The Entry Level #36
With the LCD-2s, Lefsetz rediscovers his music collection, hears new details in old favorites, and is transported to the recording studio, where he emotionally connects with artists and producers. Of "So Little Time to Fly," from Spirit's 1969 album, Clear (CD, Epic/Legacy 65002), Lefsetz writes: "It's the bass! You can hear it. As for Randy California's vocal . . . you can see him alone in the studio, singing. You can see the mixer grinning as he adds the final touches. All the elements separate, they're clearly defined. It's like putting on glasses after a lifetime of presbyopia." Lefsetz sounds like an audiophile.
He's right, too. At the moment, I'm listening to "So Little Time to Fly." Listen with me, if you like. You don't have to own a copy of the recording; you can find it, and pretty much everything else, for free on the Internet. I'm listening via YouTube, and at around 1:27, it sounds as if the guitarist (Randy California?) slides into a wrong note. Fortunately, the rest of the band holds steady. The bassist (Mark Andes?) is particularly rock-solid. Without him, I figure the song would sound like a disjointed mess.
Speaking of bass: When discussing today's popular headphones, it's pretty much impossible to avoid Beats by Dr. Dre. You know this, I know this, Lefsetz knows this. He contrasts the Audeze LCD-2 with the new Beats Studio closed-back headphone, a product so damn good-looking and downright hip that even I want it, sound unheard. "These are not fashion accessories," Lefsetz says of the LCD-2. "Not something you parade around in to demonstrate how hip and fashionable you are. Consider them the anti-Beats."
The anti-Beats? Why must we be so negative? What if you want a high-quality audio accessory that's also an attractive fashion accessory? What if you want something that demonstrates how hip you are? What if you want the same kind of attention received by those who proudly wear Beats? Lefsetz addresses none of this.
He does, however, raise the topic of price. To achieve this level of sound quality—"the most fantastic listening experience you'll have all year," Lefsetz says—you'll have to spend $1145. "But it's worth it!"
Again, Lefsetz is right. I've held the beautifully built LCD-2s in my hands, placed their luscious leather earpads over my lucky ears, enjoyed the glorious sound they make. The LCD-2s are worth the money, sure, but who has that kind of money to spend on headphones? I don't. My friends don't. I suspect most average consumers don't.
Lefsetz nevertheless makes a valiant attempt to justify the LCD-2s' high price, while also addressing the high-end audio industry's inability to attract a wider audience: "The problem is too much of this expensive stuff is purchased by tweaks who are more into the gear than the music. They quote specs and tech and your eyes glaze over and you move on."
Lefsetz almost sounds like Tinwoman, Avoider of Audiophiles, whom Art Dudley vivisects in this issue's "As We See It." Audiophiles, Tinwoman generalizes, do not listen to music, only to gear. (And she says that as if it's a bad thing!) Lefsetz's tweakers are Tinwoman's audiophiles. Lefsetz and Tinwoman are right and wrong. I agree that many hi-fi enthusiasts are more interested in the gear itself than in the music the gear is intended to reproduce, but I don't blame them for hi-fi's limited appeal. It's their money—they can spend it however they please, just as they can discuss whatever topics they please. Why should their healthy interest in technical specifications deter my desire to hear exciting music beautifully reproduced?
Lefsetz continues: "There aren't enough music fanatics exposed to this great stuff and spreading the word. . . . So I'm just making you aware of these Audezes."
With that, he almost redeems himself. I absolutely agree that too few music lovers have been exposed to true high-fidelity sound. In making more people aware of the Audeze LCD-2s, Bob Lefsetz has done a service to the high-end audio industry, hi-fi enthusiasts, and music lovers; but in focusing on one very esoteric and exclusive product, he has also missed an opportunity to introduce his readers to more versatile and affordable options, like the Skullcandy Aviator ($149.99), B&W P3 ($199.99), and Harman/Kardon CL ($199.99). In fact, he almost makes it sound as if you have to spend $1000 to enjoy a true high-quality listening experience.
"Do I expect you to spend a grand?" he writes. "No. But you should."
I neither expect you to spend $1000 nor think that you should. Unless, of course, you want to. It's your money. But I do want you to know this: In between the wildly fashionable but underachieving Beats Studio and the sonically spectacular but unapologetically inconvenient Audeze LCD-2, there exists an entire world of worthwhile headphones. Some of them actually look as good as they sound. They are free from enormous wooden earcups that require regular oiling. They can be taken with you wherever you go. You might even be able to afford them.
Allow me to make you aware of the PSB M4U 2 and Sennheiser Momentum over-the-ear headphones.
PSB M4U 2 noise-canceling headphones
Why, after 40 years of designing and building loudspeakers, has PSB (footnote 1) entered the headphone market? When I visited him at Canada's National Research Council facility in Ottawa, in May 2012, PSB's chief engineer, Paul Barton, explained that he was working to bring his knowledge of loudspeaker engineering to the relatively new and fast-growing world of portable hi-fi.
In the M4U 2 ($399.99), Barton has designed closed-back, circumaural, noise-canceling headphones with a 40mm dynamic driver in each earcup. The M4U 2 uses PSB's Room Feel equalization technology, designed to produce an open, three-dimensional sound similar to what one might experience when listening to high-end speakers in a typical listening room.
"The experience we have with sound reproduction using loudspeakers in-room has helped to shape our performance targets for headphones," Barton recently explained via e-mail. "Understanding the relationship of speakers and room interaction is important when you use headphones to listen to music intended for playback on loudspeakers in a room. Isn't this the music that most people are listening to on headphones?"
I'm not sure. I suspect that most of today's pop, rock, hip-hop, and R&B—and even much post-punk and experimental music—is recorded to sound good through headphones. "M4U" stands for Music for You. But who is You, exactly? Younger music enthusiasts, I suspect—those who've grown up on MP3s and iTunes, but who nevertheless care deeply about music and who have at least a casual interest in sound quality.
According to Barton, market and lifestyle trends dictated the M4U 2's overall look and feel: "We wanted a little fashion and a good-quality build for the person on the go." Like so many of today's headphones, the M4U 2 physically resembles Beats' ubiquitous Studio model: expandable headband; foldable, high-gloss, polycarbonate frame (available in white, black, or red); large, well-cushioned earpads. David Farrage, of the design firm DF-ID, is responsible for all of this.
The M4U 2s are indeed sturdy and very well built; they're also big and bulky. Out of the box and with the headband fully contracted, they measure approximately 7.5" W by 8" H by 2.75" D and weigh about 12oz. When I wear them, the distance between the outer planes of the earcups increases to 9.5", making me feel like an astronaut or bobblehead doll. While Barton admits that some listeners might prefer a smaller headphone, his general feeling is that an audio component's visual design must not impede its sound. "For PSB, form and function must be in perfect balance."
Although the PSBs are designed to look fashionable, their large size (and my own insecurity) kept me from wearing them in public. Nevertheless, the soft earpads easily accommodated my ears, and the headphones themselves always felt very comfortable on my head. (The average human ear measures 2.5" from top to bottom; the PSB's earpad exceeds that by about 1.25".) I could easily listen for hours without feeling any sort of pain or discomfort.
Footnote 1: PSB Speakers International, 633 Granite Court, Pickering, Ontario L1W 3K1, Canada. Tel: (905) 831- 6555. Web: www.psbspeakers.com
Source : stereophile[dot]com
Lossy Compression: the Sonic Dangers
Low bit-rate" coding, Peter W. Mitchell
"Low bit-rate" coding is the accepted new jargon for digital bit-rate compression schemes that encode audio with many fewer bits than the 1.4 million bit/second used in the CD (footnote 1). Suddenly these schemes are being discussed and evaluated everywhere. Consumer digital recording systems will use moderate bit rates (384 kilobits/second for the Philips DCC and 300kb/s for the Sony Mini-Disc), reductions of 4:1 and 5:1 from the CD bit rate. Proposed digital audio broadcasting systems will use a bit rate of 192 or 256kb/s for stereo. Other applications may use still lower bit rates, implying more drastic compression schemes and more audible compromises. You may already be listening to some of these. For example, FM stations, which traditionally have used equalized telephone lines to carry the audio signal from the studio to a transmitter several miles away, are changing over to digitally encoded studio-transmitter links.
Listening tests to evaluate low bit-rate coding systems have been underway in Sweden for about a year. In the US the Digital Audio Subcommittee of the Electronics Industries Association has launched its own program to evaluate digital bit-rate compression schemes and digital broadcasting systems. Participants in a seminar on low-bit coding at the October AES convention agreed on the importance of caution, to avoid the error of adopting a system that might later be found to have audible flaws. Bart Locanthi, former chairman of the AES digital standards committee, told of receiving a DAT copy of sample low bit-rate recordings that were used in the Swedish listening tests. He discovered false tones in the sound that had escaped detection by all of the listening panels!
Andrew Miller of the Canadian Broadcasting Corporation discussed the CBC's tests of seven low bit-rate systems and played sample recordings made through each. The variations in sound quality were shockingly large. The systems that used a moderate amount of bit-rate reduction reproduced vocal timbres reasonably well. But systems with very low bit rates caused drastic changes in the timbre of something as simple as a male speaking voice, dulling its clarity and adding severe colorations. A recording of a glockenspiel provided an even greater challenge. The systems with the lowest bit rates (below 100kb/s) produced grossly dull and distorted sound. Even the best of the tested systems altered the character of the sound slightly at high frequencies.
Of the seven designs, the one that produced the least sonic impairment was the Dolby AC-2, originally developed for satellite relays of digital audio. An earlier Dolby AC-1 system, which uses adaptive delta-modulation and operates at about half the bit-rate of the CD, has been in widespread use for several years, notably for national distribution of TV sound or networked FM broadcasts. (Historical note: The ADM circuit in the AC-1 system is a refined adaptation of the encoder used in a consumer time-delay ambience simulation system marketed by A.D.S. a decade ago.) The AC-2 system is said to provide similar sound quality with much lower bit rates, 192 or 256kb/s, the same as the European MUSICAM system.
A description of the AC-2 system outlined two reasons for its lack of obvious sonic flaws. One is that its digital filters were matched to the "masking" curve of the human ear with exceptional care, erring on the conservative side to ensure that the noise and distortion products that result from low bit-rate coding will always be inaudible. (Of course, skilled use of masking is nothing new for Dolby; every Dolby noise-reduction system from A to SR has relied on it.) The second factor is Dolby's handling of the transitions between digital data blocks—the points where the amount of data compression is altered according to the level, frequency content, and masking behavior of the incoming signal. Many digital compression systems exhibit a momentary dropout or burst of distortion at those transition points, a problem that Dolby solved with a rapid crossfade between successive data blocks.
AT&T's T1 long-distance digital transmission standard uses either fiber-optic cable or a special type of coaxial copper cable to provide a bandwidth of 1.5 Megabits/second. With digital data compression, several channels of high-quality audio can be carried on a single T1 line—or a combination of audio and computer data. Lucasfilm uses a T1 line to link accounting computers, a network of computer workstations, and several voice-grade telephone channels from Skywalker Ranch (near San Francisco) to studios in Los Angeles. They have also used Dolby AC-2 encoders to send two channels of wide-range audio over the 400-mile path. When a Dolby SR film soundtrack was encoded, transmitted to LA, and bounced back to Skywalker Ranch for comparison with the source signal, the sound was found to be nearly indistinguishable.
Lucasfilm plans to lease additional T1 lines and send all 24 channels of a soundtrack to LA for remote mixing. Other plans will give the term "remote recording" a new meaning: instead of transporting a truckload of equipment to record a concert in a church, just hang microphones, preamps, and AC-2 encoders to transmit line-level signals back to the studio and record them there while monitoring the sound in a familiar acoustic environment.—Peter W. Mitchell
Lossy Images of Audio, Robert Harley
In addition to the Audio Engineering Society's twice-yearly conventions, the organization holds special conferences on particularly timely audio topics. "Images of Audio," the 10th International conference, was held in London this past September. Although the conference's theme was audio for visual images (HDTV in particular), most of the presentations and discussions were about what's happening at the cutting edge of digital audio in general. The 17 technical presentations covered everything from digital audio in video recorders to sample-rate conversion. There was also a series of papers on bit-rate reduction and a panel discussion on these schemes, described in detail below.
The conference was unique in that it included a one-day tutorial on digital audio. Malcolm Hawksford's superb introduction included jitter, oversampling, noise-shaping, filters, and aliasing. In addition to being one of our foremost experts on digital audio, he is an excellent teacher and presenter of technical information.
During his discussion of jitter, Dr. Hawksford mentioned that "we read in magazines about attempts to reduce jitter—like the green pen on a CD." This brought howls of laughter from the audience. Dr. Hawksford quickly admonished the audience with a pointed finger and this response: "Don't be so quick...There may be more to this than you think."
Taking a decidedly contrary attitude was John Watkinson, author of six books on digital audio and video including the excellent The Art of Digital Audio (Focal Press). The thrust of his talk on digital audio data storage was that digital audio storage devices should be thought of exactly as computer data storage devices. He said that digital audio data is identical to data stored on a floppy disc, his American Express card's magnetic stripe, and other forms of data storage. I found Mr. Watkinson's insights into the fundamental principles of digital audio fascinating, but he took the opportunity to attack audiophiles, saying essentially that if the ones and zeros are the same, the sound must be the same: "Somehow I can't conceive of an audiophile [binary] 'one'."
Footnote 1: For more complete discussions of data compression, see "As We See It" in April 1991, "As We See It" in May 1991, "As We See It" in July 1994, and Tom Norton's discussion of the DTS codec in March 1995.— John Atkinson
Source : stereophile[dot]com
Electrocompaniet Classic ECD 2 D/A processor
That Electrocompaniet amplifier was impossible to find in the UK, where I then lived, and in the absence of actual experience, the reputation of its sound quality grew to mythic proportion. But when I did finally hear the Electrocompaniet amplifier at an audio show, I was not disappointed.
Which makes it all the more peculiar that, in more than 37 years of working at audio magazines, I have never reviewed an Electrocompaniet product. With this review of the company's ECD 2 digital/analog processor, which costs a dollar short of $3100, that streak of inattention has come to an end.
The ECD 2 . . .
. . . is a wider-than-usual (18.3") component, with a black-painted steel chassis and a thick acrylic front panel. Into the latter are set a blue fluorescent display on the left, a gold on/off button in the center, and, on the right, four gold buttons grouped in a diamond pattern and labeled Navigation. Of these four, the top and bottom buttons are for volume up and down (from "0" to "100" in 1dB steps), the middle two for selecting the input. The buttons are duplicated on the remote control, which also has a Standby button. When you select a source with valid data, the display shows the sample rate on its left and the volume on its right for about five seconds, after which it reverts to showing the input name.
On the left of the rear panel are the balanced and single-ended analog output jacks; and at its center, the five digital inputs: USB 2.0, and two each coaxial and optical S/PDIF. On the right are the AC inlet, trigger input and output jacks, and an RS232 port. The 18-lb ECD 2 stands on three compliant feet. Its USB input doesn't need a driver program when the ECD 2 is used with Macintosh computers; a Windows driver, and manuals for all current Electrocompaniet products, are provided on a CD-R.
Internals
The ECD 2's digital and analog circuitry are carried on a large, blue printed circuit board mounted behind the rear panel and running almost the entire width of the chassis. This board is stuffed with surface-mount parts. A large XMOS chip handles the USB input, and the digital data are fed to a Burr-Brown SRC43921, the same asynchronous upsampler chip used in the Bryston BDA-2 and Musical Fidelity V-DACII. This feeds two Cirrus 4398 multi-bit, delta-sigma DAC chips. Although the 4398 is a two-channel part, the ECD 2 has one per channel, presumably used in dual-differential mode to give a 3dB increase in dynamic range. This chip has an integral volume control, and a separate port for DSD data; perhaps an upgrade will be offered to enable the ECD 2 to decode DSD datastreams.
The analog circuitry appears to be based on discrete transistors, with local regulation for the power-supply rails and much decoupling in evidence. The power supply is based on a toroidal transformer mounted behind the front panel.
Sonics
Although its volume control makes it possible for the ECD 2 to be used straight into the power amplifier, I stuck with the traditional arrangement of feeding it to a preamplifier, in this case the Pass Labs XP-30 that I reviewed in March. I left the Electrocompaniet's volume control at its maximum setting of "100." Although I occasionally used the Marantz NA-11S1 media server's or Ayre Acoustics C-5xeMP disc player's coaxial outputs to send audio data to the Electrocompaniet, I mostly used the ECD 2's asynchronous USB input with Pure Music running on my Mac mini.
The ECD 2 is specified as handling PCM files with sample rates of up to 192kHz. As the Electrocompaniet has a DAC chip capable of decoding DSD data, I tried playing some DSD files using Pure Music. However, while these files played, it looked as if Pure Music was converting them on the fly to 176.4kHz PCM to make them compatible with the ECD 2.
I've always been a bit suspicious of upsampling converters, concerned that the benefit of driving the DAC chip with higher-sample-rate data would be offset by the possibility of mathematical imprecision in the upsampling process. But the ECD 2 allayed any such fears I might have had. In Trevor Pinnock and the Royal Academy of Music Soloists Ensemble's excellent new recording of Erwin Stein's chamber-orchestra version of Mahler's Symphony 4 (24/192 ALAC files from Linn CKD438), the acoustic around the piano in the opening movements was deliciously apparent, while the images of the solo oboe, clarinet, violin, and cello were presented more forward in the soundstage, and were well defined and stable. The single double bass that carries the responsibility of providing the work's tonal foundation was weighty yet well defined. In the first part of the symphony's third movement, Ruhevoll, poco adagio—where, after several solo instruments serially intone a forlorn melody over a pizzicato bass riff, the harmony resolves on a low bass note—the ECD 2 made sure it was goose-bump time.
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Source : stereophile[dot]com
Simaudio Moon Evolution 850P line preamplifier
Love at First Sight
When it comes to being swayed by a pretty face or curves that are just so, I'm as shallow as the next guy—and I was particularly taken with the good looks of the Moon Evolution 850P. Its case is similar in appearance to other Moon Evolution models, but in this case the visual design works especially well. The two chassis—the preamp itself, and its controller and power supply—are so proportioned that the stacked pair look simultaneously light and substantial, and the mix of finishes, textures, and shapes works perfectly. The balance of sleekness and solidity is carried over to the aluminum remote control, with which a vast array of functions has been cleverly arranged to be controlled with a small number of buttons.
Be Careful—Beauty Can Be Skin Deep
But the 850P's to-die-for looks are only the start. Like its Moon Evolution siblings, the 880M amplifiers, the 850P incorporates a full complement of Simaudio's latest proprietary technologies. One that's immediately obvious, even from the outside, is the mechanical damping system that the company calls M-Octave. It's not enough for Simaudio to isolate the audio circuits from the power supply and control functions by putting them in a separate chassis. Nor is it enough to split the left and right channels into completely separate circuit blocks, each with its own dedicated power supply. Two massive, antiresonant chassis aren't enough, nor are the spiked feet. In the M-Octave system, Simaudio decouples the audio circuit board from the chassis with eight compliant feet, each made of an elastomeric material and pre-loading chosen for their ability to damp the frequencies relevant to that part of the board. The system even suspends the rear panel's input and output connections from the chassis. A platinum-sledgehammer approach? Perhaps, but in a price-be-damned product—the Moon Evolution 850P costs $28,000—well, why not?
SimLink is a dedicated protocol the 850P uses to communicate with other Moon components. The power supply incorporates 40 examples of the unique filtering topology that Simaudio calls Independent Inductive DC Filtering, or i2DCf. The M-Ray volume-control circuitry, a particularly impressive bit of technology, contributes to both sound quality and ease of use. The circuit, which ranges from 0dB (no output) to 80dB (full output) in 530 steps, uses an optical encoder that reads the knob's position, then controls relays to select and insert in the signal path a combination of high-precision film resistors. A particularly nice touch is that M-Ray adjusts the volume in steps of different size, to allow the user to precisely and quickly fine-tune the level. From 0 to 30dB, the increment is 0.1dB. Between 30 and 80dB, holding the remote's Volume button down changes the increment to 1.0dB, to quickly make coarse adjustments; repeatedly pushing the button returns you to increments of 0.1dB, to more precisely adjust the level.
Flexibility of inputs and outputs is another area in which the 850P pays big dividends in usability: It has three balanced and four unbalanced inputs, a monitor/signal-processor loop, and two sets each of balanced and unbalanced outputs. Each input can be labeled and tailored in several ways using the preamp's onboard software. Using Simaudio's M-Lock protocol, a maximum volume can be set for each input, and that level can be offset by ±10dB. Each input can be programmed to activate either or both 12V triggers, bypassed for use in a home-theater system, or disabled entirely, to isolate and remove that block from the preamp's signal path.
In the 850P, even the basics—the mundane, nameless technologies and components—are selected or developed to be the best possible, regardless of cost. At the highest level are the fully balanced differential configuration, the two chassis, and the true dual-mono layout. The 850P uses top-quality parts: huge, custom-built toroidal transformers, and passive components sourced from single lots, measured, and matched.
Giving a Listen
I auditioned the Moon Evolution 850P in several different setups over the course of a few months, but when I sat down to do some serious listening, I began with a system built around the combination of 850P and 880M monoblocks. When I'd listened to this combo earlier this year, I'd been struck by how different the 880Ms sounded from other amps I'd heard. It took me a while to wrap my head around what they were doing, but I finally concluded that they were getting it right. The 880Ms' characteristics were obvious, regardless of what components surrounded them, but the 850P-880M combo was unquestionably better. However, when I assembled and optimized the system (see "Sidebar"), things got a bit interesting. After a round of tweaking that was quick and easy—any change made a BIG difference—I ended up with an eclectic mix of Nordost, Audience, and Stereovox cables, and a listening room that allowed sources and the entire system to sound their best.
I began my first listening session with Acoustic, by Everything But the Girl (CD, Atlantic 82395-2), an album I'd listened to and loved but had always thought a bit overprocessed. Ben Watt's and Tracey Thorn's voices were too smooth and sweet, and the guitar and piano each had a rich, burnished glow that was undeniably gorgeous, but not really what you'd hear at a live performance. I punched up track 5, a cover of Tom Waits's "Downtown Train." The music started just as I settled into my chair, and I froze. A few seconds later I hit Pause, sat back, and exhaled. Jesus, Mary, and Joseph! I thought. What the hell is going on?
I started the track again. This time, I got 15 seconds into it before I hit Pause and stopped to think. To say that "Downtown Train" sounded different from what I was used to is an understatement. It sounded like a completely different recording. Instead of the smooth, overprocessed character I'd always taken for granted as being an integral part of this recording, it sounded as if it had been assembled from voices and instruments that had each been recorded simply and in a natural space. Instead of sounding like pure cane sugar, Thorn's voice took on different characteristics as she changed pitch and phrasing, with a bit of a smoky, whiskey-like texture when her voice dropped deep into her chest.
The opening guitar strums had a sharp, clear attack, each string ringing clearly as it was struck. The mix of sounds composing each chord was clear and properly balanced, beginning with the attack of each string, and building in richness as the body and soundboard resonances built and evolved. Listening to another cover from Acoustic, of Bruce Springsteen's "Tougher than the Rest," I noted that the piano sounded much more alive and lifelike than the luscious but stylized version of it I'd grown used to hearing.
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Source : stereophile[dot]com
Listening #132
Pity the aging perfectionist, the happy diversions of whose younger days—washing records, oiling turntables, leveling equipment racks, cleaning tube pins—have now become hated chores. And this from a man who used to redo his Roksan Xerxes setup every few months, just for "fun."
Sad irony, then, that an interest in mono LPs and shellacs came upon me so late in life, for it is a field in which the experts, among whose number I count God, would have me use different stylus profiles, different tonearm setups, different step-up transformers, and different playback-equalization curves when playing different mono records. I endure in doing the first three, because those things are necessary to both my enjoyment of music and to the preservation of my irreplaceable mono records (footnote 1). But I've never gone terribly far out of my way to fine-tune playback EQ, if only because I find that I'm not nearly as sensitive to lack of tonal accuracy as I am to lack of touch, impact, scale, presence, or drama.
Oh, me of little faith.
Double, double, toil and amplitude
Yet the need for some degree of playback equalization, howsoever unrefined, is a given. Bear in mind that the groove of a non-acoustically recorded record groove is created using an electromagnetic cutter head and played back using an electromagnetic pickup head, devices that convert mechanical energy to electrical energy, and vice versa. Remember also that, in all such things, the velocity of the mechanical signal at one end and the amplitude of the electrical signal at the other are directly proportional to one another—or, to put it more simply, as frequency goes up, so must voltage. That law, first described in 1831 by Michael Faraday, poses a challenge to any electrical phonograph: Because every doubling of frequency carries with it a concomitant doubling of voltage, an electromagnetic music-storage system with a 10-octave range would require an electrical range of, say, 1 to 1024V (or 1mV to 1.024V, or what-have you). That's just crazy.
The impracticality of such a wide range is problem enough, and yet the system brings with it a second challenge, of at least equal severity: If one were to use the same electromagnetic system of inscribing a 20Hz tone as a 20,000Hz tone, the disparity in size between the two would be so great that a single groove of useful width could not possibly contain both tones. Besides, the audio industry's ability to make a pickup capable of tracing, without distortion, the enormous groove excursions required for low frequencies and yet still respond to the microscopically small modulations required for the highest frequencies is, now as then, more than a bit doubtful.
Happily, the witches who cackled around the cauldron of the recording industry in the mid-1920s hatched an ingenious scheme: Rather than abandon altogether the notion of a spiral record groove—already made popular by the earlier acoustic Gramophones and Graphophones and Victrolas, whose 2.5-octave range and purely mechanical nature posed no Faradaysical challenges—they decided that, henceforth, electromagnetic cutter heads would be fed signals in which low frequencies are attenuated and high frequencies are boosted, both at the rate of 6dB/octave. Thus driven, those cutter heads could create grooves of usefully small size in which midtones are represented by constant-velocity modulations, yet the bass and treble ranges are transformed into constant-amplitude modulations. Neat.
Everyone left the coven with the same great idea—and went home and implemented it differently. From 1925 through 1953, virtually every manufacturer of phonograph records attenuated the bass and goosed up the highs of the signals driving their cutters (for a while, Victor changed only the lows, leaving everything else at constant velocity), all by the same 6dB rate. But most record labels used different hinge frequencies to define their EQ curves. A technical survey of records from Brunswick, Clef, Columbia, Decca, HMV, Mercury, Okeh, Victor, and dozens of others would reveal over 100 different combinations of hinge frequencies. And because the very concept of high fidelity implies that every cut-and-boost going in requires a complementary boost-and-cut coming out, a tremendous challenge was laid before the makers of playback gear—who, for the most part, ignored it. Of the preamps and integrated amps made during the classic age of the mono LP, surprisingly few can be adjusted for more than just a couple of different curves.
In 1954, as orchestrated by the recently formed Recording Industry Association of America, the world's record labels signed a treaty to end the war against fidelity: They agreed to an RIAA equalization standard that endures today, wherein the bass and treble hinge frequencies are respectively set at 500.5Hz and 2.122kHz (footnote 2). Those of us who listen to modern stereo LPs must, and generally can, trust that our phono stages are designed with the correct RIAA playback EQ.
Yet we remain ill equipped to hear, at their best, the hundreds of millions of mono records that remain on Earth. A handful of pre-RIAA–friendly phono preamplifiers are still being made, two of which—Esoteric Sound's extremely cost-effective Rek-O-Kut Re-Equalizer II ($369) and the all-tube Sentec EQ10 ($2850, footnote 3)—I've discussed in the January 2009 "Listening" column. Both provide adjustable gain to suit a variety of cartridge types, and both offer user-selectable EQ curves, with separate rotary switches for bass boost and treble rolloff.
Enter the Monophonic ($1500, footnote 4), a single-channel, perfectionist-quality phono preamp with a key distinction: It is, as far as I know, the only phono stage in which equalization settings are arrived at not by means of switches but with stepless, continuously variable potentiometers, thus offering far greater potential for fine tuning and ease of use.
Designed and made in Germany by a mono enthusiast named Michael Fehlauer, the Monophonic is a solid-state preamp with gain that's user-selectable (by means of a rear-mounted switch) for moving-coil, high-output moving-coil, and moving-magnet cartridges. (Input-impedance specs for these different settings are not published.) Measuring just 4" wide by 2.5" high by 7" deep (not counting its outboard power supply), the Monophonic is built into a distinctly well-made extruded case with a smooth satin-black finish. The front panel is glossy black methacrylate, adorned only with two barrel-style knobs that wouldn't look out of place on a Gibson Les Paul guitar: bass boost on the left, treble rolloff on the right. Each knob is marked with the numerals 1 through 10, as well as lines that correspond with those numbers and the points midway between them—and the Monophonic comes with a list of suggested settings for the EQ schemes of various record labels, including the modern RIAA standard (which requires the left knob to be set at 5.0 and the right knob at 4.0).
Behind each knob stands a Bourns potentiometer, and elsewhere inside the Monophonic, enjoying pride of place, reside two op-amps, their identifying markings sanded off. There are also two voltage regulators, nine resistors, two diodes, and a bunch of capacitors—all on a notably neat, well-finished board. The Monophonic is an unambiguously modern piece of gear made in an era when PCB stands for printed circuit board, not fish food.
Footnote 1: Yes, mono records continue to be made. But with one exception—the Electric Recording Company, whose records are frightfully expensive—no company I'm aware of is mastering their discs with a real mono cutter head, opting instead for stereo cutters wherein the two channels are summed. The resulting groove is not the same as a true mono groove cut to mono-era specs.
Footnote 2: The RIAA also came up with a third hinge frequency, suggesting that mastering engineers should revert to constant velocity below 50Hz, in order to avoid interference in the form of rumble.
Footnote 3: This was recently replaced with the identically priced and newly stereo-friendly Sentec EQ11, which I have not heard.
Footnote 4: US distributor: Oswalds Mill Audio. Tel: (917) 743-3780. Web: www.oswaldsmillaudio.com
Source : stereophile[dot]com
Parasound Halo CD 1 CD player
The JA of 2008 was prescient. For the high-performance audio market, it makes a lot of sense to process digital audio data via sophisticated software running on a dedicated personal computer.
Which brings us to Parasound's Halo CD 1 CD player ($4500). Some might find it questionable to release today, as one's first digital-disc player, a machine that plays only "Red Book" CDs, rather than a universal or near-universal (non–Blu-ray) player. It is especially questionable given the market's apparent acceptance of expensive players based on one of Oppo's inexpensive universal players.
That said, I think Parasound's approach will work for music lovers whose large music collections consist mostly of CDs, who want to approach the best CD sound available today, and who aren't prepared to go whole hog into computer audio or a dedicated music server such as Meridian's, formerly Sooloos's, Control:15, both of which require an external DAC. By concentrating on CD playback and ignoring SACD, DVD-Audio, Blu-ray, and even MP3, Parasound can max out the CD 1's CD performance and deliver the goods at the relatively reasonable price of $4500—reasonable, that is, for a product that aspires to be "the best."
Like the other Halo models, the CD 1's robust, full-width case is available in black or what I call champagne silver. The review sample came in black. The overall The front panel has a disc slot (no drawer), a vacuum fluorescent display window, an On/Off switch, a button to switch analog output stages between one based on op-amps and one based on discrete components, transport controls consisting of large oval buttons for Play/Pause and Stop/Eject, and smaller round buttons for forward and back Search and Skip. The rear-panel connections are an IEC power cord inlet; an S/PDIF digital output on RCA, BNC, and TosLink (optical) jacks; and single-ended and balanced analog outputs. There are no digital inputs to allow the CD 1 to be used as a DAC. Remote control is standard; the Polarity button toggles polarity between 0° (Normal) and 180° (Inverted). All connectors are robust and of top quality, and the power cord is unusually hefty.
The CD 1 supports neither AES/EBU digital output nor word-clock sync. Parasound's Richard Schram doesn't think much of the AES3 digital-on-balanced-cables standard, which he believes is the result of an engineering compromise in the early days of digital intended to allow European broadcasters to save money by using their installed base of balanced analog microphone cables as digital-audio cables. Schram states:
"A connector carrying high frequencies must have mechanical dimensions that give it the same impedance as the impedance of the cable. The AES/EBU standard's XLR connector is simply wrong for digital audio, in all mechanical respects. The XLR was developed for analog audio, and frequencies lower than digital. When connector and cable impedances are mismatched, the result is reflections—echoes of the signal travel back and forth, corrupting the real signal. In digital audio this results in jitter, and ringing near the signal edges, which are the transitions between a 0 and a 1. The longer the cable and the higher the sampling rate, the worse the problem."
Fascinating. Barring a deathbed confession by a guilt-ridden EBU apparatchik, I think we'll never know the inside story. But I did think it important to explain that Parasound's decision not to include an AES/EBU balanced digital output jack was not penny-pinching on their part. Anyway, I am aware of no D/A converter that has an AES/EBU XLR digital input but no RCA digital input. Furthermore, I think the inclusion of a BNC jack more than makes up for the absence of an AES/EBU jack.
Inside the CD 1 are a computer optical-disc drive (CD-ROM), an R-core transformer, an analog power supply, an Intel single-board computer running Linux and DSP software from Holm Acoustics of Denmark, and a DAC board, apparently connected to one of the motherboard's USB sockets. A nearby USB thumb drive may contain the computer's own firmware. There is no cooling fan.
Features include two digital power supplies, transformer isolation for the RCA and BNC digital outputs, a linear power supply for the analog section, a choke for AC line filtering, and internal partitions of 3/8"-thick aluminum, all of which are intended to reduce noise. Parasound claims measured jitter of less than 10 picoseconds, which is remarkably low.
The CD 1 spins CDs at four times the usual speed, using the Holm Acoustics software; every bit is read at least twice, and the results must match, before being sent on to the memory buffer. Parasound claims that their proprietary software enables the CD-ROM drive to spin more quietly. I have no idea how that works, but I found the CD 1 to be very quiet.
Reading a CD's Table of Contents takes about seven seconds, so there's a bit of a lag between loading a CD, pressing Play, and hearing music. In the event of an error, the data is re-read several times before the system gives up and begins to interpolate its own approximation of the missing data. Either way, the digital data are played (or sent out via S/PDIF) from solid-state memory—which means that the Halo CD 1 can be called a memory player. Parasound's product literature seems to indicate that the buffer can hold 30 seconds' worth of music.
I had not before sought out a memory player to review, because I wanted a one-box player that also would be significantly less expensive than what was previously on offer. I believe Parasound can accomplish that owing to having its products manufactured in Taiwan, and the economies of scale enabled by its established dealer base (footnote 4).
Parasound uses Analog Devices' AD1853 DAC chip. Parasound didn't take the approach of balanced dual-differential, complementary digital architecture—not for reasons of cost, they claim, but because they believe balanced digital architecture creates timing errors between channels. After being read, the digital data are upsampled to 352.8kHz. Neither the white paper available from Parasound's website nor the owner's manual mentions the digital filter's being a minimum-phase filter, so I conclude that it is not.
Sound Quality
It didn't take too much listening for me to conclude that, for playing regular CDs, the Parasound CD 1 was quite similar in essential sound to Bricasti's justly fêted M1 DAC: Detailed yet smooth, fast yet not fatiguing, and with wonderful frequency extension at the bottom and top. This is as it should be—the Parasound costs about half the Bricasti's price, yet it is limited to CD playback and has no digital inputs or selectable digital filters (including minimum phase), or a volume control.
I think the Bricasti and the Parasound are fairly priced for what they are; the question is whether the Parasound's limited design brief meets your needs (or whether you're prepared to pay for more than one digital solution: eg, an affordable DAC such as Arcam's Sonlink, for lower-resolution, quasi-background music from Sonos and Pandora; and a higher-priced solution such as the CD 1, for when you do nothing but listen.
In addition to the Beethoven string-quartet CDs I mentioned in my last two columns, I listened a lot to the two CDs with which I began this month's column—The Bespoke Man's Narrative (CD, Mack Avenue MCD 1066), Richard Strauss's Zarathustra paired with Strauss's Don Quixote, with cellist James Kreger (CD, Guild GMCD 7204)—as well as to Requiem for a Pink Moon: An Elizabethan Tribute to Nick Drake, by lutenist and singer Joel Frederiksen and Ensemble Phoenix Munich, and many of the favorite CDs I've often written about here. I experimented with the CD 1's selectable output stages, and to my surprise had a preference for the Op-Amp setting that was so slight as to perhaps border on the imaginary.
To directly compare Parasound's Halo CD 1 and Bricasti's M1, I used The Bespoke Man's Narrative and Mary Black's No Frontiers (CD, Gift Horse G2-10002). It was close, but the Bricasti was clearly superior—not so much in tonal quality as in the solidity and dimensionality of the soundstage, that elusive quality of "thereness." The Bricasti filter that sounded closest to the CD 1 was Minimum Phase 0.
If your budget has few or no constraints, it might interest you that the Halo CD 1 performed magnificently as a transport for Bricasti's M1 DAC ($8595). No knock on Musical Fidelity's comparatively affordable M1CDT transport ($999). That said, as a transport, the Halo CD 1 produced a sound unquestionably larger in dynamic scale and in space, as well as having a lower noise floor, and more quiet between the notes. The Musical Fidelity M1CDT was unquestionably more musically engaging than my cheap'n'cheerful Denon Professional DN-961FA, a retired radio-station player, when I used it as the transport. (I've since sold the Denon on eBay.)
I think that Parasound has fulfilled its design goals for the Halo CD 1. If you're prepared to keep playing your CDs and don't care about futureproofing, this is a fantastic player for reasonable money, given the high qualities of its engineering and parts. A quickly rebadged Oppo the CD 1 is not. My vote: Class A. —John Marks
Footnote 1: My review sample of the Halo CD 1 was one of the first 25 production units. It suffered some shipping damage on the way here, evidenced by a mechanical 60Hz hum from the power-supply section. I suspect that the force of the impact loosened or relocated a power-supply component enough that it began to buzz as a result of having been knocked into an electrical field. I returned the unit to Parasound. On learning that a hard blow could have this result, Parasound redesigned the power supply around a larger inductor, revised the parts layout, applied these changes to the PCBs of the review sample and all other production units, and returned the sample to me. It performed flawlessly .— John Marks
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Source : stereophile[dot]com
