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EQUIPMENT TEST REPORTS: Hirsch-Houck Laboratory test results on the: Audio Pulse Model One time-delay system, Avid 101 speaker system, Realistic SA-2000 integrated stereo amplifier, and Shure M24H stereo/quadraphonic phono cartridge, JULIAN D. HIRSCH. ------------- By Hirsch-Houck Laboratories Audio Pulse Model One Time-delay System ![]() DESPITE the advances in the state of the high-fidelity art, the fact remains that re corded music reproduced in the home still simply does not sound like the real thing. It can sound very good-perhaps in some respects "better" than live-but, in general, it does not sound enough like an actual performance to fool anyone for a moment. One of the major reasons for this is the limitations that prevent the reproduced program from conveying the ambiance of the original concert hall or other recording acoustic. It is this sense of spaciousness, related to the reverberation and absorption characteristics of the original environment, that is so difficult to capture in a recording and to re-create subsequently at home. At one time, quadraphony seemed to be an answer to the problem. It seemed plausible that a second pair of microphones, located at some distance from the performers, would pick up a greater proportion of the reflected sound in the hall. Recorded on two additional channels and reproduced through suitably placed speakers, this should be able to impose some of the acoustic qualities of the concert hall on the sound of the listening room. Some recordings made in this manner accomplish this very well, creating a more convincing illusion of an actual performance than was possible with two-channel stereo. However, the major manufacturers of four-channel records, presumably following the dictates of the marketplace, have gone heavily into "surround sound" effects. These are certainly impressive and even fun to listen to, but they are "sound effects," farther from reality than mono records were, to say nothing of ordinary stereo discs. There are other ways besides quadraphony to restore a convincing ambiance to recorded sound. One method is to establish a separate delayed signal path, combine a number of differently delayed components into a single reverberant signal, and then play this signal through auxiliary speakers located toward the rear of the room. If the delay times, the manner in which they are blended between channels and re-circulated within each channel, the placement of the auxiliary speakers, and the balance between the direct and reverberant sound sources are all properly adjusted, the result can be astonishingly natural sound. Perhaps best of all, this process can be applied to any program source, stereo or mono, without the need for any special recording techniques. Artificial time-delay devices have been used (and sometimes abused) by the recording industry for some years. Until recently, they have been bulky and very expensive altogether unsuitable for home use. Originally they were mechanical devices, using springs or long air columns to achieve the necessary time delays, but it is now feasible to do much the same thing using electronics alone. The Audio Pulse Model One Time Delay System is an example of a unit that takes such an approach. (Audio Pulse is a division of Hybrid Systems, a well established manufacturer of analog and digital circuitry for commercial and military applications.) The Model One, its manufacturer's first consumer product, contains some ninety IC's in its active circuitry. We have used an Audio Pulse Model One for several months in several different listening rooms. Our experience has given us a pretty clear idea of what it can and cannot do (need less to say, it is not a panacea for all the ills of the recording and hi-fi worlds). It also confirmed our suspicion that any type of objective testing would be fruitless because of the difficulty of interpreting the results and the lack of any frame of reference for com parison purposes. Therefore, our evaluation of the Audio Pulse Model One will be entirely subjective. Description. The Audio Pulse Model One accepts stereo or mono inputs from the tape outputs of an amplifier or receiver (or the main preamplifier output if that is more convenient) and passes them on unmodified to the normal front-channel amplifier and speakers. It also converts these input signals from their analog form into a series of digital pulses at a sampling rate of 250,000 Hz. Before the analog-to-digital conversion, the bandwidth of the input signal is reduced to about 8,000 Hz. This is done partly because of the requirements of the digital conversion process, but also because in a typical concert hall the reverberant sound that reaches the audience has lost most of its high-frequency energy through absorption. In digital form the signals (held separate for the two channels) are passed through a series of shift registers. The shift registers form a memory system capable of storing signals for extended periods. An internal clock signal transfers the contents of each register to the next one as new pulse signals enter the delay system. The output of the last register is converted back to analog form, filtered to remove the "rough edges" from the reconstructed waveform, and made available at the output terminals of the Model One. The number of shift registers through which the signal passes (and the rate at which it is transferred between them) sets the total delay time. In the Audio Pulse Model One, there are four initial delays, varying from 8 to 94 milliseconds (ms). These delayed signals are fed back to the input of the device and recirculated to simulate the multiple reflections that take place in a real room. The recirculation takes place within each channel, and between channels through a cross-feed circuit. --------------- ![]() Physically, the Audio Pulse Model One is a black box 14 1/2 inches wide, 10 inches deep, and 4 1/2 inches high with walnut side panels. Its principal controls are a row of pushbuttons mounted vertically on an inset portion of the front panel. There are also two slider controls on the front that adjust the levels of the delayed outputs. At the left of the row of pushbuttons is the red on/off switch, followed by six gray but tons marked LEVEL MATCH. The delay circuits of the Model One have a limited dynamic range, so the incoming program levels must be set properly to avoid noise or distortion. The peak level is shown by a row of twelve LED lights located on the front panel below the LEVEL MATCH buttons. Eight of them are green, with 0 dB (maximum operating level) shown by an amber light and three red over load lights showing levels above that. The LEVEL MATCH buttons are pressed in sequence until the peak program levels do not go beyond 0 dB for any large fraction of the time. The Model One is meant to function as a unity gain device in that the output is at the same level as the input. Next is a white button marked INITIAL DELAY. When it is up, the initial time delay is 8 ms, which is followed and mixed with later delays of 22, 35, and 58 ms. This combination of delays is used to create the sense of a small-to-moderate-size room. Pressing the button to the LONG setting changes the delays to 12, 36, 56, and 94 ms to produce the effect of larger rooms-from a concert hall to a cathedral. The next five gray buttons are marked DECAY TIME. They control the relative levels of the longer delayed signals that are mixed with the initial delay component and thus vary the reverberation time of the total delayed signal. More than one of the DECAY TIME buttons can be engaged at the same time, giving a considerable degree of control over the reverberant characteristics of the sound. A table in the instruction manual lists the actual decay times (for a signal to drop 60 dB in amplitude following a transient) together with the control settings needed to produce them. The times range from 0.2 to 1.3 seconds. The final two control buttons are marked PRIM and SEC, each having DIRECT and DELAY positions. Normally the primary (input) signal passes through the device unaltered, so the PRIM button is left in its DIRECT position. The secondary signal is normally delayed, so the SEC button is pressed to its DELAY setting. If it is up (DIRECT), the secondary outputs merely feed the normal signal to all four speakers, and this is useful for initial speaker phasing. If the PRIM button is set to DELAY, some of the delayed signals (reduced in amplitude by 10 dB) are mixed with the primary signals fed to the front speakers. This can be used to add reverberation to stereo or mono programs heard through the front speakers only, or to a recording being dubbed onto tape. In the rear of the Model One are MAIN IN, PRIM OUT, SEC OUT, and TAPE IN and TAPE OUT jacks. There are also two pairs of jacks marked SHORT and LONG. These carry signals (removed from the shift registers at inter mediate delay points) which can be used with additional speakers and amplifiers to form six- or eight-channel delay systems. There is a slide switch that boosts the response below 100 Hz at a 6-dB-per-octave rate in its CON TOUR position (on the sec outputs only) to more closely approximate the reverberant frequency response measured in several well known concert halls. There are also two unswitched a.c. outlets. Price: $650. Installation. The Audio Pulse Model One is installed in a sound system exactly as if it were a four-channel decoder. There are two modes of connection possible. If the tape-monitor loop is used, the level-matching but tons can be left at the same setting since the input signal will always be at approximately the same level. However, the main volume control will then have no effect on the delay channels. Alternatively, connecting the Mod el One at the preamplifier outputs gives control over all four channels to the main volume knob, but it necessitates resetting the LEVEL MATCH buttons for any substantial change in listening level. (Of course, using the Model One with a four-channel amplifier or receiver enables you to utilize the tape-monitor loop and still maintain single-knob control over volume for all four channels.) There is a great deal of latitude in the choice of secondary speakers, which need not be identical to the primary pair. However, the mid-range characteristics of all four speakers should be reasonably similar. In the several listening rooms in which we set up the Model One, we used both conventional and multi directional speaker systems, with highly satisfactory results in every case. The Audio Pulse instruction manual recommends that the secondary speakers be placed at the sides of the room, slightly forward of the listener and preferably near the ceiling. These locations may at first appear incompatible with conventional four-channel listening. Nevertheless, in our listening evaluations, which compared the effect of the Model One with that of quadraphonic recordings decoded or demodulated in the appropriate way, we found that the side-located speakers were often quite effective for quadraphonic material, particularly of the ambiance type. Listening Tests. The instruction manual for the Model One ranks among the most useful and complete we have seen for a sophisticated consumer product such as this. It is, in effect, a basic textbook on listening acoustics as they affect our perception of a musical performance. In addition, it explains in great de tail (for those who are interested) exactly, what is happening in every part of the Model One and the effects of the controls. A study of the manual makes it plain that there is no predictable combination of operating control set tings that will give the best results in every circumstance and that each user must experiment for himself. For our part, we found that the complex interrelationships between initial delays, decay times, and secondary speaker levels is further complicated by the nature of the music being listened to. The most important thing, and one which requires some self-discipline, is to play the secondary speakers at such a level that they cannot be heard as separate sound sources. If one is aware that they are operating, the convincing "naturalness" of the total sound is lost. If you (or a guest) should doubt that the secondaries are contributing a worthwhile ambiance enhancement, simply switch them off while listening to a program. The effect is nothing less than astounding, with the sound collapsing toward the front of the room. It is hard to believe that you once thought that the flat, two-dimensional sound of stereo sounded more or less "real." Once you have heard a properly functioning time-delay system, you are not likely to be satisfied with ordinary two-channel stereo again. I am well aware that I (and others) have said very similar things about quadraphonic sound. The reason is simple: a good ambient quadraphonic recording, properly reproduced, is very similar in effect to the sound of a time-delay system. However, with the Audio Pulse Model One, you are not limited to the small number of quadraphonic recordings made with the intention of recreating the hall ambiance. Almost any stereo program will sound more "real" through the time-delay system than will the majority of four-channel records, even when reproduced properly. You are not limited to stereo either. The cross-coupling of reverberant signals between channels of the Model One, and the fact that they are non-coherent, makes it possible to create a very believable reverberant sound field with a mono program source. We played some old mono discs and found that the ambiance of the time-delay system effectively wiped out the "single-source" spatial effect. It was especially interesting to shut off the delayed signals and hear the sound collapse, not to a plane, but nearly to a point! As a general rule, the less reverberation and liveness there is in a recording, the more it can benefit from time-delay enhancement. A very "live" recording can be improved only marginally, if at all, by the secondary delayed signals. However, we found very few cases where the improvement was not worthwhile. If you walk into a room where a properly adjusted time-delay system is operating, you probably will not even be aware of its presence. Everything simply sounds natural, the way it is supposed to. You may be aware that most music systems do not sound that good, but you will hardly guess that a pair of small speakers (perhaps concealed) are responsible for the difference. Most people using the Model One will probably try long delays at first, with the secondary speakers driven at plainly audible levels, just to convince themselves that the unit really can produce cathedral-like sounds in an ordinary room (it can). That done, it is necessary to experiment at length with control set tings to get the desired effect on many types of program material. Once the operation of the Model One is mastered, it becomes simple to press a couple of buttons as required to match the music being played. To us, the most disconcerting aspect of using the Model One with FM broadcasts was the gross disparity between optimum decay times for mu sic and speech. We were repeatedly reminded of this when the announcer's voice emerged sounding as though he were speaking in a cavernous, empty auditorium. Comment. There should, at this point, be no doubt in anyone's mind that we were very favorably impressed with the performance of the Audio Pulse Model One. But it is, of course, a very expensive addition to any music system. Even if you already have a full quadraphonic set-up, there must be an investment of more than $600 in the Model One. If you are modifying a stereo system, you must add to that figure the cost of another stereo amplifier and a pair of speakers. It is certain that Audio Pulse's competitors (there are several already, and more are in the wings) will do all they can to bring the price down, but these are inherently expensive devices. Clearly, the total conversion cost will run to at least $1,000. Is the improvement worth the price? That is as much up to your ears as it is to your pocketbook. It is only fair to point out, however, that at the moment there is nothing else you could buy for $1,000 that would make as great an improvement in the sound of a really high-quality stereo system. +++++++++ Avid 101 Speaker System ![]() AVID'S Model 101 is a floor-standing speaker system that is roughly columnar in shape. The cabinet's top and bottom are made of walnut-finish wood and three sides are covered by removable brown grille cloths. The black rear surface of the cabinet is its only unfinished side. The speaker terminals are located under the base, so that connecting wires enter at floor level. [Also see the ad for this speaker: Avid 101 speakers]The Avid 101 is a somewhat un-conventional three-way system. The 8-inch woofer operates in a ported enclosure, crossing over at 2,500 Hz to a 1N-inch tweeter located just above the woofer. On each side panel, at the same height as the front tweeter, is a 2-inch-diameter cone tweeter operating above 3,500 Hz. There are no "balance" controls, the levels from the four drivers being factory set. The Avid 101 has a nominal system impedance of 8 ohms. According to the manufacturer, its moderately low efficiency makes it advisable to use amplifiers rated at between 15 and 70 watts output. The rated frequency response is 30 to 18,000 Hz ±3 dB. The Avid 101 is about 13 inches square and stands 29 inches high. It weighs approximately 40 pounds. Price: $149. Laboratory Measurements. It is a practical necessity when measuring the "frequency response" of a speaker such as the Avid 101, which radiates (at least over part of its frequency range) into a full 180-degree horizontal angle, to measure its total energy output rather than make an anechoic measurement along any arbitrary axis. Since this is our usual speaker test procedure, the Avid 101's were set up against one wall of the listening room and the microphone was placed about 15 feet in front of them. The frequency response at middle and high frequencies was recorded separately for the left and right speakers, using a swept "warble tone" signal and averaging the two response curves. The resulting curve was corrected for the known room- and microphone-response characteristics. The low-frequency response was measured with the microphone close to the woofer and then separately at the port; the two measurements were then combined with the appropriate corrections for the relative diameters. Splicing this curve to our room measurement gave a total frequency response which, in our view, is truly representative of what the Avid 101 can deliver in a normal home environment. It was a very good response curve by any standards, varying within only ±2.5 dB from 38 Hz to beyond 15,000 Hz. The only visible response variations were a slight rise above 5,000 Hz (where the side-mounted tweeters become effective) amounting to about 3 dB and a similar rise at the woofer's maximum output frequency of 65 Hz. The contribution of the port to the total bass output was limited to frequencies below 40 Hz. The sensitivity of the speaker was a bit lower than that of most ported systems, a drive level of 1 watt of random noise in the octave centered at 1,000 Hz producing a sound-pres sure level (SPL) of 89.5 dB measured at a distance of 1 meter from the speaker. The bass distortion at a 1-watt drive level was very low-under 1 percent down to 42 Hz and still an excellent 5 percent at 30 Hz. When we in creased the drive level to 10 watts, the distortion ran about twice as great, but the speaker still provided a strong fundamental output at 30 Hz. Only when we adjusted the drive level to maintain a constant 90-dB SPL at a 1-meter spacing did we come up against the bass limitations of the single 8-inch cone; the distortion was under 3 percent down to 60 Hz but climbed rapidly to 8.5 percent at 45 Hz. The tone bursts from the Avid 101 were uniformly excellent, about as good as we have seen from a conventional multi-driver system using a crossover network. The system impedance reached its minimum of 8 ohms between 100 and 400 Hz with a maximum of 30 ohms at 55 Hz and smaller peaks at 22 Hz and 3,200 Hz. Comment. The simulated live-vs.-recorded test essentially confirmed our fine measurements, especially in the uppermost octave where many speakers are deficient. In fact, the only significant departure from accurate reproduction of the original sound was an occasional thinness in the lower mid-range; this was noticeable on some musical selections but not on others. Our measurements and listening tests agreed in their indications that the Avid 101 is a very fine speaker system. Our next goal was to attempt to identify its inherent colorations and to compare it to other good speakers using a wider variety of program material than was available on our special "live-music" tape. As it happened, it was very difficult to identify any particular speaker coloration in this way. Whenever we thought we had heard something not to our liking, a switch to another speaker system usually showed that it was in the program material. Compared with some speakers we had on hand, the Avid 101 some times seemed lacking in deep bass. Further listening led us to conclude that the other speakers probably had too much bass! The 101 is notably free of boominess or tubbiness, but it has a very solid bottom end. Most of the speakers with which we com pared the Avid system over a period of sever al months sold for two or three times its price. In the few cases where one of them had a definite superiority over the Avid, it was by a small margin. And there were a surprising number of cases where a direct A-B comparison showed the 101 to be clearly superior. When test comparisons lead to this kind of result, it is safe to say that all the products being compared are of roughly similar quality. We found that the placement of the 101 was relatively noncritical, good results being obtained whether the speaker was against a wall or as much as three feet from it. Obviously, it should not be placed in a corner. If there is one adjective that applies to the sound of the Avid system, it is openness. In contrast to speakers that might have tested better but nevertheless managed to sound as if every thing was coming out of a box at a specific point in the room, the Avid's widely dispersed sound repeatedly gave it the edge in creating a feeling of natural presence. Without question, the Avid 101 is a remarkably good speaker judged by any standard. ![]() ------------ The excellent tone-burst response of the Avid 101 is demonstrated in these scope photos taken at (top to bottom) 100, 1,000, and 9,000 Hz. ++++++++++++++++++ Realistic SA-2000 Integrated Stereo Amplifier THE Realistic SA-2000 (distributed through II Radio. Shack stores) is the company's finest integrated stereo amplifier. The Realistic brand name has long been associated with components appealing to a broad segment of the population and providing good performance at a moderate price. The SA-2000 follows that tradition, but its specifications and control features place it in competition with medium-price amplifiers from many of the better-known hi-fi manufacturers.. The SA-2000 is rated to deliver 55 watts per channel to 8-ohm loads, from 20 to 20,000 Hz, with less than 0.3 percent total harmonic distortion (THD). It is a compact unit, approximately 16 inches wide, 12 inches deep, and 4 1/2 inches high in its walnut-veneer wooden cabinet; it weighs about 21 pounds. The volume- and balance-control functions are combined into two vertical slider controls at the right side of the panel. They adjust the channel levels individually and are normally moved as a pair for changing volume. Channel-balance changes can be made by a slight shift of one control relative to the other. To the left of the volume sliders is a small knob marked PERFECT LEVEL, below which is a pushbutton marked DEFEAT. These controls provide the SA-2000's loudness-compensation system, which will be described later. Across the top center of the panel are knobs for the input SELECTOR (with positions for PHONO 1, PHONO 2, TUNER, and Aux) and the detented eleven-position BASS and TREBLE tone controls which affect both channels simultaneously. Below each tone control is a three-position lever switch that gives a choice of two turnover frequencies and bypasses the tone-control circuits in its center OFF position. The available turnover frequencies are 125 and 400 Hz in the bass, 3,000 or 7,000 Hz in the treble. Flanking the turnover selectors are Low and HIGH filter pushbutton switches. Another pair of three-position lever switches below the input selector control affects the tape functions. The MONITOR switch connects the playback output from either of two tape decks-or the normal pro gram source-to the amplifier circuits. The adjacent DUBBING switch interconnects the two decks for copying a tape from either ma chine to the other. At the upper left of the panel are two blue-lit power-output meters whose logarithmic scales are calibrated from 0.01 to 70 watts (based on 8-ohm loads). Below them are individual pushbuttons to switch in the two pairs of speaker outputs. When the speakers connected to the "B" outputs are placed at the rear of the room, pressing the QUATRAVOX button drives them with an out-of-phase de rived ambiance signal for a simulated-quadraphonic effect. Another button converts the amplifier to the mono mode. There is a push button POWER switch and a headphone jack. In the rear of the SA-2000 are binding posts for the speaker outputs and phono jacks for the signal inputs and outputs. The "A" speaker outputs are duplicated by phono jacks. One of the two a.c. outlets is switched. Price: $259.95. Laboratory Measurements. Although the output transistors and their heat sinks are located entirely within the cabinet of the Realistic SA-2000, they are well ventilated and the amplifier did not become unusually warm during a one-hour preconditioning period at one third rated power and five minutes at full power that preceded our tests. At 1,000 Hz, the outputs clipped at 69 watts per channel into 8 ohms, 91 watts into 4 ohms, and 41 watts into 16 ohms. The total harmonic distortion (THD) at 1,000 Hz was 0.07 percent at 0.1 watt, de creasing to less than 0.01 percent between 10 and 40 watts output. It was 0.4 percent at the clipping point of approximately 70 watts. The intermodulation distortion (IM) was relatively constant-between 0.12 and 0.16 percent for all power outputs from 0.1 to 65 watts. Our distortion measurements across the full frequency range showed that the "distortion," especially at full power and low frequencies, contained appreciable amounts of hum (at non-audible levels) together with the harmonics of the input frequency. To separate the two effects, we used the Hewlett-Packard spectrum analyzer and measured every harmonic component strong enough to contribute to the total reading while excluding harmonics of the power-line frequency. The SA-2000 just met its specification at 20 Hz, with the rated 55 watts output, with an actual distortion of 0.3 percent. The THD decreased with increasing frequency to 0.15 percent over most of the mid-range and 0.02 to 0.07 percent between 3,000 and 20,000 Hz. At half power the mid-range THD was about 0.006 percent, increasing to 0.05 percent at 20 and 20,000 Hz. At one-tenth power it was 0.02 percent or less at middle frequencies and less than 0.1 percent at the extremes. The amplifier delivered a reference output of 10 watts with 60 millivolts at the AUX in puts, or just under 1 millivolt at the phono in puts (both of which are identical). The signal to-noise ratios, respectively 79 and 75 dB referred to 10 watts, were excellent. The phono inputs overloaded at 145 millivolts-more than sufficient headroom to handle the output of any modern cartridge. The tone controls of the SA-2000 are among the better ones we have seen in their ability to modify the response at the frequency extremes without .affecting the mid-range. The combination of a turnover frequency varying with control setting and the choice of either a 125-Hz or a 400-Hz maximum turnover frequency makes it possible to boost or cut the response by several decibels at frequencies below 100 Hz with no effect on the overall tonal balance of the program. ![]() --------------- FREQUENCY IN HZ (CYCLES PER SECOND); CONTINUOUS AND EQUIVALENT SINE WAVE WATTS/CHANNEL The filters had 6-dB-per-octave slopes, with -3-dB frequencies of approximately 60 and 9,000 Hz. Being so close to the edges of the audible spectrum, they did not seriously affect program content, but neither did they provide much noise reduction (although the Low filter is effective against subsonic rumble). The loudness compensation boosted ... ![]() ---- Heat sinks for the SA-2000 are entirely contained within the chassis (upper right). ... only the lower frequencies (under 500 Hz), with a considerable effect when the volume sliders were near the bottom of their range. The "perfect loudness" feature of the SA-2000 makes use of a secondary volume control (PERFECT LEVEL) to which we referred earlier. In use, the sliders are set to a "PL SET" mark about three-quarters of the way up. Then the PERFECT LEVEL knob is used to set the listening volume to the loudest one expects to use. After that, when the volume sliders are moved downward, the loud ness compensation is introduced properly, so that unnatural heaviness is not created at nor mal listening levels. Over the years we have seen only a handful of amplifiers which combined a loudness- compensation circuit with a means of matching the volume-control setting to an actual lis tening volume. This is the only way in which a loudness control can function as it was meant to, and the result is that the Realistic SA-2000 provides really useful, listenable loudness compensation. The RIAA phono equalization was accurate within ±0.5 dB from under 100 Hz to 20,000 Hz and rose about 1 dB in the 30- to 60-Hz range. Interaction with cartridge inductance was minimal, taking the form of a slight boost (instead of the usual loss) of output at high frequencies to a maximum of +1.2 dB be tween 10,000 and 15,000 Hz. The power meters read within about 20 percent of the actual 8-ohm output from 1 to 70 watts. They were considerably more accurate between 10 and 70 watts. Only at very low levels (where it is of minor importance) did the error become appreciable. The meters had a fast rise time, a slower decay, and little overshoot, so that they gave a useful indication of the program power level. Both meters read identically. Comment. If one were to judge the Realistic SA-2000 by measurements alone, its only "weakness" would probably be its distortion figures (0.3 percent) at full power and low frequencies. However, when one considers how unlikely it would be to encounter program content in the lowest audible octaves that had a small fraction of 1 percent distortion and yet required the amplifier to deliver its full output, the whole matter comes into its proper perspective. And this is assuming you could find loudspeakers with distortion low enough to compare with the SA-2000's, which you cannot. The SA-2000 can deliver as much power, with as little distortion, as will ever be required by the vast majority of its users. Against any of its low-frequency limitations, real or imagined, one must balance its tone-control system (definitely one of the better ones available), one of the very few loudness compensation systems that really works, a signal-to-noise ratio that compares with that of some of the most highly regarded (and ex pensive) amplifiers, power meters that are sufficiently fast-acting, accurate, and legible to give the user a good idea of how many watts he is actually delivering to his speakers, and last-but not least-an affordable price. In addition, the SA-2000 has just about all the control and operating flexibility most people could desire (separate preamplifier outputs/ power-amplifier inputs are probably the chief omission in this respect). To us, it seems that Radio Shack has done itself proud in its new "top-of-the-line" amplifier. Anyone who has a stereotypical image of Realistic as a "low-end" brand name owes it to himself to take a good look at (and listen to) the SA-2000. +++++++++++++++++ Shure M24H Stereo/Quadraphonic Phono Cartridge ![]() THE long-awaited CD-4 phono cartridge from Shure Brothers has finally made its appearance. To emphasize its compatibility with all types of discs, Shure calls the Model M24H a "2 + 4" stereo and four-channel cartridge. Unlike many CD-4 cartridges whose stereo performance (and, particularly, high level tracking ability in the audio band) leaves something to be desired, Shure's M24H is offered as a cartridge whose audio tracking abilities rival those of their Model M95ED, which is just behind the top-of-the-line V-15 Type III in performance. The M24H has a high-efficiency magnetic structure like that of the M95ED, although its winding inductance is considerably lower be cause of the necessity for a 50,000-Hz frequency response. Also, its hyperbolically shaped diamond stylus has, according to Shure; the lowest tip mass (0.39 milligram) of any CD-4 cartridge. The hyperbolic tip shape is Shure's equivalent of the Shibata and similar special styli that are designed to trace ultrasonic frequencies without causing excessive record wear. The M24H is designed to track at forces be tween 1 and 1.5 grams, with 1.25 grams being the recommended value. Its frequency response is essentially flat up to about 10,000 Hz, rising at higher frequencies to a maximum at about 30,000 Hz (the CD-4 carrier frequency). Because of its low coil inductance of 160 milli-henries, the cartridge-loading requirements are somewhat different than for other Shure cartridges (which usually give their flat test response when loaded with 400 to 500 picofarads of capacitance). In stereo operation, the M24H can be loaded by a 20,000- to 100,0000-ohm resistance in parallel with a capacitance of 100 to 250 picofarads. For CD-4 service, the recommended load is the standard 100,000-ohm input of a CD-4 demodulator paralleled with not more than 100 picofarads. Externally, the M24H resembles the M95ED, with a swing-away stylus guard on its replaceable stylus assembly. Its output of 3 millivolts at 5 centimeters per second peak velocity is compatible in stereo and four-channel. Price: $74.95. ![]() ---- FREQUENCY IN HZ (CYCLES PER SECOND) In the graph at left, the upper curve represents the smoothed, averaged frequency response of the cartridge's right and left channels; the distance (calibrated in decibels) between it and the lower curve represents the separation between the two channels. The inset oscilloscope photo shows the cartridge's response to a recorded 1,000-Hz square wave, which gives an indication of resonances and overall frequency response. At right is the cartridge's response to the intermodulation-distortion (IM) and 10.8-kHz tone-burst test bands of the TTR-102 and PEAK VELOCITY IN CM/SEC OF TEST DISC TTR-103 test records. These high velocities provide a severe test of a phono cartridge's performance. The intermodulation-distortion (IM) readings for any given cartridge can vary widely, depending on the particular IM test record used. The actual distortion figure measured is not as important as the maximum velocity the cartridge is able to track before a sudden and radical increase in distortion takes place. There are very few commercial phonograph discs that embody musical audio signals with recorded velocities much higher than about 15 cm/sec. Laboratory Measurements. The Shure M24H was tested in the tone arm of a high-quality record player using the recommended loads for each type of operation. Although it tracked our low-frequency test records nicely at less than 1 gram as Shure suggests, we found that 1.25 grams was the best force for overall operation. The lower-mid-range tracking ability was adequate but not outstanding; the cartridge was able to play only the 60-micron level of the German Hi-Fi Institute test record. The output of the cartridge was 2.7 millivolts at 3.54 centimeters per. second. The measured vertical angle of the stylus was 24 degrees. At usual recorded velocities (up to about 18 centimeters per second) the inter-modulation distortion was unusually low, measuring 0.6 to 1 percent with the Shure TTR-102 test record. However, at 22 centimeters per second and higher velocities the cartridge mistracked unmistakably. Playing the 10.8-kHz shaped tone bursts of the Shure TTR-103 test record, the M24H matched the performance of the M95ED and surpassed every other cartridge we have tested in this manner. Its repetition-rate distortion was 0.6 percent at most levels and only 0.8 percent at 30 centimeters per second. The low-frequency resonance in the tone arm we used was at 9 Hz, a near-ideal frequency. The frequency response of the Shure M24H was measured with the CBS STR 100 record in the audio band and with the JVC TRS-1003 test records in the CD-4 carrier band from 20,000 to 50,000 Hz. Although the load resistance (47,000 or 100,000 ohms) had little effect on the audio response, the capacitance did affect the output above 15,000 Hz or so. A low capacitance (75 picofarads) gave the flat test overall response, with a 2.5- to 3-dB rise at 20,000 Hz and an overall variation of ±2 dB from 40 to 20,000 Hz. A typical "stereo" load of 320 picofarads resulted in a more pronounced rise, to a maximum of +6 dB at 20,000 Hz. The channel separation was typically 25 to 30 dB at frequencies up to 10,000 Hz and higher, and it was 7 to 12 dB at 20,000 Hz with the CBS record. Measurements in the CD-4 carrier band are even more subject to test-record differences than audio measurements. Best results were obtained with the JVC TRS-1003 record, which gave an overall response variation of ±3.5 dB from 1 to 50 kHz, including a broad maximum approximating +7 dB at 30 kHz. With the unmeasured channel of the cartridge disconnected from the tone-arm wiring, the channel separation was 20 to 25 dB all the way up to 40 kHz and still a very good 13 dB at 50 kHz. The subjective tracking-ability test with the Shure TTR-110 record confirmed that the M24H was a first-rate stereo cartridge. Only a trace of "sandpaper" quality at the highest level of the sibilance section marred the otherwise perfect performance of the cartridge, which was operating at 1.25 grams. To evaluate the CD-4 performance of the cartridge, we played it through a Technics SH-400 demodulator. Having a record assortment that is probably representative of those available on the consumer market, we soon determined that the sound quality and freedom from "shattering" (a common problem in CD-4 record reproduction) were on a par with the performance of the finest CD-4 cartridges we have used, all of which are considerably more expensive than the M24H. Distortion was still heard occasionally, but only at points where it had been noted with every other cartridge as well. Comment. So far as we are concerned, Shure's claim that the M24H is a "no compromise" stereo/quadraphonic cartridge is fully justified. How else could one describe a cartridge whose stereo performance virtually matches that of the Shure M95ED (and which in most cases could not be distinguished from that of the V-15 Type III) and yet sounds, in conjunction with a modern CD-4 demodulator, at least as good as any other CD-4 cartridge on the market? While some people might consider a 1.25-gram tracking force to be at least a slight com promise, it is a fact that very few cartridges including the aforementioned Shure models cannot benefit from being operated at that force, even if they can cope with most situations at 1 gram or even slightly less. As for the rising high-frequency response, which might also seem to be less than ideal for stereo applications (CD-4 demodulators cut off the signal above 15,000 Hz anyway), we simply could not hear it as a "brightness" or other undesirable quality. To balance this, the strong output of the cartridge in the 30,000-Hz carrier range undoubtedly contributes to its excellent, distortion-free CD-4 performance. Priced between the M95ED and the V-15 Type III, the M24H is substantially less expensive than any other CD-4 cartridge of comparable quality. On the basis of performance alone, the M24H ranks very high; when value for the dollar is considered, it probably has no equal. --------------- Also see: NOISE DILEMMA--Is it possible to have full dynamic range without noise? DANIEL QUEEN
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