pauln

Recording quality is profoundly critical with nicer speakers that reveal more from the music. There are still some things that are difficult to record: - massed violins playing high and loud - large choirs singing high and loud - piano chords that consist of many notes played hard Some of the old methods of recording that only used two, three, or four mics from a distance avoided these problems. Modern close mic'ing with dozens of mics tends to raise this problem if the recording engineer isn't top notch. The problem is over-modulation of multiple complex frequencies that causes a faint "chatter" or "crackling" sound (your "metallic"?). MIC: "It's not you, it's me." Because the source is over-modulation happening in the recording microphone itself, what happens is that of say two mics recording the instruments, the mic closest to the violins for example will over-modulate but the far mic won't, so during playback the crackle is heard coming directly from the left speaker, while the apparent location of the violins is more in set from the left. This is why it is so noticeable and annoying - the crackle artifact is displaced from the instruments' location. This happens in both directions across the sound stage; any instrument section may overmod only the left or right mic, so the two channels playback the crackling sounds as if panned hard left and hard right. Only if the recording engineer allows both mics to be overmodded by the same instrument or section will the crackle take more central positions in the spatial sound stage imaging geometry. It is much easier to tolerate the crackle when it is spatially superimposed on the source instrument's position by playing back in mono mode - this forces all crackles to dead center along with all other instruments, and since the crackle is fairly low level it blends and is masked by the natural sheen of loud high massed strings, vocal phase warble of choirs, and the hammer bounce buzz of pounded piano chords, along with all the rest of the clear instruments.

I think the stock black looks classy. No idea how one might produce that finish for an unfinished pair.

I looked over there including the FAQ; it looks like the assumption is that +12dB is assumed to be enough dynamic headroom to not clip, but +12dB over what level? Average level or metered "fast"/"peak" level? I know, its -12dB down from full scale, but that is not enough. If +12dB over average, that is actually the middle of the range of recorded dynamic range (metered fast/peak over average) which means for the half of recordings that exceed that there is no more margin, and does not begin to include the additional 13dB required for transients. If +12dB over metered fast peaks, the transients are still not being fully powered. I think the dB level for the test file needs to be more like about -30dB in order to account for recordings with up to +18dB dynamic headroom above average and +13dB more for transients. This would cover just about everything except test recordings and those from a handful of specialty studios. The assumption that the listening levels are not clipping perhaps is not being met. Off to play records with milliwatts in the teens... edit... The more I think about it, the more I think I may be incorrect. Maybe the -12dB level of the file is arbitrary, just a way to calculate what the listened level power would be at full scale in order to access rated power required.

That 1.8V is what the file produces when the volume is set to one's maximum digital media listening level. The listened level is an rms voltage corresponding to standardized ballistics (called "Fast" or "Peak" on meters designed for monitoring radio modulation of adult male "announcer voice" for commercial broadcast a very long time ago. Its ballistics accumulate energy about the rate of spoken syllables, which is what vu meters, spl meters, and watt meters display). There are at least 13dB additional dynamics above that level (instantaneous transients) that don't show up in the meters. But the file is a sine wave. It is also rms but does not have any instantaneous transients - if the file is down 12dB in order to present the range to 0dB as the overhead for transients, it is not enough. Why is 1.8V multiplied by 4? Dividing by 1.4 is converting from peak to rms but the measurement is already rms when read. Can you show the actual calculation with an explanation of all the numbers and units? .

OK, but your figures appear to be 9db higher than mine... where is the other 3dB?

How are you calculating power? Why not W=(V^2) / 8 ? which would be 4mW and 2.45mW averages, 0.9W and 0.4W maxiums

Not only does the gasket provide an air tight seal and correct spacing between the driver and the horn; its inside diameter defines the design aperture for the throat. If you are operating for a while without the gasket in place, your throat aperture is larger than design, which means the horn produces less output level, which means you might turn it up to compensate but find the horn's relative level keeps sounding attenuated. Until you have a proper gasket in place with the right inner diameter, you might refrain from loud playing, and resist any loud testing or other exciting sonic experiments / investigations into why it might sound funny without the gasket.

Here is another way to separate the high drivers from the low drivers, if you have a tube amp with multiple taps on the output transformer... These methods present the high and low drivers with different voltages from the same amp through the different taps of the output transformer. Multiple configurations are presented to get different relative attenuation. "Bi-tranny tapping"

Wow, old thread... I'm still using the Sovteks that came with my Wrights 20 years ago. Birthday next month; maybe I'll get another pair of Sovtek, or maybe something else?

The old method was to use a ballpoint pen; take it apart and use the big empty section, put the pin through the hole where the ballpoint stuck out. The tricky part is to not allow lateral pressure to the base of the pin where it enters to glass. That means you don't just push the pen body and bend the pin... you have to hold the pen body hole firmly and press the hole tip in the opposite direction from the pen body movement. Like bending a wire to make a sharp angle with your fingers - if your left hand was holding the wire a foot from where you want to make the bend, you don't just grab the wire and pull with your right hand, you set your thumb at the bend location and push while pulling with your fingers... the hole in the pen body is like your thumb, it keeps the lateral pressure from translating to the base of the pin.

Please don't take this the wrong way; the designer did his homework, but don't be fooled. PWK referenced Kellogg who researched the production of bass frequencies with direct radiator speakers. To put it bluntly, a single 12 inch woofer playing a 50Hz tone loud enough that it is audible is already well into audible distortion... Kellogg needed something like 24 of these speakers to get the distortion down below detection even at low sound level. Nothing has changed since then; except that modern studios and modern listeners have agreed to believe they are producing and hearing loud clean bass... few people have ever heard true clean bass and would probably not like it without adding some grind and slam to make it sound familiar. I have had my Heresys for over 40 years and they are as good a design and implementation as it gets for that size, cleaner and clearer than just about all others, and the low end will hang in there with elevated bass EQ... but that is really just pushing them toward the normal distortion levels of other speakers.

Even the professionals evaluate the sound of speakers up for review by comparing their sound to the reviewer's "reference speakers"... instead of the sound of real live music.

just checking something...

A pair of these under the front of both speakers works nicely, they even come in brown. If you look around, there are bigger ones for industrial use that might suit larger speakers...

Because of the symmetry of the symptoms, it looks to me like there must be a systematic connection mistake on both channels... My theory - two wrongs make a right... you corrected one wrong, so what was right (two wrongs) is now wrong (one wrong and one right). I'm thinking the previous owner reversed a connection (probably on both the crossovers) and also reversed the corresponding connections (on the intermediary panels) either by accident or as a latter correction... the result was that the pair of reversals were incidental. At some point in your disconnecting (not checking for anything wrong) and reconnecting (connecting as expected to be right) you ended up correcting one from each pair of these wrong connections, so now each pairs' remaining wrong connection is no longer being corrected and showing itself. The insidious part is that your dilgence with making proper connections is probably the cause of the problem. The stock crossovers have wires from itself to an intermediary panel connection that has the provision for straps to distinguish a single full channel connection vs a dual split HF and LF connection... and then from there to the amp(s). I know it seems remedial, but all of these connections should be verified as correct. In all this is 20 connections for each channel. - the speaker connections on the crossovers (6) - the connections on the crossovers from the intermediary panels (4) - the crossover side connections on the intermediary panels (4) - the straps configuration (4) - the amp(s) side connections to the intermediary panels (2)