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How do you want your treble roll-off served?


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9 hours ago, Archimago said:

 

Thanks for the great post and link to the article! Will have to try these values next time I create some room correction filters.

 

For audio playback, we obviously need to also consider how the recording was created, especially placement of the mics. What's captured on the recording might already have applied much of that high frequency roll-off as opposed to a close mic'ed instrument where to sound more "real", we would need to implement more roll-off to simulate the larger space in a concert hall.

 

As such, maybe if we are to implement only one room-correction target frequency response, the B&K, Katz, etc. less steep roll-off at "only" around -6dB by 20kHz would be a fine compromise.

 

The response in room as opposed to concert hall is not the same. In concert halls, you are listening to 2 seconds reverberation. The so called slope in concert hall is not intentional but due to the limits of high frequencies roll off due to longer distance they travel. 

 

There are are some trade off involved in concert hall sound. Between rolled off highs and long reverberation enveloping listeners, the preference will always be the reverbs. 

 

Having said that, this only applies to full orchestra music. For other types of genre, the rolls off may not be desirable nor the long RT. 

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15 hours ago, Le Concombre Masqué said:

 I might have to fiddle between 2 convolutions filters?

 

That’s what Sean suggested. 

 

“We acknowledge this in the paper and say that the ideal target response is a moving one that will depend on the recordings you listen to.
Read more at https://www.innerfidelity.com/content/acoustic-basis-harman-listener-target-curve#pXTHEts6S5Ebqe56.99”

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1 hour ago, Le Concombre Masqué said:

yes and no. 

yes for there's no standard ; no for the statistical average is on + we want to listen to music, not go nuts, don't we?

As I published above I succeeded at reproducing "JBL Synthesis professional room correction, including subwoofers" thus  matching "the preferred in-room loudspeaker response (that) is a smooth curve from 20 to 20 kHz with about a 9-10 dB downward tilted slope." 

And I'm back being happy that way

 

excellent paper I missed, thank you

 

Agreeed. Sometimes, it doesn’t worth the time getting it too perfect and often you can’t tell the difference in the case of sloping and inclining graph. 

 

My measurements with Harbeth were sloping but with the ESL it was going upwards despite Harbeth’s aluminum dome tweeter sounded brighter to my ears. Measurements did lie. It took me a while to understand the results. 

 

Congrats for getting it right. 

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53 minutes ago, mulberry bush said:

While I wouldn’t recommend this approach for all types of music, the SQ of the aforementioned orchestral strings can be surprisingly close to what you hear in the concert hall - in terms of space, bloom, smoothness, etc.   Large scale choral music also seems to benefit

 

Let me guess... the kitchen is tiled with a lot of reflective surface?

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1 hour ago, Le Concombre Masqué said:

impressive ! but do you still get the acoustic signatures of the recording venues?

 

I have explained ( or at least tried) in other threads with diagrams. 

 

Let me try to explain some of the misconception about what actually contained in the recordings. I am only referring to recording made in a venue such as church with its own natural reverbs.

 

The acoustics signature of the recording venue is the direct sound PLUS a LIMITED ratio of the reflected sound within the critical radius of the microphone. 

 

The room acoustics signature is always captured in much lower ratio than what we hear in live performance because unlike the source which has a definite frontal direction, the acoustics signature of the room arrives from many many many different direction. 

 

The ears in a concert hall distinguish the direct and reflected sound easily due to directional cues as the reflected sound will always be delayed from the direct sound. 

 

In recordings, if you capture the complete sonic image of the 360 degrees venue’s reflected sound, then you must reproduce them in a way that it will also arrive from various directions. This is something a two speaker stereo system couldn’t do as all the sound will be fused and reproduced from the speakers. The sound will be muddy. 

 

Therefore, it is a myth to say a standard stereo recordings will contain all the information about the acoustics signature of the recording hall. 

 

It will contain the venue’s RT ( much lower ratio than the actual). It will contain the early reflection which will give some clue of the possible size of the venue. 

 

With the virtual concept hall approach, I create the venue to match the recordings so that I have the complete 2 second RT if necessary. 

 

I can play a 3ms balloon pop exactly like this in the video. I alter the the decay of the sound in my room. The sound from the main speakers is still the same 3ms balloon pop. Only the decays by 20 circling speakers reproduce the decay. I don’t believe in room equalizer. 

 

 

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35 minutes ago, Le Concombre Masqué said:

There are all kind of fears :

 

fear of being not man enough if I can't take it flat 20-20K...

fear of being not smart/wealthy enough if can't build and tune a system that I can stand without eQ and then you'd rather pay a power cord 20K $ rather than eQ...

And then there's fear of complexity ; it might shy away some to try what I suggest and I won't blame them for fear of complexity will shy me away from your thing in the first place

 

I didn’t mean to hijack your thread. I was addressing your question.

 

Room EQ or Dirac is the next natural thing if you want to improve your sound. I was into room treatment and EQ about 8 years back before accepting that the limits of stereo. Maybe, because the world class concert hall is just 15 minutes away, I was obsessed getting the most out of sound reproduction to sound like that. That was my reference. 

 

@Ralph Glasgalwho pioneered this concept already perfected stereo. He designed and patented Fisher 400CX which even today can fetch  few thousand dollars. He wasn’t afraid to try something new. 

 

As you said, fear is keeping away many audiophiles to even try 5.1 format which in my opinion is far superior to stereo. 

 

It is not complex. It only starts with two speakers and XTC. Mine is the extreme level which is a miniature copycat of Ralph’s full blown system. 

 

 

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20 minutes ago, pkane2001 said:

 

It doesn’t. It lets you reproduce the effect of the room on the sound without having to be physically in that room. So if you like the way music sounds through your speakers, you may be able to hear a similar effect through your headphones. Or, if you like the sound of a particular music hall, you might be able to reproduce it at home. 

 

Thanks. So this is all about headphones related? 

 

I am familiar with REW EQ correction and DIRAC is an extension to REW correction by employing far more sophisticated algorithm. I am still not clear about transporting a music hall response to your room. What are you really changing with the use of the filter? Is it frequency response or you also add RT?  One might work for headphones but not necessarily with loudspeakers as your end response will always be your room's acoustics signature's influence over the direct sound.

 

I am sorry to OT with this basic questions. I am confused with the purpose of the filter and the reported results.   

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44 minutes ago, pkane2001 said:

 

Well, no it's not just headphones-related, but it is much easier to hear the effect over headphones because your own room does not overlay on top of the 'recorded' room sound.

 

The filter I'm talking about is called an impulse response. It's a measurement of all the differences that the whole system, including the room, produce when subjected to a Dirac pulse. In effect, if properly measured, it contains all the distortions, reverb, reflections, FR errors, timing errors, etc. What's great about measured IR is that it can then be re-applied to any other sound through convolution. This, at least in theory, is enough to reproduce the original system and room effects without having access to either.

 

 

There is practical nil information about the workings of DIRAC. The FAQ provided some info about the process. 

 

DIRAC is a room correction software. This is the same as Audyssey or other room correction DSP. 

 

My confusion starts when when you say you use the IR to correct your headphones listening. What are you correcting in your playback chain when there is no room error with headphones?. If you were to convolute your room IR, then you are altering the original performance. This is similar to using 32 band equalizer and tailor the original sound to your taste but with IR you are listening to an equalized sound which was meant to correct the dips and peaks of the sound in your room.  

 

 

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22 minutes ago, Le Concombre Masqué said:

I read you postulate recordings convey mainly direct sounds and that you have to create virtual walls in order to have your room yield a realistic acoustic, that of a reference concert hall; I wonder how do mix your room and the acoustic of the church in that exemple (especially around 1.30): 

This being, I shamelessly published a measurement of my left channel before eQ and the after is in my signature. Which response curve, in your opinion, is closest to the one used to produce that recording up to its final stage? We know which one is closest to what Bob Katz uses and I guess guys @ MFSL or Bernie Grundman or Kevin Gray are not too far for their stuff, generally, sound great on my system.

 

Thanks for the video. This will be an amazing demo recording for me as I can hear long reverbs. 

 

To answer your question. My approach is different. I do not correct or alter the original sound. I treat the orginal sound as an instrument itself. I only change the venue where this instrument is played. A violin quartet sounds better in chamber, a choir in a large church or a lullaby in a quiet small room for the the whisper like vocal. 

 

In the video sound, I could use any of the correct 3D RT to match the main recordings, I would probably pick 2.2 second RT. I have 600ms  to 3second RT. So while I am facing the front speakers and listening to the orginal sound. I have tiny speakers around me playing 112 channels ( last count) of convoluted sound recreating the concert hall or church ambience. 

 

Having said that, your room response is excellent and I know what you hear. The declining slope was my target. I had bumped response at LF. However, I gave up on measurement after getting the Sound Lab. Due to large dispersion surface in small room I never managed to measure the response accurately. Thankfully, I used to keep recordings of all my changes and comparing both sound I realized there is much more to learn about getting the measurement done correctly. It can be misleading. 

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12 minutes ago, pkane2001 said:

 

Again, not correcting. I'm making headphones play back the sound as it would sound when listening through the speakers in my listening room. There's no reflections, reverb, no timing errors and other room distortions when using headphones. These are added by the IR convolution. This is in no way a more accurate reproduction, just a different sounding one than you'd normally hear from headphones.

 

I probably misunderstood the workings of DIRAC or any room correction for that matter. 

 

The IR is your room acoustics signature. But the moment it goes through the Minidsp processing it work inversely to produce flat (or tilted response ), to produce the desirable effect so that the dip and peak is neutralized. 

 

Imagine you are listening to the white noise. The measurement supposed to be almost flat in a perfect room. If your room is not perfect, it will not be flat. The impulse response of your room would tell were the the frequency need to be boasted or attenuated so that it will be flat. 

 

When you listen with your headphones, the white noise will be flat because there is no room interaction. Therefore, it requires no filter correction to bring it flat. However, if you use the room IR filter to correct the response than the sound that comes from the speaker will have the negative effect of the room acoustics in reverse, i.e, where there supposed to be a dip, you will have a peak and where you supposed to have peak, you will have a dip. 

 

It is you room sound but in the reverse. That is my understanding unless I have been wrong all these years. 

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7 hours ago, STC said:

 

I probably misunderstood the workings of DIRAC or any room correction for that matter. 

 

The IR is your room acoustics signature. But the moment it goes through the Minidsp processing it work inversely to produce flat (or tilted response ), to produce the desirable effect so that the dip and peak is neutralized. 

 

Imagine you are listening to the white noise. The measurement supposed to be almost flat in a perfect room. If your room is not perfect, it will not be flat. The impulse response of your room would tell were the the frequency need to be boasted or attenuated so that it will be flat. 

 

When you listen with your headphones, the white noise will be flat because there is no room interaction. Therefore, it requires no filter correction to bring it flat. However, if you use the room IR filter to correct the response than the sound that comes from the speaker will have the negative effect of the room acoustics in reverse, i.e, where there supposed to be a dip, you will have a peak and where you supposed to have peak, you will have a dip. 

 

It is you room sound but in the reverse. That is my understanding unless I have been wrong all these years. 

 

There is another function of DIRAC which will create the environment. But that works differently and in a way that’s how I use SIR2 or WAVEIR convolution engine. .

 

https://www.dirac.com/sensaround/

 

bit this is a different application.

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