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Mic processors and preamps
The microphone is the first element in the broadcast chain.
In our January issue, Trends in Technology looked at this leading element and provided some ideas and tips on making the right selection. Now that the selection has been made, what's next?
Connecting the studio mic to the on-air console or routing system is a simple task, but it's likely that you'll need to do more than that. While the console may have its own mic preamp, this preamp may not provide the quality you need or want. Using an external preamp may provide the extra edge you seek. But if you're going to add a separate preamp, consider that no two voices, and to a lesser degree, no two microphones are the same. Judicious use of mic processing can dramatically improve the sound and consistency of the voices on the air.
Mic preamps and mic processors are separate devices, but one is a subset of the other. All the mic processors I have found include a quality preamplifier. The preamp itself has an operational consideration too. While by definition a preamp adds gain to a signal, there are two approaches applied to a mic preamp design. One design makes the preamp as transparent as possible, using the lowest noise components and adding the least amount of color to the signal. The other design takes advantage of the electronics chosen to add a desirable color to the signal. One example is the tube-base mic preamp. Because most common tube designs favor even harmonics (whereas solid-state designs favor odd-harmonics), they have a unique sound, which is often described as warm. It is this sound that gives a tube preamp its aural quality, which some recordists highly covet.
With the increased use of centralized audio routing systems that use a core audio engine as the primary audio input, it may be necessary to send the mic source audio over a greater distance than a few feet. Because mic-level signals are so low, distributing mic-level audio is not a preferred method. This is another advantage to using a mic preamp.
The mic processor adds additional power to the preamp. Typical stages include a gate or expander, limiter, compressor, de-esser and equalization. Each stage can be used for a particular benefit. Just like an on-air processor, these stages of a mic processor each have their own operational benefits. To get the best use of an on-air processor you learn how each stage acts and interacts. The same is true with a mic processor. While the individual parameters may not have as many variables as an on-air processor, knowing how each stage functions will simplify the process of getting the ideal sound.
Vice president, chief engineer: Orban/CRL
“Of all the possible elements of a mic processor, which one stage can be used to reap the maximum benefit?”
If all the desired benefits could be provided by one stage, mic processors wouldn't have so many stages. Each stage contributes something unique to the final audio texture. However, if I were forced to choose one, it would be a gated compressor.
The need for EQ can be minimized by choosing a microphone with frequency response that's close to your desired EQ texture. Any competent mic preamp, used below clipping, is good enough for broadcast use. Adequate circuit headroom after the mic processor precludes the need for a peak limiter. However, no outside substitute can provide the consistency and punch of a compressor.
In an ideal world, the talent would work the mic so skillfully that no compression was required. But with all of the tasks that today's talent are expected to perform routinely, a compressor smoothes things out. Moreover, in high-energy formats, the mic chain compressor adds energy and excitement to the presentation by adding voice processing that is congruent with the processing that the recording and mastering engineers typically apply to the music that these formats play.
In any format, the mic chain's compressor provides consistent levels that make life easier on the final transmission audio processor. The transmission audio processor can be optimized for the format's music, and the mic processor's compressor can then be adjusted to complement the transmission processor's sound.
President: Omnia Audio
“What's the most common mistake made when setting up a mic processor?”
Setting up microphone processing for broadcast needs to be thought of in at least two contexts: On-air and production. There are distinct differences.
First and foremost, the main on-air processing needs to be set with respect to music, as that is usually the predominant content on the air. Because on-air processing can be set over a diverse wide range, from Bach to hard rock and anything in between, it doesn't make sense to just dial in a mic processor setting that worked at some legendary station 20 years ago. It's not possible to understand the intricacy of the former facility when working to achieve optimum performance in a new environment.
This is the key reason as to why it's critically important to set the on-air processing first and then work on the mic processing. Too often the mistaken desire to tweak the main processing is based on the combination of the last song played and subsequent jock break. Either the music will sound great but the jock doesn't cut through, or jock sounds good and the music is dull and lifeless.
Make the on-air processing sound good with music first, then lock it down and work your way back to the mic processor. Many times you'll find that the mic processor will need to be reset with respect to EQ and compression. Start with less processing/EQ and dial more in as needed. Don't work in the other direction.
Another common mistake is to set the processor based on preconceived notions. Set the processing for what sounds good, not based on how the dials are set at the station across the street. There many variables that will effect the on-air mic presentation. This is why there are not really any secret mic processor settings.
The most common mistake made in the production studio is to apply a stock setting for every user. Production facilities are set up to create whatever effected texture is required for the segment being produced. If an element is needed that replicates the on-air sound, then adjust the production mic processing to reproduce the air studio sound. If the content is to be used for an agency or multiple outside stations then adjustment should follow a more conservative path. In the end, the ears need to be the judge so that the processing is not overly done.
President: Aphex Systems
“What is your top tip for getting the best sound from a mic processor?”
It is hard to specify the top tip for getting the best sound from a mic processor because there are no ‘generic’ mic processors. Each one has its weaknesses and, hopefully, its strengths.
I believe that one of the most critical parts of a mic processing chain is the compression. I often see voice compressors set more like limiters — high compression ratios with fast attack and release characteristics.
Intelligibility is dependent on consonant recognition. Consonants, especially in languages such as English, typically have a high leading edge. If the processor crushes those edges there is a loss of intelligibility, so even if the voice is loud there is a lack of clarity. This effect is exacerbated with digital processors that use look-ahead processing.
I believe that a compressor with a lower ratio and soft knee is better. That way there is no dramatic pull back or sucking effect just above threshold. Attack times should be set so that they are slow enough to allow the natural percussives in the voice to pass through untouched or enhanced, but not so long as to create an overshoot that could overload the following input stage. Release time should be fast enough so that the signal has more in your face and a phatter presence without obvious distortion.
One very important thing to remember is that the loudness will come from your on-air processor. Do not expect your mic processor to do too much, otherwise your voice talent, your management and your listeners will complain.
Some mic processors have the option of changing the order of the various processing stages. In most cases the preset order will likely prove satisfactory, but experimentation may offer some unexpected results. Compressing an equalized sound has a different effect than equalizing a compressed sound.
Some manufacturers and dealers of mic processors and preamps
While the Resource Guide is far from a complete list, it should provide enough basic information to help you get started.
The Toolvox from Omnia features specialized compressor ratios and time constants coupled with a slow gain-riding AGC and a smart freeze. The de-esser uses Fast Fourier Transform (FFT) analysis to find and remove unwanted sibilance. The processor features three bands of parametric EQ, each with adjustable filters and shelving; built-in phase rotators and Trueverb from Waves; security features to prevent unauthorized tampering; an Ethernet connection for network control; a stereo output; and analog and digital signal outputs. The unit is set up via the Windows remote application. Presets are recalled with the front-panel controls. Presets can also be recalled across the Ethernet connection for use in other studios.
The Virtual Voice Processor (VVP) from IDT uses Fast Fourier Transform (FFT) in DSP. The FFT analyzes the signal, concentrates on the frequencies of the voice adding the effects that have been programmed. A PC can be used to customize presets via IP. All the parameters can be stored on a compact flash card and shared with other units. There are 24 contact closure inputs as well. It uses an internal sample rate of 96kHz at 24 bits. It features analog and digital inputs and outputs. Digital audio connections are selectable for 16-, 18-, 20- or 24-bit resolution at 32-, 44.1-, 48- or 96kHz.
+33 472 1819 20
The Crane Song Flamingo is a two-channel, discrete class-A microphone preamplifier. It can be operated as a transparent amplifier or it can be used to emulate vintage sounds or distinctive new ones. Gain is adjustable in 6dB increments from up to 66dB of gain. Each channel has independently switchable phase and phantom power. The 22-element VU meter with overload indicator shows levels. The Iron switch increases harmonic content on low frequencies, while the Sound switch adds second and third harmonics across the spectrum. The unit has transformerless balanced inputs and outputs.
The Apogee Electronics Trak2 begins with a discrete microphone preamplifier that is accessible via rear-panel XLRs or the front-panel XLR/TRS connectors. An insert point is included. The analog signal is converted to digital through a 24-bit converter that runs at 44.1, 48, 88.2 or 96kHz. The unit also includes Apogee's Soft Limit process to prevent digital overshoots and the Apogee Soft Saturate system, which simulates analog tape compression. Either feature can be switched in and out on one or both channels. Two Apogee Multimedia Bus (AMBus) slots are provided to add Pro Tools, ADAT, TDIF, SDIF-II, SSL Hiway and other output standards.
Also available: Mini-ME, Mini-MP
The Mackie Onyx 800R is an analog mic preamp with a 192kHz digital output. It uses the same low-noise XDR mic preamp that is used in Mackie's compact mixers. Each of the eight inputs has a variable mic input impedance control to tune each preamp to its connected mic. AES-3, ADAT lightpipe and S/PDIF outputs are provided. Each channel has a front-panel mic/line selector switch; two front-panel instrument input jacks; and individual low cut, phase and phantom power controls.
The Aphex Systems 1100 MKII builds on the 1100 and offers a wider feature set and lower noise floor. The unit has a wide dynamic range microphone preamplifier that features patented technology in a discrete class-A tube design with an integral 24-bit 192kHz A/D converter. It boasts an EIN of better than -135dBu. Connections include a stereo, optical S/PDIF and AES-3 digital audio output, and separate ¼" insert point jacks. The Mic Lim optical attenuator, located directly on the mic input line, limits the microphone's output signal according to the Mic Lim peak detector's control current, detecting the preamplifier's output signal and instructing the input attenuator to proportionately reduce the microphone's output level just enough to prevent clipping.
Also available: 207 dual-channel preamp, 1788 eight-channel preamp
The Aircorp Pro-Announcer 500PH features remote-adjustable input levels, providing equalization without further adjustments. The compressor/expander combination reduces room noise and equipment noise, while providing level control and increased loudness for the announcer. The three-section variable boost and cut equalization allows for easy setup without increasing low frequency room rumble and system hiss. Other features include symmetry correction, dynamic control coupling of the compressor and expander to eliminate flanging effects, a popless insert point for an effects device, simultaneous mic-level output and line-level output to feed the console and telephone hybrid, a remotely controllable de-esser, a buffered headphone jack for setup without being on air, a DB-25 for all logic functions and 48vdc phantom power.
The Behringer VX2496 is based on the VX2000 and provides 24-bit/96kHz performance. It features an AES-3 output and features an opto compressor, dynamic enhancer, expander, de-esser and tube simulation. The output sampling rate is adjustable and can be tied to a clock reference. There is a discrete ultra low-noise mic/line input stage with soft mute, 48vdc phantom power. The tube emulation circuitry provides tube and tape saturation sounds, and the true RMS expander offers smooth noise reduction. An opto compressor provides dynamic control and creative signal processing options. A voice-optimized equalizer was specially designed for voice enhancement.
Also available: Ultragain Mic 100, Ultragain Mic 200, Ultragain Pro Mic 2200
The Presonus Digimax 96k has class A discrete input buffers followed by a dual-servo gain stage to provide 60dB of gain with 52dB of headroom. It is electronically balanced and features phase reverse on the first two channels, as well as a 20dB pad and selectable 48V phantom power on each channel. The EQ Enhance contours the EQ curve. It uses RMS compression and peak detection to limit transients. It has eight XLR mic inputs, two 1/4" instrument inputs, eight balanced TRS analog outputs, a 24-bit ADAT lightpipe output and four stereo S/PDIF digital outputs. All outputs can be used simultaneously.
The Symetrix 628 digital voice processor uses a transformerless preamp with 20-bit A/D and D/A converters and includes a de-esser, expander/gate, compressor and parametric equalizer into a 1RU package. It stores as many as 128 processing presets with eight factory presets included. Features include independent metering of processing functions, AES-3 or S/PDIF digital output, microphone and line-level inputs and an optional remote preset controller. Remote control is via MIDI. Three seven-segment LEDs display all parameter values and preset numbers. An output level meter continuously monitors the output. Digital sample rates of 48-, 44.1- and 32kHz are selected by a rear-panel switch.
Also available: 528E and Airtools 6200
The LA Audio MPX10 features a mic or line preamplifier with DI input, downward expander, auto compressor and equalizer in a 1RU package. It includes an ultra low-noise mic pre-amp, 48V phantom power, phase reverse switch and a 75Hz high-pass filter. The compressor section includes a variable threshold and ratio with a four-mode auto-sensing attack and release. A de-esser is also included. The EQ section features two variable parametric mid frequencies with variable bandwidth and fixed HF and LF cut or boost. The meter section provides an output gain control and output level and gain reduction metering.
Also available: PS10 and MLX20
+44 20 8418 0778
The Yellowtec VIP Digital is a DSP-based mic processor with two mic inputs and 48V phantom power. The mic preamp uses a 24-bit converter and features low latency. Processing stages include a compressor, expander, AGC with freeze function, a de-esser that uses FFT analysis, parametric EQ, built-in phase rotators, VIP Verb reverb, an audio delay line and a subsonic filter. The Sound Control Software allows for drag-and-drop selection of any of the processing elements in any order. GPI and GPO provide remote control functions. Presets can be recalled from a Smartcard inserted into the front via the front-panel controls.
+49 2173 967336
The TC Electronic Gold Channel is a digitally enhanced microphone preamplifier and a signal refinement toolbox. It uses 24-bit A/D converters, and features DSP control of expander, compressor, equalizer, de-esser and an M-S encoder/decoder. As many as 100 user presets can be stored. The I/O includes an analog mic or line-level input, an analog line-level output, AES-3 input and output, S/PDIF input and output, optical S/PDIF or ADAT input and output, a word clock input and MIDI in/out/through. All audio outputs are available simultaneously. A PCMCIA card can store settings.
The DBX 286A has a mic preamp and five processors that can be used independently or in any combination. Switchable 48V phantom power, an 80Hz high-pass filter, the DBX Over-Easy compressor, a frequency-tunable de-esser, a high-frequency enhancer, low-frequency detail control, and an expander/gate are included in the feature set. Levels are shown on the meter and status LEDs. A floating, balanced XLR input accepts balanced or unbalanced inputs. An additional ¼" TRS phone jack can accept balanced or unbalanced line-level signals. An insert jack is available between mic preamp and signal processing section.
Also available: Pro Vocal, 786 and Mini-Pre
The Sonifex RB-DMA2 dual digital microphone amplifier consists of two independent, low-noise mic pre-amplifiers with AES-3 and S/PDIF digital outputs and analog outputs. The unit can be used as two independent mic amps or one mic input can be copied to both channels of the digital output. Inputs are electronically balanced. The input gain for each input is adjusted on the front panel. Each channel has a selectable high-pass filter and 48V phantom power. A TTL word clock sync input is also provided. The EIN is 130dB. A tri-color LED indicates the audio level.
Also available: RB-MA1 and RB-MA2
The SBS Mic-IT is a dual microphone preamplifier with multiple output modes. The line level outputs may be summed to create a two-input microphone mixer. A sum-and-difference mode provides an output for mid-side recording. Front-panel switches assign 48V phantom power, and a high-pass filter can be applied to one or both of the inputs. The unit is packaged in a metal case suited to mounting on rack shelves, beneath woodwork or inside rack cabinets. It features electronically balanced inputs and outputs, an EIN of -129dB, THD at 1kHz of 0.005 percent, and a maximum output level of +26dBu. Distributed in the United States by Broadcasters General Store.
Also available: Mic Lim-IT
The Great River Electronics MP-2 and MP-4 are two- and four-channel mic preamplifiers. Each channel is a transformer-coupled class A discrete solid-state design. All switches, relays and internal connectors have gold-plated contacts. Each channel has a rear-mounted, XLR, balanced input jack and a front-mounted ¼" high-impedance input jack. Each channel also has a 15dB pad, polarity control, 48V phantom power, an overload LED and a 24-position gain switch. The 1RU unit has an internal power supply. Frequency response is from 10Hz to 30kHz, ±1dB for the mic input. - Also available: ME-1NV and MP-2NV
The ATI ML200 is a compact and portable dual mic preamplifier. The outputs are servo balanced line outputs capable of delivering +22dBm. The front panel has two pushbutton switches for each channel: one selects the preamp gain range and the other selects phantom power. The front panel also has an peak-level LED indicator. The ML200 has two female XLR balanced inputs and two male XLR balanced outputs. The mic input EIN is -124dBm. The unit operates on 22Vdc to 30Vdc. Measuring 1.75" H × 5.6" W × 5.75" D it weighs 1.25 lbs. As many as three units can be mounted side-by-side in a 1RU space.
Also available: MMA800-XLR, MLA800-XLR, M100, ML200, DMA103 and System 10K.
True Systems offers the P2 Analog, which has two mic inputs and two instrument direct inputs. It features a M-S (mid-side) decoder, stereo phase correlation display, selectable high-pass filters, relay-switched signal routing and dual gain range. The unit's frequency response is from 1.5Hz to 500kHz (-3dB). The maximum output level is +31dBu with an EIN of -132dB. The THD at +26dBu is 0.0008 percent. True Systems products are distributed in the United States by Neumann USA.
Also available: Precision 8
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