SPL

The director sets the tone. The central unit of the HiFi system is the preamplifier. With remote source selection and volume control, retro-look-VU-meter, premium DA-converter and the all superior VOLTAiR technology, the Director Mk2 caters for musical blockbusters. Milled from solid is the big 45mm Volume control. Its mass together with the  motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further. The red marker LED makes the setting visible even in darkened environments. Volume RC The volume can be remote controlled with any infrared remote control, that the Director Mk2 simply learns. Learning made easy This video explains how you can use any infrared remote control to remotely control the volume and the source selection. Source of joy Up to ten (10 !) stereo sources can be connected to the Director Mk2. Six of them are for analog and four for digital input signals. The source selection is shown in the dot matrix LED display: IN 1 through IN 6 for the analog inputs. AES, USB, COAX, OPTI for the digital inputs, where the detected sample rate is displayed after about two seconds. For example U768 = USB input with 768kHz sampling rate. The source selection can also be operated remotely if the rotary switch is set to the “Remote” position. The Director Mk2 offers two pairs of balanced inputs with XLR connectors and four standard inputs with gold-plated RCA connectors. On the digital side there are four additional inputs. PCM audio is received in S/P-DIF format via coaxial Cinch, optical Toslink and balanced XLR (AES). Both PCM audio and DSD audio can be converted via USB. The DAC768 The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analog converter, which thanks to its new architecture reproduces the finest sound details. It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. The DLP120 The analog output of the DAC chip must be filtered by a low pass filter. Director Mk2 has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required. And 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. Good ol' times Two 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. Tape Monitor A tape monitor circuitry is quite retro. Who still owns a cassette recorder or a tape machine? Who still has an equalizer? To put it another way: Who would like to do it and cannot do it because no manufacturer offers a tape monitor circuit anymore? Well, those who want can again. Today’s audio levels are much higher than in the past and many recorders will be overloaded by them. In that case simply press the switch and the level to the recorder is lowered by 10 dB. To compensate, the level from the recorder is increased by 10 dB. Et voilà. Preamp & Direct Out On the output side, power amplifiers or active loudspeakers can be connected to the preamp output sockets using XLRs. The Direct Out via RCA picks up the output signal before the volume control and thus offers the possibility to connect a headphone amplifier. AMP CTL Practical are the AMP CTL sockets to which power amplifiers with 12 V trigger inputs can be connected in order to remotely switch them on and off together with the Director Mk2. Specifications: Analog Inputs & Outputs: XLR (balanced), RCA Maximum input & output gain: 32.5 dBu Input impedance (RCA): 47 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 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 (RCA), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optical SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

The director sets the tone. The central unit of the HiFi system is the preamplifier. With remote source selection and volume control, retro-look-VU-meter, premium DA-converter and the all superior VOLTAiR technology, the Director Mk2 caters for musical blockbusters. Milled from solid is the big 45mm Volume control. Its mass together with the  motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further. The red marker LED makes the setting visible even in darkened environments. Volume RC The volume can be remote controlled with any infrared remote control, that the Director Mk2 simply learns. Learning made easy This video explains how you can use any infrared remote control to remotely control the volume and the source selection. Source of joy Up to ten (10 !) stereo sources can be connected to the Director Mk2. Six of them are for analog and four for digital input signals. The source selection is shown in the dot matrix LED display: IN 1 through IN 6 for the analog inputs. AES, USB, COAX, OPTI for the digital inputs, where the detected sample rate is displayed after about two seconds. For example U768 = USB input with 768kHz sampling rate. The source selection can also be operated remotely if the rotary switch is set to the “Remote” position. The Director Mk2 offers two pairs of balanced inputs with XLR connectors and four standard inputs with gold-plated RCA connectors. On the digital side there are four additional inputs. PCM audio is received in S/P-DIF format via coaxial Cinch, optical Toslink and balanced XLR (AES). Both PCM audio and DSD audio can be converted via USB. The DAC768 The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analog converter, which thanks to its new architecture reproduces the finest sound details. It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. The DLP120 The analog output of the DAC chip must be filtered by a low pass filter. Director Mk2 has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required. And 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. Good ol' times Two 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. Tape Monitor A tape monitor circuitry is quite retro. Who still owns a cassette recorder or a tape machine? Who still has an equalizer? To put it another way: Who would like to do it and cannot do it because no manufacturer offers a tape monitor circuit anymore? Well, those who want can again. Today’s audio levels are much higher than in the past and many recorders will be overloaded by them. In that case simply press the switch and the level to the recorder is lowered by 10 dB. To compensate, the level from the recorder is increased by 10 dB. Et voilà. Preamp & Direct Out On the output side, power amplifiers or active loudspeakers can be connected to the preamp output sockets using XLRs. The Direct Out via RCA picks up the output signal before the volume control and thus offers the possibility to connect a headphone amplifier. AMP CTL Practical are the AMP CTL sockets to which power amplifiers with 12 V trigger inputs can be connected in order to remotely switch them on and off together with the Director Mk2. Specifications: Analog Inputs & Outputs: XLR (balanced), RCA Maximum input & output gain: 32.5 dBu Input impedance (RCA): 47 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 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 (RCA), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optical SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

The director sets the tone. The central unit of the HiFi system is the preamplifier. With remote source selection and volume control, retro-look-VU-meter, premium DA-converter and the all superior VOLTAiR technology, the Director Mk2 caters for musical blockbusters. Milled from solid is the big 45mm Volume control. Its mass together with the  motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further. The red marker LED makes the setting visible even in darkened environments. Volume RC The volume can be remote controlled with any infrared remote control, that the Director Mk2 simply learns. Learning made easy This video explains how you can use any infrared remote control to remotely control the volume and the source selection. Source of joy Up to ten (10 !) stereo sources can be connected to the Director Mk2. Six of them are for analog and four for digital input signals. The source selection is shown in the dot matrix LED display: IN 1 through IN 6 for the analog inputs. AES, USB, COAX, OPTI for the digital inputs, where the detected sample rate is displayed after about two seconds. For example U768 = USB input with 768kHz sampling rate. The source selection can also be operated remotely if the rotary switch is set to the “Remote” position. The Director Mk2 offers two pairs of balanced inputs with XLR connectors and four standard inputs with gold-plated RCA connectors. On the digital side there are four additional inputs. PCM audio is received in S/P-DIF format via coaxial Cinch, optical Toslink and balanced XLR (AES). Both PCM audio and DSD audio can be converted via USB. The DAC768 The highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip is used as the converter chip in the digital-to-analog converter, which thanks to its new architecture reproduces the finest sound details. It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. The DLP120 The analog output of the DAC chip must be filtered by a low pass filter. Director Mk2 has two of them: One for PCM audio and one for DSD audio, since different roll-off frequencies are required. And 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. Good ol' times Two 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. Tape Monitor A tape monitor circuitry is quite retro. Who still owns a cassette recorder or a tape machine? Who still has an equalizer? To put it another way: Who would like to do it and cannot do it because no manufacturer offers a tape monitor circuit anymore? Well, those who want can again. Today’s audio levels are much higher than in the past and many recorders will be overloaded by them. In that case simply press the switch and the level to the recorder is lowered by 10 dB. To compensate, the level from the recorder is increased by 10 dB. Et voilà. Preamp & Direct Out On the output side, power amplifiers or active loudspeakers can be connected to the preamp output sockets using XLRs. The Direct Out via RCA picks up the output signal before the volume control and thus offers the possibility to connect a headphone amplifier. AMP CTL Practical are the AMP CTL sockets to which power amplifiers with 12 V trigger inputs can be connected in order to remotely switch them on and off together with the Director Mk2. Specifications: Analog Inputs & Outputs: XLR (balanced), RCA Maximum input & output gain: 32.5 dBu Input impedance (RCA): 47 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 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 (RCA), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optical SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

Finally an analog crossover again It is difficult to optimally integrate subwoofers. At present, there are almost only digital crossovers for this purpose. But who wants to convert back and forth? You don’t need to anymore. Our Crossover sets the crossovers anew. Crossing over The crossover is divided into two section: The Low section for the subwoofer (mono or stereo) and the Mid-Hi section for the top/main speakers. The crossover frequencies for Low and Mid-Hi can be set individually from 50 Hz to 120 Hz in 6 positions. The Low filter has a slope of 24 dB per octave. The slope of the MID-HI filter can be switched from 12 dB (Butterworth) to 24 dB (Linkwitz/Riley). In Phase Audio signals should come from the speakers and subwoofers in phase. The crossover offers variable phase control for this purpose. Use the toggle switch to activate the phase control (phase 0° or phase -180°) and use the control to establish simultaneousness. If the phase control is not required, it can be switched off and thus disappears completely from the signal path. One or two? If you want to connect two subwoofers (one for L and one for R), this switch should be set to stereo. If you only use one subwoofer, switch to mono. The crossover creates a mono signal of the Low frequencies and outputs it to both Low outputs (L + R). On or off? If you prefer to run the top/main speakers full-range, you can switch off the MID-Hi filter. Get rid off it With the Low Cut filter all frequencies below 20 Hz can be filtered out with a 6dB roll-off. This way, no inaudible low frequencies get into the power amplifier, which relieves it a lot. As long as you're having fun The output volume of the subwoofer can be varied from -10 dB to +10 dB. In addition to the purely corrective adjustment of the speaker system, you can add more bass into recordings that lack low end.  It’s like saying, “Happiness is when the bass kicks in.” Good ol' times Two 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. Sources The Elector offers both balanced inputs with XLR sockets and standard inputs with gold-plated RCA sockets. Low & Mid-Hi On the output side, the power amplifiers or active loudspeakers or subwoofers are connected to the Low & Mid-Hi output sockets by means of balanced XLRs. This means that even long cable runs to e.g. mono power amps placed next to the speaker are possible without the risk of interference. AMP CTL Practical are the AMP CTL sockets, to which up to four power amplifiers with 12 V trigger inputs can be connected to switch them on and off together with the crossover. The crossover itself can also be switched on by a preamplifier like the Director Mk2 or Phonitor x. Specifications: Analog Inputs & Output: XLR (balanced) and RCA Maximum input & output gain: 32.5 dB Input impedance (RCA): 20 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -70 dBu THD (0 dBu, 1 kHz, MID-Hi filter on): 0.0008 % THD (0 dBu, 30 Hz, LOW filter on): 0.0014 % Noise (A-weighted, MID-HI output): -106 dBu Noise (A-weighted, LOW output): -104,9 dBu Dynamic range (MID-HI output): 138 dB Dynamic range (LOW output): 136.9 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. 50 VA Stand-by power consumption: < 0,3 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 278 x 100 x 330 mm Unit weight: 5,4 kg Shipping weight (incl. packaging): 7,0 kg

Finally an analog crossover again It is difficult to optimally integrate subwoofers. At present, there are almost only digital crossovers for this purpose. But who wants to convert back and forth? You don’t need to anymore. Our Crossover sets the crossovers anew. Crossing over The crossover is divided into two section: The Low section for the subwoofer (mono or stereo) and the Mid-Hi section for the top/main speakers. The crossover frequencies for Low and Mid-Hi can be set individually from 50 Hz to 120 Hz in 6 positions. The Low filter has a slope of 24 dB per octave. The slope of the MID-HI filter can be switched from 12 dB (Butterworth) to 24 dB (Linkwitz/Riley). In Phase Audio signals should come from the speakers and subwoofers in phase. The crossover offers variable phase control for this purpose. Use the toggle switch to activate the phase control (phase 0° or phase -180°) and use the control to establish simultaneousness. If the phase control is not required, it can be switched off and thus disappears completely from the signal path. One or two? If you want to connect two subwoofers (one for L and one for R), this switch should be set to stereo. If you only use one subwoofer, switch to mono. The crossover creates a mono signal of the Low frequencies and outputs it to both Low outputs (L + R). On or off? If you prefer to run the top/main speakers full-range, you can switch off the MID-Hi filter. Get rid off it With the Low Cut filter all frequencies below 20 Hz can be filtered out with a 6dB roll-off. This way, no inaudible low frequencies get into the power amplifier, which relieves it a lot. As long as you're having fun The output volume of the subwoofer can be varied from -10 dB to +10 dB. In addition to the purely corrective adjustment of the speaker system, you can add more bass into recordings that lack low end.  It’s like saying, “Happiness is when the bass kicks in.” Good ol' times Two 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. Sources The Elector offers both balanced inputs with XLR sockets and standard inputs with gold-plated RCA sockets. Low & Mid-Hi On the output side, the power amplifiers or active loudspeakers or subwoofers are connected to the Low & Mid-Hi output sockets by means of balanced XLRs. This means that even long cable runs to e.g. mono power amps placed next to the speaker are possible without the risk of interference. AMP CTL Practical are the AMP CTL sockets, to which up to four power amplifiers with 12 V trigger inputs can be connected to switch them on and off together with the crossover. The crossover itself can also be switched on by a preamplifier like the Director Mk2 or Phonitor x. Specifications: Analog Inputs & Output: XLR (balanced) and RCA Maximum input & output gain: 32.5 dB Input impedance (RCA): 20 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -70 dBu THD (0 dBu, 1 kHz, MID-Hi filter on): 0.0008 % THD (0 dBu, 30 Hz, LOW filter on): 0.0014 % Noise (A-weighted, MID-HI output): -106 dBu Noise (A-weighted, LOW output): -104,9 dBu Dynamic range (MID-HI output): 138 dB Dynamic range (LOW output): 136.9 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. 50 VA Stand-by power consumption: < 0,3 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 278 x 100 x 330 mm Unit weight: 5,4 kg Shipping weight (incl. packaging): 7,0 kg

Finally an analog crossover again It is difficult to optimally integrate subwoofers. At present, there are almost only digital crossovers for this purpose. But who wants to convert back and forth? You don’t need to anymore. Our Crossover sets the crossovers anew. Crossing over The crossover is divided into two section: The Low section for the subwoofer (mono or stereo) and the Mid-Hi section for the top/main speakers. The crossover frequencies for Low and Mid-Hi can be set individually from 50 Hz to 120 Hz in 6 positions. The Low filter has a slope of 24 dB per octave. The slope of the MID-HI filter can be switched from 12 dB (Butterworth) to 24 dB (Linkwitz/Riley). In Phase Audio signals should come from the speakers and subwoofers in phase. The crossover offers variable phase control for this purpose. Use the toggle switch to activate the phase control (phase 0° or phase -180°) and use the control to establish simultaneousness. If the phase control is not required, it can be switched off and thus disappears completely from the signal path. One or two? If you want to connect two subwoofers (one for L and one for R), this switch should be set to stereo. If you only use one subwoofer, switch to mono. The crossover creates a mono signal of the Low frequencies and outputs it to both Low outputs (L + R). On or off? If you prefer to run the top/main speakers full-range, you can switch off the MID-Hi filter. Get rid off it With the Low Cut filter all frequencies below 20 Hz can be filtered out with a 6dB roll-off. This way, no inaudible low frequencies get into the power amplifier, which relieves it a lot. As long as you're having fun The output volume of the subwoofer can be varied from -10 dB to +10 dB. In addition to the purely corrective adjustment of the speaker system, you can add more bass into recordings that lack low end.  It’s like saying, “Happiness is when the bass kicks in.” Good ol' times Two 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. Sources The Elector offers both balanced inputs with XLR sockets and standard inputs with gold-plated RCA sockets. Low & Mid-Hi On the output side, the power amplifiers or active loudspeakers or subwoofers are connected to the Low & Mid-Hi output sockets by means of balanced XLRs. This means that even long cable runs to e.g. mono power amps placed next to the speaker are possible without the risk of interference. AMP CTL Practical are the AMP CTL sockets, to which up to four power amplifiers with 12 V trigger inputs can be connected to switch them on and off together with the crossover. The crossover itself can also be switched on by a preamplifier like the Director Mk2 or Phonitor x. Specifications: Analog Inputs & Output: XLR (balanced) and RCA Maximum input & output gain: 32.5 dB Input impedance (RCA): 20 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 Ω Common mode rejection (XLR): -82 dBu Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -70 dBu THD (0 dBu, 1 kHz, MID-Hi filter on): 0.0008 % THD (0 dBu, 30 Hz, LOW filter on): 0.0014 % Noise (A-weighted, MID-HI output): -106 dBu Noise (A-weighted, LOW output): -104,9 dBu Dynamic range (MID-HI output): 138 dB Dynamic range (LOW output): 136.9 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. 50 VA Stand-by power consumption: < 0,3 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 278 x 100 x 330 mm Unit weight: 5,4 kg Shipping weight (incl. packaging): 7,0 kg

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

200 Watt Power Soak The Reducer reduces the output power of guitar or bass amplifiers. Enjoy the rich tube sound and full power amp distortion of your favorite amp even at room volume. Reduction The reducer is connected between the amplifier (head) and the loudspeaker. Here it converts the electrical energy of the amplifier into heat and transmits a reduced power to the loudspeaker. The art here is that the power reduction does not affect the sound, so play quietly with the original sound. The desired reduction is first switchable, in steps of 3 dB and then 5 dB. In the last switch position “Variable”, a load potentiometer is used to continuously reduce the level reduction from -7 dB to almost complete fade-out. Impedances The reducer can be connected to amplifiers (heads) with all common impedances. Available are 4 Ω, 8 Ω or 16 Ω. Specifications: Inputs & Outputs; 1/4" (6,35mm) TS-Jack (unbalanced) Load impedances, selectable: 4 Ω, 8 Ω, 16 Ω Maximum input power (4 Ω, RMS): 90 W Maximum input power (4 Ω, Peak): 120 W Maximum input power (8 Ω, RMS):200 W Maximum input power (8 Ω, Peak): 260 W Maximum input power (16 Ω, RMS): 160 W Maximum input power (16 Ω, Peak): 180 W Dimensions & Weight W x H x D (Width x Height w/ feet x Depth): 179 x 85 x 190 mm Unit weight: 1.85 kg Shipping weight (incl. packaging): 2.65 kg

The headphone monitoring amp and preamplifier with Phonitor Matrix. Phonitor 2 is the ultimate headphone amplifier for sound and mastering engineers who also mix and master on headphones. Speakers can also be connected, making Phonitor 2 a full-fledged monitor controller. Milled from the solid aluminum is the 45mm large volume knob. Its mass in combination with the motorized Alps RK27 “Big Blue” potentiometer further enhances the “spoon in honey” turning feeling. Remote control The volume can be adjusted with any infrared remote control, which Phonitor 2 simply learns. Just press the IR programming button on the back and then the buttons for louder and softer. Learning made easy This video explains how you can use any infrared remote control to remotely control the volume. Monitor Controller Features Sources: 2x XLR, 1x RCA Solo L/R Phase reversal for L and R Output switchable to headphones or speakers  Mute Phonitor Matrix with crossfeed, speaker angle and center level (the latter can be switched off) To The Power Amp or to the active loudspeakers it goes symmetrically via XLR sockets. Slave Thru The volume control can be switched off by means of a dip-switch on the bottom of the unit.  The Sources Two balanced and one unbalanced stereo sources can be connected to Phonitor 2. If a CD player or other HiFi equipment is connected to the RCA jacks, the consumer level can be raised from -10 dBV to 0 dBu using a dip switch on the bottom of the unit. In this way the sources are comparable in level. Mechanical VU Meter visualize the input levels, which can be switched in sensitivity with the VU Cal switch. Laterality By its very nature, hearing can be directed more to the left or more to the right. This is particularly evident when listening with headphones. That’s why Phonitor 2 has the finely resolved Laterality control. With it the center can be set perfectly. The Revolution in the headphone amplifier is the Phonitor Matrix, thanks to which you can create and evaluate your production through headphones in the same way as through speakers.  To achieve this, the super stereo effect must be corrected. It makes the placement of voices and instruments in the stereo image appear to be more widely distributed and effects like reverb and delays are overdone. 

Monitor & Talkback Controller The MTC is the complete monitoring solution. Source and loudspeaker selection, mono, dim and mute switching functions, talkback with built-in talkback microphone up to the creation of a headphone mix for the artist. Inputs The MTC offers six stereo inputs. The Musician input should preferably contain the musician’s signal, so that this signal is also available at the Musician control in the Cue Mix section. The Mix input is then intended for the DAW’s master output. Musician and Mix and 2TrA and 2TrB are balanced input stages. A mono signal should always be fed into the left channel to be heard on both channels in the phantom center. 2Tr C and 2Tr D are unbalanced and have RCA connectors. HiFi equipment can also be connected here. All inputs are additive! The MTC mixes all input signals together. The slave out stereo output always carries out the selected input signal(s) (except Musician) unprocessed. Among other things, metering can be connected here. Cue Mix As an individual mix of Musician signal and Mix signal, an analog, latency-free Cue Mix can be created for the musician. For each of the three artist statements: “I can’t hear myself”, “The mix is too loud” and “I can’t hear anything”, there is a dedicated control as an answer. At the back, the Cue Mix can be sent to a headphone amplifier in the vocal booth. Talkback A built-in electret microphone is available for talkback communication, which can either be activated with the Talk button or remotely controlled via a foot switch. The level control sets the correct level for the microphone signal. At the Talk Out output only the talkback microphone signal is send out via a balanced jack socket. The sound engineer can thus speak directly to the speakers or musicians via an active loudspeaker placed in the recording room. Control Room Three pairs of loudspeakers can also be operated in parallel on the MTC. The volume is controlled by an analog potentiometer and a 45mm full aluminium knob. With the Source Blend control an optimal monitor mix can be created quickly. The headphone output can be used to listen in parallel to the selected speaker pair.  The headphones are switched on or off with the HP On switch, so that they can remain plugged in even if they are to be muted. Specifications Analog inputs & outputs: XLR & TRS Jack (balanced), RCA: Maximum input gain (XLR, Jack): 21 dBu Maximum input gain (RCA): 19 dBu Input impedance (Jack): 50 kΩ Input impedance (RCA): 10 kΩ Output impedance (XLR): 100 Ω Common mode rejection (1 kHz): 70 dB Frequency range (-3 dB): 10 Hz – 100 kHz THD +N (10 Hz – 22 kHz, 0 dBu): 0.005 % Noise (A-weighted): -89.4 dBu Dynamic range (XLR, Jack): 110.4 dB Dynamic range (RCA): 108.4 dB Internal Linear Power Supply with Toroidal Transformer: Operating voltage for analog audio: +/- 15 V Operating voltage for relays and LEDs: +12 V Mains Power Supply: Mains voltage (selectable): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 315 mA Fuse for 115 V: T 630 mA Power consumption: max. 10 W Dimensions & Weight: W x H x D (width x height x depth): 272 x 91 x 220 mm Unit weight: 2.3 kg Shipping weight (incl. packaging): 4.4 kg

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

USB / MADI Interface Madicon is a MADI digital audio interface that connects up to 64 input and 64 output channels to the computer via USB. USB Audio Interface Maximum 64 input and 64 output channels via MADI Additional optical stereo output (Ch1/2) 24 bit / up to 192 kHz Hi-Speed USB with detector LED USB bus powered Internal/External Clock Switch External Clock Lock LED Windows & Mac Zero-latency monitoring (FPGA Monitoring Mixer) 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 necessary The sample rate is now transmitted via the MADI stream and Madisons follow automatically – very handy Monitor Mixer Control Room Mixer and six additional individual headphone mixers Intuitive and fast workflow Talkback 128-channel MADI stream metering MADI splitter – one signal can be distributed to 63 other channels Record without a DAW – directly to the hard disk! Automatic arming Displaying the remaining recording time Automatic naming of the files Solo auditions MADI Train & Device IDs This drawing explains the device ID settings when up to four Madison with SmartMADI are connected to the Madicon and a DAW in a daisy-chain connection at single, double and quadruple sample rates. Specifications: Inputs & outputs; MADI, USB, Toslink Optical MADI port: Standard 64/High Optical Toslink port (F06): Max. 192 kHz USB 2.0 High Speed, Bus-Powered, Power consumption (max): 2.5 W Dimensions & Weight W x H x D (width x height x depth): 100 x 42 x 108 mm Unit weight: 0.2 kg Shipping weight (incl. packaging): 0.45 kg

Dual-Channel Microphone Preamplifier in 120V-Technology With the Crescendo duo, we are expanding our 120V microphone preamp portfolio with a dual-channel version of the Crescendo. The Undistortable Crescendo duo convinces with the same outstanding sound and technical features as the flagship of the product range – the eight-channel Crescendo. It is almost impossible to overdrive this microphone preamplifier. It shines with absolute signal fidelity and clarity and lets microphones appear in a whole new light – because they have never been amplified like this before. 48 V  The phantom power required for the use of condenser microphones is activated with the 48V switch. VU -10 In the Crescendo duo, high gains can be used to optimally drive subsequent AD converters. Should the VU meter therefore deflect too far, you can reduce the displayed level by 10 dB. Ø Turn the polarity of a microphone signal 180° by activating the polarity reversal switch. HP A high-pass filter with 6 dB per octave reduces subsonic noise below 120 Hz. Mic Gain The gain is adjusted by a 12-step switch. The pre-amplification ranges from +18 dB to +70 dB. Output The switchable output control allows level adjustment to subsequent devices. The control range is +/- 10 dB. This results in a total gain of 80 dB.   Pad For loud input signals (including line signals) the input sensitivity can be reduced by 20 dB. Built-in Output Splitter Both microphone preamps have dual outputs. The often needed splitter is therefore already built in. Record the microphone signal on a recording system and at the same time record it on a backup system. Specifications Analog input & output XLR (balanced): Maximum input & output level: 32.5 dBu Input impedance: 10 kΩ Output impedance: 75 Ω Frequency range (-3dB): 20 Hz – 100 kHz THD + N (10 Hz – 22 kHz, 32 dB Gain): 0.002 % THD + N (10 Hz – 22 kHz, 63 dB Gain): 0.001 % Noise (A-weighted, 150 Ω, 32 dB Gain): - 94 dBu Noise (A-weighted, 150 Ω, 63 dB Gain): - 67.5 dBu E.I.N. (150 Ω): -130.5 dBu Common mode rejection (1 kHz): 88 dB Internal Linear Power Supply with Shielded Toroidal Transformer: Operating voltage for audio electronics: +/- 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. 20 VA Dimensions & Weight: W x H x D (width x height x depth): 482 x 88 x 300 mm Unit weight: 6.1 kg Shipping weight (incl. packaging): 9 kg

Stereo Monitor Controller The 2Control is an excellent sounding compact monitor controller that perfectly controls your near-field and full-range monitors. It is equipped with our crossfeed control for headphone monitoring to create mixes on phones that translate on speakers. Inputs & Speakers The 2Control, which is analog throughout, allows the connection and selection of two stereo sources. These can be monitored via two stereo speaker sets and two headphones. A third mono output is used to connect a subwoofer or mono loudspeaker. The inputs are additive! The 2Control mixes the input signals together. The volume is controlled by a high quality potentiometer (ALPS RK27 “Big Blue”) with active control and a 45mm full aluminium rotary knob. The inputs are designed for balanced analog line-level signals. The input IN1 is calibrated to a nominal level of +4dBu. The sensitivity of input IN2 can also be adjusted to a nominal level of -10dBV – ideal for connecting consumer level CD/DVD players. Headphone outputs with Crossfeed The 2 Control is equipped with two active output stages for headphones, whose level can be individually adjusted. To be able to judge a mix via headphones, the super stereo effect, which affects the placement of voices and instruments in the stereo image and also reverb rooms and the generally exaggerated effects, must be corrected. This is done by the crossfeed control, thanks to which mixes created on headphones work just as well on speakers. Crossfeed is a part of our Phonitor Matrix. In its largest expansion stage it has three adjustment parameters: Crossfeed, Speaker Angle and Center Level. In the 2Control the Center Level is preset to -1 dB and the Speaker Angle to 30°. These are the most commonly used values. The crossfeed function determines the crosstalk of the channels, the so-called interaural level difference. The intensity of the crossfeed is infinitely variable. SPECIFICATIONS :   Inputs & Outputs Electronically balanced instrumentation amplifiers Input sockets: XLR  Output sockets: XLR, Stereo TRS Nominal input level: IN 1: +4dBu, IN 2: +4dBu/-10dBV Input impedance: 22kOhm Output impedance: 75Ohm/Headphones: 22Ohm Max. input level: 21,3dBu Max. output level: 21,3dBu Volume control range: -80dB to +6,6dB Headphones: -65dB to +14,9dB    Measurements Frequency range: 10Hz to 100kHz (-3dB) CMRR: ›60dB (@1kHz, 0dBu input level, unity gain) THD&N: 0,002% (@1kHz, 0dBu input level, unity gain) Signal to noise ratio: -96dB/Headphones: -85dB (A-weighted) Crosstalk L/R: ›70dB (@1kHz) Dynamic range: 116dB Power consumption: ca. 25W    Power Supply Features: Toroidal transformer with voltage selector Fuses (slow blow): 230V/50Hz: 315mA, 120V/60Hz: 630mA    Dimensions and Weight Housing (BxHxT): 215x80x220mm Depth w. controls & sockets: 245mm Height with feet: 95mm Front height w. opened feet: 126mm Weight: 2,1kg   Note: 0dBu = 0,775V. Specifications are subject to change without notice.  

First Eight-Channel Microphone Preamplifier featuring 120V TechnologyThe Crescendo is the first microphone preamplifier which operates with an internal operating voltage of 120V. Making it absolutely unique, because with the 120V Technology it is nearly impossible to overdrive this preamplifier. The Crescendo excels with absolute signal fidelity and clarity. Rediscover your microphones, they have never been amplified like this.SUPRA op-amp Thanks to the discrete buildup with our proprietary high-voltage amplifiers, the SPL SUPRA op-amps, and thus a possible internal operating voltage of +/- 60V (120V Technology), the Crescendo provides unprecedented technical performance data. Crescendo is capable to amplify even the fastest transients, the finest details and the complete frequency range, clean and almost without any distortion.Use the Mic Gain for the setting of the pre-amplification. You can control a range from 18dB to 70dB. For high-level signals, which do not need to be amplified that much, you can use a PAD with -20dB. Thus the control range of the amplification is shifted to -2dB to 50dB.In and Output StagesThe differential amplifiers in the input stage are built-up with transistor pairs, consolidated in in a special housing. A precise matching of the transistors as well as thermal coupling guarantee a high common mode rejection and extremely low THD values. The choice of transistors and electrical resistors has a massive impact on the sound. Within countless listening sessions, the perfect components for the Crescendo were chosen – It was well worth the effort!ServoThe discrete op-amps of the output stage are able to drive long cable paths without any problems. The output stages operate as current amplification, which, with 6mA quiescent current, also provide mere Class A operation. We almost completely renounced to use coupling capacitors, to avoid their sonic disadvantages (diffuse, blurring, loss of dynamics ). However, to eliminate DC voltage, active servo circuits are used.

The headphone amplifier with preamplifier Enjoy music over headphones and over speakers equally excellently with the superior VOLTAiR technology – that’s why we developed Phonitor x. For all headphones The Phonitor x offers connections for standard headphones with jack plug and for balanced headphones a 4-pin XLR connector. Thanks to the enormous output power of 18 watts, all headphones are driven effortlessly. To the power amplifiers or to the active loudspeakers you can use the balanced XLR- for long, and the RCA sockets for short cable runs. Practical are the AMP CTL sockets to which power amplifiers with 12V trigger inputs can be connected to switch them on and off together with Phonitor x. Milled from solid is the big 45mm Volume control. Its mass together with the  motorized Alps RK27 “Big Blue” potentiometer enhances the “spoon in the honey” feeling even further. The red marker LED makes the setting visible even in darkened environments. Volume RC The volume can be remote controlled with any infrared remote control, that the Phonitor x simply learns. To do this, press the IR programming button and then the buttons for louder and quieter. Source of joy Up to six (6 !) stereo sources can be connected to Phonitor x: XLR, RCA, USB, Coax, Optical and AES/EBU Phonitor x accepts a large variety of audio connections. Balanced via XLR or unbalanced via RCA connectors. With the optional DAC768xs, three additional digital sources can be added. The DAC768xs The 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 new architecture reproduces the finest sound details. It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768kHz, which is equivalent to 16 times CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256. Good ol' times Two mechanical VU meters visualize the input levels for the left and right audio channel. Finding the middle is 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 x 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. To the right, the laterality control is switched on. The Revolution in the headphone amplifier is the Phonitor Matrix, thanks to which you can listen to music through headphones as if you were listening through speakers. The tiring super stereo effect is a thing of the past. Enjoy your music exactly the way it should be heard.  Specifications: Analog inputs & outputs: XLR (balanced) and RCA Maximum input & output gain: 32.5 dBu Input impedance (RCA): 20 kΩ Input impedance (XLR): 20 kΩ Output impedance: 75 kΩ Common mode rejection (XLR): 82 dB Frequency range (-3dB): 4 Hz – 300 kHz THD + N (10 Hz – 22 kHz, 0 dBu): 0.00085 % Noise (A-weighted): -103.8 dBu Crosstalk (1 kHz): -106 dB Dynamic range: 136.3 dB Digital Inputs: DAC768xs (optional) Coaxial SPDIF (RCA), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optical SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

RIAA Phono Preamplifier This phono preamplifier for Moving Magnet (MM) and Moving Coil (MC) pickup systems is based on the legendary Douglas-Self-Design. MM & MC There are two pickup systems for record players: Moving Magnet (MM) and Moving Coil (MC). Moving Magnet Moving magnet pickups generate a small voltage by moving a magnet between copper coils. MM pickups have a very high inductance and impedance. The Phonos offers a choice of three capacitors to correct the linearity of the frequency and phase response. The capacitances are 150pF, 220pF and 330pF. In the off-position the capacitance is determined by the capacitance of the cable. Moving Coil The MC pickup is also a small electromagnetic generator, but with the magnets firmly anchored and the coils moving between them, i.e. exactly the opposite of the MM pickup. The output voltage, however, is very low at only a few hundred microvolts and is therefore more sensitive to noise and induced hum. This makes preamplifier design a challenge, because the very low output voltage must be pre-amplified extremely noise-free. Phonos offers one of the most powerful phono preamplifiers for MC systems with up to 71.5 dB preamplification. Thanks to the 120V technology, the preamplifiers are extremely low-noise, linear and phase-locked. MC pickups have a very low inductance and impedance. This means that the effects of cable capacitance (from turntable to preamplifier) are negligible. The impedance, however, is relevant and can be set in six positions: 100, 220, 470, 2.2 k, 4.7 k, and 10 kOhm. If there is any doubt as to which impedance matches that of the pickup, the impedances can also simply be switched through until the linear frequency response is acoustically determined. Amplification The pre-amplification can be switched in three values. The normal gain value is +46 dB for MM and +67 dB for MC cartridges. In the -10 dB position, the phono produces enough output level for normal hi-fi equipment operating at a reference level of -10 dBV. In the +4 dB position, the output level is about as loud as that of modern DA converters at a reference level of 0dBfs = +15 dBu. Subsonic Filter The rumbling of the turntable’s drive mechanism is also amplified by the high amplification of the bass range due to RIAA equalization in playback. The Phonos offers a switchable subsonic filter to remove this rumble.  Connections Gold-plated RCA sockets are available for connecting the turntable and for outputting to a preamplifier such as the Director Mk2 or the Elector.  A ground connection for the turntable is designed as banana plug socket. The ground lead can be either clamped or plugged in. RIAA Equalization RIAA equalization is a specification for the recording and playback of records proposed by the Recording Industry Association of America (RIAA). The RIAA equalization curves introduced a global de facto industry standard for vinyl records in 1954. RIAA equalization uses pre-emphasis to reduce low frequencies and boost high frequencies when the record is cut (blue RIAA curve). The bass is reduced by up to 20 dB to provide more space and longer record play time. The high frequencies are boosted by up to 25 dB to increase the signal-to-noise ratio between the noise and the hot of the record media to the audio signal. During playback, exactly the opposite happens in the phono. The de-emphasis filter (red RIAA curve) compensates for the pre-emphasis and the result is a linear frequency response. Douglas Self Design The RIAA equalization of the phono was developed according to the recommendations of Douglas Self. Idea: Many small capacities instead of a few large capacities. Advantages: The capacitors charge and discharge faster, which results in better resolution and faster impulse response. Only audiophile Styroflex capacitors are used in the entire RIAA equalizer. About the person: Douglas Self is a British electronics engineer with a strong penchant for audio. He holds honorary degrees from Cambridge University and teaches psychoacoustics at the University of Sussex. Specifications: Analog Input & Output: RCA Input impedance (MM): 47 kΩ Input impedance (MC): switchable Output impedance: 75 Ω Frequency range (-3 dB): 4 Hz – 300 kHz Crosstalk (1 kHz): -80 dBu Moving Magnet (MM) Preamplifier Amplification (Norm): 46 dB Amplification (-10 dB): 36 dB Amplification (+4 dB): 50 dB Capacitance (switchable): Off, 150 pF, 220 pF and 330 pF Noise (A-weighted; 46 dB amplification): -85.3 dBu Dynamic range: 131.3 dB Moving Coil (MC) Preamplifier Amplification (Norm): 67 dB Amplification (-10 dB): 56 dB Amplification (+4 dB): 71.5 dB Impedance (switchable): 100 Ω, 220 Ω, 470 Ω, 2.2 kΩ, 4.7 kΩ, 10 kΩ Noise (A-weighted, 67 dB amplification): -61.7 dBu Dynamic range: 128.7 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. 30 VA Stand-by power consumption: < 0,3 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 278 x 100 x 330 mm Unit weight: 3.2 kg Shipping weight (incl. packaging): 4.3 kg

Performer s800 is a stereo power amplifier that delivers 2 x 285 W to 4 ohms loudspeakers. In bridge mode it delivers 450 W into 4 or 8 ohms.The technical basis of all Professional Fidelity units is the proprietary 120 V Rail Technology, which is also called VOLTAiR technology. The name makes reference to the unequalled high voltage of 120 V (+/-60 V DC) and the huge dynamic range of over 140 dB that goes with it.Features: Bi polar transistor power amplifier with VOLTAiR 120V technology2 x 285 W into 4 ohms; 2 x 185 W into 8 ohms (stereo operation) 450 W into 8 ohms (mono operation – Bridge Mode)Toroidal transformer with 855 VA40800 µF capacitance for faithful reproduction of impulse peaksGold-plated loudspeaker binding posts with protection coversNeutrik XLR connectors for balanced in- and outputsDirect slave through of the input (e.g. for bi-wiring)Input trim switch for attenuation up to -5.5 dB in 0.5 dB stepsAMP CTL connector for remote power on/off switching Overheat protection with Temp LEDDC (direct current) voltage protection with Protect LEDFront and Lid made from aluminum, housing made from sheet metalMade in GermanySpecifications:Frequency response: 1Hz-125kHz, +/- 0.5dBInputs: 1Outputs: 1Filter:  Hi Pass (50 Hz)Signal to Noise Ratio: ›130dBMaximum Level Input: +17dBuMaximum Level Output: +34dBuTotal Harmonic Distortion: 0.0005%Phantom Power: NAIInput Type: 1 x XLR & TRSOutput Type: 1 x XLR & TRS​

All essential functions in a stylish, slim housing. The Phonitor e offers Phonitor Matrix, volume remote control, connection also for balanced headphones, DA converter option and the all superior VOLTAiR technology. For all headphones The Phonitor e offers connections for standard headphones with jack plug and for balanced headphones a 4-pin XLR connector. Thanks to the high output power, all headphones are driven effortlessly. Volume with remote control The volume is adjusted with a milled all-aluminium rotary knob. The Alps RK27 “Big Blue” potentiometer provides a pleasant “spoon in honey” turning feeling. The volume can be adjusted with any infrared remote control. The Phonitor e learns their commands. To do this, press the IR programming button on the back and then the buttons for louder and softer. Source selection A total of five sources can be connected to Phonitor e: XLR, Cinch, USB, Coax, and Optical Phonitor e is equipped with RCA connectors and balanced XLR connectors for the analog inputs. The optional DAC768xs adds three more digital sources: USB, Coax and Optical. The optional DAC768xs The converter chip used in the optional digital-to-analog converter is the highly acclaimed AKM AK4490 Velvet Sound™ premium DAC chip, which thanks to its new architecture reproduces the finest sound details. It converts PCM audio with a resolution of 32 bits and a sampling rate of up to 768kHz, which corresponds to 16 times the CD resolution. Direct Stream Digital is also supported up to a resolution of DSD4 or DSD256.  The Revolution in the headphone amplifier is the Phonitor Matrix, with switchable crossfeed, thanks to which you can listen to music with headphones as if you are listening to speakers. The tiring super stereo effect and the exaggerated reproduction of reverberation rooms and effects are a thing of the past. Enjoy your music without tiring and exactly as it should be heard. The Phonitor Matrix, in its largest expansion stage, has three adjustment parameters: Crossfeed, Speaker Angle and Center Level. In Phonitor e the center level is preset to -1 dB and the speaker angle to 30°. These are the most common values. The crossfeed function determines the crosstalk of the channels, the so-called interaural level difference. There are two switch positions C1 and C2 for the intensity of the crossfeed. Specifications:   Analog Inputs: XLR (balanced), RCA Maximum input gain: 32.5 dBu Input impedance: 20 kΩ Digital Inputs: DAC768xs (optional) Coaxial SPDIF (RCA), PCM sample rates: 44.1/48/88.2/96/176.4/192 kHz Optical SPDIF (Toslink F06), PCM sample rates: 44.1/48/88.2/96/with Glass fibre

With the finest ingredients from SPL, the new flagship of SPL's channel strips establishes the basis for world-class recordings – and each of its particular processing stages can be inserted into studio environments or DAWs like analog plug-ins!The brand-new Frontliner has all tools on board to record vocals and instruments in perfect quality: it merges a preamp, a de-esser, an EQ and a compressor. Thanks to it's individual analog I/Os, all these powerful, hi-end analog processors can be inserted into any studio by the Frontliner pilot, i. e. to use them also during mixing and mastering. Thus it appears that the Frontliner is not reduced to a frontend device – especially DAW based studios get the chance to integrate high-grade analog processing at any stage of the audio production process.

The MixDream is a cascadable, 16-in-2 analog outboard mixer in a 19“/2U format. Any DAW or digital console can be expanded with active analog stereo summing and insert functionality. MixDream advantages High-grade analog summing on just 2U rack space Sophisticated, active Class A/60V stages for analog summing in the quality of the best consoles 16 balanced inserts allow for integrated analog effects with individual and overall hard bypass relays Reduction of A/D conversions (14 A/D conversions can be spared with all inserts connected). All analog tracks can be summed before A/D conversion. Channel adjustments and automation (level, panorama etc.) remain controlled from the DAW so the user loses no digital efficiency Lower DAW processor utilization rates The most efficient possible re-sampling of individual tracks with latency free monitoring Surround capable (from up to 3 MixDream units) Channel capacity expandable through linked units Sensitive and transparent stereo expansion control Analog peak limiter for impressive loudness Master inserts and switchable output transformers from Lundahl Optimized signal pathes, all switching functions via relays Proprietary differential amplifiers for each input Discrete, exceptionally low-noise power supply

The SPL Transient Designer offers a completely new technology for level-independent shaping of the dynamic response of a sound: For the first time it is possible to control the attack and sustain of a signal in a very simple way!Unlike other dynamic devices, the processing is not goverened by the signal level but rather by its dynamic characteristics, so all signals (loud and soft) are processed equally. The Transient Designer allows you to emphasise or smooth the attack and extend or shorten the sustain. For the first time it is possible to shape the dynamic path of a signal as you want, as if for example you were changing the microphone distance and position after the recording has been made. With no effort you can shorten or lengthen the attack of all kinds of percussive signals, such as the bass drum, snares, toms, congas etc. to give them more kick, or to flatten the signals. It s almost like being able to change the amount of drum damping after the recording. The same applies to virtually any other signal: Amplify or reduce the picking sound of an acoustic guitar, hold the sound of the strings longer, reduce the reverbtime of a choir, compress solo vocals, increase intelligibility, actuate the piano pedal electronically , turn down the slapbass a notch or give it even more attack, etc.Input & Output Instrumentation amplifier, electronically balanced (differential), transformerless Nominal input level: +6dB Input impedance: = 22kOhms Output impedance: < 600Ohms Max. input level : +24dBu Max. output level: +22,4dBu Minimum load ohms: 600Ohms Relay Hard Bypass Power Fail Safety

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

SPL 's Gainstation 1 very compact design allows for free placing  outside of a rack . Musicians can place the GainStation 1 easily on a rack or amplifier.To accommodate portability, the optionally available SPL GainBag ensures safe transport of the GainStation 1 and offers additional storage for cables and an average-sized microphone. For more permanent installations, up to four GainStation 1's can be rack-mounted in one optionally available 3-unit mounting frame. Features :Custom-designed and built, fully discrete, class A op-amps (no off-the-shelf stuff here). The GainStation's op-amps feature 60-Volt operating voltage twice as high as most common op-amps�for an incredible dynamic rangeAn extremely high amplifier slew rate of 200 V/microsecond, ensuring clean transmission of high-frequency signal portions and rapid transients with virtually no clippingAn almost entirely DC-coupled signal path, making signal-degenerating condensers unnecessary.An optimized layout guarantees the shortest possible signal paths, while generously proportioned grounding surfaces ensure low impedance and maximum shielding.All switching functions are handled by encapsulated relays with gold-plated contacts.All resistors are within 0.1% tolerance and were selected after extensive listening tests.A no-compromise power supply with extensive additional shielding and seven separately wound and regulated voltages. The power supply is of primary importance to audio quality; its importance should not be underestimated.A tube stage utilizing premium MKP foil condensers and a select 12 AX7 LPS tube for clear, dynamic audio.

The GoldMike Mark2 retains a hybrid solid state and tube construction to combine the best of both worlds.The transistor stage is composed of single transistors in a class A design. The circuitry is fully discrete, and each transistor is completely optimized for its specific task. You will not find any IC’s in this preamplifier stage because they cannot be optimized for this specific application to the degree we aimed for.This all new discrete class A transistor stage is a genuine innovation in the entire preamplifier market at this price level. Features :   Frequency response ‹10Hz bis 90kHz (-3dB)  THD+N (Input level -30dBu, 30dB Gain) 0,016%  Noise (A-w., R=40Ohm, 30dB Gain) -91,2dBu  Dynamic range (30dB Gain) 110dB  E.I.N. 128dBu  Max. output level (sym., XLR+Jack) +26,8dBu  Slew rate solid state stage 200V/µs

The Tube Vitalizer looks impressive, with a nod towards vintage valve equipment. It is a 2U steel box, with a bare metal front panel.There are plenty of black knobs, and three grilles revealing glowing tubes. Actually, most of the glow comes from the lamps illuminating the little VU meters.Below these, two old-fashioned black knobs are labelled "ATTENUATOR". Switches light up red when pressed in, apart from a couple of metal toggles, and there are blue LEDs to show High and Low frequency compression. When you switch on, your attention is drawn to two LEDs, proudly labelled "TUBE STATUS", with the "WARM UP" one glowing for 45 seconds before a relay kicks into "ACTIVE" mode. During this 45 second period the circuitry slowly builds up to 250 volts to increase the life of the valves. This is sensible, but I am not sure that the LEDs are really necessary. The back panel features XLR connectors and also plastic balanced jack sockets for inputs and outputs. The unit runs at +6dB instead of the more usual +4dB. Jack connectors are also set to +6dB operation, unfortunately without a switch to enable -10dB operation.

8-channel Volume Controller Volume 8 is an active DAW monitoring controller and a puristic surround preamplifier. Features High-end volume potentiometer and illuminated mute switch from ALPS; the potentiometer controls directly (no VCA or DAC circuits) The mute switch allows a quick reaction when speakers or ears need to be protected (e.g. in case of a computer crash) In contrast to digital volume control, where the bit resolution is continuously reduced as the volume decreases, the signal does not deteriorate with analog volume control. In contrast to passive volume control, active volume control does not affect the phase and frequency response. The monitoring signal sounds reliably the same at any monitoring volume. Balanced DB25 (TASCAM standard) connectors High-quality balancing stages with very high common-mode rejection (minimizing interference/interference in the signal paths) High-quality power supply unit with toroidal transformer Solid aluminium knob with 45mm diameter for smooth, sensitive operation Elegantly designed and compact housing for flexible desk positioning close to the screen. Analog Volume Control The monitoring volume is controlled analogously with an 8-gang potentiometer. If the volume is controlled digitally, for example in the DA-converter, the bit resolution decreases with decreasing volume with one bit per -6dB. Usually we listen with about -30 dB at full volume, which decreases the bit resolution by 5 dB. Active or Passive? It is claimed that passive circuits are less signal influencing and more puristic and therefore better. Well, that is not so. A passive control element such as a potentiometer in the signal path, which is not electronically buffered, influences the frequency response and phase position. Both change continuously as you change the volume. This is exactly what does not happen with an active control. Don't Panic The Mute switch has two rights of existence. On the one hand, it allows the speakers to be muted without having to touch the volume control and thus leaving the set volume.  Secondly, the Mute switch is a panic switch. Whenever the computer hangs or crashes, there is a risk that a full level signal will reach the speakers. This can be extremely loud and may damage the speakers. The mute switch can be used to quickly turn off the signal and protect the speakers and your hearing. Specifications: Analog inputs & outputs; DB25 (TASCAM standard, balanced) Maximum input & output gain: 21 dBu Input impedance: 22 kΩ Output impedance: 100 Ω Frequency range (-3 dB): 10 Hz – 100 kHz THD + N (10 Hz – 22 kHz, 15 dBu): -100 dBu Noise (A-weighted): -102 dBu Crosstalk (1 kHz): 80 dB Common mode rejection (1 kHz): 60 dB Dynamic range: 123 dB Internal Linear Power Supply with Toroidal Transformer Operating voltage for analog audio: +/- 15 V Operating voltage for relays and LEDs: +12 V Mains Power Supply Mains voltage (selectable): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 250 mA Fuse for 115 V: T 500 mA Power consumption: max. 15 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 215 x 100 x 220 mm Unit weight: 1.9 kg Shipping weight (incl. packaging): 4 kg

Stereo Volume Controller Volume 2 is an active DAW monitoring controller and a puristic high-end preamplifier. Features: High-end volume potentiometer and illuminated mute switch from ALPS; the potentiometer controls directly (no VCA or DAC circuits) The mute switch allows a quick reaction when speakers or ears need to be protected (e.g. in case of a computer crash) In contrast to digital volume control, where the bit resolution is continuously reduced as the volume decreases, the signal does not deteriorate with analog volume control. In contrast to passive volume control, active volume control does not affect the phase and frequency response. The monitoring signal sounds reliably the same at any monitoring volume. Balanced XLR input and output connectors High-quality balancing stages with very high common-mode rejection (minimizing interference/interference in the signal paths) High-quality power supply unit with toroidal transformer Solid aluminium knob with 45mm diameter for smooth, sensitive operation Elegantly designed and compact housing for flexible desk positioning close to the screen. Analog Volume Control The monitoring volume is controlled analogously with an Alps-RK 27 potentiometer. If the volume is controlled digitally, for example in the DA-converter, the bit resolution decreases with decreasing volume with one bit per -6dB. Usually we listen with about -30 dB at full volume, which decreases the bit resolution by 5 dB. Active or Passive? It is claimed that passive circuits are less signal influencing and more puristic and therefore better. Well, that is not so. A passive control element such as a potentiometer in the signal path, which is not electronically buffered, influences the frequency response and phase position. Both change continuously as you change the volume. This is exactly what does not happen with an active control. Don't Panic The Mute switch has two rights of existence. On the one hand, it allows the speakers to be muted without having to touch the volume control and thus leaving the set volume.  Secondly, the Mute switch is a panic switch. Whenever the computer hangs or crashes, there is a risk that a full level signal will reach the speakers. This can be extremely loud and may damage the speakers. The mute switch can be used to quickly turn off the signal and protect the speakers and your hearing. Specifications: Analog inputs & outputs:XLR (balanced) Maximum input & output gain: 21 dBu Input impedance: 22 kΩ Output impedance: 100 Ω Frequency range (-3 dB): 10 Hz – 100 kHz THD + N (10 Hz – 22 kHz, 15 dBu): -100 dBu Noise (A-weighted): -102 dBu Crosstalk (1 kHz): 80 dB Common mode rejection (1 kHz): 60 dB Dynamic range: 123 dB Internal Linear Power Supply with Toroidal Transformer Operating voltage for analog audio: +/- 15 V Operating voltage for relays and LEDs: +12 V Mains Power Supply Mains voltage (selectable): 230 V AC / 50; 115 V AC / 60 Hz Fuse for 230 V: T 250 mA Fuse for 115 V: T 500 mA Power consumption: max. 15 W Dimensions & Weight W x H x D (width x height incl. feet x depth): 215 x 100 x 220 mm Unit weight: 1.6 kg Shipping weight (incl. packaging): 3.7 kg

Surround Monitor Controller The SMC is the industry standard for 5.1 surround monitoring. It combines analog 6-channel volume control with speaker management for 5.1 surround and stereo sources. Source Selection & Speaker Management Two 5.1 and two stereo input sources can be connected to the SMC. This is important because in monitoring, two productions are always compared with each other. The DB25 input is used to connect the DAW balanced. In output routing, channels 1-6 should be selected for 5.1 mixing and 7/8 for stereo mixing. DVDs are often played back on standard DVD or Blu-Ray players for comparison purposes. Their consumer nominal level of -10dBV, which corresponds to a level difference of 7.8dB to the 0dBu studio level, is compensated in the SMC to make loudness comparable.  The front and rear stereo pair can each be switched to mono.  Mute All is important when the DAW is sending full-scale audio garbage due to a problem. To protect the speakers and your hearing you can quickly mute here. Each loudspeaker can also be switched on/off individually. The balanced inputs are passively looped through to the Slave Out DB25 socket. Here, for example, a multi-channel metering can be connected. The speaker outputs are balanced with TRS jacks. If mono jack plugs (TS) are used, the signal is automatically unbalanced. Specifications: Analog inputs & outputs; DB25 & TRS Jack (balanced), RCA Maximum input gain: 21 dBu Input impedance (DB25 & Jack): 50 kΩ Input impedance (RCA): 10 kΩ Output impedance (Jack): 100 kΩ Frequency range (-3 dB): 10 Hz – 100 kHz THD +N (10 Hz – 22 kHz, 18 dBu): 0.002 % Noise (A-weighted): -98 dBu Common mode rejection (1 kHz): 70 dBu Dynamic range:119 dB Internal Linear Power Supply with Toroidal Transformer Operating voltage for analog audio: +/- 15 V Operating voltage for relays and LEDs: +12 V Mains Power Supply Mains voltage (factory set): 230 V AC / 50 or 115 V AC / 60 Hz Fuse for 230 V: T 315 mA Mains voltage (selectable): 115 V AC / 60 Hz Fuse for 115 V: T 630 mA Power consumption: max. 10 W Dimensions & Weight W x H x D (width x height x depth): 272 x 91 x 220 mm Unit weight: 2,3 kg Shipping weight (incl. packaging): 4,4 kg