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A convolution based alternative to electrical loudspeaker correction networks
A convolution based alternative to electrical loudspeaker correction networks
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Old 17th October 2019, 10:26 AM   #621
fluid is online now fluid  Australia
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Originally Posted by gmad View Post
Impulse response correction can certainly achieve this, but I think it generally requires a higher degree of resolution which carries an increased risk of artifacts away from the measurement point.
I think that is one of the main reasons I find that impulse response averaging works better when the room is fairly live, you can use a longer window without getting the artefacts. What is left is the speaker and persistent room contribution. A single point measurement used for the basis of my current correction does not sound good to me.

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Originally Posted by gmad View Post
I do hope to get some feedback on this..
Why do you think a single point measurement is needed for your new correction?

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Originally Posted by Spartacus View Post
Are you referring to the number of cycles of correction, and if so how many do you think are required for an improved impulse response? Do the number of taps used have an effect?
The number of taps sets the overall length of filter (in time) and the resolution of the filter at low frequencies. A 65,536 tap filter is usually enough with a 44K sample rate. A lower tap count will make the correction less fine grained at low frequencies, which may or may not matter depending on the window lengths and other parameters. If the filter is 1 second long and the impulse is centred you can correct up to a 500ms window of time. So the two are interlinked. The downside to a longer filter is latency. You could also have a filter with lots of taps for high frequency resolution but low latency if you changed the impulse centre and used minimum phase filtering.
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Old 17th October 2019, 07:19 PM   #622
gmad is offline gmad  United States
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Hi, Spartacus:

When I used the term "resolution" I was referring to how smooth or detailed the response of the correction filter itself is. And yes, a more detailed response in the frequency domain means more resonance or time domain ringing - hence more freq cycles/periods the filter is potentially correcting over. Since impulse response inversion treats peaks and dips equally, achieving a flat freq response envelope will result in a more detailed filter than psychoacoustic based EQ.

BTW, I don't mean to imply that I feel the goal for everyone should be a flat freq response envelope or that psychoacoustic EQ is automatically superior to impulse response inversion. Response dips, while they may not "color" the timbre of a sound to the degree that peaks will, may still carry information about location, and at the very least can skew phase response. If someone has a highly unsymmetrical speaker/room setup, and is mainly concerned about the response at the sweet spot, than they may be better off with impulse response inversion. I haven't experimented with such a situation personally.

As for a sufficient resolution for impulse response cleanup, I don't personally feel that going beyond ~4.3 cycles or so (1/6 oct frequency resolution) is necessary (this would mean setting lower/upper window lengths to 19000/19 samples). For a minimum phase filter, this would require at least 8192 taps or so (for a tight squeeze and a slight loss of low freq resolution), and should be plenty for enjoying music; one could certainly push things further with the goal of having an out of body experience .

Last edited by gmad; 17th October 2019 at 07:22 PM.
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Old 17th October 2019, 07:46 PM   #623
gmad is offline gmad  United States
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Hi, fluid:

I don't think a single point measurement is absolutely needed for my config, I just don't think a multipoint avg is neccessary. The prefiltering removes a significant portion of the room's contribution, and the EQ stage ignores a significant part of what remains.

My goal is to optimize for the main listening position while minimizing filter resonance that might be noticed from other locations. I also want this process to be simple and repeatable. I'm ok with compensating for a problem that only occurs at the sweetspot as long as the compensation doesn't make things sound too bad elswhere in the room. Averaging requires more work, and may introduce resonances to the measurement that are foreign to the sweet spot.

I probably don't have as much experience with averaged measurements as you (I have tried it), however, I am able to (quasi) anechoically correct my speakers, and I can say that my current method gives me a subjectively better result with a similar overall level of filter resolution.
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