Recording Impulse Responses for Speaker Virtualization

Nov 20, 2024 at 7:35 PM Post #1,966 of 2,034
Good to know, thank you. trying to do a measurement now and i noticed every one of my headphone recordings look like this
Is there anything wrong with my right microphone that its picking up so much static?


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If you look at the recording you recorded yesterday, there is a little noise.
But it's weird that only one channel is like that, so take the microphone out of the input and try it again.
don't know much about electricity. =(
However, the noise before the start of the impulse will be cut in the impulcifier, and most of the self-noise will be recorded outside the audible range.
Maybe the gain of the input is little different?
 
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Nov 20, 2024 at 8:12 PM Post #1,967 of 2,034


If you look at the recording you recorded yesterday, there is a little noise.
But it's weird that only one channel is like that, so take the microphone out of the input and try it again.
don't know much about electricity. =(
However, the noise before the start of the impulse will be cut in the impulcifier, and most of the self-noise will be recorded outside the audible range.
Maybe the gain of the input is little different?
I have done my first 7.1 recording. I have similar bass issues to the stereo recording but this one is a bit better i think? The echo on the reverb isnt great but i think is also a bit better than the original stereo recording. How does this look to you compared to my last few? Thanks again for all the wonderful help!

Also i will say, the 4khz changes made a huge difference that i can hear
 
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Nov 20, 2024 at 8:24 PM Post #1,968 of 2,034
I have done my first 7.1 recording. I have similar bass issues to the stereo recording but this one is a bit better i think? The echo on the reverb isnt great but i think is also a bit better than the original stereo recording. How does this look to you compared to my last few? Thanks again for all the wonderful help!

Also i will say, the 4khz changes made a huge difference that i can hear
Looking at the file.
Did you record only FC, BL,BR?
I can't find SL,SR / FL,FR
 
Nov 21, 2024 at 12:21 AM Post #1,974 of 2,034
I think you should post here so other new users can see trial, error and fixes. Keep trying until you know you won’t get a better result. It’s good to have loads of measurements to be able to compare and see differences when you switch between them. I use my mine nearly everyday and it’s still mind blowing
 
Nov 21, 2024 at 12:42 AM Post #1,975 of 2,034
I think you should post here so other new users can see trial, error and fixes. Keep trying until you know you won’t get a better result. It’s good to have loads of measurements to be able to compare and see differences when you switch between them. I use my mine nearly everyday and it’s still mind blowing
Yes you are right.
I asked separately as PM because I was afraid that the writing that was not related to the measurement would be repeated due to an error in his mega link.
I will organize his files after I finish my day routine.
I'll post some graphs for him and other users to see.
 
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Nov 21, 2024 at 2:56 AM Post #1,976 of 2,034
Sure.
I downloaded the file from the @Kosorm user, organized it, and delivered it to them.
There’s nothing particularly special, but there might be parts that could be helpful for new users, so I’ll leave a record of what I did.

I opened the front-FL,FR.wav file first

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(Right Channel +3.9db , Some EQ)

I noticed an imbalance in the channels, but it seemed to be due to the microphone input gain and slight wearing deviations.
To be more certain, I also checked the headphone file.
As a result, I chose post-correction rather than re-recording.
This isn’t a big deal, but it’s a fairly common case.
When recording or advising someone, I personally recommend testing with a speaker placed in the center (mono) to adjust the insertion depth for both ears and balance the input gain.

Yes, post-correction can fix everything, of course.
However, as such issues accumulate, they can become increasingly cumbersome to deal with.
(For example, what if the left speaker has lower gain, but the right input microphone has lower gain? )

Therefore, performing a few quick tests to check if the responses from both sides are similar, verifying if the resonances are properly aligned, and ensuring that the ultra-high frequencies beyond 4kHz are not lost can usually eliminate most microphone-wearing issues.

(While proper insertion depth is important, sealing is equally critical during plug-pipe measurements. If the seal is incomplete, resonances may diminish, low frequencies may be lost, and high frequencies may also be measured inaccurately.)

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The basic response was not bad.
Since his room had a slight reverb, I applied a limit to the overall reverb, restricting it to around 200ms, and then I synthesized the low frequencies afterward.

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Of course, the Decay parameter function in Impulcifer can stabilize things to some extent, but it cannot eliminate spatial errors.
Additionally, while some issues can be corrected with EQ, others cannot. In such cases, effectively managing bass—such as with a quad-subwoofer setup or a DBA—can be quite expensive.

Therefore, it is more practical to synthesize frequencies below approximately 60Hz to 150Hz, depending on the situation.




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His speakers are decent, but I also created a file with some corrective EQ based on Spinorama data. (500~8000Hz EQ)
Generally, frequencies below 200Hz depend on the room, frequencies between 200–500Hz interact with both the speaker and the room, and frequencies above 500Hz are mostly determined by the speaker’s own response.
Of course, EQ can’t correct reflections (although time-domain EQ can be applied in BRIRs, but that’s not the point here). When applying EQ, the intended results may be mixed with the actual room-compensated sounds, and the resulting perception might vary. However, at least from my listening, it didn’t sound bad.


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His impulse response was decent, but experimentally, I delayed all the reflections from 250Hz to 12kHz by 10ms after 4ms.
This approach has been explored in some research papers, where some users reported a distorted sense of distance and space compared to the original BRIR, while others found improvements.
Generally, if early reflections arrive too quickly, they negatively affect the timbre and fail to contribute to the sense of space and distance.

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Therefore, this is the corrected outcome we can observe.
It’s not always beneficial to apply a lot of delay, as the limitations varied depending on the space.
Even in HRTF scenarios with distances over 2 meters, there are subtle changes, but since the ITD/ILD changes are small, applying delay alone can cause confusion in the brain.

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Looking at the frequency response (FR), you can clearly see that it has become much cleaner.


https://www.head-fi.org/threads/rec...r-speaker-virtualization.890719/post-18402913

It was fine when applying the logic of Dolby Pro Logic IIx for upmixing.
However, to add more interest, I added concert hall reverb to the Side and Rear channels.


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This is how it works.
Of course, applying reverb to all channels would make the sound very muffled.
However, since reverb is only applied to the upmixed, separated information, it’s somewhat different from typical experiences.
 
Nov 26, 2024 at 6:45 AM Post #1,977 of 2,034
I recently wrote a post about upmixing.
I have since revised and added to it, so I'm sharing it again.
This is the code following the Dolby Pro Logic II and IIX logic.
There are some points that were missing or changed from what I posted earlier.
Simply use the file that matches the number of channels you are using.
Place the TXT file above the Hesuvi file.
I’ve tested it with 4, 5, and 7 channels, and it worked without any issues.
If you use the latest version of the VB cable, it should work with height channel mixing for over 7 channels or more, but I haven’t tested that yet, so I haven’t included it.

https://drive.google.com/drive/folders/13MCZNX6NmDS_phPP7HnGSiTQ-gInii_e?usp=drive_link

https://drive.google.com/drive/folders/13MCZNX6NmDS_phPP7HnGSiTQ-gInii_e?usp=drive_link


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I have incorporated the delay differences between Music mode and Movie mode, and also addressed the fact that LS/RS are in-phase or out-of-phase in each mode.
I’ve also verified and included the 90-degree Hilbert phase shift.
Now, it seems to be reasonably well simulated.
Overall, it works more logically than the Pro Logic upmixing codes I’ve seen on GitHub or Google.
 
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Nov 26, 2024 at 8:31 AM Post #1,979 of 2,034
Will this work with voice meter banana I find that sound better for some reason
i don't understand what your say.
I am using a vb cable.
If you're using a voice meter for 7.1 then yes that's right.
For upmix to work, of course, multichannel has to work. If you just listen to it, it just has some effect on 2ch(FL,FR only)

@morgin
and i chagne upmix file today. check link.
 
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