Flux

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.

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.

Pure Analyzer System – The Next Generation RTA (Real Time Analyzer) System The Flux:: Pure Analyzer System is a network-driven modular RTA-system providing an extensive range of analyzer options for any imaginable audio analysis and measurement applications. For maximum flexibility and efficiency, Pure Analyzer presents the option to separate the signal acquisition from the analysis utilizing a Sample grabber plug-in, and in addition to the Sample grabber plug-in, Pure Analyzer also provide I/O support for ASIO and Core Audio.   Sample Grabber Sample grabber is a standard DAW plug-in available in all common formats (VST, AU, RTAS and TDM) that acquires the source and response signals and sends the signal through the network using the ZeroConf/Apple Bonjour protocol. You can insert as many instances of Sample grabber plug-ins as required, on the same DAW, different DAWs, and even different computers altogether, and the Sample grabber also support surround configurations. The network configuration setup for the Pure Analyzer System is entirely automatic and transparent to the user. The audio transport requires approximately 1.4Mbps for each channel at a sample rate of 44.1kHz, so a properly configured standard Ethernet 100Mbps network should be more than sufficient to handle most cases. A 5.1 configuration at 96kHz demands a little less than 20Mbps. Naturally these requirements do not apply when using both Sample grabber and analyzer on the same hardware.   Pure Analyzer Essential The foundation of the Pure Analyzer System is a standalone application, Pure Analyzer Essential, which can be extended with modules for Live, Metering/Loudness and Surround analysis, performing the graphic analysis rendering using a GPU-efficient 2D/3D graphic engine. The engine is based on the OpenGL standard optimizing the audio-to-visual experience utilizing the GPU on the built in graphics card. The Pure Analyzer Essential application receives the sample feeds from the Sample grabber and analyzes them in real time. The Pure Analyzer provides the option to receive data from several different Sample grabber plug-ins on the network, and analyze the received data in parallel by selecting the actual corresponding Sample grabber to be the current source for the analysis with a simple button click. In addition to this, the input streams from the physical ASIO/Core Audio I/O can also be selected as the current source for the analysis in the same simplistic fashion.   Main Features   Basic metering Peak meter Basic RMS meter Fully customizable scale and ballistics Phase meter Weighting filters ITU-R BS.1770-2 (K-weighting) ANSI A ANSI B ANSI C Real-time spectrum analyzer Standard FFT or unique Flux:: Spectrum representation Block size up to 64 K Configurable analysis window Rectangular Barlett Blackman Hamming Hann Amplitude or power frequency scaling Time-averaging Continuously running Fill/freeze refresh/display cycle User-adjustable length up to 200 frames Max (accumulated) or running mode Display channels’ sum or a number of channels, user-selected User-adjustable fixed range or automatic range Simultaneous display of: Raw, full-resolution curve Smoothed curve with user-adjustable detail setting Peak curve Independently adjustable curve ballistics Peak tracking with user definable frequency range and several display variations Keyboard and mouse frequency-axis zoom Fully-customizable curve visual options Spectrogram Display Color modes Power grading Frequency grading Range Min Range Max Vector Scope Internal processing up to 768 KHz Auto-scaling Fully adjustable display Blending Fading Blurring Scaling Several color modes Static color Power grading Dynamic grading Power + Dynamic grading Nebula stereo (Front projection only) Unique spectrum over space scope, providing real-time visualization of the the audio content in the stereo space-frequency domain. Auto-scaling with user adjustable ballistics Focus Control Linear or logarithmic blending Fully adjustable display Fading / persistence Blurring Scaling Choice of color modes Power grading: color varies according to signal power Dynamic grading : color varies according to signal dynamics Power/dynamic grading : combines both previous modes Frequency grading : color varies according to frequency Channel-selection with surround input content Oscilloscope Multi-channel Zero-crossing trigger

Pure Analyzer System – The Next Generation RTA (Real Time Analyzer) System  The Flux:: Pure Analyzer System is a network-driven modular RTA-system providing an extensive range of analyzer options for any imaginable audio analysis and measurement applications. For maximum flexibility and efficiency, Pure Analyzer presents the option to separate the signal acquisition from the analysis utilizing a Sample grabber plug-in, and in addition to the Sample grabber plug-in, Pure Analyzer also provide I/O support for ASIO and Core Audio. Sample Grabber  Sample grabber is a standard DAW plug-in available in all common formats (VST, AU, RTAS and TDM) that acquires the source and response signals and sends the signal through the network using the ZeroConf/Apple Bonjour protocol. You can insert as many instances of Sample grabber plug-ins as required, on the same DAW, different DAWs, and even different computers altogether, and the Sample grabber also support surround configurations. The network configuration setup for the Pure Analyzer System is entirely automatic and transparent to the user. The audio transport requires approximately 1.4Mbps for each channel at a sample rate of 44.1kHz, so a properly configured standard Ethernet 100Mbps network should be more than sufficient to handle most cases. A 5.1 configuration at 96kHz demands a little less than 20Mbps. Naturally these requirements do not apply when using both Sample grabber and analyzer on the same hardware. Pure Analyzer Live The Pure Analyzer Live provides the elements needed to assist the live sound engineer in speaker array calibration tasks: a delay finder, level, meter, transfer function magnitude, phase and coherence spectra, and impulse response, as well as snapshot facilities.Main Features Multi-Channels reference support (up to 8 with multichannel option) Individual channel selectionSummation modeMulti-Microphone support (up to 16 with multichannel option) Individual channel selectionSummation modeMeters fully customizable to make the matching as close as possible to the norm ColorScalingIntegration timeReference levelHoldSPL SPL unit with calibration toolsWeighting filters ITU-1770 (K weighting)ANSI AANSI BANSI CSignal Generator Pink noiseWhite noiseSineLinear sweepLog. sweepSample accurate Delay Finder Allow very long delay detectionTime translationDistance translationImpulse Response Real time short IR or long up to 10 secondTime averaging with confidence levelAuto RangeZoomPeak tracking with ability to send it to the delay compensationTransfer Function Analysis resolution up to 64KOverlapping controlTime averaging with Custom-able Length up to 200 framesCoherence curveDisplay typeFull resolutionSmoothedBothSmoothing detailColor and pen widthMagnitude curvedB Range (-6/6dB to -48/48dB)Auto RangeVector or Rms ModeDisplay typeFull resolutionSmoothedBothSmoothing detailColor and pen widthPhase curveDisplay typeFull resolutionSmoothedBothSpecific phases curve smoothingdetailThresholdMethodpassesUse coherence informationColor and pen widthglobal color grading optionszoomSnap Shot, unlimited (multi-project support) Transfer functionCoherenceMagnitudePhaseSpectrumSummedChannel independentImpulse ResponseAutomated Sweep Snap shot to take a snap shot with a log sweep even while using pink noiseAverage Snapshot created from any pre-selected snapshots

Pure Analyzer System – The Next Generation RTA (Real Time Analyzer) System The Flux:: Pure Analyzer System is a network-driven modular RTA-system providing an extensive range of analyzer options for any imaginable audio analysis and measurement applications. For maximum flexibility and efficiency, Pure Analyzer presents the option to separate the signal acquisition from the analysis utilizing a Sample grabber plug-in, and in addition to the Sample grabber plug-in, Pure Analyzer also provide I/O support for ASIO and Core Audio.   Sample Grabber Sample grabber is a standard DAW plug-in available in all common formats (VST, AU, RTAS and TDM) that acquires the source and response signals and sends the signal through the network using the ZeroConf/Apple Bonjour protocol. You can insert as many instances of Sample grabber plug-ins as required, on the same DAW, different DAWs, and even different computers altogether, and the Sample grabber also support surround configurations. The network configuration setup for the Pure Analyzer System is entirely automatic and transparent to the user. The audio transport requires approximately 1.4Mbps for each channel at a sample rate of 44.1kHz, so a properly configured standard Ethernet 100Mbps network should be more than sufficient to handle most cases. A 5.1 configuration at 96kHz demands a little less than 20Mbps. Naturally these requirements do not apply when using both Sample grabber and analyzer on the same hardware.   Pure Analyzer Metering The Pure Analyzer Metering provides an extensive set of measurement tools well adapted for Broadcast and Post Production facilities, presenting the metering tools for a vast range of standards such as True Peak, RMS, ITU-1770, EBU-R128 (PLOUD), history and more.   Main Features   All meters are fully customizable and configurable to conform to existing and future standards and recommendations Scaling Ballistics Reference level Peak-hold Cosmetics EBU R128 / PLOUD Momentary loudness Short-term Integrated Range ITU-R BS.1770 True Peak Default and user presets K-System VU Meter DIN PPM Nordic PPM BBC PPM EBU Real-time metering history Fully customizable Scaling Ballistics Reference level Peak-hold Cosmetics etc... Peak True Peak RMS (PPM) Dynamics (crest factor) Momentary loudness Short-term loudness Real-time metering statistics Peak: min, mean, max True Peak: min, mean, max Rms (PPM): min, mean, max Loudness: Momentary, Short, Integrated and Range True Peak incident list with customizable threshold Export to text Offline metering statistics Offline file-processing Multi-files support (with automatic channel type detection from file name) Multi-Reels Up to 16 channels Peak: min, mean, max True Peak: min, mean, max Rms (PPM): min, mean, max Loudness: Momentary, Short, Integrated and Range True Peak incident list with customizable threshold Export to text

Pure Analyzer System – The Next Generation RTA (Real Time Analyzer) System The Flux:: Pure Analyzer System is a network-driven modular RTA-system providing an extensive range of analyzer options for any imaginable audio analysis and measurement applications. For maximum flexibility and efficiency, Pure Analyzer presents the option to separate the signal acquisition from the analysis utilizing a Sample grabber plug-in, and in addition to the Sample grabber plug-in, Pure Analyzer also provide I/O support for ASIO and Core Audio.   Sample Grabber Sample grabber is a standard DAW plug-in available in all common formats (VST, AU, RTAS and TDM) that acquires the source and response signals and sends the signal through the network using the ZeroConf/Apple Bonjour protocol. You can insert as many instances of Sample grabber plug-ins as required, on the same DAW, different DAWs, and even different computers altogether, and the Sample grabber also support surround configurations. The network configuration setup for the Pure Analyzer System is entirely automatic and transparent to the user. The audio transport requires approximately 1.4Mbps for each channel at a sample rate of 44.1kHz, so a properly configured standard Ethernet 100Mbps network should be more than sufficient to handle most cases. A 5.1 configuration at 96kHz demands a little less than 20Mbps. Naturally these requirements do not apply when using both Sample grabber and analyzer on the same hardware.   Pure Analyzer Multichannel The Pure Analyzer Multichannel provides a vast range of measurement tools for the most demanding multichannel environment. In addition to standard spectrum analyzer and metering tools, an innovative set of spectrum tools, Flux:: Nebula, presenting frequency and scope over space in a fashion never seen before.   Main Features   General additions Analysis of up to 16 channels simultaneously Adds Multi-Microphone support for Pure Analyzer Live   Nebula surround (top view)   Unique spectrum over space scope, providing real-time visualization of the the audio content in the surround space-frequency domain.   Speaker mode Supports any type of surround configuration (4.0, 5.0, 5.1, 6.0, 7.0, 7.1 etc...) Music: standard music configuration (LR: -30°/-30°, Rear : -110°/-110°) Equidistant: equidistant speaker configuration Square Theatre: square with addition of rear speaker, for real theatre simulation Auto-scaling and ballistics Choice of linear and logarithmic blending Full control over display Fading / persistence Blurring Scaling Particle count Color modes Power grading: color varies according to signal power Dynamic grading : color varies according to signal dynamics Power/dynamic grading : combines both previous modes Frequency grading : color varies according to frequency Other Inter-channel phase correlation

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.

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.

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.

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.

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