Beyond the Ariel

Maybe I'm lazy, but I'd be satisfied adjusting the subwoofer level & crossover trim by feeding both channels (L+R summed) pink-noise, with the subs on, and adjusting the in-room response for the smoothest transition, and applying parametric EQ for any peaks in the 50 ~ 300 Hz region.

Note that if the pink-noise is independent for L and R channels, you will end up with a different result. But I don't think that's a valid test condition. Independent L & R noise is a better analog for random-phase ambience information, which spaced microphones will generate in real rooms.

But actual VLF musical content will be pretty well correlated, since the wavelengths are long, and the producers are often aware of mono and 5.1 compatibility considerations. So for EQ purposes, assume the VLF (and 40~100 Hz crossover region) will be mono-dominant, with only occasional excursions into L and R sides of the soundstage, and not much L-R (difference signal) content.

When it comes to EQ'ing the system, it's useful to consider average musical content, and the degree of correlation between channels. On average, most of the content is correlated and center-dominant. Stunts like an intentional 90-degree phase shift between channels is a studio special effect, and only used to briefly liven up the sound and get the listener's attention. Most of the time, the two channels are tightly phase-linked, with the majority of the energy towards the center. Go ahead and equalize for that condition, since that's where the musical energy is.

(The ambience energy is a different story ... that's often random-phase, with intentional L-R (difference signal) content used by reverb systems and also created as a side-effect by spaced microphone arrays.)
 
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Maybe I'm lazy, but I'd be satisfied adjusting the subwoofer level & crossover trim by feeding both channels (L+R summed) pink-noise, with the subs on, and adjusting the in-room response for the smoothest transition, and applying parametric EQ for any peaks in the 50 ~ 300 Hz region.
How do you apply the Parametric EQ? Presumably, your subs don’t use plate amps (lucky you!), so does your hardware equalizer that feeds your standalone subwoofer amps need to have a range that well exceeds the 20 ~ 80Hz EQ range of my plate amp? http://www.rythmikaudio.com/download/XLR2_sealed_quickguide.pdf Or might you be talking about the equalizer utility in the room EQ software?

But if it’s a hardware equalizer that you use for adjusting the subs’ level and crossover to the mains, based on the room’s (previously measured?) response, then wouldn’t you have to leave that equalizer always on in the system?

Also, if you happen to be aware of any, please recommend a reasonably priced mic, preamp and the easiest to use room measuring software for dummies.
 
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But actual VLF musical content will be pretty well correlated, since the wavelengths are long, and the producers are often aware of mono and 5.1 compatibility considerations. So for EQ purposes, assume the VLF (and 40~100 Hz crossover region) will be mono-dominant, with only occasional excursions into L and R sides of the soundstage, and not much L-R (difference signal) content.
Well, maybe not as much as you'd think. Doc Geddes and I went around about this a few years back - and I was able to show that a large number of recordings in digital format have significant stereo content in the low end. Goes against conventional wisdom, but it's what I found.
 
Salk Sound cloned a pair of Gary’s GPA Altec 416-8B midwoofers in his original sealed 3 cu ft cabinets. The box cuts the 416 off below 70Hz, however sharply I don’t know.

For a sealed cabinet, the slope is 12 dB/octave.

Below the Altecs Gary uses his pair of 15” Acoustic Elegance woofers. Unless mistaken, I believe Gary had long driven those woofers with plate amps, though I don’t know the model. But throughout all of our chats I never recalled Gary lamenting his use of these plate amps. So I had assumed that the Fc he had dialed in with the plate amp’s crossover was low enough, its the slope sharp enough and/or the plate amp’s sound quality was good enough for there to be no audible coloration by these amps of the main speakers and Amity 300B p-p mono blocks.

The plate amps were the old #300-794's from Parts Express, nothing fancy. I used their speaker-level inputs, which allowed me to keep them out of the main system's signal path. Their built-in 2nd order LP filter was set to complement the natural 2nd order slope of the sealed mid bass enclosures, somewhere around 60-70 Hz.

I see that Gary has recently swapped out the plate amps with his upgraded Parasound HCA-1500A stereo power amp. And though Gary once expressed reluctance about adding a second layer of digital conversion to his system, the pair of DSP units he installed obviously includes the crossovers needed between the Altec midwoofers and the subs.

The digital conversion only happens to the subwoofer's signal. As before, I use the sealed enclosure's natural rolloff in order to accomplish the high pass. The DSPeaker unit (Anti-Mode 8033SII) provides compensation for room modes, in addition to the same functions previously carried out by the plate amp (LP filtering, infrasonic filtering, etc.). I have no problem with the idea of using DSP in the low-frequency region.

The DSP unit receives its signal from my DAC's second set of outputs (unbalanced). The signal to the main amps is taken from the DAC's balanced outputs, and does not pass though a second layer of digital conversion.

But what most concerns me about why Gary ditched the plate amps after all this time was if it was done primarily because the extensive upgrades that he recently completed on the Amity 300B amps had made the plate amps’ presumably lesser sonics too noticeable, than they have those of the upgraded Parasound amp.

I had already been using the Anti-Mode and Parasound for some time before making the upgrades to the Amity 300B amps.

GregOh1 and virtually all Rythmik sealed sub users I’ve encountered claim they deliver excellent performance in most if not in all respects, particularly for music.

I'm sure they are excellent, certainly better than the plate amps I started with. I don't think you have anything to worry about. Integrating your Rythmik subs with your new sealed midbass enclosures should be a piece of cake. But they should receive their own signal...let your main system roll off on its own. (Man, having to misspell "rhythmic" drives me nuts!)

1.) Should the Altec midwoofer be in a sealed box larger than 3 cu ft?

Using a larger box would lower the system Q, resulting in leaner bass. Depending upon your room acoustics (and your listening preference), that may or may not be desirable.

2.) Build Gary’s crossover as is? Or must it be appropriately modified to produce the right Fc and slope, not necessarily for the Altecs to play lower than Gary's 70Hz cut off, but so that the Rythmik plate amp can cross the Altecs such that the subs start playing "…..under 60 Hz.”-as Greg's do-thereby making any plate amp coloration much less audible?

Yes, build it as is. That passive crossover is around 700 Hz, and has nothing to do with what is going on at 70 Hz. Again, don't run your audio signal through the Rythmic plate amp on its way to the main system. Let your plate amp's crossover handle the low-pass signal for the subwoofer, and let the sealed enclosures do the high-pass.

Gary Dahl
 
Well, maybe not as much as you'd think. Doc Geddes and I went around about this a few years back - and I was able to show that a large number of recordings in digital format have significant stereo content in the low end. Goes against conventional wisdom, but it's what I found.

I'm not saying that below 50 Hz is mono, although that was fairly common practice in the stereo LP days (to improve tracking with cheap ceramic phono cartridges).

What I am saying is where the majority of the energy lies on the majority of stereophonic recordings. If you plot the possible localizations of a 2-channel medium as a circle, with L+R at the top, L only at the Left side, R only at the Right side, and L-R (difference signal) at the bottom, the lower half is usually pretty empty, aside from reverberation.

Direct localizations are usually pan-potted, or imaged by the microphone array, on the upper half of the circle, and over time, the energy tends to bunch up towards the center. The hard-left and hard-right pan-pot localizations are usually restricted to a small percentage of time (or avoided completely), partially because these localizations are uncomfortable when played on headphones, and partially because they are 3 dB down relative to center when played back on mono devices.

The perception of tone (as a perception of loudspeaker coloration) is based on an impression gathered over intervals of a 5 to 10 seconds or longer, which covers a lot of musical territory, and a lot of localizations. Over time, the average is dominated by a group of localizations that are weighted towards the center. Maybe not the exact center, but close to it.

The consequence is a 2-channel, 2-speaker playback system that is equalized for single-speaker playback will not be optimized for stereophonic playback. The only time the single-speaker condition obtains is for the fairly rare, and fairly short, instrument localizations that are hard-left or hard-right ... and these are extremely rare for vocal tracks (the headphone problem mentioned above).

(The Center speaker in a 5.1 or 7.1 system is a special case, since movie dialog is typically Center only, with traveling dialog more of a special effect than standard practice. The main purpose of the Center speaker is to "pin" dialog to the center of the screen, which is important for off-center viewers. For music, some recordings benefit from a center speaker, while others are optimized for the more usual phantom center. It depends on how the recording was mastered.)
 
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An unwanted side effect of optimum equalization for 2-speaker playback is playing back monophonic recordings. In principle, they should be played through only one speaker, replicating the playback system that would have been used when the recording was originally mastered back in the 1950's. Modern low-diffraction speakers can sound kind of odd when listened to one at a time, way off to one side (or the other) of the room. The blurry localization of big 1950's loudspeakers was a positive asset in the days of mono-only source material. It "opened up" the usual closed-in mono sound ... but became unnecessary when the sound stage widened up across the entire room.

The other alternative is to use both (modern) speakers at once, with all the sound coming from a phantom center. This gives a pseudo-spacious effect, but the clarity of vocals can suffer thanks to acoustical comb-filtering.

If the playback system is 5.1 or 7.1, the solution is obvious. All that's required is a decent-sounding Center speaker.
 
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An unwanted side effect of optimum equalization for 2-speaker playback is playing back monophonic recordings.
We have an entire thread on a closely related topic.
http://www.diyaudio.com/forums/multi-way/277519-fixing-stereo-phantom-center.html

When doing room EQ with pink noise, I like to use a pseudo stereo pink noise if possible. That's just pink noise uncorrelated on each channel with about a 6-9dB mix to mono. I do this in a wave editor like Audacity or Goldwave. The resulting signal contains both stereo and mono, just like most recordings. Is it better? I dunno, but I like to think it is. ;)
 
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Blended Pink

Here is pink noise that is stereo with a strong mono center. It might be of some use to someone. It's blended to center approximately like a typical music recording. Equal blending across all frequencies, no weighting. Music is likely weighted a bit.
 

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Maybe it's because I've been spending too much time on the Klipsch forums (I am going with the K-402 mid/treble horn and straight midbass horn below), those guys love folded horns for sub-bass. The downside is they are pretty big, but space is a luxury I have.

I am curious what people think about them, like a Bill Fitzmaurice design EQ'd and using several of them to smooth in room response vs the sealed servo subs being discussed in the last few pages. I'm planning to have a pair of subs that operate up to 100 Hz, since the midbass horns will no longer be horn loading below 80 Hz. And then other subs that fill in the sub-bass/smooth/destruct room modes.
 
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I really liked the bass horn I had about 15 years ago. It had a lot of output and low distortion - sounded very clean. It was similar to a lab horn in design and size. I pointed the mouth into the corner which helped it play lower, so I more or less made it into a corner horn like a Klipsch horn, but it was much larger. I only got rid of it because I was moving around so much. I never directly compared it to a sealed sub.
 
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More Pink Noise

In case it's of use to someone, here are some new pink noise files.
There are 4 tracks in the Zip file. All tracks are 7 seconds long, they are all based on the same stereo, uncorrelated pink noise. In other words, left and right are random and not related to each other at all. In each following track they are blended more and more toward mono.
  1. The original uncorrelated pink noise.
  2. Pink noise with a mono mix. Approximately like a stereo music mix
  3. Same stereo blend as before above 100Hz, mono below 100Hz
  4. Full mono mix, both tracks identical
It's easy to hear the differences, although between stereo blend and mono bass it's subtle.
 

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Updated schematics for the Karna Kay and Symmetric Reichert amplifiers. Voltages and currents are based on measurements of the Thom Mackris Reichert SE 300B and the existing Karna amplifiers.

Hi,

For the symmetric Reichert, what is the value of the 2 capacitors connected to the 500 ohm resistor that is tied to the plates of the 1st 6SN7?

Have a wonderful rest of your day!
 

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  • Symmetric_Reichert.jpeg
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