Recording Studio Magic!

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

Combining some of our most popular products in an easy-to-use and install package, a home studio owner can get everything required to have a treated studio that will sound great and look even better. Depending on the goals or the available budget for the room you may acquire one or several boxes. Consisting of one single box containing 8 units Wavewood Ultra Lite, 2 units Multifuser DC3 and 2 tubes of Glue to install it all, VicStudio Box is an acoustic kit appropriated to treat your Project Studio. Acoustic solutions will always have a visual impact. The VicStudio Box can be acquired in three different finishes, so that you have more options available when it comes to deciding the look you want for your room. The kit includes the new Multifuser DC3, with built-in holder rips so it can be installed on a wall with a VicFix fixation system, using VicFix Mini (included), or VicFix J Profile 80mm (sold separately), or VicFix Corner (sold separately). It continues to be compatible with Flexi Glue Ultra (included), which is required for ceiling installation.

Flexi Wall is a portable, modular acoustic treatment system which can transform your working space in a matter of minutes. Its flexible panels enable you to set up a customized studio just about anywhere. The panels can also be used to change the acoustic properties of a room, allowing it to serve a variety of purposes. Flexi Wall is highly absorbent in all frequency ranges. Its design is based on Vicoustic’s top selling Flexi Panels. This unique solution can be used in a range of environments such as recording rooms, vocal booths, video editing rooms, etc.Packaging InformationUnits/Box: 3Box Dimensions (L×H×W): 620 x 620 x 500mm / 24.4'' x 24.4'' x 19.7''Box Gross Weight: 12.75kg

Flexi Screen Ultra has over the years probably evolved as the best looking mobile acoustic shield for microphones in the pro audio industry. We now take this best-selling product to the next level, by presenting an improved version with anti-scratch melamine and eight color patterns, including Metallic Copper and Gold.Made from high quality materials, the exterior surface is produced in MDF with anti-scratch melamine, which helps to create a barrier and gives an attractive visual effect. Its polyurethane structure in the interior is designed to give maximum absorption efficiency.It's the ideal portable solution for voice recording in untreated rooms or venues, which lack sufficient acoustic control. It performs on untreated and scattered room reflections, effectively isolating the sound source. The singer's energy is absorbed on the inside of the unit, producing a dryer, less ambient sound. At the same time, scattered room reflections are blocked on the outside.Flexi Screen Ultra can be used with a wide range of microphones and can be adjusted either horizontally or vertically. It's the best solution for voice recording, either for singers, musicians or podcasters.

Cinema Round Premium panels provide flexible and elegant solutions for sound control across a multitude of applications. Combining modern design with maximum acoustic efficiency, the fabric-covered Cinema Round Premium panel is commonly used to control sound reflections and excess reverberation, helping you maximize the performance of your listening space.Cinema Round Premium has a broadband performance ranging from medium-low to high frequencies. It has been proved as one of the most stable panel with a very high and linear absorption coefficient.You may also install this product with the new VicFix J Profile fixation system using VicFix Base (sold separately).

Cinema Round Premium panels provide flexible and elegant solutions for sound control across a multitude of applications. Combining modern design with maximum acoustic efficiency, the fabric-covered Cinema Round Premium panel is commonly used to control sound reflections and excess reverberation, helping you maximize the performance of your listening space.Cinema Round Premium has a broadband performance ranging from medium-low to high frequencies. It has been proved as one of the most stable panel with a very high and linear absorption coefficient.You may also install this product with the new VicFix J Profile fixation system using VicFix Base (sold separately).

Made from solid wood, the Multifuser collection is perfect for use in venues such as concert halls, hi-fi rooms and recording studios, where effective diffusion is often required, without too much absorption occurring at the same time. With its striking angled surface, the two-dimensional diffuser is based on a QRD sequence combined with changing reflection techniques (a result of the angled surface).The panel itself is made in two parts. Each part can be rotated in different directions so that a uniform, omni-directional scattering of sound is achieved, with particularly effective diffusion of mid and low frequencies. Multifuser Wood 64 works between 310Hz and 8kHz and Multifuser Wood 36 works between 470Hz and 10kHz.As well as its acoustic efficiency, the panel’s attractive appearance makes it suitable for use in a range of settings. Available in Light Brown, Black, White and two new colours: Metallic Coper and Metallic Gold.

VicTotem is a free standing, variable acoustic treatment solution to provide absorption, diffusion, bass management and even mix all in one: Fully Sound Absorption: Switch all modules to have the VMT side on the front in order to use it as a sound absorber. Fully Sound Diffusion: Switch all modules to the Melamine wood side on the front to use as diffuser. Hybrid panel: Combine VMT and melamine front modules for absorption and diffusion. Bass Trap: Place it on the corners to serve as bass trap. Expandable: It’s easy to add an extra module for added height to the unit.

Engineers searching for a full-featured monitor controller for their DAW-based studio will find much to love about the API MC531. Inspired by the center section of API's acclaimed analog consoles, the MC531 includes a 41-detent control room knob for effortless volume control, along with a generous compliment of I/O. Programmable subwoofer integration makes incorporating your sub a piece of cake. The feature-packed desktop monitor controller also includes dual headphone outputs and a built-in talkback mic. Beyond that, the MC531 is built using the exact same circuitry you'd find in a full-blown API console, yielding you low noise and crystal-clear sound. Whether you work in a large commercial space or an intimate project studio, the API MC531 provides computer-based workflows with large console-style monitor control.

Limited edition OC818 Studio set with multiple polar diagrams (Omni, Figure8, Cardioid, HyperCardioid, etc), based on the new CKR12 Double Cockpit design with world patent for double-handed ceramic rings. Multiple polar charts can be checked in addition to the traditional manual way and through the free POLARPILOT application (Android and iOS) by placing the OCR8 Bluetooth adapter on the back of the microphone, giving the user the ability to adjust 255 steps from Figure8 to Omni , controlling the pad settings. All settings through the application are done in a digital environment, but the sound of the microphone does not change, it does not change and remains proportional. A further breakthrough is the dual output with a single front-rear cap allowing you to record on two separate channels that can be further processed and change the microphone pattern after recording via the dedicated cable / the adapter included in the package for the second outlet. 

Expressive EQ with Vintage Smoothness! The magic behind the BAX EQ is decades in the making. Inspired by the classic Baxandall shelving curves — renowned by mastering engineers in the 60s and 70s for smooth, musical sloping and a unique sonic signature — the Dangerous Music team went to work creating the modern BAX EQ.This unique stereo equalizer features stepped controls for repeatability and identical stereo operation. When what you want is an expressive equalizer — backed up with audiophile accuracy — you've found it. Treat your mixes to the Dangerous Music BAX EQ.Dangerous Music BAX EQ Stereo Mastering and Mix Bus Shelving Equalizer Features:Inspired by the Classic P. J. Baxandall Shelving CurvesStepped controls throughout for repeatability and identical stereo operationBroad Q Shelving7-position High Pass and Low Pass Filters — 2-pole Butterworth configuration (12 dB/oct)8-position High and Low Frequency Select+/- 5dB Cut and Boost controls in 1/2dB steps

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

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

Mastering 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 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

The Dutch & Dutch 8c is a highly accurate active studio monitor combining revolutionary design with powerful DSP to offer perfect integration in any room with wide sweet spot and natural and detailed sound. The Dutch & Dutch 8C uses a proprietary waveguide tweeter with an acoustic cardioid midrange and boundary-coupled bass drivers and active room matching, to provide stunning acoustical reproduction in any room. Unlike traditional speakers, the 8C can be placed closer to room boundaries without losing performance. Built to provide superior detail and clarity, the 8c provides constant directivity from 100Hz upwards and has a linear frequency response from 35Hz and up (+/- 0.5dB) offering unparalleled neutrality, precision and in-room tonal balance. The subwoofer is powered by a 500Watts amplifier feeding two high-excursion 8” drivers with boundary loading to ensure uncompromised precision and impact. The cabinet is made of high-quality birch plywood surrounded by stunning solid oak exterior panels. Despite their compact footprint, the speakers include high-quality DACs, subwoofers, DSP and room matching and streaming capabilities. DSP includes controls to let you fine tune your system, emulate another loudspeaker or simply tweak the sound a little. The DSP can be controlled via its web-based application or a native App on your phone or tablet. Automatic updates are also available with new firmware provided to keep your speaker up-to-date with the latest standards and features. Overview.Completely accurate music reproduction requires more than a thoroughly optimized speaker. It requires a speaker that works together with the acoustics of the listening room.The 8c is more than just a loudspeaker. It is a unique acoustic concept because its revolutionary design provides constant directivity from 100 Hz upwards. As a result, the 8c is above and beyond any other system in terms of neutrality, precision, and in-room tonal balance.Accurate.The 8c represents our commitment to accuracy and clarity. With its constant directivity from 100Hz upwards and +/-0.5dB frequency response from 35Hz upwards, music can now be experienced exactly as it was intended. Nothing is added, nothing left out.Experience the sensation of truly accurate music reproduction, hear truth in audio like never before.
Adaptive.Tune the system to the room, not the room to the system. 8c’s adapt to the room they are in and can be placed (very) close to room boundaries without losing performance.To achieve this, 8c’s combine a proprietary waveguide tweeter with an acoustic cardioid midrange, boundary-coupled bass drivers and active room matching. The result: superb acoustical reproduction in any room,.All-in-one.Besides drivers, the 8c’s small footprint presence holds high-end DACs, amps, subwoofers and a DSP. The system also comes with room matching and streaming capabilities out-of-the-box.All components match each other perfectly and together deliver a “big sound” at 106 dB continuously from 35 Hz upwards, while keeping the signal-to-noise ratio very high. At a glance Compact DSP controlled monitor speaker 2 x 8" woofer, 1" waveguided tweeter, all individually driven Frequency response: 30Hz...25kHz +/- 1dB Phase response: minimum (best possible time coherence). Long term SPL(*): 106dB Controlled Directivity from 100Hz Size: 20x40x40cm, 8"x16"x16" (WxHxD) Weight: 26kg (57lbs) Inputs: Analogue, AES/EBU Selectable correction for free-standing, near wall or in corner DSP for position and frequency response control, speaker emulation Control via Web-based application or phone and tablet Native App

Made in France the new Solo6 is a 2-way monitor offering unrivalled transparency, definition control, dynamics and soundstage precision. Thanks to its full-range Focus Mode, Solo6 forms two monitors in one, preserving the sweet spot during the mix, saving space on the console and simplifying cabling and monitor changes.

Made in France the new Twin6 is a 2.5-way monitor offering unrivaled transparency, definition level, dynamics, and soundstage precision. It's Focus Mode uses a two-way configuration, preserving the sweet spot when mixing, saving space on the console and simplifying cabling and monitor changes.

Classic API Compression for the Modern Studio The API 2500+ Stereo Bus Compressor joins the company’s regular lineup, bringing all that formidable dynamics-shaping power to your rack in a sleek, space-saving 1U chassis. The API 2500+ model added a sophisticated Mix/Blend circuit to API’s venerable design coming form the anniversary edition. API’s flagship compressor delivers the inimitable sound of iconic API compression along with unparalleled parametric control. The result is an incredible-sounding, versatile, world-class stereo bus compressor capable of handling any audio production task. Meticulously designed to deliver a broad range of compression options and features not found on most compressors, the API 2500+ delivers warmth, clarity, and punch, whether used for nuanced dynamics control, brick-wall limiting, or anything in between. Parallel compression done right The Mix/Blend circuit found on the API 2500+ does a lot more than the simple wet/dry blend pot found on many modern compressors, giving you the option of either crossfading or parallel mixing of the compressed and uncompressed signals. First of all, the circuit can be hardwire relay bypassed entirely to deliver a purer signal than a standard blend pot simply maxed out to fully “wet” — a feature golden-eared engineers will appreciate. Next, when engaged and set to X (Cross), the Mix control lets you crossfade between fully uncompressed and fully compressed, literally increasing one while decreasing the other. In || (Parallel) mode, the MIX control effectively dials in your desired amount of compressed signal to blend in with the uncompressed signal, which remains constant. In case you’re wondering; trust us — these two active modes do sound and behave differently and you’ll quickly find uses for each. Equip Your Studio with Iconic API Compression If you’re after versatile compression, the API wrote the book. While the 2500+ is set up for stereo compression, you can also use it dual-mono. However you use it, though, you’ll have a range of compression styles to dial in — all highly musical, all incredibly usable. The 2500+ sports API’s patented Thrust circuit, a proprietary loudness contour that packs a powerful punch, imbuing each octave with the same energy (instead of half the energy as the next lowest one, as is the normal state of affairs).The Old/New Tone Type switch lets you choose between classic feedback and modern feed-forward compression styles. Whether you’re using it for drum stems, guitars, bass, vocals, or as a stereo bus compressor to pump up your mixes, the 2500+ delivers the iconic API analog sound and "slam factor" that has been relied on by recording engineers and heard on countless hits for a half century. Simply put: every serious studio needs the API 2500+. Incredibly versatile, highly musical When you choose API, you know you’re getting legendary sonic character. And with the API 2500+, you’ve got a range of character to choose from. To start, the 2500+ is equipped with API’s patented Thrust circuit before its RMS detector, which delivers punchy, old-school low end. There are also two types of compression onboard the API 2500+ — Old and New. The Old style uses the feedback type of compression in which the detector is placed after the VCA (as found on the classic API 525), while the New style uses a feed-forward type for a faster response typical of many popular modern compressors.The API 2500+ is stereo-ready, but sports a variable link between the left and right channels, so you can independently put the channels to work on your mono sources. Whatever way you use it, you’ll appreciate having API’s auto-makeup gain button on the output stage. It lets you vary the API 2500+’s threshold or ratio, while automatically maintaining a consistent output level. Sidechain input for creative ducking Every dynamics processor has a sidechain circuit that examines the signal and determines how much the VCA will reduce gain — but not every unit provides access to its sidechain. In compressors without a separate sidechain input, your main input signal is, essentially, compressing itself. With compressors that have a separate sidechain input — like the API 2500+ — you can introduce a different signal that determines the compression action on the main signal.An overt illustration of sidechaining is the “pumping” effect that’s popular in EDM production. A control signal — let’s say, a bass drum — is plugged into the sidechain input, triggering the detector and causing the main signal — a synth bass, in this example — to "duck," under each bass drum hit. This clears the way for maximum kick drum impact. Another classic use is for auto-ducking music under a voiceover, as in a radio commercial spot. Sidechaining can be used creatively in many ways, limited only by your imagination! API — the legendary American pro audio company No American pro audio company is more iconic than API. Launched in 1968 by Saul Walker, Automated Processes Inc. began building quality consoles for broadcast applications. Soon, the company’s products found favor with recording studios, and a string of industry-leading innovations (including the 500 Series module, the first VCA, computerized console automation, and more) followed. API produced the 2520 amplifier, one of the most famous op-amps in the recording industry, and their consoles and products found homes in such iconic studio facilities as the Hit Factory, Record Plant, Ocean Way, Sunset Sound, and many others.In fact, there are over 700 vintage API consoles around the globe — many of which are still in use. And the company certainly isn’t resting on its laurels; today’s API products give you the same unbeatable combination of utility, innovation, and great sound. That’s why, at Sweetwater, the name API commands reverence. API 2500+ Stereo Bus Compressor Features: * New Mix/Blend circuit with crossfade or parallel compression* Selectable ratio, attack, and release times* Expanded variable threshold between -20dBu to +20dBu* Selectable NEW or OLD switch for feed-forward or feed-back operation* Over Threshold LED illuminates when input audio crosses the set threshold* Patented Thrust filter with 3 settings for frequency-dependent sidechain control * 3 selectable compressor curve Knee settings: Soft, Medium, or Hard* Choice of Old (feedback) or New (feed-forward) compression* Preset or variable release time control* Variable Left/Right channel linking

Barefoot Sound hit a sweet spot when it introduced the Footprint01, the first in a new line of active 3-way speakers designed specifically for today’s evolving production workflows. Recording engineers, producers, composers, musicians and beat-makers as well as post-production professionals quickly recognized that the Footprint01 offered the sonic character and design features of Barefoot Sound’s high-end studio monitors, but in a smaller package and at a more affordable price.Now, the best 3-way monitor in its class has been joined by the Footprint02, a similar but smaller active nearfield speaker at an even more affordable price, offering the same uncompromising audio quality and design characteristics as the Footprint01. But these two models don’t just share each other’s looks and quality, it also the same Barefoot signature designs — Dual-Force™ technology opposing sub woofers, dual ring radiator tweeter and Multi-Emphasis Monitor Emulation™ or MEME — Technology first featured in their flagship MicroMain and MiniMain series.The Footprint02 looks a lot like its big sister: side-mounted dual opposing paper cone subwoofers; dual ring radiator tweeter and aluminum cone midrange driver set in an aluminum front sub baffle; sleek, rounded corners and edges, minimizing cabinet diffraction. The front baffle speaker configuration is instantly recognizable, with the tweeter positioned below the midrange woofer, an unusual geometry that generates the acoustic focus at the center of the cabinet.Inside they are nearly identical. The Footprint02 houses 2-way DSP crossover feeds, dual Class D amplifiers modules, as befits its reduced cabinet dimensions which is about 60 percent the volume of the Footprint01. Dual opposing 6.5-inch opposing subwoofers offering deep bass extension. The Footprint02 may be small but it is mighty, offering all the characteristics of a Barefoot Sound speaker — exemplary sound quality, 3-D imaging and wide dynamic range.Dual-Force is a core Barefoot technology, placing two subwoofers in opposition to eliminate cabinet vibrations, eliminating all vibration at the source. The only sound heard comes from the drivers, not the cabinet. The dual ring radiator tweeter is another key design component. As the wavelength of high frequency information approaches the diameter of a tweeter diaphragm, the dispersion begins to narrow, adversely affecting the image. Combining a center waveguide with the dual ring tweeter ensures a wider high frequency dispersion pattern at the listening position. Barefoot’s MEME technology enhances the versatility of the Footprint02, allowing it to fit into a broad range of applications: playback room, mobile setup, home, and production environments. MEME offers true emulations modeled on the performance and character of mix cube speakers, the infamous white coned speaker we all know well, and consumer hi-fi speakers, allowing the mixer the ability to reference all of those iconic monitors with a flick of a switch. No more need of multiple monitors cluttering up your bridge.Features:Dual-Force™ TechnologyDual ring radiator tweeterMEME™ TechnologyWide dynamic range350 Watts Total Power per monitorMore affordable price $2,750 USD a pairDescription: 3-way active monitor with MEME™ voice emulationControls: Input level stepped attenuator, MEME voice selectInput Impedance: Balanced XLR, 50k Ohm ImpedanceSensitivity: 90dB @ 1 meter with -15dBV input signal (Level control set to 0dB)Frequency Response: 42Hz – 45kHz (±3dB), 52Hz – 40kHz (±1dB)Bass Response: -3 dB @ 36Hz, Q = 0.707 Slope = 18 dB/octaveCabinet: 13 liters total internal volume, Sealed woofer and midrange enclosures, Machined aluminum baffle plate, Polyurethane acoustic damping throughoutCrossover Frequencies: 180 / 3600 Hz (Passive crossover between Midrange and Tweeter)Tweeter: 1″ ring radiator, Low distortion motor, Rear waveguide chamberMidrange: 4.0″ aluminum cone, Aluminum phase plug, Low distortion motor, +/- 3.5 mm linear excursionWoofers: 6.5″ paper cone, +/- 8.2 mm linear excursionHigh Frequency Amplifier: 150WLow Frequency Amplifier: 200WAC Power Input: 85 VAC to 265 VACPower Consumption: Idle: 2W, Maximum: 320WWeight: Speaker: 27.9 lbs each (12.66 kg)Shipping: 33.6 lbs each (15.25 kg)Dimensions HxWxD: 12.12 x 8.7 x 13.0 inches (308 x 222 x 330 mm) 

The Telefunken ELA M 250/251 was made for Telefunken by AKG after Neumann discontinued its OEM production of the Telefunken U-47. The Golden Age Premier GA-251 is inspired by this truly classic mic model. The GA-251 shares the same design goal that was set for the GA-47: to offer the classic qualities and advantages that a vintage unit in perfect condition would provide and doing so at a very low price point. Having gone through a number of prototypes that was compared to great sounding vintage units in different recording settings, everyone involved agreed that the goal was reached. The coupling capacitor is made by Solen from France. It is a high balanced film capacitor made from MKP material with pure oxygen-free copper connections. It has a very delicate and balanced sound.The capsule capacitor uses NOS Siemensantique grade polystyrene capacitor to reduce loss.The tube cathode bypass capacitor uses an ELNA OFC Cerafine high end hifi capacitor from Japan with oxygen free copper connections exhibiting a very sweet and delicate character, not the least for vocals.The other capacitors are of the low loss polystyrene type.The resistor used for setting the capsule voltage is a high quality DALE 1% metal film resistor. The result is a very stable and consistent operating level leading to better consistency between different GA-251 units. The plate resistor uses a non-inductive resistor of the United States IRC audio with pure copper connections. The sound level is rich and full bodied. The GA-251 uses a 34 mm K251 style capsule. Being a dual-sided one, it provides both cardioid and omni polar patterns.In order to get a good consistency between units, for each production run of 50 pcs GA-251 units, approximately 200 capsules is evaluatedin order to find 50 capsules with specifications that lie within the tight tolerances. The sensitivity of the front and back side of the capsule do not differ more than 0,3 dB. The sensitivity between different capsules do not differ more than 0,5 dB.The tight manufacturing tolerance will lead to a very good consistency between different GA-251 units.The capsule is specially designed to complement the electronic circuit of the GA-251 to achieve a frequency response curve with a nice balance between the high, middle and low frequencies. The result is a very smooth sound with a warm bottom and airy top end just like ELA M251. The vintage style power supply PSU uses a high power R-style transformer with a low noise level and minimum stray field.A simple but effective circuit uses dual-stage filtering for the filament voltage. The High voltage is also precisely filtered and stabilized using an ON semiconductor zener diode. The polar pattern switch is placed on the power supply so that the recording engineer can change the polar pattern between Cardioid and Omni remotely. A ground lift switch will help to solve mains power related noise issues that can appear in some cases.

The classic C800G is the only tube microphone in the world equipped with a semiconductor active cooling system that results in a higher signal-to-noise ratio compared to most other tube microphones. It has a very high-quality modern sound performance.The Golden Age Premier GA-800G is currently the only microphone to fully reproduce the C800G. It´s appearance and structure, as well as accessories and circuit principles, including its sound character, strictly follows the original design. All of the parts including the power plug and the pin definition of the power supply are the same as in the original C800G. Even though its based on the original, the circuit board has been upgraded to a ROGERS ceramic circuit board that has superior insulation parameters and offers a more stable performance. The coupling and bypass capacitors has also been upgraded, resulting in a deeper and more exact low frequency performance and a more delicate and transparent high frequency reproduction.The classic semiconductor active cooling and external cooling system from the C800G is also used in the GA-800G. This is to ensure that the GA-800G has the highest possible signal-to-noise ratio. The tube (a 6AU6) is mounted in an aluminum casing, with the semiconductor cooling chip being attached to the bottom of the case. The temperature of the tube in the GA-800G is the effectively kept down when the GA-800G is in it´s working state, the heat generated by the tube is actively dissipated. The heat that the tube generates is transferred to the reverse side of the semiconductor and transferred to the effective outer aluminum alloy heat sink at the rear of the microphone. This heat sink will then dissipate the heat from semiconductor cooling chip to the air surrounding the GA-800G.PCB board: A ROGERS ceramic gold-plated circuit board is used. This type of circuit board is usually used in high-end aerospace communication products, the cost is about twenty (20!) times higher than an ordinary PCB board.Capsule: By disassembling a C800G and handing it´s capsule to a manufacturer with more than twenty years of microphone capsule production experience, they were able to produce a capsule that matched the characteristics of the original one. The capsule used in the GA-800G can handle a higher sound pressure level than the original one and it has almost the same sound character.Tube: In order to ensure the tube will have a long life, we are using a NOS CIFTE 6AU6 tube from France, it has a maximum anode voltage rating of about 300V. The tube in this circuit has an anode voltage of 90V and will generate more heat than tubes in many other tube microphones that typically uses voltages between 35-50V. The higher voltage used in the GA-800G increases the heat dissipation so the semiconductor active cooling system is an important element in the design.Power supply: The classic 800G uses two 6AU6 tubes as rectifiers for the high voltage rail, they work as pseudo diodes by connecting all grids to the anode that outputs about 290V.After a lot of listening tests comparing a tube rectifying circuit with a solid state one, we found that we did prefer the sound of the latter so this is what we decided to use. The additional advantage is that there is no need to worry about replacing tubes in the power supply.Capacitors, resistors and solder: The metal resistors and electrolytic capacitor from Japan are of special audiophile audio types. The coupling capacitors are Siemens EPCOS from Germany. The connecting wires inside the GA-800G are oxygen-free copper wires from Japan. The solder is FELDER silver/copper-containing solder from Germany.Shockmount:  Rycote USM-VB

Titan and Atlas are world class multi-track interfaces offering flexible expansion and unsurpassable sonic clarity. Using the latest generation of Prism Sounds trusted conversion, Titan is ideal for music and sound recording, multi-tracking, overdubbing, stem-based mastering, analogue summing and critical listening applications. UNSURPASSED CLOCKING AND JITTER REJECTION Prism Sounds history in developing test and measurement equipment for the audio industry has allowed our engineers to get a much deeper understanding of digital clocking and its associated issues. DIGITAL MIXER WITH FLEXIBLE MONITORING Titan has a powerful built-in digital mixer that can be configured to provide a multitude of ultra low latency feeds, Including foldback to performers, stereo and surround outputs. FLEXIBLE I/O Titan has a maximum capability of 18 concurrent input and output channels, offering 8 analogue recording channels, 8 monitoring outputs, stereo digital input and output on a phono connector plus concurrent optical digital I/O ports. FUTURE PROOF CONNECTIVITY With a standard USB connection Titan can connect to the widest range of computers today and for the foreseeable future. Add to this the option of ProTools HDX and Dante modules for the most flexible and professional solution. MDIO MULTI-CHANNEL I/O EXPANSION In an ever changing world of I/O connectivity, the MDIO expansion slot allows your Prism Sound interface to remain relevant, increasing its lifespan and bring a much better return on your investment. Both options allow you to stack multiple Titan or Atlas providing sample accurate phase locked audio between multiple units. Specifications: Dimensions: 1U 19inch rack mountable(rubber feet and rack ears supplied) Table top (inc feet): W: 440mm D: 290mm H: 50mm Rack-mount (inc ears):W: 483mm D: 290mm H: 44.5mm Power: 90-250VAC, 50/60Hz internal power supply IEC 6A connector 35W 0.5A(T) 20mm ceramic fuse Inputs: 4 x combo connectors with XLR sockets for mic input and 6.3mm TRS jack sockets for line input (balanced or unbalanced) 1 x RCA 1 x TOSLINK for S/PDIF or ADAT Optical input Outputs: 8 x 6.3mm TRS jack sockets (balanced or unbalanced) 1 x RCA 1 x TOSLINK for S/PDIF or ADAT Optical output

For medium sized speakers and studio monitors Fine tune your sound the way it’s meant to be heard The ISO Stands are the latest generation of IsoAcoustics stands designed for studio monitors. The ISO Stands are an updated version of the popular ISO-L8R series and features a lower profile, new frame design and a newer version of isolator for improved performance. The ISO-155 isolation stand is 6.1” (155mm) wide x 7.5” (190mm) depth and is rated for monitors weighing up to 40lbs. Offering 14 variations of height and tilt, the ISO-155 will help you optimize the placement of your monitors and bring your tweeters to ear level. The IsoAcoustics patented isolation system provides superior isolation and decoupling from the supporting surface. The stands are biased to keep all energy in alignment with the speaker cones, providing greater clarity and focus. Perfect for optimizing the placement and performance of small studio monitors… An important aspect of the speaker stand design is the ability to adjust the height and tilt angle of the supported speaker enclosure to optimize its position relative to the listener and eliminate soundwave reflections off the supporting surface. In a typical Desktop Audio Workstation (DAW) configuration, the listener is positioned 3 to 4 feet from the monitors with ear level approximately 20″ above the work surface. Raising and tilting the monitors to the optimum position can be achieved in these circumstances using the IsoAcoustics stands. Patented IsoAcoustics Isolation Exciting the supporting surface The isolators manage the energy of the speaker to reduce vibrations resonating through the supporting surface to eliminate dissonant sounds in the listening area or joining rooms. Reducing internal reflections Internal reflections are vibrations reflecting back up the speaker cabinet which causes smear. Smear is a result of artifacts replicated in both channels which are perceived to be in the middle, causing the stereo image to collapse. The IsoAcoustics isolators reduce internal reflections to eliminate smear. The result is an improved stereo image of natural spatial sound. Table Tripod for Speakers / Monitors Allows optimal angle of the speakers in the direction of listening position Decouples the loudspeaker signal from the ground Two height adjustable: 76 and 210 mm Inclination adjustable: Up to 6.5 degrees Supporting surface: 155 x 190 mm Maximum load: Up to 18 kg Weight: 2.3 kg Priced per pair

For Studio Monitors, Guitar, Bass and Other Instrument Amplifiers Fine tune your sound the way it’s meant to be heard The ISO-430 isolation stand is 17” (430mm) wide x 9” (230mm) depth and is designed for Studio Monitors, Guitar, Bass and Other Instrument Amplifiers weighing up to 100 lbs. The ISO Stands are the latest generation of IsoAcoustics stands designed for Pro Audio. The ISO Stands are an updated version of the popular ISO-L8R series and features a lower profile, new frame design and a newer version of isolator for improved performance. The ISO-430 features tilt adjustment of up to 6.5 degrees to allow you to easily focus your monitors to ear level. The IsoAcoustics patented isolation system provides superior isolation and decoupling from the supporting surface. The stands are biased to keep all energy in alignment with the listening position, providing greater clarity and focus. Patented IsoAcoustics Isolation Exciting the supporting surface The isolators manage the energy of the speaker to reduce vibrations resonating through the supporting surface to eliminate dissonant sounds in the listening area or joining rooms. Reducing internal reflections Internal reflections are vibrations reflecting back up the speaker cabinet which causes smear. Smear is a result of artifacts replicated in both channels which are perceived to be in the middle, causing the stereo image to collapse. The IsoAcoustics isolators reduce internal reflections to eliminate smear. The result is an improved stereo image of natural spatial sound. Angle of slope - up to 6.5 degree max. Lifting capacity - 45 kg Dimensions - 430 x 230 x 90 mm (WxDxH) Price - per piece

Building on API’s rich heritage of extremely high-quality recording consoles, THE BOX is a small-format recording/mixing console designed for professional project studios, home studios, and production facilities of all kinds.Optimized for the digital era, THE BOX handles all the functions needed for production not provided by most DAWs, including mic preamps, input signal processing, high-quality mix bus, cue sends with talkback, monitor control, and more, without the redundant capacities of larger consoles.Most importantly, THE BOX provides the legendary “all discrete” API sound in an efficient, cost-effective package.Features:Two (2) input channels with mic/instrument/line preamp, HP filter, & integral 550A EQTwo (2) input channels with mic/instrument/line preamp, HP filter, & 500 slotTwo (2) compressors assignable to input channels or program bus with stereo linkSixteen (16) summing channels (20 channels during mix)Stereo program bus with master fader, insert, and external inputOne (1) stereo and (2) mono auxiliary sends/busesStereo cue send/bus & headphone systemPFL, AFL, and solo-in-place solo modes with stereo solo busFull-featured monitor section that supports two stereo monitor systemsTalkback systemComprehensive rear panel connections with balanced inputs and outputsIntegrated power supply 

As well as new speaker drivers (a 5” midrange and a 10” subwoofer) with high efficiency and power handling, the Trio11 Be monitor enjoys all of Focal's very latest technologies when it comes to acoustics and electronics: TDM and NIC technologies; optimized vents that promote dynamic control and perfect bass definition; high excursion, enabling even more linearity of the subwoofer...Thanks to its class G amplification, Trio11 Be delivers a high SPL (118dB at 1m) and offers great versatility of use and installation configurations in nearfield AND midfield applications. A key asset, always coming along with extremely precise sound reproduction and stereo image. With an extended bandwidth (30 Hz-40 kHz), this loudspeaker is also composed of a 3-way monitor and a 2-way monitor, just like Trio 6 Be which lets you check the mix transfer sound quality thanks to FOCUS mode.With Trio11 Be, power and versatility are waiting for you! Feel the difference!Key PointsHigh SPL: 118 dB @ 393/8'' (1m).FOCUS Mode: two monitors in one with the remote switchover.Pure Beryllium inverted dome tweeterVery low directivity, linearity, dynamics.‘W’ composite sandwich cone on woofer and subwoofer:Neutral sound, no distortion.Bass, mid-bass and treble settings on rear panel: optimum acoustical integration.Optimized decoupling of the midrange speaker driver thanks to silent blocks.Ultra-low distortion and noise.Meticulous Finishes The cabinet developed and manufactured by Focal in its cabinetry workshop in Bourbon-Lancy in France, consists of two side panels and a body in MDF measuring between 13/16” (21mm) and 13/16” (30mm) thick: an indicator of the quality of each material chosen, combined with careful attention to the finishes.Trio11 Be has two side panels in a natural dark red burr ash veneer and a black central body, for an aesthetic rendering that is also discreet and hard-wearing.FOCUS mode 2 MONITORS IN 1 Trio11 Be combines two monitors in a single enclosure. The 3-way monitor provides a flawless control of the entire audio spectrum (30 Hz – 40 kHz). The 2-way monitor, with a frequency response of 90 Hz to 20 kHz, makes it possible to check the transfer quality of mixes on systems with a limited frequency response in the bass (televisions, computers, cars, etc.).

The Neumann U-47 was produced between 1949 and 1965 meaning that the vintage original units are now very old and their performance do differ a lot depending on their condition. The different clones all have their specific sound character too. So, it´s not easy to say which sound is “the right one”. As always, not the least as far as microphones goes, it will ultimately be a matter of taste. Does the world need one more U-47 clone? The GA-47 microphone as the first product in Golden Age Projects new high end line, changing “Project” to “Premier” for this product range, the answer is “Yes!”.The Golden Age Premier GA-47 has been born from a lot of hard work and real world testing. The end result is a microphone with a very sweet character that should appeal to a great number of users and a microphone that can be used for the most demanding recording tasks. It has proved itself to be a great allrounder, this is not a “vocal-only” microphone. The GA-47 is not a product that will be built in huge quantities on a big assembly line. It´s a product that is built by hand, fifty units at a time in a small workshop by a skilled and dedicated staff. It uses very high quality components all the way to achieve the desired goal, allowing me to answer “yes!” to the rephrased question mentioned above. The capsule for the GA-47 is designed to provide the requirements of modern day recordings by mixing the character between a vintage K47 capsule and the one from a K67-style capsule with a wide frequency response and high sensitivity. The result is a truly great sound.

The Royer R-10 is a passive mono ribbon microphone designed for use in the studio and on live stages. Hand-built in our Burbank California factory, the R-10’s sound and performance are all-Royer and it handles SPLs of up to 160 dB @ 1 kHz. The R-10’s compact size and mounting system allows for flexible, unobtrusive positioning.  The R-10’s 2.5-micron aluminum ribbon element is formed with our our patented direct-corrugation process and is protected by a 3-layer windscreen system and internally shock-mounted ribbon transducer. The ribbon transducer is wired for humbucking to reject electromagnetically induced noise.  The R-10’s built-in windscreen provides superior protection from air blasts and plosives. It also reduces proximity effect (bass buildup from close miking) so guitar cabinets and acoustic instruments can be close-miked with less bass buildup. The R-10’s internally shockmounted ribbon transducer isolates the ribbon element from shocks and vibrations, increasing the ribbon element’s durability.The R-10 utilizes a David Royer custom designed transformer for high overload threshold, minimizing saturation at even extremely high sound pressure levels. You’ll never overload an R-10! The mic’s open grill design minimizes standing waves and associated comb-filtering effects and its smooth frequency response, phase linearity and lack of self-distortion make it ideal for all digital recording and live sound formats.The Sound Royer ribbon mics are legendary on electric guitar and the R-10 shows some of its best stuff on studio and live electrics, capturing all the low end, midrange warmth and punch guitarists and engineers have come to expect from a Royer. If you want more bite in the highs but don’t want to multi-mic (particularly on live stages where blending microphones can create phase-related problems), the R-10 takes EQ beautifully and we suggest experimenting with your favorite EQ unit or plugin.  The R-10 is excellent on brass and can handle close-miked trumpets, trombones and other brass instruments. Brass records naturally on an R-10, as bright as the musician plays but without the added sizzle or harshness commonly experienced when condenser mics are used on brass instruments. Close up of mountDrums are full bodied with realistic (not over-hyped) transients response, and the R-10’s figure-8 pattern conveys superb ambience and depth when used for room miking applications. A compressed R-10 in front of the kit sounds huge and punchy. R-10 recordings of violins, ukuleles, steel-stringed and nylon-stringed acoustic guitars, banjos and other stringed instruments are warmth and natural and fit into mixes easily. “Airing out” the recorded track by opening up a bit of 12K with an EQ often gives surprisingly good results.  Vocals record record warm and naturally on an R-10, and the EQ-friendliness of the R-10 allows for dialing in completely different sounds with liberal use of EQ. Try experimenting with cutting bass at 100 Hz and opening up 12K by 3 to 6 dB (or more!). Some great vocal performances were recorded on highly EQ’d ribbon mics (Sixpence None the Richer “There She Goes Again” and “Kiss Me” for example). For audio examples of EQ’d vocals, visit our Audio/Video Library. Royer’s Patented Offset Ribbon TechnologyThe R-10’s patented offset-ribbon design positions the ribbon element toward the front of the transducer, which allows for high SPL handling on the front (logo) side and the option of a brighter response when recording lower SPL sound sources like acoustic guitar, ukulele and vocals on the back side (3 feet or closer; phase reversed position). For more information on the R-series transducer design, see https://www.youtube.com/watch?v=B-KaSiYYcDU The R-10 is available in matched pairs.  Features:High SPL capabilities for electric guitar, brass and other close miking applicationsMulti-layered wind screen provides superior protection to ribbon elementInternally shock-mounted ribbon transducer gives increased durabilityPassive design and custom transformer minimize high SPL overloadHum-bucking transducer design delivers extremely low residual noiseRibbon element not affected by heat or humidityAbsence of high-frequency peaks, “ringing” and phase shiftsEqual sensitivity from front or back of the element  

The R-121 is our flagship microphone and the worlds first radically redesigned ribbon microphone. We did away with the large, heavy, fragile, classic approach to ribbon microphones and went in a completely new direction. The R-121 gives all of the warm, natural tonal response that top engineers have always loved ribbon mics for, but in a compact, light-weight, higher output, tough-as-nails package that was unheard of in a ribbon mic before the R-121.Like many of the best loved classic ribbon mics, the R-121 has a figure-8 pattern, sensitivity in the dynamic mic range, and a warm, realistic tone and flat frequency response. But thats where the similarities end. By using advanced materials and a blend of cutting edge and old-school hand-build construction techniques, the R-121 is a far more versatile and user-friendly ribbon mic that can stand up to the most demanding tasks, including live stage use. Its built solid enough for us to give it a lifetime warranty. The R-121 redefined ribbon microphones so thoroughly that Recording Magazine wrote "the Royer R-121 is destined to become one of the classic microphones of the 21st century".       FEATURESHigh SPL Capabilities               No internal active electronics to overload or produce distortion up to maximum SPL rating              Extremely low residual noise              Ribbon element is not affected by heat or humidity              Absence of high frequency phase distortion              Equal sensitivity from front or back of element              Consistent frequency response regardless of distance               

Το Dangerous 2-Bus παρουσιάστηκε το 1999 και δημιούργησε μόνο του μια νέα κατηγορία αυτή των summing mixers που βοηθούν τον ήχο των DAW  να αποκτήσουν κάτι απο την μαγεία της μίξης σε  αναλογική κονσόλα.Με ανασχεδιασμένο κύκλωμα το νέο  2-BUS+  προσφέρει αξεπέραστη στερεοφωνική εικόνα, σφικτό  ήχο και μεγάλη δυναμική περιοχή.Ταυτόχρονα στη νέα σχεδίαση περιλαμβάνονται 3 κυκλώματα με 3 διαφορετικούς ήχους που μπορεί κάποιος  να προσθέσει στο σήμα και να έχει έτσι μια παλλέτα με διαφορετικές αρμονικές, ανάλογα το κομμάτι και το στύλ μουσικής.Το Paralimit είναι ένα FET-style limiter και  το X-Former είναι ένα ζευγάρι απο  Cinemag output transformers που οδηγούνται απο ένα ειδικά σχεδιασμένο κύκλωμα. Εύκολο στη χρήση και με δυνατότητα σχεδόν απεριόριστων συνδιασμων ήχου, σχεδιασμένο απο τον θρυλικό  Chris Muth το 2-BUS+  είναι άλλο ένα προιόν Dangerous που απευθύνεται σε όσους αναζητούν πάντα τον καλύτερο ήχο για τις παραγωγές τους.Επίσης είναι σημαντικό να ξέρουμε πως μπορεί να έχουμε πολλά πετυχημένα emulation για επεξεργαστές ήχου σε ψηφιακό domain αλλά το analog summing παραμένει αδύνατο να γίνει emulation ψηφιακά.Χαρακτηριστικά:16 channels of the world’s best active analog summing, that even surpasses the original.Three distinctive analog color options: Harmonics, Paralimit and X-Former.Effortless routing and blending of analog color circuits via elegant user interface.Switchable stereo analog insert for easy outboard gear integration.Massive sounds, sacrificing no detail.Crystal clear sonic imaging and three-dimensionality.Endless headroom for modern digital signal levels.Stepped output gain control for exact recalls.Both XLR and D-Sub input connectivity.Audiophile-grade components throughout.

Simple:Hit your choice of buttons and twist the threshold for radio-ready results Smart Dyn Dual Slope Detection: Automatically limits peaks while stealthily compressing the average.Auto Attack/Release: Replace "guess" with "yes" for rapid-fire results. Transparent:Performs like an automated fader: No more drawing vocal rides.True Stereo Dual Detector: Traditional linking shares one detector, often resulting in a narrower stereo image.Immaculate VCA: Virtually no distortion, even with 20dB of gain reduction. Powerful:Internal sidechain circuitry houses the audiophile soul of the compressor:Bass Cut: Tightens the track without sacrificing the low end.Sibilance Boost: Shave sibilance and shrill histrionics with zero post-compression hangover.

Manley designed the Core reference channel strip so that you can hit record and focus on your performance, not your signal chain. It sports the quality, performance, and flexibility their VOXBOX is known for, at a price that makes it accessible to any serious artist or engineer. You get a world-class Manley mic pre, followed by a 3:1 ELOP style compressor for effortless dynamic control. Engineers at Athens Mastering are ecstatic over the active Baxandall EQ, and a fast attack FET brickwall limiter at the end of the chain protects your converters from accidental clipping. Even if you plug right in without tweaking a single knob, you're sure to get an amazing sound.Manley Core Reference Channel Strip Features:Easy to use and awesome-sounding channel strip based on the award-winning VOXBOXClass A tube mic preamplifier with Manley's hand-wound transformersDirect input uses the same circuit as the Manley SLAM! for fantastic direct-instrument recordingsELOP compressor with 3:1 ratio tames any signal transparentlyBaxandall EQ with low/high shelving and sweepable mid for flexible sonic sculptingFast attack FET brickwall limiter ensures you don't clip your converters during recordingFlexible metering can display input, output, and compressor gain reduction levels 

The API Model 500-6B Lunchbox is a six position rack designed to accept all API standard 5.25" X 1.5" modules. This allows an engineer the flexibility to bring along specialized EQ effects to any situation: 550A, 550B or 560. The Lunchbox is self-powered, with a rear panel selected AC voltage switch, and a removable power cord. The input/output jacks are XLR. The Lunchbox comes standard with a +48 volt internal phantom power supply, bussed to pin 15 for interface with the API 512C. API makes several modules for the 500-6B, such as the 512C mic preamp, the 525 compressor, the 560 Graphic EQ, and the 550B four band EQ. Combinations of these modules make perfect stereo recording packages (two 512C's and two 550B's) or vocalist's box, (one 512C, two EQ's and one 525 compressor).         FeaturesHolds 6 API standard modules48 Volt internal phantom power supplySelf-Powered, 100 to 250 Volts, 47 to 63 Hz, SwitchableRubber feet & carrying handle for portabilityRugged steel chassisPerfect for remote recording of Vocalist and InstrumentsOther uses for the versatile Lunchbox include:Multitrack Recording of Acoustic and Electronic Audio Sources.Sound Reinforcement Systems and Recording Live to Multitrack formats.Multiple facility projects where portability is a plus.Broadcast and Audio for Video Production.Custom Channel Input Strips.

The R-122V was originally developed in 2001 as a benchmark to test our R-122 designs against. Over four years of additional development, the unique R-122V has evolved into a world class ribbon microphone with unparalleled richness, depth and detail, especially in the midrange frequencies. R-122V recorded tracks reveal a lush-ness that must be heard to be truly appreciated. The R-122V vacuum tube ribbon microphone offers the ultimate in ribbon microphone performance for those who insist on the very best. The R-122V takes the Royer-pioneered concept of active ribbon microphone technology to an unprecedented level by incorporating vacuum tube electronics into the same proven transducer system used in the venerable R-121 and R-122 ribbon microphones. The high operating voltage of the vacuum tube provides a headroom capability far beyond that capable from a standard phantom power supply. In real-world use, this translates into unmatched clarity, detail and airiness.        FEATURES Vacuum tube electronics provide high output capability, optimal impedance to the ribbon element and low output impedance Extremely low self-noise Operates from a dedicated power supply High SPL capabilities for electric guitar and percussion instruments Absence of high-frequency peaks, "ringing" and phase shifts Ribbon element unaffected by impedance/load, heat or humidity Very low magnetic leakage Equal sensitivity from front or back of element Consistent frequency response regardless of distance Rear side of mic records slightly brighter when three feet or closer to sound source

The AT5040 Studio Vocal Microphone by Audio-Technica marks the debut of their flagship 50 Series of microphones. In addition to vocals, its fast transient response make it ideal for acoustic instruments. Each AT5040 is hand-assembled and individually inspected to ensure the finest quality audio capture.The AT5040 employs a four-part rectangular element to provide purity of sound. Each of the four parts function together and get proprietarily summed to output. This provides a surface area twice the size of a standard one-inch circular diaphragm. Additionally, the AT5040 features an advanced internal shockmounting system, effectively decoupling the capsule from the housing for vibration-free performance.The Audio-Technica AT5040's frequency response chart reflects the company's claim of an "extremely smooth top end with controlled sibilance." A dip of approximately 3dB occurs between 4.5kHz and 8.5kHz, a prime "s" location. The microphone's reactivity slowly tapers after the 10kHz mark. An approximate 2 to 3dB boost in response below 80Hz could create a pleasant promiximity effect on a male vocalist, then rolled off to taste in mixing.Specifications:Element Type: Fixed-charge back plate, permanently polarized condenser Polar Pattern: Cardioid Frequency Response: 20 to 20,000 Hz Open Circuit Sensitivity: -25 dB (56.2 mV) re 1V at 1 Pa Impedance: 50Ω Maximum Input Level: 142 dB SPL, 1 kHz at 1% T.H.D. Noise: 5 dB SPL (typical, A-weighted) Dynamic Range (Typical): 137 dB, 1 kHz at max SPL Signal-to-Noise Ratio: 89 dB, 1 kHz at 1 Pa Phantom Power Requirements: 48V DC, 3.8 mA typical Output Connector: Integral 3-pin XLRM-type Case Form Factor: Audio-Technica R10 Dimensions (Ø x L): 2.24 x 6.51" (57 x 165.3 mm) Weight: 20.5 oz (582 g)

Thermionic Culture is celebrating its 20th anniversary with the launch of a limited edition Thermionic Culture Vulture 20A at the 2018 NAMM Show. The 20A differs from the original in that it has much lower background noise using Sowter transformers to balance line inputs and outputs and also indented pots for easy recall. The sound is nearly identical to the original except the top end is extended to far beyond audio range making it a great mastering tool. It still has the classic Culture Vulture distortion settings with an extra extreme position on the function switch.Originally designed as a "distortion box" to simulate distortion in valve amps, the Thermionic Culture Vulture has found lots of uses beyond this. Some owners use them on drum loops, vocals, piano sounds and even across entire tracks (it is a stereo unit). Distortion figures are reduced to only 0.2% at lowest to about 99.9%. Predominant distortion can be changed from even to odd harmonics with a simple switch.Features:Warm sounds gently or create a noise like a 200 watt guitar stack with all the drivers slashedIndependent channel operationOdd or even harmonic distortion, or combination of bothAll valve design free from solid state additivesHigh impedance line input or instrument inputs4 & 7 kHz filtersOverdrive & bypass switches

Aνανεωμένη έκδοση του κλασικού πια Fat Bustard, με ενσωματωμένο INT/EXT διακόπτη στο master section και ένα ζευγάρι από αντίστοιχα jack inputs για σύνδεση ενός playback device που μπορεί να γίνει route στο monitor buss. Αυτό μπορεί χρησιμοποιηθεί και για να γίνει έλεγχος του σήματος ενός Phoenix compressor που έχει συνδεθεί στο output του Fat Bustard. Η στάθμη του σήματος EXT μπορεί να ρυθμιστεί με ένα trim pot.Το Fat Bustard είναι κάτι παραπάνω από ένα απλό mixer, παρέχοντας δυνατότητες διαμόρφωσης του ήχου με χειριστήρια bass και high cut και lift, καθώς και δυνατότητα ανοίγματος της στερεοφωνικής εικόνας. Οι είσοδοι είναι κατανεμημένες σε 4 stereo, 4 mono όλες με διακόπτη mute και οι mono με pan pots.Τεχνικά ΧαρακτηριστικάInput Impedance:8 kohm + (dependant on channel and setting), unbalancedOutput Impedance:600 ohms, unbalancedGain (ch. 1-8):0 at Attitude 1, + 15dB at Max AttitudeMaximum Output Level (MOL)+ 26 dBuDistortion : (at Att 1) 0.015% +8dBm output (at Att 3) 0.22% (at Att Max) 1+ % (dependent on how hard input valves are driven)Signal to noise (IEC weighted) 100dB below MOL (at Attitude 1)Frequency response +/-0.5 dB 9Hz to 30kHz (at Attitude 1)Max Bass Lift +11dB @ 50 HzMax Top Lift +12.5dB @ 10kHzAll audio connectors XLR wired pin 2 signal, 1&3 earthValve complement 2 x 5965, 2 x 6SN7, 1 x 12AU7

The Manley VOXBOX shatters all your preconceptions of what a mic preamp can do. It's a voice processor, but that's only the beginning. Besides the fully professional, great sounding mic preamp section, the VOXBOX also offers an innovative opto-compressor; Manley's special EQ section with the extended Pultec Mid Frequency Equalizer; and a de-esser and peak limiter section. You also have tight control over all your parameters, no matter what section your using. Top it all off with Manley's state-of-the-art design specifications and you've got an impressive piece of gear with great versatility. The VOXBOX is not only a vocal processor; it also works its magic on instruments, featuring great settings for drums, bass, guitars, keyboards, etc. The VOXBOX is easily one of the most comprehensive audio processors you can find and is highly respected in the world's audio community.Specifications:Stunning mic preamp, with opto-compressor, EQ with the extended Pultec Mid Frequency Equalizer, de-esser and peak limiter for vocal and instrument tracking applicationsManley Transformers w/nickel laminations in mu-metal cases2K ohm MIC INPUT Z w/High current 48V Phantom power built-inHi-Z (100K) Direct Instrument InputLINE & INSERT INPUTS (balanced XLR & 1/4 inch)PREAMP & EQ outputs LO-Z (50 ohm)Transformer balanced XLR outputTransformerless unbalanced 1/4 inch outputsSTEREO LINK for Compressor & De-esser/Limiter offers expandability if you get another VOXBOXSIDE CHAIN MONITOR for De-EsserLarge ILLUMINATED Sifam METER with FIVE readout modesTHD + N (1kHz @ +20 dBu): 0.3%Maximum output: +31dBuDe-Ess Notch Frequencies: 3, 6, 9, 12KHzPower Consumption: 24 wattsDimensions: 19 x 5.25 x 10 inches (occupies 3U) 

The Manley MASSIVE PASSIVE is a two channel, four band equalizer, with additional high pass and low pass filters. Passive - refers to the tone shaping part of the EQ design not using any active circuitry. Only metal film resistors, film capacitors and hand-wound inductors sculpt the sound. Super- beefy, hugely-high-headroom, Manley all-tube make-up gain amplifiers deliver your tunes into the next realm. Creating natural, organic, acoustic tone can only be done with an equalizer that shapes the signal with natural methods. The Massive Passive uses simple passive components and exploits their natural qualities rather than forcing a complex circuit to meet an arbitrary clinical or scientific goal. Manley knows that recording and mastering equalizers are used by artists for artistic goals and they balanced this design with a little more art than science. The Massive Passive is intended both for the most radical EQ sometimes needed for tracking as well as the most subtle shadings for vocals and mastering. It is designed to be a fundamentally different EQ but with the best strengths of Pultecs, choice console EQs, parametrics and graphics. The difference is that this EQ allows twice as much EQ with half the colouration. It allows massive HF boosts without sibilance problems and unbelievable fatness without mud. This is unique. Being different also gives it some quirks and idiosyncrasies that will spark your creativity.The main features of the Manley MASSIVE PASSIVE include:Perhaps the ultimate studio EQ for tracking instruments, mastering and delivering powerful tonal control with characterAll-Passive tone sculpting circuitryUnique Shelf curves use the "bandwidth" controlOverlapping and Interleaved Frequency choicesEvery band switchable to shelf or bellVacuum tube make-up gain and line driversParallel symmetrical topologyPremium components throughoutHP and LP Filters plus gain trimsEach band can be bypassed or set to boost or cut. This provides twice the control resolution over conventional EQs. It also eliminates the problem of a centre detent not guaranteeing flat responseEach band is capable of 20 dB of boost or cut combined with amazing headroom and freedom from clippingSteep LP and HP filters for maximum effectiveness and achieved passively for minimal coloration. The minimum slope is 18 dB/octave and the 18kHz filter was designed for 60 dB/octaveModular design that allows future upgrades and special functionsTransformer balanced floating outputs for ease of installation and a touch of desirable flavourIns and Outs Balanced XLR and 1/4 inch (accepts unbalanced)Level +4 dBm nominal, internal switches for -10 operationBypass Switch bypasses EQ and tube circuits (not hardwire)44 Frequencies (roughly 1/4 octave spacing)Frequency Range: 22 Hz to 27 kHzEQ Boost/Cut Range : 20 dB boost, 20 dB cutNominal Q range: 1.5 to 3 (uniquely active in shelf modesFrequency Response: +/- 2 dB: 8 Hz to 60 kHzMaximum Output @ 1.5% THD +37 dBv; + 26dBv @ 20 HzTHD and Noise (1kHz @ +4 dBm): 0.06%Noise Floor (referred to + 4dBm): -85 dB (A Weight)Dynamic Range: 120 dBTube Complement: 5751 x 2, 6414 x 4Power Consumption 72 wattsSize (3U): 19 x 5.25 x 10 inchesWeight: Unit 21 lbs. Shipping weight: 27 lbs

Introducing the MicroSub45.  Featuring Barefoot Sound’s innovative Dual-Force™ technology, the MicroSub45 brings the MicroMain45 to new power and depths.  The subwoofers boasts dual-opposing 8” subs in each of the two cabinets. The MicroMain45 has turned into a cult speaker that musicians love. A common request from users since the release of the MicroMain45 is for Barefoot Sound to build a companion subwoofer. So Barefoot has built a subwoofer as an off-the-menu item. But as demands have been overwhelming, it’s time to bring this product into the light.Introducing the MicroSub45 featuring Barefoot Sound’s innovative Dual-Force™ technology The MicroSub45 brings the MicroMain45 to new power and depths with a pair of dual-opposing 8” subs in each cabinet and Dual-Force™ Technology.The MicroMain45 is the only Barefoot that doesn’t feature Dual-Force™ Technology. But now there is a path for MicroMain45 users who want to take it to the next level. Or if you’re interested in a pair of MicroMain45s and want the option to upgrade down the road as your needs grow, this is the perfect solution!The MicroMain45s can be placed on top or below the MicroSub45, depending on your preference. The tilt feature allows the MM45 to be positioned down to the listening sweet spot and it is designed so that the stack is very sturdy and will not slip.The subwoofers is the MicroSub45s. When packaged with the MicroMain45s, it becomes the MicroStack45.  The stack features six high excursion 8” drivers, all designed to work seamlessly together and as a single unit. Each MicroSub45 features analog crossovers which allows analog signal paths into the sub and also into the MicroMain45. It crosses over at 80hz and goes down to 25hz.Features:Comes in a pairDual 8” subs in each cabinetDual-Force™ Technology1000 Watts of PowerBass response down to 25HzBalanced analog High-Pass filterSpecifications:Crossover Frequency: 80 HzSubwoofers: Dual 8″ aluminum cone with high linearity motor, +/- 13 mm linear excursionAmplifier: 500W in each cabinetAC Power Input: 85 VAC to 265 VACPower Consumption: Idle: 5W, Maximum: 765WWeight: Speaker: 45 lbs each (20.5 kg)Shipping: 54.5 lbs each (25 kg)Dimensions HxWxD: Cabinet: 14.0 x 9.5 x 13.0 inches (356 x 241 x 330 mm)Overall: 14.0 x 10.25 x 13.6 inches (356 x 260 x 345 mm)

A new version of the Phoenix soft knee stereo valve compressor, the Phoenix Standby: new features include a standby switch which is provided to extend valve and capacitor life. When in 'standby' the HT current through the valves is only 50% of the normal value. The zero level meter adjusters have now been front to the front panel to make calibration easier and to avoid having to take the unit out of the rack.The Phoenix is a twin channel 'vari-mu' device. Compression ratio starts low and increases gradually so that compression can be subtle and natural, but compression effects can occur when driven hard. Very low noise. Continuously variable controls for Gain, Attack, Release, Threshold and Output Level. Stereo link for compressing a group (eg. drums) or a complete mix.The Phoenix enhances vocals & individual instruments by being unobtrusive, gently 'warming' sounds and making solos sound more powerful. ItΉs impressive on overall mixes & can give a digital mix 'reality'. ItΉs also excellent for compression effects. Try drums with a 'thump' attack (i. e slowest) & fast release.Key Features All valve design for greater presence and naturality.Completely free from solid state additives.Transparent compression.Punchy when pushed.Drive it hard for compression effects and distortion.Absolutely flat frequency response over entire audio range and beyond.Minimal phase shift giving a crisp and warm sound.

The MicroMain27 Gen2 is a 3.5-way active system with 5 drive units housed in sealed enclosures spanning 30Hz to 45kHz with vanishingly low distortion, breathtaking dynamic range and ultra-fast transient response. The ring radiator tweeter is incredibly detailed and produces very wide dispersion out to its highest frequencies. The new 5.25″ midbass features a phalanx of advanced technology, yielding midrange detail that rivals any driver on the planet. Barefoot has teamed up with the brilliant Bruno Putzeys of Hypex to develop the powerful and completely transparent amplifier stage of the MM27 Gen2. Melding the system together is a groundbreaking new DSP crossover that has taken Thomas Barefoot more than 4 years to develop. With beautifully designed high end converts and cutting edge proprietary filtering techniques the MM27 Gen2 sound stage is seamlessly coherent and deeply revealing. MEME™ Technology: Despite the advantages of high resolution monitors, Thomas Barefoot realizes that many engineers still use their NS10M’s † and mix cubes as secondary references. These speakers have long traditions and people find them familiar and useful for focusing in on certain aspects of their mix. However, crowding one’s console with those extra boxes degrades the sound field of the primary reference monitors. Not to mention, they are no longer manufactured, they need amplifiers, cable runs, and they consume more studio space. The solution is to make the MM27 sound and translate like those speakers. With the turn of a knob one can switch from the MicroMain27′s brutally revealing “Flat” response to the warmer and sweeter “Hi-Fi” setting, generically emulating the sound of some high end consumer audio gear. More specifically the “Old School” setting closely emulates the sound of the NS10M nearfield, while the “Cube” setting emulates the mid-centric sound of classic mix cubes The idea is not to perfectly replicate every subtle quirk of these venerable old speakers. That would be impossible. But the MM27 Gen2 can capture the essence of how they behave and translate, modeling their frequency, phase and transient responses, dynamic compression and even specific distortion components. If you know how to work on these speakers you will feel very comfortable working on the new MM27′s emulation settings.  MicroMain27 Gen2 features: 3.5-way active studio monitor with integral subwoofers Four-setting DSP voice emulation Analog and AES3 digital inputs Hypex amplification Tweeter - 1" ring radiator with Advanced neodymium motor Rear waveguide chamber. Amplifier: 250W Hypex Midbass 1 - 5.25" poly/paper cone with Advanced neodymium motor , with +/- 5 mm linear excursion. Amplifier: 250W Hypex Midbass 2 - 5.25" poly/paper cone with Advanced neodymium motor, with +/- 5 mm linear excursion. Amplifier: 250W Hypex Subwoofers - 2 x 10" aluminum cone with high linearity motor, with +/- 13 mm linear excursion. Amplifier: 500W Hypex Crossover Frequencies: 100 / 600 / 3000 Hz Cabinet Dimensions: 20.5" x 9.5" x 15.5" Overall Dimensions: 20.5" x 10.4" x. 17.4"

Whether you’re tracking, mixing or mastering, the Convert-8 will deliver powerful low-end, articulate mids and transcendent highs to every channel in your analog rig. Patched into one of our award-winning analog summing units like the 2-BUS+ or the D-BOX – or to your favorite analog console – the Convert-8 will empower you to create the spacious, three-dimensional soundscapes that only real analog summing can deliver.Compatible with all of today’s digital signals (including the ubiquitous ADAT format) the Convert-8 is ready to take on any mixing or monitoring job from standard stereo and 2.1 to 5.1 and the increasingly popular 7.1 surround format. Recognizing the needs of those who don’t need more analog inputs, the Convert-8 brings today’s very best digital-to-analog conversion within reach of any studio.Our design philosophy includes a strict demand that our equipment be elegant and easy to use, and the Convert-8 exemplifies that philosophy. No menus, no cryptic multi-finger combinations to remember – just clearly labeled single-function buttons.With industry-standard reference levels of -14, -16 and -18dBFS, you can recalibrate the Convert-8 with the push of a button. While many professional converters are only aligned at the factory, the Convert lets you move flexibly between different reference levels on the fly. Anyone who’s ever fumbled around behind a rack with a “tweeker tool” while a second person watches an external meter knows the special value of our front-panel calibration buttons, especially when there’s eight channels or more!Specifications:Signal to Noise Ratio A-weighted, 20Hz to 20KHz: < 114dBSignal to Noise Ratio unweighted, 20Hz to 20KHz: < 113dBDynamic Range A-weighted, 20Hz to 20KHz: < 114dBDynamic Range unweighted, 20Hz to 20KHz: < 113dBJitter: 16ps (100Hz to 40KHz), 18ps (100Hz to 1MHz)Crosstalk rejection: > 114dBu  1kHzReplacement Fuses: USA 2 amp slo-blow for 120V Europe 1 amp slo-blow for 240VTHD+N:THD+N, 1kHz, unweighted, 20Hz to 20kHz, +4dBu out: < 94.5dB (0.00188%)THD+N, 1kHz, unweighted, 20Hz to 20kHz, +22dBu out: < 106.5dB (0.00048%)Frequency Response  96KHz sample rate:DC to 20KHz: +0, -0.25dBDC to 30KHz: +0, -0.5dBDC to 40KHz: +0, -1.0dB