SPL

GainStation 1
Models 2272/2273

Single channel microphone and instrument preamplifier

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Input makes all the difference.

Modern audio production relies increasingly on digital systems for recording and mixing processes (DAW's, digital consoles etc.). But still digital systems do not offer the same audio qualities and sound characteristics of high-end analog equipment. Especially digital equalizers and other aspects of the mixing domain cannot compete with the open, transparent sound of the best analog gear. So today more than ever, your input signal has to sound as good as possible—the quality of your original tracks will largely define the end result.

Attacking this problem on the preamplification front, with the goal of noticeably improving a signal and sending it along its path with a healthy foundation, requires a great deal of thought and engineering effort – like with the GainStation 1.

Its very compact design allows for free placing also outside of a rack (e. g. close to the microphone). Musicians can place the GainStation 1 easily on a rack or amplifier.
To accommodate portability, the optionally available SPL GainBag ensures safe transport of the GainStation 1 and offers additional storage for cables and an average-sized microphone. For more permanent installations, up to four GainStation 1's can be rack-mounted in one optionally available 3-unit mounting frame.

 

The SPL GainBag to transport GainStation 1, a microphone and cable.
 

Most notable features of the GainStation 1:

• Separated Solid state and tube preamps. The tube preamp can be used additionally to the solid state stage to add tube saturations, for clean recordings it can be switched off.
• Peak/FET Limiter
• Switchable Mic Input Impedance (200 Ohms, 1,2 & 10 kOhms)
• Custom-designed and built, fully discrete, class A op-amps (no off-the-shelf stuff here). The GainStation's op-amps feature 60-Volt operating voltage—twice as high as most common op-amps—for an incredible dynamic range.
• All switching functions are handled by encapsulated relays with gold-plated contacts.
• A no-compromise power supply with extensive additional shielding and seven separately wound and regulated voltages
• Internal 24/96 AD converter, Lundahl input transforner, SPL GainBag and rack mounting frame optionally available

In our humble opinion, the time and energy invested in the development of the GainStation 1 was well worth it. The result is an extremely compact, unusually flexible preamplifier that delivers audio quality usually found only in units costing much more. We believe we just may have set a new record in the sound per cubic centimeter category.

In detail/specs

The Channel One is a complete channel strip with all the tools onboard for recording voices, instruments and digital or analog audio sources with the highest sound quality: a transistor/tube preamplifier, a de-esser, a compressor/limiter with noise gate, an EQ section and a headphone monitor.

The concept of the Channel One is as the ideal recording tool – for hard disk tracking as well as for analog environments. Intelligent automation ensures intuitive, fast operation and guarantees the finest recording results.

Sound On Sound (UK), Paul White:
"Until you've tried something that works this well, the words 'smooth' and 'detailed' won't have nearly as much meaning as after playing with the Channel One. If you're wondering where that elusive 'produced' vocal sound is hiding, look no further – it's in here!"

The Track One is a complete mono channel strip for the high-quality recording and processing of vocals and instruments and in spite of its high level of performance, it is very competitively priced. It is ideal for the fast and reliable recording of vocals or acoustic instruments but also suitable for live performance and AV post-editing.

Features

• Preamp stages optimized for all microphone types and instruments
• SPL‘s De-Esser
• Compressor/limiter stage
• 3-band EQ stage
• Output stage with metering
• Balanced XLR and Jack connectors (parallel use possible)
• Options: A/D converter and Lundahl input transformer

Ideally suited...

• for high quality recordings
• for HD recording (with A/D converter option)
• for vocal processings in studio or live applications
• for installations, „one man show“ entertainment etc.

Increase your track power eightfold in only 2U—Thanks to exceptional headroom throughout its design and the superb signal regulation of its limters, the GainStation8 is the perfect front end for multi-track recordings or AD converters.

Nowadays typical preamps are designed on an either-or basis — either solid state or tube. With these designs the engineer has a non integrated and thus often frustrating choice between a pure, neutral solid state recording or one unavoidably influenced by tubes. While the engineer may find solutions for one application with one microphone, a preamplifier for different situations, mics and instruments must provide a commensurate range of choices—like the GainStation preamps. A combination of 63dB solid state and an independently controllable 26dB tube circuitry (which can also be deactivated) ensures a complete sound palette panorama. In any situation the engineer now can achieve his goals with one front end for best-of-both-worlds results.

• Separated Solid state and tube preamps. The tube preamp can be used additionally to the solid state stage to add tube saturations, for clean recordings it can be switched off.
• Peak/FET Limiter
• Switchable Mic Input Impedance (200 Ohms, 1,2 & 10 kOhms)
• Custom-designed and built, fully discrete, class A op-amps (no off-the-shelf stuff here). The GainStation's op-amps feature 60-Volt operating voltage—twice as high as most common op-amps—for an incredible dynamic range.
• All switching functions are handled by encapsulated relays with gold-plated contacts.
• A no-compromise power supply with extensive additional shielding and seven separately wound and regulated voltages
• Lundahl input transforner option, also for single channels or channel pairs

Balanced instrumentation amplifier design, transparent and low noise operation, extremely reliable.

High quality triple stage design with Lundahl input transformers and discrete high voltage transistors.

The DynaMaxx compressor, limiter, gate offers ease of use due to musically automated time constants and premiers the De-Compressor to regain vitality from compressed audio files.

EQ module for the mid/vocal range with 8 passive boost/cut filters and dry-/wet control.

The Charisma is a tube processor designed to improve the power, punch, dynamics and the subjective loudness of processed signals.

The new Atmos system is a complete, system independent surround miking system including two mic array options: the ASM5 indoor microphone with Brauner VM1 capsules or the new SPL Pentarray with mic capsules from Microtech Gefell for indoor and outdoor applications.

Ten years of experience in surround miking technology result in a reference system for surround sound in highest definition. The new Atmos Controller incorporates the five channel, remotely controllable preamplifier section, a LFE channel composer with level control and a headphone monitor. In contrast to the fully featured predecessor, the controller is now focused on basic features. This reduces costs dramatically for a pure, high definition preamp/microphone setup.

The preamplifier section ensures a homogeneous preamplification over all channels for a coherent imaging of rooms and source widths. The high precision preamplifiers are based upon SPL‘s Triple Gain Stage to capture the superb sound of the mic arrays in the most transparent, noiseless and uncoloured way. Lundahl transformers provide galvanic isolation and a passive 1:5 preamplification.

The ASM5 is based upon five selected and matched VM1 large diaphragm capsules. The positions of the all microphone heads are variable +/-90 degrees. The new SPL Pentarray is produced by Microtech Gefell and comes with five M930 mics with variable postions as well. Individual microphone solutions can be connected to the Atmos controller as well. The Atmos provides balanced XLR and DB25 outputs (parallel).

Area 5.1 Model 2047 Enter Area 5.1 | Features Enter Area 5.1 The Area 5.1 is a 5.1 surround microphone preamplifier featuring five matched high precision gain stage preamplifiers with motorized master/slave mic gain controls, a Sub/LFE stage to determine the composition of the sub signal and stereo downmix stages for each channel. A monitor stage allows on-location monitoring with stereo headphones. The complete Area 5.1 system contains the Area 5.1 unit and the both optionally available SPL MA5 mic array and multicore lead. Though the Area 5.1 preamplifier could be used with any microphone setup, the combination with the MA5 is recommended. This setup has proven in various live and studio situations to be perfect in capturing the 5.1 sound image of a preferred "listener's position", while its versatility is exceeded only by its ease of use. Generally any digital or analog recording system can be connected to the analog ¼" unbalanced jack or balanced XLR outputs of the Area 5.1. If, for example, an eight-track DTRS is employed, the first five tracks can be used for the L, C, R, SL and SR channels, the sub bass signal should be routed to the 6th track while track 7 and 8 can be used to record the stereo mixdown. Features Matched audiophile preamplifiers (pad, phase reverse, 48V power supply, Lundahl input transformers) High precision motorized master/slave mic gain controls Stereo downmix stages for 5.1 and stereo recordings in parallel Suited for any microphone type (matched mics recommended) First discrete 5.1 system for motion recordings with optional SPL MA5 mic array.

The Charisma is an 8-channel tube processor to improve the sound quality of digital 8-track recording systems. It brings the warmth and smooth bottom end into digital recording. The novel "Charisma" circuitry generates variable tube saturation that can be compared to the saturation effects of analogue tape recorders.

Charisma processing improves the power, punch, dynamics as well as subjective loudness. It also helps to optimise the exploitation of the headroom in digital recording and helps to prevent overloading.

The Charisma 2 is a dual channel tube processor designed to improve the power, punch, dynamics and the subjective loudness of processed signals. Designed to introduce a subjective tonal warmth and a smooth bottom end into digitally recorded material, Charisma exceeds the mere simulation of tape saturation.

Charisma also helps to prevent overload and clipping on digitally or analogue recorded tracks. Charisma gives your signals more clarity, transparency and presence. Each part of the mix, instrument or vocal can easier be distinguished and located. Using the Charisma on single instruments alone lets those signals stick out of the mix. It also helps you to place vocals in a densed playback and maintaining their presence. The Charisma is equipped with XLR-connectors and TRS stereo jacks for balanced operation.

The Transient Designer offers a completely new technology for level-independent shaping of the dynamic response of a sound: For the first time it is possible to control the attack and sustain of a signal in a very simple way!

Unlike other dynamic devices, the processing is not goverened by the signal level but rather by its dynamic characteristics, so all signals (loud and soft) are processed equally.

The Transient Designer allows you to emphasise or smooth the attack and extend or shorten the sustain. For the first time it is possible to shape the dynamic path of a signal as you want, as if for example you were changing the microphone distance and position after the recording has been made.

With no effort you can shorten or lengthen the attack of all kinds of percussive signals, such as the bass drum, snares, toms, congas etc. to give them more kick, or to flatten the signals. It’s almost like being able to change the amount of drum damping after the recording. The same applies to virtually any other signal: Amplify or reduce the picking sound of an acoustic guitar, hold the sound of the strings longer, reduce the reverbtime of a choir, compress solo vocals, increase intelligibility, actuate the piano pedal “electronically”, turn down the slapbass a notch or give it even more attack, etc.

The DynaMaxx compressor, limiter, gate offers ease of use due to musically automated time constants and premiers the De-Compressor to regain vitality from compressed audio files.

The DynaMaxx is used for unobstrusive compression or as sound tool delivering compression effects. It makes use of the new THAT 2181 VCAs in Differential-Drive technology for maximum transparency, dynamics, and lowest distortion. DynaMaxx is equipped with XLR-connectors and TRS stereo jacks for balanced operation.

"The SPL DynaMaxx succeeds pretty clearly in the performance as well as in the price/value category." TOOLS4Music 2/2003 (GER), (Review of DBX 1066, Drawmer DL-241, LA Audio GCX-20, SPL DynaMaxx)

User statement of Patrick Sanders, Midimix Recording Studio, Belgium:

"I'm totally satisfied with the SPL DynaMaxx. In my opinion, this unit captures the essence of pure "set and forget" equipment. It is so easy to use (compress ratio and gain make-up ... Hello? Where are the 35 other parameter knobs?) and nearly always sounds great, very musical and perfectly natural. When I started buying hardware for the studio, I reckoned that compressorwise I'd better play it safe and go for the old traditionals (dbx, TC Electronics). Luckily the guys of Music Store (Köln-Germany) pointed out that this compressor is a "must have", considering the price, its ease of use and its sound quality. Now I regret not having bought 3 units instead of just 1, because I find myself always returning to and depending on it. Need some punch and you don't have half an hour to waste searching for the ideal attack/release times? Just shove it through this machine, it always does the trick. Whether it's vocal tracks, drums, (bass) guitars, pianos or Hammonds I'm recording: I just wished I had more Dynamaxx compressors. Congrats SPL, great stuff!"

SPL is constantly pushing analog signal processing to its limits by combining the best possible components with perfectly optimized circuit design. As our latest achievement in this ongoing process, SPL‘s High Gain Series surpasses the hitherto accepted limits of audio signal processing and combines truly unparalleled analog sound quality with the control and convenience of digital units. Further High Gain projects are in planning, including a microphone preamplifier and a compressor, although no availability projections can be made for these currently.

PQ 2050

The PQ is a fully parametric, dual-channel 5-band equalizer. As with all SPL High Gain units, the PQ has an operating voltage of 124 V and is constructed with the best analog components available.

The Filter Stages

The individually switchable filter bands are self-contained on separate circuit boards. Contrary to most conventional equalizers, the bands are not summed via a central stage, so that the signal passes only through those filter stages which are activated.

Constant Q and proportional Q modes

Each frequency band consists essentially of two fully parametric equalizers: one for constant Q mode and one for proportional (or variable) Q mode, selectable per band. The PQ is the first equalizer to offer both modes, endowing it with double equalization power for demanding corrective and creative applications.

In constant Q mode, the selected bandwidth remains unaffected by the amplitude setting, making it the better choice for corrective applications (e.g. to eliminate unwanted frequencies). In proportional Q mode, the bandwidth is reduced as the amplitude is raised and vice versa. This more „musical“ operating mode is better suited for creative, sound-shaping applications. Controls

All variable controls are motorized and all keys are digitally controlled to provide the same programmability and repeatability of digital units. Individual settings can be stored in up to 24,000 presets.

Motorized potentiometers

The adjustment range of the Frequency, Bandwidth and Amplitude potentiometers is evenly dispersed throughout the 300-degree movement area, ensuring smooth and continuous adjustments and extreme operational security. The high resolution of each respective value (e.g. Q from 0.1-3.7, evenly distributed over 300 degrees) allows extremely precise adjustments as well as accurate repeatability.

We intentionally avoided unnecessary or confusing functions such as frequency range switching or frequency, bandwidth or amplitude multiplicators.

LINK Mode

Channel 1 and 2 can be linked for coherent stereo operation and either channel can be designated as master. In link mode, coupled bands can be activated/deactivated simultaneously, while single bands may also be released from link mode if required for individual adjustment.

Master/Slave (unit link) mode

FThe PQ offers a Master/Slave mode for surround applications or subgroup processing, in which the settings on the master unit are precisely duplicated on the slave units. Thus the PQ is the first EQ system that provides up to eight channels for coherent analog equalization of surround recordings.

Remote Control Unit

SPL is developing a remote control unit with an identical user interface to the PQ, which will be able to control up to four units—individually or in common.

Display

A central display clearly indicates all information relating to preset, storage and control functions. The control logic is clear and intuitive without menu layers, guaranteeing quick and easy operation and a minimal learning curve.

Frequency

Each channel provides five fully parametric frequency bands covering 10 Hz to 28 kHz. Each band overlaps its neighbor by one octave and can be independently activated and deactivated.

The following frequency bands are provided:
LF (Low Frequencies): 10 Hz-235 Hz
LMF (Low Mid Frequencies): 35 Hz-720 Hz
MF (Mid Frequencies): 330 Hz-7.1 kHz
HMF (High Mid Frequencies): 670 Hz-16 kHz
HF (High Frequencies): 1.2 kHz- 28 kHz

Amplitude

Amplitude settings are freely adjustable from -11.5 to + 11.5 dB. The lower range between 0 and 2.5 dB offers extremely high resolution, allowing delicate adjustments with maximum precision.

Bandwidth

Bandwidth is adjustable between 0.1 and 3.6 Q in proportional Q mode and between 0.7 and 15 Q in constant Q mode. The extremely high maximum bandwidth in constant Q mode enables tight, selective control in precision applications.

Illuminated keys The PQ features illuminated keys that illuminate brightly when activated.

LF, LMF, MF, HMF and HF keys These keys activate (key lamps lit) or deactivate (lamps off) the 10 filter bands.

Con. Q key This key switches the 10 filter bands between proportional Q mode (lamps off) and constant Q mode (lamps lit).

Additional construction features

All crucial resistors in the I/O and filter circuits are subject to ultra-tight tolerances of 0.001% to ensure clean and accurate amplifier stage performance. All circuit board tracks are double-width (1 mm) and double-thickness (70 micrometers), guaranteeing stable high-gain operation with minimum resistivity. A separate grounding track is placed between signal track pairs, virtually eliminating crosstalk for maximum signal separation.

Connections: The PQ is fitted with XLR I/O connectors.

Power for sound: SPL SUPRA components

The central component of the PQ is a fundamentally new amplification design: discrete, custom made Class A audio operational amplifiers which run on a 120V operating voltage (+/- 60 V). This amounts to over three times the operating voltage found in most high quality audio gear (+/- 15-20 V) and about twice as much as the highest voltages used in the best units currently available. This extremely high voltage allows the circuitry to process an astonishing dynamic range of ca. 150 dB and an amazing +34 dB of headroom, virtually eliminating overloading of individual filter stages—even when processing extremely high-level signals. For the first time, transistor circuits with such an impressive degree of stability and freedom from harmonic distortion can be realized. After the revival of tube units in the early ‘90s, we feel that the time may be ripe for a revival of the „transistor sound“ in the near future.

Input stages of the SUPRA components Wolfgang Neumann, founder of SPL and developer of the High Gain Series, has paid the highest attention to realizing components with high loop amplification, extremely low phase shifting and THD, combined with maximum amplification and a frequency response up to 100 kHz. A main and obvious advantage of the discrete SUPRA components is the exclusion of those parts often found in industrially manufactured standard components that are not necessary for audio processing. The SUPRA input stages are designed as symmetrical differential stages and comprise six matched high voltage transistors switched in parallel. The concept of the input stage is based on the established principle that parallel circuits are not correlated noise sources, but the wanted signals are added geometrically which decreases the overall noise. The input stages are free of coupling capacitors to exclude additional capacitor noise. The symmetrical operation voltage of +/- 60 V is delivered from a linear -80 dB high voltage power supply.

Intermediate stages of the SUPRA components The audio signal is lead to a further differential stage and from there through further processing stages to the Class-A output stage. All passive components have been tested for the highest fidelity.

Output stages of the SUPRA components Extremely low noise, high voltage output transistors are set up with a high quiescent current and excessive heat is dissipated via special cooling plates.

Technical specifications

Input impedance (balanced): 10 kOhm
(Welwyn precision resistors; transformerless)
Output impedance (balanced): 600 Ohm
(CMR trimmers, transformerless)
Overload resistance: +34 dB
Harmonic distortion:
@ -30 dBu: 0.2%
@ -20 dBu: 0.05%
@ 0 dBu: 0.01%
@ +10 dBu: 0.002%
@ +30 dBu: 0.0005%
S/N ratio: A-rated: -108 dBu
CCIR 468-3: -97 dBu
Transmission bandwidth: 8 Hz-200 kHz
Processed frequency range: 10 Hz-28 kHz
Phase: +5.5° @ 10 Hz
0° @ 1 kHz
-1.23° @ 10 kHz
-8.8° @ 100 kHz
-11.25° @ 200 kHz
Common mode rejection: › 70 dB @ 100 Hz, 1 kHz, 10 kHz
Dimensions: Standard EIA 19“ rack chassis (4 units)
Weight: 18,25 kg/40,15 lbs


Technical specifications external power supply

The PQ comes with an external linear power supply featuring a toroidal transformer for optimal audio quality and dramatically reduced inductive disturbance/interference.

Input voltage: 110-120V/60Hz or 220-240V/50Hz
Output voltage: +/- 60 V AC
Noise: › -100 dBu
Dimensions: Width: 15 cm (5 9/10 inch)
Depth: 24,5 cm (24 1/2 inch)
Height: 7 cm (2 3/4 inch)
Weight: 4,2 kg (9,24 lbs)

The Qure is a 2-channel, parametric, 3-band EQ, ideally suited for demanding tonal shaping of individual instruments, voices and complex musical signals. Mastering applications benefit from the unique tonal potential and the sophisticated filtering design that perfectly complements the fine details of the sound being processed.

The highly sophisticated circuitry of the Qure forms the basis for its outstanding tonal qualities: each filter is serially connected on separate boards with individual op-amps, resulting in excellent noise and distortion values, with a hard bypass available for each individual filter.

Features

• 2 x 3-band fully parametric EQ
• Proportional Q operation
• Variable HF and LF cut filters
• Unique QURE control
• Variable input gain (-12dB to +18dB)
• Variable output gain (-7dB to +4dB)
• Hard-Bypass for each filter band and both cut filters
• Master relay hard-bypass for entire unit
• Central Grounded Shield Layout
• SPL's Super-Balancing Hybrids (CMRR > 80dB)
• Tube warm up circuitry for prolonged tube-life
• Overrated internal toroidal power supply
• OPTION: Lundahl I/O transformers

The MK2-T (model 9739) represents the tube version of the best selling Vitalizer model, the Stereo Vitalizer MK2 (model 9526).

The MK2-T employs 12 AX 7 LPS-tubes for the mid-hi processing and in the Stereo Expander circuitry, where the summed signal is processed.Coil filtering stages complement ideally the vintage tube sound characterisitcs.

Great emphasis was laid on a harmonic tuning of the control parameters for high precision processings.

The MK2-T fits into the range of Vitalizer products between the standard Stereo Vitalizer MK2 and the high-end mastering version, the Tube Vitalizer (Model 9530).

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MMC 1
Model 2160

Multichannel Mastering Console
 

Fade In | New Technologies | The MMC 1 configuration in short | The sections in detail | Master/Output
 

As the first manufacturer worldwide, SPL has developed a stereo and multi-channel mastering console. The goal of the development was a console which is superior in audio quality to all known and foreseeable audio formats, whether analogue or digital, allowing an unrestricted reproduction of the sonic quality of SACD and DVD-Audio and to be a safe capital investment
The MMC 1 operates in the centre of a mastering environment fulfilling the tasks of speaker management, source connectivity, audio metering, track assignment, master and monitor level setting and automated insert routing of external processors.

 

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Fade In ...
Digital audio formats are subject to further development and change. The degree of incompatibility enforced by the “format war” between PCM and DSD has persuaded us to decide for a technology that is superior in dynamic range, headroom and sound quality – and that is discrete analogue technology in its most advanced implementation.

And there are further requirements speaking for the employment of high-performance analogue technology:

• The number of necessary AD/DA conversions should be reduced to a minimum. Digital sources can be connected to a digital router (i.e. of Z-Systems), which outputs the selected source through the preferred DA converter to the MMC 1. Thus it is ensured that the sound quality remains comparable and is not affected by converter differences.
   
• From a sound-aesthetical view, high-quality analogue outboard processing is superior to digital processing. The analogue concept allows for problem-free integration of those processors.
   
• Monitors and power amplifiers are mostly analogue designs. Why have another converter in that chain?
   

New Technologies
In the MMC 1 SPL’s new SUPRA operation amplifiers are used throughout. They operate at 120 V operating voltage. During a four-year period, SPL has researched this discrete operation amplifier, until the basis of a new generation of analogue audio technology was found. The SUPRA operation amplifier obtains a signal-to-noise ratio of 116dB with an headroom of 34dB. The dynamic range amounts to 150dB with a frequency bandwidth ranging to 200kHz.

With these basic specs, the MMC 1 is beyond the requirements of today's PCM digital formats up to 24 bit and 192kHz sampling rate or DSD digital format with 1 bit and 256 fs.
It is not to be expected that digital technology will offer an environment in foreseeable time, in which the MMC 1 could become a “bottle neck”.

The MMC 1 configuration in short

• In the Source section the Inputs (4x stereo/ 4x 8-channel) and Returns (8x stereo/ 8x 8-channel) can be selected.
• The Input section offers a passive router to re-assign the track configurations of the various surround formats to match the SMPTE/ITU standard. Each input channel is equipped with an On, Phase Reverse and a special Trim switch for precise gaining in 0.5dB steps from -9,5dB to +6dB.
• The Insert section is quite unique. It offers control functions for an automated patchbay, called Insert Box. Up to eight 8-channel processors can be connected to this outbreak 19” unit. The mastering engineer can store and compare up to four sequences by a push of a button. An all-over bypass switches the Insert Box in and out of processing.
• The Monitor section features a central Monitor Level control and switches for Mute and Dim levels. Two stereo loudspeaker sets and two surround loudspeaker sets of up to eight speakers can be connected to the MMC 1.
  The speaker management offers an On and a Solo switch for each loudspeaker. The Solo function operates as Solo-in-place. With the Solo-to-Center switch each speaker can be monitored through the centre speaker for better comparison. There is provision to monitor the LFE on the L/C/R. Three Mono functions (L/R; LS/RS; Lt/Rt) and two mode switches for stereo or multi-channel operation round up the Channel Selection section. The meter bridge houses a RTW Surround Monitor 10800x and eight big VU meters with superb ballistics. The remote control for the RTW Surround Monitor is already built into the console next to the calibration switches for the VUs. The VUs can be switched to either show input or output. The VUs can be calibrated to eight different values (0 dB/-2/-4/-6/-8/-10/-12/-14 dB).
• The Master section is dominated by a second eight channel level control. With the Master Level control, the overall output level of the desk to the recording sends is governed. The range of adjustment is from -10dB to +10dB.
• The Output section offers the same Trim switch for precise gaining as used in the Input section. Level differences, which may have been introduced by outboard processing, can be compensated for each channel in 0.5dB steps ranging from -9.5dB to +6dB.

The sections in detail
 

Source
The Source section is divided into two departments. On the left side are the selectors for the input sources, which undergo the mastering process. Four stereo sources and four 8-channel sources can be connected to the MMC 1.
On the right side are the selectors for the returns of the recorders, DAW, analogue multi-tracks, SACD- and DVD-Players, TV, AC3/DTS encoder/decoder etc. The Input/Return button switches the monitoring to a total of eight stereo and eight multi-channel returns.
The inputs and the returns can be attached on the back of the MMC 1 both to XLR sockets and to EDAC multi-pin sockets.
 

Input
The Input section receives the signal selected in the Source section. First the signal goes through a passive routing switch, with which the individual channels can be routed to every other channel. This function is essential regarding the different channel configurations of the surround formats.
A table above the routing switch gives an overview of the most important channel configurations:
DTS: L / R / LS / RS / C / LFE / L(t/o) / R(t/0)
Film: L / LS / C / RS / R / LFE / L(t/o) / R(t/0)
SDDS: L / LC / C / RC / R / LFE / LS / RS
The MMC 1 buss structure for all inputs and outputs follows the SMPTE/ITU track assignment:
SMPTE: L / R / C / LF / E / LS / RS / L(t/o) / R(t/0)
Note: L(t/o) and R(t/0) explained: The appendix ‘t’ means ‘total’ and refers to the automatic stereo downmix function within AC3-encoders, whereas the appendix ‘o’ means ‘only’ and stands for a separate stereo mix.
The routing selector transfers each conceivable channel configuration into the SMPTE configuration. The high-quality switch has gold-plated silver contacts with a life span of over 25.000 switching cycles.
Audio then runs through an On and a Phase Reverse switch followed by the 32-position Trim switch. The range of trimming is -9.5dB to +6dB in 0.5dB steps. The specialty of this switch is its mechanical architecture that interconnects only two contacts. Common switches route the audio through a chain of resistors switched in series. Thermal noise and tolerances are adding up. The MMC 1 Trim switch avoids that by routing the audio through only one 0,1% metal film resistor at any position.

Insert
While it is relatively easy in a stereo environment to connect processors via a patchbay and specify their sequence, it becomes complicated and time consuming when dealing with surround.
The MMC 1 features an Insert-Box to which up to eight 8-channel processors can be connected. The unique advantage of MMC 1 is that the engineer can specify up to four routings, called sequences, which can be stored and re-called.
The Insert section provides a switch for each of the eight external processors. Depressing them in a sequence specifies the signal flow through the processors. Beside each switch is a seven segment LED-display indicating the current position of the processor in the sequence.
The mastering engineer can use this feature to compare between sequences in a varying order or to compare the same type of processor like equalisers of manufacturer A with manufacturer B.
Three memory banks are available. Together with the current sequence, four different signal flows can be compared instantly. A bypass switches the Insert Box to hard-bypass.

Monitor
The audio signals returning from the Insert Box are sent to the Monitor section and coevally to the recording outputs. The Monitor section is divided in the following sub sections:

Speakers
Two stereo pairs and two surround loudspeaker sets can be connected to the MMC 1 for monitoring.

Channel Selection
Each loudspeaker can be switched on with the On switch, which is labelled with the respective loudspeaker position. Underneath them a Solo function is provided for each loudspeaker. Multiple solo is also possible.
The Solo-to-Center function allows for monitoring of each loudspeaker through the centre speaker to obtain a better comparability. Solo-to-Center can only be activated, if a speaker was switched to solo before. In a multiple solo situation the Solo-to-Center cannot be activated.
The LFE-to-L/C/R switch distributes the LFE signal proportionately on L/C/R.
In the Monitor section two switches are provided to change the MMC 1 configuration from stereo to multi-channel. The stereo switch should be pressed before starting a stereo mastering job to switch off all monitoring functions except for L/R for improved operation safety. When depressing Multi-Channel, the MMC 1 resets itself to the last multi-channel configuration.
The Monitor section offers three mono functions:
1. Mono L/R
2. Mono Rear (cannot be enabled in stereo mode)
3. Mono L(t/o)/R(t/0) (cannot be enabled in stereo mode)

Monitor Level
The Monitor Level is regulated with a genuine custom-made potentiometer with eight chambers. The MMC 1 avoids using DACs, steps ladders or VCAs for this operation. The specifications of this potentiometer are impressive. The maximum tolerance is 0,5 dB over the entire control range! And such a masterpiece deserves an appropriate optical frame: this noble potentiometers is moved with an 60 mm diameter knob of massive aluminium. The scaling is illuminated with 30 blue LEDs on a circular area of 120 mm with a pointer element from miniature orange LEDs.

Dim/Mute/Back
Three Dim levels are available (-10 dB, -20 dB and -30 dB). In addition to the Mute function a switch labelled Back is incorporated. It is used in case a Dim function is activated and afterwards Mute was pressed. The Back function immediately returns to the actual Monitor Level setting without releasing Dim and Mute.

Metering
Right of the Monitor Level potentiometers are the switch functions for the VUs and the RTW Surround Monitor. All functions of the RTW can be controlled remotely. They are placed next to the Monitor Control potentiometer for easy access. Thus the mastering engineer does not need to leave the optimal hearing position.
The VUs are likewise custom-made by a Japanese company, which manufactures VUs with the optimal ballistics. The VUs can be calibrated on eight different reference levels (0 dB/-2/-4/-6/-8/-10/-12/-14 dB). Furthermore the VUs can be switches to indicate either input or output levels.

Master/Output
The MMC 1 provides three eight channel and four dual channel recording outputs. The output section features the same 32-position switch for level trimming in 0,5dB steps as in the input section. Additionally a Master Output level is provided to regulate all eight channels simultaneously. The same potentiometer is used as for the Monitor Level regulation. With this potentiometer, extremely fine recording level settings are achievable to retrieve the very last bit of the recording headroom. Owing to a genuine potentiometer with infinite resolution a dynamic mastering is possible.

With the Atmos 5.1 Surround Miking System SPL has developed a compact location recording and mixing/premastering console fulfilling the demands of today's 5.1 recordings. Brauner's 5-channel Adjustable Surround Microphone ASM 5 is ideally complementing the Atmos 5.1 controller. Although any existing microphone set-up may be used with the Atmos 5.1, the combination of Atmos 5.1 and ASM 5 supports latest knowledge in surround recording and offers a complete system that can be set-up in a fraction of the time needed to set-up conventional systems. The fully analogue Atmos 5.1/ASM 5 Surround Miking System supports any multichannel media like DVD and SACD and it is independent from the encoding process (PCM, DSD, AC3, DTS, SDDS, ISO/MPEG2). Comprising a main unit and a separate power supply, the Atmos 5.1 occupies 5U of rack space with the PSU being just 1U. To record a coherent room sound for 5.1 has proven to be difficult. Today's recording methods involve numerous microphones (some close-up plus additional room mics) and a desk capable of handling dozens of channels. Nevertheless, due to the individual phase responses of each microphone a "real" reproduction of the room sound is hard to achieve. Other systems generate the L/R, SL/SR, centre and sub from microphones with four capsules exhibiting certain shortcomings when reproducing the recording in 5.1. The combination of Atmos 5.1 and ASM 5 has proven in various recording sessions of Live Jazz and Big Band events as well as classical recording sessions to be very accurate in reproducing the listener's position and creating an atmosphere of "being there". The Atmos 5.1 Controller 1.) The Triple-Gain-Stage Microphone Preamps The Atmos 5.1 is equipped with five matched high precision microphone pre-amps featuring SPL's new Triple-Gain-Stage microphone pre-amps to capture the superb sound of Brauner's ASM 5 microphone in the most transparent, noiseless and uncoloured way. An important feature of the Atmos 5.1 are its motorised Master Gain controls. A Master Gain switch enables motorised control over all five microphones preamplifiers by just turning one control. This is important, when re-adjustment of the pre-amplification becomes necessary while recording. Re-adjusting the pre-amplification manually may result in losing the coherence of the room sound and will cause phasing effects between the channels when reproducing in 5.1. While changing the pre-amplification the relative loudness relationships between all five microphones are maintained. The motorised Master Gain controls have proven to be most helpful in certain recording situations. The microphone preamps also feature SPL's ServoDrive-Technology which detects voltage differences (DC-offset) between the positive and negative paths of the amplifying stages. Any offset increases noise and distortion and therefore compromises the signal quality. ServoDrive minimises DC-offsets to values between 0mV and 2mV. The recorded signal contains less noise and distortion and improved tonal transparency. The mic preamps also employ Lundahl input transformers, pads, phase reverse, phantom power, low cut filters and a switchable insert, tape send/returns. All switches are luminated, high quality switches, and relays with gold plated contacts are used throughout.   2.) 5.1 Surround Panning Matrix While recording on location or mixing in the studio it may be necessary to vary the position of a microphone in the Front/Surround and L/C/R panorama to compensate for poor microphone positioning or for artistic reasons. Mixing and 5.1 panning is possible from each channel via a Front/Surround and a L/C/R pan control with divergence for correcting the L/C/R image.   3.) 0.1 Sub Channel & LFE (Low Frequency Effects) In the 0.1 Sub Channel, the sub signal can be composited from the front, surround, and centre channels. A 24dB Butterworth low-pass filter at 130Hz can be activated to only let those frequencies pass up to where localisation begins. In the mixing/premastering stage the upper corner frequency can be reduced to the value required for i.e. AC3 or DTS encoding. If the low-pass filter is not activated a mono composite of the selected mic inputs can be send to a separate sub/LFE processor.   4.) Stereo In/Out Two additional balanced inputs are provided to mix a stereo source from additional room mics or from a multichannel sub mix to the front, centre and surround channels. The STEREO OUTs allow the stereo source to be forwarded to other units for further processing or recording.   5.) ASM 5 Pattern Control/Stereo Expanders Another difficulty in 5.1 recording represents the selection of the polar patterns of the microphones in order to achieve the optimum in capturing the room atmosphere and the sound source within. There is great demand for experimentation and the Atmos 5.1 system delivers maximum freedom in doing so. The Brauner ASM 5 allows continuous adjustment of the polar pattern characteristic of each microphone from omnidirectional up to figure-of-eight. These adjustments can be made directly from the Atmos 5.1 and can be monitored while recording. Furthermore, the stereo soundstage of the front and surround channels can be widened using all-pass stereo spreading. This allows the electronic simulation of a continuously variable distance between the L/R and SL/SR microphones. The Stereo Expanders are equipped with a hard-bypass and a mono switch. Two phase meters display the L/R & SL/SR correlation both when the Stereo Expanders are active or in bypass.   6.) Monitoring Monitoring six channels represents another problem in 5.1. While recording and even more important while mixing and premastering the variation of the monitor level is delicate because level differences of just 0.5dB may cause phasing effects between the channels. SPL has developed an electronic circuitry to realise monitoring level control of all six channels simultaneously with minimum tolerance. The monitoring speakers can directly be fed from the Monitoring Outputs of the Atmos 5.1 solving the monitoring problem in the studio. When recording on location the Sennheiser Surrounder headset can directly be plugged into the Atmos 5.1. A conventional stereo headphone output is also provided allowing each bus to be monitored separately or in combination.     The Brauner ASM 5 Adjustable Surround Microphone The ASM 5 is based upon five matched microphone heads from the highly acclaimed Brauner VM 1 microphone. This microphone system represents a specific microphone set-up that has proven to be superior in blindfold tests over other surround recording techniques (ABC, IRT-Surround and other ambisonic techniques). The Brauner ASM 5 offers a large flexibility and variability to capture any sound event and offering the recording engineer a platform for experimentation and artistic freedom. Variability The L/C/R microphone heads of the ASM 5 are positioned in a rectangle triangle with each microphone being positioned 17.5 cm away from the centre. The ASM 5 allows continuous adjustment of the polar pattern characteristic of each microphone from omnidirectional up to figure-of-eight controllable remotely from the Atmos 5.1. Furthermore the positions of the all microphone heads are variable +/-90 degrees. The BRAUNER ASM 5 includes a top quality 25 m (or more if needed), 12-pair screened multicore cable to connect it to the SPL Atmos 5.1 Surround Recording Console. Together, the SPL Atmos 5.1 and the Brauner ASM 5 provide the optimum 5.1 surround miking solution for studio and live recording, outdoor recordings, mixing and post production in music, TV and movie productions.

PQ Parametric Mastering Equalizer, Modell 2050 Who needs another equalizer? Check this: 120 Volt internal operating voltage for clean, consistent audio Pure analog, discrete Class A equalization circuitry Constant Q and proportional Q equalization (selectable per band) Digital storability and total recall with motorized controls Channel link and master/slave unit link modes (deactivatable per band) Remote control unit controls up to four PQ‘s for surround processing* SPL‘s discrete SUPRA op-amps with 150 dB dynamic range Exactly the equalizer you‘ve been wishing someone would build ... The PQ from SPL. Not just another equalizer. Detailed information

MMC 2 Modell 2486 Multichannel Mastering Console on 120V rails In Short The MMC2 is the compact 19” rack format version of the fully-featured MMC1 multichannel mastering console, based on the same unsurpassed 120 volts rail to achieve optimum audio performance. Conceived as the center of a mastering environment, the MMC2 fulfills the tasks of speaker management, multiple inputs and returns connectivity, input and output trimming as well as pure analog continuously variable 8-channel master fader and monitor level setting. An automated 8 x 8 channel insert box with remote control for routing of external processors is optionally available (SPL MasterBay). In Detail

MixDream XP Model 2591 Analog Summing Device SHORT DESCRIPTION The MixDream is a cascadable, 16-in-2 analog mixer in a 19“/1U format. Any DAW or digital console can be expanded with high-grade analog stereo summing functionality. MixDream Advantages—An Overview Active analog summing on just 1U rack space—no analog mixing console necessary Sophisticated, active Class A/6oV stages for analog summing in the quality of the best consoles All analog tracks can be summed before A/D conversion. Channel adjustments and automation (level, panorama etc.) remain controlled from the DAW so the user loses no digital efficiency Lower DAW processor utilization rates Latency free monitoring Surround capable (from up to 3 MixDreamXP units) Channel capacity expandable through linked units Proprietary differential amplifiers for each input