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Everything posted by MikeRivers

  1. Monitors - or a better monitor mix. Or turn your amp volume down. Or all of the above. That's all pretty standard stuff, so what's unique about your band that keeps you from using the standard solutions? Money? Idiot mixing monitors? Nobody mixing monitors?
  2. My tool of choice for holding nearly anything small enough to solder with a soldering iron is a small Jorgensen parallel jaw wooden handscrew. With two of them, it's like having four hands. https://ponyjorgensen.com/products/specialty-clamps/
  3. It's about time somebody wrote an article like this. Guitar switches and knobs are notorious for being unmarked and in the "well, everybody knows what that switch does" category. The real meat here is in your last paragraph, except that I've never seen a manual for a guitar (maybe because I've never bought a new electric guitar) or much of an explanation on the manager's web site. Ask a clerk? Ummmm . . . maybe they know more about guitars than microphones or mixers. But thanks for the starter info.
  4. Hey, Phil - Thanks for the review. I've been trying to get one of these for a review for over a year now. Glad they're finally getting out and I'll get mine. In the spirt of the Pro Review, I have a question for you: "You'll also find two large rotary controls on the front panel. The one near the display is used to select various menu items, and has a built in push-to-select switch. The other large knob is a main output level control, and adjusts the level of the rear panel 1/4" TRS balanced +4dBm output jacks. These jacks provide a summed output of all the recorded tracks courtesy of the uTrack 24's onboard DSP mixer, or serve as main control room outputs when the unit is used as a computer audio interface. " Can you route a pair of channels from a DAW monitor mix directly to those rear panel "control room" outputs? Or do you need to return the DAW mix to a pair of channel inputs and then route that to the monitor outputs via the uTrack's DSP mixer?
  5. The problem with fretting one string to get it to the pitch of the next higher string is that frets aren't always that accurate, and even if they're spaced mathematically accurate, the bridge is in the correct place, and the instrument is set up well, you're still at the mercy of the tempered scale - unless of course you have one of those kinky oddly fretted fingerboards that attempt to compensate for what was designed for Mr. Bach's Well Tempered Clavier. One system that works better most of the time is to compare octaves. It's easy enough to hear if two notes an octave apart are identical by listening to beats and tuning so there aren't any. You'll have to fret a bit, like tuning the D string by fretting it at the 2nd fret and matching this E to the 1st or 6th string, which you've already tuned by octaves to match. Or better yet, get a clip-on tuner, tune each string to it, then correct it by ear. At a workshop at a Newport Folk Festival many years ago (back when a "festival workshop" consisted of a pole stuck in the ground in the field, with a sign on it and a chair, someone asked Reverend Gary Davis why he always tuned his B string a little sharp. He replied "Well, it's SUPPOSED to be that way.
  6. Don't artists use more than one or two microphones? So many companies have interfaces along the same model as these new Mackies and they continue to get better as far as raw audio quality. But high quality 4-mic interfaces that connect via USB seem to be as scarce as ever. I suppose if you're creative (or maybe indecisive) enough to want to record more than two channels, you're "pro" enough to have a Mac or one of the rare PCs with Thunderbolt.
  7. "Don't ever go over 0 VU when recording digitally." Digital gear doesn't have VUs or VU meters. The level you don't want to exceed, and actually, that you can never exceed, is 0 dBFS. If you're using a VU meter for reference, for example on an analog mixer or mic preamp, you need to establish a relationship between 0 VU on the analog meter and 0 dBFS on the digital meter. Usually the system is calibrated so that at 0 VU, you'll have a recording level somewhere between -14 and -20 dBFS, depending, if you have a choice, of how much headroom you want to allow. See my article entitled Meter Madness" http://mikeriversaudio.files.wordpress.com/2010/12/meter-madness_1-2_revised.pdf
  8. Here are some important details for those who have never done anything like this before. First, readers may not know what a "signal generator" is. You don't need a piece of calibrated lab equipment for this. Most DAWs and audio editing programs have a function to generate a sine wave. For something more portable (I hate to be tied to my computer workstation when doing shop tests or troubleshooting) there are web resources for downloading WAV or MP3 files of sine waves. You can transfer those to a handheld recorder and that becomes your test generator. Alternatively, there are a few apps for mobile devices to generate test tones. You'll need to be creative with cables but that shouldn't be a problem. I have a few cables with common audio connectors on one end and alligator clips on the other end to make it easy to hack together something temporary. The article doesn't state where the generator and amplifier or other device under test are to be connected. Since signal flow on a diagram is usually assumed to be left to right, so the generator should be connected to the jack on the left (3). The device under test (DUT - an amplifier or other input device) should be connected to the jack on the right (1). It really doesn't matter once you understand how a pot works, but it's best to set up a convention. As drawn, the pot will be at zero ohms (or, technically, as close to zero as it will get), when the shaft is turned fully counterclockwise with the pot oriented so that the shaft is pointing toward you. Check it with the Ohms setting on the multimeter just to be sure. Measure the generator level at the jack to be connected to the DUT, but with nothing but the meter there connected yet. Write down the voltage and then connect the DUT with the meter still connected. Then rotate the pot and stop when the voltage across the DUT input has dropped to half the open circuit voltage. The drawing show a 1 megohm pot, but some devices have in input impedance greater than that and even with the pot at maximum resistance the voltage at the DUP input won't drop to half the open circuit voltage. You can stop there and be happy that you have an input that's greater than 1 megohm, or you can start tacking 1 megohm resistors in series (between the pot and the right (1) jack) until you can get to half voltage with the pot, and then add the extra resistance to what you read when you measure the pot. You could get a 10 megohm pot, but that reduces the resolution at the low end, should you want to start measuring everything within sight, where line level inputs are typically in the 5 to 20 kilohms. Before you go off checking you mic preamp's input impedance, understand that nearly all mic inputs are differential. You can use this same technique and test setup, but you'll need to connect the generator between pins 2 and 3 of the XLR input and ignore pin 1. You'll also want to monitor the preamp to be sure it's not clipping. You may find that the input impedance varies over a small but measurable range as you adjust the input gain of the preamp. If the preamp output is connected to a DAW input, you can skip the voltmeter measurement, just set the generator output low enough so that you aren't clipping the preamp, and monitor the output level using a meter in the DAW. First check and jot down the level with the pot at zero ohms, then turn the pot so that the level drops by 6 dB (that's 1/2 the voltage) and measure the pot's resistance. -------------- Mike Rivers used to write regularly for a few magazines but they won't let him do it any more because all of his articles are too long. He has retired to the safety of on-line writing since adding another page doesn't require selling another ad.
  9. "While I don't see guitar companies doing away with the 3/4†jack anytime soon," I think you mean 1/4" jack. There was a Gibson system (the Les Paul Recording) that used a balanced connection between guitar and amplifier on an XLR connector, but they left the 1/4" jack so the guitar could be played with a conventional amplifier.
  10. "Remember those ground posts on turntables with the screw terminals where you could attach a ground wire to keep hum at bay? Have you ever seen ground posts on anything else? Of course not!" Craig's memory is a little faulty here. Turntables had a ground _wire_ that was permanently attached. The ground post, where you connected the spade lug on the other end of the turntable's ground wire was on the receiver or preamp. So convenient - you didn't have have to buy a piece of wire to ground your turntable's innards, nor did you have to solder anything or drill any holes.
  11. Always good to see Sister Rosetta Tharpe included in such a list. She should be #1, though. The clip here is pretty tame, but she's where Chuck Berry got his licks, and without Chuck Berry, where would real rock and roll be?
  12. Don't hate the saxophone - it's the first instrument of rock 'n' roll. It's OK if you hate bad arrangements.
  13. Audio Engineering Society 2014 Convention Los Angeles, California October 9-12, 2014 ©2014 – Mike Rivers The Audio Engineering Society returned to Los Angeles for the 137th annual convention for the first time since 2002 after several years of West Coast shows in San Francisco. When first moving from LA to San Francisco the exhibitors and attendees were enthusiastic about the change, but things change and it was time for a fresh venue. Attendees and exhibitors were enthusiastic about the move, and the uptick in the audio industry made this the biggest and best attended US show in several years. The technical program was strong and, while the exhibit space was fairly compact, there was still a lot to see and learn about. I didn’t return home with my usual large pile of product literature which means that while I had lots to talk about on the show floor, I didn’t find too many new products to write about, but never fear, there was some cool stuff. On with the show: Mics While there are still plenty of good, modest-priced mics around, it was refreshing to see a few new, high-grade (and expensive) mics from both new and old vendors. First up is the AT5045, a new member of Audio-Technica's flagship 5000 series, following on the heels of the AT5040 introduced last year. I guess now that there are two, it’s officially a series. The 5040 is targeted for vocals, while the 5045, being slimmer and tweaked in another direction, is aimed toward instruments. Like the 5040, the 5045 is a cardioid fixed charge (electret, though that moniker seems to have garnered a bad reputation) condenser mic with a large rectangular diaphragm. It has a rather high sensitivity of 17.7 mV (–35 dBV)/Pa and can handle loud sources, up to 149 dB SPL. The US MSRP is $1400 or $2500 for a pair. Another newcomer to the scene is Russian manufacturer Soyuz, showing two tube condenser mics, one larger “bottle” style, and one smaller “pencil” style. The small body SU-011 can accommodate the large capsule of the SU-017. The small capsule is cardioid, the large capsule is available in cardioid (standard), omni, and figure-8. Sensitivity is 16 mV (-36 dBV)/Pa. One nice thing about Russian tube mics is that Russia has never stopped production of vacuum tubes, and the ones used in these mics come from a factory that’s been in continuous operation since 1960. I don’t attempt to evaluate the sound of a microphone on a show floor, but I (and just about everyone else I talked to about these mics) was impressed with their construction and workmanship. When I commented about the really heavy brass body, I said that it reminded me of my grandmother’s old brass samovar. They reminded me that the mics come from Tula, where, also, do samovars. What’s old is new again. Neumann has re-introduced the U47 FET, a mic that helped shape the sound of recorded music in the 1960s through the 1970s. It’s the original company (well, now owned by Sennheiser) making as close to the original mic as possible. They have all of the original documentation for the mic and many of the parts suppliers are still around and are able to provide many of the exact parts used in the original mic. The diaphragm is made from PVC as was the original U47, so it will age like an original mic. So when I said “everything original except the lead in the solder,” my booth guide just grinned. I had to wonder why they decided to re-introduce this mic. Perhaps it was just to show that they could still do it (we’ll be waiting for the reviews to come in), but in reality, they felt that it was a darn good mic and there are not many still in use. While it’s possible to buy an old U47 FET, typically in the $3000-$7000 range, depending on the condition and who has recorded through it, $4,000 will get you a brand new one, up to factory specifications, with a warranty. I doubt that the new one will kill the vintage market, but it’s another source for a fine mic. Just in case you’ve never seen one, I’ve included a shot. Last but not least is the RM1 active ribbon microphone from Cliff Microphones. Cliff Hendricksen is an engineer and musician, a man after my own heart, having played fiddle and called square dances. His related background is as a loudspeaker designer, both for Altec-Lansing and Bose. He’s finally realized his dream of building a ribbon mic, and intends to market it. His experiments led him to a design with a huge magnet assembly that produces a stronger field than any mic on the market today. He says it sounds great, but it’s really hard to tell at the show. What was apparent was that it didn’t have much of a null 90 degrees off axis, an important and very useful characteristic of a ribbon mic. He seems very sincere about making a top quality mic, and he’s looking at a selling price of around $3000. If you get one, you’d better get a Latch Lake micKing stand, as the mic weighs about 8 pounds. It might be the ugliest mic at the show, though I wouldn’t hold that against it if it ends up sounding great. You be the judge. Consoles We‘re living in an era of studios that have never had a mixing console. That’s fine if you don’t mind working that way, but a console is a mighty useful thing, particularly when it comes to tracking. (hint: NO latency) We’re still seeing plenty of big and small consoles for live sound and post production, but for many years, there were pretty slim pickings if you wanted a medium size studio console. I’m pleased to see that a few more have appeared on the scene in the past couple of years, most of them coming from companies who have made such consoles in the past, and are building on that history but making adjustments in routing, inputs, and outputs for better workflow with the inevitable DAW. Last year, API brought us The BOX, a 16-input stereo mixing console with a couple of mic preamps and a frame for 500-series modules which can be inserted nearly anywhere in the signal path. For all but the smallest tracking sessions, you’ll need some outboard preamps to work with The BOX, but there’s plenty of channels for monitoring while tracking, and, if you want that API console sound in your mix, you can use it for analog summing of DAW tracks. This year, Solid State Logic (SSL) introduced the XL-Desk which, when viewed at a distance, is their take on The BOX, though it’s bigger in all dimensions (including gozintas and gozoutas) and more expensive. There are 16 mono and 4 stereo channel strips that can be used in mono when tracking. Those can be assigned to one of four stereo mix buses, of which three can be submix buses when routed to the fourth (typically the main stereo) bus. Each of the mix channels have two inputs, switchable between the recording source (mic or line) and DAW returns – the equivalent of the Line-Mic/Tape inputs on a conventional multitrack console. The first eight channels have SSL VHD (Variable Harmonic Drive) mic preamps. There’s an 18-slot 500 series frame which can be used for additional mic preamps, equalizers, or compressors. In the tradition of knowing how your chickens are counted, it’s actually a 16-slot frame and a 2-slot frame, with the 2-slot frame wired to the channel 17/18 pair, and that slot comes filled with an SSL Series G bus compressor. Normally the module directly above a channel strip is in that channel’s signal path, but there’s a separate insert/return for each channel as well as access to each 500’s input and output, so you can patch any number of on-board or outboard processors into any channel. A patchbay helps here. Those last two slots normally in line with channels 17/18 can be patched anywhere, as can the compressor There’s one stereo and two mono auxiliary sends and two stereo auxiliary (typically effect) returns. The sends can be routed to two cue buses with talkback to each cue output, a pretty flexible monitor controller that can accommodate up to 5.1 surround monitoring. To top it off, there’s a studio listen mic input with SSL’s classic “listen” compressor. There’s a lot of flexibility here if you keep your wits about you. If memory serves me correctly, it’s about $20,000 with all the module slots empty and about $10,000 more to fill them with SSL EQs. Mix and match. Remember that it’s an analog console, no automation, and no DAW controls; if that’s what you want, they still have the 900 series for about 3 times the price of the XL-Desk. Trident consoles have their own legacy. While the studio for which they were initially designed is long gone, the name and the spirit now belongs to the US marketing firm PMI, who gave us Studio Projects microphones, Joemeek signal processors, a few Trident rack mount processors, 500-series modules from the Joemeek, Studio Projects, Trident, Valley People, and Tonelux lines. At the January 2014 NAMM show, they showed a preliminary version of the Trident 88, a modular in-line monitoring console available in frame sizes of 16, 24, 32, 40, and 48 input channels, with 8 subgroup buses. Now the design is complete, parts are built, and it’s ready to ship. The mic preamp circuitry is discrete class A with optional Lundhahl input transformers. The equalizer has four sweepable bands with high and low cut filters. There are eight mono auxiliary sends. There are two insert points per channel, one which allows you to easily patch in an external mic preamp, the other of which is in the channel path in the conventional place, between the mic preamp and equalizer. Fader and mute automation will be available in groups of eight, and the channel LED meters can be optionally replaced with mechanical VU meters for the real classic look. I’m not into retro for retro’s sake, however, personally, I find VU meters to be very useful for setting and monitoring levels. The basic 16-channel version goes out the door for about $18,000. Processors & Interfaces Pulse Techniques introduced an accurate recreation of the Pultec EQ-1A3 equalizer a couple of years ago. This was a modernized version of the original EQ-1A, but in response to popular demand, Steve Jackson, “Mr. 21st Century Pultec” is now building an EQ-1A re-creation. The two equalizers use the same circuit and components, just different packaging – the EQ-1A is in a 3 rack space What’s really new and in the making, but not ready to crow about yet is a product that Pultec never made, but still was part of 1950s recording legacy. The Pultec MB-1 mic preamp was a staple in studios back in the day, and Bob Fine (Fine Recording) modified one with an electro-optical attenuator to provide limiting action. Bob’s son Tom, quite an audio historian himself, offered up his Dad’s notebooks and drawings to Pulse Techniques to re-create Bob Fine’s preamp which was his “secret sauce.” Dante seems to be leading the pack for audio over Ethernet, and Focusrite has augmented their RedNet Dante analog, MADI, AES3, and ProTools HD interfaces with an eye toward live sound applications, shrinking them from two rack spaces to one, and adding more channels, redundant power supply and redundant Ethernet connection. The RedNet MP8R, developed from the RedNet 4, is a remote controlled 8-channel mic/line amplifier with A/D converter and Dante connectivity. . The D16R provides 16 AES3 I/O channels on DB-25 connectors with one channel pair swappable to coax S/PDIF for interface with consumer devices such as CD players. One channel pair is duplicated on XLR connectors for convenience in connecting visiting gear. The D64R is a 64-channel interface between MADI and Dante. Both optical and coax bi-directional interfaces are provided for flexible connectivity to consoles or remote stage boxes. The HD32R puts a Pro Tools HD system on to a Dante network. Radial Engineering’s recent takeover of Jensen Transformers brought several new analog signal isolators to the product line. Radial has also acquired the Hafler product line with their initial products being headphone amplifiers, though they’ll be reviving a couple of the Hafler power amplifiers which were popular in studios back in the 1990s. Radial’s new Space Heater is an eight channel summing mixer with a tube “character generator” (their words, not mine, but I think it’s a good description). Each channel has half of a 12AX7 with input and output levels for controlling overdrive distortion, plus a switch to set the plate voltage to 150 volts (normal operation), 75 volts (crunchy) or a truly starved 25 volts (flat out distorted). Each channel has an input transformer that’s small enough to saturate easily for achieving the vintage transformer character. The tube circuit can be bypassed if all you need is the transformer sound, or you can go all out for stomach churning distortion. Antelope Audio introduced three new products at the show, a 32-channel mic preamp, a monitor controller, The MP32 puts 32 mic preamps in a two rack space box. It’s pretty Spartan, just a panel with 32 XLR Combo jacks on the front and four DB-25 connectors on the rear, plus a USB connector for the computer interface where all the controls are located. The mic preamp circuit is the same as is used in the Zen mult-channel USB interface (which is coming soon with a Thunderbolt port. Channels 1-4 can be switched to high impedance instrument level DI inputs, and all channels can be individually switched between mic and line (on the ¼” jacks) level, and phantom power, The preamps have a gain range of 15 to 68 dB, adjustable in 1 dB steps. Channels 1-8 have a switchable 20 dB pad for the mic inputs. There are no low cut filters or polarity switching. Maximum output level is a whopping +27 dBu. There’s nothing interesting to see on the front panel, but here’s a shot of one column of the control panel software application showing a channel with and one without the Hi-Z instrument input. Antelope Audio's Satori monitor controller is probably the most complex of the lot, so it deserves a picture of its front panel since that’s where you’ll be doing most of the work once it’s set up in your system. Nominally it’s a monitor controller, but with a lot of extras. Most obvious is switching among several monitors, which can include headphones (there are four independent headphone amplifiers), with a common volume control and trimming for equal loudness. It’s also an 8-channel analog summing mixer, talkback to headphones, and a few other tricks. There are eight stereo inputs on an assortment of XLR, ¼” TRS, and DB-25 connectors which can be assigned to any of the four stereo monitor outputs. They also appear, summed, on another pair of analog outputs. There’s an input for a talkback mic and a bass-managed LFE output for a subwoofer. In addition to comprehensive left, right and mono monitoring, the Satori can also convert left/right stereo to mid and side components for the engineer who likes to process the two components separately. Five setups can be saved as presets via the app (screen shot at the right). The Antelope Pure 2 is a “mastering quality” D/A converter that also includes a pretty darn good A/D converter and headphone amplifier. This one’s more interesting to look at around back. Routing, controls, and connectors offer means to insert an analog processor into a digital mastering chain via AES, S/PDIF, and Toslink (optical) digital I/O in addition to the USB port. With eight word clock outputs as well as word clock and 10 MHz frequency standard inputs, the Pure 2 can serve as a master word clock when several digital devices within a system are passing signals among themselves. A high precision word clock was Antelope’s first product, and they’re still looking for ways to make clocking a digital device externally sound at least as good or better than the device’s internal clock. TASCAM introduced a new line of USB computer audio interfaces, the US-2x2, US-4x4, and US-16x08 with the numbers corresponding to the number of input and output streams. What a concept! The 16x08 has eight mic/line and eight line-only inputs, with two inputs on each of the interfaces doubling as high impedance instrument directs. All of the interfaces in this series include 5-pin MIDI IN and MIDI OUT ports. All of these interfaces record up to 24-bit 96 kHz sample rate audio. The 2x2 has a single headphone output with independent volume control, the 4x4 has two independent headphone amplifiers and comes with an AC There’s a lot of bang for not too many bucks here. The 2x2 is $200, the 4x4 is $250, and the 16x08 is $300. Pete’s Place, the commune of small builders of useful audio gear is always a good place to visit to see what’s new. First up is the Blast Pad. Unlike a pop screen that’s intended for reducing plosives when working close to a mic, this one is intended to protect a sensitive mic like a classic ribbon when used in a high SPL situation. It’s a dual screen with ports around the ring supporting the front and rear screens. It’s quite amazing. You can blow through it fairly hard and not feel anything on the other side. It wasn’t set up with a mic to hear what effect it had, if any, on the sound, but when you’re recording a kick drum or bass amp from a couple of inches away, something’s going to get through. $300, but cheaper than a ribbon replacement. Developed, designed, and hand crafted by Pan60. There are a few new versions of the A-Designs REDDI tube direct box. Not a lot of details yet as only prototypes were on display, but look for a two-channel version of the original REDDI and the Black REDDI with twice the gain of the original and circuitry to emulate the classic Ampeg SVT sound. There’s an 8-channel 500-series frame on the bench that has both XLR and ¼” TRS connectors for each module’s input and output as well as an 8-channel A/D converter which could turn it into a handy expansion unit for an audio interface with digital inputs. The Electrodyne Summing Station is a 16 channel analog summing unit. It also serves as a monitor switcher, calibrated output level control, headphone amplifier, cue system with talkback, fully balanced signal path, and transformer output. The Electrodyne console had one of those unique and sought-after sounds, and the Summing Station utilizes a trick from the original Electrodyne design that includes the output transformer in a feedback loop which, on this unit, is switch-selectable for “tight” or “warm” sonic character. Recorders TASCAM had three new little recorders, but not just the same old same old. The DR-10C series (there are two models) are really tiny recorders, a bit over an inch square, that, rather than include built-in microphones, are designed to use an external handheld. lavaliere, or headworn mic. The DR 10CL/CS, which comes in two versions to match Lectrosonics and Sennheiser wiring and connector types, connects in line between the mic and body pack, with optional adapters for Shure, Sony, and “more” mics. Karl Winkler of Lectrosonics told me “They got ours right.” Of course it’s not necessary to use this model with a transmitter or belt pack. If the mic doesn’t need to go anywhere but to the recorder, you can just plug it in and go. But if you’re recording a stage act, it may be useful to have a recording as a backup in the event that a wireless gets flaky. The DR-10X is more straightforward. It has a standard XLR and would typically be plugged into a mic used for ENG or interviewing. Phantom power from a single AAA battery from a device that claims a battery life of 8-10 hours? All of the recorders in this series use a micro SD card up to 32 GB for storage, and offer a dual recording mode which records a secondary track 10 dB lower than the primary track, so as long as there’s enough headroom in the mic and signal path to the A/D converter, you won’t have digital clipping. There’s a limiter, low cut filter, automatic level control, three step mic input sensitivity switch, and instant one-button recording. Recording is mono BWF, 24-bit 48 kHz sample rate. Though tiny, the OLED display is quite easy to read when you get close enough. The TASCAM DR-44WL and DR-22WL are more typical handheld recorders, four and two channels These new recorders include all of the usually expected features. The built-in mics are now shock mounted for less handling noise, there’s a dual recording mode with a lower level backup, full four track recording (DR-44WL), 16- or 24-bit recording at sample rates up to 96 kHz as well as several MP3 bit rates, XLR combo jacks for external mic/line inputs on the 44 with mini jacks on the 22. The 22 has a selector for several preset recording situations similar in concept to what’s common with digital cameras for quick setup in point-and-record situations. Too much to describe in detail here so check the websites. JoeCo continues to stuff more features and functionality into their single rack space multitrack BalckBox recorder. The new BBR1MP integrates 24 mic/line preamps into the compact unit. Gain is specified as 60 dB with a switchable 100 Hz low cut filter and soft limiter. Sensitivity (millivolts in for 0 dBFS at maximum gain) isn’t specified, but phantom power on each channel is available at the click of a button. There’s an accessory 2 rack space breakout panel to bring the mic inputs, which are on TASCAM–wired DB25 connectors, out to 24 XLR connectors. For still further flexibility, there’s a slot for an optional Dante or MADI I/O card. The rig will operate from 12 V DC, so you can take it on a field trip. Thinking ahead, unused channels can be turned off to extend the battery life. With all of these I/O options, you should be able to record to just about anything that has an audio output. Technical Sessions As has been the practice for the past few shows, the first day’s program is devoted to technical papers, tutorial workshops, and panels. The exhibits don’t open until the second day. I like this setup because once I get out on the show floor, I usually don’t leave for a couple of hours to sit in on a technical session. I had picked out four sessions for Thursday and managed to get to 2½ of them, plus half of one on Friday morning before the exhibit hall opened. As was the case last year, the program was heavy on loudness issues; standards, lack of standards, lack of adherence to standards, and measurement procedures. Last year, Bob Katz (Digital Domain Mastering) declared that with the iTunes volume management and Mastering for iTunes guidelines, the music loudness wars were over, so this year’s loudness theme focused on film and video. Thomas Lund (TC Electronic) participated in a study of a large number of films and venues in Europe and found that loudness was spread out over quite a wide range, that few theaters adhered to the official standard procedure for setting playback level, and that many films when played according to the standard setup, far exceeded comfortable, and even safe SPL. Bringing things back around to music, Katz expressed concern that while Apple seems to have a good system to manage loudness that encourages music to be mastered with usable dynamic range, the streaming services are buggering things up now. And of course outside of the audio engineering community, everyone thinks that music downloads are dead (hard media died long ago) and how streaming is how everyone gets his music dose. Another interesting session brought Dave Derr (Empirical Labs), Dave Hill (Crane Song) and Hutch Hutcheson (Manley, Neve, and now freelance) together to discuss the design of small signal amplification. Hill designs for cleanliness while on the other hand, Derr’s products all involve distortion of some kind. Hutch brought his strong knowledge of transformers into the mix to explain how they can make for clean or dirty designs, depending on what you want, how the transformer is designed, and how it’s implemented. What they all brought to the table was that the equations and component specifications and characteristics are a good starting point, but that listening to every design change is the right way to know if you’ve achieved your design goal. The Wrap This was really a great show for attendees, presenters, and exhibitors. At 15,400 registered attendees, this was the best attended west coast show in ten years. I’m not crazy about the location – hotels and food are expensive, but the LA convention center worked well. Noisy sessions and demo rooms were located so that they didn’t provide unwanted background music for panel discussions and paper presentations. The show floor was easy to navigate, it wasn’t too hot or too cool, and bathrooms and drinking fountains were plentiful. Free WiFi was pretty skimpy, though. They should do better with that. For me, the best product of the show was the TASCAM WiFi equipped recorders. It accomplishes what I’ve always wanted in a handheld recorder, and does it in a creative and flexible way. I don’t have a nominee for worst product of the show, but I’ll have to give a nod to the Cliff RM1 microphone as the ugliest mic of the show. We’ll be back in New York next Fall. Until then, happy recording. Mike Rivers is a longtime Harmony Central contributor with a deep background in recording and covering gear. If you were a regular reader of Recording Magazine in 1995-2000, you may have read his “Oops, Wrong Button” (their title, not his) monthly series of articles on a wide range of useful topics. Mike has also written several product reviews over the years, and has two books to his credit. Mike’s done his share of recording over the past 50 years including several years with a remote truck, as Gypsy Studio, primarily recording folk festivals for locally produced albums and radio broadcast. For more of Mike’s musings on gear and recording visit http://mikeriversaudio.wordpress.com.
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