Caution: Very long post :-)
Well, it seems that my forward progress has taken a step backward, although the step backward has resulted in discoveries that will hopefully be beneficial in the end, to me and to others who may read this.
After evaluating four different speaker correction profiles on numerous recordings, all based on measurements taken with the acoustic panels I purchased placed around the measurement microphone, Ive concluded that even though those measurements, when viewed on impulse response plots over a reasonable time scale (e.g., 30 ms), looked considerably better than the measurements taken with no panels and with the panels placed around the speaker being measured, the panels were doing more harm than good. And I say that even though there were no points on the impulse response plots at which reflection amplitudes appeared to look greater with the panels placed around the mic than in the other plots, and at the great majority of points reflection amplitudes looked significantly smaller with the panels placed around the mic compared to the other two cases.
As will become clear later in this post, what I probably should have done was to place the panels up against the nearest reflective surfaces (the fireplace on the left and the large antique radio/phono on the right, as seen in my system description photos), rather than surrounding the mic with them on three sides, at a fairly close distance.
Ill first say that one of those four profiles (one of the first two I tried, with the impulse response truncation window terminated 7.2 ms after the initial sound arrival, and corrections only performed between 600 Hz and 10 kHz), on most but not all recordings sounded distinctly better than bypass mode and than the other three profiles I tried (all of which had the window terminated about 18 ms after the initial sound arrival, with corrections over a somewhat broader range of frequencies).
However I noted especially on the last two profiles I tried that the image was shifted considerably to the right, even though bypass mode (as always) was perfectly centered. And that right channel boost (the difference being in the area of 2 to 6 db depending on frequency, and occurring primarily between about 200 Hz and 1 kHz) could be clearly seen when the correction profiles were viewed in the DEQX software, with the profiles for the two speakers placed on the same graph and the scale of the vertical axis suitably adjusted.
I also found while doing further experiments with the software that the volume difference between the two channels increased dramatically in proportion to the duration of the truncation window, which really puzzled me at first, and also increased in proportion to the distance of the measurement microphone from the speaker (meaning also that it increased as the distance between the mic and the panels behind it became smaller). And the volume difference was worse for correction profiles created from the measurements made with the panels surrounding the mic than for correction profiles created from the measurements made with the panels surrounding the speaker. And, as mentioned above, I found that the issue occurred primarily (although not exclusively) in the area of 200 Hz to 1 kHz, especially around the middle of that area, with the volume difference varying considerably at different frequencies.
After a lot of study of frequency response plots, impulse response plots, and step response plots, I concluded with a fair amount of certainty that the cause of the different corrections for the two speakers was that I didnt have the panels placed in precisely the same locations when I measured the two speakers. The reason for the slightly different placements being that since I was making measurements with the panels surrounding the speaker as well as with the panels surrounding the mic, and it happened that I did the measurements with the panels surrounding the speaker last, I moved the panels aside when the first speaker being measured was moved away from the center of the room, and the second speaker was moved into that position.
Upon very close examination, the consequences of that can be seen in terms of slight differences between the timing of the wiggles of the impulse response measurements for the two speakers in the area of about 3 ms after the initial sound arrival, and can also be especially seen in the form of a roughly 1 db difference occurring at that same instant between the step response plots of the two speakers, in the cases of the measurements taken at 3 and 3.5 foot distances which I used for the correction profiles.
So, I wondered, if the issue was being caused by reflections from the panels occurring just 3 ms or so after the direct sound arrival, why would the consequences of those reflections in the correction profiles get worse as the truncation window was extended much further out in time, for instance from 7 ms to 18 ms and beyond? Im not totally certain, but I believe the answer to that is inherent in the mathematics of the Fast Fourier Transform, some variation of which I assume is what the software uses to convert between the time domain (impulse and step responses) and the frequency domain (frequency responses).
Now if I were to redo the measurements while making a point of placing the panels at precisely the same locations for both speakers, I could evidently eliminate the inter-channel differences in the corrections. However, the fact that slight differences in panel placement caused dramatic differences in calibration profiles between the speakers would seem to say that even if I were to achieve identical profiles for both speakers, that identical profile would reflect (pun intended) significant adverse effects of the panels. So for that reason, in addition to the effort that would be involved in re-measuring the speakers, Im not planning to do that. Instead Im now planning to simply try some correction profiles that Ill create based on the measurements Ive already taken with no panels in place. If those dont work out well, then Ill consider re-measuring the speakers, with the panels much further from the mic and probably placed against the reflective surfaces I mentioned earlier.
Also, if I were to try to correct the inter-channel differences using the equalization capabilities of the DEQX, given the extensive variations of those differences as a function of frequency I suspect that the effort would be extremely time-consuming, and would probably result in a less than ideal set of complementary colorations.
So although my efforts have had a bit of a setback here, its probably a good thing that I didnt make a point of placing the panels in exactly the same positions for the measurements of the two speakers. If I had done so I probably wouldnt have discovered any of this, and I would very conceivably have ended up deriving less benefit from the DEQX than I hopefully will, eventually. So, undaunted, I shall persevere and carry on. Due to various unrelated upcoming activities, my next significant update will probably be in about a week. Meanwhile, just using the DEQX in either bypass mode or with the one correction profile I mentioned as being superior to the others, provides (despite a bit of channel imbalance in the case of the latter) a modest but notable improvement on what I previously had.
Best regards,
-- Al
Well, it seems that my forward progress has taken a step backward, although the step backward has resulted in discoveries that will hopefully be beneficial in the end, to me and to others who may read this.
After evaluating four different speaker correction profiles on numerous recordings, all based on measurements taken with the acoustic panels I purchased placed around the measurement microphone, Ive concluded that even though those measurements, when viewed on impulse response plots over a reasonable time scale (e.g., 30 ms), looked considerably better than the measurements taken with no panels and with the panels placed around the speaker being measured, the panels were doing more harm than good. And I say that even though there were no points on the impulse response plots at which reflection amplitudes appeared to look greater with the panels placed around the mic than in the other plots, and at the great majority of points reflection amplitudes looked significantly smaller with the panels placed around the mic compared to the other two cases.
As will become clear later in this post, what I probably should have done was to place the panels up against the nearest reflective surfaces (the fireplace on the left and the large antique radio/phono on the right, as seen in my system description photos), rather than surrounding the mic with them on three sides, at a fairly close distance.
Ill first say that one of those four profiles (one of the first two I tried, with the impulse response truncation window terminated 7.2 ms after the initial sound arrival, and corrections only performed between 600 Hz and 10 kHz), on most but not all recordings sounded distinctly better than bypass mode and than the other three profiles I tried (all of which had the window terminated about 18 ms after the initial sound arrival, with corrections over a somewhat broader range of frequencies).
However I noted especially on the last two profiles I tried that the image was shifted considerably to the right, even though bypass mode (as always) was perfectly centered. And that right channel boost (the difference being in the area of 2 to 6 db depending on frequency, and occurring primarily between about 200 Hz and 1 kHz) could be clearly seen when the correction profiles were viewed in the DEQX software, with the profiles for the two speakers placed on the same graph and the scale of the vertical axis suitably adjusted.
I also found while doing further experiments with the software that the volume difference between the two channels increased dramatically in proportion to the duration of the truncation window, which really puzzled me at first, and also increased in proportion to the distance of the measurement microphone from the speaker (meaning also that it increased as the distance between the mic and the panels behind it became smaller). And the volume difference was worse for correction profiles created from the measurements made with the panels surrounding the mic than for correction profiles created from the measurements made with the panels surrounding the speaker. And, as mentioned above, I found that the issue occurred primarily (although not exclusively) in the area of 200 Hz to 1 kHz, especially around the middle of that area, with the volume difference varying considerably at different frequencies.
After a lot of study of frequency response plots, impulse response plots, and step response plots, I concluded with a fair amount of certainty that the cause of the different corrections for the two speakers was that I didnt have the panels placed in precisely the same locations when I measured the two speakers. The reason for the slightly different placements being that since I was making measurements with the panels surrounding the speaker as well as with the panels surrounding the mic, and it happened that I did the measurements with the panels surrounding the speaker last, I moved the panels aside when the first speaker being measured was moved away from the center of the room, and the second speaker was moved into that position.
Upon very close examination, the consequences of that can be seen in terms of slight differences between the timing of the wiggles of the impulse response measurements for the two speakers in the area of about 3 ms after the initial sound arrival, and can also be especially seen in the form of a roughly 1 db difference occurring at that same instant between the step response plots of the two speakers, in the cases of the measurements taken at 3 and 3.5 foot distances which I used for the correction profiles.
So, I wondered, if the issue was being caused by reflections from the panels occurring just 3 ms or so after the direct sound arrival, why would the consequences of those reflections in the correction profiles get worse as the truncation window was extended much further out in time, for instance from 7 ms to 18 ms and beyond? Im not totally certain, but I believe the answer to that is inherent in the mathematics of the Fast Fourier Transform, some variation of which I assume is what the software uses to convert between the time domain (impulse and step responses) and the frequency domain (frequency responses).
Now if I were to redo the measurements while making a point of placing the panels at precisely the same locations for both speakers, I could evidently eliminate the inter-channel differences in the corrections. However, the fact that slight differences in panel placement caused dramatic differences in calibration profiles between the speakers would seem to say that even if I were to achieve identical profiles for both speakers, that identical profile would reflect (pun intended) significant adverse effects of the panels. So for that reason, in addition to the effort that would be involved in re-measuring the speakers, Im not planning to do that. Instead Im now planning to simply try some correction profiles that Ill create based on the measurements Ive already taken with no panels in place. If those dont work out well, then Ill consider re-measuring the speakers, with the panels much further from the mic and probably placed against the reflective surfaces I mentioned earlier.
Also, if I were to try to correct the inter-channel differences using the equalization capabilities of the DEQX, given the extensive variations of those differences as a function of frequency I suspect that the effort would be extremely time-consuming, and would probably result in a less than ideal set of complementary colorations.
So although my efforts have had a bit of a setback here, its probably a good thing that I didnt make a point of placing the panels in exactly the same positions for the measurements of the two speakers. If I had done so I probably wouldnt have discovered any of this, and I would very conceivably have ended up deriving less benefit from the DEQX than I hopefully will, eventually. So, undaunted, I shall persevere and carry on. Due to various unrelated upcoming activities, my next significant update will probably be in about a week. Meanwhile, just using the DEQX in either bypass mode or with the one correction profile I mentioned as being superior to the others, provides (despite a bit of channel imbalance in the case of the latter) a modest but notable improvement on what I previously had.
Best regards,
-- Al