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What is your favorite house curve

The standard only covers 50-16khz. It doesn't stipulate that you have to have a brick filter at 50hz and 16khz. :)
Honestly? Personally I even despise rumble filters (highpasses) on phono preamps... ain't nobody fuck with that glorious 30Hz bass the artist and cutting engineer worked so hard for getting on that age old format at full volume.

If your speakers can't tolerate vinyl rumble and pops and everything at high volume and occasional earthquakes, then frankly you need to get better speakers. Until today it's a mystery to me how such things can be described as "dangerous for your system" in any way. :p
 
Honestly? Personally I even despise rumble filters (highpasses) on phono preamps... ain't nobody fuck with that glorious 30Hz bass the artist and cutting engineer worked so hard for getting on that age old format at full volume.

If your speakers can't tolerate vinyl rumble and pops and everything at high volume and occasional earthquakes, then frankly you need to get better speakers. Until today it's a mystery to me how such things can be described as "dangerous for your system" in any way. :p
IMD.... if your woofer is reproducing distortion (rumble) - then it will interact with the audio.... (well strictly speaking anything will interact... but at least you can cut out the undesirable irrelevant bits!) - for old style vinyl, there was filtering done during mastering, so there should be nothing down in the sub bass in any case - so a brick wall filter makes perfect sense.
 
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IMD.... if your woofer is reproducing distortion (rumble) - then it will interact with the audio.... (well strictly speaking anything will interact... but at least you can cut out the undesirable irrelevant bits!) for old style vinyl, there was filtering done during mastering, so there should be nothing down in the sub bass in any case - so a brick wall filter makes perfect sense. f
They never had any business putting sub-50hz signals on vinyl. A rumble filter is almost mandatory for most pressings that have even the slightest bit of warp, (which is most of them except for the most religiously cared for examples around).
 
They never had any business putting sub-50hz signals on vinyl. A rumble filter is almost mandatory for most pressings that have even the slightest bit of warp, (which is most of them except for the most religiously cared for examples around).
And heaven help you if your tonearm mass and cartridge compliance aren't properly matched.... then the sub 50Hz signals can make the arm "dance" across the vinyl!
 
then the sub 50Hz signals can make the arm "dance" across the vinyl!
Or bounce up and down on the LP like it was a trampoline. OMG, ROTFLMAO You should have been there.
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Honestly? Personally I even despise rumble filters (highpasses) on phono preamps... ain't nobody fuck with that glorious 30Hz bass the artist and cutting engineer worked so hard for getting on that age old format at full volume.

If your speakers can't tolerate vinyl rumble and pops and everything at high volume and occasional earthquakes, then frankly you need to get better speakers. Until today it's a mystery to me how such things can be described as "dangerous for your system" in any way. :p

It does not stiuplate brick wall filters.
 
OK. I thought about opening a rat's nest by doing this, because it is a simplification of what sometimes has to be done. Some of the points have been made earlier in this forum thread and elsewhere, but it might be useful to bring the key factors in the process to one place. The marketing of room EQ algorithms often presents the impression that all combinations of loudspeakers and rooms can be "fixed", "calibrated" or the like, by means of measurements, math and equalization. In reality, much of the "math" does not include the exceptionally complex, non-linear and occasionally capricious psychoacoustics of human listeners. A critical missing element is that humans adapt to circumstances, bringing our perceptions into acceptable territory. Loudspeakers reproduce sounds. Musicians produce sounds. Both do it in rooms. We don't feel the need to "equalize" - even if we could - the instruments and voices of live music. Two ears and a brain separate the sources from the venue, and adapt to aspects of what the environment contributes to the overall performance. The venues vary, and some are even not ideal, but we manage to appreciate the excellence of fine instruments and voices in most of them.

The special problem with sound reproducing systems is that flaws get superimposed on everything that is played through them. These monotonous colorations can sometimes be beyond the ability of humans to adapt, and they need to be identified and attenuated.

Therefore, the "right way" begins with choosing well designed, timbrally neutral, loudspeakers. If the loudspeakers exhibit audible resonances and/or frequency-dependent directivity issues, it is not likely that measurements in a room will reveal such problems and that equalization is capable of compensating for them. It is often the case that the solution is better loudspeakers. Fortunately these can be identified with good reliability from competently made anechoic measurements presented in a "spinorama" format, following the industry standard. Amir, on this site, makes such measurements and others can be found at www.spinorama.org.

This done, set them up in your room and make a steady-state frequency response measurement at the prime listening position - the stereo seat. We will be paying close attention to the frequencies below about 400-500 Hz, where adjacent boundary effects and room resonances are active. Because much of the bass in recordings is mono (all of it in LPs) drive both loudspeakers simultaneously to evaluate what is happening at low frequencies. Measure them individually to find out what is happening at frequencies above about 400 Hz. If you are using bass management and one or more subwoofers the process is the same, and of course all subs should be running simultaneously. Why? Because multiple sound sources couple energy to room resonances differently when they operate in unison.

You can repeat this at different seats to see how much seat-to-seat variation there is - often quite a lot. Averaging several of these curves is a common practice, making the curves look much smoother, but hiding some awkward realities at low frequencies. Superimposing the curves on one graph is a more useful display of what is happening in your setup. You can then choose which humps/peaks to attenuate, depending on which seats are affected. Remember, at this stage we are looking only at bass frequencies. Narrow dips, however deep, should be ignored. Broad dips can be filled in, but keep the EQ boosts below about 6 dB. Aim for a smoothish curve that is tilted slightly upward at lower frequencies.

The benefits of this exercise will apply only to the seat or seats exhibiting similar shaped curves. That is why multiple-sub methods have been developed aimed at reducing seat-to-seat variations so that one equalization can deliver improved bass to several listeners. These are discussed in detail in Chapter 8 in the 3rd edition of my book.

Above about 400-500 Hz the "early reflections" curve in the spinorama should be similar to what you have measured. If you have well designed loudspeakers the room curve might have some smallish ripples caused by acoustical interference between and among the direct and reflected sounds - these are not problems to two ears and a brain and equalization is the wrong method of addressing them if they were - that is an acoustics issue. Spatial averaging over several microphone locations tends to smooth the room curve at middle and high frequencies, thereby reducing the likelihood that an auto-EQ algorithm (or a person) might try to "fix" something that can't be fixed, or that doesn't need to be fixed. Remember, any EQ applied to a room curve modifies the direct sound, and it the the direct sound that is a key factor in determining sound quality. If you began with loudspeakers designed to have the desirable smooth and flat on-axis/listening window response, they will be degraded.

Finally, pay attention to the overall shape of the room curve. Usually, at least for conventional forward-firing loudspeakers, the room curve will tilt gently downward. If the shape deviates substantially from the early-reflections spinorama curve then one can suspect something is amiss in the acoustical treatment of the room. If listening confirms a problem, then one is free to try modifying the shape of the spectrum with broadband, low-Q, tone-control kinds of equalization. When listening to recordings we get into the circle-of confusion dilemma, where it is difficult to know where the problem lies: the playback system or the recording.

Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain. Happy landings!
Thank you! @Floyd Toole
 
I think this is not correct.

View attachment 442693

If you look at the Y-axis, the level from 50hz to 2khz is at -3dB already, and the downward slope is specified as being 1db/oct. So when it hits -6dB on the Y-axis, really we are only down 3dB. 1dB/oct means:

- 1dB at 4khz
- 2dB at 8khz
- 3db at 16khz
- 4db at 32khz

So when solving for 20khz, it should only be down 3.32dB, not 6dB.
According to that chart, a speaker with an in-room response at the listening position that is -3 dB at 50 Hz and +3 dB at 16 kHz.... is tolerable? That's laughable.
 
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Let me briefly share my latest measurements/re-confirmation of Fq-SPL in my DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo setup.

Please visit my post #1,009 on my project thread for the details...
The latest Fq-SPL (re-confirmation) of multiple amplifiers SP high-level output signals and that of room air sound at listening position: all measured by “FFT averaging of recorded cumulative DSP-processed flat white noise” (as of June 8, 2025)

Fig08_post-1009.png


Fig16_post-1009.png

As you can observe in above Fig.16, I do not like, I do not apply, too-much-smoothing on Fq-SPL spectrum which hides-out various room modes. I would rather prefer common smoothing factor (FFT size, in this case) throughout 20 Hz - 20 kHz which well visualizes various room modes (ref. my post here and here).
Furthermore...
In the end of Dr. Toole's wonderful post here #297, he wrote:
Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain.

And in his post here #315, Dr. Toole kindly wrote responding to my inquiry:
If properly done both swept tone and noise analysis should give identical answers. It is a choice. The principal difference is in the heating of the drivers in sustained tests at high sound levels - power compression. Low frequencies require longer averaging times.

Fig18_post-1009.png
 
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