Virtual Plug-ins

Already familiar with the Production Bundle? Learn how v4 is different.The MH Production Bundle is a suite of signal processing plug-ins that let you stop recording and start making records. With all the critical tools you need to shape, tame, manipulate, and master your tracks, it is the essential audio engineering toolset that no one should be without.The Production Bundle includes 10 fantastic sounding plugins that will help you bring your work to new levels:Updated MH ChannelStripLegendary channel strip processing for all DAWsNew! MH Sonic EQThe Original Digital Mastering EQLegendary Sound, Precision and VersatilityNew! MH SuperGateFinally, a gate that actually does what a gate was always supposed to do…Updated MH Dirty DelayGritty, Musical Feedback DelayUpdated MH CharacterA bucket full of old school goodness.....Updated MH HaloVerbCreate your perfect spaceUpdated MH Multiband DynamicsSometimes the best EQ is a compressorUpdated MH Multiband ExpanderExpand your horizonsUpdated MH TransientControlSophisticated Transient Modifier & WaveshaperUpdated MH Precision De-EsserA Surgical Strike On Stubborn SibilanceDifferences between v3 and v4MH Production Bundle v4 featuresNew Plug-ins included in the bundle:MH SuperGate - A new, flexible state-space based Signal Gate that finally actually does what a gate was always supposed to do.MH Sonic EQ - 6-band fully parametric Equalizer, where every band in the EQ supports 2nd to 8th order filters, using the low-noise 64-bit state-space EQ structure. This implementation is based on the original SonicEQ developed by Dr. Andy Moorer for the Sonic Solutions mastering DAW, and then ported and enhanced for the Mobile I/O DSP by Dr. B.J. Buchalter.New Functions for Pro Tools:Adds support for EQ and Dynamics graphs in Pro ToolsEnables EQ and Dynamics pages in Avid Control/EuCon when running in Pro ToolsUpdates to Character:Provides a new UI for MH CharacterProvides additional oversampling options for MH CharacterCommon changes for all plug-ins:Adds VST3 versions for all plugins on both macOS and WindowsVST3 Apple Silicon Native support for Steinberg products (Cubase, Nuendo, etc).VST3 Native Sidechain Support for our plugins that provide Sidechain input.Undo/RedoA/B snapshots and snapshot blendSoftware update trackingMore flexible resizabilityCommon controls in the header bar (Help, Resize, Pane Disclosure)All plugins support AU, AAX, VST2 and VST3 on macOS.All plugins support AAX, VST2, and VST3 on Windows.Improved Preset ManagementPlug-in Settings copy/paste between instancesPlug-in details pane (accessed by the MH Logo) that shows:About boxLinks to documentation, tech support and frequently asked questionsSoftware update information and release notesCurrent release notesPlug-in PreferencesSignificantly reduced size download for the installers on macOSAdds a user configurable themes for most pluginsAdds user selectable knob styles for many pluginsAdds user preference for band colorsAdds user preference UI colorsAdds user preference for showing/hiding knob ringsRefines layout and visuals of the UIs

Complete software package for composers, producers and studios (Download)One year access to Presonus software products including a permanent licence for Studio One ProfessionalStudio One Professional: easy to use DAW and Mastering-softwareNotion: Notation-software for the creation of sheet music, scores etc.Audio Batch Converter: a converter tool for audio filesEvery Presonus-Plugin such as Ampire, Channel Strip Collection, CTC-1 Pro Console Shaper, Fat Channel XT, Presence XT Editor and many othersMore than 100 Sound libraries, e.g. PreSonus Symphonic Orchestra, PreSonus Studio Grand, the complete Spark sample and loop collection, Tom Brechtlein Drums, Notion add-on instruments etc.Customisable PreSonus Sphere workspaces with up to 100 GB of cloud storage, online collaboration with other PreSonus Sphere users, live chats and much more.Share user presets, loops, tools and other contentConstantly growing range of exclusive tutorials, courses and instructional videosIncl. all updates during the membership periodSystem RequirementsFormat: E-MailLicense validity: 12 monthsCopy protection: Online activationSimultaneous activations: 5Windows: from 10 (64-Bit)Mac OS (64 Bit): from 10.13CPU min.: Intel Core i3 / AMD A10 / Apple M1/M2RAM min. (GB): 4min Space on HD (GB): 40 GBDisplay: 1366 x 768Additional System requirements: ASIO-comp. Audiointerface (Windows)

Ampire is PreSonus’ new State Space Modeled guitar amp/cab/stompbox simulator, included in Studio One Professional 4.6 and later. You get five new amplifier models, 16 cabinet emulations, and a pedalboard full of effects, and—in a first for PreSonus—we’ve made Ampire available for use in other DAWs in VST3, AAX, and AU formats! In short, Ampire has never sounded better, due in no small part to...

Plug ins collection in PRO TOOLS enviroment, with FOCUSRITE algorithm (ISA 110 & ISA 130). Includes plug ins: ISA 110 Equalizer, ISA 130 Multichannel Surround Compressor, ISA 130 Expander, ISA 130 Gate, ISA 130 De-esser.

Where sound becomes music! The Epure V3  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 conceivableTDM/Venue version available.For the full feature set, hover your mouse over the small images to the right above.Epure V3 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 Sonnox Fraunhofer Pro-Codec Plug-In is designed for the real-time auditioning, encoding and decoding of audio signals using Fraunhofer codecs. In the past it has not been possible to critically audition and then encode multiple formats in real time within a host DAW environment. The revolutionary Sonnox Fraunhofer Pro-Codec now makes this possible, and includes extensive monitoring tools and built-in encoding/decoding features. Using this plug-in mix engineers may produce mixes optimised towards specific target codecs, thereby ensuring maximum fidelity. Similarly, Mastering engineers may audition in the final format and produce compensated masters for final encoding and distribution. Features Supported Codecs: - mp3, mp3 Surround, AAC-LC, HE-AAC, and HE-AAC v2 - mp3-HD and HD-AAC (Lossless Codecs) Ability to select up to five codecs for simultaneous export, in real time, to encoded files Comprehensive auditioning with glitchless switching Ability to audition the difference between input signal and codec output AB auditioning in real time between codecs, or between codec and input signal ABX mode for blind statistical testing Graphical displays include: - High resolution display of the Input signal and Diff signal. - Indication of the audibility of audio removed by the encoding process - Bitstream levels and ability to compensate for overloads - Indication of filesize or datastream compression - Phase metering Off-line encoding and decoding Compatible with many digital audio workstation applications, such as: - Pro Tools, Logic, Cubase, Nuendo, Sequoia and Wavelab. (Mac and PC)

Sonnox Restore is a suite of three plug-ins designed to accurately restore audio that is impaired. Its advanced algorithms and novel features allow fast and extremely effective removal of pops, clicks, crackles, scratches, hum, buzzes and extraneous background noise from virtually any recording.Contains: Oxford DeClicker, Oxford DeBuzzer and Oxford DeNoiser (Native format only - RTAS, AU or VST)

A well defined toolbox with a selection of Flux:: dynamic processing plug-ins and eq. providing a versatile and comprehensive set of precision tools for your everyday recording and mixing session work giving you the best processor for each task within the range of dynamics and equalization processing. Epure II is our State-Of-The-Art Equalizer, a five-band equalizer using one of our most complex proprietary algorithms ensuring the superior quality ever for a digital equalizing processor.  Each section provides- An 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   To match the perfection of the sound quality a range of user interface functions to improve the workflow are provided- Copy A and Copy B duplicate the twenty settings of one slot to the other with a single click- Morphing A/B allowing for ultra fast and precise operation- X2 and /2 affect the gain value for each equalizer section- Invert, to turn the positive gain values to negative ones and vice versa 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 soundPure 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.

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 ISolo strings IIChamber Strings IChamber Strings IIOrchestral Srings IOrchestral Srings IIAppassionata Strings IAppassionata Strings IIHarpsWoodwinds IWoodwinds IISpecial WoodwindsBrass IBrass IISpecial BrassVienna Dimension BrassSaxophonesPercussionElementsVienna ImperialKonzerthaus OrganSpecial KeyboardsVienna Choir I

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.

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

Altiverb 7 Regular is the Post and music production industry standard stereo convolution reverb. It comes with the renowned full library of impulse responses. Below you can click through pictures that reveal most of Altiverb's features at a glance. Read on to find out about the features that are not visible in the interface.Altiverb 7 RegularWindows: VST, RTAS and Audio Suite.Mac OS X: MAS, Audio Unit, RTAS and Audio Suite, Universal Binaries for both Power PC and Mac-IntelPower PC VSTThe Full Altiverb Impulse Response libraryFull parameter automationan impulse response browser that replaces the impulse response popup menu.mono to mono, mono to stereo, and stereo to stereo channel configurations.supported sample rates up to and including 96 kHz.40 automation preset memories for total-recall via parameter automation.iLok or Challenge-Response authorization.Sample your own spaces or gear (Mac only)

A bad GSM connection on a busy sidewalk, a bullhorn with feedback and a helicopter overhead, or a 1952 rockabilly guitar amp in a recording studio live room: The Speakerphone audio plug-in gives you authentic speakers of any size together with their natural environments.All the walkie-talkies, distant transistor radios, upstairs TV sets, bullhorns, cell phones and guitar cabinets you will ever need. Speakerphone (Mac , Windows, iLok or challenge response) will add dial tones, operator, static, and you can select from a wealth of ambiences on either the caller or receiver's end.400 speaker impulse responses, 23 microphones, 106 'Covers' (from blankets to car trunks), 53 Altiverb rooms and outdoor spaces, 5 gigabyte of ambiences and sound FX, and 12 DSP modules from Leslie to GSM compression, conveniently presented to you in well over 500 presets.Take the tours by clicking the movies on the right.

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, Spat v3 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.Designed for surround and multi-channel use, Spat v3 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.Each of the up to eight incoming audio channels is internally mapped to a range of Virtual Sources localized in a 3D space, and connected to a room (a reverb). Up to 3 rooms in parallel are provided, presenting the option to simulate complex spaces (coupled room acoustics).Spat v3 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.Specifications:Availability Ircam Spat v3 is available in the following configurations:Native - AU / VST / AAX NativeProcessing Specifications Ircam Spat v3, as all Flux:: plug-ins, provideUp to 8 channels Input/Output.64-bits internal floating point processing.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 NativeMac OS X (Intel) - 10.7, 10.8 and 10.9 in both 32 and 64 bits.VST (2.4)AUAAX Native

The Oxford Dynamics plug-in is modeled on the extremely flexible and capable unit used in the OXF-R3 professional mixing console. Resulting from many years research into professional dynamics applications, it offers separate Compress, Limit, Expand, Gate and side chain EQ functions, with full independent control of all parameters. Features such as selectable time constant curves and variable soft compress functions allow the user to confidently tackle all common uses of compression, from subtle unobtrusive level control and mastering functions to the production of great artistic effects. The use of a feed-forward architecture with logarithmic side chain processing, making use of look-ahead techniques, ensures exemplary sonic characteristics and dynamic accuracy, with an artistic capability simply unavailable from other single units, analogue or digital. This highly sophisticated and professional product has the power and flexibility to obviate the need for many of the separate applications most users keep for specific uses.         FEATURES Channel Dynamics with separately controlled sections for Compressor, Limiter, Gate and Expander. Channel Dynamics includes fully featured 2 Band side chain EQ which can also be used in signal path. Separate Bus Compressor / Limiter with surround multi-format support and selectable Sub channel filtering and gain contribution control. Selectable linear and exponential time constant curves. Highly accurate logarithmic side chain processing. Fully variable soft ratio function for extreme programme tolerance and highly musical compression. Variable harmonic enhancement for extra loudness, presence and 'punch'. Selectable re-dithering function for word length reduction in mastering situations. Extremely low signal path noise and distortion, below -130dBr. All functions are fully automated.

The Oxford Dynamics plug-in is modeled on the extremely flexible and capable unit used in the OXF-R3 professional mixing console. Resulting from many years research into professional dynamics applications, it offers separate Compress, Limit, Expand, Gate and side chain EQ functions, with full independent control of all parameters. Features such as selectable time constant curves and variable soft compress functions allow the user to confidently tackle all common uses of compression, from subtle unobtrusive level control and mastering functions to the production of great artistic effects. The use of a feed-forward architecture with logarithmic side chain processing, making use of look-ahead techniques, ensures exemplary sonic characteristics and dynamic accuracy, with an artistic capability simply unavailable from other single units, analogue or digital. This highly sophisticated and professional product has the power and flexibility to obviate the need for many of the separate applications most users keep for specific uses.         FEATURES Channel Dynamics with separately controlled sections for Compressor, Limiter, Gate and Expander. Channel Dynamics includes fully featured 2 Band side chain EQ which can also be used in signal path. Separate Bus Compressor / Limiter with surround multi-format support and selectable Sub channel filtering and gain contribution control. Selectable linear and exponential time constant curves. Highly accurate logarithmic side chain processing. Fully variable soft ratio function for extreme programme tolerance and highly musical compression. Variable harmonic enhancement for extra loudness, presence and 'punch'. Selectable re-dithering function for word length reduction in mastering situations. Extremely low signal path noise and distortion, below -130dBr. All functions are fully automated.

The Oxford Dynamics plug-in is modeled on the extremely flexible and capable unit used in the OXF-R3 professional mixing console. Resulting from many years research into professional dynamics applications, it offers separate Compress, Limit, Expand, Gate and side chain EQ functions, with full independent control of all parameters. Features such as selectable time constant curves and variable soft compress functions allow the user to confidently tackle all common uses of compression, from subtle unobtrusive level control and mastering functions to the production of great artistic effects. The use of a feed-forward architecture with logarithmic side chain processing, making use of look-ahead techniques, ensures exemplary sonic characteristics and dynamic accuracy, with an artistic capability simply unavailable from other single units, analogue or digital. This highly sophisticated and professional product has the power and flexibility to obviate the need for many of the separate applications most users keep for specific uses.         FEATURES Channel Dynamics with separately controlled sections for Compressor, Limiter, Gate and Expander. Channel Dynamics includes fully featured 2 Band side chain EQ which can also be used in signal path. Separate Bus Compressor / Limiter with surround multi-format support and selectable Sub channel filtering and gain contribution control. Selectable linear and exponential time constant curves. Highly accurate logarithmic side chain processing. Fully variable soft ratio function for extreme programme tolerance and highly musical compression. Variable harmonic enhancement for extra loudness, presence and 'punch'. Selectable re-dithering function for word length reduction in mastering situations. Extremely low signal path noise and distortion, below -130dBr. All functions are fully automated.

The EQ plug-in is based on the OXF-R3 EQ section. It is a fully functional 5-band application with selectable shelf settings on LF and HF sections. Additionally, separate variable slope low pass and high pass filters are provided. The EQ also features 4 different selectable EQ types that cover most of the EQ styles currently popular amongst professional users, including some legacy styles which are renowned for their artistic capability. The use of novel coefficient generation and intelligent processing design provides unparalleled performance that surpasses analogue EQ in both sound quality and artistic freedom. This plug-in may well provide all the EQ you ever needed.         FEATURES Five band parametric design Selectable shelf settings on LF and HF sections LF and HF Filters providing up to 36dB/Octave slope Four selectable EQ types Fully decramped HF response Support for sample rates up to and including 192kHz(96kHz on Demo version) Fully automatable (including separate automation for A and B settings) Includes six plugins in total for optimal DSP usage(Mono and Stereo versions of EQ, Filters, and EQ and Filters)

The EQ plug-in is based on the OXF-R3 EQ section. It is a fully functional 5-band application with selectable shelf settings on LF and HF sections. Additionally, separate variable slope low pass and high pass filters are provided. The EQ also features 4 different selectable EQ types that cover most of the EQ styles currently popular amongst professional users, including some legacy styles which are renowned for their artistic capability. The use of novel coefficient generation and intelligent processing design provides unparalleled performance that surpasses analogue EQ in both sound quality and artistic freedom. This plug-in may well provide all the EQ you ever needed.         FEATURES Five band parametric design Selectable shelf settings on LF and HF sections LF and HF Filters providing up to 36dB/Octave slope Four selectable EQ types Fully decramped HF response Support for sample rates up to and including 192kHz(96kHz on Demo version) Fully automatable (including separate automation for A and B settings) Includes six plugins in total for optimal DSP usage(Mono and Stereo versions of EQ, Filters, and EQ and Filters)

The EQ plug-in is based on the OXF-R3 EQ section. It is a fully functional 5-band application with selectable shelf settings on LF and HF sections. Additionally, separate variable slope low pass and high pass filters are provided. The EQ also features 4 different selectable EQ types that cover most of the EQ styles currently popular amongst professional users, including some legacy styles which are renowned for their artistic capability. The use of novel coefficient generation and intelligent processing design provides unparalleled performance that surpasses analogue EQ in both sound quality and artistic freedom. This plug-in may well provide all the EQ you ever needed.         FEATURES Five band parametric design Selectable shelf settings on LF and HF sections LF and HF Filters providing up to 36dB/Octave slope Four selectable EQ types Fully decramped HF response Support for sample rates up to and including 192kHz(96kHz on Demo version) Fully automatable (including separate automation for A and B settings) Includes six plugins in total for optimal DSP usage(Mono and Stereo versions of EQ, Filters, and EQ and Filters)

The Inflator is a unique process that can provide an increase in the apparent loudness of almost any programme, without obvious loss of quality or audible reduction of dynamic range, yet avoiding damaging increases in the peak level of the signal. The inflator process can also bring power, presence and warmth to programme material and even provide headroom overload margin above digital maximum with a subtlety and musical character reminiscent of tube systems. Use the Inflator to produce louder mixes than you thought possible without overloads or compression pumping, or use it to add natural warmth and character to acoustic or jazz mixes.          FEATURES Increases the loudness of almost any programme material. Creates warmth, character and dynamic excitement, similar to that of analogue systems. Provides virtual headroom above digital maximum to allow percussive peaks to pass without causing signal overload. Creates artistic effects ranging from subtle tube-like harmonic characteristics for warmth, presence and 'in your face' fatness, to outright saturation distortion modelling. Features two modes of operation - direct and band split - for maximum flexibility and artistic creativity. These two modes also ensure optimal DSP usage. Digidesign Pro Tools HD, Mix compatible (RTAS version also included)

The Inflator is a unique process that can provide an increase in the apparent loudness of almost any programme, without obvious loss of quality or audible reduction of dynamic range, yet avoiding damaging increases in the peak level of the signal. The inflator process can also bring power, presence and warmth to programme material and even provide headroom overload margin above digital maximum with a subtlety and musical character reminiscent of tube systems. Use the Inflator to produce louder mixes than you thought possible without overloads or compression pumping, or use it to add natural warmth and character to acoustic or jazz mixes.          FEATURES Increases the loudness of almost any programme material. Creates warmth, character and dynamic excitement, similar to that of analogue systems. Provides virtual headroom above digital maximum to allow percussive peaks to pass without causing signal overload. Creates artistic effects ranging from subtle tube-like harmonic characteristics for warmth, presence and 'in your face' fatness, to outright saturation distortion modelling. Features two modes of operation - direct and band split - for maximum flexibility and artistic creativity. These two modes also ensure optimal DSP usage. Digidesign Pro Tools HD, Mix compatible (RTAS version also included)

The Inflator is a unique process that can provide an increase in the apparent loudness of almost any programme, without obvious loss of quality or audible reduction of dynamic range, yet avoiding damaging increases in the peak level of the signal. The inflator process can also bring power, presence and warmth to programme material and even provide headroom overload margin above digital maximum with a subtlety and musical character reminiscent of tube systems. Use the Inflator to produce louder mixes than you thought possible without overloads or compression pumping, or use it to add natural warmth and character to acoustic or jazz mixes.          FEATURES Increases the loudness of almost any programme material. Creates warmth, character and dynamic excitement, similar to that of analogue systems. Provides virtual headroom above digital maximum to allow percussive peaks to pass without causing signal overload. Creates artistic effects ranging from subtle tube-like harmonic characteristics for warmth, presence and 'in your face' fatness, to outright saturation distortion modelling. Features two modes of operation - direct and band split - for maximum flexibility and artistic creativity. These two modes also ensure optimal DSP usage. Digidesign Pro Tools HD, Mix compatible (RTAS version also included)

The Sonnox Transient Modulator is an application that allows dynamic level of signals to be modified by the transients in the programme material over time. The effect is to bring transient events in the programme forwards, or push them into to the background, such that the attacks of instruments can be accentuated or softened depending on settings.        FEATURES Radically changes the dynamics of instruments. Accentuates or flattens attacks and transients. Brings sounds forward or push them back. Increases or reduces the effects of ambience. Produces rounded and dynamic percussive effects. Hardens up and give life to dull or flat-sounding recordings and mixes, without the unwanted changes in overall timbre associated with multi-band compression techniques. Variable harmonic enhancement for extra loudness, presence and 'punch'. Increases overall modulation potential by the reduction of very short peaks.

The Sonnox Transient Modulator is an application that allows dynamic level of signals to be modified by the transients in the programme material over time. The effect is to bring transient events in the programme forwards, or push them into to the background, such that the attacks of instruments can be accentuated or softened depending on settings.        FEATURES Radically changes the dynamics of instruments. Accentuates or flattens attacks and transients. Brings sounds forward or push them back. Increases or reduces the effects of ambience. Produces rounded and dynamic percussive effects. Hardens up and give life to dull or flat-sounding recordings and mixes, without the unwanted changes in overall timbre associated with multi-band compression techniques. Variable harmonic enhancement for extra loudness, presence and 'punch'. Increases overall modulation potential by the reduction of very short peaks.

The Sonnox Transient Modulator is an application that allows dynamic level of signals to be modified by the transients in the programme material over time. The effect is to bring transient events in the programme forwards, or push them into to the background, such that the attacks of instruments can be accentuated or softened depending on settings.        FEATURES Radically changes the dynamics of instruments. Accentuates or flattens attacks and transients. Brings sounds forward or push them back. Increases or reduces the effects of ambience. Produces rounded and dynamic percussive effects. Hardens up and give life to dull or flat-sounding recordings and mixes, without the unwanted changes in overall timbre associated with multi-band compression techniques. Variable harmonic enhancement for extra loudness, presence and 'punch'. Increases overall modulation potential by the reduction of very short peaks.