[reter]

I believe the primary method for doing this is explained in the second paragraph of the Headphone Compensation section of the Impulcifer documentation:

That said, I do all my personal EQ tweaks after the BRIR has been generated by Impulcifer by applying EQ directly to the BRIR file using Audacity.

oh wait, do you mean you equalize the 7 .wav measurements of the speakers? by doing so don't you only change the actual room acoustic? in fact impulcifer should process to equalize the room to the harman curve even if you mess with the room measurement i think? maybe i'm wrong

 

[Brandon7s]

oh wait, do you mean you equalize the 7 .wav measurements of the speakers? by doing so don't you only change the actual room acoustic? in fact impulcifer should process to equalize the room to the harman curve even if you mess with the room measurement i think? maybe i'm wrong

The EQ processing I use is after any room correction has been done by Impulcifer. I've tried it both ways by using the AutoEQ-style eq.csv to adjust earlier in the Impulcifer pipeline, but I prefer to adjust the EQ after the BRIR has been completely generated since I can modify it in real time and tweak any imperfections in frequency response that are bothering me live. I don't measure my own room these days anyway; I used to do that in the past with a measurement microphone but I found it a heck of a lot faster to create BRIRs without it and the differences in the results between using a room measurement or not, it just wasn't significant enough for me to bother.

My process usually works like this:
1. Capture measurements with Impulcifer and generate the Hesuvi.wav
2. Copy that file to my hesuvi hrir folder and get some music playing while listening through the BRIR.
3. In Equalizer APO, I add an EQ vst plugin (I use Pro-Q 3) as an effect.
4. Tweak the EQ live while I'm listening to music and then save the EQ preset to my VST plugin.
5. Open the BRIR that I've been listening to with Audacity and then apply the EQ effect all tracks. Or I apply the EQ to just tracks 1, 2, 8, and 9 if I only want to adjust front left/right speakers (as if it were a regular 2.0 stereo setup).

 

[reter]

The EQ processing I use is after any room correction has been done by Impulcifer. I've tried it both ways by using the AutoEQ-style eq.csv to adjust earlier in the Impulcifer pipeline, but I prefer to adjust the EQ after the BRIR has been completely generated since I can modify it in real time and tweak any imperfections in frequency response that are bothering me live. I don't measure my own room these days anyway; I used to do that in the past with a measurement microphone but I found it a heck of a lot faster to create BRIRs without it and the differences in the results between using a room measurement or not, it just wasn't significant enough for me to bother.

My process usually works like this:
1. Capture measurements with Impulcifer and generate the Hesuvi.wav
2. Copy that file to my hesuvi hrir folder and get some music playing while listening through the BRIR.
3. In Equalizer APO, I add an EQ vst plugin (I use Pro-Q 3) as an effect.
4. Tweak the EQ live while I'm listening to music and then save the EQ preset to my VST plugin.
5. Open the BRIR that I've been listening to with Audacity and then apply the EQ effect all tracks. Or I apply the EQ to just tracks 1, 2, 8, and 9 if I only want to adjust front left/right speakers (as if it were a regular 2.0 stereo setup).

say i want to try to adjust with the .csv file, how do i generate my .csv file? do i need AutoEQ?


following your process is not different that using an equalizer directly in eq apo after the hrir? maybe more lightweight for the cpu?

 

[musicreo]

say i want to try to adjust with the .csv file, how do i generate my .csv file? do i need AutoEQ?

I think, you can just change the values in the the csv files.

following your process is not different that using an equalizer directly in eq apo after the hrir? maybe more lightweight for the cpu?

You can use filters with EQ-APO before (for 8 channels) or after (for 2 channels) Hesuvi . I did not notice any relevant cpu usage. If you want to do equalisation to any of the 16 channels of your hrir with EQ-APO you must do the convolution directly with EQ-APO and cannot use Hesuvi. The benchmark of EQ-APO can be used to save your filter settings to your hrir wave file.

 

[Brandon7s]

say i want to try to adjust with the .csv file, how do i generate my .csv file? do i need AutoEQ?

Yes, AutoEQ. It's been several years since I've tried that method so I'm afraid I can't be much more specific.

following your process is not different that using an equalizer directly in eq apo after the hrir? maybe more lightweight for the cpu?

Yes, it's basically identical except that I'm burning the EQ directly into the BRIR wave file once I've tweaked the EQ to taste. This allows me to more easily use the BRIRs on different computers (like my work laptop) without having to worry about processing overhead or installing the EQ software on that machine.

 

[morgin]

Would anyone happen to know a way to be able to use hesuvi with our BRIR’s for ps5 sound. Remote play works but I don’t think it can stream surround sound.

 

[morgin]

This might be of interest for head tracking

 

[reter]

@jaakkopasanen or someone that can help me solving this enigma

i started recording the processed hrir from my old xonar u3 audio card because i like how it sounds, i connected my behringer to the xonar output like this way

After that i started recording, looking in audacity i made sure that the channels were recorded in the right way, then i normalized with -9db then cut and corrected the position of all the channels, in the end i exported in .wav 32 PCM

The problem is that when i head the hrir in hesuvi the 7.1 test is correct but when listening to songs and other stuff the sound is like warped, left oriented or not correct at all, it's weird, it's not centered like the other processed hrir in the hesuvi folder


i attached the .zip with my audacity project with the recorded hrir and the one ready to be exported as .wav just to make sure you can see and explain to me what i'm doing wrong or what i should do to correct the hrir


please help me!

 

[musicreo]

@jaakkopasanen or someone that can help me solving this enigma

i started recording the processed hrir from my old xonar u3 audio card because i like how it sounds, i connected my behringer to the xonar output like this way

What effect are you recording from the xonar u3?

The problem is that when i head the hrir in hesuvi the 7.1 test is correct but when listening to songs and other stuff the sound is like warped, left oriented or not correct at all,

That is strange and does not make sense.

it's weird, it's not centered like the other processed hrir in the hesuvi folder

What other hrir?

i attached the .zip with my audacity project with the recorded hrir and the one ready to be exported as .wav just to make sure you can see and explain to me what i'm doing wrong or what i should do to correct the hrir

The correct channel order for HeSuVi is: [L-l L-r LS-l LS-r LB-l Lb-r C-l R-r R-l RS-r RS-l RB-r RB-l C-r ]. This is what you use in your project. For me the wav file seems to be ok.

 

[reter]

What effect are you recording from the xonar u3?

thanks for the reply, i'm recording Dolby Headphone 2 which isn't in the Hesuvi library, there's a DHT but it's a bit different and with a different reverb, more like a studio



i found the problem, as you can see all the impulse responses weren't timed correctly, you can see the start of the right channel is very delayed compared to the left channel (i'm talking about ms), i cropped all the delays and also removed the all the tails in the end of the tracks to "clean" from noises, here we go, now it's much better!

i attached the processed .wav just in case someone want to take a sneak

but still there's a weird metallic sound, it's more noticeable when i listen to vocals that's very weird, probably i have to redo the measurements or find a way to fix that damn noise the soundcard make that's dirtying my hrir


Question: i found out that metallic sound is indeed caused from the noise coming from the sound card; i recorded in every test with less volume just to avoid clipping but now i think i will record with the max MPC and Windows volume to get the best signal to noise ratio and then normalize afterwards


but still i don't understand why the recording have this much noise... i mean, when i hear directly from the sound card there isn't noise at all, i'm suspecting it has to be related due to my cables being too long

 

[Brandon7s]

thanks for the reply, i found the problem, as you can see all the impulse responses weren't timed correctly, you can see the start of the right channel is very delayed compared to the left channel (i'm talking about ms), i cropped all the delays and also removed the all the tails in the end of the tracks to "clean" from noises, here we go, now it's much better!

i attached the processed .wav just in case someone want to take a sneak

but still there's a weird metallic sound, it's more noticeable when i listen to vocals that's very weird, probably i have to redo the measurements or find a way to fix that damn noise the soundcard make that's dirtying my hrir


Question: i found out that metallic sound is indeed caused from the noise of the sound card, i recorded in every test with less volume just to avoid clipping but i think that i will record with the max volume to get the best signal to noise ratio and then normalize afterwards


but still i don't understand why the recording have this much noise... i mean, when i hear directly from the sound card there isn't noise at all, i'm suspecting it has to be related due to my cables being too long

Could be a ground loop in your signal chain. I have a couple of these things around whenever I run into them and they do a great job of eliminating that specific issue: https://www.amazon.com/Compact-Mini-Hum-Eliminator-Box/dp/B00BARTW42

 

[reter]

Could be a ground loop in your signal chain. I have a couple of these things around whenever I run into them and they do a great job of eliminating that specific issue: https://www.amazon.com/Compact-Mini-Hum-Eliminator-Box/dp/B00BARTW42

thx, how this works? where should i connect it? pardon my ignorant question

 

[Brandon7s]

thx, how this works? where should i connect it? pardon my ignorant question

Oh, no worries, I probably should have mentioned this to begin with, haha.

The hum eliminator would go between your audio interface's input and your u3's output. You'll need a 3.5mm stereo to dual 6.35mm splitter cable and a regular dual 6.35mm (aka, 1/4") cable.

You would plug the 3.5mm stereo male connector of the splitter cable into your u3 and then plug the left 6.35mm male connector from that same cable into the Channel 1-In jack of the hum eliminator, then plug the right male connector into the Channel 2-In jack. Then you would plug the one end of the dual 6.35mm cables into the Channel 1 and Channel 2 Output and then plug the other ends straight into to the front ports of your audio interface. You might have to do an audio test to make sure that your left is in the right channel. If not, just swap the left and right cables that are plugged into your audio interface, of course.

This will isolate the ground connection between that the u3 and your audio interface. If you have more than one path to ground then you get ground loops which are a very annoying hum or buzz. I'm an amateur musician and this kind of thing VERY frequently pops up when I'm connecting multiple hardware synths/pedals/speakers to my computer, since often times the audio cable is the shortest path to ground and that will get used instead of the wiring that is made for that purpose, the power cables.

 

[musicreo]

thanks for the reply, i'm recording Dolby Headphone 2 which isn't in the Hesuvi library, there's a DHT but it's a bit different and with a different reverb, more like a studio

I understand. If 5.1 is enough you could use the DH wrapper from foobar to "record" the brir digitally.

i found the problem, as you can see all the impulse responses weren't timed correctly, you can see the start of the right channel is very delayed compared to the left channel (i'm talking about ms), i cropped all the delays and also removed the all the tails in the end of the tracks to "clean" from noises, here we go, now it's much better!

Cropping by "hand" is difficult. You must enter the correct timing in Audacity. I would use a python script for the cropping and export.

You could try the attached python script with a simple gui.

This file attached is the result (start=1.1s,step=1s, crop=0.5s ). (I have only tested the "normal channel order" version not HeSuVi)

i attached the processed .wav just in case someone want to take a sneak

That sounds different (more bass). Is this a new recording?

but still there's a weird metallic sound, it's more noticeable when i listen to vocals that's very weird, probably i have to redo the measurements or find a way to fix that damn noise the soundcard make that's dirtying my hrir

I don't hear a metallic sound. I also don't hear a problematic noise level.

 

[reter]

I understand. If 5.1 is enough you could use the DH wrapper from foobar to "record" the brir digitally.

Cropping by "hand" is difficult. You must enter the correct timing in Audacity. I would use a python script for the cropping and export.

You could try the attached python script with a simple gui.

This file attached is the result (start=1.1s,step=1s, crop=0.5s ). (I have only tested the "normal channel order" version not HeSuVi)

That sounds different (more bass). Is this a new recording?


I don't hear a metallic sound. I also don't hear a problematic noise level.

thank you, yeah i did a ton of different recordings just to see what would have changed so probably that recording have more bass for that reason

anyway, Yesterday i tried plugging both the behringer and the u3 in an external hub and yeah, that noise lowered a ton, i tried recording but now i'm stuck again to the warp left-right problem, even tho i cropped; @musicreo you're probably right, i can't crop very precise, i will try your python code when i have time, hopefully today i will have time to test your processed .wav too