your post made me think about. The first post of this thread suggests using reference noise and sweeps. But then, he also suggests following the Harman curve instead of diffuse-field. I'm not sure whether the OP implied that simply using the reference noise and sweeps would get you a response that follows either curve in itself. And some have said that the equal loudness contour thing would result from that instead of a response that follows either curve. Thoughts?


 
 Our goal in this case of equalizing by ear is get a response that seems flat. Diffuse field response is not intended, but that is the result these tend to follow.
 Loudness adjustments will be needed when done by ear, especially as dependent on level. The middle and inner ear have compression mechanisms that increase the input range of sound, and also affect the transfer response when active.
  

 This is the response of the loudspeaker in the listening room used in the Harman tests, equalized to a flat response over the listening position.


 Then, that room response is measured on a not-mannequin at eardrum equivalent, with an HD 800 matched to that response.
 The curve shown here has a strong resemblance to the diffuse field response of human subjects, as measured at the eardrum. The end headphone result has more bass and less treble than the flat room-equivalent response shown here, a result also shown with a slightly different magnitude in the preferred speaker target.
  
  
 From a different Harman study.

 The blue trace is the response from microphones inside the headphone, the red trace is the response determined subjectively, with loudness adjusted pink noise bands. These are both roughly in the range of the diffuse field adjusted response of the headphone under test.
 The subjective responses don't seem to indicate a particularly "analytical" headphone, but practically every Head-Fi review of this set discusses its lack of bass, lean tone, lack of warmth, and so on.


----------



## Music Alchemist

briskly said:


> Our goal in this case of equalizing by ear is get a response that is flat. Diffuse field response is not intended, but that is the result these tend to follow.


 
  
 Good info, but it doesn't seem to define what "flat" is in this context.
  
 Perhaps you could put together your own guide to supplement the first post of the thread and show exactly what to do to achieve the desired result, in your opinion.


----------



## Lunatique

music alchemist said:


> Loudspeakers may sound better than headphones for some people, but not me. Sure, loudspeakers can do plenty of things headphones can't, and they do sound big and impressive...but also distant and impersonal. I don't connect with the music at all with them. I prefer headphones nearly 100% of the time, partially due to the more intimate presentation that makes me feel one with the music.
> 
> Here's one thing that can make headphones sound more like speakers, but it's a little pricey: http://www.smyth-research.com
> 
> ...


 
 How do you listen to your speakers though? What kind of listening space? What kind of speakers? At what distance from the listening position? Is the space acoustically treated to adequate level for critical monitoring? Is there room/speaker correction applied? All of these factors can make or break your speaker listening experience. If your setup is not optimal then it's understandable that it won't sound as good as your headphones. But a speaker system that's of high quality and properly set up, will beat the vast majority of headphones on the market with the exception of maybe the absolute apex of headphones that cost more than some cars. 
  
 I'm aware of the Smyth Research products and used to follow it. I don't feel the need to go to that extent for my headphone listening experience. I'm satisfied with TB Isone (still the best HRTF plugin for headphones I've heard to date):http://www.toneboosters.com/tb-isone/ It is created by Jeroen Breebaart, an audio engineer with some impressive credits under his belt: http://www.jeroenbreebaart.com/ It sounds realistic enough that several times in the past I had forgotten I was wearing headphones and thought I was blasting my speakers late into the night and panicked.  Many users of Isone have reported similar experiences. 
  
 You don't need to spend a ton of money to achieve a great setup with a speaker system (but this is obviously relative, since for some people even a couple thousands of dollars is a fortune). If I were to help someone put together a great speaker system in a good listening space that can stand up to critical scrutiny (that even a high-end audio professional can approve of), it'll probably cost the person about $4,000 total (this includes a pair of professional quality speakers that have amazing bang for the buck, acoustic treatment kit, and room/speaker correction). If the person is willing to go the DIY route and buy used, that total can go down even lower. 
  
 I'll give you a couple of example packages I would put together for the person:
  
*Package One:*
 Event Opal monitor speakers
 http://www.sweetwater.com/store/detail/Opal/?adpos=1t1&creative=88122131521&device=c&matchtype=b&network=g&gclid=Cj0KEQiAoby1BRDA-fPXtITt3f0BEiQAPCkqQcCFsfSPkHhA_RWSWxYY7ZExQJJOdmMAbCyzqf7tBBQaArPh8P8HAQ
  
 GIK Acoustics Room Kits
 http://www.gikacoustics.com/product-category/room-kits/
  
 IK Multimedia ARC System 2
 http://www.ikmultimedia.com/products/arc/
  
*Package Two:*
 Neumann KH120 monitor speakers
 http://www.sweetwater.com/store/detail/KH120
  
 Neumann KH805 subwoofer
 http://www.sweetwater.com/store/detail/KH805
  
 Whichever acoustic room kit of your choice here
 http://www.sweetwater.com/c673--Complete_Room_Systems
  
 Acourate room/speaker correction
 http://www.audiovero.de/en/acourate.php
  
 (There are hardware room/speaker correction products too, such as the KRK Ergo and JBL MSC1, but hardware alternatives tend to be more limited in their capabilities--at least they have been so far.)
  
 These two example packages can give you an idea of what it takes to attain a great sounding speaker system. There are other alternatives too, but the main essentials will always be:
  
 1) Quality speakers that are adequately full-ranged (I think down to 40Hz is acceptable, but it's better to get it down to 30~35Hz, and if you can afford to go down to 20Hz, even better). Also, it's important to avoid speakers that have problematic bass port resonances, since that's nearly impossible to correct. Not all ported speakers have this problem. The Opals are ported but have no such problem.
  
 2) Room/speaker correction. This is IMO even more important than acoustic treatment because some situations prevent you from being able to use acoustic treatment. Also, this is probably the number one important element, since even subpar speaker systems can be vastly improved to decent quality.
  
 3) Acoustic treatments. If your living space allows them, they will help you attain the best sound possible, and they will make the room/speaker correction system work less hard and thus decrease the likelihood of distortions caused by drastic corrections. The better your room can sound without correction, the better starting point you have when you do room measurements.
  
 As for myself, my system is this:
 Klein+Hummel O 300Ds (I lucked out and got mine on ebay for only $3,000 back in 2007, when most people in the States didn't know about Klein+Hummel. They retailed at the time for close to $7,000 for a pair. Currently, the successor to the O 300 series is the Neumann KH310. It's the updated model after Neumann bought Klein+Hummel. And some of you might know that Neumann is owned by Sennheiser.)
 http://www.soundonsound.com/sos/oct04/articles/kh300d.htm
  
 Neumann KH805
 http://www.sweetwater.com/store/detail/KH805
  
 IK Multimedia ARC System 2
 http://www.ikmultimedia.com/products/arc/
  
 DIY acoustic treatment (I don't remember exactly how much I spent on just the acoustic treatments, since the total budget included the construction of the studio. Back then commercial acoustic treatment was a lot more expensive, so I went DIY. Today, I think commercial options are cheap enough for most people.)
  
 BTW, if you want to see the entire construction process as well as the design schematics of the studio (my previous studio), here are a bunch of photos and design diagrams (keep in mind that was for a full-blown professional music production studio, and you do not need to go to such lengths for just a home sound system): 
 http://www.soundonsound.com/sos/oct13/articles/neuman-kh310a.htm
  
 So total amount spent (if I'm guesstimating how much the acoustic treatment portion cost and not counting the construction) is probably around $6,000 only. In the grand scheme of things when you look at the ridiculous amount of money some audiophile guys spent on their system (while completely ignorant of acoustics and room correction, thus their speaker systems are totally skewed anyway), $6,000 isn't much at all. And as I showed above, it's possible to attain the same level of sound quality with as little as $4,000 or less. Some of you have spent more than that on headphones and amps already.


----------



## Lunatique

music alchemist said:


> Also, I'm still curious about my question relating to whether Harman-equalized headphones sound like your speaker system in terms of frequency response.


 
 Sorry, I keep getting sidetracked by other questions. 
  
 In my experience, the Harman Target Response Curve (or Tyll's modified version of it) closely approximates what a good pair of full-range speakers sound like in an ideal acoustic space. I use it as the standard to EQ my headphones. When I listen to my EQed headphones, the frequency response is very similar to my speaker system, but of course, it's an approximation that cannot match the sense of dimensionality and visceral impact, and it's always going to sound a bit... as if the headphones are trying too hard to live up to expectations they can't quite meet. Sort of like they are straining while pretending to be a full-range speaker system. That's my impression. 
  
 If any of you live near me, I welcome you to come to my studio and listen to my system. I'm sure it would be fun, sort of like what Tyll did when he invited those people to his place and tested a bunch of headphones and then posted the videos.


----------



## Music Alchemist

lunatique said:


> How do you listen to your speakers though? What kind of listening space? What kind of speakers? At what distance from the listening position? Is the space acoustically treated to adequate level for critical monitoring? Is there room/speaker correction applied? All of these factors can make or break your speaker listening experience. If your setup is not optimal then it's understandable that it won't sound as good as your headphones. But a speaker system that's of high quality and properly set up, will beat the vast majority of headphones on the market with the exception of maybe the absolute apex of headphones that cost more than some cars.


 
  
 You're missing the point here. I prefer the _presentation_ of headphones. It doesn't matter how good the speaker system and room are.
  


> So total amount spent (if I'm guesstimating how much the acoustic treatment portion cost and not counting the construction) is probably around $6,000 only. In the grand scheme of things when you look at the ridiculous amount of money some audiophile guys spent on their system (while completely ignorant of acoustics and room correction, thus their speaker systems are totally skewed anyway), $6,000 isn't much at all. And as I showed above, it's possible to attain the same level of sound quality with as little as $4,000 or less. Some of you have spent more than that on headphones and amps already.


 
  
 That's relatively affordable. The total value of all the headphone-related gear I've owned so far (if you use the original MSRPs, and bear in mind that I'm sort of at the mid-point of my journey) is nearly $10K...and I spent two to three times as much on my music collection. I may end up coming to you for advice if/when I build a speaker system...but I love headphones so much that I'd rather focus on building the best headphone systems first.
  
 There are more members here than you may realize who think that the best headphones are better overall than the best speakers regardless of price.


----------



## Lunatique

music alchemist said:


> You're missing the point here. I prefer the _presentation_ of headphones. It doesn't matter how good the speaker system and room are.
> 
> 
> That's relatively affordable. The total value of all the headphone-related gear I've owned so far (if you use the original MSRPs, and bear in mind that I'm sort of at the mid-point of my journey) is nearly $10K...and I spent two to three times as much on my music collection. I may end up coming to you for advice if/when I build a speaker system...but I love headphones so much that I'd rather focus on building the best headphone systems first.
> ...


 
 Do you think that part of the reason you prefer the presentation might include psychological too, and not purely aural? By that, I mean the feeling that you are entitled to your privacy and don't want to announce to those within earshot what you're listening to, and feeling that wonderful immersion of being lost in your own private little world. I know that I feel it when I use headphones--it helps me disconnect from the rest of the world and enjoy my own little universe. 
  
 I'll be glad to help you or anyone with any questions you might have. I'm sure you can tell by now that I enjoy helping others and passing on knowledge (I'm actually a teacher in my "day job" right now.) 
  
 I'm sure there are members who prefer headphones to speakers just like you, and I have encountered them in my years here at head-fi. However, what percentage of them have actually heard a professional mastering grade sound system in an excellent acoustic space before? I wonder how many of them would change their mind if/when they do?
  
 Different speaker systems in different listening environments can differ wildly from each other, including the presentation, since a typical living room consumer speaker system will have different presentation than a nearfield monitoring system in a professional studio. For many people, until they have heard truly excellent speaker systems in a really good acoustic space, they really don't know how amazing speakers can sound. Most people have never even heard full-range speaker systems before in their lives, and have only been exposed to multimedia speakers, bookshelf speakers, mediocre home theater sound systems living rooms, those tall tower speakers that looks pretty with multiple bass drivers but can't even go down past 50Hz, or low-end so-called "professional monitors" that sound anything but, and all without any acoustic treatment and room correction, in rooms with severe asymmetrical layouts in both room shape and furniture placement, as well as have terrible echo problems that smears all the transients and screws up the imaging.
  
 My thinking is that until the person has actually heard a truly excellent speaker system in a good acoustic space, it's hard to say if their preference for headphones is an informed one. This isn't to say there won't be those who have heard amazing speaker systems and still prefer headphones--I'm just wondering what percentage of the people who prefer headphones fall in that category.
  
 When I have online discussions with people about these types of topics, I often think to myself, "I wish that person can hear what I'm hearing right now with my speaker system--it'll likely change that person's entire outlook about audio." That's why I say I welcome anyone who lives nearby to come and visit me and hear it for themselves.


----------



## castleofargh

there is a feeling of pinpoint accuracy on headphones, the imaging isn't realistic but it feels super precise with some headphones at least. I do not think that trumps the thrilling experience of good speakers in a treated room, but it's a very different experience and I can understand that some people prefer it. just like I know a few guys who prefer the sound of IEMs over even some of the best fullsize headphones. I could never agree, but again it can be a different experience.
  
  
 and yeah I enjoy animes more on headphones somehow ^_^(maybe some subconscious shame to be heard looking at animes at almost 40... but I doubt it as I have no self esteem
	

	
	
		
		

		
			





). but I prefer most movies on speakers.


----------



## watchnerd

To me, they're both tools.  And I can't get a good handle on a mix I'm working on without both.
  
 If I mixed entirely with headphones, I'd miss a lot of soundstaging information, 'power band' info in the upper bass / lower midrange, and mid- to macro-dynamics.
  
 If I mixed entirely with speakers, I'd miss the ability to take room acoustics out of the equation and listen for micro-dynamics.
  
 Reproduced music is an artifice...from the moment the sound hits the microphones it has been morphed by transducer colorations and distortions.
  
 The best we can hope for is to make a recording that is evocative, rather than duplicative, of reality.


----------



## Lunatique

castleofargh said:


> there is a feeling of pinpoint accuracy on headphones, the imaging isn't realistic but it feels super precise with some headphones at least. I do not think that trumps the thrilling experience of good speakers in a treated room, but it's a very different experience and I can understand that some people prefer it. just like I know a few guys who prefer the sound of IEMs over even some of the best fullsize headphones. I could never agree, but again it can be a different experience.
> 
> 
> and yeah I enjoy animes more on headphones somehow ^_^(maybe some subconscious shame to be heard looking at animes at almost 40... but I doubt it as I have no self esteem
> ...


 
 Funny you mention pinpoint accuracy. After my most recent rearrangement of placement of my speaker system (based on advice from Neumann's product manager), I started to get fooled often, thinking my 5.1 surround system is on instead of my 2.1 system. The imaging is so precise that I often think the sound is coming from the surround system's center channel or a surround channel. It is by far the most precise imaging I've ever heard.
  
 Your comment about anime made me laugh. I just turned 43, so I can relate. For me, it's K-Pop. My wife likes to tease me about my guilty pleasure. She'd walk into my studio and say, "Watching Korean chicks shaking their asses again?" or "Listening to those Korean butt-wagging girls again?" 
	

	
	
		
		

		
		
	


	




 She's not jealous or anything, since she actually looks like a K-Pop idol herself (she used to be a model and actress in China), and still looks better than most of the K-Pop idol girls half her age (I've posted my photography of her here before in a photography thread). She just likes to watch me get indignant and defend my guilty pleasure with complaints like, "But they're not all just butt-shaking chicks! Some are actually really good singers and songrwiters! Look at IU! She's uber talented and don't do the overly sexualized thing!" 
	

	
	
		
		

		
		
	


	




 
  


watchnerd said:


> To me, they're both tools.  And I can't get a good handle on a mix I'm working on without both.
> 
> If I mixed entirely with headphones, I'd miss a lot of soundstaging information, 'power band' info in the upper bass / lower midrange, and mid- to macro-dynamics.
> 
> ...


 
 Absolutely. I always check my mix/master on both studio monitors and headphones, just to cover my ass and make sure the track sounds good from both perspectives. 
  
 I think how realistic/natural and how artificial but evocative to make the mix/master depends on the style and the aesthetic sensibility of the artist and the engineers working on the track. Some really go out of their way to produce the most natural and realistic sound possible, and some don't even pretend to want that and use all the tricks possible to simply create something that's very compelling to listen to. When it comes to production, I have no stance on this--as long as it sounds great, who cares?


----------



## gregorio

> You have to compensate for human hearing.


 
  
 I'll admit to being a little confused by some of the posts in this thread. For example, the quote above. Do you mean compensate for an individual's hearing or for human hearing in general? This question might sound like semantics but it really isn't, it makes a substantial difference!
  
 If we are talking about an individual's hearing, *maybe* there is some benefit to compensating for a particular deficiency that an individual might be experiencing, say a loss of sensitivity of one or both ears in a particular frequency band, due to some illness or hearing damage.
  
 If on the other hand we are talking about compensating for human hearing in general, say compensating along the lines of the Fletcher-Munson curves, then this is something we/you should definitely NOT be doing! Sure, if you are listening ONLY to individual sine waves or sweeps, then by all means compensate for loudness contours if you wish but, if you are listening to commercial music, film or other content then you are listening to a mix created for human beings and by human beings (who obviously have human hearing). In other words, compensation for the Fletcher-Munson contours has already been built-in to the audio mixes to which you are listening. Applying EQ to compensate for a loudness contour is, in effect, compensating for the second time and is obviously going to take you a long way away from neutral. The only caveat I would make to this statement is if one listens to music at a very low level, in which case boosting the bass a little is not such a bad idea.
  
 I would also add that most commercial recording studios are not designed with a flat/neutral frequency response, most apply a "house curve". House curves vary from studio to studio but they usually include some amount (up to about 6dB) of bass boost. This is to compensate for the fact that most consumer music systems usually have a bass boost. The exception to this is dubbing theatres (where theatrical films are mixed), which are usually quite flat, with the exception of slight bass and treble roll-offs (the x-curve). However, I mention dubbing theatres only as a point of interest, it does not affect/concern consumers.
  
 G


----------



## b0ssMax

Very very interesting. I will try this out once i figure out how to do this via mac.

Thanks Lunatique for putting this together.


----------



## Solrighal

I use Sennheiser HD 650 as my favourite headphone. Do yourselves a favour and install the Sonarworks Reference 3 plug-in. It just works.


----------



## Music Alchemist

gregorio said:


> I'll admit to being a little confused by some of the posts in this thread. For example, the quote above. Do you mean compensate for an individual's hearing or for human hearing in general? This question might sound like semantics but it really isn't, it makes a substantial difference!
> 
> If we are talking about an individual's hearing, *maybe* there is some benefit to compensating for a particular deficiency that an individual might be experiencing, say a loss of sensitivity of one or both ears in a particular frequency band, due to some illness or hearing damage.
> 
> ...


 
  
 I'm surprised you missed my point entirely when I explained it so clearly.
  
 All I meant was that the raw measurements of a headphone (whether it's neutral or otherwise) are not going to look like a straight line. It's going to follow a curve. For example, here are measurements of the STAX SR-207 with the diffuse-field and Harman curves superimposed.
  

  
 You can see many other raw headphone measurements here:
  
 http://www.innerfidelity.com/headphone-data-sheet-downloads
  


solrighal said:


> I use Sennheiser HD 650 as my favourite headphone. Do yourselves a favour and install the Sonarworks Reference 3 plug-in. It just works.


 
  
 +1!
  
 Bear in mind, however, that they use a proprietary compensation curve, so it won't sound the same as Harman and diffuse-field.
  
 Here are my quick instructions for installing the plugin in foobar2000:
  


music alchemist said:


> Download and install the free trial of Sonarworks Reference 3 Headphone. (Or purchase it.)
> Download and install this VST adapter in foobar2000.
> 
> Go to Components, VST plugins and add the Sonarworks plugin.
> ...


----------



## gregorio

music alchemist said:


> I'm surprised you missed my point entirely when I explained it so clearly.


 
  
 1. I was responding to the thread in general rather than specifically your post, which I too thought I made clear.
  
 2. You appear to now be talking about having to compensate for the frequency response of the headphones, rather than having to "compensate for human hearing".
  
 G


----------



## Music Alchemist

gregorio said:


> You appear to now be talking about having to compensate for the frequency response of the headphones, rather than having to "compensate for human hearing".


 
  
 Please read my post and the preceding conversation again, then. I thought someone was implying that a headphone's measurements would look like a flat line on a graph when he was talking about flat...but by flat, he just meant neutral. I was contrasting electronics (which can measure as a literal flat line on a graph when they are neutral) with headphones (which never measure as a literal flat line on a graph). I also mentioned that if you manage to make a headphone follow a compensation curve perfectly, then the compensated (not raw) measurements will look like a flat line.


----------



## gregorio

music alchemist said:


> I also mentioned that if you manage to make a headphone follow a compensation curve perfectly, then the compensated (not raw) measurements will look like a flat line.


 
  
 Exactly, so you are talking about compensating for headphones' frequency response, not compensating for human hearing. What am I missing/misunderstanding?
  
 G


----------



## Music Alchemist

gregorio said:


> Exactly, so you are talking about compensating for headphones' frequency response, not compensating for human hearing. What am I missing/misunderstanding?


 
  
 Wow. It's like you don't even read my posts.
  
 Like I said in my last post: "I was contrasting electronics (which can measure as a literal flat line on a graph when they are neutral) with headphones (which never measure as a literal flat line on a graph)." The raw measurements of headphones do not measure as a literal flat line (like electronics do) because you have to compensate for human hearing. That is what I meant. Compensated measurements have nothing to do with that, since they are just comparing how well something follows a compensation curve.


----------



## gregorio

music alchemist said:


> The raw measurements of headphones do not measure as a literal flat line (like electronics do) because you have to compensate for human hearing. That is what I meant.


 
  
 I think I get what you meant: You are talking about compensating for measurements taken from inside a plastic/gel head, rather than compensating for human hearing?
  
 G


----------



## Music Alchemist

gregorio said:


> I think I get what you meant: You are talking about compensating for measurements taken from inside a plastic/gel head, rather than compensating for human hearing?


 
  
 No.
  
 Electronics like amps and DACs measure as a flat line when they have a neutral frequency response.
  
 Transducers like headphones do not measure as a flat line when they have a neutral frequency response, and also they never measure as a flat line either way.
  
 If the raw measurements of headphones did ever measure as a flat line, they would not sound neutral to us; in fact, they would sound awful.
  
 On another (UNRELATED!) note, if you do not understand the difference between raw and compensated measurements, here is an example.
  
 First we have the actual raw headphone measurements represented by the blue and green lines. The dotted line on the graph is the Golden Ears compensation curve superimposed on it. Actually, I may be mistaken. It says the measured frequency response is "smoothed 1/3 oct" so these may not even be the raw measurements. But it's just an example to illustrate my point.
  

  
 Now we have the compensated measurements below. It starts with the compensation curve (in this case the proprietary Golden Ears curve) as the neutral reference (thus shown as a flat line) and then shows how the headphone measurements align with it. If the headphone measurements (whether before or after EQ) followed that particular curve perfectly, then the headphone measurements would look like a flat line on the graph—but only for the compensated measurements, never the raw ones.
  

  
 I suspect that you are confusing various meanings of the word compensated without getting the context each time.


----------



## gregorio

music alchemist said:


> Transducers like headphones do not measure as a flat line when they have a neutral frequency response ...


 
  
 Mics and speakers are transducers like headphones and they do measure flat (in an anechoic chamber) when they have a neutral frequency response.
  
 G


----------



## Music Alchemist

gregorio said:


> Mics and speakers are transducers like headphones and they do measure flat (in an anechoic chamber) when they have a neutral frequency response.


 
  
 Missing the context again. -_- Forget I used the word transducers. Just pay attention to what I am saying about headphones.


----------



## briskly

music alchemist said:


> gregorio said:
> 
> 
> > I think I get what you meant: You are talking about compensating for measurements taken from inside a plastic/gel head, rather than compensating for human hearing?
> ...



These types of measurements include the frequency response of the ear canal and the diffraction of the outer ear. If you had noticed from my previous post, there was a measurement of a headphone's response from inside the headphone paired with the subjectively determined response of the same headphone.

The sound input before entering the ear should still be flat.


----------



## RRod

briskly said:


> These types of measurements include the frequency response of the ear canal and the diffraction of the outer ear. If you had noticed from my previous post, there was a measurement of a headphone's response from inside the headphone paired with the subjectively determined response of the same headphone.
> 
> The sound input before entering the ear should still be flat.


 
  
 For a headphone not necessarily, in the sense of them being like speakers at 90 degrees. If you want a signature like speakers of smaller azimuth, then you'd need a non-flat response away from the ear.


----------



## Music Alchemist

briskly said:


> These types of measurements include the frequency response of the ear canal and the diffraction of the outer ear.


 
  
 Yes, when I talk about raw headphone measurements, I am referring to the type taken by InnerFidelity with a dummy head, as that seems to be the industry standard.
  


> If you had noticed from my previous post, there was a measurement of a headphone's response from inside the headphone paired with the subjectively determined response of the same headphone.
> The sound input before entering the ear should still be flat.


 
  
 As for this post and what you have added, the information is not specific enough for me to fully understand.
  
 Elaboration is appreciated, especially pertaining to your last statement.


----------



## castleofargh

Music Al  was only talking about how a measurements without compensation would not help getting a neutral sounding signature.
 but after finding a more or less proper compensation curve (be it diffuse field, the goldenears one, whatever sonawork uses, etc),they are all still based on one model with no guaranty that the random guy will fit inside that model. so the last step of all EQ will still depends on the listener himself.
  
 I can only once again past this vid as it's IMO the clearest thing I've seen (even if a little long) about the problems/limitations that exist in any headphone measurement:

  
 the question for this topic IMO is to find out how close a sine sweep can get us? to me it's a powerful tool to get rid of massive spikes, but not an easy thing to use to get neutral sound as it will tend to lead more toward equal loudness contour than perceive neutral outside world sounds . but again it depends a lot on our own experience of sine sweeps. if one can get used to listening to it on a neutral source, then he can do a nice job at EQing a headphone.


----------



## Music Alchemist

castleofargh said:


> Music Al was only talking about how a measurements without compensation would not help getting a neutral sounding signature.
> but after finding a more or less proper compensation curve (be it diffuse field, the goldenears one, whatever sonawork uses, etc),they are all still based on one model with no guaranty that the random guy will fit inside that model. so the last step of all EQ will still depends on the listener himself.
> 
> I can only once again past this vid as it's IMO the clearest thing I've seen (even if a little long) about the problems/limitations that exist in any headphone measurement:
> ...


 
  
 My main point was simply what (I think) we all agree on: raw un-equalized headphone measurements are not a literal flat line on a graph under any circumstances.
  
 One thing I'm interested in is how to properly equalize a headphone when the frequency response is recessed, because boosting frequencies instead of reducing them can cause serious problems.


----------



## castleofargh

music alchemist said:


> My main point was simply what (I think) we all agree on: raw un-equalized headphone measurements are not a literal flat line on a graph under any circumstances.
> 
> One thing I'm interested in is how to properly equalize a headphone when the frequency response is recessed, because boosting frequencies instead of reducing them can cause serious problems.


 
 if you make yourself a poor guy's measurement rig (not the kind that costs a kidney, a cheap mic some foam and a box), then you can actually test the changes. like for speakers, EQ cannot solve all FR response problems(some parts may just nullify themselves from reflections inside the headphone, or the driver simply doesn't have what it takes to make a loud proper 15khz). with measurements you can see if there is some unnatural ringing or extreme distortions resulting from some EQ changes. as it happens a headphone can improve in those behaviors with a more even FR sometimes. that's another nail on the "EQ ruins sound" coffin.
  
 another thing to consider is of course clipping, and as a consequence, power. if your EQ ends up with +10db variations somewhere to compensate a recessed part, then most likely you will set the EQ gain to -10db to be sure there is no risk of clipping, but it also means the amp will have to be able to output those extra 10db of gain on top of what it was usually doing.
 in that case the matter of power headroom can come up with some amps, as +10db is a little more than 3 times the previous max voltage(gain X3) or 10 times the power. so drastic EQ can have more problems than just the headphone not being happy about it, and there are good reasons to favor headphones that have too much of something, so you just take it down and that's it.


----------



## icebrain1

Thanks, this seems like a pretty epic guide cant wait to give it a try on my HE-400s. I learned a ton about headphones and audio in general in this guide. 

One question, I heard some things that you should avoid boosting frequencys with Eq and you should instead decrease the other frequencys instead to make them more even. 
My question is though, can I just do all my eq Changes, then lower the EQs pregain settings by the highest ammount of DBs I added to a frequency in order prevent any cliiping or artifacting, then increase the volume with my amp. (for exampe say I add 3 db to 300hz and add 8 db to 3khz, could I just lower the total pregain by minus 8db to bring everything back but keep the ratio of my EQ the same). 

Thanks. Sorry if this has already been posted.


----------



## castleofargh

icebrain1 said:


> Thanks, this seems like a pretty epic guide cant wait to give it a try on my HE-400s. I learned a ton about headphones and audio in general in this guide.
> 
> One question, I heard some things that you should avoid boosting frequencys with Eq and you should instead decrease the other frequencys instead to make them more even.
> My question is though, can I just do all my eq Changes, then lower the EQs pregain settings by the highest ammount of DBs I added to a frequency in order prevent any cliiping or artifacting, then increase the volume with my amp. (for exampe say I add 3 db to 300hz and add 8 db to 3khz, could I just lower the total pregain by minus 8db to bring everything back but keep the ratio of my EQ the same).
> ...


 

 that's exactly how you should do it. what matters is that no signal get pushed above 0db, so +8db bass boost and -8db global gain n the EQ, that's perfectly fine as you guessed.


----------



## icebrain1

castleofargh said:


> that's exactly how you should do it. what matters is that no signal get pushed above 0db, so +8db bass boost and -8db global gain n the EQ, that's perfectly fine as you guessed.


 
 Thanks a ton, that makes it way way easier.
  
 Time to One Punch this EQ.


----------



## Joe Bloggs

I'll pitch in with my own tentative comments 

Firstly, this new guide underlines the importance of having visual feedback of the frequency being swept and adjustment of relative amplitudes (in this case via making a log sweep that tapers in amplitude with frequency--at -3dB/oct, I presume). Subsequently there have been arguments about whether a simple -3dB/oct log sweep is adequate, or whether a F-M curve equalized log sweep would be better.

I believe that a curve equalized to follow the perceived loudness ups and downs of the human ear as closely as possible would be ideal. For me however such an EQ curve does not follow the F-M curve, even at the approximate same loudness (say 80dB). The main deviation is in the bass, where the upward EQ taper is less than the F-M curve suggests. The upward taper in the treble is more than suggested by the F-M curve but I suspect this is just my ears growing old 

My EQ video in my latest EQ thread plays a log sine sweep in sync with the frequency indicator of Sinegen
http://www.head-fi.org/t/794467/how-to-equalize-your-headphones-2016-update

So you get an exact indication of the frequency being played down to the single digit Hertz. The sweep itself was also equalized by my proprietary EQ curve which looks like this:


My personal equal loudness curve for testing, in contrast: looks like this (only being "accurate" up to 12kHz, i.e. I can simply listen for equal loudness in the sweep from 20Hz to 12kHz, afterwards I have to guess at a tapering volume like Lunatique does with his -3dB log sweep):


I have no theories for why these curves should be applied, other than the latter being the curve that works for me (from years of experience) and the former appearing to be a reasonable compromise between my own curve and the published F-M curves. I haven't had anybody go through with my particular EQ tutorial so I don't know how good my curve is or isn't for the general population...


----------



## Lunatique

joe bloggs said:


> I'll pitch in with my own tentative comments
> 
> 
> 
> ...


 
 It should be clarified that my method does not rely on visual assessment of the amplitude of the test tone being monitored at all. The spectrum analyzer's only purpose in my approach is to see the actual frequency that is being played back, and the visual markings of that frequency's amplitude is completely irrelevant. This is because my method places emphasis on actually listening to the test tone and identifying dips and peaks during the log sweep and during the sequential playback of sinewave test tones. 
  
 I do not think applying the loudness curve to the EQ setting is a good idea, because the loudness curve is variable depending on the listening level, and unless you always listen to your headphones at the exact same volume, such an approach becomes useless. Maybe you can create a set of different EQ curves catering to specific listening volumes, but again, you need to know exactly what volume level is for each EQ curve, and this whole system becomes unnecessarily complex. 
  
 I feel it's much more logical to simply accept the fact that the natural physiological response we have that creates the loudness curve, is inherently part of human auditory system, and leave it at that. So just EQ your headphone according to your favorite optimal listening level (which is most likely similar to the volume level headphone measurements are done with, and that's usually at the level when our hearing is the flattest and the loudness curve is the least skewed). This means if you simply use the headphone measurements data found on authoritative sites like InnerFidelity and use the frequency response measurement graph as a visual guide to assess in conjunction with the log sweep and pre-rendered sequential sinewave test tones, you will achieve really good results. And if you were to listen at a lower level, just accept the fact that the Fletcher-Munson curve is perfectly natural, and leave it at that. When you're listening at lower volumes, you're not really after supreme fidelity, dynamics, visceral impact, etc., anyway, so save those expectations for when you do listen at optimal volume level (roughly 80~85 dB or so, which is when the loudness curve is the flattest for human hearing). And definitely don't listen louder that level because that's when dangers of hearing loss starts to kick in (depending on how loud and length of exposure). 
  
 The reason why I'm advocating this is because It's just much simpler, easier, and gets rid of a lot of unnecessary complications. Also, professional room/speaker correction systems run on the same principle. When you do a room/speaker measurement, the system will detect the volume level of the test tone being played back (as recorded by the microphone), and if the volume is too high or too low, the measurement will not work properly. This is why the measuring system has a visual guide that shows the volume level the microphone is picking up during the playback of the test tone, and you have to adjust the mic level (or the gain level on the measuring software) until the test tone is picked up by the mic at the optimal range.
  
 If you're curious about the process, here's a short video demonstrating how it's done: 

 Here's a longer video showing the same process, in more detail and with more commentary:


----------



## spruce music

I disagree with not using the F_M curve info.  While it does vary some with loudness the variance is small between 50 and 100 db.  Most of the variance is also at very low and very high frequencies. If you ignore it altogether you automatically will be including a  large error. 
  
 Not trying to start a pissing contest.  Just simple disagreement.  If what you do works and improves your enjoyment then no big deal.


----------



## Lunatique

spruce music said:


> I disagree with not using the F_M curve info.  While it does vary some with loudness the variance is small between 50 and 100 db.  Most of the variance is also at very low and very high frequencies. If you ignore it altogether you automatically will be including a  large error.
> 
> Not trying to start a pissing contest.  Just simple disagreement.  If what you do works and improves your enjoyment then no big deal.


 
 But how would you implement the loudness curve for different listening levels? Create a separate EQ curve for every single possible listening levels you might use with your headphone? 
  
 I don't know about other people, but for me, when I want to listen to music for purely enjoyment or critical monitorying, I listen at optimal level, which is also the level that headphone measurements are done at, and the flattest loudness curve. So if I create my custom EQ curve for that purpose, all I need is one EQ curve.
  
 When I'm listening at lower levels, it's pretty much always for purposes other than for purely enjoyment or critical monitoring, such as a podcast, radio talk show, or sleeping aid audio such as binaural beats, delta/beta waves, etc. For those purposes, perfect fidelity and neutral frequency response really isn't nearly as important. As long as it doesn't sound offensively sibilant or muffled, it's fine for utilitarian purposes.


----------



## spruce music

lunatique said:


> But how would you implement the loudness curve for different listening levels? Create a separate EQ curve for every single possible listening levels you might use with your headphone?
> 
> I don't know about other people, but for me, when I want to listen to music for purely enjoyment or critical monitorying, I listen at optimal level, which is also the level that headphone measurements are done at, and the flattest loudness curve. So if I create my custom EQ curve for that purpose, all I need is one EQ curve.
> 
> When I'm listening at lower levels, it's pretty much always for purposes other than for purely enjoyment or critical monitoring, such as a podcast, radio talk show, or sleeping aid audio such as binaural beats, delta/beta waves, etc. For those purposes, perfect fidelity and neutral frequency response really isn't nearly as important. As long as it doesn't sound offensively sibilant or muffled, it's fine for utilitarian purposes.


 

 I would create one based upon either the 70 or 80 phon level.  The 70 phon curve likely makes the most sense.  You rarely listen at much higher or lower average levels when fidelity is important.  So the error in the curves is going to be mostly small.  So just like you do now, you have one curve that gets very close for all your critical listening.  It won't be perfect, but the way you are doing it ignores this aspect which isn't perfect either.


----------



## Lunatique

spruce music said:


> I would create one based upon either the 70 or 80 phon level.  The 70 phon curve likely makes the most sense.  You rarely listen at much higher or lower average levels when fidelity is important.  So the error in the curves is going to be mostly small.  So just like you do now, you have one curve that gets very close for all your critical listening.  It won't be perfect, but the way you are doing it ignores this aspect which isn't perfect either.


 
 It depends on what the listening habit of the person is though. If someone is like me, who prefers leisurely music enjoyment and critical monitoring at the optimal level, and only listens at lower volume the stuff where perfect fidelity isn't all that important (such as talk show programs and podcasts), then it really doesn't matter. 
  
 However, if the person routinely likes to listen to music for enjoyment and critical monitoring at different volume levels, and is willing to take the time to create different compensation EQ curves specifically for each of those different listening volumes, then sure, why not. 
  
 There's no right or wrong in our different listening habits. The key point here is that we both agree that EQing for neutral/accurate is the ideal here. We only differ in whether we want to create custom EQ curves for all different listening levels of just for that one favorite/most often used listening level.


----------



## spruce music

lunatique said:


> It depends on what the listening habit of the person is though. If someone is like me, who prefers leisurely music enjoyment and critical monitoring at the optimal level, and only listens at lower volume the stuff where perfect fidelity isn't all that important (such as talk show programs and podcasts), then it really doesn't matter.
> 
> However, if the person routinely likes to listen to music for enjoyment and critical monitoring at different volume levels, and is willing to take the time to create different compensation EQ curves specifically for each of those different listening volumes, then sure, why not.
> 
> There's no right or wrong in our different listening habits. The key point here is that we both agree that EQing for neutral/accurate is the ideal here. We only differ in whether we want to create custom EQ curves for all different listening levels of just for that one favorite/most often used listening level.


 

 I thought you only wanted one curve.  A person can do as many as they wish.  My point would be that listening at 40 db with a curve derived from an F_M curve for 80 phon would have less error than if you don't do the F_M correction at all.  So the listening at different levels issue while true doesn't wipe out the benefits of taking F-M effects into account vs just ignoring it.


----------



## Lunatique

spruce music said:


> I thought you only wanted one curve.  A person can do as many as they wish.  My point would be that listening at 40 db with a curve derived from an F_M curve for 80 phon would have less error than if you don't do the F_M correction at all.  So the listening at different levels issue while true doesn't wipe out the benefits of taking F-M effects into account vs just ignoring it.


 
 Don't forget, when you EQ for neutrality/accuracy based on headphone measurements (such as the ones available at InnerFidelity) and applying the Harman Target Response Curve, Fletcher-Munson curve is inherently part of the equation. So my approach of using the Harman Target Response Curve as the standard already has the Fletcher-Munson curve included in the correction.
  
 BTW, the whole Fletcher-Munson curve as related to headphone measurements has already been discussed in the comments section of this article on headphone measurements: http://www.innerfidelity.com/content/headphone-measurements-explained-frequency-response-part-one#OI6BQx3oFqMqs682.97


----------



## Joe Bloggs

lunatique said:


> Don't forget, when you EQ for neutrality/accuracy based on headphone measurements (such as the ones available at InnerFidelity) and applying the Harman Target Response Curve, Fletcher-Munson curve is inherently part of the equation. So my approach of using the Harman Target Response Curve as the standard already has the Fletcher-Munson curve included in the correction.
> 
> BTW, the whole Fletcher-Munson curve as related to headphone measurements has already been discussed in the comments section of this article on headphone measurements: http://www.innerfidelity.com/content/headphone-measurements-explained-frequency-response-part-one#OI6BQx3oFqMqs682.97




So your approach is to take inverse curve of headphones measurements at e.g. InnerFidelity as the starting point for your EQ and customize your curve with your log sweep and tone testing only in terms of customizing where the sharp peaks and dips fall and how far they go?

I propose an alternative method, where you listen to your perfect speaker system and generate an equalized sweep such that you will hear all frequencies at the same loudness, when the sweep is played through your speaker system at a particular volume (preferably the volume you usually listen to music at). You would then use the same equalized sweep as reference when equalizing headphones, and you would then be able to equalize headphones without referring to outside measurements.

I have been able to do this, even though I don't have a reference loudspeaker system. My reference curve was created based on F-M curves and honed through years of experience, taking cues from measurements of headphones I have here and there. In the early years, whenever I managed to tweak a pair of phones to sound better than anything I've heard before, I'd feed my tuning results back into my personal curve by running a two-EQ chain where one is the new EQ curve for my headphones and the other is the existing custom equal-loudness curve, to be updated. These days headphones sound about as good as I can ever make them sound through EQ alone (my current frontier is speaker virtualization DSPs, like that TB Isone you're using) and I update the curve mostly to account for my changing (read: aging  ) hearing.

By this method, I can get a rough tuning of a pair of headphones / earphones I'm hearing for the first time, for which no measurements are available, in half an hour. Some listening and tuning with reference music tracks takes care of the rest. An important ability e.g. when evaluating custom IEMs, for which dummy head measurements are largely invalid anyway.

BTW, I do try to test earphones at a fixed volume level. Although it is hard to measure the exact volume being heard, as others have noted the F-M curves don't change much with 10-20dB of volume variance, which I believe I never step out of.

I mentioned seeing the exact tone being played to the single digit Hertz in reference to my EQ video compared to your method of using a spectrogram on an audio sweep, because I saw that the spectrogram you were using does not show a sharp peak at the particular frequency being played. In any case the spectrogram is only sparse notated in e.g. 1000s of Hertz from 1kHz to 10kHz.

You might also want to check out the actual Sinegen program I use, which does satisfy all our tuning requirements (being able to change frequency in real time quickly and easily without "entering numbers", and showing the exact frequency being played in easy to see numbers:
https://www.dropbox.com/s/bp4dwqfv1qtdx38/SineGen.zip?dl=0

The only thing it doesn't satisfy is being able to EQ the output in real time. I fulfil this final requirement using the published EQ video for public consumption, while privately I use Virtual Audio Cable + VSTHost to route the Sinegen output to two equalizer instances, the first being my personal equal loudness curve, the second being the EQ to be made for the earphones I'm listening to.


----------



## spruce music

lunatique said:


> Don't forget, when you EQ for neutrality/accuracy based on headphone measurements (such as the ones available at InnerFidelity) and applying the Harman Target Response Curve, Fletcher-Munson curve is inherently part of the equation. So my approach of using the Harman Target Response Curve as the standard already has the Fletcher-Munson curve included in the correction.
> 
> BTW, the whole Fletcher-Munson curve as related to headphone measurements has already been discussed in the comments section of this article on headphone measurements: http://www.innerfidelity.com/content/headphone-measurements-explained-frequency-response-part-one#OI6BQx3oFqMqs682.97


 

 Then I have misunderstood your method.  I did go back to the first post and re-read it.  Condensed is this your method?
  
 You listen to log sweeps and note areas that aren't smooth as in having higher or lower volume during the sweep.
  
 You then use jumping between sine waves to nail down what EQ will smooth the balance out so in the end it sounds smooth to you during a sweep.
  
 Have I missed where you used the Harman target curve?  Were your log sweep or sine waves pre-EQ'd for the Harman target curve?
  
 I also missed where you used Inner Fidelity or other measures as some sort of a reference for the EQ you are doing.
  
 Please explain how I am mixing this up.


----------



## Lunatique

joe bloggs said:


> So your approach is to take inverse curve of headphones measurements at e.g. InnerFidelity as the starting point for your EQ and customize your curve with your log sweep and tone testing only in terms of customizing where the sharp peaks and dips fall and how far they go?
> 
> I propose an alternative method, where you listen to your perfect speaker system and generate an equalized sweep such that you will hear all frequencies at the same loudness, when the sweep is played through your speaker system at a particular volume (preferably the volume you usually listen to music at). You would then use the same equalized sweep as reference when equalizing headphones, and you would then be able to equalize headphones without referring to outside measurements.
> 
> ...


 
 It's not just the headphone measurements and audible dips and peaks. The Harman Target Response Curve is also part of the EQ assessment. So basically, it's like this:
  
 1) Look at the frequency response measurement graph and note where the dips and peaks are and how far they deviate from the Harman Target Response Curve.
  
 2) Using the notes, EQ the headphone so its frequency response matches the Harman Target Response Curve. 
  
 3) Perform listening assessment tests with test tones and musical material, and further perfect the EQ curve so the log sweep plays back as smoothly as possible, and there are no significant audible amplitude jumps between frequency intervals. 
  
 And because Haman Target Response Curve is already a standard based on ideal "perceived" neutral/accurate response for headphones, it also includes Fletcher-Munson curve as part of the target response. This means there's no other standard needed--all are already contained in the Harman Target Response Curve. This also means there's no need to generate the pre-EQed sweep from my speakers as you suggested. To put into other words, the perfect neutral/accurate response from my speaker system with room/speaker correction applied, is pretty much what the Harman Target Response Curve is. So logically, all you need to worry about it to tweak the headphone from its default frequency response to the Harman Target Response Curve, and then use the test tones to double-check/perfect the EQ curve so you know for sure it sounds as smooth and accurate as possible. 
  
 I did notice the SineGen program from one of you numerous threads, and if I didn't already have a set of pre-rendered sinewave test tones in proper sequential interval steps, I would probably use it. But because a pre-rendered set is universal and can be used with any device and on any platform (as long as it can playback audio files), it is more ideal for people who want to test audio gear on a variety of devices and platforms.


----------



## Lunatique

spruce music said:


> Then I have misunderstood your method.  I did go back to the first post and re-read it.  Condensed is this your method?
> 
> You listen to log sweeps and note areas that aren't smooth as in having higher or lower volume during the sweep.
> 
> ...


 
  
 I guess it's not as obvious as it should be if you re-read it and still missed it. This is the part where I stressed how important the Harman Target Response Curve is in the entire equation:
  


> *Understand headphone measurements and ideal target response before we do corrections:*
> 
> Now that you've listend and noted all the areas of imbalance in your headphone, we'll start correcting those problems.
> 
> ...


 
  
 I guess I'll have to add a sentence spelling it out--that the goal is to match the Harman Target Response Curve.


----------



## castleofargh

lunatique said:


> spruce music said:
> 
> 
> > I thought you only wanted one curve.  A person can do as many as they wish.  My point would be that listening at 40 db with a curve derived from an F_M curve for 80 phon would have less error than if you don't do the F_M correction at all.  So the listening at different levels issue while true doesn't wipe out the benefits of taking F-M effects into account vs just ignoring it.
> ...


 
 I disagree with the bolded part ^_^.  F&M can be used as a tool like the old tuto by Joe Bloggs did. using it as a point of reference to EQ any source to end up sounding about the same way as another source( modulo how good we are at making an equal loudness contour consistently.)
 but in the quest toward neutral sound, which isn't equal loudness contour, but instead an attempt at hearing like we do in real life, the F&M don't even have a say in the matter. the diffuse field compensation used as base for the harman target is the signature that would get into our ears from electrically flat speakers. what we feel like we're hearing does not enter the equation when making such a curve. so no harman did not involve F&M graphs in the preset of the hd800 for the test.
  
   if you listen to your perfectly flat speaker and do an equal loudness EQ of it, you will not end up with the same sound!!! meaning that equal loudness contour just isn't the way to get a flat signature. as we assume that the flat speakers in an average room are the flat signature/preferred signature. from what I understand Spruce is suggesting to create your equal loudness contour of a gear, and then subtract to it the average F&M curve for about the loudness you used to create yours. and that should get you back to the average expected neutral signature, which hopefully isn't too far off of what you yourself find neutral. it's a different approach that could work very fine for people with average ears/hearing. (did I get it right Spruce?)
  
 a third road could be to simply take innerfidelity's graph, and reverse EQ it to get flat on his graph(so diffuse field compensated), or try to replicate harman curve instead from the raw measurements+EQ.  that would have no say at all about our individual hearing, and I wouldn't trust the sub and the trebles measurements because of how inaccurate those things are in practice. but again for the guy with the average hearing, he would end up close to the compensation he wants and if harman is indeed pleasant to most people(my attempts say yes), then going for it should yield cool results most of the time.
  
 if I had a calibrated set of speakers for a given room like you do, I would go for the Joe Bloggs method for sure. and try to get my headphones to sound like my speakers. it's because I don't have any proper reference that I venture into other methods(all of those still ending up with me fine tuning to whatever I like in the end anyway ^_^).


----------



## Joe Bloggs

joe bloggs said:


> So your approach is to take inverse curve of headphones measurements at e.g. InnerFidelity as the starting point for your EQ and customize your curve with your log sweep and tone testing only in terms of customizing where the sharp peaks and dips fall and how far they go?






			
				Lunatique said:
			
		

> It's not just the headphone measurements and audible dips and peaks. The Harman Target Response Curve is also part of the EQ assessment. So basically, it's like this:
> 
> 1) Look at the frequency response measurement graph and note where the dips and peaks are and how far they deviate from the Harman Target Response Curve.
> 
> ...




I think I got your method mostly written down, except the part where you're shooting for the Harman curve rather than flat on the InnerFidelity graph (which was also not electroacoustically flat but compensated with another curve).

So you rely on sites like InnerFidelity to measure your earphones for you, apply EQ to shape the measured response to match a certain curve on the graph (in this case the Harman curve) and finalize the EQ using your tone sweeps.

------------
In this case, what "finalize" consists of is quite arbitrary as all you know is you should smooth out peaks and nulls in the heard frequency response--even though there would be big (albeit gradual) changes in volume built into your perception of the -3dB log sine sweep.

As I noted, you also have no ability to equalize phones for which no measurements are available. I should also note that, even for phones for which there have been InnerFidelity measurements, I have seldom found that the peaks and dips measured by I.F. to actually fall at the same frequency where I hear the peaks and dips. The differences are also not systematic, i.e. I can't apply a universal correction curve to all the phones measured by I.F.
-------------
Don't get me wrong, yours and PiccoloNamek's original EQ tutorial are the most comprehensive guides I've seen to date, outstripping my half-assed efforts. You're the first guy whose writings have drawn me out to discuss the finer points of my EQ methodology. And I'll probably be totally in awe of the sound your speaker system produces. But I believe that, in your position, you can do better equalizing headphones than what you've been doing. Of course it's up to you to decide whether the effort is worth it, but being able to EQ custom IEMs (that never have any measurements) is a big part of the value proposition of these EQing efforts for me. I own three (admittedly off-brand low-end) CIEMs now, and none are what I'd call listenable without EQ. Yet I can EQ them to sound as good as any IEMs I have, better in fact, when the situation calls for isolation.


----------



## spruce music

castleofargh said:


> I disagree with the bolded part ^_^.  F&M can be used as a tool like the old tuto by Joe Bloggs did. using it as a point of reference to EQ any source to end up sounding about the same way as another source( modulo how good we are at making an equal loudness contour consistently.)
> but in the quest toward neutral sound, which isn't equal loudness contour, but instead an attempt at hearing like we do in real life, the F&M don't even have a say in the matter. the diffuse field compensation used as base for the harman target is the signature that would get into our ears from electrically flat speakers. what we feel like we're hearing does not enter the equation when making such a curve. so no harman did not involve F&M graphs in the preset of the hd800 for the test.
> 
> if you listen to your perfectly flat speaker and do an equal loudness EQ of it, you will not end up with the same sound!!! meaning that equal loudness contour just isn't the way to get a flat signature. as we assume that the flat speakers in an average room are the flat signature/preferred signature. from what I understand Spruce is suggesting to create your equal loudness contour of a gear, and then subtract to it the average F&M curve for about the loudness you used to create yours. and that should get you back to the average expected neutral signature, which hopefully isn't too far off of what you yourself find neutral. it's a different approach that could work very fine for people with average ears/hearing. (did I get it right Spruce?)
> ...


 

 Yes, you correctly understood what I was trying to say and my idea of a good approach. 
  
 The big issue with all this is of course some sort of reliable reference which is hard to come by on a DIY yourself approach for your average head-fier.  Someone very experienced or working in the pro field has a leg up on just getting it to "sound right" even without a hard reference.  I don't mean to demean anyone's attempts.  I appreciate the thought and effort put forth here.  My idea of working backwards to flat before we do other things is just to get something close to a reference. Just getting close to begin really cuts down on the work by us non-pro guys.


----------



## Lunatique

I'm still not sure why you guys are focusing on the Fletcher-Munson equal loudness curve so much, and separating it from the Harman Target Response Curve. Let's see if I can sort this confusion out.
  
 1) The Harman Target Response Curve is based on the most neutral/accurate (which has also proven to be the best sounding during the research process) speakers in an acoustically ideal room. It's important to note that whenever any kind of critical measurements are made in these researches and tests, optimal volume level is always used, which means around 80~85 dB, which is also the flattest response according to Fletcher-Munson. This mean, the Harman Target Response Curve is emulating the best sounding speakers in the best sounding room, at the best sounding volume level, where Fletcher-Munson curve has the least effect on the human hearing. On all accounts, it's bringing together all of the best attributes we want.
  
 2) Assuming you are usually listening at optimal volume with your headphones, if you EQ according to the Harman Target Response Curve, you are getting the best sound, and you're also getting the least negative impact of the Fletcher-Munson curve. 
  
 3) The Fletcher-Munson curve is only really a problem when listening at significantly low volumes. This is why some audio gear have the "loudness" button, which is basically an EQ curve that boosts the low and high frequencies to compensate for the Fletcher-Munson curve when listening at low volumes. When listening at normal volumes, you don't want to activate it, because at normal and optimal volumes, you're not being affected by the Fletcher-Munson curve much.
  
 4) Even if at optimal volume level, Fletcher-Munson curve still isn't perfectly flat by machine measurement standard, that is not a concern, because you guys have to remember that the Fletcher-Munson curve is simply part of our physiology, and your entire auditory system has been the Fletcher-Munson curve all your lives. Your brain is completely tuned to it, and it is the defacto standard you have always known. If you try to use a reverse curve to compensate for it when listening at optimal volume level, you're actually introducing anomaly that is not part of your auditory system's standard of accuracy. I guess a somewhat helpful analogy is infrared photography. When you look at infrared photography, you'll see visual information that our visual perception system cannot see normally, but it does not look natural or pleasant--it simply looks alien and strange, since our visual perception is not meant to see frequencies that's beyond what we've always been used to since birth. Our brain cannot decode and integrate the additional information and still see it as being natural and pleasing. 
  
 In other words, listening at optimal volume level of 80~85 dB is already the ideal for our auditory system, and whatever effects of Fletcher-Munson equal loudness curve at that listening level is the flattest possible for our human auditory system, and it's so ingrained in our brain that it is simply the most natural sounding to us. There's really no reason to try to force human hearing to match machine measured flatness, as that would sound unnatural to our auditory system's inherent standard of normalcy. Now, maybe if you are able to hard-wire the compensated curve for Fletcher-Munson to your hearing 24/7 for a long period of time (days, weeks, months), maybe your brain will recalibrate your auditory system and that will become the new normal, and when you deactivate that compensation curve, your original natural hearing will sound unnatural to you, but that's just not realistically possible, since we have no way to hard-wire a compensation curve to our auditory system 24/7 like that--at least not right now. Maybe in the future with technological advancements that will allow us to see, hear, taste, smell things we currently can't. it'll become possible.
  
 BTW, for those of you who worry about not being able to accurately assess what is neutral/accurate without a published measurement graph for your headphone, or don't own a trustworthy full-range speaker system in an acoustically treated room with speaker/room correction applied, one thing you can do for yourself to ensure you have a much better chance, is to educate yourself. A great resource I highly recommend is this book, which also includes an audio CD with examples to train your critical listening skills by professional standards:http://www.amazon.com/Critical-Listening-Skills-Audio-Professionals/dp/1598630237/ref=sr_1_1?ie=UTF8&qid=1455167803&sr=8-1&keywords=critical+listening
  
 I would say that for those of you who have headphones/IEMs that don't have published measurements available, just relying on your hearing while using the test tones will get you close enough that at the very least, you'll get a far better sounding frequency response than if you didn't do any EQing. Even if your perception is off due to inexperience, just being able to flatten the peaks and dips so that a log sweep plays back smoothly and sequential sinewave test tones sound relatively equal in perceived energy, you are already improving the sound of your headphone greatly. The true enemy of neutrality/accuracy isn't the macro broad tilt in frequency ranges (unless it is extremely colored, and if it is, you will hear it)--it's the micro peaks and dips that are really detrimental. Once you have at least fixed those micro dips and peaks, you're already in far better shape than if you didn't. And if you are worried that you're not getting the broad tilt of macro frequency range correct, then perhaps making an effort to build experience in critical listening and auditioning trustworthy audio gear that's professional standard, will help you get the rest of the way there. I personally feel that if you own at least one pair of headphones that's been assessed to be one of the most accurate headphones on the market by an authoritative source (such as Tyll of InnerFidelity), simply using that as your standard for deviation will be very helpful. If you can EQ it to the Harman Target Response Curve, then you're really not going to be that far off from the neutral/accurate standard. Acclaimed headphones like that almost always have published measurement data, so you're not going to be doing it all in the dark. From there on, all your other headphones can be compared to it.
  
 I think the most important takeaway I want people to remember, is that you should at the very least EQ to flatten out those obviously audible dips and peaks, and don't worry too much about the Fletcher-Munson curve or anything else (remember, it's just part of your biology and your brain is totally used to it), because if you at least do the basics, you'll already have significantly improved the sound of your headphone. It's far better to do that than nothing at all and continue to wonder and worry if you're achieving perfection. At the very least, first achieve "greatly improved" and worry about perfection later when you've built up more experience.


----------



## spruce music

lunatique said:


> I think the most important takeaway I want people to remember, is that you should at the very least EQ to flatten out those obviously audible dips and peaks, and don't worry too much about the Fletcher-Munson curve or anything else *(remember, it's just part of your biology and your brain is totally used to it)*, because if you at least do the basics, you'll already have significantly improved the sound of your headphone. It's far better to do that than nothing at all and continue to wonder and worry if you're achieving perfection. At the very least, first achieve "greatly improved" and worry about perfection later when you've built up more experience.


 
 I erased a bigger reply. 
  
 The bolded text shows a basic misunderstanding.  Yes it is part of our biology and it is the reason our brain won't lead us to a flat reference.  It is not some effect our brain is used to and can filter out.  It is the reverse.  Our brain has it built in and we can't turn it off.  This in built filter that can't be turned off is why equal loudness based EQ is not going to be a great response for accuracy.


----------



## Joe Bloggs

> 4) Even if at optimal volume level, Fletcher-Munson curve still isn't perfectly flat by machine measurement standard, that is not a concern, because you guys have to remember that the Fletcher-Munson curve is simply part of our physiology, and your entire auditory system has been the Fletcher-Munson curve all your lives. Your brain is completely tuned to it, and it is the defacto standard you have always known. If you try to use a reverse curve to compensate for it when listening at optimal volume level, you're actually introducing anomaly that is not part of your auditory system's standard of accuracy. I guess a somewhat helpful analogy is infrared photography. When you look at infrared photography, you'll see visual information that our visual perception system cannot see normally, but it does not look natural or pleasant--it simply looks alien and strange, since our visual perception is not meant to see frequencies that's beyond what we've always been used to since birth. Our brain cannot decode and integrate the additional information and still see it as being natural and pleasing.




Look, perhaps you have a preconception of what an F-M curve stands for and why you shouldn't apply one to your -3dB log sweep. So forget the word F-M. Let's call this "your custom thingamburrito curve" and explore why it's a good idea to apply it.

Right now
-You have a loudspeaker system that simply sounds perfect to you and, as you have said on numerous occasions, measures flat from 18Hz to 20kHz. So my suggestion is, why not make use of this by
1. EQing the log sweep you have until all frequencies *actually all sound as loud to you as each other* when the sweep is played at a certain music-listening volume level (the thingamburrito curve)?
Because then, whenever you come across a new headphone or even speaker system for which you do not have measurement data available, all you need to do is
2. Play your log sweep with the thingamburrito curve applied through the new headphones / speaker system
3. Adjust the EQ on your new headphones / speakers until the log sweep of (2) also has all frequencies *actually all sound as loud to you as each other* at the reference music-listening volume level

And the result would be an EQ to replicates the frequency response of your ideal speaker system in your new headphones / speakers!

Of note, *after you have created the compensation EQ in step (3), the thingamburrito curve plays no more role in your system. So there's no "hard-wiring" any curve to your ears when listening to music with your new system afterwards.* No, the thingamburrito curve is only a means to an end, to flatten all frequencies relative to each other in loudness *while doing the EQ testing* so that all you have to do is to concentrate on making all frequencies actually sound as loud as each other on the new system. It's just a means to let you copy the sound signature of your ideal loudspeaker system to any headphones, without guessing "oh this frequency sounds *this* much quieter than that frequency on these headphones now. But it only sounds *that* much quieter on my loudspeaker rig, so I guess I need to bump up this frequency by... how many dBs? "

Instead, what you're doing now is
1. Rely on 3rd party measurements to give you an EQ curve that supposedly produces the Harman curve in response when you listen to these headphones (except the measurements were made on a dummy head, and you're not a dummy head--hence the need for subsequent adjustments)
2. Use your -3dB/oct log sweep to evaluate the resulting EQ--even though you know that this does not produce a constant perceived volume even when played through the perfect headphones (e.g. "it's normal for the volume to taper down after 6kHz"--then why keep making the sweep taper at -3dB/oct?
3. Guess at the corrections that need to be made--all you know is that the curve should be correct in general terms while some sharp peak / dip adjustments need to be made. Except that may not even be true with IEMs or any other earphones where fitting may make a big difference to the actual perceived sound compared to dummy head measurements.

Yes, you're right that even with all the guesswork involved in your method, you should get much better results than no EQ. All I'm saying is that, in your position, with a perfect speaker rig, it's easy for you to do better than what you're doing now for your headphones.

And the thingamburrito curve you produce from your perfect speakers, would be of immense reference value to those of us who has no reference speaker system to compare against.


----------



## spruce music

joe bloggs said:


> Look, perhaps you have a preconception of what an F-M curve stands for and why you shouldn't apply one to your -3dB log sweep. So forget the word F-M. Let's call this "your custom thingamburrito curve" and explore why it's a good idea to apply it.
> 
> Right now
> -You have a loudspeaker system that simply sounds perfect to you and, as you have said on numerous occasions, measures flat from 18Hz to 20kHz. So my suggestion is, why not make use of this by
> ...


 

 Excellent post, and it lets him have a real reference.  His room corrected speakers.


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## Lunatique

joe bloggs said:


> Look, perhaps you have a preconception of what an F-M curve stands for and why you shouldn't apply one to your -3dB log sweep. So forget the word F-M. Let's call this "your custom thingamburrito curve" and explore why it's a good idea to apply it.
> 
> Right now
> -You have a loudspeaker system that simply sounds perfect to you and, as you have said on numerous occasions, measures flat from 18Hz to 20kHz. So my suggestion is, why not make use of this by
> ...


 
 I think maybe there's a fundamental misunderstanding of what the room/speaker correction actually does, and what is actually causing the room modes in the first place. 
  
 You can't simply do a EQed log sweep of my speaker system in my studio space and then use it as a reference, because that EQ curve would only be applicable to those exact problematic room modes. The critical point to understand here, is that my speaker system only measures perfectly flat in an anechoic chamber. As soon as you take it out of that very esoteric environment, it is no longer flat, because of room modes. In other environments, it'll take special room design, acoustic treatment, room/speaker correction to get them to sound neutral/flat again.
  
 Room modes are highly variable from room to room, depending on the size, dimensions, and construction material. There are specific mathematical equations that can calculate the room modes, and the instructions for speaker placement you read a user manuals of high-end professional monitoring systems is based on this math. I'll give you a simple example: Lets say your speakers are placed 1 meter away from the front wall of your listening space (your room). Mathematical calculations will tell you that you'll get a null at 86 Hz. This kind of null is extremely difficult to correct because no amount of boosting the EQ at 86 Hz will reverse the null to flatness. The reason is because the null is caused by sound wave energies reflecting off the front wall of the room and canceling out the same sound wave coming out of the speakers drivers at the 1 meter position. At different distances there will nulls at different frequencies, and this is why it's extremely important to carefully place your speakers at an optimal position, or add a subwoofer to fill in the null (as long as the null is happening below the crossover point of the subwoofer). But the subwoofer itself has to also be placed in an optimal spot so it doesn't have detrimental nulls of its own. (Nulls are far worse than peaks, because nulls can't be boosted while peaks can be flattened.) And all of that is just the front wall. If we factor in reflections from the side walls, the ceiling, the rear walls, the floor, and all the furniture, you can see how it quickly turns into to cluster***** of epic proportions that wreaks havoc on the sound of your beloved sound system. Your perfectly accurate professional monitoring system now ends up sounding totally skewed and completely unreliable. This is what most people's speaker systems sound like, and they don't even realize it. 
  
 Acoustic treatment can help correct those nulls and peaks caused by room modes, but unless you spend a huge amount of money on tons of acoustic treatment (including specifically tuned bass traps to target specific low frequencies) or your studio is designed from the ground up with mathematical equations that naturally avoids room modes (for example, side walls that are slanted and thus don't cause severe reflections), you're still going to end up with enough problematic room modes that drive you crazy. Before I finally broke down and bought the subwoofer, I had to deal with a null at 65 Hz that's plagued me for years. 
  
 So, this is the point where room/speaker correction comes to the rescue. But as you can see, when you use the room/speaker correction product to measure your listening spot (with the measuring mic), the resulting EQ correction is compensating for room mode problems unique to that room and to that exact listening position (move the position even by a few inches and you could end up with completely different room modes!). The "perfectly neutral/accurate" sound is only possible when the room/speaker correction is applied. You turn it off and it's no longer neutral/accurate. 
  
 Logically, it's not possible to just EQ a log sweep for the speaker system and expect to use it as the reference, because:
  
 1) The neutral/accurate sound I'm getting from my speaker system depends on the room/speaker correction software, which applies correction based on the measurement taken by the measurement mic at the listening position. That EQ curve is unique to that listening position only, and nothing else. Every time I move my speakers to another placement or move my desk placement, I would need to do a new set of measurements and end up with a new EQ curve for the new position. So obviously, none of this can be applied to any headphones since the headphones aren't suffering from those specific room modes.
  
 2) If I deactivate the room/speaker correction software (which means the custom EQ curve it uses is bypassed), then what I'm hearing with my speakers is the room mode problems caused by the shape/size/dimension/construction material of my room. It's equally useless to try to create an EQed log sweep in this state, because the EQ curve will only be compensating for the nulls and peaks caused by the acoustics of my room, which obviously doesn't apply to headphones. 
  
 This is why the only way to compare headphones to my speaker system, is by listening critically and assess deviations, and then try to correct them. However, I can tell you guys that when using published measurements and EQing to the Harman Target Response Curve, and then using my ears to do more EQ adjustments with the test tones and musical material, I already get so damn close to my speaker system that when I do A/B comparisons of the EQed headphone with my speaker system (with the room/speaker correction applied), the headphone sounds good enough that even if they don't sound exactly the same, it is close enough in general tonal character that I don't even bother wasting more time and energy trying to make them sound even more similar. Headphones won't ever sound as dimensional and visceral as my speaker system, but in terms of just frequency response, all my EQed headphones sound close enough that I never feel dissatisfied with the results. I would assume that kind of result is going to sound pretty damn good to most people here at head-fi--in fact, to some the results are already beyond what they imagined they could ever achieve with those headphones.
  
 It's funny--all this discussion makes me want to invite all of you to my place and just hang out and do some listening, testing, enjoy some music and good food (and drinks, if you partake). 
	

	
	
		
		

		
		
	


	




 None of you live close to me?


----------



## Joe Bloggs

No, I'm saying, the way to do get an equal loudness log sweep with your speakers, is to have *BOTH* the room correction EQ engaged AND an additional thingamburrito filter tuned in.

You know, like this?



From *right* to left: Sinegen, thingamburrito equal loudness filter*, room correction EQ (or in this case, headphone correction EQ).
*a really old filter setting that's very different from what I use these days; in particular, the bass boost now is much much less.

Surely your system is capable of more than one EQ in a row in the signal chain?

Regardless of our disagreements you're a great asset to our forum and I'd love to hang out at your place. But I'm in Hong Kong :]


----------



## Lunatique

joe bloggs said:


> No, I'm saying, the way to do get an equal loudness log sweep with your speakers, is to have *BOTH* the room correction EQ engaged AND an additional thingamburrito filter tuned in.
> 
> You know, like this?
> 
> ...


 
 OK, I get what you're after now. You're thinking that the speakers with the room/speaker correction EQ curve applied, and then also add a Fletcher-Munson equal loudness curve for a desired volume level (for me it would be 80~85 dB), then simply use that to represent a full-range speaker system that is neutral/flat (because it's been corrected by room/speaker correction software), and also have the Fletcher-Munson curve built in too (And we can do the same with the set of sinewave test tones too.). Then the resulting log sweep will be played back by the speaker system and recorded with an accurate measurement mic, and the recording of the EQed log sweep will not be used to EQ headphones to make them sound just like the speaker with the corrections applied. I'm pretty sure I understand you this time. 
  
 The problem with that approach, is that it really doesn't significantly decrease the amount of time and energy required to get great results, or give you distinctly better results, The headphones themselves are inherently flawed and have individual frequency response problems that include both macro broad frequency range tilts as well as micro dips and peaks. The amount of time you have to spend on correcting those dips, peaks, and tilts is still going to be significant, with lots of listening, assessing, tweaking, comparing, etc. The only time you've saved is the first step of looking at a published measurement and then doing the first initial EQ correction of getting it close to the Harman Target Response Curve. That initial step really doesn't take long, since there's a visual guide to follow. It's the listening/assessing/comparing with your ears that's the most time-consuming. 
  
 For headphones that do have published measurements, I already explained that I feel the Harman Target Response Curve represents how an ideal pair of speakers sounds like in an ideal acoustic space, so I trust that standard and feel fine using it as a good starting point for the first pass of EQ correction. 
  
 For headphones that don't have published measurements, I can see how your suggestion might be useful. But you still have to deal with the time-consuming process of listening, assessing, comparing, and tweaking. 
  
 BTW, I should mention that I don't bother with CIEMs because returning and reselling is a real pain in the @ss. Just dealing with IEMs in general is already frustrating enough. Adding that extra layer of complication is just too much.


----------



## icebrain1

joe bloggs said:


> No, I'm saying, the way to do get an equal loudness log sweep with your speakers, is to have *BOTH* the room correction EQ engaged AND an additional thingamburrito filter tuned in.
> 
> You know, like this?
> 
> ...





Sorry for going realy off topic on a very interesting conversation. 
But Joe Bloggs, do you an alternative to vst host. As I cant get to run on my pc for the life of me. (windows 10). I manged to get savihost (vsthost, but for only 1 vst) running but i don't think it allows me to make priorities of certain vsts and theyll just run in tangen. 


Thanks, as ive been looking for days now but it seems vsthost is the only program that lets you use system wide VST plugins. Any info greatly appreciated. 


Thanks and good listening.


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## Joe Bloggs

Lunatique
You got it mostly right, except the part about mic recording the double-EQed sweep isn't necessary. It just needs to sound right (flat throughout the range) to your ears, and then the sweep with *only the thingamburrito EQ* (no room correction EQ, no playing and recording) can be used directly to tune other headphones.

With my method I clocked in tuning an unknown cheapo set of IEMs, in the middle of a noisy convention, in less than 45 minutes. The resulting tuneup brought the perceived going price of the unknown IEMs from the actual going price of single digit US dollars (*low* single digit US dollars) to around 200USD, craptastic build and all. 

It's also the basis for my Viper4Android tour with an unknown pair of IEMs that's turned quite a few heads back then:
http://www.head-fi.org/t/726569/review-tour-somic-mh412-viper4android-the-put-up-or-shut-up-review-and-tour

And yes, CIEMs have a pretty poor value proposition--especially when you consider that even demo sets for audition, if you're lucky to get your hands on one, may not sound anything like what you finally get. They would have been a complete non-starter for me if I were not able to tune their sound after the fact. As it is, they are valuable earphones for me to have in noisy Hong Kong  Also, the ones I sport right now are simply $60 custom earmolds wrapped around a $6 set of IEMs (the Philips SHE3580 series I brought to fame)  They did require a completely new EQ profile after being put into the custom earmolds.

icebrain1, Live Professor is a robust alternative. But what exactly goes wrong when you try running VSTHost? Have you tried running VSTHostRegClean first? It's easy for a VSTHost installation to bork itself at times. What I do is maintain an "initial" setting that has no plugins loaded, turn OFF the engine, THEN load an actual preset with all my presets, THEN turn on the engine. If the VSTHost installation is irreversibly borked, I would download it from the official site again, unzip it to a new directory, run VSTHostRegClean from there (so registry entries are cleared and point to the new VSTHost), then start the new VSTHost.


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## icebrain1

@Joe Bloggs
Thanks. 
The problem I have with it is that I cant even to get the program to start. The moment I start it seema to work but in about 10s I get a wierd error and it crashes. I tried the regclean, but only after I installed it. Ill give your suggestions a try and let you know how it works out. 

Thanks a ton. 
Good listening.


----------



## BiggerHead

I'm not the most experienced headfier here but I've got a long history on more technical matters than this and I've paid quite a bit of attention to this so I'm going to butt in here and give my take on some of this, similar to another thread about measurements elsewhere yesterday.
  
 If you stand at Niagra falls, and let's just pretend that niagra falls is white noise, lets' imagine that a perfect microphone placed at some spot _not_ inside a head would measure a flat spectrum.  Ok this is a real thing, now you go and stand where the mic was and  you hear it.  *However you hear it, that's how it really should sound.*  *(every time I make this star it will mean give or take the bass boost room correction, see my comment below)  It doesn't sound like a flat spectrum.  *It sounds like the inverse of an equal loudness curve,* which incorporates some product of curves, basically HTRC*: "transfer function" from diffuse field(or free or whatever) to ear canal and BRC: Brain Response Curve (I just made that up).
  
 Now if you have a perfect speaker system (easier than headphones so lets start there),  and play back your Niagra recording it should _not  _sound flat to you because the real Niagra didn't sound flat to you.  So if you do a frequency sweep with your speakers you should _not _eq the result to a flat sounding curve.  With reference speakers of course (give or take room effects) you won't have to eq it at all, but if you eq it, it should* be eq'd so a mic in the listening position would pick up a flat spectrum.  Now if you play back niagra, the mic will pick up a flat spectrum just like there was in front of Niagra.  If you put your head there, you'll have the same input to your head as you had at Niagra, a flat spectrum, and you'll hear the same thing, not flat.  If you played an equal loudness curve, that by definition would sound flat*.  I think maybe this is what Joe Bloggs does, but not what he said to do.  (I'm intentionally oversimplifying diffuse field, free field and directional stuff, fine, but if you would have been facing Niagra, maybe that all works out reasonably ok).
  
 Ok so you put on a set of headphones and you want to hear that same thing.   Tyll measures what sound gets inside his dummy ear.  Someone said he has to correct that for human hearing and someone else objected.  Well he has to correct it for how sound gets in your ear, if he wants  flat on his graph to mean Niagra sounds real.  Of course that's not correcting the headphone response for human hearing for a couple of reasons.  1) It's not measuring "the headphone response".  That's actually impossible.  There's no such thing.  There is only the response produced by the headphones, measured somewhere, in some setting, but anyway, this is far from "the headphone response".  It's measuring "the headphone response" modified already by part of the human hearing system, the ear canal.  2) It's only modified by _part_ of the human hearing system, not the BRC (neural bits).  So it's correcting the headphone response modified by part of the human hearing system... for that part of the human hearing system that already modified the actual measurement.  That really isn't "correcting the headphone response for human hearing".  This really is an important point I think**. 
  
  
 So why not just measure the headphone response with a mic outside the ear?  Because that won't/shouldn't be the same as a diffuse field niagra recording either and while it doesn't depend as strongly on ear shapes, the relationship between that and the sound at your eardrum probably does depend significantly on headphone shapes.  Now you'd need an HTRF for every headphone. This is a problem speakers didn't have and is the reason speakers can be measured outside of the dummy head.  I think this is an important detail often missed.
  
 So.. he measures inside the ear and wants to have a flat input to sound to the mic like the HTRC.  If that happens and he subtracts (divides... but subtract if we're working in db ie log scale) the HTRC he should see a straight line.  Of course that doesn't happen because headphones aren't perfect.  He still subtracts though and we see how close it gets.
  
 So back to eq'ing...  I certainly also didn't understand (and still don't) in the original post how the HTRC was meant to be used, step by step, in the process, but I just can't see how it could be if you're using hearing as the judge.  The HTRC is the sound in your inner ear from a flat spectrum at Niagra*.  It's not what you hear.  What you hear at Niagra is the inverse equal loudness curve, probably because of nerve and brain issues beyond that.  So how can you use a human hearing based eq system and also use HTRC as any useful reference?  This doesn't make sense to me.  What would make sense to me is an equal loudness curve played on speakers should sound equally loud in all frequencies (that's what it means).  And that same curve played in headphones should sound the same, ie equally loud in all frequencies, in headphones.  We should NOT be correcting measured heaphone spectra with equal loudness curves, that's a different issue and common false claim, but we aren't talking about measurements here.  So this actually seems very simple to me.  Just play equal loudness curves and make them sound like equal loudness regardless of what equipment they are played on.  After all, speakers should sound the same as headphones right?  Which equal loudness curve should you play?  That's the hard part.  It should be the one corresponding to the volume you are hearing when you do the eq.  Your head will modify appropriately for other volumes.  So you need a way to sort that out. 
  
  
 * About that room bass.  I think it's neat that the industry has finally realized that people listen to music in their living room and the brain expects that, but the brain hears many directional cues of where the walls are with speakers.  With headphones that doesn't happen and I'm somewhere else anyway.  I don't want it then, and I think it's wrong and is just justification to play into people's apparent enjoyment of bass (which is fine, in the recording).
  
 ** I hope/think Tyll is still using his own measurements with his own dummy for his own version of an HTRC.  This means that he's correcting the mic in his dummy head with the HTRC for his dummy head.  There actually shouldn't be much caveat left about head to head variation.  He makes it sound right on his dummy head.  Of course it will sound different in your ear canal, but everything sounds different in _your _ear canal.  It's _supposed to.  _Niagra does too. There's no problem here unless the resonances set up between your ear canal and a particular headphone create a strong effect that doesn't exist normally without headphones _and also_ that doesn't exist in the other head.


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## BiggerHead

.. and white noise is probably a bad example because it brings in other technical points but please ignore that and take it for how it's meant in the context of the argument... a real source that when measured by the mentioned mic, produces tones at all frequencies (one at a time if you like), in equal volume.


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## Joe Bloggs

biggerhead said:


> There's no problem here unless the resonances set up between your ear canal and a particular headphone create a strong effect that doesn't exist normally without headphones _and also_ that doesn't exist in the other head.




The rest of your post went a bit over my head :blink: , but this last part, I think it does happen a lot of the time.


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## spruce music

I think the short version of the Niagra post is this.
  
 If you EQ your speakers for a log sweep equal loudness result using your ears, take that EQ and modify it until you get log sweep equal loudness in your phones. You will now have an EQ that makes your phones have the same response as your speakers. 
  
 Is that the essence BiggerHead?
  
 If you don't like your speakers or don't use speakers then you have a bootstrap problem.


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## AudioBear

This has been a very interesting discussion in which I have now become totally confused. Too many good ideas and alternative approaches. 
  
 Why wouldn't it work to simply put on your phones and use individual tones and log sweeps to create an equalization curve that to us (subjectively) sounds flat?  This eliminates Niagra Falls, speakers, rooms, transfer factors, diffuse fields, Ty's measurements and everything else.  It combines all of those into a curve that our brain interprets as flat.  Lunatique has already provided us the files to download. 
  
 EDIT:  what I guess I am asking is whether the idea is to create physically flat response curves or perceptually flat response curves.   When I read other threads I chuckle at how many in this hobby are going the opposite direction by looking for headphones and IEMs that are fun, warm, analytical, bright, etc but never for those that are accurate.


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## spruce music

audiobear said:


> This has been a very interesting discussion in which I have now become totally confused. Too many good ideas and alternative approaches.
> 
> Why wouldn't it work to simply put on your phones and use individual tones and log sweeps to create an equalization curve that to us (subjectively) sounds flat?  This eliminates Niagra Falls, speakers, rooms, transfer factors, diffuse fields, Ty's measurements and everything else.  It combines all of those into a curve that our brain interprets as flat.  Lunatique has already provided us the files to download.
> 
> EDIT:  what I guess I am asking is whether the idea is to create physically flat response curves or perceptually flat response curves.   When I read other threads I chuckle at how many in this hobby are going the opposite direction by looking for headphones and IEMs that are fun, warm, analytical, bright, etc but never for those that are accurate.


 

 A perceived flat response absolutely guarantees a non-flat measured response.  Which guarantees a lower subjective fidelity playback. 
  
 A measured flat response guarantees a playback with subjective fidelity.  Maybe not subjective preference in sound, but subjective fidelity to the music.


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## AudioBear

spruce music said:


> A perceived flat response absolutely guarantees a non-flat measured response.  Which guarantees a lower subjective fidelity playback.
> 
> A measured flat response guarantees a playback with subjective fidelity.  Maybe not subjective preference in sound, but subjective fidelity to the music.


 

 I get that but I'm not convinced perceptually flat isn't what we want with phones.  I see a fundamental difference between speakers, rooms and correction than with phones and IEMs.  We can measure the sound that is incident on our ears from speakers and correct that to flat.


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## BiggerHead

spruce music said:


> I think the short version of the Niagra post is this.
> 
> If you EQ your speakers for a log sweep equal loudness result using your ears, take that EQ and modify it until you get log sweep equal loudness in your phones. You will now have an EQ that makes your phones have the same response as your speakers.
> 
> ...


 
  
 Yes.  I took aim at a few other misconceptions as I see it, but this is exactly right*.  This is simple.  Of course you can cheat, and just prepare a source sweep made with a pre-programeed equal loudness curve if you don't have the speakers to bootstrap through, and then eq that to flat in your headphones.  But the point is the right thing when using ears is equal loudess, not HTRC.  HTRC is not what a human perceives.  It's what a mic in your ear would perceive.  It's the wrong curve unless you're using measurements.
  
 *Edit: (one important caveat in the technique as you described it ... the correct eq for your headphones now, maybe it was obvious, is the _difference_ eq applied in the second step, not the total eq.  You don't want to leave the equal loudness eq in your already flat response speakers, and you don't want to leave it in the headphones either.  We are just using it as a tool to get a benchmark.)


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## BiggerHead

joe bloggs said:


> The rest of your post went a bit over my head
> 
> 
> 
> ...


 
  
 Maybe, but I think it's a smaller effect than either by itself.  Headphones do setup resonances, and different ears do propogate the sound differently, but Tyll's technique is fine with that.  The only correction it misses is the extext to which the particular ears and particular headphones combine to make the sound say very near the driver, very different in one ear and another ear.  The propagation inward (edit: to the extent it can be thought of just as a propagation inward) is properly accounted for, even considering ear to ear variation.


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## spruce music

audiobear said:


> I get that but I'm not convinced perceptually flat isn't what we want with phones.  I see a fundamental difference between speakers, rooms and correction than with phones and IEMs.  We can measure the sound that is incident on our ears from speakers and correct that to flat.


 

 You may be conflating two different ideas here. 
  
 With speakers truly flat response is perceived as way too bright and thin sounding.  A number or reasons one being close miking of recordings is unnatural.  Another being in large concerts you are far enough away the highs are absorbed by passage thru the air.  So a slight gentle downward tilt typically will sound like a balance more like what we hear in real life.
  
 Same is true in headphones.
  
 The difference in perceptually flattening response with EQ is we aren't talking about that.  We are talking about a roller coaster response varying over 10 db if we do it that way.  Do that and play music and it won't sound perceptually flat in the end.


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## BiggerHead

Audio bear spruce music is dead on right.  aA flat sweep doesn't sound flat to a human and thus shouldn't, period, with any playback device.  You an actually measure speakers and headphones, the same, way.. in the ear, and use the HTRC for both.  Again, though don't confuse measurement and listening based correction.  Listening is not HTRC.  It's equal loudness.


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## BiggerHead

So including my edit in response above I guess you could say I proposed two main techniques.
  
*Bootstrap technique*
  
 version a)
  
 1) Get reference class speakers (in a reference class room).
 2) Eq them so a sweep sounds flat to you.
 3) Record the eq values.
 4) Play this eq on your headphones.
 5) Eq it further to sound flat
 6) Record the HP eq values.
 7) Subtract 3 from 6.  This is your headphone eq. 
 The problem here is this works for graphic eq.  For parametric, you'll have to subtract the eq curves and then try to parameterize that subtracted curve, which is maybe not so easy.  Maybe good software can do this?
  
 so version b)
  
 1) Get reference class speakers.
 2) Prepare tones at many frequencies so that they sound flat to you on the speakers
 3) Play the tones on your headphones.
 4) Eq the headphones so they sound flat.
 5) Caveat, you probably have to take out small spikes before or after with sweeps like in the O.P.
  
  
*Equal loudness curve technique*
  
 1) Play on your headphones a sweep or tones already biased for a generic equal loudness sweep at the right volume.
 2) Eq that to sound flat on your headphones.
 This assumes your personal equal loudness curve is reasonably generic, but it doesn't require reference speakers in a reference room.


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## Joe Bloggs

BiggerHead, version (a) and (b) are superceded by the two-EQ technique:
1. Get reference class speakers in a reference room.
2. EQ them so a sweep sounds flat to you (with a parametric EQ or otherwise). Call this your personal equal-loudness EQ.
3. Play this EQed sweep through your headphones.
4. Adjust a second EQ until it sounds flat (with a paremetric EQ or otherwise). This already is your headphone EQ.

I illustrated this process in this pic:

From *right* to left: tone generator, personal equal-loudness EQ, headphone EQ.

Added bonus, the EQed sweep of (3) can be exported for others to play with. I made my EQ guide with a video that plays an EQed tone sweep of (3) in time with the video footage that shows the exact frequency being played at any instant. All you have to do is to find a video player that supports EQ and adjust the EQ until the sweep sounds flat. Doesn't get much simpler than that.
http://www.head-fi.org/t/794467/how-to-equalize-your-headphones-2016-update

This would be the "Equal loudness curve technique" but with my personally-tuned equal loudness curves (modified in the hopes of working for a larger audience)


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## BiggerHead

Sorry, maybe I misunderstood.  I think you're saying your software allows you to pile eq's on top of each other.  If so, then yes this works.  If you're just saying start with the first eq and modify it some more and that's it, then I would disagree strongly.


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## BiggerHead

deleted.


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## BiggerHead

Replied too hastily, fixed.  Anyway, I think we're agreeing on one thing now, HTRC or whatever it's called only comes into play if microphones are involved.  We might be agreeing on the rest too.


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## BiggerHead

So I'll try again and see if I can get Joe Bloggs method (c) in context



Bootstrap technique

version a)

1) Get reference class speakers (in a reference class room).
2) Eq them so a sweep sounds flat to you.
3) Record the eq values.
4) Play this eq on your headphones.
5) Eq it further to sound flat
6) Record the HP eq values.
7) Subtract 3 from 6. This is your headphone eq. 
The problem here is this works for graphic eq. For parametric, you'll have to subtract the eq curves and then try to parameterize that subtracted curve, which is maybe not so easy. Maybe good software can do this?

so version b)

1) Get reference class speakers.
2) Prepare tones at many frequencies so that they sound flat to you on the speakers
3) Play the tones on your headphones.
4) Eq the headphones so they sound flat.
5) Caveat, you probably have to take out small spikes before or after with sweeps like in the O.P.


Joe bloggs c)
1) Get reference class speakers (in a reference class room).
2) Eq them so a sweep sounds flat to you
4) Play this eq on your headphones.
5) Using a second equalizer, not touching the first one, eq this to be flat on your headphones
6) The second equalizer has your headphone eq in it. Remove the first equalizer and and enjoy your headphones.

(c is correct in my opinion but I'm only 70% certain it's what Joe bloggs meant)

Equal loudness curve technique

1) Play on your headphones a sweep or tones already biased for a generic equal loudness sweep at the right volume.
2) Eq that to sound flat on your headphones.
This assumes your personal equal loudness curve is reasonably generic, but it doesn't require reference speakers in a reference room.


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## Joe Bloggs

Yes, this is what I mean.


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## BiggerHead

Yay! 
	

	
	
		
		

		
			




  
 This really isn't that complicated.  Flat sound is (edit) NOT (darn it) supposed to sound flat.  We make something that is supposed to sound flat using a system that sounds already like it's supposed to, and then we fix the headphones to make that sound flat on them too.


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## castleofargh

c) is pretty much a a) that is functional. I don't see the need for 2 solutions when it's the same thing. unless I misunderstood something, get rid of a).
  
 and I don't really get what you're supposed to do with b). make one tone, compare it to 1khz, change the loudness and record it at that loudness? it looks super complicated for no real reason. seems like forcing the equal loudness contour into the tones instead of and EQ. I just don't see myself ever doing this.


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## BiggerHead

Right, a is maybe useful if you only have a hardware graphic equalizer and no software solution.  b probably isn't very useful since there is a free software solution for c.  c is indeed the best.


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## BiggerHead

So we've settled on this:
  
 JB bootstrap technique:

 1) Get reference class speakers (in a reference class room).
 2) Eq them so a sweep sounds flat to you
 4) Play this eq'd sweep on your headphones.
 5) Using a second equalizer, not touching the first one, eq this to be flat on your headphones
 6) The second equalizer has your headphone eq in it. Remove the first equalizer and and enjoy your headphones.
  
 Specifically Joe Bloggs uses Elictri-Q to create two software equalizers.
 If you only had one equalizer you'd have to find a way to find the difference of two eq settings or possibly to prepare source tones in equal apparent loudness as input to step 5.

 Equal loudness curve technique

 1) Play on your headphones a sweep or tones already biased for a generic equal loudness sweep at the right volume.
 2) Eq that to sound flat on your headphones.
 This assumes your personal equal loudness curve is reasonably generic, but it doesn't require reference speakers in a reference room.
  
 In a practical sense this second technique could be done in JB's way by programming the first eq with the generic equal loudness curve, and then tweaking a second eq as in step 5 of the first method.


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## AudioBear

biggerhead said:


> == Equal loudness curve technique
> 
> 1) Play on your headphones a sweep or tones already biased for a generic equal loudness sweep at the right volume.
> 2) Eq that to sound flat on your headphones.
> ...


 
 I probably didn't say it right but the equal loudness technique is what I was driving at in my prior post.  Step 2 "to sound flat" puts our perceptions in place of an instrumental measurement and I don't see any way around that.  Thanks to all, the discussion has really helped so far.


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## Joe Bloggs

audiobear said:


> biggerhead said:
> 
> 
> > ==
> ...




Our way around this is to pre-EQ the tone sweep with the inverse of our perception biases (e.g. we perceive a 100Hz to be x dB quieter than a 1000Hz tone so we make 100Hz x dB louder in the sweep) so that your perception using the pre-EQed sweep matches instrumental measurements using a flat sweep.


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## davidsh

With SineGen it is possible to readily switch between 2 sine tones with a click. OP should perhaps look into that?
EQ'ing is indeed very complicated and full of trade-offs no matter what method you use, there is really no practically achieveable ideal, though each method will have pros and cons. I do, however, find the JB method as recently discussed quite appealing in general. Pick your poison. 

Even using reference speakers in a treated room can't be considered ideal if you seek fidelity (due to stereo crosstalk and phase issues for example, among other things), and when done fussing over that there's the source material to consider too, which will be sub-par as well, which makes it a moot point doing any better than a reference class speaker setup.
Sometimes I entertain myself by trying to design/think out the perfect fidelity situation when playing back recorded sound, and consider viable approximations. My idea of fun


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## BiggerHead

By the way, there is no step 3.  Step 3 was just wrong.


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## Lunatique

It should be pointed out that it's no trivial matter to "get reference class speakers (in a reference class room)," and unless you are willing to spend the time and money and do a lot learning about acoustics and acoustic treatment and room modes, it's very hard to achieve that goal. It took me many years to finally achieve it, and I've been at it since 2007 or so. In those years I read books on recording studio design/construction and acoustic treatment, asked questions on pro audio forums, corresponded with audio engineers who work for the company that manufactures the speakers, and went through lots of trial and error. 
  
 The biggest obstacle to overcome is the room itself. Very very few people have a readily available room in their home that's suitable for critical listening, or can afford to (or have spouse permission) transform the available space into an ideal listening room. Too often the available listening space is severely asymmetrical in shape, layout, and/or have highly problematic acoustic properties (such as room dimensions that create severe room modes that even acoustic treatments can't do much about). In my first studio, I had to construct the room from the ground up as shown in these photos: 
  
 http://www.ethereality.info/ethereality_website/about_me/images/workspace/cloud_pagoda/cloud_pagoda-design_construction.htm
  
 In my current studio space, I had to take an open living room and build a new wall/door to close it off so I can get my studio space:
  
 http://www.ethereality.info/ethereality_website/about_me/images/workspace/lincoln/finished/lincoln-studio.htm
  
 (The entire left side of my current studio was originally open, with only a couple of support columns.)
  
 And even having done all that, I still had room modes that could not be corrected fully. Acoustic treatment and room/speaker correction can only do so much if you have severe nulls caused by room modes.
  
 The last bit of the missing puzzle was to add a subwoofer and then move the speakers as close to the listening position as possible (but still within the recommended distance), and make sure the speakers are far enough from the front wall so that rear reflection cancellation causing the null happened at a frequency lower than the crossover point for the subwoofer, so the subwoofer can fill in that null. (Meanwhile the subwoofer must be placed at an optimal position so itself isn't creating a null too). Only after having done this last bit, was I finally able to achieve a truly neutral/accurate sound.
  
 So it needs to be said that one shouldn't assume that "get reference class speakers (in a reference class room)" is going to be easy. The only reason you should consider it is if you actually want a killer speaker system and a great sounding room to enjoy listening in, and are willing to spend the time and money achieving that goal. Otherwise, for just headphone EQ'ing, the method I described in my original post is plenty good enough.


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## Lunatique

davidsh said:


> With SineGen it is possible to readily switch between 2 sine tones with a click. OP should perhaps look into that?
> EQ'ing is indeed very complicated and full of trade-offs no matter what method you use, there is really no practically achieveable ideal, though each method will have pros and cons. I do, however, find the JB method as recently discussed quite appealing in general. Pick your poison.
> 
> Even using reference speakers in a treated room can't be considered ideal if you seek fidelity (due to stereo crosstalk and phase issues for example, among other things), and when done fussing over that there's the source material to consider too, which will be sub-par as well, which makes it a moot point doing any better than a reference class speaker setup.
> Sometimes I entertain myself by trying to design/think out the perfect fidelity situation when playing back recorded sound, and consider viable approximations. My idea of fun


 
 I explained in a previous post that pre-rendered sinewave test tones are more universal because they can be used with any device and platform as long as they can playback audio files. SineGen is great but it's not as universal across all platforms and devices. 
  
 IK Multimedia's ARC System 2 corrects time-domain and phase issues too, not just frequency response. Take a look: http://www.ikmultimedia.com/products/arc/


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## davidsh

Can you get the software without measurement mic? 

Presently, I use the free equalizer APO.


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## Lunatique

davidsh said:


> Can you get the software without measurement mic?
> 
> Presently, I use the free equalizer APO.


 
 Without the measurement mic the software is useless, because the software is calibrated for the measuring mic that comes with the ARC System. All other similar products are the same as far as I know--they all come with their own calibrated measuring mics.


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## davidsh

Perhaps should've specified that I have the UMIK-1, which is a fairly cheap calibrated measurement mic with individual calibration file, basically.
My main problem with digital correction is that I don't quite understand the math behind it. Not that I've tried to except conceptually, though. As a student of physics I might get to learn more about that, eventually.


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## BiggerHead

lunatique said:


> It should be pointed out that it's no trivial matter to "get reference class speakers (in a reference class room)," and unless you are willing to spend the time and money and do a lot learning about acoustics and acoustic treatment and room modes, it's very hard to achieve that goal. It took me many years to finally achieve it, and I've been at it since 2007 or so. In those years I read books on recording studio design/construction and acoustic treatment, asked questions on pro audio forums, corresponded with audio engineers who work for the company that manufactures the speakers, and went through lots of trial and error.
> 
> The biggest obstacle to overcome is the room itself. Very very few people have a readily available room in their home that's suitable for critical listening, or can afford to (or have spouse permission) transform the available space into an ideal listening room. Too often the available listening space is severely asymmetrical in shape, layout, and/or have highly problematic acoustic properties (such as room dimensions that create severe room modes that even acoustic treatments can't do much about). In my first studio, I had to construct the room from the ground up as shown in these photos:
> 
> ...


 
  
 Hi Lunatique.
  
 I won't argue with that, but what's your point?  The idea here was to find a way to do eq that anyone can do.  Do you think this will be worse than nothing?  Do you think the second method will get closer?  That's not obvious to me.  In your original post I think you intended something closer to the second method I listed.  That's probably ok too.  I do think it's important to use the equal volume curves though, not the harmon curve which isn't directly related to hearing.


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## Joe Bloggs

davidsh said:


> Perhaps should've specified that I have the UMIK-1, which is a fairly cheap calibrated measurement mic with individual calibration file, basically.
> My main problem with digital correction is that I don't quite understand the math behind it. Not that I've tried to except conceptually, though. As a student of physics I might get to learn more about that, eventually.




Basically the system plays sine sweeps or other measurement-oriented signals that can be mathematically transformed into an impulse response. Said transform is then performed on the waveform measured by the measurement mic to produce the impulse response of the system. A deconvolution impulse for the impulse response is the final result of the system
https://en.wikipedia.org/wiki/Deconvolution

which is from now on applied to all music to be played through the system. The system impulse response is now the deconvolution impulse convolved by the system's own impulse response, which theoretically yields a perfect impulse.

Of course, some impulse response aberrations are well-nigh impossible to correct digitally, and still others may simply be artifacts of the measuring system. Still other aberrations are highly position-dependent; correcting for them 100% at the mic position can result in huge pre-ringing artifacts if your actual ear positions are just centimetres away. The digital correction software should in theory account for these failings and produce a realistic deconvolution impulse that does not attempt a mathematically perfect deconvolution but one that produces the most pleasing listening result.

MY problem with measurement-mic corrections is that, having become so adept at finding and nulling resonance peaks and dips with Sinegen and my two-EQ setup, I find that the peaks and dips as measured by my mic, as best I can position it, does not even align that well with the peaks and dips I hear in the bass in frequency and amplitude. I would find the frequency of the peaks and dips to be ever so slightly off and the wild noise-like frequency plot not to be really reflected in my listening. I suppose it could be because I'm replacing my seat at the computer and my body with a tripod and mic, which affects the acoustics. But then I have no idea what would constitute the ideal replacement "seat+mic", acoustically speaking, let alone the means to build such an object. At any rate, seeing as it didn't even get the amplitude right, I'm highly skeptical that its time-domain echo-cancellation voodoo would work either. So I stick with tuning by ear for my loudspeaker system as well, which has the additional benefit of having a set-up time of practically nil, compared to several minutes of mic positioning, sweeping and calculations.


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## BiggerHead

> Otherwise, for just headphone EQ'ing, the method I described in my original post is plenty good enough.


 
  
 Ok... so that was the point.  But on this I disagree.  I had the second longest post here so you may have missed it, but your method seems (I think nobody can quite tell) to rely on the HTRC but without involving any microphones, and that's just not right.  The only the thing HTRC is a "target response curve" for, is a microphone in an ear canal (probably a fake one at that), not the perception in your brain.


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## Lunatique

Quote:


joe bloggs said:


> Basically the system plays sine sweeps or other measurement-oriented signals that can be mathematically transformed into an impulse response. Said transform is then performed on the waveform measured by the measurement mic to produce the impulse response of the system. A deconvolution impulse for the impulse response is the final result of the system
> https://en.wikipedia.org/wiki/Deconvolution
> 
> which is from now on applied to all music to be played through the system. The system impulse response is now the deconvolution impulse convolved by the system's own impulse response, which theoretically yields a perfect impulse.
> ...


 
 Have you actually used ARC System or Audessey's MultEQ (the technology that is licensed to ARC System)? There are a number of different hardware and software products that do room/speaker correction (such as ones from JBL, KRK, and other companies), and they are not the same in terms of capability and sophistication. There are plenty of reviews of these products on the web that you can read, from both consumers and professionals, and if you read them, you'll see that ARC System 2 (emphasis on version 2, which is vastly superior to version 1) is the easiest to use and gives the best results. There were lots of skeptics out there before they actually experienced ARC System 2, citing the exact same points you did and then some, saying that there's no way ARC System 2 can do what it promises, yet after actually using it, they had to admit they were floored by what it can do and they can't figure out why it works as promised. So until you have actually experienced something for yourself, all the skepticism are merely unanswered questions that could easily be answered if you just gave it a chance. If you are ever in Northern California, come visit me and experience ARC System 2 in person. Or, just spend the $210 (current lowest price) and find out for yourself. You can always sell it if you don't want to keep it. People here buy and sell audio gear constantly anyway.
  


biggerhead said:


> Ok... so that was the point.  But on this I disagree.  I had the second longest post here so you may have missed it, but your method seems (I think nobody can quite tell) to rely on the HTRC but without involving any microphones, and that's just not right.  The only the thing HTRC is a "target response curve" for, is a microphone in an ear canal (probably a fake one at that), not the perception in your brain.


 
  
 You know what's really discouraging about all the time and energy I've spent over the years here at head-fi trying to help people? It's that all this time, I hoped I could help just a small fraction of the huge number of members here, but after all I've done, I've had only a handful of people who contacted me and told me they actually went through the steps I've taught and was stunned by the results. Some said they were skeptical but the result spoke for itself, and they were very happy and grateful to have their headphones transformed, able to sound like something they didn't even think they could afford, or ever achieve by just continuing their perpetual treadmill of buy and sell headphones and amps and hoping they'd finally find that "synergy" that so many misinformed/ignorant members here constantly talk about, as if amps are single fixed-preset EQs to be used to compensate for the shortcomings of specific headphones.
  
 With over 300,000 members here, if I could help just 1% of people to find sonic bliss in their headphones, that would be 3,000 people. But what's very frustrating, is that instead of being able to help as many people as I can, I mostly just get people wanting to debate instead of actually trying what I suggested and then letting the result speak for itself. What's even worse is that the people who do the debating are often not even professional audio engineers, yet they are rejecting the research done by those who have spent many years working on these issues in dedicated labs and studios with multi-million dollar budgets and a team of expert audio professionals who know far, far, far more about these subjects.
  
 People want to debate and reject the Harman Target Response Curve and don't want to even try to use it, and it's extremely frustrating for me to see that. Do you guys even realize how much time, money, expertise, research, experiment, professional equipment, etc. goes into all that work? None of you will ever have that kind of budget, time, equipment, or expertise to equal the research these experts have done--not even all of you put together, yet all you want to do is reject their findings and don't even make an effort to try and incorporate the results of their research into your own experiments. I can't help but see it as a form of misguided hubris and ignorance. 
  
 In my original post, I began by saying how it's been an impossible task to educate and help the community here because so much misinformation is regurgitated repeatedly that the members are drowning in a lot of misconceptions cooked up by people who are not experts on these subjects, yet their limited understanding, lack of experience, professional equipment, research methodology, budget, etc., are somehow never questioned, and their rejection of research done by those far more qualified are never questioned either. It's as if everyone wants to believe they are smarter and more informed than the experts who have spent hundreds of thousands to millions of dollars working with teams of audio engineers for many years on these issues. 
  
 My question to all of you is this: Do you actually want to improve the sound of your headphones and be able to enjoy them much more than you do now, as well as be able to achieve a much higher quality of sound without having to spend far more money on fast diminishing returns? If so, are you willing to actually carry out the steps I laid out in my original post exactly, and then actually hear the result for yourself before you reject the whole thing outright without having even tried it? 
  
 Those of you who are willing to at least try it, I will absolutely f--king guarantee that you will hear significant improvements in your headphones, and for some of you, the result you get will likely even end your headphone journey because you will finally attain the kind of sound quality you have been searching for, without having to continue the treadmill of never-ending buying and selling of headphones and amps to find that elusive "end game" setup. Even if the results are not 100% perfection, you'll at the very least still be far closer to 100% than you ever were without having tried my method. Shouldn't that be a compelling enough reason for you to at least try? It doesn't even cost you any money and will only take you a few hours at most. So instead of engaging in debates that go around in circles, why not just try it and then after you have the results, we can actually talk about the results and go from there? If you don't even try, there are no results to discuss, and all of this debate doesn't actually lead to any real action and results that can be assessed. I can even post the EQ curves I have created for the headphones I do have, so those of you who have the same headphones can use my EQ curves and hear for yourself how much they improve the sound of those headphones.


----------



## AudioBear

Some of us are listening. You can count one who will try what you have laid out. What I haven't gotten my head around is if I understand it all but ic the result is as promised that doesn't matter.


----------



## davidsh

Looking further into EQing and room correction is definitely on my to do list, though mostly focused on speakers for now because that is where I spent most of my serious listening time.
  
 I have tried correcting by ear before and I found it quite hard to get to something that sounded much better than without EQ with headphones. I didn't, however, spent the time and energy that was required to do it properly.
  
 Doing simple measurements and corrections have indeed improved my speakers by a big notch because my room is crappy and untreated. But the sound is very far from perfect, more like 'passable' in my book. Ironing out the worst peaks in the bass really helped tremendously. Doing some EQ in the higher registers also got rit of some of the muffled-ness. I'm glad the speakers only set me ~400$ back as the room seems to be what really messes with the sound as one would expect.


----------



## BiggerHead

Well Lunatique I guess we're all  just not smart enough to understand your volumes of brilliance you have poured out. I don't think more extolling of your efforts and expounding on your frustrations of the failing of us lay people in five thousand words is going to fix that.  Maybe actual simple step by step instructions for that which you say we should follow would.  Also, if a teacher really wants to teach, as you act like you do, he will answer questions and will have patience to review the "students" efforts.  99% of your post is after all "teaching" and not in fact instructing, so I would think you would be happy to do this.  I laid out a very solid model for how to get the same sound in a room, and then in headphones, as what you get in _reality_.  Yes it relies on some idealisms, like that it's even possible to eq your stereo correctly. But from those ideal assumptions it also creates some understanding of what all these curves are really for.  namely:  HTRC is a reference for how a flat spectrum should look when recorded on a mic in an ear canal.  The inverse of the equal volume spectrum is a reference for how flat recording should be percieved in your head.
  
  
 And you won't even answer a simple question:
  
 Why are you using HTRC as a reference for what the brain should hear, when it is designed as a reference for what a microphone should hear when positioned in the ear canal.. ie not the same thing at all? 
  
 I'm not the first person to point this out.  Before anyone understood that your "steps" (which are very hard to find clearly laid out anywhere in your many words) were meant to use HTRC it was pointed out to you that you'd end up with an equal volume eq.  You can't fix an equal volume eq with an HTRC correction.
  
 I think that poster was dead on right.  YOU can use your steps because YOU have a different idea in your head about equal volume after years of experience.  It might work for you but it doesn't work for anyone else.  Never mind that I can't even make sense of the steps. 
  
 Why don't you separate your book on frequency spectra  from a simple step by step saying what it is you think we should do.  Who know, maybe then my questions will be answered, but I doubt it.
  
 I find it surprising (ok maybe not) that you find it surprising that nobody comes running to the master sensei.


----------



## Lunatique

biggerhead said:


> Well Lunatique I guess we're all  just not smart enough to understand your volumes of brilliance you have poured out. I don't think more extolling of your efforts and expounding on your frustrations of the failing of us lay people in five thousand words is going to fix that.  Maybe actual simple step by step instructions for that which you say we should follow would.  Also, if a teacher really wants to teach, as you act like you do, he will answer questions and will have patience to review the "students" efforts.  99% of your post is after all "teaching" and not in fact instructing, so I would think you would be happy to do this.  I laid out a very solid model for how to get the same sound in a room, and then in headphones, as what you get in _reality_.  Yes it relies on some idealisms, like that it's even possible to eq your stereo correctly. But from those ideal assumptions it also creates some understanding of what all these curves are really for.  namely:  HTRC is a reference for how a flat spectrum should look when recorded on a mic in an ear canal.  The inverse of the equal volume spectrum is a reference for how flat recording should be percieved in your head.
> 
> 
> And you won't even answer a simple question:
> ...


 
 Actually, I did lay out the exact step-by-step in the original post. I even bolded each section's focus, and the actual steps starts at *"Here's how to use the tools."* The next section after that, "*Understand headphone measurements and ideal target response before we do corrections," *is a primer on why headphone measurements are important and why the Harman Target Response Curve is the standard to shoot for. The section after that, "*Now, lets start correcting the imbalance,"* also gives step-by-step instruction on exactly what to listen for, the order the corrections should be done in. Then finally, in the section "*Testing with musical material," *I even provided all the musical material that I use to test audio gear with, and describe exactly what problematic spots to listen for, down to the exact minute and second where the snippet of interest starts. So what exactly don't you understand. Did you actually try and do the steps I laid out in detail?
  
 As for the Harman Target Response Curve, I thought I had already explained numerous times in this thread, but no one is listening and continues to ask the question as if I haven't explained yet. I'll explain again now:
  
 Fletcher-Munson is supposed to be part of our auditory system. It's misguided to try to compensate for it in your EQ if you are playing back at optimal volume (80~85 dB), because at that volume level, the equal loudness curve is already at the flattest it can be for our hearing, and it's what our auditory system is used to. If you try to boot the bass and treble to reverse it, you'll end up with way too much bass and treble and it will sound really colored. The only time you should be applying the equal loudness curve is when listening at low volumes, which is exactly why audio gear manufacturers often put a "loudness" button on their receivers or other audio gear--it's for that exact reason. It is only meant to be used to make quiet listening sound fuller by booting the bass and treble. It's as simple as that. Don't over-think and make it more complicated than it has to be.
  
 So if you're listening at optimal volume level, you do not need to even think about the Fletcher-Munson curve. Which means all you need to do now, is to make your headphones sound like great speakers in a great room, which is what HTRC is all about. This is why I say you simply need to follow the HTRC in your EQ (and then double-check by ear with test tones and musical material to make sure everything is smooth and correct). 
  
 Now, if you inherently disagree with any of the above, I suggest you simply just email/telephone Dr. Sean Olive (the audio engineer behind the HTRC) and debate with him, and let him explain to you all the years of research and experiments that he's done to arrive at the conclusion he has. Here are the various links where you can easily contact him:
 http://www.aes.org/aes/seanolive
 https://www.linkedin.com/in/seanolive
 http://seanolive.blogspot.com/
  
 You can also contact Tyll Hertsens and have debates with him about all this and let him convince you, since he's one of the most respected and knowledgeable headphone experts in this community:
 http://www.innerfidelity.com/


----------



## BiggerHead

I don't need to email him at all because I've never seen where he or anyone else claimed that the harmon target response curve is supposed to have anything to do with what you hear.  It is supposed to be
  
 a TARGET
 CURVE
 for the
 RESPONSE
 of
 a  **MICROPHONE** measurement.
  
 not your brain.  The fact that it is so incredibly different from the FM curves just proves how much your brain is doing.  You don't hear an HTRC curve in response to a flat free field input. You hear and inverse equal loudness curve, which is exactly by definition, what you should hear, and nothing else.
  
 The fact that the FM curve at 80db is flatter than 60 isn't very interesting.  Fine, use the 80db curve and EQ at 80db if you want.  If it's nearly flat, no problem.
  
 But using the HTRC is wrong.  Which is kind of fine because while I did see all your stuff about how important it is, I didn't see where or how your steps actually say to use it.  I'll look again, but I couldn't find it the first time through.


----------



## BiggerHead

From your link:
  
 "The plot above shows the frequency response measured at the ear drum (DRP) of a flat in-room speaker response (dashed green line) and the new target response curve developed in the research.  These curves are essentially what should be seen with ideal headphones in the raw (uncompensated) measurements on InnerFidelity headphone datasheets.
 Read more at http://www.innerfidelity.com/content/headphone-target-response-curve-research-update#jjsa59M8JOWJXvO8.99"
  
 Those datasheets are measured with microphones... in the ear drum.
  
 This does NOT say that playing an inverse HTRC will sound flat to your brain.  It should record as flat in your ear drum.  Not the same thing. If you want to refuse to acknowledge or respond to that fact, that's your business I guess.


----------



## BiggerHead

So I'm going to try to give you the benefit of the doubt that this is a communication problem:
  
 "2) Using the notes, EQ the headphone so its frequency response matches the Harman Target Response Curve"
  
 This is the ONLY place in your actual instructions of what to DO (not what to understand) where you mention HTRC.
  
  The only thing I can see now is that you do tell us (not in the original instructions)
  
 "1) Look at the frequency response measurement graph and note where the dips and peaks are and how far they deviate from the Harman Target Response Curve."
  
  
 So is your idea that I should look at say tyll's measurements, and if I see a 10db dip at 4 khz... then I should play a 4khz tone and eq it up by 10db to correct for the MEASURED deviation from HRTC at that frequency?  I'm I supposed to use my ear to raise it 10db?(weird)  If so you don't need notes at all. Just use the measurements which are _already _relative to the HTRC (so you don't actually need to understand anything about HRTC or even use it) and make an eq curve that matches them and take the inverse. done.  
  
 But that's using _measurements.  _ 
  
 If you're supposed to use your brain perception, then HRTC has essentially nothing to do with it.  HRTC is for measurements.


----------



## Joe Bloggs

Lunatique's method is as follows:

1. Find out what the Harman target response curve looks like on the Innerfidelity compensated measurements (I haven't seen Lunatique show this. I have seen miceblue produce such a curve in a chatroom once but damned if I can find it now. It appears one would have to eyeball the compensation curve comparison graph here http://www.innerfidelity.com/content/first-test-estimated-harman-target-response-curve-various-headphones and draw the curve yourself)
2. Make an EQ to remodel the frequency response of the headphones as measured by IF to fit the target curve. (if one were going after I.F.'s own target curve, all you would have to do at this point is to create an upside down plot of the compensated FR graph in a paragraphic EQ, but since Lunatique is after the Harman curve it's rather more complicated)
3. Use the tone sweeps provided to find out any peaks and dips in the frequency response as heard by your own ear and modify the EQ to remove those bumps. It should be noted that because he does not incorporate (personal or generic) equal loudness curves (or perhaps I should say *perceptual target curves*) in his tone sweeps, it is known that these sweeps should not sound like constant loudness throughout the sweep even when played through perfect headphones. Therefore the goal in this step is to remove local peaks and dips in the heard response while preserving the overall slope of the heard response (tapering down in perceived volume towards the high treble, among other things).

He says it's a simple method and wishes people would just get down to trying it. I would note that step 1 and 2 are still an unknown quantity, but can be substituted with simply inverting the I.F. FR curve of your headphones for now if you want "simple".

What gets me is, I have put out an even simpler guide just a few days before his:
http://www.head-fi.org/t/794467/how-to-equalize-your-headphones-2016-update

Conceptually it boils down to just one step:
1. Play my video of the tone sweep and EQ the audio until the sweep sounds constantly loud throughout the range when played through your headphones.

That's because the audio sweep in my video DOES include a (what I hope to be) generic equal loudness curve (or should I say *perceptual target curve*), such that ideal headphones playing the audeio sweep in my video *should* sound constantly loud throughout the range to the, eh, generic listener.

Nevertheless, for this one step I have provided much more detailed step-by-step guidance in my thread, including a walkthrough for how to use the equalizer I'm using...

...and I have received even fewer responses than Lunatique says he's got.

So to his bemoanings I say "cry me a river". Or, "misery loves company"...


----------



## Lunatique

biggerhead said:


> So I'm going to try to give you the benefit of the doubt that this is a communication problem:
> 
> "2) Using the notes, EQ the headphone so its frequency response matches the Harman Target Response Curve"
> 
> ...


 
 What Joe Blogg just posted is pretty much right, so I'm not going to repeat what he wrote.
  
 Keep in mind that what I'm advocating is to use THREE separate elements: measurement data, HTRC, and using test tones/music. So your assertion that it's all only based on measurement is incorrect. The last step of using your hearing and test tones/music is crucial because you cannot be sure the published measurement used a headphone that sounds identical to the one you own, since there could be manufacturing inconsistencies. That's why you need to actually use your ears (which means also your brain) and actually listen and make sure everything sounds smooth and rich, and full, and detailed, without bloat, mud, harshness, dullness, thinness, etc. 
  
 I'm here to help those who are willing to listen and give new experiences and ideas a chance, and then listen to the results themselves. Until you have actually carried out the steps and listened to the final results, you have no real concrete evidence that it won't work and won't sound good--you're just arguing for the sake of arguing, without actually carrying out the experiment to see for yourself. 
  
 You keep focusing on this whole "HTRC has nothing to do with brain perception" thing, and you are wrong. HTRC was developed to create what our brains have perceived to be the best and most satisfying sound, modeled after the best sounding speakers in the best room. It isn't just measurements--it includes psychoacoustic considerations. They actually tested many people and recorded their preferences for what sounds the most satisfying, and the final result is NOT equivalent to microphone measurement of a flat sounding pair of speakers--this is clearly stated in all the articles about HTRC online. I'm not the one who did all the years of research and experiments on all that, but I already gave you the contact information for the person who did--someone who is readily reachable online and has public profiles and contact information. You can choose to avoid contacting him and have him educate you on all the things you don't understand and hold misguided ideas on--that is your choice. I'm sure Dr. Olive would be more than happy to talk to you, since that is his passion and area of expertise. I don't know if you are afraid to be proven wrong or something, but if not, just contact him and learn something. Shoot him an email or just point him to our posts in this thread. It only takes you a few minutes and it'll clear up everything. He'll be able to explain it all far better than I could. You can give all the excuses you want for why you don't need to contact him, but it simply sounds like avoidance. You have a great opportunity here to educate yourself and learn things that could be very eye/ear-opening that will change your understanding of something that you seem passionate about. So why won't you do it?


----------



## Joe Bloggs

What it boils down to is, BiggerHead thought your method is invalid because the Harman curve ought to be applied to mic measurements of frequency response, not to perceptual tone sweeps meant to be listened to. But your method *is* applying the Harman curve to mic measurements. The tone sweeps come in a later step and does not make use of Harman. So there is no disagreement.


----------



## BiggerHead

lunatique said:


> You keep focusing on this whole "HTRC has nothing to do with brain perception" thing, and you are wrong. HTRC was developed to create what our brains have perceived to be the best and most satisfying sound, modeled after the best sounding speakers in the best room. It isn't just measurements--it includes psychoacoustic considerations. They actually tested many people and recorded their preferences for what sounds the most satisfying, and the final result is NOT equivalent to microphone measurement of a flat sounding pair of speakers--thi


 
  
 I realize that the HTRC isn't designed to actually give a traget reponse based on a flat speaker.  I said this with about 20 astereisks in my original post.  It is though the target response, based on whatever reasoning they use, FOR, TO BE COMPARED TO, what the microphone reading for the headphone is.  My most recent simplified argument with flat spectra was to make a point, but anyway, we are talking about relatively flat spectra anyway.    I don't have time now.  I'll go back one more time and read what Joe said (you certainly didn't clarify).   It sound to me like basically this is 
  
 a) start by programming the inverse of a tyll's response curve measured realative to the HTRC.
 b) Fix the remaing sharp peaks.
  
 You certainly did not make any of this clear in the OP.  You keep saying try it.  What I'm saying is I can't try what is clear as mud.
  
 As for the target resxponse not being flat.  I disagree with that anyway.  I don't need a listening room bass boost in my headphones.  I'm not in a listening room when I listen to headphones.   this notion that the headphones should sound liek a good room was an ASUMPTION  of the HTRC, not a conclusion of it.  The conclusions was that people think a good room (not headphones) should have a bass boost.


----------



## BiggerHead

OK, so basically joe boggs confirms my strong attempt at finding a reasonable interpretation of the OP.  I cannot find anywhere where the OP actually says to do step 2 of Joe boggs latest post, or step one of mine.  Maybe lunatique could kindlly point me to a quote of that part.

Anyway, fine. Use an inverse Tyll, HTRC corrected measurement as a starting eq.  Of course that's reasonable, but hardly a simple way for anyone to find their own eq for THEIR particular cans and hardly enlightening.  It's just using one Tyll already found and fixing it a little.

 Mine don't even have measurements of that quality.  And.. as said, I disagree with the bass boost.

I do like Joe boggs guide quite well, but I suppose we could say it comes down to a difference in opinion.  His FM technique takes into account everything the microphone method does minus the "preferences" aspect of things like bass boost which you're welcome to color back in to your personal liking.


----------



## BiggerHead

Reading in bits and pieces today...
  
 I like "perceptual target curve" PTC as JB called it.  Although in practice we can't print PC's to compare to PTC's so it's easier to use the inverse and target a flat PC.
  
*So some definitions:*
*F*: Flat curve (mathematically 0 in additive log math)
  
*PTC*: Perceptual target curve, target perceived response from a flat source (normally used in inverse).
  
*ECTRC*:  Ear canal TRC  target for mic measurement in an ear canal from a flat source. HTRC is an example of one of these.
  
*ATRC*: Air target response curve, target for mic measurement in a listening room from a flat source as needed to make a "desirable" sound field in that room (not a flat sound).  The harmon version (*HATRC*) includes the "bass boost" and other "preferences" for a how room stereo should sound.
  
*HTRF* (did I get those letters right?): Head transfer response function: Response for mic measurement in an ear canal from a flat air spectrum (free field seems to essentially match latest thinking, but never mind that for now).
  
*ELC*: equal loudness curve, air (let's say free field?) spectrum which produces equal volume perception in brain.
  
*BRC*: Brain response curve: perceived response curve to a flat spectrum at the ear canal... transfers an ECTRC to a PTC   (maybe better to call it a BRTF, transfer function, oh well).
  
  
*some relationships then:*
 F =0 so can be ommitted below, but I include it for effect:
  
 PTC=F+ATRC+HTRF+BRC   (note that this separates perceptual preferences in overall sound signature, ATRC, from transfer functions which are non negotiable, the perceptual preferences are even separated from the perceptual transfer function for individual frequencies which is perceptual but more objective).  The HTRC is NOT including BRC; it's including ATRC.)
  
 ECTRC (HTRC being an example) = F+ATRC+HTRF  
     or, for the harmon versions specifically:
 HTRC=F+HATRC+HTRF
           So an HTRF is just an ECTRC with the ATRC set to zero, which is why Tyll just replaced his prior use of HTRF's  with HTRC's, which just added the HATRC.
  
 and we see a
 PTC=ECTRC+BRC
  
 so a "harmon PTC" would be:
 HPTC=HTRC+BRC
  
 More interestingly:
 -ELC (note the negative)=F+HTRF+BRC
           = PTC-ATRC  
           =ECTRC+BRC-ATRC
  
 using the harmon examples:
 -ELC=HTRC+BRC-HATRC
  
 So an inverse ELC is just the HTRC propogated from the ear canal to the brain, minus the subjective room sound preferences!
  
 ..which is what JB's EL method is using, an inverse harmon target response curve, transformed to the percieved curve, without the harmond stuff... or with the preference for a non flat room sound set to zero(flat). 
  
 Note that the ATRC curve is NOT objective even in the sense of defining how volumes of individual tones are perceived by the brain.  That's done by the BRC which is included already in the inverse ELC.  It is a non-fidelity subjective term related not to perception of tone volumes but to subjective preferences for overall sound mix in a room, and is the only difference between the ELC method and HTRC methods other than shifting the measurement point through the BRC. 
  
 And in my opinion these sound preferences if they are truly universal should be in either a) in the recording or b) are something we expect to hear in a room, not in (possibly closed) headphones.. that I think take most of us away from the room we're in.  
  
 So that's all sorted (if I had no typos).. hah.


----------



## Malfunkt

This is a great thread. Thanks Lunatique and everyone contributing.
  
 I've done a small bit of amateur engineering before, and have always been interested in musical composition and production. So I'm somewhat familiar with these techniques.
  
 I tried Joe Bloggs suggestions as well, and using Adobe Audition ran his test pink noise video.
  
 Have to say, this is really difficult, and Lunatique you likely have a more experienced ear for this. With my MDR-7520 I couldn't pick out any obvious peaks, well aside from its large bass boost (which I'm happy with) its a pretty smooth ride all the way up to around 8khz where I detected a spike. I just wasn't really that comfortable making any significant adjustments. Felt like the headphone was pretty damn near perfect. A number of engineers have said the same online, that this headphone is very close in presentation to calibrated room monitors.
  
 Again, I also had difficulty calibrating my LCD2, though casual observations of my own have brought attention that the upper registers of this headphone aren't quite perfect in presentation. 
  
 I'll try your HD650 setting though, as it seems pretty close to my own assessment of where it needs EQ. Of headphones I've owned, the 650 has been closest to rendering binaural recordings of nature as realistic as possible, which for myself seems like a great reference (barring that the original recordings are meticulous) as we are intimately familiar with these ambient soundscapes.
 Sounds like you have a studio reference with a calibrated room and monitors. That right there is enough to give you the ear and experience with music to know where you might need to make adjustments. Harder probably for those that don’t have this.
  
 Although I'm somewhat 'studio minded' I don't bother EQing my headphones often. My LCD2 seem almost perfect in presentation, my ears adjusting to their 'darker', relaxed but still incredibly clear sound.

 One thing I find a bit amusing, that so many people are chasing ‘synergies’ with DACs, amps, and mods when a small EQ adjustment could remedy many headphone issues and a fraction of the price. EQ won’t fix distortion, ringing or other issues, but neither will cables, amp pairings, etc. 
  
 Another claim I see often, is that a member may have heard many pieces of equipment, and this is only relevant to a degree, as you may hear obvious technical capabilities, but overall reference for neutrality won’t necessarily get better by hearing more pieces of gear. Having a reference for how music and sound is reproduced on a calibrated system and room space.

 All this said, one can enjoy their gear, and using their own ears can evolve a taste for what their listening preferences are - even if these aren’t perfectly neutral or balanced - they are subjectively valid. Would just caution users that before spending major cash, may consider experiencing such a reference, as it might be helpful in guiding them to a more audiophile grade level of sound or help them decide that it isn’t that important.


----------



## Lunatique

malfunkt said:


> This is a great thread. Thanks Lunatique and everyone contributing.
> 
> I've done a small bit of amateur engineering before, and have always been interested in musical composition and production. So I'm somewhat familiar with these techniques.
> 
> ...


 
 You might want to also take a look at my LCD-2 EQ curve here to get an idea of how much tweaking was required to get it to sound more neutral/accurate: http://www.head-fi.org/t/551426/my-eq-curves-for-lcd-2-hd650-m50-and-007mk2
  
 You and I totally agree on the silliness of spending so much time and money on different amps, DACs, and headphones, hoping and praying that they just happen to cancel out each other's shortcomings and create "synergy." But I suspect a lot of people really enjoy all that, and to them it's just as much about playing with gear. I'm far more pragmatic and don't have a collector's mentality, and I have other passions in life that are more meaningful to me than buying/trading/selling electronics. For me, It's always been about simply achieving the most neutral/accurate/satisfying sound, and once I attained that, I stopped with the never-ending treadmill of buy, trade, sell, and simply focus on my passion for music. And the only reason I was able to get off that tiresome treadmill was because of EQing.


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## Malfunkt

lunatique said:


> You might want to also take a look at my LCD-2 EQ curve here to get an idea of how much tweaking was required to get it to sound more neutral/accurate: http://www.head-fi.org/t/551426/my-eq-curves-for-lcd-2-hd650-m50-and-007mk2
> 
> You and I totally agree on the silliness of spending so much time and money on different amps, DACs, and headphones, hoping and praying that they just happen to cancel out each other's shortcomings and create "synergy." But I suspect a lot of people really enjoy all that, and to them it's just as much about playing with gear. I'm far more pragmatic and don't have a collector's mentality, and I have other passions in life that are more meaningful to me than buying/trading/selling electronics. For me, It's always been about simply achieving the most neutral/accurate/satisfying sound, and once I attained that, I stopped with the never-ending treadmill of buy, trade, sell, and simply focus on my passion for music. And the only reason I was able to get off that tiresome treadmill was because of EQing.


 
 Thanks Lunatique, I'll have a look.

 I think there are just different paths. If you start a path is music, and more specifically, as a recording or mastering engineer you are going to learn a whole different side of audio. If those on head-fi had a different experience - such as using a mixing board (analog or digital), applied EQ or dynamic compression to audio, or have gone through a bit of the recording process - I think their take on audio would be a lot different.

 For many, playing with their gear, is in a way learning sound engineering (to a degree), but one that is slow, and expensive and not necessarily accurate. You could save yourself years and many dollars just by knowing some solid audio engineering basics. I really feel that some people are 'chasing the dragon' sometimes. Further, manufacturers may be taking advantage of consumer ignorance and exploiting areas of ambiguity.
  
 I think sites like Innerfidelity are moving in the direction towards measurement, but they should not be the lone voice. Would be great to hear from some recording industry heavy weights and I'm not talking the 'artists', but the mastering and recording engineers behind the magic to help set the record straight. Would be great to have a video having one of them explain audio calibration and in turn how it could apply to headphones and what one might look for in a reference.


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## davidsh

I would love to get my hands on some good room correction software/a solution that will allow me to apply the result system wide on windows 10. I got a calibrated measurement mic. Presently, I use Equalizer APO, which also has support for FIR filters.
  
 Ideally, I think what I need is something that does
  
 Measurements -> FIR filter 
  
 the best way, and allows me to import the FIR filter into EQ APO. What are the options? Is this actually what I want?
  
 How much better is FIR compared to IIR? 
  
 Generally, I really need some ressources on how to measure my speakers and what to do with the measurements. Presently just using REW


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## Lunatique

davidsh said:


> I would love to get my hands on some good room correction software/a solution that will allow me to apply the result system wide on windows 10. I got a calibrated measurement mic. Presently, I use Equalizer APO, which also has support for FIR filters.
> 
> Ideally, I think what I need is something that does
> 
> ...


 
 This is probably what you're looking for:http://realtraps.com/art_measuring.htm


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## BiggerHead

I get that there a many details that real engineers in the field are familiar with for making sound come out enjoyable in practice using realistic gear, but it's easy to also rationalize fixes to real world situations or simply habits until reference has no meaning. 
  
 If the microphone on stage doesn't pick up the same bass a person in the audience would, it's obviously not the job of a reference playback system to fix what wasn't recorded.  You might though want your (non-reference) stereo to do that.  Better yet, the recording should probably be modified to fix the problems inherent in the recording system.  Reality is reality though.
  
 We also should not be defining the _meaning_ of a flat recorded spectrum as being an intention to produce a bass boosted free field spectrum.  That's just doesn't make sense, and in this sense I don't think innerfidelity is going the right direction. We can define the meaning of the recording however we want of course but there is no sensible choice other than that flat is meant to be flat in some suitably defined listening-room-agnostic sense.
  
 I am quite sure that people with good "ears" do mentally separate effects of the music from effects of the room to some extent.  This is like how our brains interpret color.  The measured color of light seen reflecting from a printed photo of an orange shirt  is drastically different when viewing the photo in indoor light and outdoor light.  However our brain is aware of both the shirt and the lighting environment where we hold the photo, and we deconvolve the two to interpret the color as being pretty close to the same either way, even though by measurement, it's not!    So it's perfectly reasonable, even necessary to allow that a spectral measurement of the photo is different indoors and out, and if you tried to correct the photo for the viewing environment, you would actually cause a distorted perception!  
  
 For this reason I think it's perfectly reasonable to allow a listening/playback room to do a bit of what it does, including boosting bass in a measurement, if that's what the room does.  
  
 But that's not a reference definition of what the recording fundamentally means and it's probably not even a definition that will be suitable for every room you listen in.  That's allowance for an environment-based presentation of the recording.  Headphones though are a very different playback environment and as I've said before, I think the HTRC *assumption *(not conclusion) that headphones should sound like a good listening room, is just not right.  As I recall Tyll is a bit of a bass head, and I certainly don't mean him any disrespect, but maybe he's finding ways to rationalize it a little?  Certainly not intentionally, and maybe not at all, but anyway, I don't like intentionally distorting the notion of reference, the very definition of a what a recording is meant to mean.


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## Lunatique

biggerhead said:


> I get that there a many details that real engineers in the field are familiar with for making sound come out enjoyable in practice using realistic gear, but it's easy to also rationalize fixes to real world situations or simply habits until reference has no meaning.
> 
> If the microphone on stage doesn't pick up the same bass a person in the audience would, it's obviously not the job of a reference playback system to fix what wasn't recorded.  You might though want your (non-reference) stereo to do that.  Better yet, the recording should probably be modified to fix the problems inherent in the recording system.  Reality is reality though.
> 
> ...


 
 No, Tyll is not a basshead. In fact he's known for not being a basshead. He likes mainly neutral sonic signature with some warmth but enough detail and airiness in the high treble. If he wants a little bit of elevated bass it's only to emulated the visceral feel of speakers. He has learned to appreciate how some sonic signature that can be fun, including bass-heavy ones, but that is not his natural preference. 
  
 You assumption that our ears and brain can acclimate to a room's sound is partially correct. It is true only up to a certain point, but once we get past minor to somewhat moderate coloration, the brain is no longer capable of adjusting itself to that much coloration. It's just like if you take the lamp in a room and then put in a RGB LED light bulb that can change color with a controller. You can push it slightly towards red and if it's not too severe, our brain will adjust to it, just like it does to warm tungsten lights or greenish fluorescent lights, but once you push it far enough towards red, all the relative differences between the hues starts to disappear and past a certain point, all you can see is just red and you cannot see other colors anymore. (I know this stuff because I teach art to visual artists that work in Hollywood special effects and video games, as well illustrators and photographers). 
  
 The only way you can tell how bad the room mode is in any given room, is to actually measure it at the listening position. You can "think" that maybe it sounds decent enough, but unless you are very experienced and knowledgeable in acoustics and audio, you're likely wrong, and ought to do measurements to be sure. 
  
 Have you ever actually been in an acoustically ideal room, such as a professional mastering grade studio, or a very well designed (and likely expensive) listening room such as some rich audiophiles listening room? How about just someone who's done extensive acoustic treatment and applied room/speaker correction? See, if you don't have experience with any of that, your opinions and assertions have only speculation to stand on. You seem to be quite passionate about this stuff, and I suggest you follow your passion and actually become educated about this stuff. I can recommend you books on acoustics and recording studio design, books with audio CDs that teach you critical listening for audio professionals, as well as how to approach local professional audio facilities to request a tour and possibly even get an internship so you can really learn something from a pro audio studio.


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## BiggerHead

Yes, I have been in a professional recording studio.  Simply being in one, with no sound at all, is a sonic experience.


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## BiggerHead

oh.. but we're talking about the bass boost from a reference class room, that's the boost at issue in the HTRC and the one that I think you and I both agree, should probably be there, based on the actual "lighting" of a very good room, not a lousy  one with red light bulbs so to speak.


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## Lunatique

biggerhead said:


> Yes, I have been in a professional recording studio.  Simply being in one, with no sound at all, is a sonic experience.


 
 How about actually sitting down in the audio engineer's chair and then listening to a selection of musical material and hearing how music you're familiar with actually sounds in a mastering grade room?
  
 BTW, a lot places might seem like "professional recording studio" to the lay person, but in reality they might be anything but. Many smaller studios are far from mastering grade, using very cheap monitors in inadequate sounding rooms and improper placements.
  
 This is what professional mastering studios looks like (and some cost millions of dollars):
 https://www.google.com/search?q=mastering+studio&source=lnms&tbm=isch&sa=X&ved=0ahUKEwjrmd3m-oXLAhWEKGMKHalOCRQQ_AUIBygB&biw=1270&bih=1464#tbm=isch&q=professional+mastering+studio
  
 And you might be interested in reading this:http://www.recordingconnection.com/reference-library/recording-entrepreneurs/how-much-does-a-music-studio-cost-2-0611/


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## Lunatique

biggerhead said:


> oh.. but we're talking about the bass boost from a reference class room, that's the boost at issue in the HTRC and the one that I think you and I both agree, should probably be there, based on the actual "lighting" of a very good room, not a lousy  one with red light bulbs so to speak.


 
 I really wish you lived close to me. I would love to have you come over to my studio, and I'll play some music for you on my full-range system and you can hear for yourself just how powerful and visceral the bass sounds and feels. And then we can put a pair of headphones on you that's tuned to Harman Target Response Curve, and you can hear for yourself why it works the way it does, and why that is the result of extensive research. All this talk on paper (or more correctly, typing posts in forums) mean very little without actual experience hearing the sound and comparing them with your own ears.


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## BiggerHead

Just to be clear, we're talking about a 5 or 10db (I forgot at the moment) smooth bass boost.  That's all.  That isn't going to turn a cheap system into a reference one or vice versa.  It's just going to boost the bass 5 or 10 db.  Seriously.


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## BiggerHead

> but overall reference for neutrality won’t necessarily get better by hearing more pieces of gear. Having a reference for how music and sound is reproduced on a calibrated system and room space.


 
  
 Surely you can also get in some sense the most important reference for neutrality by having experience hearing the real thing?  not how it's _reproduced _ anywhere, maybe most relevant for symphonic and orchestral music.
  
 Of course this gives a reference for neutrality of the whole process including recording, not necessarily of the playback equipment by itself.


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## gregorio

Quote:


lunatique said:


> Have you ever actually been in an acoustically ideal room ...


 
  
 I know you weren't aiming your question at me but I'll answer anyway  It depends on what you mean by "ideal"? I've been in a couple of extremely expensive anechoic chambers, which are pretty much ideal for freq response/room modes but very far from ideal as far as mastering is concerned.
  


lunatique said:


> ... such as a professional mastering grade studio ...


 
  
 Yes, quite a few, also quite a few project studios, well designed listening rooms, a couple of expensive audiophile rooms and in addition, I also have my own mastering grade studio but regardless, the two quotes of yours above are in fact NOT synonymous. By definition a mastering room cannot also be an acoustically ideal room! For starters, one sits directly in front of large reflective surface, which pretty much negates any possibility of an "acoustically ideal room", even assuming that a mastering studio would be "ideal" without a console (or other audio equipment), which would not be a correct assumption anyway. Some of the mix rooms I've worked in have price tags in the $10m+ range, my own is closer to $250k and sounds pretty good. None of them are ideal though, even the $10m ones have a few peaks and troughs here and there, peaks and troughs which change as you move position. The $10m ones are generally closer to ideal than mine, although as biggerhead mentioned, one does acclimatise. With enough experience, one can eventually align one's hearing to compensate and reduce translation errors, between say my studio and a world class one. It's only a reduction in translation issues though, not an eradication. BTW, many of the studio images you linked to demonstrate quite severe potential acoustic problems, all that reflection causing equipment, at least some of which probably has cooling fans which of course also affect the noise floor. In my case, most of that equipment is placed in another, acoustically isolated room, leaving only what is necessary in the studio to control that equipment:
  

  
 I'm not quite sure what you mean by "minor to somewhat moderate colouration". Swings of 6dB (or more) at certain frequencies are common in studios, even mastering grade ones, and in comparison to the freq response of say a moderate quality consumer DAC, a 6dB swing would be a very large colouration. I'm not saying that EQ'ing cans is absolutely the wrong thing to do, in some cases I think it might be of benefit, even ignoring the fact that some people might prefer the sound of "wrong". What I'm saying is that tweaking has it's limits, beyond which it's counter-productive and very often audiophiles unwittingly exceed those limits. One way they commonly (and unwittingly) exceed those limits is by making incorrect assumptions such as: professional mastering grade studios are ideal acoustic environments or that mastering is directly related to taking advantage of an ideal acoustic environment. Such incorrect assumptions could easily lead to the logical conclusion that EQ'ing their room/headphones to their idea of "ideal" will get them closer to the intentions of the artists/engineers, whereas in reality, it could just as easily (if not more easily) be getting them further away!
  


> Originally Posted by *Malfunkt* /img/forum/go_quote.gif
> 
> Would be great to hear from some recording industry heavy weights and I'm not talking the 'artists', but the mastering and recording engineers behind the magic to help set the record straight. Would be great to have a video having one of them explain audio calibration and in turn how it could apply to headphones and what one might look for in a reference.


 
  
 I don't think in practice that would help much. Professional calibration comes after professional design, construction and acoustic treatment, then some measuring/testing, then usually a little more treatment and then, final level calibration and maybe a little tweaking with EQ here and there. Those couple of final steps are relatively meaningless without the other steps. My personal opinion and that of other industry professionals (including the heavyweights) is that virtually none of this directly applies to headphones and the general opinion/advice within the industry is not to mix or master on headphones. While headphones are sometimes employed briefly, just to check, the vast majority of the professional/commercial mastering of audio is done with monitors rather than with headphones. That's why you can't find a video of one of them explaining how audio calibration could apply to headphones.
  
 G


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## Lunatique

gregorio said:


> Quote:
> 
> I know you weren't aiming your question at me but I'll answer anyway  It depends on what you mean by "ideal"? I've been in a couple of extremely expensive anechoic chambers, which are pretty much ideal for freq response/room modes but very far from ideal as far as mastering is concerned.
> 
> ...


 
 I don't think anyone would  call anechoic chambers ideal, since they're famous for making people feel really uncomfortable due to the extreme silence and complete lack of reflections. 
  
 The photos I linked to was just a very broad and general search result from google images with the term "professional mastering studio," and it was just to show BiggerHead what they tend to look like. If I were to specifically choose mastering facilities as golden standards, then I'd be more selective. 
  
 The reflective surfaces such as the mixing desk/metering bridge do get considered by speaker manufacturers during the design process, which is why in user manuals, they often give instructions on how to adjust the EQ settings on the backs of the speakers to suit specific environments and speaker placements. There are different suggested settings for placing in corners, on the metering bridge, freestanding without obstruction, flush/next to front wall, etc. 
  
 It's extremely difficult to get a really neutral/accurate sounding room, since that will require designing/constructing from the designs created by experts in acoustics, with non-standard shaped rooms (such as slanted side walls and ceiling), tuned bass resonance traps, etc. There's no doubt that some mastering facilities will have some coloration, but they are still far better than some average Joe's bedroom, living room, garage, etc. Also, room/speaker correction products have made a big difference in how neutral/accurate the playback sounds at the listening position, so as long as the room doesn't have really significant nulls, the correction can get extremely good results. My own standard has always included that final step of room/speaker correction, and that is the standard I judge my headphones by.
  
 You're right about headphones and how many audio pros feel about them, but in the last several years opinions about headphones have been shifting, coinciding with the recent advancements in headphone technology and sound quality, as well as more sophisticated crossfeeds and HRTF plugins. Some audio pros are getting more comfortable with headphones, especially the higher-end Stax systems, amps aimed at audio pros like the SPL Phonitor and Grace design m902, higher-end headphones from Audez'e, Hifiman, Sennheiser, plugins like Redline Monitor, TB Isone, etc. Bob Katz's recent involvement with headphones at InnerFidelity is part of that noticeable shift. Here's one where he talks about EQing headphones and the Harman Target Response Curve: http://www.innerfidelity.com/content/big-sound-2015-bob-katz-eqing-headphones-harman-target-response#lgR1tm6s3SeYdDEO.97


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## BiggerHead

Probably nobody would say an anechoic chamber is an ideal/optimal listening environment, but that doesn't mean it isn't the definitive listening environment.  I think a definitive speaker in that definitive environment should produce measured sound exactly equal to the source signal, probably for any listening angle (so maybe the definitive speaker also isn't ideal). If this isn't the definition of the meaning source signal, then what is? We're saying a listening room is not.
  
 So that comes back to headphones.  Ok, so anechoic chambers aren't ideal because they don't have echoes.  What does that make headphones then? But of course headphones present stereo entirely different from speakers and they don't need echoes.   They retain the original phasing and separation of the echoes originally recorded.
 So why are we trying to reproduce frequency effects of reproduction room echoes in headphones?  Again, this notion was an assumption of the HTRC, not a conclusion.  Before you can ever accept conclusions of a rigorous argument, you have to accept the assumptions.  I don't.


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## gregorio

lunatique said:


> I don't think anyone would  call anechoic chambers ideal, since they're famous for making people feel really uncomfortable due to the extreme silence and complete lack of reflections.
> 
> ... so as long as the room doesn't have really significant nulls, the correction can get extremely good results.


 
  
 Any room with reflections will have significant nulls (and sums), that's why an anechoic chamber can be considered "ideal" acoustically; No reflections, therefore no interaction of those reflections with the direct signal from the speakers. I agree though, an anechoic chamber is far from ideal in practice. Even ignoring the comfort issue you mentioned, it's certainly impractical for consumers to create anechoic chambers and from a mixing/mastering perspective it wouldn't work either, the lack of reflections might give a perfect freq response but would cause other issues such as over compensation with the use of reverb, under-panning, etc. In practice, although absorption of reflections is a major acoustic treatment tool, diffusion is also important in good studio design; to avoid the pitfalls of an anechoic chamber while randomising the reflection interactions. Not "ideal" but as good as it gets because "ideal" as far as studio design is concerned is a contradiction and therefore not attainable.
  


lunatique said:


> The reflective surfaces such as the mixing desk/metering bridge do get considered by speaker manufacturers during the design process, which is why in user manuals, they often give instructions on how to adjust the EQ settings on the backs of the speakers to suit specific environments and speaker placements. There are different suggested settings for placing in corners, on the metering bridge, freestanding without obstruction, flush/next to front wall, etc.


 
   
 Agreed. However, you are implying these features provide a cure, when in reality they provide only a partial/compromised treatment.
  
 Quote:


lunatique said:


> It's extremely difficult to get a really neutral/accurate sounding room, since that will require designing/constructing from the designs created by experts in acoustics, with non-standard shaped rooms (such as slanted side walls and ceiling), tuned bass resonance traps, etc. There's no doubt that some mastering facilities will have some coloration ...


 
  
 Not really, I would say that "there's no doubt that ALL mastering facilities will have some colouration"! Each of the treatments mentioned will reduce the amount of colouration but even all of them in combination will not cure it. For example, my room was designed by an expert acoustician (I'm lucky in that one of my best friends is a recently retired expert acoustician/commercial studio designer and what should have cost many tens of thousands, actually cost me nothing), it was constructed with asymmetrical walls/ceiling, a suspended floor, has custom; tuned bass traps, broadband absorbers and diffusers. My room is very good, it's better than many pro studios, very significantly better than most project studios or well designed listening rooms and in a completely different league to "the average Joe's bedroom" but it's still far from "ideal"! I don't believe the average audiophile has any notion of how truly horrendous the acoustic response of the average room really is. They often talk about being able to identify differences which in effect means they can hear tenths, hundredths or even thousandths of a dB while at the same time appear completely oblivious to room acoustics which probably have one or more freq response swings of 30dB or so, several of around 20dB and numerous swings of around 10dB. All the acoustic designs/construction/treatments combined can very significantly reduce these averages and the very best studios reduce them to below 6dB but that's still, relatively speaking, very significant colouration compared to much of what is discussed in audiophile circles.
  
 The aim of studio design is to create a working environment conducive to producing quality audio. The best studios achieve this aim exceptionally well but in practice this doesn't mean a perfectly flat response (even if that were possible, which it isn't) and indeed, it commonly means a deliberately non-perfectly flat response because as I've mentioned, many/most top studios employ a "house curve". This means that studio design is not therefore pure science, it is at least partly an art, as ultimately, it's based on subjective determinations!
  


> Originally Posted by *Lunatique* /img/forum/go_quote.gif
> 
> My own standard has always included that final step of room/speaker correction, and that is the standard I judge my headphones by.


 
  
 I don't, off the top of my head, know of any commercial studios which don't include that final step. Likewise, I don't know of any commercial studios which only employ that final step! The general rule of thumb for studio design is that one should not attempt to use EQ for more than about 10% of the required acoustic treatment.
  


lunatique said:


> You're right about headphones and how many audio pros feel about them, but in the last several years opinions about headphones have been shifting, coinciding with the recent advancements in headphone technology and sound quality, as well as more sophisticated crossfeeds and HRTF plugins. ... Bob Katz's recent involvement with headphones at InnerFidelity is part of that noticeable shift. Here's one where he talks about EQing headphones and the Harman Target Response Curve: http://www.innerfidelity.com/content/big-sound-2015-bob-katz-eqing-headphones-harman-target-response#lgR1tm6s3SeYdDEO.97


 
  
 I don't believe there has been a "noticeable shift". Although I need to put that statement in context. Yes, many more producers/engineers are mixing on headphones but that's because there are far more project studios today, studios which do not have the resources to achieve even a half decent acoustic room response and therefore even quite expensive headphones represent a very cheap practical solution to the problem of extreme freq fluctuations due to room acoustics. However, it's still a compromised solution for a number of reasons (even with the use of sophisticated crossfeeds and HRTF) and therefore not the preferred solution for commercial studios.
  
 I did not interpret Bob Katz's statements as you appear to have:
  
 1. BK did not stick to the Harmon Target Curve, he just used it as a starting point. From there, he used his intimate knowledge of the freq response of recordings he'd actually mastered, to subjectively adjust the HTC to match. In other words, according to BK, the HTC was not quite right. The difficulty for the average audiophile is that applying the HTC *may* get them closer (to the artists/engineers intentions) but then applying a subjective adjustment might do the exact opposite. The average audiophile does not have BK's reference knowledge of what the master should sound like and is therefore effectively adjusting/matching to their personal preference, which is just as likely to be further away than closer to the artists/engineers intentions.
  
 2. Although BK concluded an obvious improvement in the headphone's FR, I doesn't represent a "noticeable shift" in his position, I believe that's maybe just wishful thinking on your part. As far as I'm aware, BK still masters in his mastering suite with monitors, he has not "shifted" to mastering with headphones.
  
 A couple of points I'd like to re-iterate/re-phrase:
  
 A. Mastering is about using the accuracy/revealing nature of the mastering environment to identify errors/issues which may have been missed in the recording/mixing studio. It's about making adjustments to the final mix, so that it sounds as good as possible on the target audiences' systems, not so that it sounds as good as possible on the mastering suite's system. It would obviously be counter-productive to adjust the mix to take full advantage of a mastering suite's system because no consumers have a mastering suite system and the mastering engineer would therefore be moving further from the actual goal of mastering (making the mix sound as good as possible on the target audience's systems).
  
 B. As counter-intuitive as this may at first appear; with the exception of it's revealing nature, a mastering suite is designed to represent the FR response of the average consumer listening environment/system! One consumer listening environment might have, for example, a +12dB boost at say 180Hz, another is just as likely to have a -12dB cancellation at 180Hz, the average would be 0dB at 180Hz. This is just example and of course could be any almost any dB or Hz and is going to occur numerous times throughout the spectrum. The average though is going to be 0dB, even though no average individual consumer is actually going to comply with this average, just as no individual family actually has 2.4 children. This isn't quite the end of the story though, because despite various unpredictable peaks and troughs, most consumer systems/environments generally induce a bass boost. So, the average throughout the bass range is generally not 0dB but somewhat higher, this is the main reason why most studios have "house curves" rather than attempt to be flat.
  
 The obvious conclusions of these points should be: 1. Attaining a flat response might provide certain worthwhile benefits at certain freqs but in other respects will actually take you further away from the intentions of the artists/engineers. 2. The actual goal/point of mastering already compensates (albeit to a compromised extent) for the traits of consumer systems/environments. Compensating a mix which has already been compensated is also obviously not going to get you closer to the intentions of the artists/engineers and lastly 3. Putting these two points (and the others I've made) together, it should be obvious that there is no "easy/reliable way to achieve the ideal sound", even assuming that the most ideal sound (with a particular system/environment) has not already been achieved by a combination of the system manufacturers and the mix/mastering engineers.
  
 G


----------



## Malfunkt

Wow thanks for the long posts Gregorio. And what you describe actually makes sense. 

Mastering engineers have their work cut out for them. Well some do depending on music style. With so many people listening to Beats headphones - perhaps these should also be reference? Lol I kid (sort of). 

Music has been mastered for compromised listening environments. Probably less than 1% have a highend system with acoustically treated room. How many listeners have actually even place their speakers properly or even sit in the middle of a stereo field. 

I think the headphone trend is actually just starting. Perhaps new technologies will allow a better reproduction of environmental audio. It's going to be a huge need for virtual reality gaming and media. Some manufacturers such as AKG are already experimenting with this (can't remember the headphone but it uses an impulse to measure the response from your ears inside the cup). Effectively, we should be able to map ones physical features (think of a higher end Kinect), which would be mapped to a virtual avatar. This way you'll be able to move in virtual environments and also simulate the effects of reflections within the environment but also off of your own body. 

But that's Star Trek thinking right there. And probably many decades away.


----------



## gregorio

malfunkt said:


> Wow thanks for the long posts Gregorio.


 
  
 No problem. I see mixing and mastering mentioned here quite a lot, and rightly so, it's arguably the single biggest determining factor of sound quality and as this website is effectively all about sound quality, many of the arguments/discussions often come down to mixing and mastering, even when the participants might not realise it. With it being mentioned so much, I feel it's important to explain what it actually is and to dispel some of the myths/incorrect assumptions about it.
  


malfunkt said:


> Mastering engineers have their work cut out for them. Well some do depending on music style.


 
  
 I don't think style/genre really comes into it much. There is generally a little less required with the mastering of classical music but even that depends on a number of factors. There's very little in it between most of the other genres, they all have their quirks, considerations specific to the genre but I wouldn't say one was generally easier or required less work than another.
  


malfunkt said:


> With so many people listening to Beats headphones - perhaps these should also be reference? Lol I kid (sort of).


 
  
 That's not so silly as you think it sounds. My job (when mastering music), as I've mentioned, is to make the mix sound as good as possible on the target audiences' systems. If a significant portion of the target audience is going to be listening on Beats headphones, then I'm going to take into consideration the characteristics of Beats headphones. I'm not going to master on Beats headphones or even master specifically for them, as that would compromise the master for everyone else but it is likely to have some influence on some of the decisions I take.
  


malfunkt said:


> This way you'll be able to move in virtual environments and also simulate the effects of reflections within the environment but also off of your own body.
> 
> But that's Star Trek thinking right there. And probably many decades away.


 
  
 Yes, it is rather Star Trek thinking. Just dealing with the reflections "within the environment" is a far more complex task than most realise. Many would have far more respect/understanding of the complexities of human perception if they had to work with reverb units solidly for a few weeks! Reverb units have many parameters, some have very obvious affects and others appear very subtle or esoteric, at least to start with! Without going into a huge essay on the minutiae of reverb operation, suffice it to say that often times they are creating up to 3,000 reflections per second and processing each one individually. Even so, they're only partially effective because the brain is so efficient at (sub-consciously) building a perception out of all that audio data. In 5.1 (and higher) systems, current reverb unit capabilities are very limited, often requiring workarounds and/or compromises in what we want to achieve. In other words, we're still some way off achieving fully capable/functional reverb units even with a fixed perspective 5.1 system! Until we can get this first step sorted out, it's hard to see how we can even start to think about variable perspective surround sound and, throwing in reflections off one's own body is of course just an additional complication. So yes, probably many decades away before it can be done completely convincingly.
  
 G


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## Lunatique

gregorio said:


> I don't believe there has been a "noticeable shift". Although I need to put that statement in context. Yes, many more producers/engineers are mixing on headphones but that's because there are far more project studios today, studios which do not have the resources to achieve even a half decent acoustic room response and therefore even quite expensive headphones represent a very cheap practical solution to the problem of extreme freq fluctuations due to room acoustics. However, it's still a compromised solution for a number of reasons (even with the use of sophisticated crossfeeds and HRTF) and therefore not the preferred solution for commercial studios.
> 
> I did not interpret Bob Katz's statements as you appear to have:
> 
> ...


 
 The "noticeable shift" I mentioned was more about how today's audios pros are not as scathingly critical of headphones as they were decades ago, due to the advancement of headphones in the last decade or so. Obviously those who have quality speaker monitors will prefer them whenever possible, just as I do, since speakers just sound more visceral, dimensional, and dynamic overall. I would never choose headphones over speakers unless I had no choice (such as if the wife taking a nap in the next room), but because we have pretty good headphones these days, when I do have to wear them, I don't feel too severely limited by it--especially with the custom EQ curves I've created to make them sound more neutral/accurate. 
  
 In my EQ tips, I stressed that the Harmon Target Response Curve is only one of the steps, and the person must also use his ears and listen to the test tones (log sweep, sinewave tones at regular intervals, pink noise) as well as familiar musical material that can expose potential problems. 
  
 Logically and logistically, it makes sense to do mastering while targeting a neutral sounding system, because the range of different possible system is too wide. While it's true that many consumers are using speakers and headphones that that have bass emphasis, but at the same time the opposite is true, with many people that don't even think twice about listening to music using the tinny sounding speakers of their laptops and smartphones, or the speakers of their large screen TV which can't reproduce very low bass. Then there are the public spaces that have PA speakers that are mid-range focused. So it's a tug of war in all these different directions, and I personal feel that the wise thing to do is to use neutral response as the target, so that whatever coloration happens will be limited to each consumer's choice of playback system, and not so much a built-in one that's baked into the music. Of course, there's creative license and the so-called "house sound" is more of an artistic statement. I think a good mastering engineer knows where that line between logistical needs and artistic statement is, and the art of it all is riding that fine line and getting a sound that plays back well in as wide a range of systems as possible while also satisfying the client. 
  
 BTW, what kind of room/speaker correction are you using in your studio? How do you like it? What other competing products have you tried and what did you think of them?


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## gregorio

I have already presented responses to all the points made in your most recent post. We're therefore starting to go around in circles and unless anything new comes up this will be my last response:
  
 Quote:


lunatique said:


> The "noticeable shift" I mentioned was more about how today's audios pros are not as scathingly critical of headphones as they were decades ago, due to the advancement of headphones in the last decade or so.


 
  
 No, I've already mentioned that more of today's audio pros work in project studios, which often makes the use of headphones the only affordable, though compromised, solution. Obviously those audio pros are not going to undermine the quality of their work by being "scathingly critical" of the way they work, with headphones. The "noticeable shift" is therefore not "due to the advancement of headphones in the last decade or so". The advancements have merely narrowed the gap somewhat, not eliminated it.
  


lunatique said:


> In my EQ tips, I stressed that the Harmon Target Response Curve is only one of the steps, and the person must also use his ears and listen to the test tones (log sweep, sinewave tones at regular intervals, pink noise) as well as familiar musical material that can expose potential problems.


 
  
 You seem to have misunderstood point #1 in my previous post. Let me put it slightly differently: How familiar one is with musical material is completely irrelevant in all cases except one; where that familiarity means you have an accurate reference against which to make comparative adjustments. This is where it gets tricky/impossible for the audiophile because without access to the mastering facility which created the master or at least a very similar mastering grade environment, there can be no accurate reference. The only reference is the personal taste of the audiophile, not that of those who created the content.
  


lunatique said:


> Logically and logistically, it makes sense to do mastering while targeting a neutral sounding system, because the range of different possible system is too wide. While it's true that many consumers are using speakers and headphones that that have bass emphasis, but at the same time the opposite is true, with many people that don't even think twice about listening to music using the tinny sounding speakers of their laptops and smartphones, or the speakers of their large screen TV which can't reproduce very low bass. Then there are the public spaces that have PA speakers that are mid-range focused.


 
  
 If a smartphone, laptop or TV can't reproduce low freqs, which I agree they can't, then it doesn't really matter what I do with the low freqs because those consumers won't be able to hear it anyway! I can therefore aim my adjustments in the low freqs to those consumers who do have the equipment to hear those lower freqs and that generally means some bass boost. BTW, I also agree that PA speakers are very mid-range focused, to help the intelligibility of speech (public announcements). However, when dedicated to music playback, they are most commonly coupled with subs. Not a perfect solution by any means but irrespective, the subs are usually balanced with the speakers in such a way as to cause the same basic scenario, of boosted bass.
  
 In reality, I usually don't just ignore the fact that a particular target group may include significant numbers of consumers who will be listening with devices which can't reproduce bass freqs. There are a number of tricks/techniques which can help to partially mitigate this scenario, rather than just leaving a big harmonic and/or rhythmic hole where the bass should be for those consumers. Generally though, these tricks/techniques are applied in such a way as to have minimal affect on the low freqs themselves.
  
 I agree with your logic. Based on your information/assumptions, what you are saying does make logical sense. My point is that your information/assumptions are incorrect or at least partially incorrect and therefore what you are saying is NOT entirely logical!
  


> Originally Posted by *Lunatique* /img/forum/go_quote.gif
> 
> So it's a tug of war in all these different directions, and I personal feel that the wise thing to do is to use neutral response as the target, so that whatever coloration happens will be limited to each consumer's choice of playback system, and not so much a built-in one that's baked into the music.


 
  
 This raises two points:
  
 1. Yes, there are commonly compromises which have to be made to account for "all these different directions" (different consumer listening scenarios) but: A. There are still some generalities which can be applicable to the vast majority of a target audience, without being too detrimental to those outside that target demographic and B. It's not common that we have to make a single master to cover absolutely everyone and absolutely all possible listening environments. It's quite common, even required standard practice in some cases, to create more than one master, precisely for those situations where one single master would have to be too severely compromised for some other significant target demographic/s.
  
 2. If, as mentioned, the goal is to get closer to the intentions of the artists/engineers, then your personal feelings of the "wise thing to do", of how music should be mastered, are completely irrelevant. The only thing which is relevant, is how the artists/engineers did in fact mix/master the work, not how you think they should have, which by definition is your intention rather than that of the artists/engineers! It's your music system and of course you are free to adjust it to your personal tastes, to compensate for how you feel music should be mastered but you can't have it both ways. With no accurate reference, there's a very slim chance that your personal intentions just happen to coincide with the artists'/engineers' exact intentions, especially as your assumptions on mastering are not entirely accurate.
  
 G


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## gregorio

lunatique said:


> BTW, what kind of room/speaker correction are you using in your studio? How do you like it? What other competing products have you tried and what did you think of them?


 
  
 Whoops, I forgot to answer this question. I've already detailed the basic tools used for the correction of my room; design, construction, various acoustic treatments and a little EQ. I have not used any software correction solutions (such as Dirac Live, Audyssey, etc.). I couldn't implement them in my studio setup, even should I wish/need to. For a home environment, I was quite impressed with what Dirac Live was capable of. Not in anyway a desirable alternative to proper acoustic treatment but generally a worthwhile solution where proper acoustic treatment is impractical (say where a wife does not want her sitting room filled with diffusers, broadband absorbers, helmholtz resonators, etc!).
  
 There are a few similar products starting to appear on the market for pro audio use. However, they appear aimed more at the home/project studio market than to my segment. I'm not against them in principle and do try to keep up with developments but currently I'm loath to make any changes to an environment I know and trust, unless I'm pretty certain it will definitely improve my situation (rather than just give the illusion of an improvement). It would take considerable time and effort to install, test and satisfy myself of a real improvement, studio time which I don't currently have, baring in mind my studio is currently carefully calibrated to translate well.
  
 G


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## Koukol

Wow...what a great thread!
 I'm have to pour over this thread later to absorb everything.
  
 I hope I can ask something related regarding EQing.
  
 For while now I've been having a problem with shrillness around the 2K range with every HP I've used (many including the HD650's)
 Some recording do sound great though.
 I think I have ear damage or might be suffering from neurological problems (I just had an MRI) or maybe I own too many brick walled CDs 
  
 In JRiver and Audacity I've tried to eliminate the offending frequency without success.
 I can't reduce it without destroying the upper range creating a huge veil.
  
 Is there an EQ VST that will allow me to De-Harsh yet not veil the upper range?
 If yes...what frequency should I reduce and boost?


----------



## Lunatique

gregorio said:


> No, I've already mentioned that more of today's audio pros work in project studios, which often makes the use of headphones the only affordable, though compromised, solution. Obviously those audio pros are not going to undermine the quality of their work by being "scathingly critical" of the way they work, with headphones. The "noticeable shift" is therefore not "due to the advancement of headphones in the last decade or so". The advancements have merely narrowed the gap somewhat, not eliminated it.
> 
> 
> You seem to have misunderstood point #1 in my previous post. Let me put it slightly differently: How familiar one is with musical material is completely irrelevant in all cases except one; where that familiarity means you have an accurate reference against which to make comparative adjustments. This is where it gets tricky/impossible for the audiophile because without access to the mastering facility which created the master or at least a very similar mastering grade environment, there can be no accurate reference. The only reference is the personal taste of the audiophile, not that of those who created the content.
> ...


 
 We do agree on many points, but just with some different emphases. 
  
 One thing you didn't address is the use of test tones to test of a system's accuracy in frequency response, and it is the most critical part of my method--even more so than using musical material. By using logarithmic sweep, pink noise, sinewave tones spaced at regular chromatic intervals from 20 Hz to 20 KHz, you can EQ your system to be a lot more accurate than without EQ. Of course, the less EQing you do the less possible distortion you'll suffer, so it's always best to first tackle the problem by choosing the ideal room shape, furniture placement, speaker/listening position placement, and using acoustic treatment, and only use room/speaker correction as the final step.
  
 I'm going to assume that when you first put together your studio, you've checked its accuracy with the various test tones I mentioned, since you mentioned you do have calibration for your system. I can't imagine you, or any other audio professional, playing back a log sweep and hearing obvious spikes and dips in the frequency response and then just shrug and not do anything about it. Or if you play back a chromatic sequence of sinewave test tones and hearing that a particular tone (let's arbitrarily say 200 Hz) is significantly louder than the neighboring frequencies, you just mentally note it but don't try to fix it somehow at the system level and instead choose to address it individually in every single piece of music the facility will ever work on.
  
 So if we can agree that audio professionals do have an objective baseline standard for accuracy that we try to aim for, then even if no one reaches 100% accuracy, at least when we are all aiming for it and adjusting our systems to the same objective standard using test tones, we'll be within closer vicinity of each other's system than if we don't do those adjustments. And that is the logic behind why I say if headphone enthusiasts want to get closer to that objective standard of accuracy, EQing is the best way to get there (provided the headphone being used isn't horrendous to begin with). Then from there, the person can choose to add more to the signal chain such as a crossfeed or HRTF plugin like TB Isone, to get the headphone to sound closer to speakers. Again, none of this promises 100% identical playback as heard in the original mastering facility or the original intention of the mastering engineer, but at least it is mitigating severe colorations that's outside the range of what the mastering engineer would consider ideal. If given a choice, I can't think of any reason why any mastering engineer would prefer someone to listen to his mastering on a pair of severely colored headphones, instead of that same headphone having been calibrated to sound more neutral and also using quality crossfeed/HRTF to sound more like speakers. 
  


koukol said:


> Wow...what a great thread!
> I'm have to pour over this thread later to absorb everything.
> 
> I hope I can ask something related regarding EQing.
> ...


 
  
 2 KHz isn't supposed to sound shrill, since it's a bit below the sibilance range. This is what I would do if I were in your shoes:
  
 Take a parametric EQ and create a single narrow band (usually labeled Q or Octave, set to narrowest possible setting), and then set it to about -6 to -12 dB and then sweep it across the entire frequency range back and forth slowly. When you hear that shrillness disappear--that's the offending frequency. A narrow bandwidth should not affect other frequencies so there's no reason why you'd create a veil elsewhere. Once you're sure you've found the offending frequency, adjust the gain until that frequency sounds balanced with the other frequencies. You might also have to adjust the bandwidth too, since we're starting with the narrowest setting. When it sounds balanced with other frequencies (especially neighboring frequencies), you shouldn't hear shrillness. If you are still hearing shrillness, you might want to get your ears checked, since like you suspect, there might be problems with your hearing.


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## Koukol

Thanks, Lunatique
  
 Yes, that's the technique I already use.
 Perhaps I focusing on the wrong frequency or as you said something _is_ wrong with my hearing.
 I usually have problems with keyboards, female vocals and bighting electric guitars.


----------



## Lunatique

koukol said:


> Thanks, Lunatique
> 
> Yes, that's the technique I already use.
> Perhaps I focusing on the wrong frequency or as you said something _is_ wrong with my hearing.
> I usually have problems with keyboards, female vocals and bighting electric guitars.


 
 The part that's strange, is when you said that bringing down 2 KHz would create a veil in the high frequency range. I suspect it's because you're not using a narrow enough bandwidth. Unless you meant 20 KHz, which I doubt, since most people can't even hear 20 KHz (once past teenage years). 
  
 Are you using pink noise to test with? What about log sweep? Sinewave test tones in chromatic intervals? I posted all of them in the original post so you can download them and use them.


----------



## Koukol

lunatique said:


> koukol said:
> 
> 
> > Thanks, Lunatique
> ...


 

 Yes, it's can be anywhere from 1 to 4 K (not 20)
 The problem is I can't find the right balance with the bandwidth.
 It's as though I have to accept the harshness or the veil.
  
_However_, I noticed something new today which may explain my hair pulling.
  
 I used your recommendation of using ReaQ and had some wonderful results.
 But once I clicked apply (in Audacity) and listened to the results they were nothing like I had set.
 The volume even dropped by one or two decibels.
 This could explain why my end results are horrible on my DAP.
  
 So now I'm wondering if my Audacity settings are wrong or if the program doesn't work properly with 3rd party VST's.
  
 And yes, I have to take a moment to understand how to use the tools you posted...cheers


----------



## Joe Bloggs

Koukol Have you tried my video EQ tutorial? 

And yeah, all the VSTs I have work fine with Audacity, so I don't know what's up with that...


----------



## Koukol

joe bloggs said:


> @Koukol Have you tried my video EQ tutorial?
> 
> And yeah, all the VSTs I have work fine with Audacity, so I don't know what's up with that...


 
 I'll have to look into the settings then.
 Is this the tutorial, Joe?
  
 http://www.head-fi.org/t/615417/how-to-equalize-your-headphones-advanced-tutorial-in-progress


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## Joe Bloggs

Koukol It's this one http://www.head-fi.org/t/794467/how-to-equalize-your-headphones-2016-update

It should help you pinpoint your actual problem frequency easily. You can also play with the dB and bandwidth / Q setting on the equalizer you use until the problem frequency is toned down by the correct amount with the appropriate steepness of filter.


----------



## gregorio

lunatique said:


> We do agree on many points, but just with some different emphases.


 
  
  Agreed. Although the differences between us are quite small, they lead to a quite significant difference in our final conclusion.
  
 Quote:


> Originally Posted by *Lunatique* /img/forum/go_quote.gif
> 
> ... you can EQ your system to be a lot more accurate than without EQ. ... I can't imagine you, or any other audio professional, playing back a log sweep and hearing obvious spikes and dips in the frequency response and then just shrug and not do anything about it.


 
  
 In practice, just shrugging and not doing anything about it, is a very common response! Of course, it's not quite that simple in reality. We would first identify the cause of those dips and spikes, as that will suggest an appropriate treatment. Commonly, EQ is NOT an appropriate treatment! In the case of a dip caused by a cancellation for example, EQ is typically not an effective treatment because EQ boosting simply increases the amount of energy equally for both the direct sound and the reflections causing the cancellation, resulting in a net gain of very little or nothing at all. Absorption or the re-direction (diffusion) of those cancelling reflections would be very substantially more effective but, in the case of the reflections being caused by say the mixing console, we obviously can't cover the console in absorber or diffuser panels. There's really not much option other than just shrugging and doing nothing about it! Even in the case of spikes, EQ is sometimes no more than a band-aid rather than a cure. Ideally, we need to think in terms of the time domain itself, rather than just the timing of reflections and the resultant affect on freq response. If a spike is caused by some sort of resonance (or ringing) for example, then we not only have some amount of signal summing but also a substantial increase in the duration of that ringing freq, IE. Not just a freq problem but a time/duration problem. Just using EQ as the treatment may lower the average amount of energy at a particular freq, to the point where the response looks flat but it hasn't addressed the time/duration issue. In other words, to counteract the increase in total energy due to the longer duration of that energy at a particular freq, we've reduced the total energy so our freq response looks flat but if we were to take a snapshot of a particular instant then that freq would have significantly less energy (be a dip). That's why a "waterfall" plot is a useful measurement tool, in addition to just a standard freq response plot. Absorption would probably be the best solution here, but again, applying absorption maybe a practical impossibility.
  
 Shrugging and doing nothing about it is the typical option for problems above about 800Hz, although there shouldn't be too many really serious problems due to the initial design, construction and treatment. Higher freqs are particularly sensitive to very small changes in position. What may have been a 5dB dip at say 1.5kHz may become a 5dB boost, just by moving the measurement mic an inch or two. We obviously can't tune a listening point to just a square inch. Even if we could position our head that accurately all the time, we have two ears which are more than an inch apart! How do you treat that with EQ? 
  
 From all this, a few things should be apparent: 1. Acoustics is one of those audio rabbit hole areas; the more you investigate, the deeper you realise the hole goes! 2. EQ is both a blunt and frequently ineffective acoustic treatment tool. 3. A flat freq response is only part of the picture. It's entirely possible that a "flat" mix/mastering room is neither particularly accurate, particularly neutral nor conducive to producing quality audio, even if creating a "flat" room were attainable in the first place!
  
  Quote:


lunatique said:


> So if we can agree that audio professionals do have an objective baseline standard for accuracy that we try to aim for ...


 
  
 Ah, but this our biggest point of disagreement! There are two elements to my disagreement: The first, I've addressed before and in more detail above. There have been some fairly extreme solutions to the issue of attaining an accurate/neutral response while avoiding the even worse pitfalls of an anechoic chamber, here's an example of such an extreme mastering room solution:
  

  
 While covering almost every inch of the studio in quadratic diffusers probably gives an amazing result, the reflective surface closest to the mastering engineer (and directly between him and the monitors), the console, is obviously not covered in quadratic diffusers. So however flat/neutral/accurate this mastering suite is, it's still probably some way off "ideal". This mastering suite is obviously substantially different from the pictures you previously linked to of other mastering studios and would presumably sound at least somewhat different.
  
 The second element of my disagreement is subjectivity, the personal preference/s of the mastering engineer. Although not a bass-head, I do like a little more bass than average and my tendency would therefore be to add a little too much bass to my masters. I sometimes counter this by adding a few dB of bass to my b-chain when mixing or mastering. Some other engineers add even more, most a little less. Obviously, this is all subjective rather than objective. It's a subjective observation that I tend to prefer a little more bass than others, a subjective determination of how much and a subjective determination of whether to counter it with just personal awareness or by actually altering my b-chain.
  
 Putting these two elements together, I disagree that there is an "objective baseline standard for accuracy". IMO, there is a "subjective baseline standard" for what constitutes an environment conducive to good mastering and typically that means a fairly inaccurate freq response both deliberately and due to unavoidable circumstance. There is no objective baseline standard, all mastering studios are audibly different and all mastering engineers are individuals and at least somewhat different. I've been in some wonderful mastering rooms and also some which I felt were poor enough to preclude me from producing my best work but which don't preclude (and actually aid) other engineers to produce top quality results and, that's even in those cases where my idea of "top quality" is actually identical to another mastering engineer's! What you are "trying to aim for" is based on a fallacy that mastering suites (and mastering engineers) adhere to some objective standard. The actual target you are "trying to aim for" is a creation of your own imagination, of what you think/feel a mastering suite should be, not what they actually are!
  
 I don't dispute your (or anyone else's) right to EQ your equipment however you wish. I also don't dispute that the results of that EQ may very well sound better to you and possibly even to me. What I'm saying is that acoustics, mastering suites and the personal act of mastering itself is a warren of rabbit holes, a complex set of objective and subjective variables. Reducing this to a single variable, reliably/easily treated with a single and rather blunt tool (EQ), is over-simplifying the issue to the point that it's just as likely to be counter-productive. I realise that system tweaking is an integral part of audiophilia for many and therefore any advice against tweaking is an anathema. The question is, what is our goal and if that goal is not tweaking itself, does tweaking get us closer to that goal? In this particular case, if our goal is to create a sound we personally like, then tweak away. If our goal is to try and experience what the artists/engineers intended then the answer is not so simple; tweaking may get us closer or it may take us further away but crucially, we're never going to know unless we're lucky enough to visit the mastering studio where the music was mastered!
  


koukol said:


> For while now I've been having a problem with shrillness around the 2K range with every HP I've used (many including the HD650's)
> Some recording do sound great though.
> I think I have ear damage or might be suffering from neurological problems (I just had an MRI) or maybe I own too many brick walled CDs


 
  
 My advice is to visit an audiologist. Not only will they help you to identify exactly where you have a hearing weakness, making it much easier for you to attempt to EQ around the problem but they may also be able to identify why you have a weakness and/or refer you to a consultant, who in turn might be able to treat the cause and/or help reduce the possibility of further deterioration. If I were you, I would be visiting an audiologist at my earliest opportunity!
  
 G


----------



## Koukol

Thanks guys.
 I guess I better see an Audiologist
 I've never had this problem before and I've been listening with HP for about 40 years.
  
 btw~Reaper is giving me more accurate results.
 I'm wondering if I downloaded the right Audacity version now.


----------



## Lunatique

gregorio said:


> In practice, just shrugging and not doing anything about it, is a very common response! Of course, it's not quite that simple in reality. We would first identify the cause of those dips and spikes, as that will suggest an appropriate treatment. Commonly, EQ is NOT an appropriate treatment! In the case of a dip caused by a cancellation for example, EQ is typically not an effective treatment because EQ boosting simply increases the amount of energy equally for both the direct sound and the reflections causing the cancellation, resulting in a net gain of very little or nothing at all. Absorption or the re-direction (diffusion) of those cancelling reflections would be very substantially more effective but, in the case of the reflections being caused by say the mixing console, we obviously can't cover the console in absorber or diffuser panels. There's really not much option other than just shrugging and doing nothing about it! Even in the case of spikes, EQ is sometimes no more than a band-aid rather than a cure. Ideally, we need to think in terms of the time domain itself, rather than just the timing of reflections and the resultant affect on freq response. If a spike is caused by some sort of resonance (or ringing) for example, then we not only have some amount of signal summing but also a substantial increase in the duration of that ringing freq, IE. Not just a freq problem but a time/duration problem. Just using EQ as the treatment may lower the average amount of energy at a particular freq, to the point where the response looks flat but it hasn't addressed the time/duration issue. In other words, to counteract the increase in total energy due to the longer duration of that energy at a particular freq, we've reduced the total energy so our freq response looks flat but if we were to take a snapshot of a particular instant then that freq would have significantly less energy (be a dip). That's why a "waterfall" plot is a useful measurement tool, in addition to just a standard freq response plot. Absorption would probably be the best solution here, but again, applying absorption maybe a practical impossibility.
> 
> Shrugging and doing nothing about it is the typical option for problems above about 800Hz, although there shouldn't be too many really serious problems due to the initial design, construction and treatment. Higher freqs are particularly sensitive to very small changes in position. What may have been a 5dB dip at say 1.5kHz may become a 5dB boost, just by moving the measurement mic an inch or two. We obviously can't tune a listening point to just a square inch. Even if we could position our head that accurately all the time, we have two ears which are more than an inch apart! How do you treat that with EQ?
> 
> ...


 
 I understand the pain of nulls from room modes too well. I dealt with it for a long time and couldn't do much about it, since EQing a null in an acoustic environment doesn't help much--it just puts unnecessary strain on the drivers. I finally had to add a subwoofer to fill out that null (and luckily the null was below the crossover point of the subwoofer). This is one area that headphones have an advantage, since the acoustic space it taken out of the equation, and that also makes EQing easier since you're not fighting against some stubborn null that can't be fixed.
  
 Audessey's room/speaker correction technology involves time-domain correction too, not just frequency response: https://audyssey.zendesk.com/entries/20352398-Time-Domain-correction-explained
  
 It also uses multiple measurements within the listening area, to accommodate for the movement of the head while seated at the listening position: https://audyssey.zendesk.com/entries/73287-How-does-MultEQ-apply-room-correction-
  
 As for your most important point regarding subjectivity vs objectivity, I think maybe we can look at it from another angle. I agree that mastering engineers have their own subjective taste, since it's as much art as it is science. But let's simplify the issue down to one basic point, which is this:
  
 If given a choice, wouldn't most mastering engineers prefer that the person listening to their masters is using a system that's more neutral/accurate than significantly colored? Regardless if that person can know or achieve the same sonic signature as the mastering facility, or even know what the mastering engineer really intended subjectively, does it not make logical sense that a neutral/accurate system is ultimately more desirable, because it will be able to playback the widest range of different subjective tastes from different mastering engineers without veering off that cliff of a basic standard for fidelity?
  
 I know you'll probably bring up the possibility of corrections maybe causing more harm than good, but that is not a foregone conclusion, since it depends on multiple factors such as the gear being corrected, the tool used for correction, and the knowledge/skill of the person doing the correction. Generally speaking, my stance is that trying to achieve a more neutral/accurate sounding playback system is overall a good thing, and the more care taken with the process and tools and the better the gear is, the more positive the outcome will be. From your previous posts you seem to more or less agree with this.
  
 So it appears the only real remaining issue I'm trying to see if we can agree on, is whether neutral/accurate sonic signature is ultimately more desirable than significantly colored ones (which unfortunately is far too common in consumer audio).


----------



## gregorio

lunatique said:


> Audessey's room/speaker correction technology involves time-domain correction too, not just frequency response ...


 
  
 Yes, I've got a basic grasp of how the technology works. However, given the current state of audio processing capabilities, the practical implementation limitations present within consumer AVRs and the limitations of the average consumer speaker system, I can't see how this technology can apply enough corrections to provide a completely corrected, flat response. Having said this, I have heard the audyssey and Dirac systems and there's no doubt in my mind that for the average consumer they do provide a considerable improvement and indeed I did recommend their use earlier in this thread. We do need to put this improvement in context though, the context being, numerous truly horrific acoustic problems to start with, the level of which most consumers, even including many audiophiles, is far, far worse than they would ever suspect. A situation only complicated and worsened with 5.1 (or higher) home cinema systems.
  


lunatique said:


> It also uses multiple measurements within the listening area, to accommodate for the movement of the head while seated at the listening position ...


 
  
 Again, your use of the word "accommodate" implies a cure. Indeed, it may provide a cure in some freq ranges but I'm highly sceptical it can provide a cure throughout the spectrum, a general improvement sure but not a cure. Logically, it must work on some sort of average (albeit a complexly constructed one) and apply correction to that average rather than attempt to perfectly correct for each listening position individually. This would have to involve some degree of compromise, even to the point of making one particular listening position worse, to improve the majority of the other listening positions measured. Here in the science forum I need to clarify that this is just my personal opinion. I cannot easily integrate a consumer AVR into my studio setup to run the tests required to really understand what is going on under the hood and substantiate my assumptions with real evidence.
  


lunatique said:


> This is one area that headphones have an advantage, since the acoustic space it taken out of the equation, and that also makes EQing easier since you're not fighting against some stubborn null that can't be fixed.


 
  
 Agreed. However, that's not the end of the story, we don't just eliminate all the complex variables of room acoustics, we exchange them for another set of variables and importantly, a set of variables which are extremely difficult to objectively measure, unlike the variables of room acoustics. EQ'ing headphones is near impossible for serious consumers on any basis except subjectivity and, like with speakers and rooms, EQ is again only part of the picture.
   
 Quote:


lunatique said:


> If given a choice, wouldn't most mastering engineers prefer that the person listening to their masters is using a system that's more neutral/accurate than significantly colored?


 
  
 The answer to this is an emphatic "no"! Or more precisely, the answer is "yes", given a specific set of conditions which rarely exist. The answer could only be "yes" if the vast majority of those listening were using a system which is relatively flat/neutral. If it were an equal number or a significant minority, that would be problematic. Under those conditions, I personally would be looking to create a master for consumers somewhere between flat/neutral and the usual "significantly coloured" (or rather, inversely "significantly coloured"). In other words, a master somewhere in the middle and somewhat compromised for both groups of listeners within the target group! Currently, those listening with flat/neutral systems are part of a very small minority, an almost insignificant minority and therefore I can create a master with fewer compromises for that vast majority, fewer concessions to the flat/neutral group. I'm of course rather over-simplifying my approach to mastering but this is the basics of the equation.
  
 There are some circumstances where those specific conditions mentioned above do exist or exist up to a point. For example, mixing audio for cinema. Cinemas are calibrated (albeit not entirely flat and with a fair margin of error) and of course our mix (dub) stages are similarly calibrated, meaning no counter colouration is required. A broadly similar situation existed with SACD. SACDs were relatively expensive, as were the players and neither were the players portable. The logical inferences of these facts, from a mastering perspective, is that: SACD consumers generally had higher quality playback systems than the average consumer (if they spent a significant sum on a SACD player and on content, they probably also had significantly better than average speakers), far better listening environments (SACDs were not playable in cars, trains or other portable scenarios) and SACD consumers would tend to listen more critically, IE. Were more likely to be concentrating on the listening experience, rather than just playing music in the background while doing something else. These inferences made this a rather specific target group and while it didn't necessarily make a huge difference as far as neutrality/flatness were concerned, as even higher end consumer systems are still coloured (although maybe slightly less so, on average) but it did make a significant difference in other aspects of mastering. The amount of audio compression applied perhaps being the most obvious (but not only) example. For this reason, although intrinsically no better than 16/44.1 as far as resolution, dynamic range or any other aspect of sound quality is concerned, SACD does somewhat represent the pinnacle of mastering as far as critical listening with a high quality system is concerned.
  
 G


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## Joe Bloggs

gregorio said:


> For this reason, although intrinsically no better than 16/44.1 as far as resolution, dynamic range or any other aspect of sound quality is concerned, SACD does somewhat represent the pinnacle of mastering as far as critical listening with a high quality system is concerned.
> 
> G




... as long as you don't need any bass redirection or digital room correction, because SACD allows for no such things. :mad:

And as long as the recording and mastering engineers are under no grandiose illusion of being able to mix the whole thing without any PCM-domain DSP. I've heard one such "pure" recording and I could only facepalm.

http://www.head-fi.org/t/782131/why-high-res-audio-is-bad-for-music-take-2


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## gregorio

joe bloggs said:


> ... as long as you don't need any bass redirection or digital room correction, because SACD allows for no such things.
> 
> 
> 
> ...


 
  
 I wasn't making a case FOR SACD, personally I think it's a flawed format which offers no fidelity, resolution or audible benefits over 16/44.1. The reason I mentioned SACD is simply because from a mastering perspective, it presented a highly targeted consumer demographic. There is absolutely no technical reason why the exact same end result could not be achieved with 16/44.1: Simply create two 16/44.1 masters, one standard version and another more "purist" version with more dynamic range (less compression) and generally better suited to more critical listening on higher quality systems. This "purist" version could be called the "hi-fidelity" version or, just as legitimately as say 24/192 or SACD, a "high-def" version. The only reason this doesn't currently happen has got nothing to do with any actual format limitations of 16/44.1 but purely because of the perceived marketing limitations. It's (rightly or wrongly) perceived as more difficult to differentiate, from a marketing point of view, a standard priced 16/44.1 version from a higher priced version in exactly the same format. There is only the marketers' word that one version is worth more than another version but with SACD or 24/192 the marketers have got bigger numbers which they can use to support a difference, regardless of the reality that those bigger numbers don't actually result in any better quality.
  
 I don't believe there are hardly any serious, knowledgeable professional mastering engineers under the "grandiose illusion" you are talking about or indeed the grandiose illusion of 24/192 either. There may appear to be more than there actually are though, I personally know some who publicly support the marketers' claims, even though they're well aware it's BS, because they have families to support and can't risk upsetting/loosing their clients by undermining their marketing strategies.
  
 G


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## Ike1985

Guys, im trying all this for the first time and i'm lost at the beginning. UNDERSTAND that I know basically nothing about EQ'ing. I dont know how to listen for spikes or dips. I think I may have identified spikes based on harshness/grating sound in the high frequencies but am completely lost at finding dips in bass frequencies. 

I'm using jriver on mac, the provided test tones and the marvelGEQ plugin as in EQ in jriver.

Help please


----------



## Lunatique

ike1985 said:


> Guys, im trying all this for the first time and i'm lost at the beginning. UNDERSTAND that I know basically nothing about EQ'ing. I dont know how to listen for spikes or dips. I think I may have identified spikes based on harshness/grating sound in the high frequencies but am completely lost at finding dips in bass frequencies.
> 
> I'm using jriver on mac, the provided test tones and the marvelGEQ plugin as in EQ in jriver.
> 
> Help please


 
 I replied to you via PM, but I"ll post my replies here so others can benefit from it too:
  
 Are you using the test tones I posted? If you listen to the log sweep, you should hear clearly that as the tone sweeps from 20Hz to 20KHz, there will be obvious spike/dips--especially in the 1KHz~10KHz region. You can't miss it--it's very, very obvious. If your ears have no problems and you can discern differences in volume, then you will hear it. 
  
 Same with the sine wave test tones. If you playback the test tones sequentially, you will hear the differences in amplitude at certain frequency intervals. Again, this is usually the most obvious from 1KHz to 10KHz region. Some frequencies in that range will sound especially sharp/loud, while some will sound duller/quieter. Those can be the spikes/dips, but you're not necessarily listening for harshness/dullness--you're listening for relative volume (how loud or quiet each tone "feels" to you when compared to each other). 
  
 As for bass frequencies, are you using a frequency response measurement graph as instructed? It's much easier if you have one (search at InnerFidelity to see if your headphone has been measured). When you can see what the measurement is, it puts into what you hear into much clearer context. It takes some practice to discern the relative difference in energy level between bass frequencies, but if you go with your gut instinct and "feel" which tones sound louder or quieter, you can get pretty close. But keep in mind that 40~50Hz is supposed to "feel" stronger relatively than neighboring bass frequencies, but not overwhelmingly so.


----------



## Ike1985

lunatique said:


> I replied to you via PM, but I"ll post my replies here so others can benefit from it too:
> 
> Are you using the test tones I posted? If you listen to the log sweep, you should hear clearly that as the tone sweeps from 20Hz to 20KHz, there will be obvious spike/dips--especially in the 1KHz~10KHz region. You can't miss it--it's very, very obvious. If your ears have no problems and you can discern differences in volume, then you will hear it.
> 
> ...




I cannot hear bass from 16Hz to 20Hz unless i increas volume, does that mean I need an EQ bump in those frequencies?


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## Lunatique

ike1985 said:


> I cannot hear bass from 16Hz to 20Hz unless i increas volume, does that mean I need an EQ bump in those frequencies?


 
 You're not supposed to be able to hear frequencies beflow 20 Hz--you're supposed to mainly "feel" the low frequency vibration instead. 
  
 Looks like you need to learn some basic lessons about audio. I highly recommend you google search terms like "learn audio basics" or "music production basics." You really should have at least some basic understanding of audio otherwise you're going to be making all kinds of dumb mistakes (but it's good that you're asking, so we can help you avoid those mistakes).


----------



## audioholics

Lunatique,
I don't get the idea with first EQing the headphones and then EQing accordig to the target curve to actually afterwards flatten the response. If you already have flat sound with the first EQ than it is flat according to your hearing, but afterwards if you apply the target curve EQ and then try to flatten that sound again, practically you are neutralizing the target curve. Am I right or am I missing some point here? As I can notice what Tyll Hertsens explains, NAD VISO HP50 are headphones with almost neutral FR that is following the Harman curve which creates the feeling of relatively flat sound. Harman curve pracically compensates for the 20-50 Hz dip, then between 3-4 kHz dip and around 8 kHz peak.


----------



## Lunatique

audioholics said:


> Lunatique,
> I don't get the idea with first EQing the headphones and then EQing accordig to the target curve to actually afterwards flatten the response. If you already have flat sound with the first EQ than it is flat according to your hearing, but afterwards if you apply the target curve EQ and then try to flatten that sound again, practically you are neutralizing the target curve. Am I right or am I missing some point here? As I can notice what Tyll Hertsens explains, NAD VISO HP50 are headphones with almost neutral FR that is following the Harman curve which creates the feeling of relatively flat sound. Harman curve pracically compensates for the 20-50 Hz dip, then between 3-4 kHz dip and around 8 kHz peak.


 
 Looks like you mixed up the process a bit. 
  
 It's actually very straightforward. The end goal is to match the Harman Target Response Curve, as that is the current ideal according to headphone experts. So during the entire process, just keep that in mind as the end goal.


----------



## audioholics

By testing audio according to the Harman curve, it gives somehow weird results in the 3-4 kHz field by boosting them too much. After flattening according to my hearing, I get logarithmic flat, pink noise actually. It looks like Harman curve isn't appropriate for every ear or one must implement it logarithmically (with -3db slope).


----------



## Lunatique

audioholics said:


> By testing audio according to the Harman curve, it gives somehow weird results in the 3-4 kHz field by boosting them too much. After flattening according to my hearing, I get logarithmic flat, pink noise actually. It looks like Harman curve isn't appropriate for every ear or one must implement it logarithmically (with -3db slope).


 
 Make sure you're not confusing the measurement graphs of the "with compensation" and "without compensation." For example,Tyll's headphone measurement graphs are "after compensation," which means the ideal target curve should look more like a very steady and gentle slope of -10 going from 2 KHz to 20 KHz. There is no big rise at 3 KHz shown in a compensated curve. Compare the measurements shown on this page, where the blue and red curves are compensated, and the rough looking gray curves are before compensation: http://www.innerfidelity.com/content/headphone-measurements-explained-frequency-response-part-two#lQj6g6oGKtuyso3M.97


----------



## Lunatique

FYI, Tonebooster (creator of Isone) released a very useful headphone plugin called Morphit. It corrects your headphone's frequency response, simulate other famous headphones, and more.
  
 Check out the thread I just created for it here: http://www.head-fi.org/t/832543/tonebooster-morphit-correct-your-headphones-frequency-response-simulate-other-famous-headphones-and-more


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## DivineCurrent

I posted something in the computer audio forum last month that I thought people would find use for it on here. 
  
 http://www.head-fi.org/t/829549/equalizing-headphones-to-your-ears-with-in-ear-binaural-microphones


----------



## Lunatique

achelgeson said:


> I posted something in the computer audio forum last month that I thought people would find use for it on here.
> 
> http://www.head-fi.org/t/829549/equalizing-headphones-to-your-ears-with-in-ear-binaural-microphones


 
 That is really cool. I didn't know binaural microphones were readily available for purchase even on sites like Amazon. That really helps simplify the process of measuring and EQing headphones.


----------



## DivineCurrent

lunatique said:


> That is really cool. I didn't know binaural microphones were readily available for purchase even on sites like Amazon. That really helps simplify the process of measuring and EQing headphones.



Yeah it's a fun thing to experiment with. Although equalizing a headphone to be flat right at the ear canal entrance is probably not the most accurate thing to do, since what we perceive as neutral doesn't measure perfectly flat sound at the ear. Now an interesting thing to test would be to measure a flat speaker with these microphones, preferably in a room with little reflections, then match the response of the headphones to that.


----------



## Lunatique

achelgeson said:


> Yeah it's a fun thing to experiment with. Although equalizing a headphone to be flat right at the ear canal entrance is probably not the most accurate thing to do, since what we perceive as neutral doesn't measure perfectly flat sound at the ear. Now an interesting thing to test would be to measure a flat speaker with these microphones, preferably in a room with little reflections, then match the response of the headphones to that.


 
 Measuring speakers that way would have the benefit of including each individual's HRTF, instead of using just a measuring microphone placed at the listening spot (which is what IK Multimedia's ARC System does). 
  
 I wonder if there's a way to measure the individual's ear canal's resonance and then factor that into the overall measurement?


----------



## RRod

achelgeson said:


> Yeah it's a fun thing to experiment with. Although equalizing a headphone to be flat right at the ear canal entrance is probably not the most accurate thing to do, since what we perceive as neutral doesn't measure perfectly flat sound at the ear. Now an interesting thing to test would be to measure a flat speaker with these microphones, preferably in a room with little reflections, then match the response of the headphones to that.


 
  
 This is essentially what the Realiser has you do: measure speakers and headphones using the same binaural mics; it then does all the grunt work of making filters to change the cans into the speakers. You can also use the mics to "turn" one set of headphones into another using the same procedure.
  


lunatique said:


> I wonder if there's a way to measure the individual's ear canal's resonance and then factor that into the overall measurement?


 
  
 You would need a probe mic. But since the effect of the canal is (as far as I've read) independent of direction, it would seem you would only need to measure it for those cans inserting into the canal.


----------



## DivineCurrent

I'm curious about the mic that the Realiser uses, I can't find anything like it available. Apparently it is surrounded by foam so you can easily insert it into your ears, and probably get better measurements than what I'm getting with my mics.

I'm also wondering if something like a Stax ear speaker would benefit from EQ more than the HD650. Sure it's got better distortion measurements, but do you guys think EQing an HD650 and either a planar or electrostatic headphone to have the same frequency response would show noticeable differences?


----------



## RRod

achelgeson said:


> I'm curious about the mic that the Realiser uses, I can't find anything like it available. Apparently it is surrounded by foam so you can easily insert it into your ears, and probably get better measurements than what I'm getting with my mics.
> 
> I'm also wondering if something like a Stax ear speaker would benefit from EQ more than the HD650. Sure it's got better distortion measurements, but do you guys think EQing an HD650 and either a planar or electrostatic headphone to have the same frequency response would show noticeable differences?


 
  
 I tried finding specs on it, but couldn't find anything in the A8 manual. It almost seems like they put them together themselves. I guess I should ask in one of the Realiser threads. I use the same binaural mics from your link. I find I have to use the windscreens to get consistent placement, else they slide around like hot butter.
  
 Yes, I think EQing any headphones with significantly different FR will result in immediate differences. If the headphones are low distortion and their response is minimum phase, then a minimum phase EQ should technically be all you need to turn one into the other. There are practical limitations, since you can get some pretty steep dips in the FR due to interactions between your cans and your ears.


----------



## castleofargh

clearly we all buy a Realiser just for the mics. ^_^
  
 for the ear canal resonance, it seems to be mostly a matter of length, at least that's how many people treat it. not that I have any idea how to measure my ear canal without ending up with a hole in my ear drum.
 now some of the newest dummy heads can have ear canal bend, so maybe it's not as insignificant as many made it to be? IDK. we need some of the guys who played with cadavers and couplers to get that kind of answer. so I'd rather just fool around in the 2.5/3khz area with an EQ and settle where it feels right to me. not very objective but I don't know what else to do.


----------



## RRod

castleofargh said:


> clearly we all buy a Realiser just for the mics. ^_^
> 
> for the ear canal resonance, it seems to be mostly a matter of length, at least that's how many people treat it. not that I have any idea how to measure my ear canal without ending up with a hole in my ear drum.
> now some of the newest dummy heads can have ear canal bend, so maybe it's not as insignificant as many made it to be? IDK. we need some of the guys who played with cadavers and couplers to get that kind of answer. so I'd rather just fool around in the 2.5/3khz area with an EQ and settle where it feels right to me. not very objective but I don't know what else to do.


 
  
 It's not that it's insignificant, it's whether its affect changes based upon the relative position of the transducers. For transducers external to the pinnæ data suggest that the effect of the canal is constant. For IEMs, which insert into the canal, you would need a probe mic measurement at the eardrum to get the true story. This sucks because I think we'd all like to be able to get proper HRTF effects with IEMs, but probe mic measurements are a different bear than measurements at the meatus.


----------



## RandyE

Hi, I just registered after reading up on all of this for a while. Main purpose was to flatten my new DT990 pro's a bit since they are kind of bright in the high end.
  
 Managed to pinpoint the exact spikes, at least the main ones, and level them out.
  
  
 For those who have the DT990 pro and wish to slap some EQ on, these are the settings that will eliminate the three prominent spikes in their freq. response :
  
  
*filter 1 : 150Hz, -3db, Q=1.2*
*filter 2 : 6450Hz, -3db, Q=4.1*
*filter 3 : 8875Hz, -7db, Q=5.2*
  
  
*It looks like this on the parametric : *http://imgur.com/LAZ0r3D
  
  
 The way I pinpointed these was first by following this guide, which led me to these big spikes plus some smaller ones.
 By adjusting the Q and levels a bit I was able to sum those few little spikes within these three adjustments.
  
 I noticed only the one fairly gentle (read: low Q) boost of about 3-4db between 75Hz and 500Hz for the lows and lowmids.
 In the highs above 2K there were these two very big spikes between 5K and 7.6K, and between 8K and 10K.
  
 I used the Frequency Response Data from Goldenears to verify that what I was hearing and finding corresponded to their data, which it did rather perfectly.
  
  
 My conclusion, personally, is that in the lows and mid lows, there isn't that much of an issue at all,
 and representation between say 100 and 500Hz is quite flat although a bit boosted across the band.
  
 But then starting @ about 5-5.5K two huge spikes, which to me explains away their overly brightness...
  
  
 You can set these filters in f.e. Equalizer APO or any other kind of EQ to have a more 'true' flat response.
  
 I found that by using only these three filters the end-result is even better than when EQ'ing all the little spikes and cuts, ending up with almost 15 cuts.
  
  
  
 Try it if you have these Beyers, I thought it a nice idea to provide these for others, since nowhere could I find it for the DT990 pro.
  
  
 Cheers.


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## jonathane40

This method seems very interesting. Would anyone who has the Sony xba-n3 be able to follow these steps and report back the results and the best way to eq these headphones as per this method? I can't do it for now because I only have the iem with a balanced cable that has a 4.4mm male plug and can only use it with my DAP. I'm not able to connect these to my computer!

Thanks!


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## Zapp_Fan (Jan 18, 2018)

I haven't read the entire thread, but I'm actually big proponent of using sine tones, although it's definitely necessary to use a sweepable generator and not static tones.  Static tones are next to worthless for this, I'd say.

I have this one bookmarked: http://www.szynalski.com/tone-generator/

One point about the FR alterations you posted for the DT 990s... in my experience you can't necessarily trust peaks >5Khz as manufacturing tolerances can cause variation between individual headphones of the same model.  So I would not do that without at least verifying they have a peak there using a sine sweep.  

For the 990s probably they're manufactured with good enough QA and tight enough tolerances that you won't see wild variation in the treble, but you never know.  Cheap headphone transducers are practically each a unique snowflake in terms of all the craziness you get in the upper treble range.


----------



## bigshot

I agree with you 100% Zapp Fan. Manufacturing tolerances are rarely published, but they often make a big difference, especially in high end headphones. I had a chance to correspond with the fella who designed my headphones and he said that they were aiming for a +/- 1dB sample variation. He told me that most really good headphones have +/- 3dB. I'm sure less expensive headphones have even broader tolerances. The sample variation can end up more important than the target curve if it gets to be too wide.


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## Zapp_Fan (Jan 18, 2018)

bigshot said:


> I had a chance to correspond with the fella who designed my headphones and he said that they were aiming for a +/- 1dB sample variation. He told me that most really good headphones have +/- 3dB. I'm sure less expensive headphones have even broader tolerances.



Can confirm, a good but mainstream-quality headphone may have +/- 3dB specified by the factory.  You have to go out of your way to do better than that.  +/- 1dB is the preserve of really high end stuff.

And unless otherwise specified, that value probably pertains to the averaged frequency response between ears, channel matching is another question altogether.  (That brings up another point - don't be afraid to EQ each channel separately as pretty much all headphones have some L/R variation, the cheaper the headphone, the more imbalance...)

Although we'd all like one DT990 to be perfectly identical to the next, the truth is some variation is allowed and expected in all manufacturing processes, and you can have a tight tolerance, or a tight budget, but not both...


----------



## tili

is there a way to add peak filters to left and right channels separately? seems like my hearing is pretty unbalanced in some frequencies


----------



## Lunatique

tili said:


> is there a way to add peak filters to left and right channels separately? seems like my hearing is pretty unbalanced in some frequencies



There might be some software and hardware EQs out there that allow separate left and right channel settings. You're going to have to do some research into which brand/models (but make sure they are parametric EQs).


----------



## tili

I realised EQ APO (Peace) has this so I use that


----------



## coolkwc

I think i'm a big fans of 'neutral' for every electronics devices that i bought, be it headphone, monitor, DSLR. I'm an electronics engineer, so maybe this is my nature to pursue 'truth' and accuracy for everything.

I bought a colorimeter to calibrate my monitor, i bought a USB dock and expensive piece of AF software to calibrate my DSLR lens at each focal point and distance, i bought the expensive sonarworks calibrated profile for my ATH-m50x, been using APO parametric EQ to tune my speaker.

But i already gave up to 'educate' others how to appreciate neutral, because they simply won't listen. They will refer all EQ tuning = colouring = distortion. The so called 'audiophile' will pursue higher end cabling and equipment to tame the peak or dip in the frequency curve which = non sense in my eye. At the end different people different opinion, is better to save time to share useful info with those who agree and have the same mindset with you rather than wasting time to argue, and importantly save time to enjoy our own music.

Anyway, i can listen to a properly 'balanced' or 'neutralised' headphone and speaker for prolong time without hearing fatigue.


----------



## bigshot

"the purity of precious bodily fluids" to quote Stanley Kubrick


----------



## Lunatique

coolkwc said:


> I think i'm a big fans of 'neutral' for every electronics devices that i bought, be it headphone, monitor, DSLR. I'm an electronics engineer, so maybe this is my nature to pursue 'truth' and accuracy for everything.
> 
> I bought a colorimeter to calibrate my monitor, i bought a USB dock and expensive piece of AF software to calibrate my DSLR lens at each focal point and distance, i bought the expensive sonarworks calibrated profile for my ATH-m50x, been using APO parametric EQ to tune my speaker.
> 
> ...



It really is kind of sad and depressing, because all the efforts we make are only appreciated and understood by a small minority, which is the reason why I kind of stopped making these types of posts. Also, I figured I've already posted enough about these topics, and some of my threads have gotten enough attention that anyone wanting the knowledge can just find them.


----------



## castleofargh

coolkwc said:


> I think i'm a big fans of 'neutral' for every electronics devices that i bought, be it headphone, monitor, DSLR. I'm an electronics engineer, so maybe this is my nature to pursue 'truth' and accuracy for everything.
> 
> I bought a colorimeter to calibrate my monitor, i bought a USB dock and expensive piece of AF software to calibrate my DSLR lens at each focal point and distance, i bought the expensive sonarworks calibrated profile for my ATH-m50x, been using APO parametric EQ to tune my speaker.
> 
> ...


the big issue aside from ignorance, is that there is no neutral for headphones the way we define one for speakers, or to calibrate your monitor. the cause being that headphones don't provide sound in a natural way. with speakers, the sound comes kind of like it would from an actual instrument, it interacts with the room, bounces on our body/head and reaches our ears in similar fashion to any other sound source in the room.
for a screen calibration it's the same. we look at the screen like we would look at anything else. there is no need for magic trick to fool us, just send the same tone and we'll see the same tone. 
now for headphones, we bypass the room and the reflections on the body and head. the ears are included but with a sound source at 90° or close to that on angled headphones. so the cues we get from the sound bouncing on the ear are for that angle, not for the singer in front of us.  so even if we disregard interaural cues, head movements and all those fun HRTF related stuff, and focus only on basic FR, neutral for a listener requires an extra compensation tailored to that listener's body. good will is hardly enough to achieve that for the random guy. I think products like Sonarworks are certainly a step in the right direction, because no matter how different we are, we still tend to have a head on our shoulder and heads smaller than Mars Attack aliens'. so while not exact, chance several aspects of the calibration go in the right direction for most people. but it will be real neutral only to people who fit the model they picked. for the rest, some more or less significant EQ is still necessary. hence attempts to find our own neutral like proposed in this topic. 

I don't disagree with your message, a better balanced frequency response should be the first priority to audiophiles instead of gimmicks and expensive toys to get HI-FI silence at -100dB. my point is only that I understand why most would just give up. they get overwhelmed by all there is to know and all the personal work to put into setting up an EQ properly.


----------



## DivineCurrent

castleofargh said:


> the big issue aside from ignorance, is that there is no neutral for headphones the way we define one for speakers, or to calibrate your monitor. the cause being that headphones don't provide sound in a natural way. with speakers, the sound comes kind of like it would from an actual instrument, it interacts with the room, bounces on our body/head and reaches our ears in similar fashion to any other sound source in the room.
> for a screen calibration it's the same. we look at the screen like we would look at anything else. there is no need for magic trick to fool us, just send the same tone and we'll see the same tone.
> now for headphones, we bypass the room and the reflections on the body and head. the ears are included but with a sound source at 90° or close to that on angled headphones. so the cues we get from the sound bouncing on the ear are for that angle, not for the singer in front of us.  so even if we disregard interaural cues, head movements and all those fun HRTF related stuff, and focus only on basic FR, neutral for a listener requires an extra compensation tailored to that listener's body. good will is hardly enough to achieve that for the random guy. I think products like Sonarworks are certainly a step in the right direction, because no matter how different we are, we still tend to have a head on our shoulder and heads smaller than Mars Attack aliens'. so while not exact, chance several aspects of the calibration go in the right direction for most people. but it will be real neutral only to people who fit the model they picked. for the rest, some more or less significant EQ is still necessary. hence attempts to find our own neutral like proposed in this topic.
> 
> I don't disagree with your message, a better balanced frequency response should be the first priority to audiophiles instead of gimmicks and expensive toys to get HI-FI silence at -100dB. my point is only that I understand why most would just give up. they get overwhelmed by all there is to know and all the personal work to put into setting up an EQ properly.



I am a huge fan of all the stuff Sonarworks has done. They are very underrated IMO. They have also given me a baseline to create my own EQ curves with. The only limitation with this is you can't really fix a headphone with a lot of harmonic distortion very well. You can't give the 40 mm dynamic HD 600 driver the undistorted flat bass of Audeze planar magnetic drivers. However, in terms of tonality, Sonarworks does extremely well with most headphones I've tried, and on the flat curve setting, they all sound somewhat balanced and natural. Definitely a step in the right direction.


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## bigshot

I've found that when it comes to my own ears at least, if the response is balanced, I tend to be more forgiving for other small problems. (I guess unless they sit right in the middle of the sweet spot for hearing around 2kHz.) Balanced always comes first. That's the bulk of the battle.


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## bcaulf17 (Apr 1, 2018)

Thank you, this was very helpful. My 770 Pro 32 ohm check off just about all of these boxes  Sub bass is good (can't really tell if it's rolled off from some of these, but the Massive Attack sample made me feel light vibrations lol), those tracks called bright don't sound too bright from these samples. The only anomaly is Rivers Of Love, I could hear sibilance around that 0:54 mark. Here's Where The Story Ends at 1:10 didn't sound very sibilant at all though. Weird...

Anyway, thanks!


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## deama

Hey so I've been trying to create a balanced EQ for my earphones, but problem is, I don't quite know what to do. I've read the first post here, but I'm still fairly confused. How am I supposed to be able to adjust the EQ by listening to the logarithmic sweep if I have no reference point for neutral sounds or how the sweep is supposed to sound "neutral"? I've never had headphones or even tried any, that were above £100, so I'm fairly certain I won't be able to adjust by just feeling.

Is there perhaps a way I could buy some sort of device that accurately produces a logarithmic sweep, record that on my microphone, then EQ balance my microphone, then tape my earphones next to the microphone, record another sweep, then adjust the earphones? Would that work?


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## Lunatique

deama said:


> Hey so I've been trying to create a balanced EQ for my earphones, but problem is, I don't quite know what to do. I've read the first post here, but I'm still fairly confused. How am I supposed to be able to adjust the EQ by listening to the logarithmic sweep if I have no reference point for neutral sounds or how the sweep is supposed to sound "neutral"? I've never had headphones or even tried any, that were above £100, so I'm fairly certain I won't be able to adjust by just feeling.
> 
> Is there perhaps a way I could buy some sort of device that accurately produces a logarithmic sweep, record that on my microphone, then EQ balance my microphone, then tape my earphones next to the microphone, record another sweep, then adjust the earphones? Would that work?



The most important part is to use your ears, because even if something measures flat/neutral, your unique physiology could still be hearing a skewed frequency response. There are a number of causes--anything from hearing loss, birth defect, to wax buildup that hasn't been cleaned out in a long time (but in this case you CAN clean your ears). 

When you listen to a sweep, what you're listening for is a spikes and nulls in the volume throughout the sweep. This is often the most common in the high-mids and lower-treble, as most headphones suck at producing those frequencies smoothly. Ideally, during the sweep, you should hear the same volume/energy throughout, instead of it getting louder and quieter at various spots. When you listen to the sweep, you need to be looking at a spectral analyzer so you can see what frequencies are the uneven ones, then you can use a sine wave generator or sine wave tones people have made (at regular intervals from 20 Hz to 20 KHz) so you can compared how loud/quiet they are to each other, and then use EQ to adjust until all the frequencies sound at the same volume/energy to your ears. Then you listen to some musical material to check and see how they sound. If they don't sound right, double-check your EQ and make sure the sweep is constant in volume/energy and when you click to play the sine wave tones randomly, they all sound about the same volume. Normally, if you can get all that done, whatever you listen to should sound as close to ideally neutral to your ears.


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## deama

Lunatique said:


> The most important part is to use your ears, because even if something measures flat/neutral, your unique physiology could still be hearing a skewed frequency response. There are a number of causes--anything from hearing loss, birth defect, to wax buildup that hasn't been cleaned out in a long time (but in this case you CAN clean your ears).
> 
> When you listen to a sweep, what you're listening for is a spikes and nulls in the volume throughout the sweep. This is often the most common in the high-mids and lower-treble, as most headphones suck at producing those frequencies smoothly. Ideally, during the sweep, you should hear the same volume/energy throughout, instead of it getting louder and quieter at various spots. When you listen to the sweep, you need to be looking at a spectral analyzer so you can see what frequencies are the uneven ones, then you can use a sine wave generator or sine wave tones people have made (at regular intervals from 20 Hz to 20 KHz) so you can compared how loud/quiet they are to each other, and then use EQ to adjust until all the frequencies sound at the same volume/energy to your ears. Then you listen to some musical material to check and see how they sound. If they don't sound right, double-check your EQ and make sure the sweep is constant in volume/energy and when you click to play the sine wave tones randomly, they all sound about the same volume. Normally, if you can get all that done, whatever you listen to should sound as close to ideally neutral to your ears.


Oh ok, that makes more sense. Do you happen to have a logarithmic sweep sample? The link you posted in the first post is dead.


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## Lunatique

deama said:


> Oh ok, that makes more sense. Do you happen to have a logarithmic sweep sample? The link you posted in the first post is dead.


https://www.youtube.com/results?search_query=sine+wave+sweep


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## deama

Lunatique said:


> https://www.youtube.com/results?search_query=sine+wave+sweep


Oh those work fine too? I thought you needed a super high sample quality of it, ok thanks.


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## Lunatique (Apr 16, 2020)

deama said:


> Oh those work fine too? I thought you needed a super high sample quality of it, ok thanks.


Nope, you don't need super high quality. If you can hear it clearly, it's fine. In fact, the audiophile world too often becomes the audiofool world, wasting money on super high sampling rate and bits and lossless audio, when vast majority of the people in the community can't even reliably tell the difference between the high resolution version and decently encoded mp3 in double-blind tests. When the differences require you to focus so intensely as if you're taking some kind of exam you need to pass, and often you're not even sure you really heard a difference, it's just complete overkill and a waste of money. This is when people need to step back from being audiofools.

BTW, don't use a sweep that takes too long. Use one that's short enough to get through in about 30 seconds or so.


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## bigshot

Remember to account for the range where human hearing is more sensitive in the upper mids, low highs.


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## deama (Apr 16, 2020)

I can't seem to do it, I ended up with something like this for my £20 earphones after several hours of playing with it:
https://imgur.com/wWBpIQE

I listened to music but it sounds very muffled, so do voices.

I basically used this:
https://www.szynalski.com/tone-generator/
and basically either boosted or lowered the Hzs that sounded too high/low.

Is there just some device I could buy or something? I don't think I'm good at this at all; it would help a lot if I had some sort of point of reference, but I've never heard music with a neutral EQ, never been to a concert either.

Oh perhaps you have an EQ profile for cheap earphones that I could try and see if they work for my ones?


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## Lunatique (Apr 16, 2020)

I have updated the original post with a new link to the test tones I uploaded, and updated the YouTube links as well.

The forum only allows 30 media links per post, so I'm moving the ones past #30 here, and I'll provide a link to this post from the original post.

*For fun/enjoyment (these can be used for a variety of reasons, but they're just fun to listen too):

02 Risingson / 04 Inertia Creeps / Angel* - These three tracks from Massive Attack's Mezzanine album are just so fun to listen to on a well-tuned system. Very atmospheric and dramatic.




*09 - hot fuji* - Another fun track to listen to with a hard hitting sound.


*06 - 夏雲 (Summer Clouds)* - A lively string arrangement that's also a joy to listen to. Great for listening to the mids.


*01 - Shoudo Satsuriku* - Percussion driven track that mixes traditional Japanese instruments with modern instruments.


*04. 融了鐘時間 (Melted Clock)* - Fun to listen to for stereo imaging.


*03 Dreaming In Colour* - A very nice electronic track with complex arrangement that starts off with repeated sub-bass sweep tones.


*02 - MISTY LOVE* - The snare drum sound in this track is good for detecting too much brightness. It is sharp, but it shouldn't sound piercing and grating. Also, that sixteenth note hi-hat on the left channel--once the full arrangement kicks in, it tends to get totally buried if the frequency response is not well-balanced. Even on neutral sounding systems, it will sometimes get buried by the rest of the arrangement, but it will come back here and there. If you just can't hear it at all, then the treble is too recessed.


*05 - Medicine Mix* - A good track for solid thudding bass kick.


*Jesper Kyd - Hitman Contracts CD 1 [2004] - 05 - Slaughter Club* - Really enjoyable full-on electronic track that's got a driving momentum.


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## Lunatique

deama said:


> I can't seem to do it, I ended up with something like this for my £20 earphones after several hours of playing with it:
> https://imgur.com/wWBpIQE
> 
> I listened to music but it sounds very muffled, so do voices.
> ...


There's no way to just use any EQ curve and apply to your headphones and expect it to work. The EQ has to be surgically tweaked for your exact headphone model, because every model has different frequency response. 

There are apps out there that has EQ curves for the most popular headphones, but I don't think they contain EQ curves for any cheap headphones that aren't prominently known. You can check to see if yours is included:
https://www.sonarworks.com/reference/headphones
https://www.toneboosters.com/tb_morphit_v1.html

I updated the original post with links to the test tones and YouTube videos. Maybe repeat the process again with those updated links and see if it helps. Make sure you understand everything I had wrote before moving on to the next section. For example, you must remember that the high-mids has much more perceived energy at the same amplitude compared to other frequencies, and you can't overcompensate by over-correcting it, or else you end up with a sound that's not bright enough.


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## deama (Apr 16, 2020)

Lunatique said:


> For example, you must remember that the high-mids has much more perceived energy at the same amplitude compared to other frequencies, and you can't overcompensate by over-correcting it, or else you end up with a sound that's not bright enough.


That's probably the biggest issue. The curve I posted above somewhat works, it fixes the annoying "sting" some songs have that I hate, but end up making the whole thing muddy, probably overdid it as the bass sounds too high, I think. So how would I fix that part? Do I just need to keep in mind that the high-mids must be lowder than what I'm hearing?

EDIT: I have an audio technica ATH-M20x pair that I found and EQ profile thing for in one of the links you gave me. Ok now I at least have a reference point. Yeah, the sound does sound pretty "neutral", pretty nice actually.j

The software has a free trial, but I was hoping to be able to rip the EQ profile from them and use it on my EQ APO, any idea if you can do that?


----------



## Lunatique

deama said:


> That's probably the biggest issue. The curve I posted above somewhat works, it fixes the annoying "sting" some songs have that I hate, but end up making the whole thing muddy, probably overdid it as the bass sounds too high, I think. So how would I fix that part? Do I just need to keep in mind that the high-mids must be lowder than what I'm hearing?
> 
> EDIT: I have an audio technica ATH-M20x pair that I found and EQ profile thing for in one of the links you gave me. Ok now I at least have a reference point. Yeah, the sound does sound pretty "neutral", pretty nice actually.j
> 
> The software has a free trial, but I was hoping to be able to rip the EQ profile from them and use it on my EQ APO, any idea if you can do that?



It helps if there's a measurement graph for the headphone that you can look at, but if it's not a popular enough headphone, no one would have done a measurement and posted it online, and you can't do a good enough measurement yourself without professional equipment. 

I don't know of any way to rip the EQ curve, and it's not something head-fi would support anyway, because it's similar to pirating. 

Since you have the M20X and it's in their profiles, you can use it as reference to do the EQ curve for your other headphones.


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## headdict

deama said:


> That's probably the biggest issue. The curve I posted above somewhat works, it fixes the annoying "sting" some songs have that I hate, but end up making the whole thing muddy, probably overdid it as the bass sounds too high, I think. So how would I fix that part? Do I just need to keep in mind that the high-mids must be lowder than what I'm hearing?
> 
> EDIT: I have an audio technica ATH-M20x pair that I found and EQ profile thing for in one of the links you gave me. Ok now I at least have a reference point. Yeah, the sound does sound pretty "neutral", pretty nice actually.j
> 
> The software has a free trial, but I was hoping to be able to rip the EQ profile from them and use it on my EQ APO, any idea if you can do that?


If you're lucky, your earphones might be included here: https://www.head-fi.org/threads/eq-settings-for-700-headphones.885196/

I once had an EQ profile that worked for me, but was not usable on my preferred player and there was no way of extracting the parameters. So I played a sine sweep using the EQ, recorded the output and analyzed the resulting frequency response in Audacity. Then I was able to reconstruct the EQ from scratch. It was not perfect, but good enough for me. I'm no lawyer, so can't tell you if this procedure would be legal in your case.


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## deama

headdict said:


> If you're lucky, your earphones might be included here: https://www.head-fi.org/threads/eq-settings-for-700-headphones.885196/
> 
> I once had an EQ profile that worked for me, but was not usable on my preferred player and there was no way of extracting the parameters. So I played a sine sweep using the EQ, recorded the output and analyzed the resulting frequency response in Audacity. Then I was able to reconstruct the EQ from scratch. It was not perfect, but good enough for me. I'm no lawyer, so can't tell you if this procedure would be legal in your case.


I looked at the list there and found mine:
https://github.com/jaakkopasanen/AutoEq/tree/master/results/rtings/avg/Audio-Technica ATH-M20x
But what does that Q mean? It looks quite different from the one I got from the free trial.


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## bigshot

EQ is a process, not a destination. You can dial in the perfect curve from a theoretical standpoint, but it might not suit the way you actually hear or your tastes. I'd suggest looking at published curves for particular models as a starting place. Once you find the proper correction for your model, if something doesn't sound right, make a small change and listen to music with it and see if it's better. If not, go back to the published curve and try to figure out what would improve it. If it does help, try a little more in that direction and see if it is moving towards better sound. Use the published curve as a baseline to make your improvements upon. Take baby steps and evaluate. Don't make changes in more than one part of the range at one time, and don't make huge changes. Slow and steady wins the race. Also make sure you correct subtractively... instead of boosting something that is low, subtract from everything around it. You don't want to boost above zero or you will get clipping distorting your sound. If at the end, the overall curve is too low, you can boost it all equally across the range so the highest peak isn't over zero.


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## bigshot

deama said:


> But what does that Q mean? It looks quite different from the one I got from the free trial.



Q is the width of the correction. It only applies to parametric equalizers.


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## gargani

bigshot said:


> Q is the width of the correction. It only applies to parametric equalizers.


When looking at the numbers for Q. Does a higher number mean a wider range of correction and a lower number mean a narrower range of correction?


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## sander99

gargani said:


> When looking at the numbers for Q. Does a higher number mean a wider range of correction and a lower number mean a narrower range of correction?


Other way round. With a high Q you can create a narrow peak or dip.


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## deama

Alright so I managed to take the graph they gave for that free trial software, used a script to make the window transparent and through-clickable, then I just traced the EQ points on EQ APO and ended up with a very accurate EQ graph. So one problem solved.

Now I need to do an EQ tweak thing for my earphones... at least I have a reference point now.


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## deama

Also, I managed to find my earphones on the github repo, however that github repo I don't think has 100% got the right neutral balance going for it, at least when I tried the config for the audio techinca m20x. I'll try going through the Hz stuff again tomorrow or so, see if I can get it better.

The profiles from sonarworks seem to be the best, but their software isn't great.


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## gargani

sander99 said:


> Other way round. With a high Q you can create a narrow peak or dip.


Got it. Thank you.


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## deama

I was also wondering, is it worth it to EQ balance a microphone too? Any links to a repository that has a list like with the headphones one?


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## bigshot

Most microphones are pretty flat. A lot of EQing goes on in the mix, but that is mostly to weave sounds in and around each other so they don't fight.


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## Lunatique

bigshot said:


> Most microphones are pretty flat. A lot of EQing goes on in the mix, but that is mostly to weave sounds in and around each other so they don't fight.


There are some mics out there that are voices in a specific way to sound pleasing for certain applications, such as recording vocals, guitars, drums, etc., so it's a good idea to look up the measurement data for the specific mic you want to get. I have an Audio-Technica condenser mic that's significantly brighter than neutral, while my Shure SM7 is much more neutral. Ideally, you want to get what's called a "measurement" mic, as they are especially designed to be very flat and neutral, as they are used for measuring sound. For example, I use IK Multimedia's ARC System 2, and it comes with a measurement mic that's very accurate, but it is a model that's rebranded and could also be bought from the original brand. I don't recall which, but a little investigative work will turn the name up.


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## bigshot

I should have said pro mikes are mostly pretty flat. Consumer mikes can vary.


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## Lunatique (Apr 18, 2020)

bigshot said:


> I should have said pro mikes are mostly pretty flat. Consumer mikes can vary.


I think it depends on application. The mics I mentioned are all pro audio mics meant for professional studio recordings. Often mics that are "good for vocals" or any other specific applications will be tuned to not be neutral. For example, having "tube-like warmth" is often a desirable trait in popular professional mics, so you need to know about the specifics of the mic you want to use if you want to EQ it.



deama said:


> I was also wondering, is it worth it to EQ balance a microphone too? Any links to a repository that has a list like with the headphones one?



There are plugins that can emulate various famous microphones, so those have build in EQ curves for each preset. You just specify which model/brand of mic you're using, and then pick the mic you want to emulate. It works similarly as the Sonarworks and Morphit plugins.

It's going to be hard to try to measure and EQ a mic on your own, because first of all, whatever you're using to play test tones for the mic to record will have to be able to play back test tones with perfect neutral accuracy at the mic's position, and that in and of itself is already a great challenge. If you just want a very neutral/accurate mic, just buy a measuring mic and be done with it.


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## bigshot (Apr 19, 2020)

I've never worked with colored mikes and if an engineer suggested it to me, I would probably ask them why they couldn't add coloration in the mix. My goto mike for voice is a Neumann U-87, and it is pretty flat. The trick to miking is mike placement, not the coloration. Likewise the tube mic pres I've worked with are stone flat. They cost as much as a nice car, they had better be! Gregorio knows more about this stuff than I do. I just supervise the recording and mix, I don't engineer it. But I expect recording to be flat and clean. If I want to adjust, I do that in the mix.


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## Lunatique

bigshot said:


> I've never worked with colored mikes and if an engineer suggested it to me, I would probably ask them why they couldn't add coloration in the mix. My goto mike for voice is a Neumann U-87, and it is pretty flat. The trick to miking is mike placement, not the coloration. Likewise the tube mic pres I've worked with are stone flat. They cost as much as a nice car, they had better be! Gregorio knows more about this stuff than I do. I just supervise the recording and mix, I don't engineer it. But I expect recording to be flat and clean. If I want to adjust, I do that in the mix.


That's my general stance about audio gear as well--if I wanted coloration, I prefer to add it myself, and my gear shouldn't have some kind of permanent coloration built into it. I recently spent a lot of time researching and trying a bunch of bass combo amps, and it was very disappointing because all the big name brands you'd find at Guitar Center had significant coloration in their products. Eventually, I found the most neutral/accurate one in Phil Jones products (I ended up with a Flightcase BG-150). 

But it's surprising how many people in the pro audio world favor build-in coloration, selecting gear based on the unique voicing of specific amps, mics, mixing consoles, studio monitors, or whatever, because their unique colorations are deemed pleasing and/or useful.


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## deama

Oh, I might have to drop the mic stuff then, as I wouldn't like to spend much money onto it, and I doubt there's any EQ specification for my £40 microphone.


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## bigshot

That's why most pro microphones capture full range sound clean. You can always take away in the mix, but you can't put back what isn't there.


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## Lunatique

deama said:


> Oh, I might have to drop the mic stuff then, as I wouldn't like to spend much money onto it, and I doubt there's any EQ specification for my £40 microphone.





bigshot said:


> That's why most pro microphones capture full range sound clean. You can always take away in the mix, but you can't put back what isn't there.


Your safest bet for any kind of critical measuring using a mic, is to get a measuring microphone, designed specifically to be extremely flat in frequency response. It is not recommended to just use any professional mic because they are not designed to be ruler flat. Here are some examples of measuring mics:
https://www.sweetwater.com/store/search.php?s=measuring+microphone

The one that came with my ARC System 2 is one of those but IK Multimedia took the branding off when they included it in the ARC System package.


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## deama

Lunatique said:


> Your safest bet for any kind of critical measuring using a mic, is to get a measuring microphone, designed specifically to be extremely flat in frequency response. It is not recommended to just use any professional mic because they are not designed to be ruler flat. Here are some examples of measuring mics:
> https://www.sweetwater.com/store/search.php?s=measuring+microphone
> 
> The one that came with my ARC System 2 is one of those but IK Multimedia took the branding off when they included it in the ARC System package.


Oh, the price is not as high as I thought.
Would this one work?
https://www.sweetwater.com/store/detail/ECM8000--behringer-ecm8000-measurement-condenser-microphone

So basically I use the above measuring microphone instead of my normal one, and then just apply some light EQ for whatever purposes I need, correct?


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## castleofargh

Sometimes you can find the version from Sonarwork that comes with individual calibration file. If you can find those at the same price, that's not a bad idea. 
I haven't followed the conversation but I assume you're aware that this ECM8000 needs an ADC with XLR input and phantom power? 
Some microphones comes conveniently as an all in one USB device(so the ADC is in the mic), which is very cool, but some room calibration software might not work with those for some reason(if you're only going to bother with basic frequency response, you don't have to care about that).


----------



## Lunatique

deama said:


> Oh, the price is not as high as I thought.
> Would this one work?
> https://www.sweetwater.com/store/detail/ECM8000--behringer-ecm8000-measurement-condenser-microphone
> 
> So basically I use the above measuring microphone instead of my normal one, and then just apply some light EQ for whatever purposes I need, correct?


Can you describe what you're trying to achieve? I'm assuming you want to use the mic to do recordings (voice, instrument), and then you're going to apply EQ to mimic the sonic signature of other microphones with desirable characteristics? If that's what you are trying to do, I wonder if it's just as feasible to simply EQ your current mic to be flat, then apply the emulation EQ curve. But that might not be necessary--I'll explain later. 

The difference between applying EQ to headphones and to microphones, is that one is playback gear, and one is recording gear. One is free from room mode and acoustics issues, and one is not. This makes microphones a lot more complicated. Your mic placement in your recording space and whether the space is acoustically treated will have impact on your recording. The type of mic you use and the mic pattern you choose (some mics allow you to switch the pattern) will also affect the results. There are things you can do like using a mic shielding baffle, treating your room, etc., to get as clean of a recording as possible. 

Very often the most desirable mics used in recording studios are not necessarily the flat sounding ones. Recording engineers choose mics and mic-preamps according to their coloration, and match them to different needs. For example, often a warm sounding mic is desirable for certain vocals, or certain mics are used to mic the bass drum, some are better for cymbals, or for the toms, or the snare. Some are great for amped electric guitar but not for acoustic guitar, and so on. If professional microphones are all just flat this wouldn't be the case. So depending on what you're trying to do, ultimately, you need to know what characteristic you are after and why. But if you know that, then you can simply just EQ that tone yourself based on the need. But I suppose if you want to save time, you can get one of those mic emulation plugins and choose among the models they have in the presets. But going that route, I'm assuming it'll be easier if the mic you're recording with is one that's a very popular and ubiquitous model, as those plugins likely will be basing their emulation on those very popular mics as the starting point. You'll need to check out the specifics of those plugins to be sure.


----------



## deama

Lunatique said:


> Can you describe what you're trying to achieve? I'm assuming you want to use the mic to do recordings (voice, instrument), and then you're going to apply EQ to mimic the sonic signature of other microphones with desirable characteristics? If that's what you are trying to do, I wonder if it's just as feasible to simply EQ your current mic to be flat, then apply the emulation EQ curve. But that might not be necessary--I'll explain later.
> 
> The difference between applying EQ to headphones and to microphones, is that one is playback gear, and one is recording gear. One is free from room mode and acoustics issues, and one is not. This makes microphones a lot more complicated. Your mic placement in your recording space and whether the space is acoustically treated will have impact on your recording. The type of mic you use and the mic pattern you choose (some mics allow you to switch the pattern) will also affect the results. There are things you can do like using a mic shielding baffle, treating your room, etc., to get as clean of a recording as possible.
> 
> Very often the most desirable mics used in recording studios are not necessarily the flat sounding ones. Recording engineers choose mics and mic-preamps according to their coloration, and match them to different needs. For example, often a warm sounding mic is desirable for certain vocals, or certain mics are used to mic the bass drum, some are better for cymbals, or for the toms, or the snare. Some are great for amped electric guitar but not for acoustic guitar, and so on. If professional microphones are all just flat this wouldn't be the case. So depending on what you're trying to do, ultimately, you need to know what characteristic you are after and why. But if you know that, then you can simply just EQ that tone yourself based on the need. But I suppose if you want to save time, you can get one of those mic emulation plugins and choose among the models they have in the presets. But going that route, I'm assuming it'll be easier if the mic you're recording with is one that's a very popular and ubiquitous model, as those plugins likely will be basing their emulation on those very popular mics as the starting point. You'll need to check out the specifics of those plugins to be sure.


Oh right ok, thanks.


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## Lunatique

deama said:


> Oh right ok, thanks.


BTW, someone had just asked something similar on gearslutz (one of the most popular forums for pro audio folks), and there's a lot of discussion about it--similar to what I've told you, but much more in-depth: https://www.gearslutz.com/board/low-end-theory/1303304-mic-preamps-they-really-needed.html


----------



## Blackwoof

Been testing out differing levels of low shelf on my ER3SE/ER4SR. With a +6db 125Hz the bass sounds like what my Westone W40 could do, Which is great since it only 1 BA.


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## Metalomaniac

I'd like to delve deeper into this to get my 2nd dan black belt in audiophilia, but I'm not sure my gear is suitable. I have Hiby R5 Android DAP. It doesn't have a parametric EQ. I listen almost exclusively on Spotify, which doesn't have one, either. 

Any options?


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## bigshot (Jul 9, 2020)

The biggest problem is finding a decent equalizer. I don't think they make a good one for a portable rig- only a home setup. Probably because they figure no one really cares about fine tuning sound quality when they're in the street or on a train. For home use, MiniDSP makes some good stuff. I have one of their units on my wish list. If you find something useful, let me know.


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## Axxelred

Interesting first post. Just posting to save this thread.


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## SenorChang8

Axxelred said:


> Interesting first post. Just posting to save this thread.



There’s a useful “watch” function to save specific threads and forums.


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## Axxelred (Oct 23, 2020)

I looked but I didn't found it.


Edit : thanks it's fine now


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## Joe Bloggs

Metalomaniac said:


> I'd like to delve deeper into this to get my 2nd dan black belt in audiophilia, but I'm not sure my gear is suitable. I have Hiby R5 Android DAP. It doesn't have a parametric EQ. I listen almost exclusively on Spotify, which doesn't have one, either.
> 
> Any options?


Yes!  R Android players now have a Convolver function which lets you modify your audio output with an impulse response.  If you EQ a base impulse with a parametric EQ on your computer, you can then use this on the R5.  Details here near the bottom of the post:  https://www.head-fi.org/threads/r5-a-new-android-dap-by-hiby.912566/page-209#post-15892903


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## rmsanger

Hoping to get some advice:

I currently have a variety of headphones that I source from my turntable and Fiio X7 DAP. I don't use a pc connection so using a windows based eq won't work. I also don't want to get something like an RME ADI-2 pro dac that has a built in EQ as I already have an R2R dac that I love. Finally the Fiio X7 has a built in eq function but it requires you to use the onboard dac; if you switch to lineout via Coax it overrides any onboard eq settings.

So that leaves me with adding a new piece of hardware into my chain specifically for eq functionality. I could go something like this:

Pioneer GR-860 graphic Equalizer - https://classifieds.ksl.com/listing/61796700
Schiit Loki - https://www.schiit.com/products/loki
Bellari Audio Exciter - https://www.amazon.com/Bellari-SE560-Sonic-Exciter-Enhancer/dp/B07FDQLVTY
Bellari EQ570 - https://www.amazon.com/Bellari-Audio-EQ570-Equalizer/dp/B07PPQ3XWM


But I want to be careful an not introduce anything detrimental into the sound-chain like THD as I've already got a pretty clean setup. Does anyone have any advice or suggestions on these or other options?


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## bigshot

I wouldn't worry about distortion with a digital EQ. Just pick the one with the most flexible features (parametric with the most bands).


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## rmsanger

bigshot said:


> I wouldn't worry about distortion with a digital EQ. Just pick the one with the most flexible features (parametric with the most bands).



Do you think this one for $100 would be fine? 

Pioneer GR-860 graphic Equalizer - https://classifieds.ksl.com/listing/61796700


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## bigshot (Nov 13, 2020)

That isn't a digital EQ. That is analog. And it isn't a parametric equalizer. It's graphic. You want a digital parametric equalizer. The other ones you've linked to aren't really equalizers. They're fancy tone controls. You want something like a MiniDSP. You'll need to figure out what kind of digital in and out you can use with your DAP. It might get complicated.

For your turntable, you're stuck because it is analogue. The best quality analogue EQ you can get would be best for that. I have a RANE myself but DBX makes good ones too. A dual channel 31 band would work well. But you are going to need to raise the output of your turntable to line level to run it through the equalizer. You could take line level out of your DAP and run it through a good analog equalizer too. It wouldn't be optimal, but it would work.

Someone else might know of a portable equalizer. I've always wanted one, but haven't found anything I liked yet.


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## rmsanger

Could you make a few suggestions then of a digital parametric eq or minidsp you would recommend?


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## bigshot (Nov 13, 2020)

It depends on what kind of digital in and out you have. Do you have HDMI in and out? The EQ patches in-between the player and the amp. If you have multiple inputs, you will need a preamp to act as a switcher in front of the equalizer.

I use a AV Receiver with built in 5 band parametric on every channel. But I don't think that would work for  headphones. Older receivers had tape loop inputs and outputs that you could use to patch an equalizer into and have it work with every input. But I haven't seen anything like that in many years.


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## rmsanger

bigshot said:


> It depends on what kind of digital in and out you have. Do you have HDMI in and out? The EQ patches in-between the player and the amp. If you have multiple inputs, you will need a preamp to act as a switcher in front of the equalizer.
> 
> I use a AV Receiver with built in 5 band parametric on every channel. But I don't think that would work for  headphones. Older receivers had tape loop inputs and outputs that you could use to patch an equalizer into and have it work with every input. But I haven't seen anything like that in many years.



My chain is:

Fiio X7 DAP (lineout/coax) ->  Holo Audio Spring KTE dac (XLR / RCA) -> Phonitor E / Holo Mommoth KTE -> Headphones


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## bigshot

A digital equalizer would go between your DAP and your DAC. You would need a equalizer that has coax in and out. That may be tough. Someone else might know of something that would work. I'm afraid I don't.

You're probably going to have to go analog with a dual 31 band EQ and take the XLR output of the DAC and plug the equalizer into the amp with XLR. With that route, the more you spend, the better precision you get. The cheapest would probably be the DBX 231s at about $150. The high end would be the DBX2231at about $550. You'll want to look over the features of the DBX equalizers and see what suits you. The higher end ones might have limiters and noise reduction you don't need.


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## rmsanger (Nov 13, 2020)

bigshot said:


> A digital equalizer would go between your DAP and your DAC. You would need a equalizer that has coax in and out. That may be tough. Someone else might know of something that would work. I'm afraid I don't.
> 
> You're probably going to have to go analog with a dual 31 band EQ and take the XLR output of the DAC and plug the equalizer into the amp with XLR. With that route, the more you spend, the better precision you get. The cheapest would probably be the DBX 231s at about $150. The high end would be the DBX2231at about $550. You'll want to look over the features of the DBX equalizers and see what suits you. The higher end ones might have limiters and noise reduction you don't need.



Have you heard of Klark Teknik DN370 as it seems to get very good reviews on the pro audio forums (gear slutz).  I would assume this would be down the same area as the DBX eqs.

https://reverb.com/item/31188350-klark-telnik-dn370

https://www.klarkteknik.com/product.html?modelCode=P0ACA


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## bigshot

I don't know anything about that brand, but it looks like it would do the trick well.


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## colonelkernel8

rmsanger said:


> Have you heard of Klark Teknik DN370 as it seems to get very good reviews on the pro audio forums (gear slutz).  I would assume this would be down the same area as the DBX eqs.
> 
> https://reverb.com/item/31188350-klark-telnik-dn370
> 
> https://www.klarkteknik.com/product.html?modelCode=P0ACA


That Klark Teknik one looks beautiful. I love that clean industrial look.


----------



## Reticular

@Lunatique, man, i had to register to say thank you for that first post! Thank you!

Why you ask? I have pretty old Sennheiser PX100 headphones, and my monitors (M-Audio BX8) are kinda wanky in the room i am in right now, i did "EQ" of the room, but i dont feel confident. I think i made better mixes with my old Technics Hifi, anyhow...i knew my headpohones were bass heavy, i was trying to reference stuff, but was not satisfied, i needed "something" to hold on so i can get at least 65% of the mix on the right track and make the best of what i have right now. 

I got some graphs from headphone test site, did EQ thru Peace Equalizer yesterday, went thru your track and descriptions, did some moves and ended up feeling pretty confident for doing some referencing.

This really helped me. Cheers


----------



## arar

Hope this is an okay thread to ask this and yall won't mind the minor bump: I have the CA Honeydews as my "main"--or rather, only--IEMs right now. I'm interested in getting the Shozy SCB2s, but they're a bit expensive to blindly buy right now. Crinacle (for example) has the FR graphs for both IEMs. Could I EQ the Honeydews to sound potentially similar to the SCB2s to get an idea how they might sound, based on the graphs alone? If so, what would be the best way to go about it? I suppose Peace does have the EQ graph too so I could just start adjusting things by hand until it looks somewhat similar, but that seems laborious.


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## gregorio

arar said:


> Could I EQ the Honeydews to sound potentially similar to the SCB2s to get an idea how they might sound, based on the graphs alone?



Not sure you’ll get and particularly helpful answers to your question. Unless someone has tried to EQ match those specific IEMs to the SCB2s and accurately measured the results, there’s no way to know. Otherwise, the answer to your question is “possibly”. 

The way to try, is to compare the graph of your IEMs to the graph of the SCB2s and compensate for the differences with EQ. This could get you roughly in the ball park but could just as easily lead you astray. For example the EQ could induce distortion in the music files or your Honeydews that wouldn’t exist with the SCB2s, it would also make volume matching only a rough guess and there could be other issues.

G


----------



## arar

gregorio said:


> Not sure you’ll get and particularly helpful answers to your question. Unless someone has tried to EQ match those specific IEMs to the SCB2s and accurately measured the results, there’s no way to know. Otherwise, the answer to your question is “possibly”.
> 
> The way to try, is to compare the graph of your IEMs to the graph of the SCB2s and compensate for the differences with EQ. This could get you roughly in the ball park but could just as easily lead you astray. For example the EQ could induce distortion in the music files or your Honeydews that wouldn’t exist with the SCB2s, it would also make volume matching only a rough guess and there could be other issues.
> 
> G



Gotcha, I figured that's likely the case. Thanks! I might try messing around with it for a bit, but I'm not gonna make any assumptions on the SCB2s based on it.


----------



## bigshot

When you say, "Could I EQ the Honeydews to sound potentially similar to the SCB2s to get an idea how they might sound, based on the graphs alone?" I'm assuming you're talking about how they sound to your ears in a general sense, not precise measured matching. My guess is that yes, you could get one set of IEMs to sound very much like another set of IEMs as long as the response of the set you're eqing has sufficient bass extension. You would want to use response graphs from the same source to avoid differences in correction curves that might have been used. That should give you a general idea of what the difference in response sounds like to you. It never hurts to experiment and try.

I doubt you could make IEMs sound like headphones that way though.


----------



## Joe Bloggs

arar said:


> Hope this is an okay thread to ask this and yall won't mind the minor bump: I have the CA Honeydews as my "main"--or rather, only--IEMs right now. I'm interested in getting the Shozy SCB2s, but they're a bit expensive to blindly buy right now. Crinacle (for example) has the FR graphs for both IEMs. Could I EQ the Honeydews to sound potentially similar to the SCB2s to get an idea how they might sound, based on the graphs alone? If so, what would be the best way to go about it? I suppose Peace does have the EQ graph too so I could just start adjusting things by hand until it looks somewhat similar, but that seems laborious.


If you are looking at two sets listed on autoEQ from the same agent e.g. crinacle, easy enough to load the two sets of EQ data onto excel and subtract the EQ for the SCB2 from that for the Honeydew and use that as your emulator EQ


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## SoNic67 (Feb 14, 2022)

I am very puzzled about "Harman target curve" - to me that sounds really bad and I am not sure why is so popular.
I have applied compensation curves with AutoEQ and various results from here: *https://github.com/jaakkopasanen/AutoEq/tree/master/results*
I have used Grado SR60i, AKG K701, HIFIMAN Deva Pro for those tests. The results were strange with all of them, for me mids were too strong and I was bothered especially by the high-end loss.
I know about the history, but I am not convinced of the "science" behind it. Mainly because is based on "a panel of 10 trained listeners" that were not specified.
Education and genre preference has a lot to do with how music is perceived by the brain. All those "listeners" seemed to prefer the female voice and really hated the cymbals/high hat.
To me that's not good at all...

Am I alone in this?


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## bigshot

The first Harman is very close to perfect for me.


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## SoNic67

I am curious... Would it be possible to know what's your age (as a range if you want)? Maybe that makes the difference?


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## castleofargh

SoNic67 said:


> I am very puzzled about "Harman target curve" - to me that sounds really bad and I am not sure why is so popular.
> I have applied compensation curves with AutoEQ and various results from here: *https://github.com/jaakkopasanen/AutoEq/tree/master/results*
> I have used Grado SR60i, AKG K701, HIFIMAN Deva Pro for those tests. The results were strange with all of them, for me mids were too strong and I was bothered especially by the high-end loss.
> I know about the history, but I am not convinced of the "science" behind it. Mainly because is based on "a panel of 10 trained listeners" that were not specified.
> ...


1/ The Harman curve for headphones was the result of a dozen different papers over several years, not 10 trained listeners(which were their own people). the link to his blog shows the beginning of their work on headphones, just looking at more recent articles on the same blog would clearly show that. They tested for ages, on different continents, they gave people some simplified bass and treble EQ to play with, they tested some objective hypothesis, they tried to simulate one headphone on another one to remove the risk of people recognizing the headphone or being biased in some way by the comfort or weight... I think the first target came out from 250 listeners, and the all series of experiments probably goes beyond a thousand listeners.
It's by far the most involved and solid series of studies on a preferred headphone FR that were publicly(at least for a time) available.

2/ The result is a statistical one. Nowhere did they claim that it would or should please you or me in particular. When cool dude Olive discussed one of their headphone following the target pretty tightly, he had stats on how many would prefer it over something else, and that was 64%. It leaves plenty of room for you and your personal taste. 

3/ The curve is from a specific measurement rig(that basically nobody posting graphs on this forum is using), and Autoeq is an aggregate of graphs that Jaako collected online. So it wouldn't be at all surprising to find out deviations from target of varying amplitudes on most compensations. It is also going to be the case for most reviewers showing the Harman curve or using it to compensate their graphs. They may, or may not use the right curve for their system. Even then, the pairs of headphones you're using probably has a few dB here and there that differ from the pair measured(+pad wear+...).

4/ Listening level(cf. equal loudness contour), age and hearing damage.

5/ Just personal taste from a lifetime of more or less conventional experiences. 




SoNic67 said:


> I am curious... Would it be possible to know what's your age (as a range if you want)? Maybe that makes the difference?



I'm 45 and I approve this message.. I mean, my preferred response is pretty much right between DF and Harman, so both were a very good starting target for me that I would just fine-tune with EQ. The amount of bass I like, changes greatly depending on the type of headphone and I'm guessing the quality of the seal on *my* head(Something clearly mentioned by Olive). I have some intuition that the wider the driver, the less I feel the urge to get boosted low freqs(for whatever reason).
All in all, I guess you can interpret my case as validation that the Harman curve is pretty good for me, and also that it isn't. I'm not difficult.


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## SoNic67 (Feb 14, 2022)

I am of the same age and like the highs/treble un-corrected. I always liked the treble either slightly boosted or on zero, since I was a teenager listening to LPs and reel-to-reel.
I can't stand the "Harman" curve because it takes a dive after 10-13 kHz, especially when compared with the boosted mids.
If the recording engineers EQ that program in one way, why do we find that "people" prefer a different EQ?
I get the bass-boost, if the headphones have an attenuation in that region, especially open-back versus closed-back. But not liking the treble? 
Is that because they listen mostly to compressed streaming formats (that usually cut that region) and when they hear those treble, they find them "un-natural"?

It's kind of mind boggling to me.


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## sander99

SoNic67 said:


> If the recording engineers EQ that program in one way, why do we find that "people" prefer a different EQ?


Recording engineers mix music for speakers. Because there is no one unique objective flat headphone curve for everyone they can not really do anything else.


sander99 said:


> Headphones skip a part of your HRTF, loudspeakers don't, and sounds in the real world coming from a distance don't. That's why there exists no objective neutral for headphones, but objective neutral for loudspeakers does exist (although the latter depends on the room as well as the loudspeaker).





sander99 said:


> If you were talking about any other audio component, even loudspeakers, but not headphones then everything you say sounds reasonable.
> Only with headphones there is one complication: there is no objective flat or neutral for everyone.
> If you are listening to sounds coming from your environment, from a certain distance, for example to the drum you mentioned then the sound is subjected to your personal head related transfer function (hrtf) filtering. (The sound is bouncing off your torso, bending round your head, bouncing of your pinnae, bending into your ear canal, etc.)
> The resulting filtering of the audio differs per person. If you use headphones a part of this filtering is skipped. Because the part that is skipped differs per person one and the same headphone will sound objectively different to different people.


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## SoNic67 (Feb 14, 2022)

sander99 said:


> Recording engineers mix music for speakers.


So my Klipsch reference speakers have less treble than my headphones? Or more mids? That's why I have to apply that strange 20dB curve on headphones (that accentuates mids and lowers the highs)?
Also HTRF is the same for over-ear headphones and speakers, because it's the same ear (not IEM's). As for bouncing on walls and body... that is not considered while mastering, don't they use they use near-field monitors?

Makes no sense to me. Anyway, what do I know?
I just tested the Harman target curves and, to me, they sound bad.


----------



## sander99

SoNic67 said:


> Also HTRF is the same for over-ear headphones and speakers


HRTF is different for different directions. And that by the way gives the brain important information about the direction of sound.
For most people blowing highs straight into the ears from the side will result in more highs reaching the eardrums compared to highs coming from 30 degrees left or right from center (dead ahead).
Sound bouncing of the walls is part of room acoustics, not HRTF.
Sound bouncing of your own head is part of your HRTF.
Sound engineers by the way anticipate a bass bump in the consumer's playback system that often is there due to room acoustics, they use a monitor system with a bass bump.


----------



## hakunamakaka

Curious how they come up with 64% of people preferring harman curve. It can be very dependant on the music genre that person listens. Matching different set of headphones is very limited and in my view is only possible with the ones that sounds similar to begin with


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## ST33L

Great thread! Thanks


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## SoNic67 (Feb 14, 2022)

I am listening to more of classic/prog rock and pop music. I can't imagine how people can enjoy the cymbals muted, it's like something is missing.
The HRTF argument, IMHO, is just that - an argument.
I don't hear more highs from my headphones than from my Klipsch R-51M speakers. Or should those speakers be muted on treble part too?
My theory is that people's brains get used to certain equipment response curve and then, when presented with something better, they feel that they need to bring it down to their brain comfort level, where they are familiar with. I guess for me, that "average" of opinions doesn't cut it. It would be interesting to know what was their home equipment and music sources.

PS: My 2 cents, not a scientist. Listening to Deep Purple's "Smoke On The Water" right now. "Child In Time" is next". FLAC.


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## bigshot (Feb 14, 2022)

SoNic67 said:


> I am curious... Would it be possible to know what's your age (as a range if you want)? Maybe that makes the difference?



I'm 62, but that wouldn't explain your description of what Harman sounded like to you. I can hear mids as well as you can, and the balance in the mids is what makes Harman so good for me. That's the most important part of the frequency range to get right.

It could be that the music you're listening to has recessed high end and needs boosting. Analog tape back in the 70s rolled off at the top end and LPs rolled it off further. Try listening to a few good DDD recordings and see if it is still a problem.


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## Lunatique (Feb 14, 2022)

Personally, I find the Harman Curve a bit too bright and can get uncomfortable in the upper-mids. This is why I only use it as a basic starting point, but then EQ according to how loud each frequency band sounds, which then tailors the curve to my physiology and hearing. This is the best approach IMO, because we all have different hearing. For example, you might be more sensitive to, let's say, 8 KHz, while I might have a bit of hearing loss in that range, so we should EQ that region according to how loud we hear in that frequency band, and we'll both end up with the most accurate/neutral sounding result for us individually, although they would be different EQ curves.


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## SoNic67 (Feb 14, 2022)

bigshot said:


> I can hear mids as well as you can, and the balance in the mids is what makes Harman so good for me.


I get the correction of mids, if the headphones are not that linear in that portion. But the deliberate boost of mids is not so good. I can hear well enough without that hump.


bigshot said:


> Try listening to a few good DDD recordings and see if it is still a problem.


It's not that, and even if it was, why is the Harman attenuates that portion even more, at -10dB???
I am using Foobar2000 and it's spectrum shows the bandwidth. Hard to screen shoot... but here are a couple:


----------



## castleofargh

SoNic67 said:


> I am of the same age and like the highs/treble un-corrected. I always liked the treble either slightly boosted or on zero, since I was a teenager listening to LPs and reel-to-reel.
> I can't stand the "Harman" curve because it takes a dive after 10-13 kHz, especially when compared with the boosted mids.
> If the recording engineers EQ that program in one way, why do we find that "people" prefer a different EQ?
> I get the bass-boost, if the headphones have an attenuation in that region, especially open-back versus closed-back. But not liking the treble?
> ...



I believe the state of things above 10kHz is mostly "mehhh, whatever". First, the body of research has mostly been done on a standard that wasn't accurately simulating human ear above that frequency(not even a statistical averaging of one). Another problem is how high frequencies(short little things), are easily affected by the smallest physical change, from placement of the headphone to ear shape.
Second problem: What is your idea of neutral sound on those graphs? spoiler, a flat line is not it. 
For us random consumers, to get around those problems and stop going for the false intuitive answers that cripple the audiophile hobby, we need to go read a all lot, or to see our headphone(ideally, our own pair) measured on that rig. Then it's much easier to at least correlate the graph and how we feel about it. But that brings me back to my previous post and the variables we don't control or don't know where to find the references used.

The good news is that if you enjoy more treble, you're 100% free to add more.
Just to be sure, @jaakkopasanen did you strictly apply the delta between a graph and the Harman target at all frequencies?





hakunamakaka said:


> Curious how they come up with 64% of people preferring harman curve. It can be very dependant on the music genre that person listens. Matching different set of headphones is very limited and in my view is only possible with the ones that sounds similar to begin with


It's from this paper I think:
Segmentation of Listeners Based on Their Preferred Headphone Sound Quality Profiles​From memory they end up with 64% out of 130 listeners being happy with the target(because it's the one they picked as best within X headphones?), and like 20% picked others but are grouped as basically wanting less bass because that's what the trend shows for the FR of the headphones they preferred. while the rest is said to want more bass(the smaller group but for similar reasons). 
I wouldn't bet money on me and that summary being entirely right, but that's how I seem to remember it.


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## SoNic67 (Feb 14, 2022)

castleofargh said:


> What is your idea of neutral sound on those graphs? spoiler, a flat line is not it.


Why not? I exposed above my considerations: Flat on headphones sounds closer to my speakers than Harman target" corrected.
I don't see the point. So some random people "voted" that this "sounds better" and no we are taking it as gospel? People that for all purposes might have been used to listen to Spotify free, with iPods or Beats?


----------



## sander99

SoNic67 said:


> I don't hear more highs from my headphones than from my Klipsch R-51M speakers. Or should those speakers be muted on treble part too?


You seem to have missed the point: No, those speakers should not be muted in the treble part, the treble is attenuated by bending around your head into your ears (if the speakers are positioned at about 30 degrees or less left and right of center). The headphones blow the highs straight into your ears from the sides, so to compensate the headphone's frequency response rolls off in the highs. At least that is on average and approximately what happens. It is entirely possible that for you as an individual something else is happening, and that the harman curve is not suitable for you.


SoNic67 said:


> I don't hear more highs from my headphones than from my Klipsch R-51M speakers.


Ideally you should hear the same amount. But not everyone can achieve that with the same headphones (not without additional EQ).


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## bigshot (Feb 14, 2022)

I have a really good speaker system, and Oppo PM-1s, which are very close to Harman. The Oppos don't sound like my speaker system (no headphones do) but they have the same general response balance. My only deviation from Harman is a -2dB cut from 2-3kHz. That frequency range is where human hearing is the most sensitive. It also might vary a lot from person to person, I don't know. But try doing a small cut right in there and see if the flinch response goes away.

Klipsch speakers are horn loaded and have a tendency to project upper mids and some of the treble a bit more than most speakers. But I doubt that is affecting your comparison much.

I just thought of something... What measurements of your cans did you use to calibrate to Harman? Could the curve you used be compensated?


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## sander99

SoNic67 said:


> Why not? I exposed above my considerations: Flat on headphones sounds closer to my speakers than Harman target" corrected.


I could interpret the above sentence in various different ways. To make things more clear: Do you have some specific headphones in your mind (which ones?), and do you mean that they sound closer to your speakers without EQ-ing them towards the Harman curve (entirely possible)? And/or do you mean that they measure absolute flat (not relative to some target response curve)(not very likely I think)?


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## hakunamakaka (Feb 14, 2022)

castleofargh said:


> It's from this paper I think:
> Segmentation of Listeners Based on Their Preferred Headphone Sound Quality Profiles​From memory they end up with 64% out of 130 listeners being happy with the target(because it's the one they picked as best within X headphones?), and like 20% picked others but are grouped as basically wanting less bass because that's what the trend shows for the FR of the headphones they preferred. while the rest is said to want more bass(the smaller group but for similar reasons).
> I wouldn't bet money on me and that summary being entirely right, but that's how I seem to remember it.



So harman curve was matched against Original headphone tuning or something specifically made for this test ? Either way this is nothing as it is a very small number to make any conclusions from the test. It doesn't even cover huge range of headphones. Taste in music and preferred genres can be very dependent on location and time. To have valuable findings you would need at least 100k or even million of user inputs. It is possible through web poll where users could compare their headphone against the sound matched towards harman curve. You would be surprised that some of the headphones would play in favor for their original tuning than the one matched towards harman curve


----------



## castleofargh

SoNic67 said:


> Why not? I exposed above my considerations: Flat on headphones sounds closer to my speakers than Harman target" corrected.
> I don't see the point. So some random people "voted" that this "sounds better" and no we are taking it as gospel? People that for all purposes might have been used to listen to Spotify free, with iPods or Beats?


Those graphs you posted are clearly a sort of raw measurements based on an ear simulator rig. They are not supposed to be flat for something that feels flat to the ear. I asked that question to you because I was wondering if maybe you got the wrong idea about those graphs as it is not specifically written on them if or what type of measurement it is(although a big bump near 3kHz is usually a solid indication of RAW measurement on an ear simulator).

Maybe read this and see if it helps.
https://www.stereophile.com/content/innerfidelity-headphone-measurements-explained

Pay particular attention to this:






The green line is supposed to illustrate what calibrated speakers in a pretty close to ideal room would look like on that measurement rig. One of the hypothesis from the guys at Harman at the beginning was that listeners would prefer that sound from headphones(kind of like you seem to do), so they took a HD800 and EQed it to give(on the dummy head) the same FR as their flat speakers in their listening room.
That particular listening experiment turned out to sound too bright to listeners. But even then you can see that the black line, which is the initial Harman curve, isn't all that different from the sound of flat speakers. Because of course we wish for a sound that feels pretty neutral. but it doesn't look like a flat line. It isn't supposed to.


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## someguyontheinternet

SoNic67 said:


> So some random people "voted" that this "sounds better" and no we are taking it as gospel? People that for all purposes might have been used to listen to Spotify free, with iPods or Beats?


Harman used groups of people with different experiences and found that rankings were stable across experience groups. The experience of the listener did not seem to have any bearing on the order of the ranking, but did seem to have an effect on how far apart the rankings would be.
For example the random person would give scores between 4 and 8 while the trained listeners would give scores between 2 and 9. However the order of the ranking was stable between experience groups (also nationality and age groups).


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## gregorio

SoNic67 said:


> [1] Also HTRF is the same for over-ear headphones and speakers, because it's the same ear (not IEM's). [2] As for bouncing on walls and body... that is not considered while mastering, don't they use they use near-field monitors?


1. The ear and HRTF is the same but the sound reaching the ears is different. Much of what's reaching the ears when listening to speakers is reflections ("bouncing on walls" and other surfaces) and those reflections have relatively little high freqs, unless the walls, ceilings and floor of your listening environment are made of glass or ceramic tiles. In addition, there's the direction of the sound with the speakers in front of you and the absorption and reflection from your skull and pinnae.
2. That's not correct. Firstly, mid-field monitors are almost always used while mastering, although some mastering engineers may also have near-fields for reference/checking. Secondly, even when using near-fields, you still get reflections, plus the effects of the body and the skull and pinnae.


SoNic67 said:


> [1] Makes no sense to me. Anyway, what do I know?
> [2] I just tested the Harman target curves and, to me, they sound bad.


1. That's because you don't know or don't understand the science/reliable evidence.
2. There's three possible reasons for that:
A. You haven't applied the target curves correctly.
B. You fall outside the probability distribution for whom the Harman target applies. EG. You are not among the 64%.
C. You have little/no access to a reliable reference and value a type of sound signature that's actually relatively poor. Therefore "good" is incorrectly assumed (and subjectively perceived) as "bad" and vice versa. This is particularly common in the audiophile world, because of all the misleading marketing audiophiles are subjected to, which affects their sound quality/value judgements and perception.


SoNic67 said:


> [1] My theory is that people's brains get used to certain equipment response curve and then, when presented with something better, they feel that they need to bring it down to their brain comfort level, where they are familiar with. [2] I guess for me, that "average" of opinions doesn't cut it. [3] It would be interesting to know what was their home equipment and music sources.


1. Sure and there's science/reliable evidence to support that theory. In fact, it's quite common that audiophiles exposed to a much more accurate system (say a commercial studio for example) will criticise it, relative to their own, poorer system.
2. Maybe, as per "B" above. Or maybe you are subject to "C" above, and that "average of opinions" would "cut it" if you re-educated/acclimatised to a better reference.
3. It varied and in some cases would not be relevant anyway, because some of the subjects were sound/music engineers accustomed to very high quality listening environments.


SoNic67 said:


> Why not? I exposed above my considerations: Flat on headphones sounds closer to my speakers than Harman target" corrected.


That's very possibly because the Harman target is designed to get closer to the freq response of the sound hitting your ear drums from an accurate system/listening environment, it's NOT designed to get closer to YOUR speakers!


SoNic67 said:


> [1] I don't see the point. So some random people "voted" that this "sounds better" and no we are taking it as gospel? [2] People that for all purposes might have been used to listen to Spotify free, with iPods or Beats?


1. Firstly, no one is taking it "as gospel". We're taking it as (by far) the most reliable evidence that applies to the majority of people (but not necessarily all). Secondly, you don't seem to realise you are contradicting yourself. Presumably because you don't understand what "random" means in the context of a scientific study or group of studies. Studies are commonly deliberately "randomised", so that different demographics are represented. EG. Members of the general public, professionally trained listeners such as engineers and musicians, younger and older people, males and females, etc. ...
2. This is the contradiction: You stated "random people" but then stated that the test subjects "might have been used to Spotify free, with iPods or beats", which isn't random people, it's a specific sub-group! As the test subjects were randomised (over various studies), some of them probably were "used to listen to Spotify free, with iPods or Beats" but certainly not all or even the majority. Others were used to listening to high quality systems and even professionally tuned systems/environments.


hakunamakaka said:


> Curious how they come up with 64% of people preferring harman curve. It can be very dependant on the music genre that person listens.


No it can't, because that 64% was arrived at across a range of different genres.


hakunamakaka said:


> Matching different set of headphones is very limited and in my view is only possible with the ones that sounds similar to begin with


You seem to be missing the basic fact that the Harman target curve is for the HP output, not the input!


hakunamakaka said:


> To have valuable findings you would need at least 100k or even million of user inputs.


That's nonsense. One can have "valuable findings" from a sample size of just 10 or so test subjects. However, one would not necessarily have high confidence that those "findings" applied widely. As sample size increases so does the confidence level but this relationship is NOT linear, there are diminishing returns beyond a certain point. This is a very well studied area in science because huge amounts of money and countless lives are dependant on it: Market research for products, advertising and politics, plus drug/medical research for example. In most cases "valuable findings" with a high level of confidence can be obtained with sample sizes in the few hundred to a few thousand range and even in the most extreme cases, where lives are at stake throughout the globe (phase 3 vaccine and drug studies/trials for example) then roughly 1,000 - 20,000 is sufficient. So where do you get your 100k - 1 million from? Again, just another example of you simply making up whatever supports your agenda, regardless of whether it contradicts the demonstrated/established science!

@SoNic67 looking at the demographics of the Harman curve, you would seem to fall into the 21% sub-group of older (or female) subjects, who demonstrated a perception of significantly lower requirement of bass and therefore inversely, more high bass, mids and treble. In other words, the Harman curve for your demographic is very roughly about 5dB lower than the general curve.

G


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## SoNic67 (Feb 15, 2022)

castleofargh said:


> The green line is supposed to illustrate what calibrated speakers in a pretty close to ideal room would look like on that measurement rig.





someguyontheinternet said:


> Harman used groups of people with different experiences and found that rankings were stable across experience groups.


See, that's what I don't get. IMO that line is not how speakers are "calibrated", but how they said their averaged group preferred to hear them. To me that's not only unbelievable, but when I applied those correction to my headphones, it sounded bad.

As all those people preferred to hear the voce clear and hated the cymbals/high-hat/tambourines, etc... That's probably formed by years of listening to mp3's (and similar lossy encoded) and that formed their brain into thinking that' show music should sound! Music is not only ears and transducers, is also brain.








sander99 said:


> Do you have some specific headphones in your mind (which ones?), and do you mean that they sound closer to your speakers without EQ-ing them towards the Harman curve (entirely possible)?


I have tried the curved on Grado SR 60i, AKG K701, and HIFIMAN Deva Pro.
Compared to what I hear trough my Klipsch RM-51M speakers.

I know that some people say that the horn-loaded AL tweeters on Klipsch sound "too harsh", but then they also probably never listened to a live rock band, to hear how those cymbals should really sound. Again, the same brain bias of what are you used to hear...


gregorio said:


> That's because you don't know or don't understand the science/reliable evidence.


Maybe. But to me this doesn't sound like science, it sounds more like a voting system. 130 people voted, and we are all supposed to conform to the result of that "voted" result. That's not really science, is more like sociology. It's more of how people brains got mangled by listening to decades of poorly compressed streamed music, FM radio and mp3's. Ah, and bad SiriusXM (my case, when I drive). Plus response in the cars are really screwed up - I have tested my car JBL/Harman system frequency response and it had a "bump up" at about 8kHz. People in US get probably used to that kind of response too.

I didn't see any "science" that can explain me those two bumps, besides statistics. The mixing in studio (near or mid field) is not *that* different from home or headphones!
If it was it true, it would imply that, at our hoses, or in our cans, the mids are attenuated and the treble are accentuated compared to studio and the curve reverses that. No way that's true!
PS: I typed above my equipment.


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## someguyontheinternet

SoNic67 said:


> As all those people preferred to hear the voce clear and hated the cymbals/high-hat/tambourines, etc... That's probably formed by years of listening to mp3's (and similar lossy encoded) and that formed their brain into thinking that' show music should sound! Music is not only ears and transducers, is also brain.


I don't think that this is reasonable to assume, because the test groups included audio-salespeople, audio gear reviewers, trained listeners and "normal" people. The experience range was quite varied so this MP3 assumption is not likely to hold.
The research shows that between these groups the ranking order of headphones was consistent. Every group put the same headphone at 1st place, 2nd place, etc down to last place.


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## gregorio

SoNic67 said:


> See, that's what I don't get. IMO that line is not how speakers are "calibrated", but how they said their averaged group preferred to hear them.


That's why you don't get it! You are talking about the speaker output while we (and Harman) are talking about the ears' input, two completely different things! The green line represents the ears' input when the speakers are calibrated flat. Castleofargh has already explained this to you. In other words, that green curve is typically what arrives at the ears and is what the brain perceives as flat.


SoNic67 said:


> [1] To me that's not only unbelievable, [2] but when I applied those correction to my headphones, it sounded bad.


1. The way you seem to understand it, is unbelievable to everyone else too! The problem isn't with the science but of your understanding (misunderstanding) of it.
2. And how did you apply it to your HPs? The amount of correction required would obviously depend on the freq response (output) of your HPs to start with.


SoNic67 said:


> Maybe. But to me this doesn't sound like science, it sounds more like a voting system.


But it is science, so if it doesn't sound like science to you, then clearly you don't know what science should sound like. Admittedly, psycho-acoustics often isn't as clear cut as some other scientific areas but it's still science!


SoNic67 said:


> 123 people voted, and we are all supposed to conform to the result of that "voted" result. That's not really science, is more like sociology.


Firstly, 123 people didn't vote. The Harman curve is the result of multiple papers/studies. Secondly, we are NOT all supposed to conform to that result, only 64% are. Again, this has already been explained to you!


SoNic67 said:


> It's more of how people brains got mangled by listening to decades of poorly compressed streamed music, FM radio and mp3's. Ah, and bad SiriusXM.


Now you're just making up nonsense to support your misunderstanding. Unless of course you've got some reliable evidence to support your assertion? Again, some/many of those tested were professionally trained listeners, whose "brains got mangled" by working for many hours every day of the week with high quality systems/monitoring environments.


SoNic67 said:


> I didn't see any "science" that can explain me those two bumps.


Exactly! There is in fact old, very well established and uncontested science. So clearly, the issue is what YOU "didn't see", and NOT the science itself! For example, the first bump is due to the resonant frequency of the ear canal, this has been known and demonstrated/proven for roughly a century or so.


SoNic67 said:


> The mixing in studio (near or mid field) is not *that* different from home or headphones!


It depends on what you mean by "*that*", there certainly are significant differences. Have you ever spent much time in a high quality commercial studio?


SoNic67 said:


> If it was it would imply that, at our hoses, or in our cans, the mids are attenuated and the treble are accentuated compared to studio. No way that's true!


How does it imply that?

G


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## SoNic67 (Feb 15, 2022)

gregorio said:


> The green line represents the ears' input when the speakers are calibrated flat.


That is not ears input, that's just a voted preference. It means squat, for reasons that I described - people used to FM radio, streaming lossy files...


gregorio said:


> For example, the first bump is due to the resonant frequency of the ear canal, this has been known and demonstrated/proven for roughly a century or so.


The correction curves actually ACCENTUATE that frequency. If it was to correct any resonances, the curve would have to be the other way!
And there is no reason in the curve for the DROP after the 10 kHz.
That's the target that is "desired" by those EQ programs:









T


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## hakunamakaka

gregorio said:


> That's nonsense. One can have "valuable findings" from a sample size of just 10 or so test subjects. However, one would not necessarily have high confidence that those "findings" applied widely. As sample size increases so does the confidence level but this relationship is NOT linear, there are diminishing returns beyond a certain point. This is a very well studied area in science because huge amounts of money and countless lives are dependant on it: Market research for products, advertising and politics, plus drug/medical research for example. In most cases "valuable findings" with a high level of confidence can be obtained with sample sizes in the few hundred to a few thousand range and even in the most extreme cases, where lives are at stake throughout the globe (phase 3 vaccine and drug studies/trials for example) then roughly 1,000 - 20,000 is sufficient. So where do you get your 100k - 1 million from? Again, just another example of you simply making up whatever supports your agenda, regardless of whether it contradicts the demonstrated/established science!
> 
> 
> 
> G



Now you comparing against drug tests which has general use case, but human brain works differently than organs.

When peoples taste and preferences are involved you actually need way larger amount of inputs. Check how IMDB's movie scores fluctuates when they have small amount of voters. Plus you need to multiply it against all various types of headphones and music genres available, because conditions will change once you swap gear and genre. You need to be very specific with an outcome too.

The only route to find the most enjoyable sound is individually through trial & error. There isn't any test or machine available which would be able to tell you what fits you best


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## SoNic67 (Feb 15, 2022)

What I don't get is why are we accepting that albums mastered in studio have to be rendered on our systems with *higher mids and lower treble*.
To me, the excuses that our "home" equipment is not adequate (really, are we even going there?), or that our ear channels are worse than the ones of audio engineers that mastered that music (mine have "resonances", but theirs don't), are just not good enough.

Just because a bunch of random people decided that's how they like their music. And we call that a "standard".


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## gregorio

SoNic67 said:


> I know that some people say that the horn-loaded AL tweeters on Klipsch sound "too harsh",


Missed this previously. I actually heard Klipsch "Reference" speakers not that long ago and thought they were pretty good for home/consumer speakers, although I agree they're maybe a little too harsh. Horn tweeters make perfect sense for far-field monitors/speakers, not so much for home use though.


SoNic67 said:


> but then they also probably never listened to a live rock band, to hear how those cymbals should really sound.


How much time have you spent listening to a live rock band, both in a studio and on stage, before recording/amplification/mixing and after? How much do you know about "_how those cymbals should really sound_"? If you're going to attempt an "appeal to authority" then at least make sure you have more than those you're arguing with. However, as an "appeal to authority" is a fallacy anyway, it's obviously not acceptable as an argument to start with!


SoNic67 said:


> Again, the same brain bias of what are you used to hear...


And therefore, also the bias of what you are used to hearing, which according to you are speakers, that are somewhat harsh!


SoNic67 said:


> That is not ears input, that's just a voted preference.


No it's not! Again, the green line represents the in ear (dummy head) measurements of flat speakers in an acoustically treated room. The black line represents the preferred HP output, that closely correlates with the flat speakers in ear measurements. How many times?


SoNic67 said:


> What I don't get is why are we accepting that albums mastered in studio have to be rendered on our systems with *higher mids and lower treble*.


No one else gets that either and the reason is because no one apart from you is suggesting that! The Haman curve is for headphone (only) output in order to perceptually have the SAME bass, mids and treble as would be measured in ear in a studio (treated acoustic room). This is acheved by having slightly lower mids (not as you falsely state higher mids) and treble. You see that the black line is lower than the green line in the mids right?


SoNic67 said:


> To me, the excuses that our "home" equipment is not adequate (really, are we even going there?), or that our ear channels are worse than the ones of audio engineers that mastered that music (mine have "resonances", but theirs don't), are just not good enough.


What excuses, the one's you've just made up and are arguing with yourself about?


SoNic67 said:


> Just because a bunch of random people decided that's how they like their music. And we call that a "standard".


Yes, just because a large bunch of randomised people under controlled conditions reported their preference/subjective observations. And "yes" that's exactly how we define many audio standards, even some internationally agreed and legally binding ones!

G


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## SoNic67 (Feb 15, 2022)

gregorio said:


> If you're going to attempt an "appeal to authority" then at least make sure you have more than those you're arguing with.


LOL, appeal to authority. Let me try that for real then:
So. based on your arguments, the mixing engineers have no clue what they are doing, those treble need to be squashed by -10dB and the mids have to be elevated +10dB.
Gotcha.


gregorio said:


> And "yes" that's exactly how we define many audio standards, even some internationally agreed and legally binding ones!


Should we include Flat Earth Society in voting if Earth is round?


gregorio said:


> The Haman curve is for headphone (only) output in order to perceptually have the SAME bass, mids and treble as would be measured in ear in a studio (treated acoustic room).


They were not result of any measurements. It's just a "preference" curve.
And the studio engineers don't mix for treated rooms, they know their job. 
Also... I thought this was about headphones? Hos the treatment of a room would affect them?


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## gregorio

SoNic67 said:


> LOL, appeal to authority. Let me try that for real then:


Nice try, again. But unfortunately you've got it badly wrong, you didn't listen to the fact that an appeal to authority is a fallacy and is therefore not acceptable here AND you didn't listen to the advice to make sure that you actually are the authority, because ...


SoNic67 said:


> So. based on your arguments, the mixing engineers have no clue what they are doing, those treble need to be squashed by -10dB and the mids have to be elevated +10dB.


Err, I am a mix engineer, that's how I've earned my living for nearly 30 years. I do know what I'm doing and I don't need to squash the treble by 10dB or boost the mids by 10dB because room reflections, my body, skull, pinnae and ear canals already do that. Again, you don't seem to understand even the basic concept of the Harman curve or anything about psycho-acoustics.


SoNic67 said:


> Should we include Flat Earth Society in voting if Earth is round?


Sure, if you believe that the earth being round is just an aural perception? The rest of us rational people do not believe that though, because of the science/reliable evidence!


SoNic67 said:


> They were not result of any measurements. It's just a "preference" curve.


Wrong, have you even read what the Harman curve is and what the green line represents?


SoNic67 said:


> And the studio engineers don't mix for treated rooms, they know their job.


Yes, I obviously do know my job, what do you know about my job?


SoNic67 said:


> Also... I thought this was about headphones? Hos the treatment of a room would affect them?


Huh, that's what the Harman headphone curve is based on. Commercial music/audio is created in treated rooms, that's what recording and mastering studios are, didn't you know that?

G


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## castleofargh

SoNic67 said:


> See, that's what I don't get. IMO that line is not how speakers are "calibrated", but how they said their averaged group preferred to hear them. To me that's not only unbelievable, but when I applied those correction to my headphones, it sounded bad.


I don't know what to say to this. How is your opinion relevant about what that graph actually is? Right below the graph in the article from Tyll(I linked and suggested you read), you can find:


> Green dashed line is the ear drum response of a speaker that measures flat in the room. Black line is the subjectively derived preferred ear drum response for headphones.


As annotation to go with the graph.

And below he goes again:


> The graph above shows the ear drum response as measured on a dummy head at the normal listening position between a pair of speakers. The green dashed line shows the ear drum response for a speaker that has been equalized flat at the listening position. The black line shows the adjustment away from flat while wearing headphones that most people chose as more pleasing.



Clearly the green line has nothing to do with group preferences, the black line does(although it was only an early version of the curve).  The paper discussing that particular experiment to setup a listening test with various targets, DF,FF, the default FR of the headphone, and that flat speaker calibration being neither free of reverb nor overly reverberant(that would then measure as the green line when applied to the headphone) is called:
Listener preference for different headphone target response curves​I can't find it in open access. But the test itself just concluded that people preferred the frequency response like that of the flat speakers over other older standards. And I don't know if it was in this paper or later, they seemed to find it too bright. better than the other options, but too bright. That's only because of that realization that the black line came to be a thing.

So far you've misinterpreted and probably badly tested a bunch of things, I hope that this can somehow point you in the right direction. Because regardless of enjoying the Harman curve or not, which is nothing more than personal taste, if you don't understand the data at all, how can you pass any sort of judgement about it?


----------



## bigshot

We've created a new logical fallacy, "reducto ad argumentum". I'm not going to dive into the back and forth because it doesn't look profitable, but I will toss in a question...



SoNic67 said:


> See, that's what I don't get. IMO that line is not how speakers are "calibrated", but how they said their averaged group preferred to hear them. To me that's not only unbelievable, but when I applied those correction to my headphones, it sounded bad.



Did you subtract the response curve of your own headphones before you added the Harman curve, or did you just add the Harman curve without any correction?

Headphones don't generally have a flat response. Every manufacturer has their own target curve that they apply to their cans. If you add the Harman curve on top of that, you'll double your response curves and come up with weird results.

Likewise, not every measurement of headphones published on the internet is the same. Some are compensated as Castle points out in 288 and some aren't. So it's possible that the subtraction you need to do before calibrating to Harman might be off.

The third option is just that your personal physiognomy is an outlier from the Harman Curve. Not everyone thought the Harman target sounded best, just most of them did. You could be among the rest of the sample that didn't like the Harman curve.

Personally, I think Harman is a good place to start. It works well as a baseline. Then EQ in small corrections at a time over a long period of time to try to improve it for your own ears. There is no "one size fits all" response curve with headphones like there is for speakers. There's just a "one size fits most". You have to experiment and learn what works best for you.


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## SoNic67

Personally, I look at the Harman curve above and it screams at me "FM radio", or "mp3's at 120kBps".

I guess some people like to hear only part of the music... oh, well. More power to them!


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## bigshot (Feb 16, 2022)

I have no idea what you’re talking about. Harman is just a response curve. It doesn’t involve dynamic compression nor does it introduce compression artifacts. I think you’re just expressing bias here, not information.

This isn’t a football game where we cheer on a team. The goal here is to optimize sound and Harman is one tool for doing that. It’s effective for most people.


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## SoNic67

Maybe my allegory was lost to you. OK, my bad.
Again, you don't address the actual complaint - why are the mids raised and treble squashed? There is not reason why a studio mastered program to be destroyed like that.


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## bigshot

You can calibrate speakers in a room and it will sound correct for everyone, but headphones are different. The sound doesn't inhabit a space. It's directed straight into your ear canal. Harman may sound like that to you, but the balance may be right for someone who has a different shaped head and inner ear. Harman isn't "bad". In fact it sounds good to most people. But for about a third of people, it isn't right. You fit in that category it seems.


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## SoNic67

Over the ear headphones always sounded like the speakers to me. This is not about IEM coupling.
And even the Harman curves for speakers confirm that - they have the same profile.

This argument that we need to raise the mids +10dB and lower the treble 10dB because "ears are different" is not right.

Harman curve it's just a preference, and a bad one for that. That's because it implies that the mastering engineers make unpleasant sounding records...


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## bigshot (Feb 16, 2022)

I don’t think you understand what I’m saying.


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## castleofargh

SoNic67 said:


> Maybe my allegory was lost to you. OK, my bad.
> Again, you don't address the actual complaint - why are the mids raised and treble squashed? There is not reason why a studio mastered program to be destroyed like that.








Those graphs are like a language you don't know how to read. At this point I'm convinced it has absolutely nothing to do with Harman, or EQ. You just have no idea how headphones are measured and decided to misinterpret the hell out of some RAW graphs.

Did you never see RAW measurements of headphones you owned?


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## SoNic67

Please show me the RAW graphs that have a -10dB down dip at mids that needs to be corrected by that +10dB hump.


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## bigshot (Feb 16, 2022)

You would understand better if you listened better. Did you follow the instructions in the first post in this thread when you calibrated to the Harman curve? I'm sensing that you don't have the knowledge to be able to actually do that. I think you applied the Harman curve on top of your headphones' built in manufacturer curve. You essentially doubled the Harman bump in the midrange because your cans were already corrected for that. And that huge boost in the upper mids masked all the treble. (I know you won't understand what I'm saying here, but other people might.)

And of course as Castle has been trying to explain to you... Flat response is not reflected on raw measurements as a flat line.


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## gregorio

SoNic67 said:


> I guess some people like to hear only part of the music... oh, well. More power to them!


You mean more power to you. You're the one who seems to have a serious hearing problem because:


SoNic67 said:


> Over the ear headphones always sounded like the speakers to me.


Well they don't to the vast majority of people. Humans do not have acoustically transparent skulls, are almost always able to easily differentiate high levels of room reverberation from no reverberation and can easily tell the difference between headphones and speakers.


SoNic67 said:


> This argument that we need to raise the mids +10dB and lower the treble 10dB because "ears are different" is not right.


What argument that we need to raise the mids by 10dB and lower the treble by 10dB? No one apart from you has suggested that!


SoNic67 said:


> Please show me the RAW graphs that have a -10dB down dip at mids that needs to be corrected by that +10dB hump.


What correction with +10dB hump? You don't seem able to grasp the simple (and obvious) fact that the Harman Target Curve is a TARGET curve, it is NOT a correction curve! As an example, here is the raw graph of the AKG 701 without any EQ correction, which you've stated you own:



In this example, you would need to apply a dip to the mids (very roughly about -5dB) in order to match the Harman Target Curve, NOT a +10dB hump. A +10dB hump would take you even further away from the target! 


SoNic67 said:


> Again, you don't address the actual complaint - why are the mids raised and treble squashed?


Your "complaint" has been addressed numerous times - because that is what room acoustics, the human torso, skull, pinnae and ear canal do!


SoNic67 said:


> There is not reason why a studio mastered program to be destroyed like that.


If you don't have a human torso, skull, pinnae and ear canals then it wouldn't be destroyed like that (with speakers) or need to be destroyed like that (with HPs)! But then if you don't have a torso or skull, how are you going to wear your HPs?

G


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## punkedrock

Okay it's easy use a sonynw-zx500 to break it into SQ. This gives it the reprucussionable acoustic properties needed for a driver to go loud once it's ready, you won't need an amplifier. I recommended a SONY DAP. The best current one? Probably the SONYNWZX507.

NOW - onto your second dilemma, the 2.2khz curve is for audio voice. SO - lower that all the way down. And level everything else appropriately; starting with 10,000Hs+ so that the highs are clean without white noise being noticeable and annoying.

NEXT control midbass - then bass. Then guitar, then dial in that chorus if you can't hear any sound on the audio regarding voices. REMEMBER step 1.

After that.... just tune to the instruments your trying to isolate and promote without over extending / under surging them.


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## gregorio

punkedrock said:


> Okay it's easy use a sonynw-zx500 to break it into SQ. This gives it the reprucussionable acoustic properties needed for a driver to go loud once it's ready, you won't need an amplifier. ….



Sorry, I have no idea what you are trying to say. It almost looks like some translation software has messed your post up. 

G


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## gordonli (Jul 27, 2022)

If it could help someone else, I just want to recommend software I have found helpful in equalizing my sets to neutral.

I have been using the NCH Tone Generator to great success (some kind of freemium, Windows, Mac, iOS, Android), or as a completely free alternative the Tolvan Data Tone Generator (Windows).

(Edit: I now only use the Tolvan Data Tone Generator. Its coarse and fine sliders are very useful, as is the quick entering of frequencies in the box below the sliders.)

The NCH Tone generator can output grey/pink/white noise which you can use to first set your volume to your comfortable/max listening SPL. Then you should proceed with its variety of options of generating a sine wave and changing its frequency, the following being the ones I use alternately and with a variety of settings:

log sweep - can set start/end frequencies and sweep length in ms (e.g. sweep slowly or quickly, from 20hz-20khz, or from 20hz-1khz, or problem areas, etc.)
continuous tone - can use page up/down or numpad +/- to step in intervals of semitones, or by any amount of hz you set (e.g. reducing the hz interval when deciding where to center a cut)
you can even play multiple sines of different hz simultaneously (which I have only tried briefly, could be useful to set the balance equally between the different freqs)
I do this on top of a base auto EQ to get me quickly in the ballpark, from 20hz-5khz to a Harman-ish target (e.g. I use an equal loudness compensated Harman target to 65 SPL). My individual resonance peak lies above 5khz around 7khz, so that is where I limit the auto EQ. You can auto EQ higher; you can use any Harman-ish target; this step is flexible because in the end you adjust the frequency balance to what sounds flat to you (or your preference).

Don't feel pressured to neutralize your set in one sitting. Like a rolling pin over dough, gradually over time, with every nudge (and overlaid band if your EQ can afford it) of EQ you can turn the FR into whatever you like.


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## DivineCurrent

gordonli said:


> If it could help someone else, I just want to recommend software I have found helpful in equalizing my sets to neutral.
> 
> I have been using the NCH Tone Generator to great success (some kind of freemium, Windows, Mac, iOS, Android), or as a completely free alternative the Tolvan Data Tone Generator (Windows).
> 
> ...


That's pretty interesting, I'll have to try that generator out.
I've been using SineGen for a few years now for sine sweeps, in order to pinpoint peaks on my headphones and IEMs. https://sinegen.en.lo4d.com/windows 
I basically do it all manually, sweep throughout the whole audible frequency range and check for any obvious elevations or dips. I usually ignore the dips unless they are very severe.
This is especially useful for fixing the timbre on certain headphones such as the Focal Clear, which has narrow peaks in the 6kHz and 10.5 kHz range causing that "metallic" timbre. I know using narrow Q value filters is sometimes frowned upon for whatever reason (I don't really know the scientific explanation why, something to do with affecting the phase which I just can't hear), but it works very well for me. In particular, I have the Dunu Zen Pro which has a few sharp narrow peaks from 7-12 kHz, most are about 6 dB higher than the rest of the frequency range. Using narrow EQ filters smooths this IEM out very nicely, allowing the excellent technical performance of the Dunu Eclipse style driver to shine through.


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## gordonli

@DivineCurrent Nice recommendation. Maybe its just me but these programs can be hard to find, tucked away in dark internet corners


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