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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.
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
Ultrasone IQ In-Ear-Headphones, made of superior metal die casting quality, 2-way hybrid technology: high performance BA-Driver + 8 mm dynamic driver. Frequency response: 17 - 21000 Hz20 Ohms impedance106 dB SPLOne 1,2 m cable with microphone & remote control and one 1,2 m behind-the-neck cableincl. caseflight adapterOMTP adapter for smartphones5 pairs of silicone and 2 pairs of TX-100 Comply foam earbud tipsWeight: 10 g (without cable)
Cello microphone holder for DPA 4099 Cello microphone, black
DPA IMK 4061 - miniature microphone set (Type 4061-BM),Includes 1 x 4061 microphone, 1 x XLR adapter (Type DAD6001), includes holders for strings, universal surface mounts with double sided tape, a magnet mount and foam windscreens.
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
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.
Pure Compressor II produces a wide range of compression from ultra clean subtile compression to classic heavy pumping. It’s up to your artistic choice not to the technology's limitations. New functions have been added to the original Pure Compressor. External side chain inputs are now available. New dynamic detection algorithms are available. The MS Width control manages stereo signals. A Hold parameter and a Threshold for Hysteresis, complement the settings. Classic modes and Feedback modes are now available for signal level detection, completing the legacy Solera mode. Various new profiles can be selected for detecting the dynamics of the signal. 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. Pure Compressor 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 slots To 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 Automation The 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 Pure Compressor II, as all Flux:: plug-ins, provide Up 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) AU RTAS* *The RTAS version requires ProTools 7 or later.
Pure DCompressor II allows you to restore the original dynamics of a sound. It is very useful for heavily compressed signals. Pure DCompressor II reverts your signal to a more natural state. New functions have been added to the original Pure DCompressor. External side chain inputs are now available. New dynamic detection algorithms are available. The MS Width control manages stereo signals. A Hold parameter and a Threshold for Hysteresis, complement the settings. Classic modes and Feedback modes are now available for signal level detection, completing the legacy Solera mode. Various new profiles can be selected for detecting the dynamics of the signal. 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, efficient and 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.Pure DCompressor 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 slots To 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 Automation The 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 Pure DCompressor II, as all Flux:: plug-ins, provide Up 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) AU RTAS* *The RTAS version requires ProTools 7 or later.
Pure DExpander II enhances the low levels of sounds (amplifying the signal below the threshold). The specialization information is magnified. Pure DExpander also makes your sound more compact. The “Angel’s Share” algorithm allows heavy processing whilst maintaining an organic character. New functions have been added to the original Pure DExpander. External side chain inputs are now available. New dynamic detection algorithms are available. The MS Width control manages stereo signals. A Hold parameter and a Threshold for Hysteresis, complement the settings. Classic modes and Feedback modes are now available for signal level detection, completing the legacy Solera mode. Various new profiles can be selected for detecting the dynamics of the signal. 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, efficient and 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. Pure DExpander 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 slots To 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 Automation The 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 Pure DExpander II, as all Flux:: plug-ins, provide Up 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) AU RTAS* *The RTAS version requires ProTools 7 or later.
The exclusive "Angel's Share" algorithm developed by FLUX:: allows you to remove unwanted noise or reverberation without adding a processed character to your sound.Pure Expander II produces a wide range of expansion processing from subtle expansion to hard noise-gate. New functions have been added to the original Pure Expander. External side chain inputs are now available. New dynamic detection algorithms are available. The MS Width control manages stereo signals. A Hold parameter and a Threshold for Hysteresis, complement the settings. Classic modes and Feedback modes are now available for signal level detection, completing the legacy Solera mode. Various new profiles can be selected for detecting the dynamics of the signal. 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, efficient and 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. Pure Expander 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 slots To 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 Automation The 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 Pure Expander II, as all Flux:: plug-ins, provide Up 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) AU RTAS* *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
Full Pack II Recording, Mixing, Remixing, Mastering - The complete range of Flux:: plug-ins in one big bundle..Alchemist aiming at professional mastering and re-mastering applications, the intricate design of Alchemist suggests a superior workflow and sound perfection within the field of dynamics processing, well adapted for the high demands of today’s pro-audio industry. Alchemist presents the option to operate as a single broadband processor as well as a multiband processor providing up to five individual bands, with the possibility to process up to eight channels simultaneously when inserted on a multi-channel (surround) bus. Each of the five band-sections provides - Flux:: core dynamics processors; Compressor, De-Compressor, Expander and De-Expander, all operating in parallel, as well as our exquisite BitterSweet transient processor.- MS (mid/side) control.- Auto Delay (Introduces a delay line in the processed signal allowing zero attack time).- Selectable level detection mode and dynamics detection profile- Dynamics processing envelope control (Attack, Hold, Release with optional Auto mode).- Input / Output gain control. Epure II a five-band state of the art equalizer using one of our most complex proprietary algorithms, ensuring the most superior quality achievable in digital equalizing processing. Each section provides- Individual bypass- Filter type that can be set to low cut, low shelving, peak, hi shelving or hi cut- Gain control with the range -24 dB to +24 dB- Frequency range covering the whole bandwidth- Q factor adjustable from 0.1 to 100 Solera II primarily designed for mastering and re-mastering applications, though heavily used as a regular dynamics processor with it's superior sonic quality and distortion free operation in many professional studios all over the world. 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 and the side chain features a three-band equalizer to generate frequency sensitive processing. Syrah a new generation dynamics processor, which handles the dynamics in a way that allows you to be creative, without a complicated user interface. 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. Pure Series is the very core of the Flux:: dynamics derived from our flagship Solera II and provided as five single plug-ins for maximum versatility with a classic “one tool for each task” approach to slim the workflow without compromising quality or function. Pure Compressor produces a wide range of compressions from ultra clean subtle compressions to classic heavy pumping ones. It's up to your artistic choices not to the technology limitations. Pure DCompressor allows you to restore the original dynamic of a sound. The DCompressor is superior for heavily compressed signal adding some of that natural feeling to the sound. Pure Expander produces a wide range of expansion process from subtle expansions to hard noise-gate. Using Flux:: proprietary Angel's Share and Hysteresis algorithms Pure Expander allows you to remove unwanted noises or reverberation without adding a processed character to your sound. Pure DExpander enhances the low levels of the sound, magnifies the spatialization information and is also very useful when you want a compact expander sound. The Angel's Share algorithm permits some heavy processing though still keeping an organic character with the sound Pure Limiter is using exquisite proprietary Flux:: algorithms generating a release-envelope producing no artifacts on the processed sound achieving a transparent limiting. A dramatic increase of the average audio level is easily achieved without damaging the perceived audio quality of the material.The Full Pack II bundle contains the following plug-ins: AlchemistEpure II (Native)Solera IIPure Compressor IIPure DCompressor IIPure DExpander IIPure Expander IIPure Limiter IISyrahFreeware Included with the bundle Stereo ToolBitterSweet IIProcessing Specifications 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.
Custom pedestal , comprised of 5 adjustable layers of reinforced dense MDF. Stacking in removable segments allow of tweeter adjustable height. The re-unforced pedestal also allows for solid bass transfer.
The American Audio VLP-2500 power amp gives you power to spare at a stunningly low price. The VLP-500 is rated at 750W RMS per channel at 4 ohms, 500W RMS per channel at 8 ohms, and 1,500W RMS bridged mono at 8 ohms. Frequency range is 20Hz-20kHz. TRS and XLR inputs, binding post, and Speakon outputs.Specifications: 1100W RMS Per Channel 4 Ohms, 1kHz, 0.1% THD 700W RMS Per Channel 8 Ohms, 1kHz, 0.1% THD 2200W RMS 8 Ohms, 1K Hz, 1% THD (Bridge Mode, Mono) Total Harmonic Distortion: Less Than 0.5% (20Hz - 20kHz 8 Ohms) Frequency Response: (+/-05db, rated output Power, 8 Ohms): 20Hz - 20K Hz Slew Rate: 20V Per usec Damping Factor 8 Ohm: 450 Impedance: 20K Ohms balanced / 10K Ohms unbalanced Cooling System: 2 high speed cooling fans Special Features: High/Low Pass Filter / Input Sensitivity Selector / Ground Lift / Limiter Dimensions (LxWxH): 17.75" x 19" x 3.5" / 453 x 482 x 88mm (2 rm spaces) Weight: 50 lbs. / 23 Kgs.
This rugged, ATA-style case is a great way to protect your Dave Smith instruments. Custom made to fit the Prophet '08, the case is also a perfect fit for the Evolver Keyboard when used with the removable accessory tray (included). A built-in accessory compartment holds the power supply.Features include:Made from rugged molded polyethyleneHeavy-duty aluminum valanceHalf the weight of standard wooden flight casesEPS foam protective interiorChrome-plated, locking latchesDeluxe plush interiorAccessory compartment for power supplyIncludes accessory tray for use with Evolver KeyboardWeight: ~13 lbs. (~5.9 kg) empty; ~33 lbs. (~15 kg) with Prophet '08 Keyboard; ~29 lbs. (~13.2 kg) with Evolver KeyboardDimensions: 14.6" W x 40.2" L x 6.74" H ( 37.1 cm x 102.1 cm x 17.1 cm)
The Distressor is an automatic gain (or volume) control device (AGC in engineering terms) designed for pro audio (music) applications. Basically, it electronically controls the volume of just about any source in a very pleasing, and "musical" manner - adding fullness, intelligibility, and especially in the Distressor's case - excitement. This type of device is often called a "limiter" or "compressor" by audio industry people. Its most probable uses will be in recording studios, live sound situations, movie sound production, and radio broadcast production.
Empirical Labs ELDS Derresser, Module for API-500 System, Freq. Response is 3 Hz to 120 kHz (-3dB points), Dynamic thomann Range - 115 dB from maximum output (.5%THD soft clipping) to minimum output. Transformerless noise floor typically – 93dBu, Distortion Ranges between .0035% and .01% depending on mode and settings, DC coupled Input and Outputs, High quality audio caps used internally, Input Balanced, thomann Output single ended but with balanced impedance, Pin 2 hot, Input Impedance is 20k Ohm. Output impedance is less than 51 ohms, D-S Dynamics Time Constants - Attack time is less than 1mS. Release .04 sec, fixed internally, Power Consumption – 8 thomann Watts max. 4 Watts Typical
Empirical Labs EL9 Mike-e, digital controlled Mic Pre-Amp with compressor, Instrument Input, Phase Switch, Freq. Response is 3 thomann Hz to 200 kHz on (No CompSat). CompSat is 3Hz - 150Khz. Optional Transformer output is 6Hz–80kHz, Signal to Noise - 130 dB signal to Noise.at 40dB gain. Maximum output is +28dBu, Distortion Ranges between .0006% and 15% depending on mode and settings, Transformer thomann coupled Input, Active and Transformer Outputs available, Input Impedance is 600 Ohms. Output impedance is less than 38 ohms., Attack/Decay Ranges - .9 - 100mS attack. .15 - 1 Sec., Power Consumption 15 Watts Max.
When compared to other handheld vocal microphones, the KMS 105 offers a superior resolution of the voice, with extended frequency response and accurate transient detail. Sibilance problems common to other handheld condenser mics are eliminated. No foam is used in the basket, thus eliminating any clouding or muffling of the sound. Also, by using a DC-polarized studio condenser capsule, all the nuances of the voice come through with subtlety, power and focus.The K 105 capsule was developed from the K 50, found in the KMS 150, KM 150 and KM 185 microphones. The K 105 features a supercardioid pattern, achieved by specially modifying the response of the K 50 pressure gradient transducer. The frequency response characteristic is developed in the acoustic realm, allowing for accurate representation of transient details found in music and speech, and thus transmitting a very open and detailed capture of the human voice. Due to the tight supercardioid polar pattern and the coloration-free off axis response, a very high level of gain before feedback is achieved. This polar pattern offers the highest degree of isolation while still allowing for a generous sweet spot in front of the microphone. The dynamic range of the KMS 105 is 132 dB, allowing reproduction of the artists full expression, without restraint. The maximum SPL is 150 dB and with a low self noise of 18 dB-A, a greater gain can be used without risk of adding noise to the mix. This way, the artist is able to use the microphone at a greater distance therefore maintaining a wide range of creative freedom during the performance. Because of the superior resolution and linear high frequency response, it is very easy for the artist using in-ear monitors to identify whether or not he or she is on mic with the KMS 105. And by employing a transformerless output circuit, this microphone can incorporate long cable runs with no loss of signal quality or high-frequency response. Most of the available vocalist microphones rely on the attenuating effects of foam layers inside the basket. Although this is certainly a cost-effective procedure, the detrimental effects of this foam layer can be shown quite easily. Treble frequencies are attenuated, and furthermore the polar pattern can be deformed. These drawbacks can be eliminated by selecting open mesh constructions. With careful dimensioning and selection of different mesh sizes, excellent pop and wind attenuation can be delivered. In the KMS 105 a combination of four wire meshes with different mesh sizes is used. The chosen combination yields a very constant polar pattern, while maintaining the capsules acoustic frontal frequency response. Naturally, the outer microphone basket is hardened steel, due to the required ruggedness for on-stage use.Features:Neumann sound on stageExcellent transparency for vocals/ speech and as an AnnouncerΆs mic for broadcasting/dubbingHigh SPL capability at 150 dB and a low self noise of 18 dB-AWithout off-axis colorationTransformerless outputEspecially suited for in-ear-monitoringSupercardioid polar pattern with excellent feedback rejectionEffective pop shielding without any side effectsSet includes SG 105 Stand clamp and Padded nylon bagSpecifications:Acoustical operating principle: Pressure gradient transducerDirectional pattern: supercardioidFrequency range: 20 Hz - 20 kHzSensitivity at 1 kHz into 1 kohm: 4.5 mV/PaRated impedance: 50 ohmsRated load impedance: 1000 ohmsEquivalent SPL CCIR 468-3: 28 dBEquivalent SPL DIN/IEC 651: 18 dB-AS/N ratio CCIR 468-3: 66 dBS/N ratio DIN/IEC 651: 76 dBMaximum SPL for THD 0.5%: 150 dBMaximum output voltage: 12 dBuDyn. range of the mic. amplifier DIN/IEC 651: 132 dBSupply voltage: 48 V ± 4 VCurrent consumption: 3.5 mAMatching connector: XLR 3FWeight: approx. 300 gDiameter: 48 mmLength: 180 mm
Heavy-duty speaker wall mount stand. The directional mounting bracket allows both horizontal and vertical adjustment. Available in black.Width: 124 mmHeight: 250 mmDepth: 380 mmWeight: 2.2 KgMaximum load: 35 Kg
Avid Artist Color, DAW Controller with 3 optical trackballs, 3 optical trackwheels, 6 touch-sensitive rotary encoders, 6 high-resolution OLED displays, transport and navigation controls, copy/paste grade buttons for memory banks 1-4, Ethernet-Connection. Dimensions : 500 x 238 x 30mm, Weight: 2.5 kg, for WIN 7 and MAC OSX
JAM is a premium digital converter featuring PureDIGITAL technology which delivers your guitar’s true tone to your iPad or Mac without compromise. JAM is the first high quality interface that makes a digital connection to both iOS devices and Mac. You will notice the sonic difference between JAM and other similar products immediately. PureDIGITAL instrument connection for pristine sound qualityDesigned specifically for electric guitar and bassNo configuration, just plug in and playEasily adjust input levels for any instrumentMulticolor LED for status indication and input level monitoringMade for GarageBand, Logic, and MainStageWorks with iPad, iPhone 4 and Mac
Ircam HEar v3 provides faithful reproduction of a stereo or surround mix with a pair of conventional stereo headphones. It relies on proven technology to model the various phenomena that occur when playing back audio material through a loudspeaker system.This allows monitoring a full surround mix in situations when a surround-capable environment is not available or practical. Another typical use of HEar is doing precise checking of a mix, which is convenient with headphones as these provide a ‘surgical’ and very detailed, microscope-like rendering of the audio. HEar can also prove very useful in a project studio context, and whenever noise isolation is a concern, as it helps achieving a more realistic sound environment.Binaural technology Binaural technology encompasses methods for recording, processing, synthesizing and reproducing sound that are specifically designed to preserve tridimensional localisation properties.In order to mimic the impression of a sound originating from a given incidence, it is sufficient to filter a mono signal, which on its own lacks any kind of spatial information, with both left and right HRTF filters. This constitutes the foundation of binaural synthesis.Please note that the resulting signal is only meant to be listened to with headphones, and isn’t designed for a conventional stereo loudspeaker setup.
VERB Session is based on the same fine technology used in the acclaimed IRCAM VERB algorithmic room acoustics and reverberation processor, with a modular construction employing a recursive filtering reverb engine, reproducing and synthesizing the specific acoustical characteristics of any spatial sound environment, developed utilizing the results of over a decade of intense research by the Acoustic and Cognitive Spaces Team at the IRCAM R&D centre in Paris.Tailored for simplicity, with a fast paced workflow well-suited for situations where the perfect result has to be achieved within seconds, VERB Session presents the ultimate solution whether you are a seasoned session engineer or a demanding broadcast and post-production mixer.
Complete mastering solution by Flux, containing five plugins for dynamic range adjustment and tonal balance.Solera implements 4 different detectors and envelop generators in parallel and simultaneously.Compressor, de-compressor, expander and de-expander which can all be independently contolled. In addition, here are some of the key features that make our plug-in the best available for dynamic processing in the digital domain : An analog like metering. RMS values are more useful than dB-fs when setting a dynamic processor. The "Angel's Share" parameter takes into account the signal for intelligent relaxation of the compressor action. The "Hysteresis" parameter allows compressing and de-compressing, independently of the sound level and can be mixed with the standard compression scheme. The Auto Release features an unique algorithm that avoids typical pumping effect. A delay line can be inserted in the signal path to produce a null attack time. The A/B Compare and Morphing section enables ultra fast and precise operations. Dry Mix function to achieve the full control of the dynamic processing amount.Epure features for every section: Individual bypass. The filter type can be set to low cut, low shelving, peak, hi shelving, hi cut The gain be set from -24 dB to +24 dB The frequency range covers the whole bandwidth The Q factor can be adjusted from 0.1 to 1Epure also features five unique functions to make equalization even more easy: Copy A and Copy B duplicate the twenty settings of a slot to the other in one click X2 and /2 affect the gain value for every equalizer section Invert turns the positive gain values to negative ones and vice versaEpure includes internal Input /output routings. A channel group management allowing to set many channels or individual channel setting is also available from the new setup window. From this window, up to 4 MS encoders and up to 4 MS decoders can be engaged if required. Combined with our exclusive A /B morphing system, these functions enable a global and efficient control of the equalization curve in a second. Pure Limiter is supposed to be the very last stage of your audio processing chain. It uses the exclusive FLUX technologies to generate a release envelop producing no artefact on the processed sound.A transparent limiting dramatically increasing the average audio level can now be achieved without damaging the perceived audio quality.
A comprehensive toolbox with the elemental components needed to take control over the dynamic processing when recording, mixing or mastering. Pure Compressor produces a wide range of compressions from ultra clean subtle compressions to classic heavy pumping ones. It's up to your artistic choices not to the technology limitations. Pure Expander produces a wide range of expansion process from subtle expansions to hard noise-gate. Using Flux:: proprietary Angel's Share and Hysteresis algorithms Pure Expander allows you to remove unwanted noises or reverberation without adding a processed character to your sound. Pure Limiter is using exquisite proprietary Flux:: algorithms generating a release-envelope producing no artifacts on the processed sound achieving a transparent limiting. A dramatic increase of the average audio level is easily achieved without damaging the perceived audio quality of the material.
Tetra takes the award-winning sound and features of Mopho, multiplies them by four, and packs them in a box less than half an inch larger!Tetra has multiple personalities. It is a four-voice, analog poly synth, a sort of “mini Prophet.” It's a four-part, multitimbral synth with separate outputs, essentially four Mophos in one very compact box. And it’s a voice expander for other Tetras or the Prophet ’08.Affordable, fully programmable poly synth with a 100% analog signal path.Classic, real analog sound—including legendary Curtis analog low-pass filter.Four-part multitimbral capability with four separate outputs.Combo Mode for huge unison patches, stacked sequences, and "modular-style" poly sounds.Expandable: poly chain with other Tetras, Prophet '08, and Mopho for expanded polyphonyJust 7.9" x 5" (20.07 cm x 12.7 cm).Free editor for Mac OS and Windows.
The Fundamentals In Dynamic Processing in a comprehensive bundle by Flux. Pure Compressor is the compressor section of the Solera. It uses the exclusive Angel's Share and Hysteresis algorithms developed by FLUX:: Pure Compressor produces a wide range of compressions from ultra clean subtle compressions to classic heavy pumping ones. It's up to your artistic choices not to the technology limitations. Pure Expander is the expander section of the Solera. The exclusive Angel's Share and Hysteresis algorithms developed by FLUX:: allows you to remove unwanted noises or reverberation without adding a processed character to your sound. Pure Expander produces a wide range of expansion process from subtle expansions to hard noise-gate. Pure DCompressor is the de-compressor section of the Solera. It uses the exclusive the Angel's Share and Hysteresis algorithms developed by FLUX:: Pure DCompressor allows you to restore the original dynamic of a sound. It's very useful for heavily compressed signal. Pure DCompressor adds some naturalness. Pure DExpander is the de-expander section of the Solera. It uses the exclusive the Angel's Share and Hysteresis algorithms developed by FLUX:: Pure DExpander enhances the low levels of the sound. The spacialization information is magnified. Pure DExpander also makes your sound more compact. The Angel's Share algorithm permits some heavy processing while keeping an organic character. Pure Limiter is supposed to be the very last stage of your audio processing chain. It uses the exclusive FLUX technologies to generate a release envelop producing no artefact on the processed sound. A transparent limiting dramatically increasing the average audio level can now be achieved without damaging the perceived audio quality.
Guitarist John Scofield has blazed a fearless trail, moving effortlessly in and out of bop, fusion, rock, funk, blues, and soul. Since becoming an Ibanez Signature artist in 2001, John’s trusted JSM100 has been at the ready, waiting for the next adventurous musical turn in the road. neck type JSM Mahogany set-in neckbody Flamed Maple top/back/sidesfretboard Bound Ebony fretboard w/Acrylic & Abalone block inlayfret Jumbo frets w/Prestige fret edge treatmentbridge Gotoh® 510BN bridgetailpiece Gotoh® GE101A tailpieceneck pickup Super 58 (H) neck pickup (Passive/Alnico)bridge pickup Super 58 (H) bridge pickup (Passive/Alnico)factory tuning 1E, 2B, 3G, 4D, 5A, 6Estring gauge .013/.016/.022/.032/.042/.052nut Half-Bone/Half-Brass nuthardware color Goldcase/bag Hardshell case included
G-DRIVE mobile 500GB FW/USB
G-DRIVE mobile 750GB FW/USB
A powerful analog filtering and distortion unit with a huge frequency range and a killing TUBE overdrive behavior. Independent low pass, band pass and high pass filters with resonance, frequency and ADSR (positive - negative) controlsThese are switchable between parallel or serialThe second filter has a 6 octave switch, and is synchronizable with the first filter1 ADSR generator for these filters with an external output for use with more filterbanks or modular systems, triggered by the incoming signal, by a second sounds source, or by a gate1 FM input for the filters, internal or external audio, LFO or CV1 VCA overdrive with AR generator also1 ring modulator input, internal or external audio, LFO or CV
Θρυλική αναλογική μονάδα φίλτρου / distortion με τεράστια συχνοτική περιοχή, παραμόρφωση σε στυλ λάμπας, envelopes και πολλές επιλογές modulation / routing.Χαρακτηριστικά:Ανεξάρτητα φίλτρα Lowpass, Bandpass και Highpass με ρυθμιζόμενο resonance, cutoff και θετικό/αρνητικό ADSRΠαράλληλη ή σειριακή τοπολογίαΤο δεύτερο φίλτρο έχει διακόπτη 6 octave και μπορεί να συγχρονιστεί με το πρώτοADSR generator με έξοδο που μπορεί να χρησιμοποιηθεί σε modular συστήματαΕίσοδος FM για τα φίλτρα, εσωτερική ή εξωτερική, LFO ή CVVCA overdrive με AR generatorΕίσοδος Ring Modulation, εσωτερική ή εξωτερική, LFO ή CV
The system 270 consists of a stereo amplifier SA-270, an AM/FM stereo tuner, ST-270 and a CD-player CD-270. The performance of this system outbeats the 250 system due to upgraded units and the design has also been upgraded. A system remote control for all functions is supplied with every unit, which can work as perfect with any other branded equipment, but in a system, all of them are matched for giving the best performance for the money. TECHNICAL SPECIFICATIONS:Channel Separation: >60dBCharacteristics: THD 99dBType: CD PlayerWeight: 4.0 kg
This is the ultimate Vienna Symphonic Library anthology, from A to Z. 23 Collections, covering the symphonic orchestra and a lot more. This package includes every Vienna Instruments DVD Collection the Vienna Symphonic Library has ever released, except the Vienna Special Edition. With its unsurpassed number of over one million pristine sound recordings, this tour de force is by far the largest compilation of samples on the planet. And still, the power of Vienna Instruments lies in its simplicity, giving quick and easy access to this wealth of instruments and articulations, and rendering authentic results at the simple touch of a key. Includes: Solo Strings I Solo strings II Chamber Strings I Chamber Strings II Orchestral Srings I Orchestral Srings II Appassionata Strings I Appassionata Strings II Harps Woodwinds I Woodwinds II Special Woodwinds Brass I Brass II Special Brass Vienna Dimension Brass Saxophones Percussion Elements Vienna Imperial Konzerthaus Organ Special Keyboards Vienna Choir I
This is the ultimate Vienna Symphonic Library anthology, from A to Z. 23 Collections, covering the symphonic orchestra and a lot more. This package includes every Vienna Instruments DVD Collection the Vienna Symphonic Library has ever released, except the Vienna Special Edition. With its unsurpassed number of over one million pristine sound recordings, this tour de force is by far the largest compilation of samples on the planet. And still, the power of Vienna Instruments lies in its simplicity, giving quick and easy access to this wealth of instruments and articulations, and rendering authentic results at the simple touch of a key. Includes: Solo Strings I Solo strings II Chamber Strings I Chamber Strings II Orchestral Srings I Orchestral Srings II Appassionata Strings I Appassionata Strings II Harps Woodwinds I Woodwinds II Special Woodwinds Brass I Brass II Special Brass Vienna Dimension Brass Saxophones Percussion Elements Vienna Imperial Konzerthaus Organ Special Keyboards Vienna Choir I
