The 2Control with its unique Crossfeed control allows to judge a stereo panorama on headphones just like with loudspeakers. The unit's concept is consequently based upon duality: Two channels, two sources, two loudspeaker sets, two headphones—2Control.
The speakers and headphones monitoring center
The 2Control combines loudspeaker and headphone monitoring in a compact, user-friendly and high quality control device. Control near-field monitors, full-range speakers and headphone monitoring for two listeners from one central unit!
Analog volume control for your DAW
A majority of D/A converters and sound cards provide nothing in the way of analog level monitoring control, and this means the necessity of varying signal levels at the converter outputs. The result is a lowered bit rate in the monitoring signal, which can lead to commensurate loss of audio quality.
Two excellent headphone amplifiers
Friends, colleagues, customers: in the end there’s always someone else who wants to listen in, too. With the 2Control you also have the second headphone under control – and can rely on first-class quality.
Crossfeed control
With it's innovative crossfeed control, the 2Control allows to adjust the stereo width on headphones. Now headphone monitoring can be compared to speaker monitoring, making it possible to also mix on headphones.
Extend the range of headphone applications
Some exemplary benefits of the crossfeed control: at home, nocturnal engineers now have an alternative to full range monitors for judging stereo imaging in their mixes. Furthermore, the headphone monitoring is not restricted for analytical listening only, but can serve as full stereo monitoring alternative. And the compact housing makes it easy to transport the 2Control to have familiar monitoring conditions at foreign places.
High-grade balancing stages
We use custom made balancing stages to drive long distances reliably and for first class common mode rejection values. Note that only balanced connections exclude hum and interferences. We recommend to establish balanced connections wherever possible – especially connections over long distances (e. g. to speakers).
Unbalanced connections (e. g. RCA, TS Jack)
You can establish unbalanced connections easily and without adaptors – for example from CD-Players with RCA outputs or to (HiFi) power amplifiers with RCA inputs. It is important to pay attention to the correct polarity of the three XLR wires.Connections to RCA and TS Jack inputs or outputs are always unbalanced. Connections to TRS inputs or outputs may be balanced or unbalanced. In any case we recommend to use readily configured cables from XLR to the respective RCA or TS/TRS connector to dispense with adaptors. Ask your dealer for configured cables. With the XLR pin configuration diagram in our manuals any audio expert can ensure to select or configure appropriate cables.
Separate mono speaker output
In addition to two stereo speaker sets you can also connect a mono speaker or a subwoofer. Tip for working with headphones: Listen with subwoofer to enjoy the physical impact of the low end.
Dual Band De-Esser 500 Series module
Sibilance reduction by phase cancellation
Very effective - very unobtrusive
2-band processing: Hi-S and Lo-S
Only 2 controls - easy and fast to use
De-S intensity LED display
Male/Female voice selection
ON/bypass switch
Signal LED
Single-Slot 500 series rack module
Made in Germany
Unlike traditional compression methods, this procedure is much more unobtrusive and simplifies control to one single parameter. SPL‘s De-Esser quickly became a standard reference among recording studios, broadcast stations and live sound engineers.
The Dual-Band De-Esser expands on this concept by making use of two frequency bands that can be used independently or jointly. Two de-esser stages increase processing effectiveness without introducing audible artifacts. Focused processing with high and low bands makes it possible to process sibilant sounds with great precision. Furthermore, input signals are automatically adjusted so the processing is uniform, regardless of the distance between source and microphone. The Male/Female modes adapt processing in the lower band to male or female voices.
Now, the Dual Band De-Esser DeS is also available in 500 Series format.
High Band and Low Band De-Essing
You can use the two processing stages separately or jointly. They are connected in series as independent de-esser modules. The Low-Band De-Esser (Lo-S) is set fi rst in the chain. If both de-essers are engaged, there is interaction between them: a signal already processed with the Low-Band De-Esser is different from the raw material that the High-Band De-Esser would otherwise process.
Application: Voice (Male/Female)
The Male/Female switch allows you to adjust the Low-Band De-Esser to the type of voice being processed. These values have been determined by practical experience, so that the processor adapts better to gender. Nevertheless, you cannot take for granted that these settings will suit every single male and female voice.
Consider the Male/Female function as an additional tool to help you set the Low-Band De-Esser more precisely according to your needs. Always trust your ears to find the best settings.
Transient Designer 500 Series module
Extended original Transient Designer circuitry
NEW: Parallel MIX control - blend between "Wet" and "Dry"
ATTACK: amplify or attenuate attacks
SUSTAIN: Prolong or shorten sustain
Differential Envelope Technology (DET)
Output level control
ON/Bypass switch
Signal LED
Single-Slot 500 series rack module
Made in Germany
Working with the Transient Designer is very simple: Attacks can be amplified or attenuated and sustain may be prolonged or shortened. However, the possibilities for studio and live application are seemingly endless.
Technical foundation is SPL‘s Differential Envelope Technology (DET) which allows level-independent dynamic processing by calculating differences in generated envelopes. These envelopes are always tracking the curve of the original signal to provide optimal results in every moment of the music. So only two controls per channel are required to allow the user to completely reshape the attack and sustain characteristics of a sound.
Thanks to the new TDx feature MIX (parallel mix) you can continuously blend between the processed and the unprocessed signal. Thus, the range of functions is extended even further and with the three parameters ATTACK, SUSTAIN and MIX, which offer an intuitive operation, the options of designing transients reach a new dimension of great variety.
Applications
The Transient Designer is ideally suited for use in professional recording, project or home studios and in sound reinforcement applications. Applied to single instruments or loops the Transient Designer allows you to create entirely new sounds and/or effects.
The following examples are given as suggestions and examples. You will find more examples in the manual. The described procedures with specific instruments can of course be transferred to others which are not mentioned here.
Drums & Percussions
Processing drum and percussion sounds is probably the Transient Designer’s most typical range of application, both from samples to live drum sets:
Emphasize the attack of a kick drum or a loop to increase the power and presence in the mix by increasing ATTACK.
Drums Ambience
If your drums happen to sound as if the room mics have been placed in a shoe closet, the Transient Designer can immediately turn that sound into the ambience of an empty warehouse. Just send the room mic through the Transient Designer module and crank the ATTACK control to emphasize the first wave.Now slowly increase SUSTAIN values to bring up an “all-buttons-in- 1176-sound“ room tone—but without pumping cymbals.
Bass: Staccato vs. Legato
Speaking of bass: Imagine a too sluggishly played bass track ... you may not have to re-record it: Reduce the SUSTAIN until you can hear clear gaps between the downbeats—the legato will turn into a nice staccato, driving the rhythm-section forward.
Transient Designer 500 Series module
Extended original Transient Designer circuitry
NEW: Parallel MIX control - blend between "Wet" and "Dry"
ATTACK: amplify or attenuate attacks
SUSTAIN: Prolong or shorten sustain
Differential Envelope Technology (DET)
Output level control
ON/Bypass switch
Signal LED
Single-Slot 500 series rack module
Made in Germany
Working with the Transient Designer is very simple: Attacks can be amplified or attenuated and sustain may be prolonged or shortened. However, the possibilities for studio and live application are seemingly endless.
Technical foundation is SPL‘s Differential Envelope Technology (DET) which allows level-independent dynamic processing by calculating differences in generated envelopes. These envelopes are always tracking the curve of the original signal to provide optimal results in every moment of the music. So only two controls per channel are required to allow the user to completely reshape the attack and sustain characteristics of a sound.
Thanks to the new TDx feature MIX (parallel mix) you can continuously blend between the processed and the unprocessed signal. Thus, the range of functions is extended even further and with the three parameters ATTACK, SUSTAIN and MIX, which offer an intuitive operation, the options of designing transients reach a new dimension of great variety.
Applications
The Transient Designer is ideally suited for use in professional recording, project or home studios and in sound reinforcement applications. Applied to single instruments or loops the Transient Designer allows you to create entirely new sounds and/or effects.
The following examples are given as suggestions and examples. You will find more examples in the manual. The described procedures with specific instruments can of course be transferred to others which are not mentioned here.
Drums & Percussions
Processing drum and percussion sounds is probably the Transient Designer’s most typical range of application, both from samples to live drum sets:
Emphasize the attack of a kick drum or a loop to increase the power and presence in the mix by increasing ATTACK.
Drums Ambience
If your drums happen to sound as if the room mics have been placed in a shoe closet, the Transient Designer can immediately turn that sound into the ambience of an empty warehouse. Just send the room mic through the Transient Designer module and crank the ATTACK control to emphasize the first wave.Now slowly increase SUSTAIN values to bring up an “all-buttons-in- 1176-sound“ room tone—but without pumping cymbals.
Bass: Staccato vs. Legato
Speaking of bass: Imagine a too sluggishly played bass track ... you may not have to re-record it: Reduce the SUSTAIN until you can hear clear gaps between the downbeats—the legato will turn into a nice staccato, driving the rhythm-section forward.
SPL Channel One Mk2
SPL design represents the common origin of channel strips Frontliner, Channel One and Track One.The preamp stageprovides three optimised inputs: The microphone input features 48V phantom power, phase reverse function and a subsonic filter. The low-impedance Line input has a precise balancing stage for connecting studio equipment. The low-noise, high-impedance instrument input is easily accessible on the front panel.De-EsserThe renowned SPL De-Esser subtly yet effectively removes offending sibilants by reliably eliminating only the “S” frequencies with its unique phase-cancelling processing technology. Auto-Threshold ensures constant processing even with varying microphone distances. All you need do is to turn the knob until the sibilants are gone.Insert (Send/Return)The Insert (Send/Return) allows for insertion of external processors or can be used to record the pure preamplifier signal.Compressor/LimiterThis section is based on SPL’s Double-VCA-Drive® circuitry for outstanding noise and distortion characteristics. The compressor is extremely easy to use via a single knob. But the signal-dependent automation not only allows for set and forget operation, it also grants very muscial results in any situation. Actually it is much more difficult to destroy a recording with wrong compression settings.Noise GateThe noise gate operates flawlessly - reverbs are recognized and not cut off.Three band EQThe Channel One provides a new EQ design especially optimized to process vocals, acoustic and electronic imstruments. Low, Mid Hi, and Air-Bands allow for both corrective adjustments and/or effective sound designing processing. Gadget: a distortion control might be useful sometimes to apply for decent distortions to signals that deserve it.Headphone monitor stageProvides an individual mix for the singer/speaker - latency problems from digital systems are avoided.
Speaker & Subwoofer
You can listen to your production via two switchable pairs of speakers. A subwoofer is also switched on and off with speaker pair A.
In the middle position of the switch, the speakers are switched off, which is convenient because you don’t have to turn the volume control back.
Two Stereo Inputs
Two analog stereo inputs can be connected to the Control One for sources such as a DA converter, CD player, mixing console, synthesizer or analog tape machine/tape deck, which can also be listened to mixed together (1+2).
Line Out
The selected input signal is output unchanged at the Line Out. This allows the Control One to be looped between devices without “losing” an output.
If, for example, several people are to listen in via headphones, a headphone amplifier can simply be connected to the Line Out. You can also record the signal analog or send it to a mixer for further processing.
Three Monitoring Modes
are offered by the Control One. In the middle is the default setting “Stereo”. To check the mono compatibility set the switch to “Mono”. The special feature is the channel swap function:
L/R > R/L
reverses the stereo image. L/R becomes R/L. This is especially important and extremely time-saving when you are searching your sound library for samples in video dubbing that should match a scene with direction of movement.
If the direction is not correct, you usually have to load the sample into the DAW to switch channels before you can hear if the sample fits.
With the L/R > R/L function this is no longer necessary. You can now simply swap the direction of movement while prelistening samples in the library.
The Headphone Power Amp
Poor sound over headphones?Not enough juice?
Not here. Control One delivers a rich headphone sound and it can get really loud.
The output stage of the headphone amplifier is designed as a push-pull amplifier in class AB mode. The bipolar transistors share the amplification of the positive and negative half-waves, which produces a higher gain and a higher output voltage than in Class A operation, where only one transistor amplifies both half-waves.
The output stage transistors are thermally coupled and thus run particularly coherently, which contributes to a consistent and stable sound image.
The power supply has a buffer circuit with low source resistance, ensuring generous current reserves even when driving low-impedance headphones.
The Revolution
in the headphone amplifier is the Phonitor Matrix, with adjustable crossfeed, thanks to which you can create mixes with headphones that sound the same on speakers.
The Phonitor Matrix in its largest expansion stage has three parameters: Crossfeed, Speaker Angle and Center Level.
In the Control One, the center level is preset to -1 dB and the speaker angle to 30°. These are the most commonly used values. The crossfeed function determines the so-called interaural level difference. The intensity of the crossfeed is fully variable. At the beginning of the control path the crossfeed circuit is not in the audio path. Dialing the control in switches on the crossfeed circuit via relay (hysteresis circuit). The audio signal therefore does not pass through the crossfeed stage if this is not desired.
The Crescendo is the first microphone preamplifier which operates with an internal operating voltage of 120V. Making it absolutely unique, because with the 120V Technology it is nearly impossible to overdrive this preamplifier. The Crescendo excels with absolute signal fidelity and clarity. Rediscover your microphones, they have never been amplified like this.
With the Crescendo duo, we extend the SPL 120V technology microphone preamp portfolio with a two-channel version of the SPL Crescendo.
Crescendo duo excels with the same outstanding sonic and technical features as the flagship of the product line – Crescendo. It is nearly impossible to overdrive this preamplifier. It excels with absolute signal fidelity and clarity and lets microphones appear in a whole new light – they have never been amplified like this before.Besides the switchable +48V phantom power for condenser microphones, Crescendo duo offers a high-pass filter to eliminate low frequency noise, the ability to reverse the polarity of the signal and a PAD to lower the input level. The two large VU meters visualize the output level of the amplified signal. The ability to reduce level, shown on the VU meter, by 10dB makes it easy to display the dynamics of high levels. A highlight of the Crescendo duo is the individual control of Mic Gain and Output Gain. This way, the microphone signal is amplified as best as possible without wasting a single dB.
The signal can be picked up at the rear via two parallel XLR outputs per channel. For example, two different recording systems can be used (back-up) or two different signal paths can be fed.
The Crescendo duo is the perfect start into the world of the SPL 120V technology microphone preamplifiers.
The Crescendo duo is perfectly suited for the most demanding stereo recordings, but is also ideal for mono signals such as vocals. It is the ultimate front end for all professional recording studios.
Typ: De-Esser
Kanalanzahl: 2
Bedienelemente: S-Reduction Regler, Auto Button, Female Button, ON Button, Power Switch, Ground Lift
Anzeige(n): 2x S-Reduction-LED-Anzeige
Anschlüsse: 2x XLR (Input), 2x XLR (Output), 2x 6,3 Klinke (Input), 2x 6,3 Klinke (Output)
Bypass: schaltbar
Besonderheit(en): S-reduktion nicht durch Bandspezifische Kompression, sondern durch Beimischung des Phasengedrehten spezifischen Frequenzbandes, wodurch sich der S-Laut akustisch auslöscht
Spannungsversorgung: 115/230 V AC, 50/60 Hz
Abmessungen: 482 x 44 x 237 mm
Lieferumfang: Prozessor, Netzkabel
Produktionsland: Deutschland
Expansion Rack
Rack-mount housing for the Phonitor 2 + 1x4 Switch
Thanks to the Expansion Rack's built-in 1x4 Switch, the Phonitor 2 is expanded with three additional loudspeaker outputs, making it the perfect analog 120V monitor controller.
It allows you to listen to three user-definable analog stereo signals with headphones or up to four stereo speaker pairs all with the best-possible quality.
Plus, the comprehensive controller functions of the Phonitor 2 (L/R phase reverse, L/R Solo, Mono, Mute) are readily available for use with all connected playback systems.
Dimensions and Weight
W x H x D: 19" x 3.46" x 11.8" (482 x 88 x 300 mm)
Weight: 6lbs (2.7kg)
Expansion Rack
Rack-mount housing for the Phonitor 2 + 1x4 Switch
Thanks to the Expansion Rack's built-in 1x4 Switch, the Phonitor 2 is expanded with three additional loudspeaker outputs, making it the perfect analog 120V monitor controller.
It allows you to listen to three user-definable analog stereo signals with headphones or up to four stereo speaker pairs all with the best-possible quality.
Plus, the comprehensive controller functions of the Phonitor 2 (L/R phase reverse, L/R Solo, Mono, Mute) are readily available for use with all connected playback systems.
Dimensions and Weight
W x H x D: 19" x 3.46" x 11.8" (482 x 88 x 300 mm)
Weight: 6lbs (2.7kg)
This preamplifier combines Class-A Clean Gain with sound-shaping Tube Gain in a compact design that allows for free placing close to the microphone or on top of an amplifier.
Features:
Separated Class-A and tube preamps.
Tube saturation from subtle to brutal
A tube stage utilizing premium MKP foil condensers and a select 12 AX7 LPS tube for clear, dynamic audio
Peak/FET Limiter
Polarity Reversal
48-volt phantom power,
LED level indicators.
Switchable Mic Input Impedance (200 Ohms, 1,2 & 10 kOhms)
Custom-made, fully discrete, Class A op-amps
60V DC Audio Rail operating voltage—twice as high as most common op-amps—for an extended dynamic range.
Extreme amplifier slew rate of <40V/µs, ensuring clean transmission of high-frequencies and rapid transients
All switch functions are handled by encapsulated relays with gold-plated contacts.
Optimized layout guarantees the shortest possible signal paths, while generously proportioned grounding surfaces ensure low impedance and maximum shielding
All resistors are within 0.1% tolerance and were selected after extensive listening tests
A no-compromise power supply with extensive additional shielding and seven separately wound and regulated voltages
With Gemini, M/S processing enters the SPL Mastering series. Gemini is an M/S Encoder and Decoder. Mid signals (voice, snare, bass …) can clearly be seperated from side signals (guitar, spatial sounds, cymbals …) and can individually be processed. When working on the sum signal, M/S coding often is the best way to specifically get access to individual elements within a mix.
Gemini also provides the possibility to work on the stereo panorama. With the Balance control you can position the mid signal within the stereo panorama. With the Trim control, you can adjust the level of the mid signal in relation to the side signal. In connection with the Stereo Width control, the mixing ratio of both channels can be adjusted.
An Elliptical Filter, which can cut low frequency ratios of the side band is also provided.
SPL Goldmike MK II
Like its successful predecessor, the GoldMike Mark2 retains a hybrid solid state and tube construction to combine the best of both worlds. The transistor stage is composed of single transistors in a class A design. The circuitry is fully discrete, and each transistor is completely optimized for its specific task. You will not find any IC’s in this preamplifier stage because they cannot be optimized for this specific application to the degree we aimed for. This all new discrete class A transistor stage is a genuine innovation in the entire preamplifier market at this price level.
Frequency response ‹10Hz bis 90kHz (-3dB)
THD+N (Input level -30dBu, 30dB Gain) 0,016%
Noise (A-w., R=40Ohm, 30dB Gain) -91,2dBu
Dynamic range (30dB Gain) 110dB
E.I.N. 128dBu
Max. output level (sym., XLR+Jack) +26,8dBu
Slew rate solid state stage 200V/µs
The GoldMike MK2's new features provide an amazing tonal flexibility and versatility, making it a perfect dual-channel frontend for any modern production environment.
Discrete Class A solid stage
Tube drive in three different intensity levels
Flair presence enhancement in two switchable settings
Switching inserts
Pre-output limiter stage (extremely fast diode-based operation, perfect for A/D converter protection)
VU metering with three different, switchable display ranges
Front panel instrument input, separate rear-side microphone and line inputs
Phantom power, phase reverse, pad and high-pass filter (50Hz)
Options include 24/192 AD converter and in/output transformers by Lundahl
The GoldMikeMK2 is the result of all our experience and knowledge from its successful predecessor and represents an improved, more flexible, and more modern version.
We have added technologies that have already proven to provide superior tonal results in other SPL products. For example, the instrument input is designed around a class A impedance converter which premiered in the SPL GainStation. A significant contribution to the success of the predecessor was the hybrid preamplifier concept. It combines transistor and tube preamplifier stages and specifically utilizes their technological and sonic advantages: The efficiency and low noise of a discrete transistor stage is enriched with musical coloration of a tube stage.
The transistor stage is composed of single transistors in a class A design. The circuitry is fully discrete, and each transistor is completely optimized for its specific task. You will not find any IC’s in this preamplifier stage because they cannot be optimized for this specific application to the degree we at SPL aim for. This all new discrete class A transistor stage is a genuine innovation in the entire preamplifier market at this price level.
The degree of tube pre-amplification is no longer the fixed +6 dB value of its predecessor. The GoldMikeMK2 sports the selection of three different tube pre-amplifications: +6 dB remains the standard complemented by +12 dB and +18 dB. This allows creative variety with the tube saturation and limiting effects.
We have also added an extra setting to the popular Flair circuitry to give more flexibility to the presence improvement. Because the GoldMikeMK2 is a dual channel preamplifier, its channel separation is of fundamental importance for the stereo image. Therefore the print-board layout is symmetrically mirrored around the power supply in the center from channel 1 to channel 2. This results in an identical signal flow to supporting a naturally balanced stereo image. Additionally the print-board provides extra-large ground areas maximizing the shielding against crosstalk and interference.
Hermes revolutionizes Mastering. Now it is possible to route an audio signal through up to eight dual-channel processors in any order. User definable presets allow the comparison of complex processing chains with just a flip of a button.
In addition, Hermes has two integrated parallel mix stages that work with any of the eight processors allowing for comparison of two compressors with different parallel mix settings. The parallel mix stages are stored with the processing chains.
Hermes routing is entirely passive using gas-capsuled and gold-plated high-end relays. All active electronics like the I/O stages and the Parallel Mixes use SPL's proprietary and unequaled 120V Technology.
Hermes speeds up the workflow in mastering in ways that were previously extremely difficult, and makes the most out of your existing mastering gear. Repatching to hear a simple change is a thing of the past with Hermes – you can change processor sequences on the fly, store them, and compare settings instantly. All with real switches, relays and no software application.
Features:
Mastering router for 8 dual-channel processors
Two Parallel Mix stages
Create, store and compare complex processing chains
With gain compensation
Fast comparison of up to 4 processing chains
Can be assigned to any device
Fully passive routing using
120V Technology
Gold-plated and gas-capsuled relays
Label mode to enter device names
Features:
With the Headphone Monitoring Amp HPm the SPL Phonitor Matrix is making headway into the world of 500 series rack modules. Each of the two headphone outputs is fed with a separate amplifier to allow independent operation of two headphones with high-quality amplification.The Phonitor Matrix – derived from the High End Headphone Amplifier ‘Phonitor’ – eliminates the super-stereo-effect inherent in traditional headphone amplifier designs. In cases where speaker monitoring is impossible or just as an alternative thereto the Phonitor Matrix allows a speaker-like listening experience on headphones. With the Phonitor Matrix you create better mixes on headphones.Both the Crossfeed and Speaker Angle can be individually adjusted to achieve a headphone playback sounding equal to your loudspeaker playback.The adjustable Center Level supports the mix engineer to correctly level the phantom center signals when mixing on headphones.The HPm can easily be inserted between the DAW and a monitor controller because the input signals of the HPm are slaved through to the outputs of the 500 series rackPhase Inversion of left or right channel as well as a Stereo/Mono/Mute switch round off the feature set of the HPm.
The IRON Mastering Compressor is a variable-bias limiter/compressor from the basic concept. Through the integration of new technologies, this concept gets significantly improved.
IRON works on the principle of the bias controlled remote cutoff tube. Sharp-cutoff tubes with a steeper characteristic are parallelly connected to these.
Depending on the level of the signal amplitude, one or another tube is in charge for signal processing/limiting. The positive result is a well-balanced sound and more controllable settings of the parameters.
All four tubes in the IRON are a specially matched set to the overall system.
IRON
Mastering Compressor
The IRON Mastering compressor combines not only the sonic virtues of legendary vintage tube compressors with the advantages of the High Dynamic 120V operating voltage in a single unit. It is also perfect for the needs of modern mastering studios and sets a new benchmark in terms of tube compressor technology, with the innovative implementation of a parallel dual-tube circuit.Thanks to the especially conceived Mu-Metal iron transformers, the signal of each channel is split across two different twin-triode tubes. The combination of the different response curves of both tubes results in a transparent and musically pleasant compression. Additionally, peak signals of the control voltage are limited by a feed-forward resistive vactrol-opto-isolator. Thus, the output signal remains lively even with a high gain reduction. The compression is only noticeable with extreme settings.The Tube Bias has three different settings. Together with the Input Gain of up to +/- 12dB and the Threshold control, it allows the compression behavior of the tubes to be perfectly adapted to any material.The Attack and Release parameters have six different settings ranging from Slow to Fast. The times are not constant, they vary according to the Rectifier circuit selected: There are six different Rectifier settings available with different diodes (germanium, silicon, LED, mixed).The Sidechain EQ is yet another option to adjust the Iron to the signal being processed in an optimal way. It allows you to choose between an external EQ or one of the four internal EQ presets. The EQ presets have a complex frequency response and have been conceived with different program material in mind. Only the control signal is affected, not the actual audio signal. In the Off position, the Sidechain EQ is inactive.After the compression stage, the signal can be boosted or attenuated up to +/- 12dB with the Output Gain, which makes the Iron easy to integrate into any signal chain.All functions can be adjusted via switches or, in the case of the Threshold, with a detented potentiometer, making it easy to replicate exact settings.In stereo Link mode, the Threshold, Tube Bias, Attack, Release, and Rectifier parameters, as well as the Sidechain EQs, are controlled with the right side (channel 2) of the unit. In contrast to the Dual-Mono mode, in stereo Link mode the compressor works with a sum signal. Thus, the stereo image is wider when working with the Dual-Mono mode on stereo material.The compressor has an additional passive 120V equalizer to give the output signal the finishing touches. You have two different EQ presets to choose from: AirBass and Tape Roll-Off. The AirBass mode gently boosts high and low frequencies approx. 2dB. For its part, the Tape Roll-Off mode is based on the frequency response of a tape machine and uses a high and a low-cut filter.Both the left and right channels have separate illuminated activating switches. Further, in Auto Bypass mode you can use the Interval control to set a time frame during which the compressor automatically toggles back and forth between the processed and unprocessed signals. This makes AB comparisons between the original and compressed signals a breeze, besides making it easier to assess the settings in context.The ergonomic design and clear arrangement of the control elements make working with the compressor more intuitive, while its shallow housing makes the Iron perfectly adaptable to tabletop rack stands.Mastering is not the only domain where the IRON sets new standards. It can also be used to process individual instruments, like vocals, bass, guitar, strings, etc. The IRON is also an excellent option for subgroups.
Operating Principles of a Compressor
The basic operating principles of a compressor/limiter can be easily explained. The level of an audio signal is reduced according to the specified attack time and ratio whenever it exceeds a given threshold. This reduction ceases when the release time elapses, while the compressed signal is amplified with the make-up gain. Compressors basically differ from each other in the technology used. This technology - tubes, opto, FET, or VCA - is what gives a compressor its particular character. Some units sound soft and silky, some sound pounding, while some others make sound fatter, and there are those that make sound clearer, harder or more percussive. The trick resides in how the unit is technically designed, in the signature of the maker. Different compressors with the exact same settings might work and sound completely different. They provide different sounds for different applications and music styles.Nowadays, the compressor has become a key element when it comes to provide dynamicsand punch to any production. The number of compressors available is huge and it‘seasy to succumb to the promises made by software emulations and analog recreations ofvintage gear as the perfect solution. Unfortunately, many of these emulations and recreationsdiffer quite a bit form their original counterparts. You must simply accept that the components used today, like the transformers, tubes and all other passive elements, are different to the ones originally used and that they can‘t be digitally emulated. No software (DSP-emulated compressors) or hardware replica will ever be able to sound like the original. An authentic sound can only be achieved with the original unit.
Innovating Compression
The IRON mastering compressor was conceived as a variable-bias limiter/compressor right from the start. However, the implementation of new technologies results in many improvements. Its basic operating principle as a variable-bias tube compressor was loosely inspired by the sonic and technical operation of Fairchild, Collins and Gates compressors, which used remote cut-off of tube biasing to achieve a well-balanced, well-compensated and musical compression. However, the IRON compressor features a second sharp cut-off tube, a medium-variable bias triode, in its circuit design. This tube is connected in parallel to the remote cut-off tube and it has a considerably steeper characteristic curve. The tube used to process the signal depends on the amplitude of the latter. This results in a more well-balanced sound and more controllable settings of the parameters. The pair of parallel connected tubes has been specially matched for the IRON. In order to guarantee that tube selection and pairing is perfect, we use the Weigl Roe Test for PC. The optimal selection of the tubes guarantees that all IRONs have the same sonic characteristics. Moreover, we use Lundahl custom-made balanced high-level dual-coil mu-metal iron transformers in the signal flow of the variable-bias tubes, which add to the overall sound. The second new technology implemented is the independent feed-forward resistive opto-isolator in the control path of the variable-bias tube circuit. Its function is to limit signal peaks and, thus, get a smaller THD (Total Harmonic Distortion) within the variable-bias tube section. The result is a silkier, more homogeneous sound in the higher frequencies of the music signal. This circuit has its own rectifier in the signal path. The optical control element does not work in the sense of an audio limiter, like in a conventional opto-compressor. It is built-in in the control path of the parallel connected variable-bias tube, not in the audio path itself. Time control parameters, like attack and release time, are adaptedand fixed to the variable-bias tube circuit. The IRON compressor works as a feedback compressor when set to the variable-bias tube circuit and as a feed-forward compressor when set to the opto-control circuit.Thirdly, the complex rectifier circuit is also worth mentioning, since it is the basis for tube control. You can use the six-position switch to choose either of the six different control characteristic curves of the diodes within the rectifier. Given the specific characteristic curve of its elements, the combination of germanium, silicon and LED diodes produces different behaviors and characteristics for the Attack and Release times. Hence, compared to most compressors, the application scope of the IRON is clearly enlarged, resulting in new possibilities regarding the processing of music material. The fourth exceptional feature is the comprehensive logical relay circuit that perfectly links both channels together, making the right channel the Master regarding Release, Attack, Threshold, Rectifier, Tube-Bias and Side-Chain EQ settings.
120 Volt Technology
SPL‘s goal was to push analog signal processing to the limits. That‘s why we combined the best possible components with a high-grade optimized circuit design. We have been using the in-house developed 120 volt technology - the highest-ever operating voltage used for audio applications - in all our products of the mastering series for years. Some of the most highly respected mastering studios today revolve around SPL consoles and signal processors from our mastering series (Bob Ludwigs Gateway Mastering & DVD in the USA, Simon Heyworth‘s Super Audio Mastering in the UK, Galaxy Studios in Belgium, and the legendary Wisseloord in the Netherlands, for instance). The 120 volt technology is based on op-amps developed internally by SPL‘s founders together with chief developer Wolfgang Neumann. The IRON features the most advanced generation of these op-amps. They boast better tech specs thanks to the thermal behavior optimization they underwent under the hands of Bastian Neu. Ultimately, the supply voltage is key for the overall dynamic response of a processor. Voltage is to an electrical circuit what cylinder capacity is to an internal combustion engine. You can‘t replace cylinder capacity with anything else, except more cylinder capacity.
The SPL Madison offers the pure SPL analog sound for every MADI equipped digital console, audio router and PC MADI card.
With 16 input and 16 output channels in a single rack space, the Madison combines state of the art signal conversion with highly reliable clock and MADI processing. And as a true SPL unit it is simple to operate, easy to expand and, of course, an awesome sounding conversion system.
Special Features
16 AD and 16 DA converters in 1U
Superb sounding converters with a sophisticated analog section on 36 V audio rails
+24 dBu professional output level
Best price/performance value
Up to 64 I/Os on 1 MADI port (4 units)
Latency-neutral MADI chaining
44.1 to 192 kHz, ±10% vari-speed
SPL clock-shop for jitter-free operation
Ultra-compatible, always operational MADI IO with ultra low conversion latencies
Four directly selectable reference levels: 15/18/22/24
Simple operation and setup right from the front panel controls
Ergonomic, distance-readable meters for clear level status indication of any I/O channel
Fanless, absolutely silent low-power design (max. 30 W)
Redundant power supply optionally
Four analog, 8-channel multicore DB 25 sockets (TASCAM industry standard)
Made in Germany
Why MADI?
Broadcast and live professionals rely on the still-unsurpassed MADI since about 20 years now – with jitter-free, rock-solid transmission of many audio channels per single wire and low-cost cabling with lengths far above one mile, MADI simply offers unique advantages.
Why Madison?
As a company with a rich heritage in analog audio engineering, it has always been our idea to put all our knowledge into the analog sections of a converter. Excellent converter technology requires an appropriate analog environment, and that is where a true difference can be made – a difference against existing solutions, and a difference in sound of course.
We have designed an analog audio section on 36 V audio rails, reaching output levels of 24 dBu. The result is simply amazing. With its straight-forward operation, flexible expansion, and pristine sound, the price per channel ratio of the Madison may set a new benchmark.
And for the first time, SPL’s analog sound is coming to digital environments.
A good fit everywhere
Thanks to its rich I/O equipment, the Madison makes it a breeze to integrate professional analog gear into larger multichannel infrastructures. With four popular reference levels, operational sample rates between 44.1 and 192 kHz, multicore analog I/O, fibre MADI I/O and word clock I/O, the Madison simply fits everywhere.
Up to four units can be connected to a single MADI port on a digital console, a MADI PC card, or any professional audio routing infrastructure. The MADI port offers excellent and near-zero-latency digital transmission of up to 64 I/Os with galvanic insulation. And since MADI is the emerging new studio I/O standard, the Madison is the perfect interface for home and project studios, commercial recording studios and scoring stages.
So whereever you may have a need for reliable and pure conversion of multiple analog audio channels, the Madison meets your demand brilliantly.
Mercury is the first Mastering DA converter in 120V rail technology.
Mercury is a stereo digital to analog converter that fulfills the highest demands both technologically and tonally.A total of seven digital inputs, two AES/EBU, two SPDIF, two TOSLINK as well as USB can be connected to Mercury. AES input 2 also supports Dual-Wire (DW) mode.Each digital sources has its dedicated and illuminated switch for instant selection and fast comparison.
€2,590.00*
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