Don Richard

Read the reviews. $20K will barely get you started. Nonsense! You can get an excellent set of used wires for that price from them.

As I looked at over this site, one thought kept recurring - "A fool and his money are soon parted" Regarding all Klipsch speakers, this site says: "While thought of as high quality by non-audiophiles these are NOT quality speakers. Painful to listen to." This same site sells used speaker cables, Oracle V1.2 Spades for only $14,072 for a 10 foot pair. This "non-audiophile" would rather spend $14K on a whole lot of things besides freakin' speaker cables. Like a complete sound system, for instance. Anyone who would pass up a pair of Klipsch Heritage, new for half of $14K, to purchase a set of speaker cables for $14K is certanly no audiophile. They would not even be an "audiophool". They would be a complete, utter fool and I am sure that listening to anything that they had to say would truly be a truly painful experience.

AC/DC

The type of woofer that you use in a Khorn makes a big difference. Years ago I put some Electro Voice 15WK woofers from Georgian horns in my Khorns. These woofers were much more impressive looking than the stock K-33s. The magnet structure on the 15WKs was so large that they barely fit into the woofer chamber. They sounded so terrible that I removed them and went back to the original K-33s with less than an hour of listening.

If I'm a sockpuppet for Roy, do I get the employee discount?[]

Design it to do that or purchase such a unit that meets those specs.

The choices you mention will result in lower distortion and higher efficiency. These are easily audible, beneficial effects. Extending the response into the ultrasonic region is unnecessary to achieve those benefits. Ultrasonic performance on a tweeter is a marketing ploy designed to entice those who buy "per spec".

I love organ music. It shakes the walls and stirs the soul. Virgil Fox was a virtuoso musician.

If you read the current information on the Heritage Klipschorn, on this website, you will see that the walls of the room forming the final horn flare is clearly mentioned, as it has been in every brochure that I have seen, going back to the early 1960s. PWK undoubtedly approved, if not actually wrote, this description. In "How To Build Speaker Enclosures" by Alexis Badmaieff and Don Davis, dated 1966, the authors say, "Figs 5-7A, 5-7B, and 5.7C illustrate the dual path the horn takes and how it is mated to the corner of a room to provide the final flare of the mouth. The listener actually sits inside the horn". Don Davis is a well known audio engineer whose credits include working with Richard Heyser and Crown Corporation to bring Heyser's Time Delay Spectrometry instrument to the audio engineering community. He also founded SynAudCon, a company that trained engineers in the use of TDS equipment, equalizers, and studio design. His newest edition of "Sound System Engineering" is considered a bible for audio engineers. In The Dope From Hope, Vol.2, No. 12, dated 10 November 1961, PWK writes: "Back in 1948 we were aware of an interference effect from the right and left sides: a 40 deg. off-axis microphone placement under anechoic conditions (such as an inside corner outdoors) results in a deep response dip at about 280 cps. Also under similar conditions with a hard floor or ground there is an interference between the horn radiation and its mirror image below ground, and a microphone at a 4 foot height will show a dip at 350 cps. These are typical standing wave phenomena: in spite of such effects the system radiates power and does so smoothly if all precautions are observed. In fact our claim of "10 dB peak-to trough ratio" can be bettered by several decibels". The Khorn bass horn is composed of multiple conical segments. There ain't a curve to be seen anywhere in that enclosure. If you are suggesting that the Khorn has insufficient low band efficiency, you are just plain wrong. This has absolutely not been my experience, nor that of anyone who has listened to Khorns in a decent listening room. You got that part right. Any horn can output below Fc, with reduced efficiency and increased modulation distortion. I never said they could not.

Not to be annoying, but that's not data...that's just some dude throwing out numbers. I'm not saying he's right or wrong, but that's not proof of concept. How would one measure end correction anyway? Compare measured results against a prediction? Why not just call the prediction wrong in the first place End correction is not a novel idea. Rayleigh and Olsen mention it and give the math in their texts. They explain how end correction is different for the different types of horn flares.They also describe how an acoustic horn may be designed. There you will see that the flare rate of an acoustic horn is the primary determinant of LF cutoff. This is confirmed in the more recent writings of Klipsch and Leach on that subject. Note that if you follow the different authors' methods of horn design you will get slightly different final results. This is because a horn may be optimized for varying design objectives such as efficiency, size, bandwidth, smoothness of response, maximum output, pattern control, etc. The early horns were designed for maximum efficiency to go with the low power amplifiers of the day. This generally results in large sizes and frequency response limitations. Horns work best when certain ratios are observed. One rule of thumb that I have seen in more than one place is that a horn's path length exceed 1/4 wavelength of the lowest frequency to be reproduced. When shortened too much the device is no longer a horn and cannot be expected to have the characteristics of a horn. This may help explain the results you weren't expecting. Regarding your original post, two of the best ways to achieve good results for your application would be a bandpass-horn enclosure or a conical bass horn. In a bandpass horn one balances the back chamber volume with a front chamber volume like a bandpass subwoofer. The output slot is then horn loaded. For the best writeup on conical bass horns: http://www.danleysoundlabs.com/pdf/danley_tapped.pdf Read the section on Synergy Horns for design info and pictures of what loading a driver near the mouth of a conical horn looks like. Most concert horn loaded enclosures for bass duty these days use either bandpass horns, conicals, or tapped horns. Turbosound, EAW, Yorkville, Danley are some of the companies using these types of bass horns for sound reinforcement. They are not as efficient as the old exponential horns but they are still more efficient than direct radiator enclosures and they take up less room than vented enclosures for the same output. Hornresponse easily models bandpass horns, but the tapped horn and conical bass horn design features are lacking or incomplete.

Transient response is defined by the upper frequencies detectable by the ear, in this case, the human ear. By definition, the ear cannot detect what it cannot hear. If those frequencies that steepen the rise time cannot be heard, no improvement in transient response will be detected. Single frequencies are precisely the information that the ear provides to the brain for decoding into the sounds that we hear. The sounds that we hear are complex waveforms made up of many single frequencies. The ear behaves exactly like an FFT, breaking down the complex waveform that describes that complex waveform at that instant in time into the individual frequencies that comprise that waveform. Any single frequency can exist only as a sine wave. Not only does the ear present various single frequencies of various amplitudes simultaneously to the brain for decoding, but the frequency scale is presented logarithmically, just like the frequency display on an FFT graph. So the FFT machine's output correlates very well to the psychoacoustics of hearing sounds.

So what you are saying is that the only way you would know that is to look at a graph of impulse response? Because you can't hear it? Hmmmm.

Two out of three ain't bad[]

The LF cutoff of a horn is dependant on the flare rate. What you are referring to applies to open-ended and closed-ended resonators such as organ pipes. Horns used as loudspeakers are non-resonant in their designed operating range. The Fc is not lowered. Making the mouth of a foreshortened horn the proper size allows that horn to load the driver better at lower frequencies. http://ldsg.snippets.org/HORNS/design.html This is one explanation of end correction as applies to horn loudspeakers. Googling end correction yields many hits, some having application regarding the organ pipes that you referred to earlier. Edit: The concept of end correction in the design of acoustic horns is not a novel one. The treatises on acoustic horn design by Rayleigh and Olsen both mention it and provide formulae for the calculation thereof.

I had a similar problem. What I did was to remove the sheetrock and pack the space between the studs with fiberglass insulation thick enough so that it became compressed when the sheetrock was reinstalled. This was done on both of the corners that the Khorns fit into.

It is not the function of any reproduction transducer to excite standing waves in a room. This does happen despite best efforts. Actually, the mouth of a horn may be beyond the physical dimensions of the horn's enclosure. Designers of horns apply what some call end correction to account for the true mouth size of foreshortened horns. The particle velocity of a sound wave at the physical mouth structure is quite high, and in the case of the Khorn this wavefront curves around the front face of the enclosure creating a virtual horn segment large enough to cause horn loading to below 40 Hz.

So a direct radiator is superior to a Khorn because one can hear the breakup modes of the driver better in a DR enclosure? This can be construed as distortion and more distortion isn't better, at least not for me.

The way I approach the issue is to realize that, since both tweeters that you described extend beyond the limits of human hearing, the one with the 24kHz response will not audibly improve the transient response of the loudspeaker system if it is swapped for the one that goes to 20 kHz. Just exactly how is this happening? There is no audible mechanism in humans that can cause what you are talking about.

Technically correct, practically irrelevant. Harmonics of the fundamental are what gives rise to steeper rise times (pun intended). A measurement mic may have the necessary frequency response to detect the frequencies which would steepen the rise time of a complex waveform displayed on a graph. An FFT done on that waveform would separate and display the amplitude of those frequencies comprising that waveform at that instant in time. If these frequencies (harmonics) that cause the rise time to steepen are beyond the range of a human's hearing, no information will be presented to the brain which would cause any detection of improved transient response. If one is interested in drawing curves on a sheet of paper, the transient response is improved by measuring the tweeter with the wider bandwidth. Any audible difference, if any, between that tweeter and the other would be caused by other factors. Considering the information given above, those factors have not yet been determined accurately or completely, in the above case. Considering that different people have different hearing limits, blanket statements on the audibility of transient behavior are completely incorrect. What is correct is the analogy of human hearing to an FFT display. The psychoacoustic mechanism of human hearing hearing behaves like an organic FFT because the cilia that move in response to the vibrations of the eardrum are frequency-specific. The cilia that are moving stimulate the auditory nerves, and the brain is then presented the varying amplitudes of the frequencies comprising the waveform at that instant in time. The brain then decodes that information which we percieve as sound. If no cilia are stimulated there will be no information presented to the brain pertaining to those frequencies so we cannot percieve "better transient response" in that case.

No

Most commercial manufacturers of small to medium point and shoot sound reinforcement systems have gone to vented cabinets for that very reason.

Bi-wiring may be effective in the case of a speaker with long stroke woofers. The back EMF from the woofer can interfere with treble reproduction. Running separate wires to the HF and LF from the amplifier reduces this because the LF draws current only from the bass frequencies and the HF draws current from the treble frequencies, even though all frequencies are presented to both. Note that the so called bi-wires that split two outputs from one wire that goes back to the amplifier just a few inches from the speaker end, are completely bogus and are no better than a single wire hooked up conventionally. None of the Klipsch Heritage speakers move the cone enough to cause back EMF problems. As has been mentioned, bi-amping is a more effective method of improving performance with about any speaker but only if aligned and calibrated properly. YMMV

And here's the Audio Engineering Society's proposed standard which, in part, addresses jitter: http://mixonline.com/news/audio_aes_standards_committee/index.html Seems like a lot of attention being paid to a non-issue.

Here's one piece of studio equipment that solves the jitter problem: http://mixonline.com/products/review/audio_apogee_electronics_big/index.html

Jitter sounds like roughness or graininess, shows up most noticeably on female singers. It ocurs generally with separate transports and D/A converters on playback. Perhaps your D/A box is well buffered and doesn't have such problems. There was a website that demonstrated the difference, and it was plainly audible even thru computer speakers. I don't remember the site, but I will post it when the memories return.