Friday, May 15, 2015
API Vision Attracts Business To 25th Street Recording
Analog 64 channel API Vision console still impressing clients and engineers after six successful years.
Oakland, California’s 25th Street Recording opened their doors with an all-discrete, fully-analog, 64-channel API Vision Console six years ago. Today, the console is still keeping 25th Street booked and busy. Among those who have called the studio home are all-star producer Joe Chiccarelli, when he is in town, ‘80s icon Sheila E., Norteño mega-band Los Tigres Del Norte, and guitar hero Joe Satriani. Twenty-fifth Street draws in high-profile clients with the help of the well-earned reputation and versatile capabilities of the API Vision.
Studio manager John Schimpf says there’s one thing his diverse clients all have in common: “Everyone who comes through here loves the API Vision. In this business, people don’t hesitate to complain, but I never get complaints about the Vision.” Schimpf and studio owner Dave Lichtenstein were drawn to the console in part because of API’s sonic heritage – which appeals to artists and engineers in equal measure.
John Cuniberti, Satriani’s long-time engineer says he has nostalgia for the API sound: “I grew up on vintage API consoles. The Vision is a zero-compromise, full-blooded embodiment of the classic API sound,” a sound which he believes “strikes the perfect balance between ‘organic personality’ and ‘high-fidelity’.” During his time at 25th Street, Cuniberti explains that the Vision was a “critical component” of the overdub process. “The Vision was the tool that effectively told us whether we were done with a song or not.”
Part of Schimpf’s confidence is due to the Vision’s flexibility. Schimpf asserts that the console, “really shines when we have big input sessions,” and believes that flexibility is welcoming to everyone who wants to use it, regardless of their experience level. “Engineers are free to create a custom signal flow, but the default configuration is straightforward and intuitive.” That was put to the test with Los Tigres’ latest recording Realidades.
The band’s Grammy-winning engineer Alfonso Rodenas says “When we recorded at 25th Street Recording, I used its patch bay to move EQs and preamps all over the place. It integrated beautifully with 25th Street’s outboard gear and allowed me to easily print the effects I wanted to print – and to hear, but not print – the effects that I didn’t want to commit to. It helped to create the beautiful sound of Realidades.” Rodenas continues, “I love the easy workflow of the Vision. It contributed to the inspired atmosphere in which the band performed at its best.”
Schimpf is looking forward to many more years with the Vision. “The audio quality is excellent,” he added. “It’s an amazing piece of sonic equipment. In all, the API Vision is an inspiring instrument that fosters creativity and delivers solid-sounding recordings.”
To hear music you needed to go to a live performance. Eventually sheet music was printed and available to buy. If you liked the song, you bought the sheet music … if you were lucky enough to have an instrument and could play you could actually hear the song.
The piano (or other musical instrument) was an important part of home entertainment.
In the late 1800s, Thomas Edison developed the idea of moving a piece of tin foil under a needle that was attached to something like a stretched out balloon. When he spoke into the piece of balloon, the attached needle vibrated and those vibrations were stored in the sheet of tin that he moved under the needle. He invented the Phonograph and in 1887 formed the first record label, selling records that were cylinders with sound scratched along the outside, played on a hand-cranked device.
Emile Berliner (who had a hand in the invention of the microphone) patented a flat disk system that was better than the tube, called the Gramophone. His system eventually incorporated a spinning flat disk with sound scratched in a spiral played back on systems with needles connected to stretched-balloon type membranes that were themselves connected to large open flaring horns (like a tuba) to help the sound waves radiate out in a single direction with extra resonance from the horn itself. The system was crank-wound, and elaborate springs and gears would then spin the disc at a constant speed.
Eventually, sound was being captured by microphones and stored magnetically on steel wire magnetic recorders, which used spools of wire that would follow a path from a “supply” spool to a “take up” spool, passing a record head that stored the magnetic sounds onto the wire and a play head that could read the magnetic signals back from the wire. This was accomplished using electro-magnetic transducers rather than early technologies that utilized physical transfer of sound energy.
Steel wire recorders were developed using technology that was first proposed in the late 1870s, and were used at times to send secret messages (for example, in a shipment of piano wire).
In the 1950s, oxide-based magnetic tape replace steel wire as the material to store magnetic signals onto. Tape used magnetically sensitive particles glued to a wide piece of plastic, which allowed for more focused and controllable recording and eventually the ability to record multiple bands (tracks) rather than a single sound.
Magnetic tape recorders utilized many of the same features as steel wire recorders, including supply reel, take-up reel, record head, playback head, and tape path.
Magnetic tape can be saturated, which means that if the tape is overloaded it will compress the sound. Magnetic tape has a “hiss” on playback. Different methods used to reduce the noise include dbx and Dolby, which crank up the hiss while recording and then drop it back down when playing back (reducing the tape hiss along with the “cranked” hiss). “Single-ended noise reduction” devices do not change the recorded sound, but rather will gate the high frequencies on playback.
Les Paul (yes, the Les Paul) created a magnetic tape recorder with a sync head (a record head with limited playback abilities). This was the invention of sound on sound recording. Previously, if you were listening back to something that was playing back and recorded something new, the new would not be at the same physical point along the tape because the playback head and the record head were in two different physical locations. The sync head meant that you could listen back to sound and record new sounds at exactly the same point along the tape.
Suddenly, we have multi-track recording, allowing people to record up to eight different tracks individually and listen to the tracks on newly developed equipment called mixers that controlled the volume of the tracks both going into and out of the machine, and also mixed those sounds together at those controlled volumes.
Suddenly, you could re-perform one part on one instrument rather than be forced to re-perform the entire song with all the musicians. You could even erase and replace small parts of individual tracks rather than have to re-do everything, as long as the sounds were separated enough when you recorded them that each track contained only the sound of one instrument. Replacing small parts involved going into and out of record at specific times, which was called punching in and punching out.
Isolating the instruments, even if it was just using a separate microphone for everyone rather than a single common microphone, had other benefits. By raising or lowering the volume of the microphones it was possible to either enhance or in some cases create dynamic interaction between the instruments.
Tom Dowd used this possibility to further the expressive nature of the music he was recording and mixing, and was the first modern recording engineer.
Engineers used to be only technicians who told musicians and producers what they could not do in order to make sure that records had usable grooves that would allow a needle to properly play without skipping or jumping. There were no moving mics, no riding levels while recording, nothing but documenting whatever happened to happen in the room with the microphones in it.
When a producer was forced to use Tom Dowd because his usual engineer was booked, he was able to make more expressive music as a result of having an engineer that would manipulate the equipment as required by the music, rather than change the music to fit the equipment limitations. This was the beginning of Tom using technology to enhance the creativity of the music he was working on.
He isolated instruments for better control later. He pioneered the fader console, with devices to control sound such as EQs to change tone or limiters to control volume built into the console channels. He was a musician as well as a tech, so everything he did served the music. He was also a nuclear physicist involved in the development of the atomic bomb when he was young (in fact since the work he did was top secret and could not be discussed in schools or industry he did not pursue a physics career after the army as planned but went back into music).
Tom built relationships with the people he worked with, overcoming the typical expectation that an engineer was just a technician without creativity. He was able to do this in different musical styles, and he became instrumental in very important music with pivotal artists throughout the years.
Tom Dowd brought out the best out of the people, the songs, and the sounds.
Overdubbing means adding new parts to pre-recorded ones. This meant that it was no longer necessary for all musicians to play at the same time. A band could record one day and the singer could record the next day. Since the singer was now on a separate track, the singer could continue to re-perform the song and re-record the track until they were satisfied with their performance.
Now that recordings were an artificial combination of sounds rather than capturing a natural music occurrence, you had to mix the sounds together to simulate either a natural sound environment or even to create a new sound environment. Normal dynamics that would take place between people performing music together had to be simulated, because now the people were performing at different times, or even in different locations.
Once music was stored on tape, people started to edit, which means to cut it up and move whole sections or individual parts around. Mono tapes were edited long before multi-tracks, but with multi-track recording it became possible for new tracks to be either newly recorded or flown (played from another tape machine) from other performances.
Things moved on fairly the same for a while, recording and overdubbing microphones and sound generating instruments onto individual tracks of magnetic tape recorders and then mixing those tracks through the separate channels of a mixing board into a cohesive combined sound.
Then came digital.
Digital tape recorders (DTRs) first looked and operated like analog magnetic tape recorders, with a supply reel, tape path, take-up reel, etc. Since DTRs recorded digital information onto the tape rather than actual magnetic signals it was only a matter of time before computer technology allowed you to record without the tape, directly to a computer hard drive.
Pro Tools was the first vitual digital tape recorder (meaning it was tapeless) that existed completely within a computer. Early Pro Tools was very limited in quality and capabilities, but with the introduction of non-destructive editing, music production was changed forever.
Now music could be edited with a click of a mouse instead of a flick of a razor blade. And you now have undo. That’s right, undo in an industry that had always involved permanent decisions with physical tape and razors rather than backed up computer files.
Once digital audio was in a computer rather than a tape machine, it was easy to start to manipulate it. Moving, quantizing, replacing and harmonizing sounds became as easy as clicking on a button. Auto-Tune (a program that fixes out-of-tune vocals) is responsible for many of the “in tune” vocals heard today. Before harmonizers and Auto-Tune, you actually had to be able to sing in order to be a singer. Now you only need to look and dance well and the music part can be fixed automatically. Click.
Original Pro Tools systems cost tens of thousands of dollars. These days you can get much more powerful systems that actually work well in home computers for hundreds.
Now everyone with a home computer is an artist/musician/producer/engineer. The age of the “prosumer” is here.
BAE Audio Announces Latest Member Of 500 Series Family
The BAE 1073 sound is now available in a diminutive package
Building on its tradition of creating vintage preamplifiers faithful to the original design specifications, BAE Audio announces the newest family member among its 1073 series: the 1073MPL.
The 1073MPL is the result of more than two years of research and development, making the sound of the 1073 available in a small package. The new 1073MPL delivers the same class A sound that BAE preamplifiers are known for, all in a compact and cost effective 500 series package.
For equipment manufacturers, designing and building 500 series modules is often ripe with challenges, given the significantly reduced physical space. As a result, compromises are often made with componentry, wiring and circuitry. This is not the case with the 1073MPL, which is hand-wired and assembled in California using the same Carnhill/St Ives transformers that BAE is known for. This module lives up to the reputation of its larger, rackmounted brethren and delivers authenticity down to the last detail, from its original Marconi knobs to its sonic signature across the frequency range.
“Despite its small footprint, this unit is packed with features and delivers an immediately recognizable sound,” commented Mark Loughman, BAE Audio president. “It is also very versatile — its impedance switching capability and high impedance DI make it useful for use with a range of different microphones, as well as guitars, synths and other instruments.”
BAE Audio has been building the 1073 for 15 years. While the company continues to innovate with products like the 1073MPL, it takes a ‘no-compromise’ approach to build and manufacturing, with each of its products meeting the company’s exacting standards of quality and authenticity.
Similar to the rest of the BAE 1073 line, the 1073MPL contains a trinity of core elements helping it produce the ‘vintage’ sound that is associated with classic recordings of the 1960s and 70s. These include the aforementioned Carnhill transformers — imported from England and hand assembled at the company’s plant in North Hollywood. Loughman says that the company strives to educate consumers at every juncture on the importance of insisting on these components, “otherwise, corners have been cut,” he says.
Frequency Response: 10Hz to -3dB at 55kHz
Line Input Impedance: 10k ohms
Output Impedance: 65 ohms
Common Mode Rejection Ratio: 100dB min @ 60Hz
Maximum Output Level: +27.4 dBu @ 600Ω
Gain dB: 0 to 71 dB
The 1073MPL is available now through BAE Audio’s network of authorized dealers, and carries a retail price of $899.
A handy guide to greatly reduce the likelihood of hum and radio frequency interference (RFI) in your studio system
You patch in a piece of audio equipment, and there it is: HUM!
This annoying sound is a common occurrence in sound systems. Hum is an unwanted 60 Hz tone—50 Hz in Europe—maybe with harmonics. If the harmonics are especially strong, the hum becomes an edgy buzz.
Your sound system also might be plagued by RFI (Radio Frequency Interference). It’s heard as buzzing, clicks, radio programs, or “hash” in the audio signal.
RFI is caused by CB transmitters, computers, lightning, radar, radio and TV transmitters, industrial machines, cell phones, auto ignitions, stage lighting, and other sources. This article looks at some causes and cures of hum and RFI. Following these suggestions goes a long way in keeping your audio clean.
Hum And Cables
One cause of hum is audio cables picking up magnetic and electrostatic hum fields radiated by power wiring in the walls of a room. Magnetic hum fields can couple by magnetic induction to audio cables, and electrostatic hum fields can couple capacitively to audio cables. Magnetic hum fields are directional and electrostatic hum fields are not.
Most audio cables are made of one or two insulated conductors (wires) surrounded by a fine-wire mesh shield that reduces electrostatically induced hum. The shield drains induced hum signals to ground when the cable is plugged in. Outside the shield is a plastic or rubber insulating jacket.
Cables are either balanced or unbalanced. A balanced line is a cable that uses two conductors to carry the signal, surrounded by a shield (Figure 1). On each end of the cable is an XLR (3-pin pro audio) connector or TRS (tip-ring-sleeve) phone plug.
Figure 1. A 2-conductor shielded, balanced line.
Each conductor has equal impedance to ground, and they are twisted together so they occupy about the same position in space on the average.
Hum fields from power wiring radiate into each conductor equally, generating equal hum signals on the two conductors (more so if they are a twisted pair). Those two hum signals cancel out at the input of your mixer, because it senses the difference in voltage between those two conductors—which is zero volts if the two hum signals are equal. That’s why balanced cables tend to pick up little or no hum.
An unbalanced line has a single conductor surrounded by a shield (Figure 2). At each end of the cable is a phone plug or RCA (phono) plug. The central conductor and the shield both carry the signal.
They are at different impedances to ground, so they pick up different amounts of hum from nearby power wiring. There’s a relatively big hum signal between hot and ground that results in more hum than you get with a balanced line of the same length.
Figure 2. A 1-conductor shielded, unbalanced line.
Sometimes it’s impossible to avoid long unbalanced cables, and some cables used between pieces of equipment are unbalanced. An unbalanced line less than 10 feet long usually does not pick up enough hum to be a problem.
Wherever you can, use balanced cables going into balanced equipment. Keep unbalanced cables as short as possible (but long enough so that you can service them). Check inside cable connectors to make sure that the shield and conductors are soldered to the connector terminals. Route mic cables and patch cords away from power cords; separate them vertically where they cross. This prevents the power cords from inducing hum into the mic cables.
Also keep audio equipment and cables away from computer monitors, power amplifiers, lighting dimmers and power transformers.
Another major cause of hum is a ground loop: a circuit made of ground wires.
It can occur when two pieces of equipment are connected to the building’s safety ground through their power cords, and also are connected to each other through a cable shield (Figure 3).
The ground voltage may be slightly different at each piece of equipment, so a 50- or 60-Hz hum signal flows between the components along the cable shield. It becomes audible as hum.
Also, the cable shield/safety ground loops acts like a big antenna, picking up radiated hum fields from power wiring. For example, suppose your mixer’s power cord is plugged into a nearby AC outlet.
The musicians amps are plugged into outlets on stage. So the mixer and amps are probably fed by two different circuit breakers at two different ground voltages.
When you connect an audio cable between the mixer and power amps, you create a ground loop and hear hum. To prevent ground loops, plug all audio equipment into outlet strips powered by the same breaker. (Make sure the breaker can handle the current requirements).
Figure 3. A ground loop.
Run a thick AC extension cord from the stage outlets to the mixer, and plug the mixer’s power cord into that extension cord. That way, the separated equipment chassis will tend to be at the same ground voltage—there will be very little voltage difference between chassis to generate a hum signal in the shield.
Caution: Some people try to prevent ground loops by putting a 3-to-2 safety ground lifter on the AC power cords. Never Do That. It creates a serious safety hazard.
If the chassis of a component becomes accidentally shorted to a hot conductor in its power cord, and someone touches that chassis, the AC current will flow through that person rather than to the safety ground. Lift the shield in the receiving end of the signal cable instead, and plug all equipment into 3-pin grounded AC outlets.
Figure 4. Lifting the shield from the pin-1 ground in a male XLR connector.
Let’s explain the signal ground lift in more detail. The hum current in a ground loop flows in the audio cable shield, and can induce a hum signal in the signal conductors.
You can cut the audio cable shield at one end to stop the flow of hum current. The shield is still grounded at the other end of the cable, and the signal still flows through the two audio leads inside the cable.
So, to break up a ground loop, disconnect the cable shield from pin 1 in line-level balanced cables at the male XLR end (Figure 4). You can either cut the shield, or plug in an inline audio cable ground-lift adapter.
Removing the shield connection at one end of the audio cable makes the connection sensitive to radio-frequency interference (RFI). So solder a 100 pF capacitor between the shield and XLR pin 1 (Figure 5). This effectively shorts RFI to ground, but is an open circuit for hum frequencies.
Figure 5. Supplementing the lifted shield with a capacitor prevents RFI.
Some engineers create a partial ground lift by placing a 100 ohm resistor between the cable shield and male XLR pin 1 (Figure 6. next page). This limits the current passing through the cable shield but still provides a good ground connection.
Label the XLR connector “GND LIFT” so you don’t use the cable where it’s not needed. For example, mic cables must have the shield tied to pin 1 on both ends of the cable. The ground lift is only for line-level cables.
Here’s another way to prevent a ground loop when connecting two balanced or unbalanced devices. Connect between them a 1:1 isolation transformer or hum eliminator.
Even if your system is wired properly, hum or RFI may appear when you make a connection. Follow these tips to stop the problem:
Unplug all equipment from each other. Start by listening just to the output of your studio monitors PA speakers. Connect a component to the system one at a time, and see when the hum starts.
Remove audio cables from your devices and listen to each device by itself. It may be defective.
Partly turn down the volume on the amps, and feed it a higher-level signal from your mixer (0 VU maximum).
Do not wire XLR pin 1 to the connector-shell lug because the shell can cause a ground loop if it touches grounded metal. If you are sure that the shell won’t touch metal, wire XLR pin 1 to the shell lug to prevent RFI.
Try another mic. Some dynamic mics have hum-bucking windings.
If you hear hum or buzz from an electric guitar, have the player move to a different location or aim in a different direction. Magnetic hum fields are directional, and moving or rotating the guitar pickup can reduce the coupling to those fields.
If the hum is coming from a direct box, flip its ground-lift switch.
Figure 6. A ground lift using a 100 ohm resistor and a 100 pF capacitor.
Turn down the high-frequency EQ on a buzzing bass guitar signal.
If you think that a specific cable is picking up RFI, wrap the cable several times around an RFI choke (available at Radio Shack or other electronics supply houses). Put the choke near the device that is receiving audio.
Install high-quality RFI filters in the AC power outlets. The cheap types available from local electronics shops are generally ineffective.
Connect cable shields directly to the equipment chassis instead of to XLR pin 1, or in addition to pin 1. Some equipment is designed this way to prevent the “pin 1 problem”. The cable shield should be grounded directly to the chassis - - not connected instead to a ground terminal on a circuit board inside the chassis.
Periodically clean connector contacts with Caig Labs DeoxIT, or at least unplug and plug them in several times.
By following all these tips, you can greatly reduce the likelihood of hum and RFI in your audio system. Good luck!
AES and SynAudCon member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.
Brentwood, Tennessee studio already making plans to expand console modules and automation
Grant Goddard is not new to the music industry, but his studio “Bikini Atoll Sound” recently got its first bit of API gear. In addition to owning Bikini Atoll, Goddard is also a professional sound designer. He saw a 1608 installed in his studio this spring, and he’s already looking forward to expanding it.
“The 1608 allows for better recordings, warmer sound, bigger, fuller mixes, and smoother work flow. I’m planning on more modules, another 16-channel chassis, and plan to add automation in the near future.” The 1608 was an easy choice for Goddard to make, but he did have some help from API dealer Robb Zenn at Alto Music in New York.
Goddard explains, “The 16-channel 1608 was the best choice for me as my goal was to start with a very high-quality professional console that offers the ability to expand as my studio, workload and budget increase. I was impressed with the API reputation for longevity and service, and needed a centerpiece for my studio that wouldn’t be outgrown any time soon.”
Goddard’s studio is located in Brentwood, Tennessee, so the 1608 has gotten to cut its teeth on “a handful of promising up-and-coming local Nashville bands such as Sky Temple Blues, Chris Firebaugh y Los Diablos en Fuegos, and Dane & the Aquatic.” While Goddard describes himself as a “lover of all types of music”, he says that the studio tends to record rock and blues the most. In the past, Goddard has worked with major artists across all genres, including Mos Def, Fergie, Slash, Smashing Pumpkins, and Kid Cudi, although he is “confident the 1608 would be instrumental in helping to elevate the overall quality of the productions at Bikini Atoll,” no matter who they have in the studio. The 1608 has its “obvious” advantages, according to Goddard, who defines those as “ease-of-use, reliability, clarity, and headroom, headroom, headroom.” Another hidden advantage he has noticed since the 1608 was installed is that “the console has brought a new energy, which has greatly inspired and enhanced our creativity at the studio.”
New Compact Powered And Ultra-Compact Mixers From Studiomaster At Prolight + Sound
The portable mixer series is aimed at gigging musicians, sub-mixing, small venue install and corporate and other applications
Studiomaster presented its new VISION series compact analogue mixers – in both powered and passive variants – for the first time in Europe at Prolight + Sound.
Vision is positioned immediately beneath the top of the range Horizon series, offering its larger sibling’s design and construction, operational and audio performance, in an even more compact, price competitive form.
VISION is available in unpowered (VISION 8) and 2x500W powered (VISION 1008) and 2x1000W (VISION 2008) variants, featuring lightweight Class-D amplification. All models feature 8 channels – 6 mic and 2 stereo – the same twin DSP effects as on the Horizon, and 9-band graphic EQ.
Introducing VISION, Studiomaster and Carlsbro assistant general manager / marketing manager, Patrick Almond said: “Like the Horizon series, VISION immediately distinguishes itself from the mass of small-form plastic mixers flooding the market, with superior build quality, design and performance. It comes in at a very attractive price point, well below Horizon but offering the same seriously pro quality, rich analogue sound and robust construction that will provide years of serious touring service.”
Also making its European debut ClubXS, is a price / spec competitive portable mixer series, aimed at gigging musicians, sub-mixing, small venue install, corporate and other enterprise applications is also debuted.
Two models, the XS8 and XS10, feature onboard FX, integrated USB / SD card stereo media player, USB / SD card recording from main mix, Bluetooth connection for playback from mobile or media devices, 60 mm faders, balanced XLR outputs, control room output, two sends and a stereo return. All mic input channels have 3-band EQ, hi-pass filter and built in compressor. Line input channels feature 2-band EQ; all channels feature channel mute and two Aux controls.
The XS8 has four mic and two stereo inputs, and the XS10, six mic and two stereo inputs. An internal switched-mode power supply and phantom power on all mic inputs are also featured.
Show Central: New Products & News From Prolight + Sound 2015
Welcome to ProSoundWeb’s coverage of the ongoing Prolight + Sound & Musikmesse 2015 in Frankfurt, which presents the latest products from the worlds of event technology and musical instruments.
Preliminary figures show a reported total of 2,257 exhibitors from 57 countries on hand this year, up from the 2,216 exhibitors reported in 2014.
While Musikmesse has remained stable in terms of exhibitor numbers despite a volatile market, Prolight + Sound, which is celebrating its 20th anniversary, reflects the buoyant mood of its sector and has set a new exhibitor record, with 928 companies showing products and services.
“The ups and downs of companies in the musical-instrument sector represent a distinct contrast to the booming event-technology industry. We aim to take individual account of the opportunities and challenges of these sectors and develop the two events accordingly,” states Detlef Braun, a member of the executive board of Messe Frankfurt.
The exhibitors include key players from the fields of sound and stage technology, audio-visual media technology, systems integration, and lighting. Prolight + Sound is also characterized by a high level of internationality: 61 percent of the exhibitors come from 41 foreign countries – after Germany, the most represented nations are China, Great Britain, the United States, the Netherlands, Taiwan, Spain and France.
We’ll be providing updates regularly here on the “show central” page, so be sure to check back often for the latest.
Audient Announces New iP14 Audio Interface At Prolight + Sound 2015
Bus-powered unit provides two channels of Audient console microphone preamps delivering 10 inputs/4 outputs with Burr-Brown AD converters
Audient has introduced the iD14, a new bus-powered audio interface offering two channels of Audient console microphone preamps delivering 10 inputs/4 outputs with Burr-Brown AD converters, housed in a desktop package.
“The Class A mic pres used are exactly what you’ll find in our stand-alone units and consoles, and the best you’ll find in any interface,” says Audient technical director Tom Waterman. “These, alongside the class-leading converters make iD14 an absolute audio powerhouse. You won’t find the audio performance of iD14 on any other audio interface in its price range.”
Proprietary ScrollControl turns the iD14’s volume encoder into a virtual scroll wheel, providing hands-on control. “With the touch of a button, you can adjust compatible DAW hosts, plug-in parameters, iD14’s mixer app and even scroll through your iTunes library and web browsers—just like you’re adjusting a piece of hardware,” Waterman explains.
When not using ScrollControl, the iD button can also be assigned to enable iD14’s console-style monitor control features, such as DIM, CUT, Polarity Reverse, Mono Sum, Talkback and Cue Mix monitoring, just as with the larger Audient iD22.
“The audio performance of iD14 is the best of any bus powered interface in its class,” Waterman adds. “We spent a long time optimizing the Burr-Brown converter layout, auditioned each building block during extended listening tests and paid very careful attention to the multi-stage regulated power supplies to match the sonics and performance of our acclaimed iD22 on bus power.”
“iD14 is one of the few audio interfaces in its price range that offers ADAT expandability,” he continues. “So adding external mic pres, such as the (new) ASP800 can give you up to 10 inputs—perfect if you need to record drums or if you simply need more channels.”
The iD14 will retail at $299 (MAP). Ship date will be announced soon.
As any electric string instrument player knows, there are a number of different types of pickups, and within each category there’s a tremendous variation in possible tone.
This excerpt from The Ultimate Guitar Tone Handbook explains the nine factors that affect how a pickup sounds. The next time you’re in the market for one, keep these in mind so you can better tailor the pickup to your needs.
Just like most things in life, something that seems so simple on the outside is very intricate on the inside and a pickup is no exception. Here are the numerous factors that contribute to a pickup’s sound.
1. The number of turns or winding. This is the number of turns of wire around the bobbin of the pickup. The more turns, the louder the pickup, but the worse the high-frequency response becomes. The number of turns is measured by the electronic resistance of the wire, which is measured in ohms. The higher the ohms value, the hotter the pickup but the less high-frequency response you’ll have. Humbucking pickups have more resistance than a single coil because there are more turns of wire, which is why they’re hotter and have less high end.
2. Type of wire used. The diameter and insulation determines the number of windings that can fit on a bobbin, which will determine the resistance, which determines the output, etc.
3. Type of winding method used. We’ll look at this a bit closer in a bit, but many of the pickups in the early days of the electric guitar were wound by hand, which meant that there were more or less than the required number of windings on the bobbin, and an uneven wind would also affect the capacitance of the pickup, which can cause a peak in the frequency response. This problem was virtually eliminated when manufacturers switched to machine winding, but while every pickup was now the same, some of the magic that occasionally came from a hand-wound pickup also disappeared.
4. The type of magnets used. Although Alnico (a blend of aluminum, nickel and cobalt) is the alloy of choice for most pickups, occasionally you’ll find pickups made of other materials such as ceramic or neodymium. This will affect the strength of the magnetic field.
5. The strength of the magnets used. Magnets used for pickups are categorized by strength on a scale of two to five with five being the strongest. A stronger magnet will produce a louder and brighter sound while a weaker one will produce one that’s warmer.
6. The magnet height. How close the individual magnets are to the strings will determine how loud that string is. On pickups that have adjustable pole pieces that’s not so much of a problem, but on pickups with fixed pole pieces (like a Fender Strat or Tele) that could cause a slight imbalance in the string output. As an example, prior to the late 1960s, most guitarists used a wound G string, so the fixed height of the magnets on a Strat were different to compensate.
7. Pickup cover. Metal covers on humbuckers can cause a resonance that results in feedback problems at high volumes. That’s why many of the early rockers removed their pickup covers, and why many guitars and pickups are sold that way today.
8. Pickup potting. Many pickups are sealed in wax to eliminate vibration induced signals that make a pickup microphonic. The heat from the hot wax can weaken the magnet though, thereby changing the pickup’s sound.
9. Potentiometers. Although not exactly a part of the pickup itself, the volume and tone pots are part of the electronic circuit along with the pickup and can affect the sound. The higher the resistance of the pot, the more high end will pass. Fenders use 250k ohm pots, Gibson uses 500k, and many other manufacturers use 1 Meg pots.
There are other factors such as winding direction, magnetic polarity, and the type of bobbins used, but their contribution to the final sound is subtle at best.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website, and go here to acquire a copy of The Ultimate Guitar Tone Handbook.
The MP8d follows the Zen Studio and MP32 mic preamp from Antelope Audio.
For Musikmesse 2015, Antelope Audio [hall 5.1, booth E41] introduces a new addition to its family of class-A mic preamps: the MP8d eight-channel mic preamp with A/D conversion.
During exhibition hours between April 15 and 18, Antelope Audio will showcase all the latest products and feature limited time show specials including a special 5% discount on any of its current portfolio, valid until May 15, 2015.
MP8d is an eight-channel class-A microphone preamplifier, which boasts Antelope Audio’s integrated A/D conversion. The signature sound is accomplished thanks to Antelope’s Acoustically Focused Clocking jitter management algorithm and Oven-Controlled crystal oscillator, along with analog and digital circuit board design, powered by a proprietary power supply.
“MP8d is the latest addition to Antelope’s expanding eco-system concept, with the goal of maintaining a stable and coherent audio signal throughout the entire recording, mixing and mastering process,” says Igor Levin, ceo and founder of Antelope Audio. “MP8d now adds a product to the Antelope lineup that provides clean and transparent microphone gain, with precise A/D conversion, along with an optimized analog drive stage to gently handle overloads, allowing the engineer to capture the purest sound quality, even when recording hot signal levels,” Levin adds.
The MP8d mic pres are combo XLR/TRS connectors, which accept both mic and line level signals. Hi-Z inputs are associated with channels 1 and 2, and are accessible via dedicated TRS connectors on the front panel. Inserts for connecting your favorite effects outboard gear are available on the rear panel. A headphone amp is accessible from the front panel to provide the convenience of integrated monitoring. The analog connectivity is accomplished with a D-Sub 25 connector directly outputting the mic signals at line level, making MP8d a companion device for Antelope’s Orion32 and Zen Studio audio interfaces.
Multiple digital connectors offer a wide variety of options. MP8d utilizes Antelope’s custom-built USB chip and PC/Mac drivers, allowing 24-bit/192 kHz streaming of 24 simultaneous I/O. In addition S/PDIF, AES/EBU, TOSLINK, ADAT and MADI connections expand the product compatibility and allow for connection to other outboard gear or DAW, as well as cascading two or more MP8d units, expanding the channel count.
MP8d relies on the software control concept developed for Orion32 and Zen Studio. The user interface with color-coded drag and drop signal routing, multiple mixers and powerful DSP engine with on-board effects, makes the device suited to various recording applications. Programmable presets allow for recall of preferred configurations, while ergonomic aluminum knobs on the front panel provide precise mic gain adjustment in real-time.
Brings experience in sales management and as a recording artist, engineer and studio owner to the position
Looking for a great position in pro audio? Seeking the most qualified candidates? It all starts with the ProSoundWeb Jobs Center.
Vintage King Audio has named Dan Scalpone as sales manager, working directly with company co-founder and sales director Mike Nehra.
“Vintage King’s track record has earned a lot of customer trust,” says Scalpone. “I plan to follow in that tradition and build on that foundation as we continue to grow. I never forget what it was like building a studio and being passionate about making music.”
With over a decade of experience in sales and sales management, Scalpone has worked in nearly every facet of the pro audio industry. As a composer, recording artist, engineer and studio owner of 18 years, he brings that knowledge to his new position at Vintage King.
Taking over the position of sales manager allows Scalpone to open up the company’s sales team to new opportunities. He states that he intends to start “by building on my background and giving our sales team some fresh perspectives on taking care of our customers and helping the company grow.”
Vintage King’s reputation has always been built on active participation in the pro audio community. Scalpone sees this as a big strength and looks to further develop the connection between engineers, producers, studio owners and recording artists.
SSL Announces Upgrades To Duality And AWS Studio Consoles
New platform unites automation in the analog console domain with DAW-based workflow
Solid State Logic has announced the release of new Duality δelta and AWS δelta consoles, marking the introduction of δelta-Control (δ-Ctrl), a new analog console automation platform for SSL studio technology that unites the automation of the analog domain with a DAW-based workflow.
All new Duality and AWS consoles will be δelta models, and console upgrades are available for all preceding models.
At the heart of δ-Ctrl is a native AAX/RTAS/AU/VST/VST3 plug-in that allows automation of the console as if it were a DAW plug-in. The automation system in the DAW is used to record and playback control data from the faders and switches on the console as an alternative to the legacy console automation system.
The δ-Ctrl plug-in is inserted into a DAW mixer audio channel, with the plug-in then receiving and sending control data from an assigned console channel, VCA or Master fader via a high speed Ethernet network connection. Audio on the DAW track passes through the plug-in slot unprocessed so the plug-in can be combined with other DAW plug-ins.
The console fader, mutes and relevant switches are represented as plug-in parameters in a δ-Ctrl plug-in GUI and their automation data is recorded to the automation lane of the selected DAW channel. In playback the plug-in converts the stored automation data from the DAW into δ-Ctrl messages and routes these back to the console via the SSL Logictivity Network connection.
Automation data can be viewed and edited as normal plug-in data in the DAW tracks. The console fader bbsolute and trim values are saved using the same dB law as your selected DAW automation so that the “Paste Special” command can be used to copy existing DAW fader data into the plug-in.
The Paste Special feature is available in Pro Tools and in most (but not all) DAWs. δ-Ctrl retains all the key operational elements of SSL’s Signature Mix System with features such as JOIN, REVISE, MOTORS OFF and SNAP Override all actioned from dedicated front panel switches. The MOTORS OFF mode offers the full non-moving-fader SSL VCA mix experience as preferred by some re-mixers. A new MOTORS OFF, Touch Write mode emulates the SSL G Series Mix system Immediate Pickup (IP) option.
Duality δelta also receives a visual refresh with an new style for its Channel Displays interface and a new graphic design-based Eyeconix library.
Audient ASP880 Enhances Producer Charlie Andrew’s Toolkit
Simplicity, size and sound quality make the ASP880 a favorite tool in his kit
Mercury Prize-winning producer Charlie Andrew, who rose to fame producing Alt-J’s first two albums has recently added an Audient ASP880 eight-channel mic pre to his setup.
Currently in his Brixton studio, working on an album with four-piece Mancunian band Money, he is using the new mic pre alongside his old ASP008 8-channel mic pre.
“I’m a big fan of simplicity,” continues Andrew. “The gain is basically pretty much all I put my hands on, and the phase buttons to make sure when I’m recording drums I can quite quickly flip phase and see if I’ve got phasing right.
“Having the analogue to digital converters is a huge help, too,” he adds. “We’re going to go to a pub to record one of the Money songs on an upright piano to get a bit more atmosphere. Having a 1U ‘interface’ with the mic pres and the converter, you can quite easily plug them into the back of something without having to lug a massive rig around. I’ll definitely be using the ASP880 for that. The mic pres are very transparent, very clear and very low noise – they’re just beautiful pieces of kit.”
With the latest albums from Marika Hackman and Sivu also credited to him, Charlie Andrew’s star is certainly in the ascendant. Audient wishes him the best of luck with it all.
Full Compass Announces Cristin Livezey As Vice President Of Finance & Procurement
Company veteran now managing purchasing department in addition to financial accounting
Looking for a great position in pro audio? Seeking the most qualified candidates? It all starts with the ProSoundWeb Jobs Center.
Full Compass Systems has announced that Cristin Livezey has expanded her role to vice president of finance and procurement; in addition to managing the company’s financial accounting, she will now manage the purchasing department as well.
Livezey joined Full Compass as a corporate controller in 2008 and helped to drive initiatives resulting in record sales. In 2010, she was promoted to VP of finance and has since been dedicated to deepening outside financial relationships, optimizing internal processes and controls, and managing cash flow to achieve profitable growth for the company.
“Combining these two departments will improve efficiencies and continue a high-level of support for our vendors,” states chief operating officer Doug Carnell. “Cristin’s new role is the result of careful consideration and our goal to continually improve service to both our customers and our vendors.”
Cymatic Audio Announces MADI Module Expansion For uTrack24
Add 24 channels of digital audio to your uTrack24 through optical or BNC coaxial connections
Audio interface developer Cymatic Audio announces that it is planning to release the uTrack24 MADI Module, a user-installable MADI (Multichannel Audio Digital Interface) option card for its 19-inch rack-mountable uTrack24 24-track recorder, player, and interface…
Cymatic Audio’s uTrack24 24-track recorder, player, and interface handles recording and audio playback without computer constraints. It records directly onto USB media plugged into the front panel. 24 three-colour LEDs show the level of each channel while the same LEDs can be used as a 24-segment level meter for one channel individually. An LCD screen shows additional information and allows editing of parameters via a push encoder. Lastly, large illuminated front panel-positioned transport controls help with recordings and overcoming onstage and off-stage lighting conditions.
Integration of the uTrack24’s multichannel audio I/O with MADI-based audio systems will soon be reality with the release of the uTrack24 MADI Module, making it possible to record 24 separate audio tracks from the output of MADI-equipped digital consoles and stage boxes. 24 channels of uTrack24-delivered pre-recorded digital audio can be output to a digital console’s inputs. Integration and interconnections are made through the uTrack24 MADI Module’s onboard optical in and out and (BNC) coaxial connectors. The uTrack24’s 24 analogue outputs (on standard 25-pin D-Sub connectors) can still remain active while using the uTrack24 MADI Module’s digital outputs.
Connect a uTrack24 MADI Module-equipped uTrack 24 to a MADI-equipped digital console for 24-channel virtual soundcheck playback or 24-channel soundcheck recordings — all at the push of an illuminated button.
The uTrack24 MADI Module will be shipping in Summer 2015 and available to buy from Cymatic Audio’s global network of dealers and retailers with projected pricing significantly lower than €400.00 EUR/$400.00 USD.
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