SPL

The SPL IRON Tube Mastering Compressor comes in a new revision with features improving operation and sound. This updated hardware revision can be recognized by the “IRON v2” labeling on the serial number sticker. IRON v2 now has Side Chain Inserts with XLR sockets instead of Side Chain Inputs with TS sockets only. See picture below.SPL IRON 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.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.Balanced Side Chain Inserts are now standard using send and return XLR sockets instead of Side Chain Inputs with TS sockets only. That makes it easy to add an equalizer of choice into the control voltage path to create individual filter curves to focus the response of the compression to certain frequency ranges.IRON v2 is now equipped with Wathen CryoTone 12AX7-WCM Long Plate tubes. These are the perfect match to the parallel connected 12AU7 tubes in its circuit, which results in an improved sound and longevity.COMPRESSORThe 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 dynamics and punch to any production. The number of compressors available is huge and it‘s easy to succumb to the promises made by software emulations and analog recreations of vintage gear as the perfect solution. Unfortunately, many of these emulations and recreations differ quite a bit from 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.VARIABLE BIASThe 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.TUBE BIASThe Tube Bias has three different settings. Together with the Input Gain of up to ±12 dB 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).MU-METAL TRANSFORMERMoreover, 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.VACTROL PEAK LIMITERThe 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.RECTIFIERSThirdly, 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.STEREO LINKThe 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.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.AIR BASS AND TAPE ROLL-OFFThe 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.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.EASY RECALLAll functions can be adjusted via switches or, in the case of the Threshold, with a detented potentiometer, making it easy to replicate exact settings.SIDECHAIN EQThe 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.OUTPUTAfter 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.AUTO BYPASSBoth 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.FEATURES:Variable bias tube compressorParallel dual tube circuitTube selection and pairing with Weigl Roe TestSpecially conceived Mu-Metal iron transformersFeed-forward resistive vactrol-opto-isolatorSix different Rectifier settings availableSidechain that allows you to choose between Off, four sidechain-filter presets or an external sidechain signal.All functions can be adjusted via switches or, in the case of the Threshold, with a detented potentiometerIn 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.Additional passive 120 V equalizer with two presetsHigh Dynamic 120 V operating voltageAuto Bypass modeMade in Germany

Eight-channel mic preamp with discrete transistors, analog gain, 48V phantom power, PAD, phase inversion, high-pass filter, ADAT optional, 192kHz, robust and sleek designThe SPL P8 is a high-end, eight-channel microphone preamplifier designed to offer pristine audio quality for multi-channel recording. Each of the eight preamps is built with discrete technology, ensuring a signal path free from integrated circuits (ICs). This design, utilizing individual transistors, is tailored for audiophiles and professionals who prioritize sound purity, contributing to the unit's open and transparent sonic characteristics. The preamplifiers provide a flexible gain range from +8 dB to +62 dB, making the P8 suitable for a wide range of microphones, including those requiring higher gain.Key features include individual controls for each preamp, such as a 48V phantom power switch for condenser microphones, a 20 dB PAD for handling loud sources, a high-pass filter to eliminate subsonic noise, and a phase inversion function. For digital workflows, the P8 can be equipped with an optional eight-channel ADAT converter, allowing seamless connection to digital audio interfaces. This versatility makes the P8 ideal for both analog and digital recording environments, delivering unparalleled audio performance while maintaining ease of integration into various studio setups.FeaturesEight Preamplifiers: Equipped with eight discrete microphone preamps for high-end sound quality.Gain Control: Adjustable gain from +8 dB to +62 dB for handling various microphone types.Phantom Power: Each channel has switchable 48V phantom power for condenser microphones.Signal Processing: Includes PAD switch (20 dB reduction), high-pass filter, and phase reverse for each preamp.Analog & Digital Outputs: Offers both analog outputs and an optional ADAT digital output for direct digital connectivity.Durable Build: Features a robust aluminum and steel construction for long-lasting use.

SPL Pre One OverviewEnjoy the open and transparent sound of the SPL Channel One Mk3 in a compact, dual-channel preamp with the Pre One, a desktop unit with two discrete preamps. With a wide gain range of 8 to 62 dB and optional Hi-Z instrument input, you can draw out the full glory of nearly any input source, from condenser mics to passive instruments.Each preamp features an independent 6 dB per octave high-pass filter for subsonic rumbles, switchable 48V phantom power, and a 180° polarity reversal switch. Each preamp features a selector, so you can choose Flair or Contour, analog EQ presets, and on the rear panel, you have a channel-selectable -20 dB input pad for overly hot signals. Each preamp features a balanced XLR 3-pin mic input and its own dedicated 1/4" TRS balanced output. Preamp 2 also includes a 1/4" TS instrument input (which automatically deactivates the mic input on preamp 2). This design allows the Pre One to process stereo signals if desired. Like all SPL products, the Pre One is manufactured in Germany.Gain ControlThe gain control can be used to adjust the preamplification. The control range ranges between +8 and +62 dB, so even truly demanding microphones can unfold their full potential. Each preamp gets its own large gain control knob. A green signal LED is provided for both preamps to indicate that a signal is present.Switchable FunctionsA high-pass filter with 6 dB per octave is available in both preamp sections and reduces subsonic noise below 80 Hz when activated. This low-cut filter can be used for the microphone inputs as well as for the instrument input on the second amplifier.Both preamplifiers have a 48V switch. This switch activates the required phantom power of 48 volts for the use of condenser and active dynamic microphones. The phase reverse function reverses the polarity of the signal. After pressing the switch, the phase is reversed by 180°.Perfect Sound with Flair and ContourFor each of the two preamps, Flair and Contour provide an analog EQ preset to perfect the sound. When Flair is activated, it gives the signal a beautiful range of high frequencies; silky highs is the key word here. When Contour is activated, the signal perfectly blends into a compact mix while retaining its natural presence and strength.Input PadThe pad switches for each preamp section are located as DIP switches on the rear of the Pre One. These switches attenuate the signal of the microphone input by 20 dB, so that even very high levels can be processed with the Pre One. This is the case, for example, with loud percussion or brass recordings.Power SupplyGood sound always starts with the power supply. This is where you lay the foundation for the performance of the entire device. Although SPL supplies a 12V plug-in power supply unit, the secondary power supply inside generates a voltage level of +/-17V for the analog audio sections with which professional levels of up to 22.5 dB can be achieved. Elaborate low-drop operating-voltage regulation provides more juice and regulates cleanly even when the maximum operation voltage is reached.In the BoxSPL Pre One Dual-Channel Microphone Preamplifier (Black)12V Power SupplyLimited 2-Year Manufacturer WarrantyKey FeaturesFor Studios and AudiophilesTwo Discrete, Transparent PreampsMic and Hi-Z Input on Preamp 2Dual 8 to 62 dB Gain Controls6 dB per Octave HPF per Channel48V Phantom Power and Phase ReversalSwitchable Flair and Contour EQ Presets-20 dB Pad per Mic InputSeparate 1/4" TRS Line Out per PreampMade in Germany

Ο Phonitor 3 DAC είναι ένας ενισχυτής ακουστικών, ελεγκτής παρακολούθησης και DAC με τεχνολογία 120V, που προσφέρει την απόλυτη λύση τόσο για παραγωγούς όσο και για ηχολήπτες και μηχανικούς mastering που δουλεύουν με ψηφιακές πηγές ή σε DAW. Το ενσωματωμένο DAC τον καθιστά ιδανικό κεντρικό σημείο παρακολούθησης.Είτε πρόκειται για USB, AES/EBU ή S/PDIF, το DAC με αναλογικό SLP120 μετατρέπει ψηφιακά σήματα PCM με ανάλυση έως 32 bit και συχνότητα δειγματοληψίας έως 768 kHz, ενώ μετατρέπει και σήματα DSD με ανάλυση έως DSD256.Στην αναλογική πλευρά, ο Phonitor 3 DAC, βασισμένος στην τεχνολογία 120V της SPL, προσφέρει την ίδια ποιότητα παρακολούθησης όπως οι μεγάλες mastering κονσόλες της SPL, τόσο σε ηχεία όσο και σε ακουστικά. Η αναλογική τεχνολογία Phonitor Matrix επιτρέπει τη μίξη και το mastering μέσω ακουστικών με την υψηλότερη ποιότητα, προσφέροντας την ίδια αίσθηση του στερεοφωνικού πεδίου όπως με ηχεία.Εξαιρετικός ήχος – για ακουστικάΟ Phonitor 3 προσφέρει σύνδεση για ακουστικά μέσω στερεοφωνικού βύσματος στην μπροστινή πλευρά. Χάρη στην τεράστια ισχύ εξόδου, μπορεί να οδηγήσει κάθε τύπο ακουστικών με άνεση, αποδίδοντας έναν έντιμο, λεπτομερή και ζωντανό ήχο.Εξαιρετικός ήχος – και για ηχείαΣτην πίσω πλευρά της συσκευής, ο Phonitor 3 προσφέρει ένα στερεοφωνικό έξοδο, κάνοντάς τον έναν υψηλής ποιότητας ελεγκτή παρακολούθησης για οποιαδήποτε κατάσταση. Μπορούν να συνδεθούν ενεργά ηχεία ή παθητικά με ενισχυτή.Παρακολούθηση στο υψηλότερο επίπεδοΟ Phonitor 3 προσφέρει την ίδια βασική τεχνολογία – SPL 120V – και την ίδια ποιότητα σήματος όπως οι μεγάλες mastering κονσόλες SPL DMC και MMC.

Optimize Your Audio with the Stereo VitalizerThe SPL Stereo Vitalizer Mk3-T harmonic exciter is an indispensable tool for enhancing your audio. Based on research in the fields of psychoacoustics and audiometry, SPL’s unique, patented Vitalizer circuitry heightens detail and transparency, boosts perceived loudness, and beefs up bass response for powerful, radio-ready mixes that blast right out of the speakers. The key to the Stereo Vitalizer Mk3-T’s sonic magic is that it works both in the frequency and time domains through a series of minimal temporal offsets of loud frequencies. As a result, superimposed audio transients are unmasked to unveil a whole new level of clarity in your mixes. For decades, recording engineers have used subtle harmonic excitement to help elements sit perfectly in the mix. Now, with the SPL Stereo Vitalizer Mk3-T harmonic exciter, you can too.An indispensable sound-design toolSPL's Vitalizer technology is known by engineers worldwide as an effective way to polish audio with highly musical results. Use the Drive control to fine-tune the signal level for precise interaction with the filter network and bring out the best parts of an audio signal. The Stereo Vitalizer Mk3-T is unsurpassed as a sound-design tool. It adapts the frequency spectrum to the curves of equal loudness, heightening the perception of loudness. It can enhance intelligibility in vocals and augment the soundscape of a complete mix — and more. Low frequencies are subtly shifted to prevent them from masking transient details, giving your tracks and mixes more clarity than ever. The integrated Bass Comp focuses on the bass range so you don't overload downstream equipment in your signal chain. So, even if you use the Stereo Vitalizer Mk3-T to bolster your bass tracks, you still have the dynamic control you need to keep signal levels in check.The cure for iron-poor mixesPrevious versions of the Stereo Vitalizer have proved their mettle in top studios for years. Now, the third-generation Stereo Vitalizer Mk3-T ups the ante with a new design that uses a higher internal audio voltage of +/-18V, resulting in enhanced dimensionality and increased detail. Processed through the Vitalizer Mk3-T, you’ll notice your mixes bloom with vitality and richness of detail that makes them more accessible to the ear. They brim with enhanced depth and width, more deeply layered and widely spread apart. Authoritative, natural, and lively-sounding, your mixes will translate better to the world outside your studio, holding up well on real-world playback systems where lackluster mixes go to die. So do your mixes a favor. Energize them with the SPL Stereo Vitalizer Mk3-T.SPL Audio: supercharged audio productsFormed in Germany in 1983, SPL Audio has been at the forefront of the pro audio industry since its inception. After developing proprietary 120V Technology in the ’90s, SPL decided to make this groundbreaking invention the heart of all its high-end products, and this incredibly powerful yet transparent sound has earned many devotees at Sweetwater and beyond. Other revelatory inventions SPL has become known for include level-independent envelope processing and using phase cancellation in auto-dynamic de-essers for reducing sibilance. Sweetwater is proud to offer audio professionals SPL’s distinguished lineup of innovative products.

Get That Authoritative, Professional SoundNow in its third iteration, the ever-popular SPL Track One MK3 Premium channel strip has been extensively updated with upgraded specs and a more intuitive design optimized for the modern recording studio. This version of the Track One MK3 Premium is spec'd with premium Lundahl input and output transformers for that authoritative "big-iron" sound that leaps right out of the speakers. From broadcasting and voiceovers to music recording, podcasts, and live events, Track One MK3 Premium always delivers superior sonics, enhancing vocals and instruments from any source, be it mic, line, or DI. The stellar mic preamp provides plenty of gain for your mics and instruments, and SPL's acclaimed De-Esser is onboard for taming sibilant vocalists. Optimize your dynamics with the compressor/limiter, sweeten your tone with the 3-band EQ, and you've got radio-ready sound from the SPL Track One MK3 Premium channel strip.Your solution for fast, high-quality recordingsWe talk to many musicians at Sweetwater who need to be able to produce professional recordings quickly — guitarists and bassists for hire, engineers making radio spots, the list goes on. The beauty of the SPL Track One MK3 Premium channel strip is that it's amazingly fast to dial in a great sound with virtually any source. The built-in processing is perfect for adding that extra studio polish without having to patch in external processors or bounce audio through plug-ins.Capture better vocals with the built-in De-EsserThe SPL De-Esser is considered an essential tool in countless studios, and it's built right into the SPL Track One MK3 Premium channel strip. It's known for removing harsh sibilance from vocalists in a natural, musical way, serving up transparent results even at extreme settings. One thing you'll find especially useful is the SPL De-Esser's automatic threshold-adjusting function. It basically compensates for a singer's varying distance from the mic, applying de-essing even with the signal is lower so that compressing afterward doesn't bring lower sibilants back up in level.Compressor/Limiter controls dynamics and adds characterThe Track One's compressor/limiter is an extremely easy-to-use one-knob design utilizing time constants (including attack and release) that are automated adaptively to the input signal, allowing for simple set-and-forget operation. Pressing the Limiter button switches the compressor into limiter mode, with the threshold being set by the compression knob. Like the compressor, SPL’s soft limiter is unobtrusive and delivers consistent, highly musical-sounding results. A separate makeup gain knob is provided for maintaining proper gain structure in your signal chain.Combine two for stereo operationFor maximum flexibility in your studio, get two SPL Track One MK3 Premium channel strips and link them for stereo compression. You'll be able to record beautifully wide stereo tracks and get true stereo compression with just one set of controls. Best of all, the EQs will still be independent, so you can fine-tune the stereo image simply by using slightly different EQ settings on each side.SPL Audio: supercharged audio productsFormed in Germany in 1983, SPL Audio has been at the forefront of the pro audio industry since its inception. After developing proprietary 120V Technology in the ’90s, SPL decided to make this groundbreaking invention the heart of all its high-end products, and this incredibly powerful yet transparent sound has earned many devotees at Sweetwater and beyond. Other revelatory inventions SPL has become known for include level-independent envelope processing and using phase cancellation in auto-dynamic de-essers for reducing sibilance. Sweetwater is proud to offer audio professionals SPL’s distinguished lineup of innovative products.

Channel One Mk3The perfect front-end for the modern producer.For over 20 years, Channel One has been a synonym for a high-quality and extremely musical recording and mixing channel strip.In the newest Mk3 version, this classic has been thoroughly revised and, in addition to a higher internal audio voltage (now +/-18 V) for even better, more detailed sound, a further improved preamplifier section, an integrated Transient Designer, a Tube Saturation stage and a Mic A/B comparison option for two microphones and other great features that raise the modern recording and mixing studio to a new level of quality. With de-esser, compressor and equalizer, all the important tools of a real channel strip are still on board. Whether it’s a microphone, line, or instrument signal, the Channel One Mk3 makes any source sound like a professionally recorded signal.The new design of the SPL Studio Series perfectly highlights the sonic qualities of this 3rd generation Channel One.Packed with must-have toolsThe SPL Channel One MK3 Premium channel strip packs a wealth of features, including a discrete preamp, 3-band EQ, compressor/limiter, noise gate, and de-esser, making it the ideal front end for your DAW. Plug into this stellar-sounding channel strip and enjoy the world-class sounds that you've heard in your head but have been unable to achieve in your studio. Whether you want to record vocals, instruments, or any other audio source, the SPL Channel One MK3 Premium channel strip will imbue your recordings with the lustrous, expensive sound you've been craving.

The puristic summing mixer with Class A technology.As an analog 16-in-2 Class A summing device, MixDream XP Mk2 offers impressive sound quality and brings together single tracks in a musical way. The spatial depth of the stereo stage and the harmonic coherency of the summed signals make mixing through an analog summing device a very special experience. The Mk2 generation of the MixDream XP offers the same outstanding audio quality as its predecessor thanks to the internal audio operating voltage of +/-30 V. In addition to the new design and an even better feel, the MixDream XP Mk2 offers another new feature: the individually switchable -18 dB level attenuation for the last four stereo channels. This is how the analog summing gets the best out of every mix!Analog SummingAnalog summing with the MixDream XP Mk2 brings intense depth, precise localisation and impressive spatiality. Individual instruments mix with smooth transitions. For the analog summing, individual tracks from the computer are sent via D/A converters into the MixDream XP Mk2. These are summed in the MixDream XP Mk2 and then recorded again as a stereo signal via an A/D converter in the DAW software. The MixDream XP Mk2 is designed for stereo summing of up to 16 audio tracks in the analog domain. In combination with digital production systems, the advantages of analog and digital techniques can thus be ideally combined: digital convenience is complemented by the sound quality that until now only the best analog consoles have been able to deliver. Conceptually, the reduction to high-quality analog summing without panorama controls and faders offers the great advantage of being able to retain the entire scope of computer automation. So when using the MixDream XP Mk2, the usual workflow with the DAW does not have to be changed, all techniques and options of a DAW can still be used without any restrictions.MonoThe first four channel pairs of the MixDream XP Mk2 (1/2 to 7/8) can be switched from stereo to mono via the Mono switch (red status LED lights up). This is important for mono signals that should be positioned exactly in the center of the mix (kick, snare, lead vocal, bass, …). In stereo mode, the two channels are generally panned hard left/ right. This basic setting is useful for all stereo signals. Panorama settings and automation of the DAW remain perfectly unchanged.-18 dBThe level of the last four channel pairs of the MixDream XP Mk2 (9/10 to 15/16) can be attenuated by 18 dB via the -18 dB switch (blue status LED lights up). Low level signals in the mix can thus be played back and get converted with a higher level, therefore with a higher digital resolution, from the DAW. This preserves more details and sonic characteristics. Tip: This can work wonders for digital reverbs (especially plug-ins)!OutputThe Output control adjusts the output level of the MixDream XP Mk2. The control range can be set from -10 dB to +5 dB. With the output control a perfect control for following AD-converters can be performed. Since the main and monitor outputs are configured in parallel, the output control always affects both output pairs.Variable OutputThe output control is switched on or off with the Variable Output switch. When the Variable Output switch is deactivated, the input level is maintained at 1:1 (Unity Gain).

Premium DAC with VOLTAiR technologyDiamond is the perfect DAC and preamplifier for everyone who plays music exclusively from digital sources. Diamond offers connectivity for six digital sources and an external word clock. The superior VOLTAiR technology and the premium DAC with the unequalled DLP120 combined with an analog volume control allowing for highest resolution at any volume makes the Diamond an one-of-a-kind DA converter that delivers finest soundstages with an unrivalled analog feel.Everything under controlThe volume is adjusted with a solid, milled aluminum, rotary knob. The Alps RK27 “Big Blue” potentiometer gives a pleasant “spoon in the honey” rotary feel.Source of joyUp to 6 digital stereo sources can be connected to Diamond.Diamond offers each a stereo USB and AES input as well as each two optical and coaxial stereo inputs. Via USB, both PCM audio and DSD audio can be converted. PCM audio is received in S/PDIF format, both coaxial (RCA), optical (Toslink F06) and balanced (AES/EBU). The source selection is done via the selection control on the front. The selected input is also shown in the dot matrix LED display: USB, COA1, COA2, OPT1, OPT2 or AES. After approximately two seconds, the display shows the type of clock source and the detected sampling rate.Always in timeDiamond can not only synchronize to the clock integrated in the digitally fed source signal. As a specialist for digital players Diamond offers a word clock input. The source of the clock signal can be selected via the CLOCK switch.Source = Diamond synchronizes to the digital clock in the source signal.Word = Diamond synchronizes to the digital clock present at WORD IN.Well informedThe DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer, CD player or an external word clock is connected. The detected sampling rate and the selected clock source are shown in the dot matrix LED display. For example, U768 is an acronym for a clock signal in the USB stream with a sampling rate of 768 kHz.All variable – or Slave ThruDiamond provides two stereo output pairs for connecting power amplifiers or active loudspeakers. One pair of outputs with XLR, the other with RCA outputs. Both have the same output signal, which is controlled by the volume control on the front panel. The volume control for both outputs can be individually switched out of the signal path via DIP switch on the back of the device. For example, a headphone amplifier with its own volume control can be connected to one of the outputs.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Sounds goodWith all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. Many important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltageThe 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.The 120V technology achieves exceptional technical specifications and sonic benefits. Technically, in terms of dynamic range, signal-to-noise ratio and headroom. Sonically, in terms of richness of detail and an absolutely relaxed listening experience. By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.VOLTAiR is a composition of the terms Volt and Air. Volt is the unit for electrical voltage and Air stands for the unlimited space the music can breathe in. It symbolizes the perceived limitless dynamics resulting from high audio voltage.ComparisonMost audio devices work with an internal operating voltage of +/-15 volts and can thus process a maximum input level of +21.5 dBu. If a DAC, for example, has an output level of +22 dBu at 0 dBFS, level peaks of the music material would already cause overloads in the input stage of the device. All components in the audio device often operate at their limits. The result is an unsteady sound that causes stress and faster ear fatigue.SPL devices with VOLTAiR technology can handle input levels of +32.5 dBu thanks to the higher internal operating voltage of +/- 60 volts – thus offering 12 dB more headroom. All components consequently operate continuously in the optimum operating range. The result is a very pleasant, natural and relaxed sound experience. So you can enjoy your music in every detail.

Premium DAC with VOLTAiR technologyDiamond is the perfect DAC and preamplifier for everyone who plays music exclusively from digital sources. Diamond offers connectivity for six digital sources and an external word clock. The superior VOLTAiR technology and the premium DAC with the unequalled DLP120 combined with an analog volume control allowing for highest resolution at any volume makes the Diamond an one-of-a-kind DA converter that delivers finest soundstages with an unrivalled analog feel.Everything under controlThe volume is adjusted with a solid, milled aluminum, rotary knob. The Alps RK27 “Big Blue” potentiometer gives a pleasant “spoon in the honey” rotary feel.Source of joyUp to 6 digital stereo sources can be connected to Diamond.Diamond offers each a stereo USB and AES input as well as each two optical and coaxial stereo inputs. Via USB, both PCM audio and DSD audio can be converted. PCM audio is received in S/PDIF format, both coaxial (RCA), optical (Toslink F06) and balanced (AES/EBU). The source selection is done via the selection control on the front. The selected input is also shown in the dot matrix LED display: USB, COA1, COA2, OPT1, OPT2 or AES. After approximately two seconds, the display shows the type of clock source and the detected sampling rate.Always in timeDiamond can not only synchronize to the clock integrated in the digitally fed source signal. As a specialist for digital players Diamond offers a word clock input. The source of the clock signal can be selected via the CLOCK switch.Source = Diamond synchronizes to the digital clock in the source signal.Word = Diamond synchronizes to the digital clock present at WORD IN.Well informedThe DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer, CD player or an external word clock is connected. The detected sampling rate and the selected clock source are shown in the dot matrix LED display. For example, U768 is an acronym for a clock signal in the USB stream with a sampling rate of 768 kHz.All variable – or Slave ThruDiamond provides two stereo output pairs for connecting power amplifiers or active loudspeakers. One pair of outputs with XLR, the other with RCA outputs. Both have the same output signal, which is controlled by the volume control on the front panel. The volume control for both outputs can be individually switched out of the signal path via DIP switch on the back of the device. For example, a headphone amplifier with its own volume control can be connected to one of the outputs.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Sounds goodWith all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. Many important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltageThe 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.The 120V technology achieves exceptional technical specifications and sonic benefits. Technically, in terms of dynamic range, signal-to-noise ratio and headroom. Sonically, in terms of richness of detail and an absolutely relaxed listening experience. By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.VOLTAiR is a composition of the terms Volt and Air. Volt is the unit for electrical voltage and Air stands for the unlimited space the music can breathe in. It symbolizes the perceived limitless dynamics resulting from high audio voltage.ComparisonMost audio devices work with an internal operating voltage of +/-15 volts and can thus process a maximum input level of +21.5 dBu. If a DAC, for example, has an output level of +22 dBu at 0 dBFS, level peaks of the music material would already cause overloads in the input stage of the device. All components in the audio device often operate at their limits. The result is an unsteady sound that causes stress and faster ear fatigue.SPL devices with VOLTAiR technology can handle input levels of +32.5 dBu thanks to the higher internal operating voltage of +/- 60 volts – thus offering 12 dB more headroom. All components consequently operate continuously in the optimum operating range. The result is a very pleasant, natural and relaxed sound experience. So you can enjoy your music in every detail.

Premium DAC with VOLTAiR technologyDiamond is the perfect DAC and preamplifier for everyone who plays music exclusively from digital sources. Diamond offers connectivity for six digital sources and an external word clock. The superior VOLTAiR technology and the premium DAC with the unequalled DLP120 combined with an analog volume control allowing for highest resolution at any volume makes the Diamond an one-of-a-kind DA converter that delivers finest soundstages with an unrivalled analog feel.Everything under controlThe volume is adjusted with a solid, milled aluminum, rotary knob. The Alps RK27 “Big Blue” potentiometer gives a pleasant “spoon in the honey” rotary feel.Source of joyUp to 6 digital stereo sources can be connected to Diamond.Diamond offers each a stereo USB and AES input as well as each two optical and coaxial stereo inputs. Via USB, both PCM audio and DSD audio can be converted. PCM audio is received in S/PDIF format, both coaxial (RCA), optical (Toslink F06) and balanced (AES/EBU). The source selection is done via the selection control on the front. The selected input is also shown in the dot matrix LED display: USB, COA1, COA2, OPT1, OPT2 or AES. After approximately two seconds, the display shows the type of clock source and the detected sampling rate.Always in timeDiamond can not only synchronize to the clock integrated in the digitally fed source signal. As a specialist for digital players Diamond offers a word clock input. The source of the clock signal can be selected via the CLOCK switch.Source = Diamond synchronizes to the digital clock in the source signal.Word = Diamond synchronizes to the digital clock present at WORD IN.Well informedThe DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer, CD player or an external word clock is connected. The detected sampling rate and the selected clock source are shown in the dot matrix LED display. For example, U768 is an acronym for a clock signal in the USB stream with a sampling rate of 768 kHz.All variable – or Slave ThruDiamond provides two stereo output pairs for connecting power amplifiers or active loudspeakers. One pair of outputs with XLR, the other with RCA outputs. Both have the same output signal, which is controlled by the volume control on the front panel. The volume control for both outputs can be individually switched out of the signal path via DIP switch on the back of the device. For example, a headphone amplifier with its own volume control can be connected to one of the outputs.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Sounds goodWith all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. Many important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltageThe 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.The 120V technology achieves exceptional technical specifications and sonic benefits. Technically, in terms of dynamic range, signal-to-noise ratio and headroom. Sonically, in terms of richness of detail and an absolutely relaxed listening experience. By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.VOLTAiR is a composition of the terms Volt and Air. Volt is the unit for electrical voltage and Air stands for the unlimited space the music can breathe in. It symbolizes the perceived limitless dynamics resulting from high audio voltage.ComparisonMost audio devices work with an internal operating voltage of +/-15 volts and can thus process a maximum input level of +21.5 dBu. If a DAC, for example, has an output level of +22 dBu at 0 dBFS, level peaks of the music material would already cause overloads in the input stage of the device. All components in the audio device often operate at their limits. The result is an unsteady sound that causes stress and faster ear fatigue.SPL devices with VOLTAiR technology can handle input levels of +32.5 dBu thanks to the higher internal operating voltage of +/- 60 volts – thus offering 12 dB more headroom. All components consequently operate continuously in the optimum operating range. The result is a very pleasant, natural and relaxed sound experience. So you can enjoy your music in every detail.

Dual-Channel Microphone Preamplifier with 120V technologyCrescendo duo combines the precise sound that German Rundfunk-Technik is famous for with the innovative SPL 120V technology. The internal operating voltage of 120 volts allows Crescendo microphone preamplifiers to break completely new ground. This results in microphone preamps that sound detail-rich, vivid and equally honest.Crescendo duo lets microphones shine in a whole new light – because they have never been amplified like this before.The UndistortableThanks to the SPL 120V technology, it is almost impossible to overdrive this microphone preamplifier.Crescendo duo is a dual-channel microphone preamplifier with 120V technology. Both channels feature the same range of functions. Besides phantom power (48 V) for condenser microphones, polarity reversal, a high-pass filter and a PAD switch, they provide the possibility to adjust the VU meter to display very high levels – so they leave nothing to be desired. The gain can be adjusted comfortably and precisely via a 12-step switch. With the switchable output control, the output level of the Crescendo duo can be perfectly adapted to following AD converters. Two outputs per channel allow the signal to be sent to two different devices in parallel. In addition to the extensive feature set and the outstanding technical characteristics, Crescendo duo convinces with a sonic authenticity that is second to none.The 120V technologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps.Sounds goodWith all SPL devices we develop not only according to plan, but also by ear. Many important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.

Eight-Channel Microphone Preamplifier with 120V technologyCrescendo 8 combines the precise sound that German Rundfunk-Technik is famous for with the innovative SPL 120V technology. The internal operating voltage of 120 volts allows Crescendo microphone preamplifiers to break completely new ground. This results in microphone preamps that sound detail-rich, vivid and equally honest.Crescendo 8 lets microphones shine in a whole new light – because they have never been amplified like this before.The UndistortableThanks to the SPL 120V technology, it is almost impossible to overdrive this microphone preamplifier.Crescendo 8 is an eight-channel microphone preamplifier with 120V technology. All eight channels feature the same range of functions. Besides phantom power (48 V) for condenser microphones, polarity reversal and a PAD switch, they provide the possibility to adjust the VU meter to display very high levels. The gain can be adjusted comfortably and precisely via a high-quality potentiometer. Two outputs per channel allow the signal to be sent to two different devices in parallel. In addition to the extensive feature set and the outstanding technical characteristics, Crescendo 8 convinces with a sonic authenticity that is second to none.Built-in Output SplitterAll eight microphone preamps have two parallel outputs each. This allows the signal to be sent to two different devices in parallel. One output each is available as XLR line output. The respective second outputs are available as a DB25 jack. These can for example be ideally connected to a multi-channel AD converter, with the for this kind of device type typical DB25 input. An additional external signal splitter for back-up systems during a recording session is therefore no longer necessary.Perfection in detailThe input differential amplifiers are constructed with pairs of transistors, each of which is combined in a housing. Precise matching of the transistors and thermal coupling ensure high common-mode rejection and extremely low THD values. The choice of transistors and resistors has a decisive influence on the sound. In countless listening sessions the optimal components for Crescendo were selected – the effort paid off!More sound through innovationOur 120V SUPRA operational amplifiers work as current amplifiers, which also offer pure class A operation with over 6 mA quiescent current. We largely dispense with coupling capacitors in order to avoid their sonic disadvantages (diffuse, sloping, loss of dynamics). In order to eliminate DC voltage components active servo circuits are used.

The headphone amplifier, monitoring controller and DAC with 120V technology Phonitor 3 DAC is not only the ultimate headphone amplifier and monitoring controller with 120V technology – the integrated DAC makes it the perfect monitoring centerpiece for demanding producers as well as sound and mastering engineers working with digital sources or in the DAW. Whether if USB, AES/EBU or S/PDIF – the integrated DAC with analog SLP120 converts digital PCM audio signals with a resolution of 32 bit and a sampling rate of up to 768 kHz. It converts DSD signals with a resolution of up to DSD256. On the analog side, the Phonitor 3 DAC, based on the SPL 120V technology, offers the same monitoring quality as the big SPL mastering consoles – on loudspeakers and headphones. The analog Phonitor Matrix allows mixing and mastering on headphones in the highest quality – with the same spatial perception of the stereo stage as on loudspeakers.Great sound – for headphonesThe Phonitor 3 DAC offers connection for a standard headphone with a stereo jack plug on the front. Thanks to the enormous output power, all kind of headphones are driven effortlessly. Thus, it plays out the advantages of the SPL 120V technology and rewards the listener with an honest, detailed and at the same time vivid sound experience.Great sound – also for loudspeakersOn the rear of the device, Phonitor 3 DAC offers a stereo output which makes it a high-quality monitoring controller for any monitoring situation. Active speakers with an integrated power amplifier or passive speakers in combination with a power amplifier can be connected at this output.Monitoring at the highest levelPhonitor 3 DAC offers the same key technology – SPL 120V technology – and signal quality as the big SPL DMC and MMC mastering consoles.Headphones or SpeakersThe Output switch – simply switch from headphone to speaker playback. In the center position (Mute), no signal reaches the outputs. The VU meters light up red.Digital & analog stereo inputs – free choiceUp to six stereo sources can be connected to the Phonitor 3 DAC. The source is selected via the Source switch.Analog inputsOn the analog side, two analog stereo inputs are offered, each with two XLR line inputs. Professional analog players with balanced outputs and line level can be connected here.Digital inputsOn the digital side, the Phonitor 3 DAC offers four digital stereo inputs.USBThe USB input (type B) can be used to directly connect the DAW. No driver installation is required for operation with a Mac or an iDevice. Via USB, the Phonitor 3 DAC converts PCM audio signals with a resolution of 32 bits and a sampling rate of up to 768 kHz as well as DSD signals with a resolution of up to DSD256.S/PDIF, coaxialThe Input S/PDIF coaxial can also receive a two-channel signal (PCM audio) with sampling rates up to 192 kHz and a bit rate of 16 bits to 24 bits. Unbalanced, 2-wire 75-ohm coaxial cables with RCA connectors are used for signal transmission.S/PDIF, opticalThe S/PDIF optical input can also receive a two-channel signal (PCM audio) with sampling rates of up to 192 kHz and a bit rate of 16 bit to 24 bit. The input jack is an F05 jack. This interface is better known under the Toshiba brand name TOSLINK.Tip: You should attach importance to the quality of the optical fiber. With inexpensive plastic optical fibers, transmissions with quad sampling rates (176.4kHz/192kHz) may have faults. In this case a glass fiber optic cable should be chosen.AES/EBUThe AES input provides an input jack which corresponds to a standard XLR input (XLR, female). An AES signal can transport two encoded PCM audio channels with a maximum sampling rate of 192 kHz and a bit rate of 16 bits to 24 bits. This standard is defined under IEC 60958 Type I. Balanced, 3-core, 110 Ohm “twisted pair” cables with XLR plugs are used for connection.The DAC – the best is just good enoughThe converter chip in the Phonitor 3 DAC´s digital-to-analog converter is the highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip, which thanks to its architecture reproduces the finest sound details. It converts PCM audio at 32-bit resolution and a sampling rate of up to 768 kHz, which is 16 times the resolution of a CD. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256.SLP120The analog output of any DAC chip must be filtered by a low pass filter when entering the analog world. In contrast to all other DACs in the world, the low pass filters here are built using SPL 120V technology, which provides a plus in dynamics and headroom and sound. The DAC in the Phonitor 3 DAC is equipped with the SLP120 – an analog “single low pass” filter with 120V technology.Milled from solidThe massive 45mm volume control knob milled from aluminum is a haptic highlight. Its mass together with the Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further and provides perfect control of the monitoring level.Monitoring without any compromisesThe Phonitor 3 DAC is a full-fledged monitor controller:Source selectionPhonitor Matrix with Crossfeed, Speaker Angle and Center Level (the latter can be switched off)Mono/StereoLaterality controlSolo L/RPhase inversion for L and RL/R SwapOutput switchable to headphones or speakersMuteMono or stereo – a matter of settingThe Mono/Stereo switch can be used to select whether the output signal of the Phonitor 3 DAC is a regular stereo signal or whether it is summed to a mono signal.Finding the real centerNaturally, hearing can be directed more to the left or more to the right. This becomes particularly clear when listening on headphones. Therefore, the Phonitor 3 DAC has the uniquely finely resolved laterality control, which rebalances the stereo image on the monitoring side. The Mono/Stereo switch can not only be used to switch between stereo or mono playback. This switch is also used to activate or deactivate the laterality control.Solo L/ROnly the right or left channel of the audio signal should be played back? The Solo L/R function makes it possible. If the triple selection switch is in the “L” position, only the left channel is played back – if it is in the “R” position, only the right channel is played back. In the middle position “Off” the Solo function is switched off. The regular stereo signal consisting of the left and right channel is thus present at the output of the Phonitor 3 DAC.L/R SwapA special feature is the channel swap function: L/R Swap. This function inverts the stereo image. Left becomes right and right becomes left. This is especially important and extremely time-saving, when you are monitoring 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 judge whether the sample matches the image. With the L/R Swap function, this is no longer necessary. You can now adjust the direction of movement on the Phonitor 3 DAC while pre-listening the sample library.Phase inversion for L and RWith this switch the phase of the left or right channel of the audio signal can be inverted. If the three-way selection switch is in the “L” position, the left channel is inverted – if it is in the “R” position, the right channel is inverted. In the middle position “Off” this function is switched off. At the output of the Phonitor 3 DAC the regular stereo signal is thus present.M/S – Mid or Side?By using the mono/stereo and L/R phase inversion switch in combination, it is also possible to only monitor the mid or side signal. When the switch is set to “Mono” and phase inversion is active for L (or R), only the side signal is played back. If the phase inversion is switched off, the mono signal corresponding to the “M” signal is played back. Separate monitoring of the M and S signals has become a widespread standard for many mixing and mastering engineers.Everything at a glanceTwo mechanical VU meters visualize the input levels for the left and right audio channels. The “VU Cal” switch can be used to reduce the signal visualized on the VU meters by -6 dB or -12 dB. Thus, the VU meters operate in the ideal display range even at very high input levels.The Revolution: Phonitor MatrixThe Phonitor Matrix is the revolution in the headphone amplifier.One of the unique features of SPL headphone amplifiers is the SPL Phonitor Matrix. It enables mixing and mastering engineers to create perfect mixes on headphones, which will translate perfectly to all types of stereo speaker systems. But the Phonitor Matrix is not only designed for professional use in the studio. It also enables the hifi enthusiast to enjoy music on headphones, like if it was played back on speakers.How is this possible?Music is normally produced and mixed for playback on stereo speakers. Listening on headphones is different from listening on loudspeakers. The biggest difference is the lack of crossing signals of the sound signal from the left speaker to the right ear and from the right speaker to the left ear.These crossing signals are missing in conventional headphone listening, because there are no signals crossing from one side of the headphones to the other. This results in an unnaturally wide stereo stage and sound sources are not played back at their actual position in the stereo spectrum. The SPL Phonitor Matrix can correct this false stereo image with an analog circuitry.Summed up:The SPL Phonitor Matrix corrects the false representation of the stereo sound image, which makes it much easier to find the right decisions for mixing and mastering engineers. The hifi enthusiast can experience the music, like it was originally mixed and recorded. So nothing stands in the way of a successful and long mixing session on headphones.Preamp Out or Direct OutWith a DIP switch on the rear of the device, the volume control for the rear stereo output can be switched out of the signal path. But that’s not all. It is also possible to output the Phonitor Matrix to the stereo output on the rear panel of the device – regardless of whether this is configured as Preamp Out (with volume control) or Direct Out (without volume control).More powerIf headphones have a rather low sensivity – no problem. With DIP switch 1 on the rear of the device, the level at the headphone output can be increased by 12 dB.Sounds goodWith all SPL devices we develop not only according to plan, but also by ear. All important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The 120V technologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps.

Mastering Equalizer This is a unique equalizer – for several reasons. It has two times five fully parametric filter bands, each of which can be switched between constant Q and proportional Q! And there’s sheer unlimited headroom and a very special kind of EQ sound – thanks to 120V technology. Stepped Potentiometers Recall of all settings is easy thanks to the potentiometers with 41 steps. Cut/Boost & 1/4 Gain Each filter band can be boosted or cut up to 20 dB. If only minimal changes are required, the 1/4 Gain switch can be used to reduce the boost or cut range from +/-20 dB to +/-5 dB. Frequencies We have designed the frequency distribution to overlap widely between the bands, so that problematic frequencies can also be addressed with two bands. Bandwidth Q The bandwidth Q determines the steepness of bell characteristic. A small Q is a wide bandwidth and a high Q a narrow bandwidth. The labeling of the scale is divided into two parts. The values with white background refer to the proportional Q mode and the values without background refer to the constant Q mode. On/Off The orange button is used to switch the individual filter band on and off.  Constant & Proportional Q The blue button is used to toggle the Q mode. In constant Q mode, the amplitude is constant regardless of the selected bandwidth. This is ideal for eliminating interfering frequencies. In proportional Q mode, the amplitude is proportional to the bandwidth. It decreases with increasing bandwidth and vice versa. At the smallest bandwidth setting, the maximum amplitude value is +/- 20 dB, while the maximum amplitude decreases to +/- 2.8 dB at the largest bandwidth setting. This control behavior simplifies sensitive, creative processing and is musically very useful, since high amplitudes become increasingly unusable with increasing bandwidth. Auto Bypass For an objective evaluation of the edited music program, it makes sense not to switch between the original signal and the edited signal yourself, but to leave this to an automatic system. It is also an advantage that, for an optimal evaluation of the processing, one does not have to move from the stereo center and can concentrate fully on the program. With the help of the Interval control the switching time window can be determined. Turning it clockwise extends the time interval. Link The PQ is designed as a completely separate dual-mono, two-channel equalizer and can individually process two mono music programs simultaneously. It is of course also possible to edit a stereo program (left/right). If the Link function is activated, the functions LF, LMF, MF, MHF, HF On/Off and Con. Q/Prop.Q on both sides are switched together by the buttons on one side. This makes it possible to activate or deactivate a filter band or the Q characteristic on both sides of the equalizer with a single button operation. In Link mode, the right side controls the left side as factory preset. However, this can be adapted to individual working habits. If the channel switch of a channel is pressed until it flashes, this channel controls the other side from that moment on. Channel On/Off The orange button switches the entire channel on and off. Specifications: Analog inputs & output: XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 10 Hz – 100 kHz THD & N (+30 dBu, 1 kHz): 0.0005 % Noise (A-weighted): -94.3 dBu Dynamic range: 135 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 3.15 mA Fuse for 115 V: T 6.3 A Power consumption: max. 100 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 177 x 300 mm Unit weight: 15.2 kg Shipping weight (incl. packaging): 18.7 kg

Το SPL MTC Mk2 παίρνει όλα όσα έκαναν τον προκάτοχό του μεγάλη επιτυχία και το ανεβάζει σε υψηλότερο επίπεδο. Αυτός ο Monitor Controller διαθέτει ένα κουμπί στάθμης ήχου με πολύ σταθερή αίσθηση και μπορεί να συνδεθεί με 3 ζευγάρια ηχείων monitor και ένα Subwoofer. Περιλαμβάνει διακόπτες Mono και Dim, μεμονωμένους διακόπτες Mute για κάθε ηχείο monitor και ακουστικά. Tο MTC Mk2 περιλαμβάνει πολλές εισόδους - εξόδους I/O για 3 ζευγάρια ηχείων monitor, και προσφέρει τέσσερις στερεοφωνικές πηγές εισόδου. Η ειδική έξοδος μέτρησης στάθμης και το calibration της ρύθμισης στάθμης εξασφαλίζουν τον απόλυτο έλεγχο στην αναπαραγωγή του ήχου.Το εξαιρετικό σύστημα Talkback του MTC Mk2 περιλαμβάνει ενσωματωμένο μικρόφωνο, καθώς και αποκλειστικό έλεγχο στάθμης του Talkback, αυτόματη εξασθένιση μίξης και έλεγχο hands-free μέσω ενός ποδοδιακόπτη. Έχετε επίσης έναν κορυφαίο ενισχυτή ακουστικών με δύο υποδοχές εξόδου ακουστικών και ειδικά χειριστήρια στάθμης για κάθε έξοδο. Συμπληρωματικά, το MTC Mk2 περιλαμβάνει την περίφημη τεχνολογία Phonitor Matrix της SPL για απόδοση που μοιάζει με ηχείο κατά την ακρόαση με ακουστικά. Το MTC Mk2 περιλαμβάνει +/-18V εσωτερική τάση λειτουργίας, εξαρτήματα και υψηλής ποιότητας και τροφοδοτικό με μεταγωγή.

Τα επαγγελματικά projects απαιτούν μια ευέλικτη αλυσίδα monitoring. Ο SPL Marc One είναι εδώ για να σας την εξασφαλίσει. Αυτός ο εξαιρετικός sound&recording controller, διαθέτει δύο εξόδους ηχείων καθώς και έξοδο για subwoofer, δηλαδή ό,τι χρειάζεστε για να εξασφαλίσετε ότι οι μίξεις σας θα ακούγονται παντού καλά! Διαθέτει επίσης δύο στερεοφωνικές αναλογικές πηγές εισόδου συν μια ψηφιακή πηγή μέσω USB, οι οποίες δρομολογούνται απευθείας σε μια έξοδο σήματος line μόλις επιλεχθούν. Το Marc One περιλαμβάνει τρεις ρυθμίσεις monitor ( Stereo , Mono και Channel SWAP )για να ακούτε σχολαστικά τις μίξεις σας και έναν υψηλού επιπέδου ενισχυτή ακουστικών που περιλαμβάνει το αναγνωρισμένο Phonitor Matrix της SPL. Το Marc One της SPL , ανήκει στο επίκεντρο του χώρου σου. Τα project σου θα σε ευχαριστούν για αυτό !

Τα επαγγελματικά projects απαιτούν μια ευέλικτη αλυσίδα monitoring. Ο SPL Control One είναι εδώ για να σας την εξασφαλίσει. Αυτός ο εξαιρετικός sound&recording controller, διαθέτει δύο εξόδους ηχείων καθώς και έξοδο για subwoofer, δηλαδή ό,τι χρειάζεστε για να εξασφαλίσετε ότι οι μίξεις σας θα ακούγονται παντού καλά! Διαθέτει επίσης δύο στερεοφωνικές αναλογικές πηγές εισόδου οι οποίες δρομολογούνται απευθείας σε μια έξοδο σήματος line μόλις επιλεχθούν. Το Control One περιλαμβάνει τρεις ρυθμίσεις monitor ( Stereo , Mono και Channel SWAP ) για να ακούτε σχολαστικά τις μίξεις σας και έναν υψηλού επιπέδου ενισχυτή ακουστικών που περιλαμβάνει το αναγνωρισμένο Phonitor Matrix της SPL. Το Control One της SPL , ανήκει στο επίκεντρο του χώρου σου. Τα project σου θα σε ευχαριστούν για αυτό

Transient Designer Module The Transient Designer TDx brings level-independent dynamics processing to the world of the 500 Series. On/Bypass With On/Bypass you switch the module on or off (bypass). Signal LED The Sig LED (Sig) indicates whether an audio signal is present at the input and exceeds a level of -20 dB. This LED indicator is used as an aid to quickly identify whether a signal is arriving at the TDx in a complex studio cabling system. Attack Attack can be used to boost or cut the transient phase of a signal by up to 15 dB. Positive attack increases the amplitude of the transient response. Negative attack attenuates the amplitude of the transient response. Sustain With Sustain you can lengthen or shorten the decay phase of a signal by up to 24 dB. Positive Sustain extends the decay. Negative Sustain shortens the decay. Mix The mix control allows for continuous crossfade aka parallel mix between processed (WET) and unprocessed (DRY) signals. Output With the output control the output gain can be attenuated in order to optimally drive subsequent devices. The Differential Envelope Technology Our Differential Envelope Technology (DET) allows dynamic signal characteristics to be processed by forming the difference between envelopes and thereby making them level-independent. The envelope trackers align the working processes with the natural signal characteristics. This ensures optimum results for every moment.  Specifications:  Input & Output: balanced Maximum input & output gain: 21 dBu Input impedance: 20 kΩ Output impedance: 150 Ω Common mode rejection: > 82 dB Frequency range (-3 dB): 10 Hz – 100 kHz THD & N (0 dBu, 10 Hz – 22 kHz): 0.03 % Noise (A-weighted): -92.3 dB Dynamic range: 113 dB Dimensions & Weight Unit weight: 0.65 kg Shipping weight (incl. packaging): 0.9 kg

Dual-Band De-Esser Module Our well known De-Esser as a dual-band version built into a single module for the 500 Series. On/Bypass With On/Bypass you switch the module on or off (bypass). Signal LED The Sig LED (Sig) indicates whether an audio signal is present at the input and exceeds a level of -20 dB. This LED indicator serves as an aid to quickly recognize whether a signal is arriving at the DeS in a complex studio cabling system. Hi-S The Hi-S control adjusts the intensity of the S sound reduction in the upper frequency band. The center frequency for sibilance detection is 11.2 kHz with a bandwidth of 3 kHz. Lo-S The Lo-S control adjusts the intensity of the S sound reduction in the lower frequency band. Male/Female Voicing The Voice switch is used to adjust the low-band de-esser to the voice characteristics. The switch changes the center frequency for the sibilance identification: Female: 7.6 kHz, Male: 6.4 kHz. The processing bandwidth in the low band is 1.44 kHz. De-Esser Tech Talk De-Essing by Phase Cancellation SPL has developed a circuit technology that combines efficient de-essing with maximum ease of use. De-essing already during recording spares a lot of time in comparison to the subsequent search for the S-sounds in post production. The DeS automatically adjusts itself to the relevant frequencies and hones in on them so that only the range of the S-sound is processed and neighbouring frequencies remain untouched. This range is phase-inverted and mixed back into the original signal, which acoustically deletes the S-sound. The DeS therefore works unobstrusively and sound-neutral. Auto-Threshold The DeS features our Auto-Threshold which automatically adjusts the threshold if the input level fluctuates due to varying distance to the microphone which leads to fluctuating input levels. With Auto-Threshold engaged, the de-essing intensity remains constant at the value you set – a great help not only for untrained speakers/singers in the studio and live. With conventional de-essers using compressor technology, the processing intensity decreases as the distance to the microphone increases to which compressors or limiters then react by letting the S sounds reappear. The disadvantage of weaker de-essing – bad enough in itself – therefore also leads to undesired effects in the subsequent processing. Auto-Threshold prevents these disadvantages from occurring in the first place. Specifications: Input & Output: balanced Maximum input & output gain: 21 dBu Input impedance: 20 kΩ Output impedance: 150 Ω Common mode rejection: -82 dB Frequency range (-3 dB): 10 Hz – 100 kHz THD & N (0 dBu, 10 Hz – 22 kHz): 0,03 % Noise (A-weighted): -92.3 dB Dynamic range: 113 dB Dimensions & Weight Unit weight: 0.65 kg Shipping weight (incl. packaging): 0.9 kg

Με καινοτόμο Phonitor Matrix, μετατροπέα DA 32-bit και εξαιρετική ποιότητα ήχου. Για εμάς ο ήχος είναι πάντα σε πρώτο πλάνο και για να αποκτήσετε έναν τέτοιο ήχο, χρειάζεστε εμπειρία και τεχνική φινέτσα. Έχουμε ενσωματώσει και τα δύο στη σειρά Phonitor One. Το στάδιο του προενισχυτή Δύο ενισχυτές Burr-Brown OPA 2134 SoundPlus™ παρέχουν την προενίσχυση. Αυτά είναι ειδικά σχεδιασμένα για κυκλώματα audiophile και χαρακτηρίζονται από εξαιρετικά χαμηλό θόρυβο και ελάχιστη παραμόρφωση μόνο 0,00008%. Χάρη σε ένα πραγματικό στάδιο FET, το OPA 2134 επιτυγχάνει ρυθμό περιστροφής 20 V/μs, καθιστώντας το έναν λειτουργικό ενισχυτή υψηλής ταχύτητας. Αυτό έχει άμεσα θετική επίδραση στη ζωντάνια της αναπαραγωγής. Επιπλέον, πυκνωτές φιλμ χαμηλής χωρητικότητας εγκαθίστανται στη διαδρομή σήματος ως πυκνωτές παράκαμψης των πυκνωτών ζεύξης, γεγονός που εξασφαλίζει ακριβέστερη μετάδοση των στοιχείων του σήματος υψηλής συχνότητας. Οι εξαιρετικοί ηλεκτρολυτικοί πυκνωτές Panasonic «Electric Double Layer» χρησιμοποιούνται ως πυκνωτές σύζευξης. Προσφέρουν την υψηλότερη χωρητικότητα αποθήκευσης από όλες τις τεχνολογίες πυκνωτών.

Για εμάς ο ήχος είναι πάντα προτεραιότητα. Ομολογουμένως, όταν ο ήχος είναι καλός, μας αρέσει να ακούμε. Για να φτάσεις σε αυτό το αποτέλεσμα, χρειάζεσαι εμπειρία και τεχνική δεξιότητα και έχουμε ενσωματώσει και τα δυο στο Series One . Tην προενίσχυση αναλαμβάνουν δύο Burr-Brown OPA 2134 SoundPlus™ .Είναι ειδικά σχεδιασμένοι για audiophile αποδόσεις και χαρακτηρίζονται από εξαιρετικά μικρό θόρυβο και ελάχιστη παραμόρφωση με δείκτη μόνο 0.00008%. Λόγω του πραγματικού FET , το OPA 2134 επιτυγχάνει έναν ρυθμό περιστροφής 20V/µs, καθιστώντας τον έναν γρήγορο ενισχυτή υψηλής ποιότητας.

Our nonplusultra headphone amplifier.The Phonitor xe is the nonplusultra standalone headphone amplifier without any compromises. Phonitor Matrix, remote volume control, retro-look VU meter, the optional premium DA converter and the all-superior VOLTAiR technology make the Phonitor xe one of the best headphone amplifiers of our time.Great Sound – for all headphonesThe Phonitor xe offers connections for a standard headphone with a stereo jack plug and for balanced-driven headphones using a 4 pin XLR plug.Thanks to the enormous output power, all headphones are driven effortlessly.Thus, it plays out the advantages of the SPL VOLTAiR technology and rewards the listener with an honest, detailed and at the same time vivid sound experience.Twice as niceThe Phonitor xe provides headphone outputs on the front and on the rear. On the front, you switch between F (front) and R (rear). This way, your favorite headphones remain firmly connected on the rear – without visible cabling on the front.Milled from solidThe massive 45mm volume control knob milled from aluminum is a haptic highlight. Its mass together with the motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further.The red marker LED ensures good visibility of the volume parameter even in darkened environments.Remote controlThe volume control can be remotely controlled with any infrared remote control.The Phonitor xe learns to communicate with it with the simple push of a button.Learning made easyUsing Phonitor x as an example, this video explains how you can use any infrared remote control to remotely control the volume.Source of joyUp to six (6 !) stereo sources can be connected to the Phonitor xe:XLR, RCA and via the optional DAC768 also USB, COAX, OPTICAL and AES.Phonitor xe accepts almost all audio connections. Balanced via XLR or unbalanced via RCA connectors.With the optional DAC768, four additional digital sources are added. From USB to professional AES format, no wishes remain unfulfilled.The DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer or CD player is connected. No further settings on the Phonitor xe are necessary.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the optional digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Good ol' timesTwo mechanical VU meters visualize the input levels for the left and right audio channel.With the VU switch you can optimize the display for different signal levels.Finding the middleis not always easy. By our very nature, hearing can be leaning more to the left or more to the right. This is especially obvious when listening with headphones.That’s why Phonitor xe has the uniquely finely resolved laterality control. With it, the center can be found perfectly.With the MODE switch you can choose between stereo or mono playback. This switch also activates the laterality control.The Revolution: Phonitor MatrixThe Phonitor Matrix is the revolution in the headphone amplifier.Thanks to the Phonitor Matrix, music can be experienced on headphones as if it was played on speakers. The tiring super stereo effect is a thing of the past.Enjoy your music without listening fatigue and exactly the way it should be heard.Music is normally produced and mixed for playback on stereo speakers.Listening on headphones is different from listening on loudspeakers. The biggest difference is the lack of crossing signals of the sound signal from the left speaker to the right ear and from the right speaker to the left ear. These crossing signals are missing in conventional headphone listening, because there are no signals crossing from one side of the headphones to the other. This results in an unnaturally wide stereo image and the various sound sources of the audio signal are not localized as the sound engineer intended them to. This effect is often referred to as “super stereo effect”. During conventional listening on headphones, our brain can balance the false representation of the playback to a certain extent – but this is very exhausting.The SPL Phonitor Matrix can correct this false stereo image with an analog circuitry. The Phonitor Matrix therefore not only ensures a correct representation of the stereo image, but also a relaxed listening experience.The two main parameters of the Phonitor Matrix are Crossfeed and Angle:Crossfeed determines the crossing signals of the channels, the so-called interaural level difference.Angle determines the opening angle of the stereo image, the so-called interaural time difference.Center of AttentionTo make the listening experience even more perfect, the level of the center of the stereo image needs to be attenuated when the Phonitor Matrix is active. This ensures that not only the position of all sound sources is correct but also their volume.In the Professional Fidelity Phonitor devices this value is set to a fixed attenuation of -1 dB, which is the best choice for getting an authentic representation of the sound stage.Sounds good!With all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. All important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltage.The 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.

Our nonplusultra headphone amplifier.The Phonitor xe is the nonplusultra standalone headphone amplifier without any compromises. Phonitor Matrix, remote volume control, retro-look VU meter, the optional premium DA converter and the all-superior VOLTAiR technology make the Phonitor xe one of the best headphone amplifiers of our time.Great Sound – for all headphonesThe Phonitor xe offers connections for a standard headphone with a stereo jack plug and for balanced-driven headphones using a 4 pin XLR plug.Thanks to the enormous output power, all headphones are driven effortlessly.Thus, it plays out the advantages of the SPL VOLTAiR technology and rewards the listener with an honest, detailed and at the same time vivid sound experience.Twice as niceThe Phonitor xe provides headphone outputs on the front and on the rear. On the front, you switch between F (front) and R (rear). This way, your favorite headphones remain firmly connected on the rear – without visible cabling on the front.Milled from solidThe massive 45mm volume control knob milled from aluminum is a haptic highlight. Its mass together with the motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further.The red marker LED ensures good visibility of the volume parameter even in darkened environments.Remote controlThe volume control can be remotely controlled with any infrared remote control.The Phonitor xe learns to communicate with it with the simple push of a button.Learning made easyUsing Phonitor x as an example, this video explains how you can use any infrared remote control to remotely control the volume.Source of joyUp to six (6 !) stereo sources can be connected to the Phonitor xe:XLR, RCA and via the optional DAC768 also USB, COAX, OPTICAL and AES.Phonitor xe accepts almost all audio connections. Balanced via XLR or unbalanced via RCA connectors.With the optional DAC768, four additional digital sources are added. From USB to professional AES format, no wishes remain unfulfilled.The DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer or CD player is connected. No further settings on the Phonitor xe are necessary.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the optional digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Good ol' timesTwo mechanical VU meters visualize the input levels for the left and right audio channel.With the VU switch you can optimize the display for different signal levels.Finding the middleis not always easy. By our very nature, hearing can be leaning more to the left or more to the right. This is especially obvious when listening with headphones.That’s why Phonitor xe has the uniquely finely resolved laterality control. With it, the center can be found perfectly.With the MODE switch you can choose between stereo or mono playback. This switch also activates the laterality control.The Revolution: Phonitor MatrixThe Phonitor Matrix is the revolution in the headphone amplifier.Thanks to the Phonitor Matrix, music can be experienced on headphones as if it was played on speakers. The tiring super stereo effect is a thing of the past.Enjoy your music without listening fatigue and exactly the way it should be heard.Music is normally produced and mixed for playback on stereo speakers.Listening on headphones is different from listening on loudspeakers. The biggest difference is the lack of crossing signals of the sound signal from the left speaker to the right ear and from the right speaker to the left ear. These crossing signals are missing in conventional headphone listening, because there are no signals crossing from one side of the headphones to the other. This results in an unnaturally wide stereo image and the various sound sources of the audio signal are not localized as the sound engineer intended them to. This effect is often referred to as “super stereo effect”. During conventional listening on headphones, our brain can balance the false representation of the playback to a certain extent – but this is very exhausting.The SPL Phonitor Matrix can correct this false stereo image with an analog circuitry. The Phonitor Matrix therefore not only ensures a correct representation of the stereo image, but also a relaxed listening experience.The two main parameters of the Phonitor Matrix are Crossfeed and Angle:Crossfeed determines the crossing signals of the channels, the so-called interaural level difference.Angle determines the opening angle of the stereo image, the so-called interaural time difference.Center of AttentionTo make the listening experience even more perfect, the level of the center of the stereo image needs to be attenuated when the Phonitor Matrix is active. This ensures that not only the position of all sound sources is correct but also their volume.In the Professional Fidelity Phonitor devices this value is set to a fixed attenuation of -1 dB, which is the best choice for getting an authentic representation of the sound stage.Sounds good!With all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. All important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltage.The 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.

Το Phonitor x, είναι ο απόλυτος ενισχυτής ακουστικών. Προσφέρει συνδέσεις για ακουστικά τόσο balanced όσο και unbalanced, ενώ με ισχύ εξόδου 3.7 W, η απόδοσή του είναι εντυπωσιακή.  Το επαναστατικό σύστημα Phonitor Matrix, προσφέρει την πιό ρεαλιστική ακρόαση για ακουστικά, αφού δημιουργεί την εντύπωση ακρόασης μέσω ηχείων. Το Phonitor x, δεν είναι μόνο ενισχυτής ακουστικών, αλλά και ένας εξαιρετικός προενισχυτής, όπου μπορείτε να συνδέσετε έναν τελικό ενισχυτή ή ενεργά ηχεία μόνιτορ Προδιαγραφές: Ενισχυτής ακουστικών και προενισχυτής, τεχνολογίας VOLTAiR 120 V Rail,  Ξεχωριστές συνδέσεις για ακουστικά και τελικό ενισχυτή balanced και unbalanced,  Κατάλληλο για όλα τα ακουστικά με αντίσταση μεγαλύτερη από 10 ohms,  Είσοδοι balanced και unbalanced (XLR και RCA),  Προαιρετικά ψηφιακές εισόδους: USB, coaxial and optical,  Προαιρετικά μετατροπέας DAC: 192 kHz / 24 Bit,  Έξοδοι balanced και unbalanced (XLR και RCA),  Phonitor Matrix για προσομοίωση ακρόασης ηχείων,  Έλεγχος Laterality: super-fine balance,  Διακόπτης Mono,  Έλεγχος στάθμης (ελέγχεται και με τηλεχειριστήριο),  Μέγιστη ισχύς εξόδου: 2 x 3.7 W (120 ohms),  Απόκριση συχνότητας: 10 Hz - 300 kHz (-3 dB),  THD+N: 0.00091 % (HP), 0.00085 % (Line),  Dynamic range: 135.5 dB (HP), 136.3 dB (Line),  Σύνδεση AMP CTR με τον τελικό ενισχυτή Performer (για έλεγχο του διακόπτη on/off),  Made in Germany Χαρακτηριστικά: Απόκριση Συχνότητας: ‹10Hz - ›200kHz (-3dB) Αριθμός Εισόδων: 2 Αριθμός Εξόδων: 2 & 1(Headphones) Δυναμική Περιοχή: 129,5 dBΑ Ενισχυτής Ακουστικών: ΝΑΙ Συνδετήρες Εισόδου: 2 x XLR Συνδετήρες Εξόδου: 2 x XLR 1 x TRS Τύπος Ενίσχυσης: Ενεργή  

Our nonplusultra headphone amplifier.The Phonitor xe is the nonplusultra standalone headphone amplifier without any compromises. Phonitor Matrix, remote volume control, retro-look VU meter, the optional premium DA converter and the all-superior VOLTAiR technology make the Phonitor xe one of the best headphone amplifiers of our time.Great Sound – for all headphonesThe Phonitor xe offers connections for a standard headphone with a stereo jack plug and for balanced-driven headphones using a 4 pin XLR plug.Thanks to the enormous output power, all headphones are driven effortlessly.Thus, it plays out the advantages of the SPL VOLTAiR technology and rewards the listener with an honest, detailed and at the same time vivid sound experience.Twice as niceThe Phonitor xe provides headphone outputs on the front and on the rear. On the front, you switch between F (front) and R (rear). This way, your favorite headphones remain firmly connected on the rear – without visible cabling on the front.Milled from solidThe massive 45mm volume control knob milled from aluminum is a haptic highlight. Its mass together with the motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further.The red marker LED ensures good visibility of the volume parameter even in darkened environments.Remote controlThe volume control can be remotely controlled with any infrared remote control.The Phonitor xe learns to communicate with it with the simple push of a button.Learning made easyUsing Phonitor x as an example, this video explains how you can use any infrared remote control to remotely control the volume.Source of joyUp to six (6 !) stereo sources can be connected to the Phonitor xe:XLR, RCA and via the optional DAC768 also USB, COAX, OPTICAL and AES.Phonitor xe accepts almost all audio connections. Balanced via XLR or unbalanced via RCA connectors.With the optional DAC768, four additional digital sources are added. From USB to professional AES format, no wishes remain unfulfilled.The DAC automatically picks up the sampling rate and resolution of the digital playback source. No matter if a streamer, computer or CD player is connected. No further settings on the Phonitor xe are necessary.The DAC768The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the optional digital-to-analogue converter, which thanks to its architecture reproduces the finest sound details.It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768 kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. In contrast to the DAC 768xs, the DAC768 not only offers an AES/EBU digital input, but also the SPL DLP120 with VOLTAiR technology.The DLP120 (Dual Low-Pass)The analog output of the DAC chip must be filtered by a low pass filter. Phonitor xe has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required.In contrast to all other DACs in the world, the low pass filters here are built using VOLTAiR technology, which provides a plus in dynamics and headroom and sound.Good ol' timesTwo mechanical VU meters visualize the input levels for the left and right audio channel.With the VU switch you can optimize the display for different signal levels.Finding the middleis not always easy. By our very nature, hearing can be leaning more to the left or more to the right. This is especially obvious when listening with headphones.That’s why Phonitor xe has the uniquely finely resolved laterality control. With it, the center can be found perfectly.With the MODE switch you can choose between stereo or mono playback. This switch also activates the laterality control.The Revolution: Phonitor MatrixThe Phonitor Matrix is the revolution in the headphone amplifier.Thanks to the Phonitor Matrix, music can be experienced on headphones as if it was played on speakers. The tiring super stereo effect is a thing of the past.Enjoy your music without listening fatigue and exactly the way it should be heard.Music is normally produced and mixed for playback on stereo speakers.Listening on headphones is different from listening on loudspeakers. The biggest difference is the lack of crossing signals of the sound signal from the left speaker to the right ear and from the right speaker to the left ear. These crossing signals are missing in conventional headphone listening, because there are no signals crossing from one side of the headphones to the other. This results in an unnaturally wide stereo image and the various sound sources of the audio signal are not localized as the sound engineer intended them to. This effect is often referred to as “super stereo effect”. During conventional listening on headphones, our brain can balance the false representation of the playback to a certain extent – but this is very exhausting.The SPL Phonitor Matrix can correct this false stereo image with an analog circuitry. The Phonitor Matrix therefore not only ensures a correct representation of the stereo image, but also a relaxed listening experience.The two main parameters of the Phonitor Matrix are Crossfeed and Angle:Crossfeed determines the crossing signals of the channels, the so-called interaural level difference.Angle determines the opening angle of the stereo image, the so-called interaural time difference.Center of AttentionTo make the listening experience even more perfect, the level of the center of the stereo image needs to be attenuated when the Phonitor Matrix is active. This ensures that not only the position of all sound sources is correct but also their volume.In the Professional Fidelity Phonitor devices this value is set to a fixed attenuation of -1 dB, which is the best choice for getting an authentic representation of the sound stage.Sounds good!With all devices of the Professional Fidelity series we develop not only according to plan, but also by ear. All important components are installed on the circuit boards using Through-hole technology. This way we can ensure that we can use the best sounding components.The VOLTAiR TechnologyThe 120V technology is our reference technology. The 120V technology is unique in the world. It operates at a DC voltage of 120 volts. This is four times that of IC-based semiconductor op-amps. In our Professional Fidelity series, we refer to this unsurpassed technology as VOLTAiR technology.The highest possible audio quality requires the highest possible audio operating voltage.The 120V technology works with +/-60 V. To be able to handle such a high voltage, we have developed special proprietary operational amplifiers that can operate with a DC voltage of +/-60 V: the SPL 120V SUPRA operational amplifiers. This high voltage would destroy conventional components and operational amplifiers.By the way, the “120V” in the name of the technology has nothing to do with the local mains voltage from the mains power socket. This is about the operating voltage inside the device with which the audio signals are processed.The mains voltage from the mains power socket is transformed to the required secondary voltage in the device’s internal linear power supply with toroidal transformer. Rectifiers convert this AC voltage into DC voltage required in the audio device.

Mastering Equalizer This is a unique equalizer – for several reasons. It has two times five fully parametric filter bands, each of which can be switched between constant Q and proportional Q! And there’s sheer unlimited headroom and a very special kind of EQ sound – thanks to 120V technology. Stepped Potentiometers Recall of all settings is easy thanks to the potentiometers with 41 steps. Cut/Boost & 1/4 Gain Each filter band can be boosted or cut up to 20 dB. If only minimal changes are required, the 1/4 Gain switch can be used to reduce the boost or cut range from +/-20 dB to +/-5 dB. Frequencies We have designed the frequency distribution to overlap widely between the bands, so that problematic frequencies can also be addressed with two bands. Bandwidth Q The bandwidth Q determines the steepness of bell characteristic. A small Q is a wide bandwidth and a high Q a narrow bandwidth. The labeling of the scale is divided into two parts. The values with white background refer to the proportional Q mode and the values without background refer to the constant Q mode. On/Off The orange button is used to switch the individual filter band on and off.  Constant & Proportional Q The blue button is used to toggle the Q mode. In constant Q mode, the amplitude is constant regardless of the selected bandwidth. This is ideal for eliminating interfering frequencies. In proportional Q mode, the amplitude is proportional to the bandwidth. It decreases with increasing bandwidth and vice versa. At the smallest bandwidth setting, the maximum amplitude value is +/- 20 dB, while the maximum amplitude decreases to +/- 2.8 dB at the largest bandwidth setting. This control behavior simplifies sensitive, creative processing and is musically very useful, since high amplitudes become increasingly unusable with increasing bandwidth. Auto Bypass For an objective evaluation of the edited music program, it makes sense not to switch between the original signal and the edited signal yourself, but to leave this to an automatic system. It is also an advantage that, for an optimal evaluation of the processing, one does not have to move from the stereo center and can concentrate fully on the program. With the help of the Interval control the switching time window can be determined. Turning it clockwise extends the time interval. Link The PQ is designed as a completely separate dual-mono, two-channel equalizer and can individually process two mono music programs simultaneously. It is of course also possible to edit a stereo program (left/right). If the Link function is activated, the functions LF, LMF, MF, MHF, HF On/Off and Con. Q/Prop.Q on both sides are switched together by the buttons on one side. This makes it possible to activate or deactivate a filter band or the Q characteristic on both sides of the equalizer with a single button operation. In Link mode, the right side controls the left side as factory preset. However, this can be adapted to individual working habits. If the channel switch of a channel is pressed until it flashes, this channel controls the other side from that moment on. Channel On/Off The orange button switches the entire channel on and off. Specifications: Analog inputs & output: XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 10 Hz – 100 kHz THD & N (+30 dBu, 1 kHz): 0.0005 % Noise (A-weighted): -94.3 dBu Dynamic range: 135 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 3.15 mA Fuse for 115 V: T 6.3 A Power consumption: max. 100 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 177 x 300 mm Unit weight: 15.2 kg Shipping weight (incl. packaging): 18.7 kg

Passive Mastering Equalizer In the PASSEQ we have revived the concept of the passive equalizer that was used in the 50’s and 60’s. The charming sound of this vintage technology combined with the modern 120V technology results in a very special equalizer. Only one filter​ All control functions are part of only one passive filter. There are three frequency ranges: Low Freq / Mid Freq / High Freq Each with a cut control on the left and a boost control on the right. In the middle is the output control. Design A passive filter has no amplification stages. It can therefore only be lowered. If one frequency is to be boosted, all the others must be lowered. Behind the filter there is a stage that makes up for the lowered overall level. Since low noise is particularly important here, since up to 20 dB can be made up for, the 120V technology comes into its own. The entire passive filter (variable resistor, capacitor and coil) provides a very nice sound characteristic. Besides the choice of components, the charging behaviour of the capacitors and the saturation behaviour of the coils play a major role in this. The resulting relative inertia compared to potentially very fast active filters is the reason for a pleasant, musically advantageous sound characteristic. We would choose smoothness and transparency as well as strikingly silky highs and pithy basses as a description that suits our perception. During fine-tuning, through component selection, in countless listening sessions, the emphasis was on achieving the most musical sounding curves possible, which, for example, need fear no comparison with a Pultec EQ from the 1950/60s. Only without all the disadvantages of a 60 year old original, such as the very high background noise, but above all the very limited frequency selection. Another highlight of the PASSEQ is the HF+ band, which has been extended by the frequencies 25 kHz and 35 kHz and sounds so incredibly good that you don’t want to switch it off. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -60 dBu Frequency range (-3 dB): 10 Hz – 200 kHz Crosstalk (1 kHz): -108 dBu THD + N (+30 dBu, 1 kHz): 0.0012 % Noise (A-bewertet): -95.2 dBu Dynamic range: 134.5 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 500 mA Fuse for 115 V: T 1 A Power consumption: max. 15 VA Dimensions & Weight W x H x T (W x H x D (width x height x depth) x Höhe x Tiefe): 482 x 132 x 300 mm Unit weight: 10.2 kg Shipping weight (incl. packaging): 14 kg

Mastering D/A Converter Mercury is the first mastering DA-converter in 120V technology. It offers seven digital stereo input sources, all of which can be synchronized with each other or with a word clock. The additional variable output qualifies Mercury as a monitor controller. Seven Digital Sources Mercury provides connections for a total of seven digital stereo input sources.  For each digital source there is a dedicated and illuminated tactile switch for quick selection and quick comparison. The detected sample rate is shown in the display. One USB input and two coaxial, two optical and two AES/EBU inputs are available. AES input 2 also supports Dual-Wire (DW) mode. Sync The display shows the synchronization status and the detected sampling rate. Any source except USB can be synchronized to another source or to an external Word Clock. If the sync “Source” is selected in the display, each source uses its own sync code embedded in the signal. O dBfs Holding the Sync button for more than two seconds Mercury will change to the 0dBfs display. Mercury offers all reference levels in dBu corresponding to a 0 dBfs (full scale) digital signal: +14, +15, +16, +17, +18, +20, +22 and +24 dBu  Fix & Var The FIX OUT is an analog balanced stereo output which can be calibrated to all common reference levels (see 0 dBfs). Mercury offers a variable analog balanced stereo output. This output provides the same signal as the FIX OUT, but the level is continuously adjustable. Thus Mercury can also be used as a state-of-the-art monitor controller. Variable Output This analog potentiometer is used to control the level of the VAR OUT output. We use the ALPS RK27 “Big Blue” potentiometer with a nice “spoon in honey” feel and excellent channel tracking. So not only the auditory but also the haptic experience of volume control is at the highest level. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -108 dBu THD (0 dBu, 1 kHz): 0.000992 % Noise (A-weighted): -102.5 dBu Dynamic range: 135 dB Digital Inputs, DAC768 AES/EBU (XLR), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Coaxial SPDIF (Cinch), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optisch SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

16-Channel Mastering Monitor Controller The MC16 is designed to monitor surround and immersive audio projects (Dolby Atmos®, Auro 3D®, etc…) in the analog domain, offering the same unrivalled quality as 120V stereo monitoring. 16 Speakers Sixteen buttons are used to activate the corresponding loudspeakers. The 16 inputs of Input 1 and Input 2 are routed directly to the outputs to ensures that an output routing of a DAW session always matches the routing in the MC16. The buttons allow you to insert a label with the exact name of the loudspeaker. 2 x 16 MC16 offers two 16-channel inputs to quickly compare a current project with a reference production. Monitor Level This analog potentiometer is used to adjust the monitor level. We developed a 16-level potentiometer, with which we could realize the volume control in the analog domain, because we did not want to accept a digital control, which is afflicted with resolution deficits and limited dynamic range. Presets You can save speaker setups to three memory locations. So you can quickly call up a 5.1 or an 11.1 or a 9.1.6 configuration. In solo mode, three more memory locations are available with the same keys. For example, you can create solo groups for all front speakers and all ceiling speakers and all rear speakers. Specifications: Analog inputs & output: XLR & DB25 (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -100 dBu THD + N (+24 dBu): > 112 dB Noise (A-weighted): -102.4 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 70 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 88 x 300 mm Unit weight: 8.4 kg Shipping weight (incl. packaging): 12 kg

Mastering Compressor As a mastering compressor, IRON is inspired by the legendary vintage tube compressors à la Fairchild, Collins or Gates and has been developed in a way to give modern productions more energy and power with the advantages of the 120V technology. Input & Output Input and output levels are switchable in a control range of +/- 12dB. An increase of the input level leads to more intensive compression. When mastering full scale material, the reduction can be useful. The output is used to adjust the level ratios of input and output. Attack & Release The time parameters Attack and Release are switchable in six steps from Slow to Fast. The times vary depending on the rectifier circuit selected. Side Chain EQs With the side chain EQs, compression can be focused on frequency ranges. In addition to an inserted EQ, there are four internal EQ presets with a partly complex frequency response. Rectifier The tube stage is controlled by a rectifier circuit. There are six diode control characteristics generated by germanium, silicon and LED diodes or combinations thereof. Variable-Bias Limiter/Compressor The IRON Mastering Compressor is a variable-bias (variable µ) limiter/compressor from the basic concept. IRON works on the principle of the bias controlled remote cutoff tube. Parallel to this, a medium-variable-bias triode (sharp-cutoff tube) with a steeper characteristic is connected. The tube bias voltage (BIAS) can be adjusted in three steps. The tubes are integrated into special Lundahl manufactured balanced high-level double core mu-metal-iron transformers. AirBass & Tape Roll-Off The compressor is followed by a passive filter network with two equalizer presets: AirBass and Tape Roll-off. The AirBass filter boosts low and high frequencies by about 1.5 dB and the Tape Roll-Off filter emulates the typical frequency response of a tape machine. Link All functions can be set via switches and the threshold via a rasterized potentiometer which makes recall easy. The settings of Threshold, Bias, Attack, Release, Rectifier and Side-Chain-EQs are transferred from the right side (channel 2) to the left side in Link Mode, in which the compressor operates with summed control signals. Auto-Bypass Channels 1 and 2 are activated with the large illuminated IDEC switches. In Auto Bypass mode, the channels are automatically switched on and off, which leads to a more objective assessment. The Interval control is used to adjust the time window continuously between two and 20 seconds. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Frequency range (-3 dB): 10 Hz – 40 kHz Common mode rejection (0dBu, 1kHz): > 80 dB Crosstalk (1 kHz): -108 dBu THD (+10 dBu): 0.002 % Noise (A-weighted): -98 dBu Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 132 x 300 mm Unit weight: 11 kg Shipping weight (incl. packaging): 14 kg

Mastering D/A Converter Mercury is the first mastering DA-converter in 120V technology. It offers seven digital stereo input sources, all of which can be synchronized with each other or with a word clock. The additional variable output qualifies Mercury as a monitor controller. Seven Digital Sources Mercury provides connections for a total of seven digital stereo input sources.  For each digital source there is a dedicated and illuminated tactile switch for quick selection and quick comparison. The detected sample rate is shown in the display. One USB input and two coaxial, two optical and two AES/EBU inputs are available. AES input 2 also supports Dual-Wire (DW) mode. Sync The display shows the synchronization status and the detected sampling rate. Any source except USB can be synchronized to another source or to an external Word Clock. If the sync “Source” is selected in the display, each source uses its own sync code embedded in the signal. O dBfs Holding the Sync button for more than two seconds Mercury will change to the 0dBfs display. Mercury offers all reference levels in dBu corresponding to a 0 dBfs (full scale) digital signal: +14, +15, +16, +17, +18, +20, +22 and +24 dBu  Fix & Var The FIX OUT is an analog balanced stereo output which can be calibrated to all common reference levels (see 0 dBfs). Mercury offers a variable analog balanced stereo output. This output provides the same signal as the FIX OUT, but the level is continuously adjustable. Thus Mercury can also be used as a state-of-the-art monitor controller. Variable Output This analog potentiometer is used to control the level of the VAR OUT output. We use the ALPS RK27 “Big Blue” potentiometer with a nice “spoon in honey” feel and excellent channel tracking. So not only the auditory but also the haptic experience of volume control is at the highest level. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -108 dBu THD (0 dBu, 1 kHz): 0.000992 % Noise (A-weighted): -102.5 dBu Dynamic range: 135 dB Digital Inputs, DAC768 AES/EBU (XLR), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Coaxial SPDIF (Cinch), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optisch SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

16-Channel Mastering Monitor Controller The MC16 is designed to monitor surround and immersive audio projects (Dolby Atmos®, Auro 3D®, etc…) in the analog domain, offering the same unrivalled quality as 120V stereo monitoring. 16 Speakers Sixteen buttons are used to activate the corresponding loudspeakers. The 16 inputs of Input 1 and Input 2 are routed directly to the outputs to ensures that an output routing of a DAW session always matches the routing in the MC16. The buttons allow you to insert a label with the exact name of the loudspeaker. 2 x 16 MC16 offers two 16-channel inputs to quickly compare a current project with a reference production. Monitor Level This analog potentiometer is used to adjust the monitor level. We developed a 16-level potentiometer, with which we could realize the volume control in the analog domain, because we did not want to accept a digital control, which is afflicted with resolution deficits and limited dynamic range. Presets You can save speaker setups to three memory locations. So you can quickly call up a 5.1 or an 11.1 or a 9.1.6 configuration. In solo mode, three more memory locations are available with the same keys. For example, you can create solo groups for all front speakers and all ceiling speakers and all rear speakers. Specifications: Analog inputs & output: XLR & DB25 (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -100 dBu THD + N (+24 dBu): > 112 dB Noise (A-weighted): -102.4 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 70 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 88 x 300 mm Unit weight: 8.4 kg Shipping weight (incl. packaging): 12 kg

Mastering Compressor As a mastering compressor, IRON is inspired by the legendary vintage tube compressors à la Fairchild, Collins or Gates and has been developed in a way to give modern productions more energy and power with the advantages of the 120V technology. Input & Output Input and output levels are switchable in a control range of +/- 12dB. An increase of the input level leads to more intensive compression. When mastering full scale material, the reduction can be useful. The output is used to adjust the level ratios of input and output. Attack & Release The time parameters Attack and Release are switchable in six steps from Slow to Fast. The times vary depending on the rectifier circuit selected. Side Chain EQs With the side chain EQs, compression can be focused on frequency ranges. In addition to an inserted EQ, there are four internal EQ presets with a partly complex frequency response. Rectifier The tube stage is controlled by a rectifier circuit. There are six diode control characteristics generated by germanium, silicon and LED diodes or combinations thereof. Variable-Bias Limiter/Compressor The IRON Mastering Compressor is a variable-bias (variable µ) limiter/compressor from the basic concept. IRON works on the principle of the bias controlled remote cutoff tube. Parallel to this, a medium-variable-bias triode (sharp-cutoff tube) with a steeper characteristic is connected. The tube bias voltage (BIAS) can be adjusted in three steps. The tubes are integrated into special Lundahl manufactured balanced high-level double core mu-metal-iron transformers. AirBass & Tape Roll-Off The compressor is followed by a passive filter network with two equalizer presets: AirBass and Tape Roll-off. The AirBass filter boosts low and high frequencies by about 1.5 dB and the Tape Roll-Off filter emulates the typical frequency response of a tape machine. Link All functions can be set via switches and the threshold via a rasterized potentiometer which makes recall easy. The settings of Threshold, Bias, Attack, Release, Rectifier and Side-Chain-EQs are transferred from the right side (channel 2) to the left side in Link Mode, in which the compressor operates with summed control signals. Auto-Bypass Channels 1 and 2 are activated with the large illuminated IDEC switches. In Auto Bypass mode, the channels are automatically switched on and off, which leads to a more objective assessment. The Interval control is used to adjust the time window continuously between two and 20 seconds. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Frequency range (-3 dB): 10 Hz – 40 kHz Common mode rejection (0dBu, 1kHz): > 80 dB Crosstalk (1 kHz): -108 dBu THD (+10 dBu): 0.002 % Noise (A-weighted): -98 dBu Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 132 x 300 mm Unit weight: 11 kg Shipping weight (incl. packaging): 14 kg

Mastering Router with dual Parallel Mix Hermes revolutionizes mastering. Now it is possible to have an audio signal processed by up to eight 2-channel processors in any order that can be stored. At the push of a button you can compare up to four complex processing chains. Connect, route, store, compare Hermes accelerates mastering workflows in an unprecedented achievable way. The time-consuming replugging of the patchbay alone had made true comparison almost impossible. Now you can change, store and compare processing chains in seconds. All with real buttons, relays and without additional software. Simply push the processor buttons A to H in the order in which the audio signal should pass through the processors and store it.  All processors can be named for a better overview. The name always appears briefly in the display when an insert button is pressed. Free routing of two parallel mix stages Hermes has two parallel mix stages that can be freely assigned to each of the eight processors and stored with a processing chain. This means that even two compressors with different parallel mix settings can be compared quickly and easily. The routing is completely passive with gas-encapsulated and gold-plated high-end relays. All active stages, such as the input/output stages and the parallel mix stages work in 120V technology. Sends & Returns The stereo channels are strictly separated on two boards to achieve the highest possible channel separation. The inserts are connected with XLRs for the first four stereo processors and with DB25 for the second four stereo processors. Specifications: Analog inputs & output; XLR & DB25 (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Frequency range (-3 dB): 4 Hz to 300 kHz Common mode rejection: -75 dBu Crosstalk (1 kHz, 0dBu): -130 dBu Crosstalk (1 kHz, 0dBu, P-Mix on): -110 dBu Noise (A-weighted): -121 dBu Noise (A-weighted, P-Mix on): -104 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 88 x 300 mm Unit weight: 9 kg Shipping weight (incl. packaging): 12 kg

Mastering M/S Processor Gemini is the first mid/side processor in 120V technology. In mastering, M/S processing is often the best way to get targeted access to individual elements of a mix. M/S ? The decomposition of the stereo signal in mid (voice, snare, bass …) and side (guitars, pads, cymbals …) allows their separate processing. The center signal (Mid) is the sum of the left and right channel of the stereo signal (L+R), i.e. the “in phase” part of the stereo signal. The side signal (Side) is the difference of the left and right channels of the stereo signal (L-R), i.e. the “out-of-phase” part of the stereo signal. Gemini can also be used to enlarge or reduce the stereo image. An elliptical filter for shifting the low frequency component from the side to the mid is also on board. The Mid When the M/S button is activated, the input signal (L/R stereo) is split into a mid and side signals. The exact center position can be adjusted with the Balance control and the exact level with the Trim control. Both functions are switchable to be in the signal path only when in use. Exclusive monitoring of the center is possible when the solo function is activated. Processors can be inserted to process the mid signal and activated with the Insert button.  The Side The side signal also has a solo function and a switchable insert for external processing.  In addition, the elliptical filter can be used to shift the bass frequencies from the side to the mid. This improves the punch and precision of low frequencies. The Stereo Width control changes the level of the side signal, with a level increase broadening the stereo image. These two functions can also be switched on to be in the signal path only when used.  I/Os and Inserts All inputs and outputs as well as sends and returns are balanced using Neutrik XLR connectors. Processors can be connected to the inserts for external processing. Especially interesting is the integration of the Gemini into the Hermes Mastering Router. In this way, processing chains can be created that combine classic L/R processing with M/S processing. Gemini & Hermes The M/S encoder and decoder stages of the Gemini can each be connected to an insert of the Hermes and freely positioned within the processing chain. Processors that are in a chain between M/S encoder and decoder process the mid/side signal. These processors can also easily be mono devices. In the example above, Insert Send A is routed to Gemini for M/S encoding. The mid/side sends of the Gemini go back to Return A of the Hermes. Send B goes from Hermes back to the Gemini to be encoded back to L/R, which flows from the output of the Gemini back to Return B of Hermes.  For example, it is possible (see figure above) to place the M/S encoder at processing position 3, an equalizer at processing position 4 for separate processing of the mid and side signals, and then the M/S decoder stage of the Gemini at processing position 5 to generate an L/R stereo signal again. If a compressor is now to be used as an additional device for M/S processing, it can simply be set to processing position 5 in the chain, and the M/S decoder stage would move to processing position 6 accordingly. Specifications: Analog inputs & output: XLR (balanced) Maximum input & output gain: 32,5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -79 dBu Frequency range (-3 dB): 10 Hz – 150 kHz Crosstalk (1 kHz): -108 dBu THD + N (0 dBu, 1 kHz): -110 dBu Noise (A-weighted), all stages active): -98,2 dBu Dynamic range: 135 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 500 mA Fuse for 115 V: T 1 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 44 x 300 mm Unit weight: 5 kg Shipping weight (incl. packaging): 8,1 kg

Mastering Router with dual Parallel Mix Hermes revolutionizes mastering. Now it is possible to have an audio signal processed by up to eight 2-channel processors in any order that can be stored. At the push of a button you can compare up to four complex processing chains. Connect, route, store, compare Hermes accelerates mastering workflows in an unprecedented achievable way. The time-consuming replugging of the patchbay alone had made true comparison almost impossible. Now you can change, store and compare processing chains in seconds. All with real buttons, relays and without additional software. Simply push the processor buttons A to H in the order in which the audio signal should pass through the processors and store it.  All processors can be named for a better overview. The name always appears briefly in the display when an insert button is pressed. Free routing of two parallel mix stages Hermes has two parallel mix stages that can be freely assigned to each of the eight processors and stored with a processing chain. This means that even two compressors with different parallel mix settings can be compared quickly and easily. The routing is completely passive with gas-encapsulated and gold-plated high-end relays. All active stages, such as the input/output stages and the parallel mix stages work in 120V technology. Sends & Returns The stereo channels are strictly separated on two boards to achieve the highest possible channel separation. The inserts are connected with XLRs for the first four stereo processors and with DB25 for the second four stereo processors. Specifications: Analog inputs & output; XLR & DB25 (balanced) Maximum input & output gain: 32.5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Frequency range (-3 dB): 4 Hz to 300 kHz Common mode rejection: -75 dBu Crosstalk (1 kHz, 0dBu): -130 dBu Crosstalk (1 kHz, 0dBu, P-Mix on): -110 dBu Noise (A-weighted): -121 dBu Noise (A-weighted, P-Mix on): -104 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 88 x 300 mm Unit weight: 9 kg Shipping weight (incl. packaging): 12 kg

Mastering Console The DMC is the heart of the mastering studio. The converters, the players, the DAW, the recorders and the speakers are connected to them. The audio quality of the console is therefore crucial. And like all devices of the mastering series, the DMC also works on unparalleled 120V operating voltage. Inputs & Sources Audio signals that shall be processed are connected to the Inputs and audio signals for comparison or playback are connected to the Sources. Four stereo inputs pairs are available for each. Each input channel can be switched on/off, phase reversed and trimmed in 0.25 dB steps. A stereo input can be switched to mono. The large potentiometer for the recording gain allows sensitive gain rides. If it is not used, it should not be activated. This keeps the entire control stage outside the signal path. Monitoring & Inserts Three pairs of speakers can be operated for full-range, mid-field and near-field monitors. Speakers D is for a mono loudspeaker. Preferably a Phonitor headphone amplifier is connected to the Phonitor HP Out, whereby it can be programmed whether the loudspeaker is automatically switched off during headphone operation or not. The same applies to the subwoofer. It can be programmed with which monitor it is paired and activated together. The insert return can also be trimmed in 0.25 dB steps to compensate for the finest level differences after an insert chain. The Hermes Mastering Router is preferably connected to the inserts, with which up to eight stereo processors can be connected and freely routed and stored. In the monitoring section, each loudspeaker channel can be solo’ed and phase reversed. The large monitoring level potentiometer has an illuminated indicator so that the monitor level can be easily assessed even from a greater distance. A loudness compensation (+/- 10 dB) can be activated. Alternatively it can be linked to the Insert, Sources or Inputs switches allowing loudness compensation at any stage. DMC & MC 16 Yeah, it looks kind of wild. Makes sense, though. The DMC can be linked to the MC16 for surround and immersive audio applications. Dolby Atmos® or Auro 3D® projects can also be realized. The MC16 is the world’s first analog 16 channel monitoring controller. Mastering studios often use the L/R speakers and subwoofer for stereo and multi-channel monitoring. The combination of DMC with MC16 makes it unnecessary to replug these speakers. Pressing the Speaker A button for two seconds transfers the volume control of L/R and Sub to the MC16 and pressing it again brings it back to the DMC. In this way, stereo and surround projects in a studio can easily be performed one after the other. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dB Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz to 300 kHz Noise (A-weighted, +24 dBu, Rec Out, Insert On): -101.6 dBu Noise (A-weighted, +24 dBu, Speaker Out): -103.9 dBu THD & N (+24 dBu, Rec Out, Insert On): > 112 dB THD & N (+24 dBu, Speaker Out): > 108 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 132 x 300 mm Unit weight: 10.6 kg Shipping weight (incl. packaging): 14 kg

Mastering M/S Processor Gemini is the first mid/side processor in 120V technology. In mastering, M/S processing is often the best way to get targeted access to individual elements of a mix. M/S ? The decomposition of the stereo signal in mid (voice, snare, bass …) and side (guitars, pads, cymbals …) allows their separate processing. The center signal (Mid) is the sum of the left and right channel of the stereo signal (L+R), i.e. the “in phase” part of the stereo signal. The side signal (Side) is the difference of the left and right channels of the stereo signal (L-R), i.e. the “out-of-phase” part of the stereo signal. Gemini can also be used to enlarge or reduce the stereo image. An elliptical filter for shifting the low frequency component from the side to the mid is also on board. The Mid When the M/S button is activated, the input signal (L/R stereo) is split into a mid and side signals. The exact center position can be adjusted with the Balance control and the exact level with the Trim control. Both functions are switchable to be in the signal path only when in use. Exclusive monitoring of the center is possible when the solo function is activated. Processors can be inserted to process the mid signal and activated with the Insert button.  The Side The side signal also has a solo function and a switchable insert for external processing.  In addition, the elliptical filter can be used to shift the bass frequencies from the side to the mid. This improves the punch and precision of low frequencies. The Stereo Width control changes the level of the side signal, with a level increase broadening the stereo image. These two functions can also be switched on to be in the signal path only when used.  I/Os and Inserts All inputs and outputs as well as sends and returns are balanced using Neutrik XLR connectors. Processors can be connected to the inserts for external processing. Especially interesting is the integration of the Gemini into the Hermes Mastering Router. In this way, processing chains can be created that combine classic L/R processing with M/S processing. Gemini & Hermes The M/S encoder and decoder stages of the Gemini can each be connected to an insert of the Hermes and freely positioned within the processing chain. Processors that are in a chain between M/S encoder and decoder process the mid/side signal. These processors can also easily be mono devices. In the example above, Insert Send A is routed to Gemini for M/S encoding. The mid/side sends of the Gemini go back to Return A of the Hermes. Send B goes from Hermes back to the Gemini to be encoded back to L/R, which flows from the output of the Gemini back to Return B of Hermes.  For example, it is possible (see figure above) to place the M/S encoder at processing position 3, an equalizer at processing position 4 for separate processing of the mid and side signals, and then the M/S decoder stage of the Gemini at processing position 5 to generate an L/R stereo signal again. If a compressor is now to be used as an additional device for M/S processing, it can simply be set to processing position 5 in the chain, and the M/S decoder stage would move to processing position 6 accordingly. Specifications: Analog inputs & output: XLR (balanced) Maximum input & output gain: 32,5 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -79 dBu Frequency range (-3 dB): 10 Hz – 150 kHz Crosstalk (1 kHz): -108 dBu THD + N (0 dBu, 1 kHz): -110 dBu Noise (A-weighted), all stages active): -98,2 dBu Dynamic range: 135 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 500 mA Fuse for 115 V: T 1 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 44 x 300 mm Unit weight: 5 kg Shipping weight (incl. packaging): 8,1 kg

Mastering Console The DMC is the heart of the mastering studio. The converters, the players, the DAW, the recorders and the speakers are connected to them. The audio quality of the console is therefore crucial. And like all devices of the mastering series, the DMC also works on unparalleled 120V operating voltage. Inputs & Sources Audio signals that shall be processed are connected to the Inputs and audio signals for comparison or playback are connected to the Sources. Four stereo inputs pairs are available for each. Each input channel can be switched on/off, phase reversed and trimmed in 0.25 dB steps. A stereo input can be switched to mono. The large potentiometer for the recording gain allows sensitive gain rides. If it is not used, it should not be activated. This keeps the entire control stage outside the signal path. Monitoring & Inserts Three pairs of speakers can be operated for full-range, mid-field and near-field monitors. Speakers D is for a mono loudspeaker. Preferably a Phonitor headphone amplifier is connected to the Phonitor HP Out, whereby it can be programmed whether the loudspeaker is automatically switched off during headphone operation or not. The same applies to the subwoofer. It can be programmed with which monitor it is paired and activated together. The insert return can also be trimmed in 0.25 dB steps to compensate for the finest level differences after an insert chain. The Hermes Mastering Router is preferably connected to the inserts, with which up to eight stereo processors can be connected and freely routed and stored. In the monitoring section, each loudspeaker channel can be solo’ed and phase reversed. The large monitoring level potentiometer has an illuminated indicator so that the monitor level can be easily assessed even from a greater distance. A loudness compensation (+/- 10 dB) can be activated. Alternatively it can be linked to the Insert, Sources or Inputs switches allowing loudness compensation at any stage. DMC & MC 16 Yeah, it looks kind of wild. Makes sense, though. The DMC can be linked to the MC16 for surround and immersive audio applications. Dolby Atmos® or Auro 3D® projects can also be realized. The MC16 is the world’s first analog 16 channel monitoring controller. Mastering studios often use the L/R speakers and subwoofer for stereo and multi-channel monitoring. The combination of DMC with MC16 makes it unnecessary to replug these speakers. Pressing the Speaker A button for two seconds transfers the volume control of L/R and Sub to the MC16 and pressing it again brings it back to the DMC. In this way, stereo and surround projects in a studio can easily be performed one after the other. Specifications: Analog inputs & output; XLR (balanced) Maximum input & output gain: 32.5 dB Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection: -82 dBu Frequency range (-3 dB): 4 Hz to 300 kHz Noise (A-weighted, +24 dBu, Rec Out, Insert On): -101.6 dBu Noise (A-weighted, +24 dBu, Speaker Out): -103.9 dBu THD & N (+24 dBu, Rec Out, Insert On): > 112 dB THD & N (+24 dBu, Speaker Out): > 108 dB Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 60 V Operating voltage for relays and LEDs: + 12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 1 A Fuse for 115 V: T 2 A Power consumption: max. 40 VA Dimensions & Weight W x H x D (width x height x depth): 482 x 132 x 300 mm Unit weight: 10.6 kg Shipping weight (incl. packaging): 14 kg

Surround Monitor Controller The SMC 7.1 expands analog surround monitoring to 8 channels and offers Solo, Solo-In-Place and Solo-To-Center as well as a headphone jack. Inputs & Speaker Management Two 7.1 and two stereo input sources can be connected to the SMC 7.1. This is important because in monitoring, two productions are always compared with each other. A subwoofer can be added to a stereo input by activating the LFE button. Each loudspeaker can be switched on and off individually, and the SMC 7.1 remembers the last speakers used for each source. The Solo-In-Place function now turns the loudspeaker on/off switches into solo buttons. This allows each loudspeaker to be heard alone at its position. Solo-To-Center places the selected loudspeaker on the center speaker, so that it can be listened to exactly from the front in case critical listening is required. An alternative output is available for connecting a pair of stereo loudspeakers.  The 45mm all-aluminium rotary knob moves an 8-level potentiometer.  Mute is especially important when the DAW is sending full-range audio garbage due to a problem. Mute can be used quickly to protect speakers and hearing. As an alternative to the speakers, headphones can be used for monitoring. The headphones can be switched on and have their own volume control. The SMC 7.1 can be installed in the Expansion Rack to be screwed into a 19-inch rack. The Expansion Rack is equipped with a switch that can passively route the stereo output signal of the SMC 7.1 to four stereo outputs. In this combination, the SMC 7.1 is expanded by three additional stereo outputs. This allows a 7.1 loudspeaker array and full-range, mid and nearfield stereo speakers to be controlled and monitored. Specifications: Analog inputs & outputs; XLR & DB25 (balanced) Maximum input: 22 dBu Input impedance: 20 kΩ Output impedance: 100 Ω Frequency range (-3 dB): 10 Hz – 150 kHz THD +N (10 Hz – 22 kHz, 0 dBu): 0.001 % Noise (A-weighted): -93 dBu Crosstalk (1 kHz): -92 dB Common mode rejection (1 kHz): 82 dB Dynamic range: 115 dB Standard Headphones Output, 6.35 mm (1/4") TRS Jack Wiring: Tip = Left, Ring = Right, Sleeve = GND Frequency range (-3dB): 10 Hz – 80 kHz Crosstalk (1 kHz): -70 dBu Total harmonic distortion (0 dBu, 1 kHz, 40 kΩ load): 0.001 % Noise (A-weighted): -93 dBu Dynamic range: 115 dB Internal Linear Power Supply with Toroidal Transformer Operating voltage for analog audio: +/- 18 V Operating voltage for relays and LEDs: +12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Sicherung für 230 V: T 500 mA Sicherung für 115 V: T 1 A Power consumption: max. 30 W Dimensions & Weight W x H x D (width x height x depth): 278 x 100 x 330 mm Unit weight: 4.1 kg Shipping weight (incl. packaging): 5 kg

Surround Monitor Controller The SMC 7.1 expands analog surround monitoring to 8 channels and offers Solo, Solo-In-Place and Solo-To-Center as well as a headphone jack. Inputs & Speaker Management Two 7.1 and two stereo input sources can be connected to the SMC 7.1. This is important because in monitoring, two productions are always compared with each other. A subwoofer can be added to a stereo input by activating the LFE button. Each loudspeaker can be switched on and off individually, and the SMC 7.1 remembers the last speakers used for each source. The Solo-In-Place function now turns the loudspeaker on/off switches into solo buttons. This allows each loudspeaker to be heard alone at its position. Solo-To-Center places the selected loudspeaker on the center speaker, so that it can be listened to exactly from the front in case critical listening is required. An alternative output is available for connecting a pair of stereo loudspeakers.  The 45mm all-aluminium rotary knob moves an 8-level potentiometer.  Mute is especially important when the DAW is sending full-range audio garbage due to a problem. Mute can be used quickly to protect speakers and hearing. As an alternative to the speakers, headphones can be used for monitoring. The headphones can be switched on and have their own volume control. The SMC 7.1 can be installed in the Expansion Rack to be screwed into a 19-inch rack. The Expansion Rack is equipped with a switch that can passively route the stereo output signal of the SMC 7.1 to four stereo outputs. In this combination, the SMC 7.1 is expanded by three additional stereo outputs. This allows a 7.1 loudspeaker array and full-range, mid and nearfield stereo speakers to be controlled and monitored. Specifications: Analog inputs & outputs; XLR & DB25 (balanced) Maximum input: 22 dBu Input impedance: 20 kΩ Output impedance: 100 Ω Frequency range (-3 dB): 10 Hz – 150 kHz THD +N (10 Hz – 22 kHz, 0 dBu): 0.001 % Noise (A-weighted): -93 dBu Crosstalk (1 kHz): -92 dB Common mode rejection (1 kHz): 82 dB Dynamic range: 115 dB Standard Headphones Output, 6.35 mm (1/4") TRS Jack Wiring: Tip = Left, Ring = Right, Sleeve = GND Frequency range (-3dB): 10 Hz – 80 kHz Crosstalk (1 kHz): -70 dBu Total harmonic distortion (0 dBu, 1 kHz, 40 kΩ load): 0.001 % Noise (A-weighted): -93 dBu Dynamic range: 115 dB Internal Linear Power Supply with Toroidal Transformer Operating voltage for analog audio: +/- 18 V Operating voltage for relays and LEDs: +12 V Mains Power Supply Mains voltage (selectable, see fuse chamber): 230 V AC / 50; 115 V AC / 60 Hz Sicherung für 230 V: T 500 mA Sicherung für 115 V: T 1 A Power consumption: max. 30 W Dimensions & Weight W x H x D (width x height x depth): 278 x 100 x 330 mm Unit weight: 4.1 kg Shipping weight (incl. packaging): 5 kg

Υψηλής πιστότητας ενισχυτής ακρόασης για ακουστικά.Διαθέτει: Τεχνολογία 120 volts Κουμπί Mono Διακόπτη εξασθένισης (Dim) Κουμπί Solo (αριστερά - δεξιά) Αντιστροφή φάσης Ενδεικτικά στάθμης VU Σύνδεση εισόδου / εξόδου balanced XLR Πλήρης προσομοίωση ακρόασης μέσω ηχείων Ρύθμιση γωνίας των ηχείων Ρύθμιση στάθμης ‘‘κέντρου’’ του ήχου Έξοδος ακουστικών (εμπρός μέρος) Αριθμός Εισόδων: 2 Συνδετήρες Εξόδου: 2 x XLR 1 x TRS Τύπος Ενίσχυσης: Ενεργή

Επιτραπέζιος ελεγκτής ακρόασης Stereo με σύστημα ενδοεπικοινωνίας (talkback). Διαθέτει: Αναλογικό έλεγχο στάθμης stereo Κουμπί Mono Διακόπτη εξασθένισης (Dim) Σύστημα ενδοεπικοινωνίας με ενσωματωμένο μικρόφωνο (talkback) 3 στερεοφωνικές εξόδους ηχείων Έξοδος Cue-Mix Επιλογή πηγής ήχου 6X Stereo (4 balanced και 2 unbalanced) Ενισχυτής ακουστικών Ιδανικό σύστημα ακρόασης σε οποιοδήποτε επαγγελματικό ή home στούντιο. Αριθμός Εισόδων: 6 Ιδιαίτερα Χαρακτηριστικά: Ιδανικό σύστημα ακρόασης σε οποιοδήποτε επαγγελματικό ή home στούντιο Συνδετήρες Εξόδου: 3 x XLR Τύπος Ενίσχυσης: Ενεργά

16+16 channel AD/DA converter and MADI interface Madison offers what hardly any other manufacturer in this price segment has to offer: 16 AD/DA converters. With SmartMADI and the Madicon this becomes an expandable multi-channel studio interface of the first order. MADI transmission distance up to 2 km MADI I/O latency: 1 sample Optical MADI socket, type SC 56 & 64 channel Hi/Lo modes accepts Varispeed up to +/-12,5% Sync to MADI, Wordclock or Internal Device ID for daisy chaining 16 AD and 16 DA converters 24 bit / up to 192 kHz Level lights per channel in easily readable blocks of four OdBfs: 15, 18, 22 or 24 dBu Fanless and therefore particularly suitable for noise-sensitive environments Optional redundant power supply unit Madison together with the Madicon support SmartMADI MADI becomes more user-friendly, simple and modern SPL’s Madicon and Madisons can now be connected in series (daisy chaining) like Audio-Over-IP devices, and are easier to configure and more reliable. An additional MADI router for more than two MADI devices is not required The sample rate is now transmitted via the MADI stream and Madisons follow automatically – very handy Specifications: Analoge Input & Outputs; DB25 (balanced) Maximum input & output gain: 24 dBu Input impedance: 20 kΩ Output impedance: 75 Ω Common mode rejection (1 kHz): 82 dB Crosstalk (1 kHz, 48 kHz SR): -110 dB Dynamic Range (A-weighted, 48 kHz SR): 115 dB Digital I/O; MADI-Port MADI optical, Type SC: 44.1/48/88.2/96/176.4/192 kHz Varispeed: ±12.5% in 56 CH mode, -12.5% in 64 CH mode Reference Levels: 15dBu, 18dBu, 22dBu, 24 dBu Wordclock input, BNC, terminated impedance: 75 Ω Wordclock Level (Input TTL/CMOS5/CMOS3): 3.3 V S/N (A-bewertet, 48kHz SR): -115 dBFS S/(N+D) (-1 dBfs, 48kHz SR): -102 dBFS Passband response (-0.02dB, 48kHz SR): 22 kHz Passband Ripple (Decimation-LPF, 48kHz SR): +/- 0.005 dB Stopband Attenuation (Decimation-LPF, 48kHz SR): 100 dB Crosstalk (1 kHz, 48kHz SR): -110 dBu Internal Linear Power Supply with Shielded Toroidal Transformer Operating voltage for analog audio: +/- 17 V Operating voltage for digital audio: +12 V, +5 V BTU (British Thermal Units, typical): 65 BTU/h BTU (British Thermal Units, max): 110 BTU/h Mains Power Supply Mains voltage: 90 – 264 V AC, 50/60 Hz Hz Power consumption: max. 35 W Dimensions & Weight W x H x D (width x height x depth): 482 x 44 x 260 mm Unit weight: 3.05 kg Shipping weight (incl. packaging): 5 kg