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Χρωματικό κουρδιστήρι υψηλής ακρίβειας. Χρησιμοποιεί ηλιακές κυψέλες για να φορτίζει τη μπαταρία (φορτίζει και σε φωτισμό δωματίου). Έχει ιδιαίτερα ανθεκτικό περίβλημα από σιλικόνη και διαθέτει ειδικό ''κλίπ'' για εύκολη μεταφορά. Το TC-1S, διαθέτει ενσωματωμένο μικρόφωνο καθώς και είσοδο για ηλεκτρική κιθάρα / μπάσο, καθώς και επίσης σύνδεση USB, για να φορτίζει τη μπαταρία, όταν δεν υπάρχει φώς.Τεχνικά χαρακτηριστικά του TC-1S Κούρδισμα έως 12 νότες Εύρος αναγνώρισης A0-C8 (27.5 - 4186.010006 Hz) Εύρος κουρδίσματος 437-446 Hz, με βαθμίδες του 1Hz Ακρίβεια αναγνώρισης 1 εκατοστό (A = 440 Hz) Γενικά χαρακτηριστικά του TC-1S Ενσωματωμένο μικρόφωνο, mono, παντοκατευθυντικό 6.3mm μονοφωνική είσοδος με υποδοχή για όργανο ή μικρόφωνο. Επαναφορτιζόμενη μπαταρία μαγγανίου - λιθίου η οποία φορτίζεται είτε μέσω του ενσωματωμένου ηλιακού panel, είτε μέσω USB σύνδεσης Υποδοχή USB τύπου Mini-B (μόνο για φόρτιση της μπαταρίας) Διάρκεια χρήσης της μπαταρίας : περίπου 6 ώρες (η ακριβής διάρκεια εξαρτάται από τις συνθήκες χρήσης) Εξωτερικές διαστάσεις : 96mm Χ 18mm X 40mm (εξαιρώντας την προστατευτική σιλικόνη) Βάρος : 51 γραμμάρια Θερμοκρασίες χρήσης : από 5 έως 35 βαθμούς C. Ενσωματωμένα αξεσουάρ : οδηγίες χρήσης, λουράκι με "κλιπ"
A suite of no less than three ingenious processors using advanced technology, developed during the last decades, by the sound analysis/synthesis team at IRCAM. These precision tools provide truly innovative signal transformation algorithms allowing the user to alter sound characteristics such as; pitch, timbre and duration, using a highly intuitive user interface.Transformer - Real-time Voice and Sonic Modelling ProcessorTRAX Transformer v3 is based on an augmented phase vocoder technology and a cutting edge transformation algorithm, allowing for manipulating characteristic properties of a voice such as gender, age and breath, and on any other sound; expression, formant and pitch.Cross Synthesis - Spectral Envelope Morphing ProcessorTRAX Cross Synthesis v3 utilizes a phase vocoder (the amplitude and the frequency/phase spectra), to morph the spectral characteristics of two sounds. The amplitude and frequency/phase spectra can be blended continuously and since the features that are used here are strongly nonlinear, the sound morphing is nonlinear as well, offering a way to create a wide range of new exciting and unusual sound effects.Source Filter - Enhanced Sound FilteringTRAX Source Filter v3 is based on a signal model decomposing the signal into a time envelope describing the energy/loudness contour of the sound, and a spectral envelope describing the spectral colour of the sound timbre. The energy contour and the spectral colour of the source sound, can be continuously blended with the spectral colour and energy contour of an arbitrary filtrating sound, allowing transformation that extend well beyond the more common source filter morphing effect.Specifications:Availability The Ircam Trax v3 plug-ins are available in the following configurations:Native - AU / VST / AAX NativeProcessing Specifications The TRAX v3 plug-ins, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix DSP based).Sampling rate up to 384 kHz for Native (AU/VST/AAX).Compatibility Windows - Vista, 7 and 8 all in both 32 and 64 bits*. (XP v.2.4 / 32 bit only)VST (2.4)AAX NativeMac OS X (Intel) - 10.7, 10.8 and 10.9 in both 32 and 64 bits.VST (2.4)AUAAX Native
Room Acoustics and ReverberationVERB is an algorithmic room acoustics and reverberation processor. It has a modular construction, employing a recursive filtering reverb engine, reproducing and synthesizing the specific acoustical characteristics of any spatial sound environment.Following a decade of intense research by the Acoustic and Cognitive Spaces Team at IRCAM, the IRCAM VERB introduce state of the art techniques for room acoustics simulation utilising advanced perceptive models, concealing the complexity behind the actual reverb algorithms and allowing for a flexible and intuitive experience for the user. The IRCAM VERB provides eight input and output channels, presenting the option for reverberation processing in multi-channel and surround formats. A built in Input/Output (I/O) routing matrix provides instant flexibility when setting up the I/O in relation to the physical audio monitoring in the control room.Signal Processing Features Input/Output gain controls for adjusting the levels before and after processing.Dry Mix control (Dry/Wet with gain compensation) allowing for parallel processing or for blending some of the original "peak" into the processed signal.Preset/Parameter slotsTo enhance the workflow the two Preset/Parameter slots, A and B, can be loaded with two full set of parameters at the same time. Apart from saving each preset, a "Global Preset" containing both the A and B settings, and the position of the "Morphing Slider", can be saved.Parameter Morphing Slider with AutomationThe Morphing Slider provides morphing between the parameter settings of slot A/B allowing for really creative and useful real-time tweaking. Enabling the Automation control button exposes the Morphing Slider to the host automation.Processing Specifications Verb, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix DSP based).Sampling rate up to 384 kHz for native (AU/RTAS/VST). Compatibility Windows - XP, Vista, 7 all in both 32 and 64 bits.VST (2.4)RTAS*Mac OS X - 10.4, 10.5, 10.6 in 32 bits.VST (2.4)AURTAS**The RTAS version requires ProTools 7 or later.
Flux / IRCAM next generation audio tools: IRCAM SPAT - The complete Room Acoustics Simulation and Localisation Solution With more than a decade of research performed by the Acoustic and Cognitive Spaces Team at IRCAM, being at the forefront of scientific and technological innovations, the SPAT is the most advanced and sophisticated tool for room acoustics simulation and localisation ever designed, managing both spatialisation (source localisation) and room acoustic simulation in a truly consistent and visually logical way. SPAT introduce state of the art techniques for room acoustics simulation utilizing advanced perceptive models, concealing the complexity behind the actual algorithms, allowing for intuitive and accommodating user interaction capabilities. Designed for surround and multi-channel use, SPAT presents the option to setup the output arrangements providing a variety of stereo and surround configurations, including subwoofer configuration. With eight input and output channels available in SPAT, configurations up to 7.1 and 8.0 are feasible. IRCAM VERB - Room Acoustics and Reverberation VERB is an algorithmic room acoustics and reverberation processor. It has a modular construction, employing a recursive filtering reverb engine, reproducing and synthesizing the specific acoustical characteristics of any spatial sound environment. Following a decade of intense research by the Acoustic and Cognitive Spaces Team at IRCAM, the IRCAM VERB introduce state of the art techniques for room acoustics simulation utilising advanced perceptive models, concealing the complexity behind the actual reverb algorithms and allowing for a flexible and intuitive experience for the user. The IRCAM VERB provides eight input and output channels, presenting the option for reverberation processing in multi-channel and surround formats. A built in Input/Output (I/O) routing matrix provides instant flexibility when setting up the I/O in relation to the physical audio monitoring in the control room. Transformer - Real-time Voice and Sonic Modelling Processor TRAX Transformer is based on an augmented phase vocoder technology, and a cutting edge transformation algorithm, allowing for manipulating characteristic properties of a voice such as gender, age and breath, and on any other sound; expression, formant and pitch. These voice transformation algorithms have already been used, with great success, in international and French cinema productions. (Farinelli, Vatel, Tirιsia, Les amours d'Astrιe and Cιladon). Cross Synthesis - Spectral Envelope Morphing Processor TRAX Cross Synthesis utilizes a phase vocoder (the amplitude and the frequency/phase spectra), to morph the spectral characteristics of two sounds. The amplitude and frequency/phase spectra can be blended continuously and since the features that are used here are strongly nonlinear, the sound morphing is nonlinear as well, offering a way to create a wide range of new exciting and unusual sound effects. Source Filter - Enhanced Sound Filtering Processor TRAX Source Filter is based on a signal model decomposing the signal into a time envelope describing the energy/loudness contour of the sound, and a spectral envelope describing the spectral colour of the sound timbre. The energy contour and the spectral colour of the source sound, can be continuously blended with the spectral colour and energy contour of an arbitrary filtrating sound, allowing transformation that extend well beyond the more common source filter morphing effect.
Be it a serious problem with the dynamics in a mix, a delicate mastering assignment, or a situation with complicated dynamics, Alchemist provides everything you need in order to accomplish. Designed with professional mastering and re-mastering applications in mind, the intricate design, superior workflow, and sound perfection, makes Alchemist tailored for the high demands of today’s music and pro-audio industry.ProcessingAlchemist operates as a single broadband processor or as a full-scale multiband processor offering up to five individual bands. Each band presents a complete dynamic processing section including; compressor, expander, de-compressor and de-expander, all working in parallel employing independent Angel’s share and Hysteresis parameters for each section.Angel's share and Hysteresis are key features of all Flux:: dynamic processors, using the dynamic range content of the signal and not just the signal levels as standard processors do. Angel’s share controls the amount of auto-ratio determined by the signal dynamics and by the manual ratio setting. Hysteresis controls the amount of auto-threshold determined by the signal dynamics and by the manual threshold value.Specifications:Availability Alchemist v3 is available in the following configurations:Native - AU / VST / AAX NativeMerging VS3 MassCore/NativeProcessing Specifications Alchemist v3, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix & Ovation MassCore/Native)Sampling rate up to 384 kHz for Native (AU/VST/AAX).Compatibility Windows - XP, Vista, 7, 8 and 8,1 all in both 32 and 64 bits.VST (2.4)AAX NativeMerging VS3 (Only in Windows 7 32/64 bits)Mac OS X (Intel) - 10.7, 10.8 and 10.9 in both 32 and 64 bits.VST (2.4)AUAAX Native
Where sound becomes music! The Epure II is a five-band equalizer with an algorithm designed to deliver the most superior quality achievable in digital equalizing processing today. The logical and comprehensive user interface includes a range of quick "shortcut" functions to enhance and simplify the user workflow, allowing for ultra fast and precise operation.For surround/multichannel operation up to eight channels of I/O, four MS decoders/encoders and internal processing groups are provided, all routable using the built in routing matrix.A powerful go-to processor for your day-to-day session work, and a sharp-edged surgical precision tool for the most demanding equalizingtasks conceivable TDM/Venue version available.For the full feature set, hover your mouse over the small images to the right above. Epure II Band Features Each of the five band-sections providesFilter type selectionlow cutlow shelvingpeakhi shelvinghi cutFrequency range covering the whole bandwidthQ factor adjustable (from 1.00 to 100.00)Gain control (ranged -24 dB to +24 dB)BypassEpure II User Interface Features To match the perfection of the sound quality of the Epure II, we have provided a range of user interface functions to improve the workflow.Multiply gain (x2) affecting the gain value of the band-section.Divide gain (/2) affecting the gain value of the band-section.Invert, converting the band-section positive values to negative and vice versa. Signal Processing Features Master gain control.Bypass routes the incoming signal direct to the output for a true smooth transition between processed and clean signal. Preset and Parameter Handling Preset/Parameter slotsTo enhance the workflow the two Preset/Parameter slots, A and B, can be loaded with two full set of parameters at the same time. Apart from saving each preset, a "Global Preset" containing both the A and B settings, and the position of the "Morphing Slider", can be saved. Parameter Morphing Slider with AutomationThe Morphing Slider provides morphing between the parameter settings of slot A/B allowing for really creative and useful real-time tweaking. Enabling the Automation control button exposes the Morphing Slider to the host automation.Processing Specifications Epure II, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix DSP based).Sampling rate up to 384 kHz for native (AU/RTAS/VST).Compatibility All major native formats and TDM (Venue D-Show compatible)* are supported.Windows - XP, Vista, 7 all in both 32 and 64 bits.VST (2.4)RTAS*Mac OS X - 10.4, 10.5, 10.6 in 32 bits.VST (2.4)AURTAS**The TDM/RTAS version requires ProTools 7 or later.
The complete dynamic processing solution for superior sonic quality and distortion free operation within all fields of demanding audio processing. Primarily designed for mastering and re-mastering applications, making it a really versatile workhorse in the studio.Four dynamic sections work in parallel, compressor, de-compressor, expander and de-expander. Three release modes (manual, auto, and advanced) are available. In advanced mode the minimum and maximum release values are user adjustable. An Auto Delay function allows zero attack times by introducing a delay line in the processed signal. The side chain features a three-band equalizer to generate frequency sensitive processing. Angel's Share is the parameter controlling the amount of auto-ratio determined by the signal dynamics and the manual ratio setting. Hysteresis is the parameter controlling the amount of auto-threshold determined by the signal dynamics and the manual threshold value. Angel's Share and Hysteresis are key features of all Flux:: dynamic processors because they make use of the dynamic range content of the signal, and not just the signal levels as standard processors do.Many new functions have been added to the original Solera. External side chain inputs are now available. New dynamic detection algorithms are available. The Hysteresis and Angel's Share parameters are now independent for each dynamic section. Each dynamic section now has a Hold parameter and a Threshold for Hysteresis. Classic modes and Feedback modes are now available for signal level detection, completing the legacy Solera mode. A Maximum mode can be engaged on the Hysteresis control, allowing processing that takes into account maximum values of both the standard and the Hysteresis detection schemes.Our exclusive A / B morphing system, allows instant and efficient global control of processing. The preset manager allows you to save global presets made up of two different presets including the A / B morphing slider position. Note also that factory presets are provided for every plug-in. Solera II manages digital audio up to 384 KHz, and up to 8 channels. This is the native version of the plug-in for AU, RTAS and VST.The DSP based Pyramix version is available from the Merging Technologies sale network.For the full feature set, hover your mouse over the small images to the right above. Signal Processing Features Input/Output gain controls for adjusting the levels before and after processing.Phase invert of the processed signal.Dry Mix control (Dry/Wet with gain compensation) allowing for parallel processing or for blending some of the original "peak" into the processed signal.Clipper, a discrete peak limiter at the very last stage of the processing chain rounding off the peaks smooth and musically.Bypass routes the incoming signal direct to the output for a true smooth transition between processed and clean signal.Preset and Parameter Handling Preset/Parameter slotsTo enhance the workflow the two Preset/Parameter slots, A and B, can be loaded with two full set of parameters at the same time. Apart from saving each preset, a "Global Preset" containing both the A and B settings, and the position of the "Morphing Slider", can be saved. Parameter Morphing Slider with AutomationThe Morphing Slider provides morphing between the parameter settings of slot A/B allowing for really creative and useful real-time tweaking. Enabling the Automation control button exposes the Morphing Slider to the host automation. Processing Specifications Solera II, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix DSP based).Sampling rate up to 384 kHz for native (AU/RTAS/VST). Compatibility All major native formats are supported Windows - XP, Vista, 7 all in both 32 and 64 bits.VST (2.4)RTAS*Mac OS X - 10.4, 10.5, 10.6 in 32 bits.VST (2.4)AURTAS**The RTAS version requires ProTools 7 or later.
When we started sketching our new plug-in project, our aim was to create a versatile and truly musical dynamics processor, which handles the dynamics in a way that allows you to be creative, without a complicated user interface.The result is Syrah, a new generation dynamics processor utilizing real time dynamic detection and level dependent processing, providing adaptive dynamic capabilities, which mean that Syrah is always trying to adapt to the music and to the beat of the material.Using parts of our exquisite ‘BitterSweet’ technology, our new adaptive-dynamics technology, and our well-recognized level independent dynamics processing, Syrah will be well suited as a creative versatile processor for recording and mixing, as well as for delicate and demanding mastering tasks.User Interface and WorkflowAs you may notice, the controls are not the usual suspects found on a dynamics processor, instead, the controls provided typically affect more than one parameter in the underlying algorithms, with everything carefully tweaked allowing for creative processing still ensuring the finest sound achievable.The built in preset manager and the preset morphing slider, provides instant and intuitive control of all parameters and controls. In a second, with a simple one-click operation, everything is copied from one of the two preset slots to the other. even during playback. Except for only A/B comparing two set of parameters, the morphing slider will allow for mixing them, and if desired, record the morph with the host automation.With the built in preset manager, you can save a complete snapshot (called a Global Preset) with all the settings from both of the preset slots, as well as the position of the preset morphing slider, allowing for instant recall of your morphing set.Specifications:Availability Syrah v3 is available in the following configurations:Native - AU / VST / AAX Native+AAX DSP - AU / VST / AAX Native / AAX DSPMerging VS3 MassCore/NativeProcessing Specifications Syrah v3, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.Sampling rate up to 384 kHz DXD (Pyramix & Ovation MassCore/Native)Sampling rate up to 384 kHz for native (AU/VST/AAX).Compatibility Windows - XP, Vista, 7, 8 and 8,1 all in both 32 and 64 bits.VST (2.4)AAX Native/DSPMerging VS3 (Only in Windows 7 32/64 bits)Mac OS X (Intel) - 10.7, 10.8 and 10.9 in both 32 and 64 bits.VST (2.4)AUAAX Native/DSP
The U5 high-voltage DI-preamp combines a unique passive tone selector with a variable gain preamp and filter. The high input impedance accepts a wide variety of signal levels and instruments from acoustic guitars to high-output active bass guitars and keyboards. The U5 direct box is loaded with sonic character and user features professionals demand. FEATURESPure Class A, 100% discrete design Smooth musical detail and sonic excellence Variable gain preamp to +30dB. Dual microphone and line outputs Hi-Z 3,000,000 ohm input impedance Very low noise -100dB High headroom +30dB Six tone bank selections. DC coupled Class A output for superior bass High cut switch eliminates noise LED active signal indicator Headphone monitor output jack Low distortion less than 0.5% THD and IMD High level 400 watt speaker input Active to thru selector - ground isolation switch 100% discrete power supplies for audio path Internal toroidal power supply for low noise Long lasting, stainless steel hardware
Τype: MajorTone: AChromatic: No
The cardioid e 935 is a fully professional vocal microphone, developed to cut through high on-stage levels.Pick-up pattern : CardioidFrequency response (microphone) : 40.....18000 HzSensitivity in free field, no load : (1kHz) 2,8mV/Pa bei 1 kHz = -51dB (0 dB = 1V/Pa) = -71 dB (0 dB = 1V/ubar)Nominal impedance : 350 OhmMin. terminating impedance : 1000 OhmDimensions : 47 x 181 mmWeight : 330 gPolar Diagramm : Frequency Response :
The e 902 is a dynamic cardioid instrument microphone, which was designed for lowest bass signals with very high sound pressure levels.It is suitable for kick drums, bass guitar-amps, tuba, and other bass instruments. Pick-up pattern : cardioidFrequency response : 20...18000 HzSensitivity in free field, no load : (1kHz) 0,2 mV/Pa; (@ 60 Hz): 0,6 mV/PaNominal impedance : 350 OhmMin. terminating impedance : 1000 OhmDimensions : 128,5 x 60 mmWeight : 440 gFrequency Response :
The supercardioid e 906 instrument microphone was especially designed for guitar amplifiers, but it is also an excellent choice for percussion and brass instruments. Pick-up pattern : supercardioid Frequency response : 40...18000 Hz Sensitivity in free field, no load : (1kHz) 2,2 mV/Pa Nominal impedance : 350 Ohm Min. terminating impedance : 1000 Ohm Dimensions : 55 x 34 x 134 mm Weight : 140 g Polar Diagramm : Frequency Response :
The e 914 is a high grade condenser microphone for very demanding applications. Its outstanding sound properties qualify the e 914 for highly sophisticated tasks.The e 914 is a perfect microphone for ambitious recordings and live performances.Its main areas of application are acoustic guitars, cymbals, percussion, overhead, orchestras, grand pianos etc Pick-up pattern : cardioid Frequency response : 20...20 000 Hz Sensitivity in free field, no load : (1kHz) 7 mV/Pa Equivalent noise level : 19 dB(A) Attenuation : 0, -10, -20 dB Max. Sound pressure level (aktiv) : 137, 147, 157 dB Nominal impedance : 100 Ohm Min. terminating impedance : 1000 Ohm Dimensions : 24 x 157 mm Weight : 198 g Polar Diagramm : Frequency Response :
By virtue of its design and features, the MD 431 II ranks amongst SennheiserΆs most exceptional microphones and is suitable for vocal, speech and broadcasting applications. Colour: black, sound inlet basket: refined steel. Pick-up pattern : super-cardioid Sensitivity in free field, no load : (1kHz) 2,0 mV/Pa +- 3 dB Nominal impedance : 250 Ohm Min. terminating impedance : 1000 Ohm Dimensions : d 49 x 200 mm, Griff d 31 mm Weight :ca. 250 g
The e 865 is a fully professional super-cardioid vocal microphone utilising the very best in condenser microphone technology to achieve new standards of quality and sound. The microphone capsule insorporates special blast/pop protiction plate to shield the diaphragm from excessive sound pressure levels. Pick-up pattern : super cardioid Sensitivity in free field, no load : (1kHz) 3 mV/Pa Max. Sound pressure level (aktiv) : 150 dB Nominal impedance : 200 Ohm Min. terminating impedance : 1000 Ohm Dimensions : d 47 x 193 mm Weight : w/o cable 311 g Frequency response (microphone) : 40.....20000 Hz Phantom powering : 12 - 48 V
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Straight microphone stand with tripod leg base and add-on telescopic boom. Minimum Height : 960mm - Maximum Height :1500mm. Boom Lenght : 750mm. Weight :2,65 Kg
Universal Audio 4-710d Twin-Finity, four-channel microphone/line preamplifier, Dual-path 285-volt Class-A tube and transimpedance solid-state preamps, Phase-aligned tone-blending of tube and solid state circuits, selectable 1176-style compression circuitry on each channel, four additional line inputs feeding eight channels of pristine analog-to-digital conversion. 24-bit A/D conversionsample rates up to 192 kHz48 Phantom Power per Channel enableDigital output via dual ADAT optical and AES/EBU DB-25 connectors8-channel soft limiter (switchable for all channels)20Hz-100kHz +/- 0.2dBu, max. Gain +70dBu4x Mic-In XLR, 4x Line-In XLR4x Line-In TR 1/4", 4x Hi-Z In4x Send/Return (1 per Mic-Channel)4x Line-Out XLR, 19" 2HE
The Retro Sta-Level is a replica of the legendary 1956 Gates Sta-Level. The Sta-Level dominated the sound of hit radio in the 1960's. Now these super musical sounding compressors have found their way into today's hits. The Retro Gold Edition celebrates fifty years of the products existence. The Sta-Level uses the coveted 6386 tube in a classic G.E. circuit design. Alternatively, the Gold Edition Sta-Level uses a pair of commonly available 6BJ6 tubes (stock). The customer has the choice to use their own 6386 they provide or the 6BJ6 provided with the purchase of the unit. The choice is yours and the sound is undeniably true to the original either way. Specifications Gain Reduction Threshold: Continuously adjustable from +24dBm to -20dBm Available Gain Reduction: 40dB Total System Gain: 35dB (with standard input and output pads) Distortion: 1 percent or less from 0-30 dB gain reduction Frequency Response: ± .25dB from 20-20,000 Hz Noise: -70dB or better referenced to gain reduction threshold Output Level: Continuously adjustable from -40 dBm to +18 dBm Input Impedance: Accepts low impedance active or transformer balanced 0-600 Ohms Output Impedance: 600 Ohm transformer balanced, terminated and resistive pad coupled Tube Complement: 2-6BJ6), 1-12AT7, 2-6V6GT, 1-6AL5, 1-OB2, 1-5Y3 Power Requirements: Three prong IEC - Switchable 115-230 VAC - 50/60 Hz Mounting Requirements: 3U metal rack 5.25" high, 19" wide, 8" deep
Vintage Tube Amplifiers and Custom-Designed Transformers set the soundstage. True Passive Equalization creates Bountiful and Musical Equalization curves. The new RETRO 2A3 is designed as an ultra High-Fidelity Program Equalizer whose sonic benefits go beyond the equalization process. Accurate to the industry-standard Pultec EQ performance, the RETRO 2A3 adds HF boost frequencies in several new sweet spots. Carefully implemented HF boost circuits capture the signature sound of the original passive equalizers. Use these settings to make individual tracks fit in place. Make a vocal shine and add distinct presence. There are many satisfying applications, most notably the stereo mix buss. A new Subsonic Filter allows you to do peaking low frequency boosts that tame the excessive subsonic energy exhibited in the original design if thatΆs what you want. Utilizing the interstage transformer, the filter not only reduces the subsonic energy but also provides distinct transformer tonal characteristics for tracking and mixing. The filter has settings of 40 and 90 Hz with a peaking response and sharp cutoff. The 40 Hz setting has no apparent loss of lows as it adds excitement to the 35-40 Hz region that is golden in many listening environments. The smoothing effect is pleasantly apparent throughout the midrange. It is easily switched-out for applications that require tight, accurate low frequency response. In consideration of the limited real estate in your rack, the RETRO 2A3 packs two EQ channels into one 2U space with surprising freedom of movement, ease of adjustment and familiarity. Use the two channels for stereo, separately on independent tracks or cascaded for extended equalization possibilities. The channel separation exceeds 70 dB. The RETRO 2A3 passive equalizer circuits do not rely on amplifier feedback, which can cause a harsh and clinical sound in typical active equalizer designs. By incorporating pure passive Capacitor - Inductor based equalization; the RETRO 2A3 sounds so natural and effortless even with extreme boosts and cuts. We also gave special attention to the Pultec-style bass boost/bass cut method that is essential to a powerful bass and kick drum EQ. This involves simultaneous low frequency boosts and lower-mid cuts that can scoop out the mud and add real punch. Quite simply, there are benefits to having the RETRO 2A3 in line without even equalizing, as the tube amplification has itΆs own unique musicality. The RETRO 2A3 Dual Channel Tube Program Equalizer provides a very useful palette of colors and textures ready for tracking, mixing and mastering. Industry-standard Passive EQ with authentic feel, controls and equalization curves Additional HF Boost Frequencies for more precise control Easily recallable 100-position knob scales on boost and cut controls A ganged interstage-coupled Subsonic High-Pass Filter provides Peaking LF Boosts with alternate tonal characteristics Transformer Balanced and fully-floating 600 Ohm Input and Output to eliminate ground-induced hum and noise LF Boost Settings of 20, 30, 60 and 100 Hz Complementary LF Cut settings of 20, 30, 60 and 100 Hz HF Boost Settings of 1.5 kHz, 3 kHz, 4 kHz, 5 kHz, 6 kHz, 8 kHz, 10 kHz, 12 kHz, 14 kHz and 16 kHz Complementary HF Cut Settings of 5 kHz, 10 kHz and 20 kHz Vintage Class AB tube amplification for unity-gain make-up Equalization Bypass Switches Size 2U Height: 19" wide x 3.5" tall x 9" deep XLR input and output connections IEC power connector 115/230V 50-60 Hz Switchable AC Mains Uses readily available 12AX7(2) and 12AU7(2) electron tubes Tube substitutions are possible with internal unity gain calibration and jumper selectable 6 or 12 volt tube configuration Design minimizes tube heat generation High stability components used throughout Steel Chassis built to last Manufactured in the USA Full factory support Specifications Signal to noise ratio of greater than 76dB Channel Separation greater than 70 dB Frequency response is flat within 1 dB from 20 Hz-20,000 Hz Harmonic Distortion is less than 1% from 20 Hz-20,000 Hz
The Switchcraft SC600 Dual Adapter Box combines many of the most commonly used audio adapters into one, easy to use device. Each side of the box functions separately, and features a 3.5mm (1/8") stereo jack, a 1/4" stereo jack, two 1/4" mono jacks, and two RCA jacks.Any of the connections on the SC600 can be used as inputs or outputs, enabling you to adapt audio from one device to another. The SC600 can also be used as a passive splitter to send audio from one device to up to three devices with different connections. For example, plug the audio output of your laptop into the 3.5mm (1/8") jack, then access the stereo signal from the two 1/4" mono jacks and the two RCA jacks. This would enable you to run audio from your laptop into a mixing console or house sound system and a cd recorder at the same time.
The new Aurora line of digital audio converters from Lynx Studio Technology offers unprecedented audio quality and control in single-rack space eight and sixteen channel models. The Aurora 8™ and Aurora 16™ feature 192 kHz analog to digital and digital to analog conversion with front panel control of all routing and sample rate options.Extended functions in Aurora are accessible via computer with the Lynx AES16 or by infrared using compatible laptops and handheld Pocket PCs. The rear panel has Clock and MIDI In and Out connectors and an LSlot bay, for the optional use of ADAT and other audio interfaces.Simultaneous 16 Channel Analog I/O and 16 Channel AES/EBU I/O24 Bit / 192 kHz Mastering Quality A/D and D/A conversion192 kHz AES/EBU I/O Supporting Single and Dual Wire ModesSingle Rack Space ConfigurationExtensive Remote Control Capability via Lynx AES16, IrDA and MIDILSlot™ Expansion Slot for FireWire®, ADAT®, ProTools|HD® and Future Interface OptionsOn-board Digital Mixer Provides Flexible I/O RoutingWord clock I/O with Lynx SynchroLock™ Jitter Reduction Technology
Καλάμι boom 0,84 έως 3m. 850gr με standard πάσσο 3/8
The Bricasti Model 7 is a modern high resolution digital design, utilizing a stunning array of the latest DSP processors, provides a platform for the long overdue next step in reverb processing algorithms. A separate fully differential analog section and dedicated transformer based linear power supply provide the finest analog specifications of any product of its kind. The Bricasti Model 7 features an exceptionally strong stainless steel chassis, and a tooled aluminum front panel, combine with a classic high visibility display and straight forward human interface, to complete an enduring design that is intended to fulfill its role, now and into the future. Each design element of the Model 7 is a carefully considered statement of our vision of what the evolution of reverberation processing in its most classic form can be. With a deep appreciation of the best designs which precede it, and a passion for moving the science of reverberation forward, the Model 7 provides a palette of sounds that encompass the familiar as well as new expressions in the art. Listen to the new reference in reverb processing; it will bring new life to your art, in a way unimagined by any process before it.
Aluminium boom pole precision machined 3m, 850gr
Lipinski Powered Stands feature the new L-401 power amplifier built into the stand rather than the speaker, enabling short speaker cable length while allowing easier amplifier setup, especially in a surround environment.Cut for the total height of 30" (75 cm) and 36" (90 cm) to drive treble straight to the discerning ear of a listener at his/her sitting position. Just as all Lipinski Sound products, the stands are stable, rigid, and of fine quality.
Lipinski Powered Stands feature the new L-301 power amplifier built into the stand rather than the speaker, enabling short speaker cable length while allowing easier amplifier setup, especially in a surround environment.Cut for the total height of 30" (75 cm) and 36" (90 cm) to drive treble straight to the discerning ear of a listener at his/her sitting position. Just as all Lipinski Sound products, the stands are stable, rigid, and of fine quality.
Lipinski Powered Stands feature the new L-401 power amplifier built into the stand rather than the speaker, enabling short speaker cable length while allowing easier amplifier setup, especially in a surround environment.Cut for the total height of 30" (75 cm) and 36" (90 cm) to drive treble straight to the discerning ear of a listener at his/her sitting position. Just as all Lipinski Sound products, the stands are stable, rigid, and of fine quality.
Lipinski Powered Stands feature the new L-401 power amplifier built into the stand rather than the speaker, enabling short speaker cable length while allowing easier amplifier setup, especially in a surround environment.Cut for the total height of 30" (75 cm) and 36" (90 cm) to drive treble straight to the discerning ear of a listener at his/her sitting position. Just as all Lipinski Sound products, the stands are stable, rigid, and of fine quality.
TASCAM's HS-8 is the first solution for professional multi-track recording and playback to solid-state media. From studio surround recording to post production playback of HS-P82 location recordings, the HS-8 fits a variety of multitrack roles first pioneered by TASCAM's DA-88 recorders. Also true to TASCAM tradition, audio is top-quality throughout with up to 192kHz/24-bit recording. Like the HS-P82, the HS-8 uses a color touch-screen interface to access settings and tracks. A mixer is built in for monitoring, and the stereo mix can be recorded live as a separate track. Audio is recorded as Broadcast WAV files to Compact Flash media, with SMPTE timecode available on various interfaces. A pair of CF card slots is available for data mirroring or continuous recording. The RC-HS20PD remote control allows operation from a separate room over RJ-45 connection. TASCAM's HS-8 is the perfect DTRS successor, with the instant access, ease-of-use and sound quality studio and post professionals demand. Features: 8-channel solid-state audio recorder Records to Compact Flash media 8-channel 96kHz/24-bit Broadcast WAV file recording 4-channel recording at 192kHz/24-bit resolution 8-channel recording plus stereo mix track at 44.1/48k resolution Internal stereo mixer (level/pan) * BWF file format with iXML metadata Multi-channel flash start Dual CF card slots for mirroring or continuous recording Color TFT touch panel interface 5-second pre-record buffer Cascade function for multiple unit operation Physical specifications: (8) balanced analog inputs on 25-pin D-sub connector (8) balanced analog outputs on 25-pin D-sub connector XLR stereo analog input and output for channels 1 and 2 (8) AES/EBU digital I/O on 25-pin D-sub connector * XLR stereo AES/EBU I/O for channels 1 and 2 8-channel ADAT optical digital in and out Video/word clock BNC input Word clock BNC out/thru SMPTE timecode BNC in/out PS/2 keyboard input RS-422/RS-232C serial control input Parallel control port RJ-45 LAN control (10/100/1000) on locking connector RC-HS20PD Remote input USB 2.0 host connector for transfer to flash memory drive 1/4" stereo headphone output
The Dave Smith Instruments Prophet '08 Pot Edition Eight-Voice Analog Synthesizer offers an alternative with 38 of the 52 front panel parameters controlled by pots (potentiometers); rotary encoders are still used for the remainder of the controls.The Dave Smith Instruments Prophet '08 Pot Edition is exactly the same as the non-pot version. The difference is strictly in the controls themselves. Encoders are “endless”—they have no minimum or maximum limit. The advantage to that is that when you edit a preset parameter, the change begins at the preset value and increases or decreases, depending upon the direction turned.Pots typically have about 270° of travel and have definite limits. The advantage to that is that you know, by feel, where the minimum and maximum limits are and you can sweep through the entire value range in less than one full turn.The Prophet Ά08 PE features three edit modes for the pots: Relative, Passthru, and Jump. In Relative mode, the value change is relative to the preset value. In Passthru mode, turning the pot has no effect until after the edited value equals the preset value (that is, until the edited value “passes through” the stored value). Jump mode uses an absolute value based upon the position of the pot when edited; turn a pot and the value jumps immediately from the stored value to the edited value.The Dave Smith Instruments Prophet '08 Pot Edition Main Features Include: 38 of the 52 front panel parameters controlled by pots Encoders are “endless”—they have no minimum or maximum limit Three edit modes for the pots: Relative, Passthru, and Jump Pots typically have about 270° of travel and have definite limits Two analog oscillators per voice Classic Curtis analog low-pass filters Extensive modulation routing capabilities Four-on-four splits and layers with separate stereo outputs for each layer Arpeggiator, gated 16 x 4 step sequencer, and LFOs all syncable 5-octave keyboard with semi-weighted action, velocity, and aftertouch Spring-loaded pitch wheel and assignable mod wheel 256 fully editable Programs (2 banks of 128) with 2 Layers (2 separate sounds) in each Program 16 x 4 gated step sequencer Arpeggiator 2 digitally controlled analog oscillators (DCOs) per voice with selectable sawtooth, triangle, saw/triangle mix, and pulse waves (with pulse-width modulation), and hard sync White noise generator 1 Analog Curtis low-pass filter per voice, selectable 2- and 4-pole operation (self-resonating in 4-pole mode) 3 Envelope Generators: filter, VCA, and assignable (four-stage ADSR + delay); Envelope 3 can loop 4 LFOs Glide (portamento): separate rates per oscillator Analog VCAs MIDI In, Out, Thru, and Poly Chain Main stereo audio output: 1/4" unbalanced Output B stereo audio output: 1/4" unbalanced Sustain pedal input: accepts normally on or normally off momentary footswitch Pedal/CV input: responds to expression pedals or control voltages ranging from 0 to 5 VDC (protected against higher or negative voltages) Headphone output: 1/4" stereo phone jack Includes power supply for 110V – 240V AC operation (13-15 VDC, 400 mA) and operation manual.
Avid Artist Transport, Weighted, optically encoded jog wheel and shuttle ring, 7 ergonomic transport/navigation keys with multi-color status LEDs, 6 programmable Soft Keys, 16 numeric keys for time code/marker navigation, Control multiple applications & workstations via Ethernet, Supports HUI and Mackie Control protocols. Dimensions: 231 x 238 x 30 mm (WxDxH), Weight: 2.4 kg.
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Bass frequency resonance is one of the most difficult acoustic issues to solve in any room. Until now, the most effective options have generally involved using bass traps that are specifically designed, or customized to fit the room. However, these solutions can often be expensive and invariably involve complex measuring procedures. ThereΆs also little scope for flexibility if changes are made to the listening space, such as moving furniture or gear.Vari Bass is designed for use in any professional recording space, home studio, or hi-fi room. Made from wood and high-density acoustic foam, the bass trap can be tuned between 50Hz and 100Hz by simply rotating its wooden top (which has a series of resonant pots) until the problem disappears. A scale is included to show the exact frequencies on which Vari Bass is acting. The unitΆs stand-alone structure enables it to be repositioned as required and used in different rooms.Advised Units per Rom AreaVary bass should be place here the pressure is maximum. 7 to 11 m2 - 4 Units (2 Box) 11 to 16 m2 - 6 Units (3 Box) 16 to 22 m2 - 8 Units (4 Box) For rooms more than 22 m2, modes formation should be measured and analyzed.
The Phantom Classic, like its predecessor the Phantom C (Cardioid), embodies reduction to the absolutely essential without compromise in performance. The design philosophy of this large diaphragm FET microphone - which works purely in cardioid characteristic, guarantees a sound quality otherwise found only in our tube microphones. Its capsule is based on the ideal of the VM1 giving the Phantom C a very natural and pristine sound character.The Phantom's success was most recently augmented through the release of the limited edition Phantom AE (anniversary edition). This edition was released to mark the occasion of the company's 10-year anniversary and was comprised of 1000 nickel colored as well as 1000 black microphones. After the paramount success of the limited Phantom anniversary series, we now deliver the legitimate successor of the Phantom C with the Phantom Classic.Specifications:Equivalent Noise: < 11 dB A (IEC651)Signal to Noise: > 83 dB (1 Pa/1 kHz-Cardioid)Sensitivity: 28 mV / Pa-CardioidPattern: CardioidFrequency Range: 20 Hz - 22 kHzMaximum SPL: 142 dB SPL 0,3 % THDPower Supply: 48 V phantom power
First class components, excellent dynamics and exceptionally low self noise exemplify Brauner microphones. This of course, holds true for the Valvet, a tube microphone based on the already legendary VM1, which provides a unique, finely balanced character.Our objective with the Valvet was to create a tube microphone that would be ideally suited to be used as a spot microphone through small construction and the use of a newly developed microphone suspension.The Valvet produces excellent recordings even under difficult acoustic conditions due to its directional characteristics, leaning slightly towards hyper-cardioid while maintaining a high level of back rejection. In omni though, it becomes the perfect choice for ambience recordings. In 1999, the Valvet received the SSAIR Award for best microphone in its category as the VM1 did the year before.The Valvet, boasting the same naturalness and openness as the VM1, produces recordings with wonderful spatial qualities and pleasing proximity and warmth. The Valvet uses the most common directional characteristics - omni and cardioid - and has a phase reverse switch for the use with non-standard pre-amplifiers.Specifications:Equivalent Noise: < 11 dB A (IEC651)Signal to Noise: > 83 dB (1 Pa/1 kHz-Cardioid)Sensitivity: 28 mV / Pa-CardioidPattern: Omni, CardioidFrequency Range: 20 Hz - 22 kHzMaximum SPL: 142 dB SPL 0,3 % THDPower Supply: 115 V or 230 V
The VM1 was the fulfillment of Dirk Brauner's dream to build his vision of the perfect sounding tube microphone. A microphone that captures the spirit of the classic vintage tube microphones and at the same time, meets the high technical standards of today's modern microphones. The VM1 was awarded the SSAIR Award in 1998, marking the beginning of a tradition of excellence in design philosophy, engineering and precision.In our humble opinion, no other microphone has played a greater role in the renaissance of the tube microphone. As the archetype of all Brauner microphones it continues to serve as the inspiration for all Brauner products. The VM1 has a completely natural, modern sound with the highest resolution of sonic detail. Its transparent open character is predestined for vocal and instrumental recordings and is the first choice microphone of top artists and sound engineers all over the world. The power supply of the VM1 allows infinite adjustments of the directional characteristics as well as a 10 dB reduction through a switchable attenuator.Specifications:Equivalent Noise: < 11 dB A (IEC651)Signal to Noise: > 83 dB (1 Pa/1 kHz-Cardioid)Sensitivity: 28 mV / Pa-CardioidPattern: all, infinitely variableFrequency Range: 18 Hz - 24 kHzMaximum SPL: 142 dB SPL 0,3 % THDPower Supply: 115 V or 230 V
Whereas the VM1 represents the natural sounding part of the VMA (which combines two microphones in one), the VMX offers the full character sound of the VMA in a single microphone and yields highest detailed resolution with a soft and silky top end. Slight accentuation of the low mid and bottom end as well as a soft upper mid range grant this reference quality tube microphone a unique and completely charming sonic character.The VMX excels in vocal and narration recordings through its well-balanced tonal appearance and pleasant proximity. It adds its own character without recognizably tainting the sound source. The VMX is also eminently suitable for instrumental recordings, through the reproduction of the finest details; giving string instruments for example the noticeable appearance of vicinity and warmth. Maximum flexibility in the fine-tuning of the respective recording is achieved through the possibility of infinitely adjustable directional characteristics.Specifications:Equivalent Noise: < 11 dB A (IEC651)Signal to Noise: > 83 dB (1 Pa/1 kHz-Cardioid)Sensitivity: 28 mV / Pa-CardioidPattern: all, infinitely variableFrequency Range: 20 Hz - 20 kHzMaximum SPL: 142 dB SPL 0,3 % THDPower Supply: 115 V or 230 V
Super Bass Extreme Ultra’s elegant wooden front is based on Vicoustic’s flagship Wavewood panel. Appropriate for corner mounting, it provides effective low frequency absorption between 60-125Hz, and delivers maximum effectiveness between 75 -100Hz.The wooden front panel has two functions, providing sufficient mid-high frequency absorption to control corner reflections without deadening the sound, while simultaneously acting as a diffuser. When the sound pressure is at its maximum, Super Bass Extreme Ultra’s internal membrane transforms high-pressure fluctuations into air motion. The membrane sympathetically vibrates over a frequency range of 75-100Hz, causing the air to pass through a layer of high-density foam absorbing the low frequencies.Highly recommended for smaller rooms with low frequency issues, Super Bass Extreme Ultra can be used in different corner positions. Besides its aesthetics, it’s extremely practical, with a modular structure allowing further units to be added as intended.
New extensible arm microphone stand featuring fixed elevation arm, fixed desktop stand and clip-on base desktop stand. Suggested for radio installations and dubbing studios. Available in matt black (DST260) finish.
AKG WMS 45 Perception Wireless Vocal SystemThe SR 45 receiver provides professional XLR and 1/4” jack outputs. The audio level and squelch threshold are adjustable on the receiver.The handheld transmitter features a spring-steel wire mesh front grill to protect the dynamic transducer from the hardships of onstage use. Its cardioid polar pattern ensures maximum gain before feedback and makes your voice cut through any mix.The system includes an SR 45 receiver, HT 45 handheld transmitter, universal power supply with US/UK/EU adapter, SA 45 stand adapter, and one AA size dry battery.AKG WMS 45 Perception Wireless Vocal System features:30 Mhz selection bandwidth (depending on local frequency plans)8 hours of operation with a single aa size batteryGain control on handheld transmitterLow battery indicatorNoiseless on/off/mute switch on handheld transmitterIncludes: SR 45 receiver, HT 45 handheld transmitter, universal power supply with US/UK/EU adapter, SA 45 stand adapter, and one AA size dry battery
Featuring a tight polar pattern and tailored-for-voice frequency response, the RODE Procaster is perfect for every application where a great sounding, rugged microphone with superior ambient noise rejection is demanded.RΨDE Procaster main features include:Broadcast quality soundHigh output dynamic capsuleBalanced, low impedance outputInternal shock mounting of capsule for low handling noiseInternal pop-filter to reduce plosivesRobust, all metal constructionAcoustic Principle: DynamicDirectional Pattern: CardioidFrequency range: 75 Hz - 18 kHzOutput impedance: 320OSensitivity: -56 dB re 1 Volt/Pascal (1.6 mV @ 94 dB SPL) +/- 2 dB @ 1kHzWeight: 695gDimensions: 214mmH x 53mmW x 53mmD
