Arkytype

It would appear your Yamaha has the YPAO feature which will automatically adjust the "frequency response" of your loudspeakers. If your Yamaha RX-V663 didn't come with a measuring microphone, this one might be worth checking out. It is the model specified for other YPAO receivers. At ten bucks, it's a steal. http://cgi.ebay.com/YAMAHA-YPAO-Optimizer-Microphone-Setup-RECEIVER-SEALED_W0QQitemZ190309926488QQcmdZViewItemQQimsxZ20090525?IMSfp=TL090525153002r19323 The RS SPL meter suffers from less-than-ideal circuitry and a non-linear capsule. There are several web sites which have posted the correction curves for the RS unit as well as sites which have instructions for modding the RS unit for flatter response. Here's a link to one of several threads discussing the YPAO and RS. http://www.hometheatershack.com/forums/spl-meters-mics-calibration-sound-cards/1652-ypao-mic.html As for using the RS meter, use the C-weighting scale and set the meter ballistics to Slow. For general system testing, you'll probably prefer 1/3-octave pink noise over discreet tones. Single tone testing can be misleading depending upon your room's "acoustics" and microphone placement. Pink noise is also easier on the ears! Be sure to take readings at several locations around your listening position. If certain bass frequencies sounds and/or measures low or high, move the SPL meter a few feet laterally away from that spot. If the reading increases/decreases you are probably measuring the high and low sound pressure levels associated with low frequency room modes. Here's a neat room mode calculator with a 3-D representaion of the peaks and nulls associated with room modes. http://www.mcsquared.com/modecalc.htm One last suggestion--don't try to boost the EQ to "fill a hole" in the response. If you are sitting in a room mode null, (you'll probably run out of amplifier power trying to fill a 30 dB "notch" at 50 Hz) try a different speaker/listening position setup or (better yet) use some diagphragmatic bass treatment to smooth out the overall bass octaves. Lee

Here's yet another DFH showing how to derive a center channel. If you have a Class D amplifier such as based upon the Tripath chipset, you cannot connect the "grounds" or minus terminals together. Lee Vol. 2 #13.pdf

OK, looks like you can't post more than one .pdf at a time. Here's the DFH article. Lee Dope From Hope.pdf

Here's a Dope From Hope which shows two more useful circuits for deriving a center channel. If you think the use of a center channel is relatively new, William B. Snow of Bell Labs was issued patent number 2,137,032 on November 15, 1938 for Sound Reproducing System. This patent is is noteworthy on several counts. Snow discusses the use of signal delay and volume level to spatially localize a sound source. Helmut Haas didn't "discover" this until 1946. Snow also advocates the use of three or more loudspeakers spaced across the soundstage. If you have the complete Dope From Hope set, read the Symposium on Auditory Perspective written in 1933. It took another two decades before "stereo" became a household word. Lee Snow Patent.pdf

Here's the paper PWK wrote describing how to derive a center channel from a left and right signal. The circuit shown is only appropriate for amplifiers having output transformers. Lee PWK IRE paper.pdf

Joe, Chances are you are experiencing stiction (static friction?) which is a mechanical squealing sound caused by the binder (that holds the magnetic particles to the polyester) coming into contact with the tape guides and heads. I've experienced it with Ampex 456 mastering tape recorded in the early '80s. The tapes had been sitting on a shelf for 8-10 years and the client wanted a CD copy of the performance. It took several passes and a bunch of Q-tips, unwaxed dental floss (for the guides) and Xylene to get a clean playback. BTW, Xylene or Xylol is the original Ampex head cleaner and is sold at hardware stores, Lowe's and Homeless Depot. It will melt most plastic but will disolve oxide or binder buildup on tape guides and heads better than alcohol. Here's one of many articles on the Web that addresses the issue. http://palimpsest.stanford.edu/byform/mailing-lists/av/2002/11/msg00003.html Lee

Gil, et al Here's the link to the AlternaTIFF site. http://www.alternatiff.com/ Once you've downloaded the AlternaTIFF viewer, be sure to click on the test page. If you don't see the proper results, you'll have to change a setting in your Windows Explorer. Here's the link to the gubmint's patent office search page. http://patft.uspto.gov/ Notice you can also search for patent applications such as Roy's. A far simpler approach to searching for and saving patents is to go to Google's patent search site. http://www.google.com/patents It will allow you to directly download the entire patent in one swell foop as opposed to one page at a time from the gubmint's site. I've found that Google's patent search engine can miss all or some of the patents issued if you search only by the inventopr's name. Adding other info such as state , city or assignee (hee hee, he said assignee). Lee

Wes Phillips' review of the Boulder 865 integrated amplifier in the April issue of Stereophile includes this intriguing sentence: Eventually, new speakers occupied my large listening room---Klipsches with a sensitivity of 95 dB that I will be reviewing in due course. Unfortunately, that rules out the P-39F which has a rated sensitivity of 99 dB @ 2.83V/1m. The closest Palladium with the 95 dB sensitivity is the P-37F rated at 96 dB. Other Klipsch models which have the closest sensitivity are the RF-52 & VF-35 @ 96 dB each and the F-2 @ 95.5 dB. Lee

You might want to look at the Behringer CX3400. It offers almost identical features/performance for $40.00 less (street price) than the Samson. http://www.americanmusical.com/Item--i-BEH-CX3400-LIST The main problem with both of these units is that you can only delay the LF. If you have a LaScala, Belle or Klipschorn, you'll definitely need to delay the mid and HF with respect to the LF. IMHO the next step up is not the EV DX-38 but the Behringer DCX-2496. This is a DSP-based crossover with every bell and whistle at 1/4 the cost of the EV DX-38. http://www.americanmusical.com/Item--i-BEH-DCX2496-LIST If you are averse to using front panel menu trees (and who isn't), you can download a free interface and control everything from your laptop. Here's a linkl to the Sync manual. http://behringer-en.custhelp.com/cgi-bin/behringer_en.cfg/php/enduser/fattach_get.php?p_sid=lKEkErrj&p_li=&p_accessibility=0&p_redirect=&p_tbl=9&p_id=54&p_created=1137311343&p_olh=0 Lee

There are three lamps which might float your boat: 1893, 1889 or 1816. These are rated to operate between 13-14 volts and might be dim enough. OTOH, if your pilot light is behind a frosted or jeweled panel-mounted assembly, you might try using some theatrical lighting gel (or colored cellophane) between the lamp and panel. Here's a link to a Mouser catalog page. Scroll down to the T-3 1/4 Miniature Bayonet Base Lamps to see your choices. Just click on any lamp to see the price. http://www.mouser.com/catalog/637/103.pdf Here's a link to an old GE lamp catalog that will no doubt provide hours of amusement. http://www.stevenjohnson.com/ads/ge-ml-catalogo01.htm Lee

You should only measure "zero" ohms between the neutral and ground bus at the service entrance which is the box the meter loop is attached to. At this point (and this point only) the neutral and grounding conductors should be bonded together. There should be a ground rod driven close to the meter box and connected to this common buss. Inside the breaker box in your house the neutral and grounding busses are separate; so while you might measure a low resistance between them, it won't be "zero" because you are reading the resistance of the ground (literally) and the neutral wire running back to the bonding point. There should also be a ground rod connecting the ground buss in the breaker panel. The only time a grounding conductor will be carrying and current is when there is a fault. Hopefully the breaker will trip before injury or damage occurs. Our electric coop suggested (after I had cad welded three buried eight-foot copper ground rods!) that the ground wires from the meter loop to my service panel be connected together. I suspect, my house ground is better than their single ground rod hence the suggestion. Since the wiring is in an underground conduit from the house to the pole, it wasn't possible to retrofit a new ground. If you have aluminum wiring feeding your breaker panel and it is over five years old, turn off the main breaker feeding your panel, clean the bare ends of the feeder wires (two hots and a neutral).Then treat the ends (as well as the inside of the lugs with Noalox. It will act as an anti-oxidant and anti seizing agent. . Lee

Mark, Back in the mid-'70s, one of the Klipsch sales reps (Don Peterson?) had a VW bus with a pair of LaScalas in the back!!! Not much soundstage width but they could sure modulate one's internal organs...... Lee

Great posts, everyone. If replacing your audio equipment's power cord makes a difference, just think what improvements can be made by ripping out drywall and replacing the wiring from your stereo's power receptacle to your home's breaker panel. Don't stop there, you might as well replace that dark-sounding circuit breaker feeding your stereo gear with an audiophile-rated breaker. You know, the one with the zero-crossing detector and healing crystals. That leaves getting the power company to replace that sound stage-narrowing aluminum wire with 99.9999999999% oxygen-free shielded copper cable from the pole-mounted transformer to your house. After all those tasks are complete, you'll lie awake at night wondering, "Is the output impedance of my coal-fired power generation station low enough?" As someone posted earlier (to paraphrase), we know next to nothing about the Nature of Electricity........... Lee P.S. I have found that black painted clothes hangers offer less-wrinkling and creasing of the music than the unpainted hangers. Somehow, the soundstage appears to be hanging about 12 inches from the front wall and 65-68 inches above the floor. Sorry.

It might make a difference but it won't be because of the reasons Rick Schultz avers in the video. Let's say you are currently using a #10 dual-zip OFC stranded copper wire. Schults claims that the signal from the power amplifier is divided up into each of the 423 X 2 individual strands of copper and that each of the strands has a different resistance. He then claims that the individual resistances create (to quote) "time and phase distortion" of the signal. That is, the signal will arrive at the loudspeaker with 423 different time delays. BS If you watch the other videos, you'll see his company (like many snake oil merchants), produces several cures for no known audio diseases. There is one DIY freebie worth a laugh. Does your stereo soundstage seem to be drifting to the left or right? According to Schultz, all you need to do is tighten the speaker mounting screws!! Yep, checkout The Oddiophile Episode 1 to see how!! As long as you have sufficiently low DC resistance, capacitance & inductance, it matters not which type cable you use although large gauge solid wire will offer more challenges with termination and cable dressing. I would recommend using twisted pair cables to reduce the susceptibility of EMI. Lee

Lee, I have a BX-300 purchased new lo these many years ago. This is a well-made three motor deck with a dual capstan transport. One capstan shaft is slightly smaller diameter-wise and the tape is gently stretched as it passes over the three heads. One transport problem you may encounter is the friction idler wheels will slip in FFWD and RWD. Nakamichi's fix was to change to a gear-driven system. If you are experiencing FFWD or RWD problems and your deck doesn't have the mod, just let the tape "play" to the end or try to find some rubber cleaner/rejuvenator such as Rogersol. Here is a link to the service manual. http://www.eserviceinfo.com/downloadsm/25051/NAKAMICHI_BX-300.html Lee

Coytee, Think of power response (PR) as the average of many anechoic frequency response measurements taken in a 360 degree sphere around a loudspeaker. For sound system designers, the PR data is far more useful than a single axial frequency response curve especially when designing an array for specific coverage angles in a large venue. Of course, directivity curves, polar plots and other data are used as well in the design process. The performance of a "perfect" loudpeaker (measurement-wise) in an imperfect acoustic space will suffer the indignities of uncontrolled reflections and poorly spaced room modes. If you are satified (at least for now :>)) with the loudspeakers you have but aren't satified with how they sound, try to quantify what you are hearing or missing. If the bass is excessively boomy (or thin sounding) and relocating the loudspeakers to a different wall or moving them away from (or toward) the wall doesn't help, you probably need to invest in or construct some diaphragmatic-type bass treatment. OTOH if the bass balance seems about right but the highs seem too prominent, try some wall-mounted absorption products. As for your last questions regarding how clarity, intelligibility, spatial clues correlate with a loudspeaker's PR; while there may be a parallel, e.g. a flatter power response might signify higher speech intelligibility, I don't think PR data alone can be used as an acoustic benchmark. I wouldn't rely only on PR data to make a purchasing decision. Besides, how many manufacturers even publish PR data for their products? If you read the articles linked below, you'll learn that a loudspeaker's PR and frequency response don't necessarily correlate. The author of the first link's article notes that a loudspeaker with a flat PR won't necessarily have a flat axial frequency response and conversely, a loudspeaker with a flat axial frequency response may not have a flat PR. Kinda like the saying, while all Port is wine, not all wine is Port. The author of the second link's article raises the oft-overlooked issue of how should a loudspeaker be energized when making axial frequency response measurememnts---constant voltage or constant power source? The third link is to a paper, Basic Acoustics (written by John Vanderkooy) and is a good reference (with dumbed down math) for understanding acoustics and loudspeakers. http://svconline.com/mag/avinstall_sound_power/ http://svconline.com/mag/avinstall_ambiguous_frequency_response/ http://www.linkwitzlab.com/LF-SL/Basic_Acoustics_P773.pdf IMHO the acoustic arrival time from the loudspeaker drivers (woofer, squawker, tweeter) are of at least equal importance to the on-axis frequency response. I'm not just referring just to the arrival time difference caused by the physical front-to-back separation of say a K-55 diaphram from that of a K-77. The acoustic arrival time from an individual driver can vary with frequency. The effect is that the driver is moving toward and away from the listener in a frequency dependent manner. Now that can have a marked (an measurable) effect upon clarity, intelligibility, spatial clues, etc.!! Richard C. Heyser, father of Time Delay Spectrometry (TDS), wrote Determination of Loudspeaker Signal Arrival Times: Parts 1-3 which was published in the Audio Engineering Society's Journal in 1971. By using the then-new TDS measurement techniques he invented while working at JPL, Heyser makes the case for coherent arrival times in the design of a loudspeaker system. If there is interest in reading some of his insightful papers, I can post a few. Lee Lee

Tony, Back when I'd visit Paul on a semi-regular basis (mid '70s), I take a bottle of George Dickel Ivory Label (90 proof). After a little taste "test", he'd pronounce the Tennessee sour mash whiskey to be, "Smooooth!" Thanks for the fond memory. Lee

HDBRBuilder (and other forum members) might confirm this but I seem to remember that you are supposed to remove the four metal "feet" from the bottom of all Klipsch loudspeakers. The "feet" were used at the factory only to allow sliding the loudspeakers on the factory floor to the next stage for painting, testing, boxing, et al. By removing the "feet" you are effectively and efficiently coupling the box to the floor. Lee

From Sound on Sound magazine: The Nearfield Monitor — A Brief History The term 'nearfield monitor' was an invention of the early '80s. It just about predates the explosive rise of the home and project studio and was originally the term applied to auxiliary monitors that sat on the meterbridge in large commercial studios, and were supposed to reflect the sound of typical home audio or TV speakers. One speaker originally defined the breed: the Auratone 5C. The Auratone was, and is, little more than a five-inch 'full-range' driver screwed into a small cube-shaped enclosure. It had little pretence to audio accuracy or wide bandwidth, and was simply intended to provide a reference for the likely sound of recordings when reproduced on an AM radio, or via a TV. So the Auratone was not really a 'nearfield' in the sense that we understand the term now, but it did set a precedent for auxiliary monitors, and prepared the ground for the second nearfield icon — the Yamaha NS10M. The early '80s also saw the rise of freelance 'celebrity' engineers, and I suspect it was one or two of these, carrying a few items of favoured gear from studio to studio, that first introduced the NS10M to the world. As studios began to realise that equipping with favoured gear helped to attract the celebrities, NS10Ms began to pop up everywhere, taking up a position on the meterbridge next to the Auratones. The role of the NS10M, however, was not to mimic the low-fi performance of a TV speaker, but to offer a level of performance and sound that reflected that of a domestic hi-fi. In fact, I believe the Yamaha was derived from a domestic hi-fi product — which, in the context of the question asked at the start of this article, is perhaps significant. But the NS10M had something more going for it. Probably by accident, it displayed a pretty characterful tonal balance and this perhaps helped it become the nearfield benchmark, as material mixed on NS10Ms sounded 'wrong' on anything else. The balance of the NS10Ms also resulted in many a discussion about the exact brand of tissue paper that should be draped over the tweeter in order to dull the balance a little. So despite becoming the industry-standard nearfield monitor, the NS10 has always provoked derogatory mutterings about its sound and tonal balance. What's more, the niche it opened up was soon crowded by countless competitors. We now live in different times. The huge studios, if not quite heading the way of the dinosaurs (there'll always be a role for recording spaces the size of tennis courts, and mixers that could do with a Burger King at the halfway point), have long been under threat from small-scale recording spaces and control rooms. And being very much smaller than of old, the typical control room now has little space for vast main monitors. These days, the nearfield has had a promotion. More often than not, it's now out on its own, the top dog. The Little Rock studio used a pair of the Auratones perched on the meter bridge of the MCI console. The idea was that if the commercial or song sounded good on these, the program material would sound good on the typical car speaker. I don't think the term "near field" was in the parlance of the recording engineer back then. For a couple of hundred bucks you could invest in a pair of AKG, Sennheiser, Grado or Sony headphones that will put most loudspeakers to shame. If you'd like some advice on acoustic treatment of your project studio, send me a PM. While I'm usually up to my arse in alligators project-wise, I'll try to steer you to a better-sounding studio. Lee

I have near fields for critical listening and removing the room artifacts, but they are horrible at more than about 3' away. That statement indicates your control room has some potential acoustic problems. If the "near field" monitors sound bad at 3 feet, I'm not sure any full-size monitor will sound different. If you aren't controlling early reflections and using low frequency trapping, you will probably be disappointed with a speaker that has a very wide dynamic range, good imaging and a flat response, but that has a broad sweet spot. Mixdown monitor loudspeaker choice these days is a highly subjective one. In the "old days" the studio version of the Altec Lansing Voice of the Theater was the loudspeaker of choice. Why? Not because it was particularly accurate, but because it played loud and there weren't that many commercial choices out there that would fit in a control room. In the 1980s, JBL became the defacto standard. Klipsch has never enjoyed any appreciable foray into the recording studio market for reasons that we don't need to go into here. The Cornwalls installed in the Little Rock studio sounded really good. They were separated about 20 feet and toed in slightly to cross at the engineers mix position. They were mounted upside down and the tweeters were slightly above ear level. Depending upon what you are recording, a pair of Cornwalls with upgraded crossover network caps may be all the loudspeaker you need. La Scalas and Belles are heavy and have big footprints. La Scalas vs. Belles: If I already owned the Belles, I'd upgrade them with Al Klappenberger's Trachorn, a Beyma tweeter and extreme slope networks. For vocals, the trachorn easily outperforms the La Scala's K-400 or the Belle's K-700? mid horn. If you are recording a lot of synthesized music with extended bass, you'll need a subwoofer. Otherwise, you'll overcompensate below 50 Hz with bass EQ. Lee

You might be thinking of ARCA--they had a pair of birch Cornwalls in the studio I designed around 1981. There were a pair of Heresys in the main studio for playback. They were driven by a Crown D150 amplifier. Lee

I'd start with 750 degrees. If you are soldering small components to a PC board, the iron will recover heat loss quicker than if you are soldering a piece of copper braid to a terminal. For the larger jobs that act as heat sinks, just crank up the heat a bit before soldering those large components. Looks like your unit has an LED indicator which may illuminarte when the tip is being heated. Just use that to judge how high to set the thermostat. Your soldering iron tip will appreciate a lower temp as high prolonged heat will oxidize the tip beyond recovery. I will amend my previous post to say I do keep a small tin of solder paste next to my solder station to occasionally tin the tip during a long night of breadboarding. BTW, I use 63/37 solder (63% tin and 37% lead) also called eutectic solder. 63/37 melts at a lower temperature and the melting point is just that--there is no range of temperatures at which it melts. It also seems to harden a bit faster than 60/40. Dean is correct about a good mechanical connection being essential. Invest in some high quality spring return electronic pliers (I use Diamond) and do not cut anything but copper wire with your side cutters! Lee

'Bout the only thing I'd quibble with is his failure to make a good mechanical connection with the leads prior to soldering. Us old timers were taught that if the solder joint fails, a good mechanical connection would help keep the device from failing. Another technique I've used for many years is to tin the ends of an insulated wire before soldering it to a terminal or tab. A lot of insulation won't take prolonged heat before melting and pre-tinning eliminates the time needed for the solder to flow into a bare connection. If you roll your own mike cables, fill the XLR connector's solder cups with solder and pre-tin the cable leads. If you are soldering capacitor or resistor leads, use a scotchbrite pad to remove the oxide from the bare wire. I'm not a big fan of using solder flux or paste unless I can't clean the parts being soldered. If you plan to build a kit or solder on a regular basis, invest in a good temperature controlled iron. Weller is my solder station of choice. There are a dozen tips you can choose from to match the job. With the advent of the ROHS (Reduction Of Hazardous Substances), I have been stocking up on lead-based solder. They can have my roll of 63-37 Kester solder when they pry it from my cold dead fingers..... Lee

Hey Dee, Here's a great link that'll show you perzactly how to wire audio connectors. See pages 20-27. As for your setup, I assume you are running unbalanced audio connections. Balanced will require a three conductor 1/4" plug aka TRS or stereo plug. http://www2.yamaha.co.jp/manual/pdf/pa/english/mixers/PM1000E_1.pdf Lee

Ok, definitely 1199 tubes. I get my boxer shorts at K-Mart in Cincinnati....