Virtual Plug-ins

What makes a hit sound so good? If it’s on the Billboard charts, chances are it was tracked, mixed, and mastered using Waves. Musicians 1 is an outstanding collection containing several of our most coveted processors. Look at what you get: delay, chorus, phasing, flanging, compression, gating, limiting, double-tracking, level maximization, and more. Let Musicians 1 open your ears to new worlds of sound!

Whether he's remixing for superstars or putting out his own GRAMMY-nominated singles, few artists are as adept as Morgan Page when straddling the line between electronic dance beats and pop sensibilities. From chart-topping powerhouses like Katy Perry and Madonna to industry darlings Tegan & Sara and Icona Pop, Morgan Page has been called upon by every brand of hit-maker to paint their songs with new colors and sounds for the club scene.As in any medium, electronic musicians need a palette to work from that can deliver equal parts creative inspiration and technical refinement. That's why Morgan Page relies on Waves EQs, dynamics and effects as his primary set of production tools. Hand-picked by Morgan himself, the Morgan Page EMP Toolbox contains nine powerful Waves audio processors he uses most in his mixing and mastering sessions, ideal for electronic music producers of every shade of the spectrum.Nine powerful plugins for electronic music productionHand-picked by Morgan himself based on the plugins he uses mostIncludes EQs, dynamics and effects for mixing and mastering

With over 100 stellar processors and over 250 component plugins, Mercury features more Waves in one package than ever before! From time-domain effects to equalizers, from dynamics control to noise reduction, from guitar sound modeling to the sound of classic analog components, there has never been such a comprehensive collection as this. The Mercury Collection offers more effects, more channel components, more mixdown tools, more processors than ever before in one place. Finally-the world of Waves in one massive collection. Waves Mercury.

Your tracking is complete and your mixes are good to go. Now let Masters give your recording that competitive edge. With linear phase filters for breathtaking transparency and Waves IDR™ Increased Digital Resolution, you’re guaranteed maximum sound, translucent and clear. If you want to hear what these processors can do, just turn on Top 40 radio. Whatever you hear ― chances are, it was mixed and mastered using Masters.

With over 30 plug-ins covering everything from EQ, reverb, and modulation to compression, gating, and limiting, Waves Live Bundle gives you all the processing power you need to take it to the stage!

From U2 to Pussycat Dolls, fromMary J. BligetoBlack Eyed Peas, from Snow Patrol and Counting Crows to Sheryl Crow and Jellyfish, fromGoo Goo Dolls and Panic at the Disco! to Beck and The Rolling Stones, from John Mayer and Keith Urban to Green Day and Lady Gaga…Iconic Grammy-winning producer/engineer Jack Joseph Puig and his personal palette of tones, shades, and hues have illuminated and empowered a kaleidoscope of projects that have spanned commercial record making. His vivid and vibrant productions have defined and redefined the sound of contemporary pop, rock, country, and urban music, time and time again.

MultiRack is a software host that lets FOH and monitor engineers run multiple, simultaneous instances of the same award-winning Waves plugins used in recording studios and mixing rooms the world over. The world’s best-sounding reverbs, equalizers, compressors, limiters and delays are at your fingertips, without the limitations of hardware and at a fraction of the cost.Features:Take studio sound to the stage and back againQuick and simple setupA fraction of the cost of hardwareFull recall, full controlSet and save presets and snapshots per song and song sectionsAdvanced side-chaining capabilitiesWindows- and OS X-compatibleReal-time low-latency plugin (MultiRack SoundGrid only)Simultaneous processing and recording capabilities (MultiRack SoundGrid only)Plugin chain presets saved on MultiRack can be loaded in StudioRack, and vice versa.

Four-time GRAMMY award-winning mixing engineer Manny Marroquin takes the Waves Signature Series to a newlevel with six hybrid plugins that bring his unique workflow to your studio. Alongside his personalized versions of tried-and-true favorites like EQ, reverb, delay and distortion, Manny introduces the innovative new Tone Shaper and Triple D plugins which are destined to become mixing essentials as well. Combining inspiration from his impressive collection of unique and boutique hardware, together with the fruits of his imagination, the Manny Marroquin Signature Series gives you direct access to the same tools Manny uses to mix hit after hit after hit.Includes:Manny Marroquin ReverbManny Marroquin DelayManny Marroquin Triple DManny Marroquin Tone ShaperManny Marroquin DistortionManny Marroquin EQ

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.

Phoenix II is an update of Crane Song's suite of TDM plug-ins designed to emulate the unique properties of a magnetic tape machine. Phoenix II runs both NATIVE and DSP on Avid's PT 10 AAX format. Phoenix II uses 32 bit floating point math and has a lower noise floor than the original release.Designer Dave Hill has crafted this software with the same attention to detail he used to create the electronics for ATR Service's much heralded "Aria" discrete tape recorders.  His intimate knowledge of analog electronics, as well as decades of experience as a recording engineer has spawned a very useful, musical suite of plug-ins.  The Phoenix II process not only incorporates the nonlinear saturation characteristics created by magnetic tape itself, but also includes the interrelation of an analog tape recorder's record/reproduce electronics and equalization curves.  Phoenix II is a ground-up application derived from HEDD technology, and specifically engineered and optimized for Avid's PT 10 architecture.Phoenix II has five different tape-analog characteristics, The type is selected with a switch for easy comparison between the types, and the brightness is also selected with a switch.  Gold is the position where the color is approximately flat in frequency response, with Sapphire being a brighter, and Opal being a warmer tonality.  A level control determines the amount of the Phoenix II process integrated into the audio signal, and an input trim determines "how hard you hit the tape." Because the DSP process is level dependent, the input level trim control has been improved and an output trim control has been added.   The input trim can also be used to prevent clipping in the rare cases where clipping may be a problem and bringing up the level of a track for an increased amount of color.  This can be useful on material that is hitting close to or at digital zero and on material that has a low recorded level. When the input trim and output trim controls are at 0dB, (no change in gain) and no tape process is being added to the sound, the plug-in is bit accurate, meaning the output exactly matches the input. 

PHOENIX is a TDM plug-in suite for Pro Tools HD that is designed to put analog color into the digital domain by increasing apparent loudness without increasing gain (e.g., by modeling analog tape compression). The result is that added realism that is missing in many modeled synths, EQs and digital samplers. The five different "flavors" of PHOENIX can result in a much more musical and satisfying sound.Each one of the five different TDM plug-ins emulate different tape characteristics. Each one also incorporates a color change button allowing three choices to modify the process per plug-in. The Gold button is the position where the color is approximately flat in frequency response, with Sapphire being a bit brighter and Opal having a warmer tonality. A level control determines the amount of the PHOENIX process integrated into the audio signal, and an input trim determines "how hard you hit the tape." Because the DSP process is level dependent, reducing the input level will also cause a change in sound.The input trim can also be used to prevent clipping in the rare cases where clipping may be a problem. This can be useful on material that is hitting close to or at digital zero. When the input trim is at 0dB, (no change in gain) and no tape process is being added to the sound, the plug-in is bit accurate, meaning the output exactly matches the input.The PHOENIX suite contains five separate plug-ins:Luminescent is the most neutral sounding process of the five.Iridescent has a similar magnetic character, but with a fatter bottom and midrange. This plug-in is the most similar to the tape knob on HEDD-192.Radiant is characterized by a more aggressive compression curveDark Essence is even more aggressive. (The effect is a color with a wider frequency range--when used on a vocal Dark Essence can reduce sibilance problems by increasing the apparent loudness of the rest of thesignal.)Luster starts more gently than the other four processes, but becomes as aggressive as Dark Essence when the process is at full scale.

Peacock can be used on a stereo bus to create the classic vinyl sound.  On individual tracks it is also very useful in creating large fat sounds, smoothing of the harmonic content and making a nice smooth vintage sound on vocals and other instruments.There are three main controls; Harmonic, Dynamic and Color.  The Harmonic and Dynamic controls interact and control the level of the harmonic distortion.  The Color switch changes the character / interaction of the controls and sets the maximum amount of midrange / LF color and the HF compression characteristic.  The Dynamic control is the tricky one, what is does is time modulate the distortion components which are very frequency dependent due to the RIAA curve.  There are two main components that cause time modulation in a vinyl record.  The first is due to the cutter and the playback stylus not having the same shape, this is called tracing distortion the other is tracking distortion due to the playback systems miss-alignment and the inability to perfectly follow the grove in the record. The Dither control adds noise that is the spectrum of record noise.  It modulates some of the internal functions an adds dither to the audio path at a level for 16 bit ditheringColor = Gold is an extremely close match to the test material which is a record and the original wave file.  It has been optimized at 96K but will work fine at other sample rates. The Silver setting is lighter setting and coloring increases in the order of the settings; Silver, Gold, Rich, Fat, Deep.   Silver is also the brightest setting.  The amount of time modulation with the Dynamic control is level dependent and has a maximum range.  It is more of a matter of finding the control position that is optimum.   Pushing it to higher levels creates ugly sounds.  The code prevents this. 

Digital recording is now the standard for professional music production. But many engineers and musicians long for the days of analog recording, and for good reason. Analog Tape Machines have a SOUND. Big, fat, warm, deep, rich, and exciting are just some of the terms used to describe what recording onto a piece of magnetic tape sounds like. Most of all, tape is more musical.To capture the sound of tape is to capture a living, breathing sound that dynamically reacts to the music going on to it. Slate Digital Chief Technology Officer Fabrice Gabriel utilized his electrical engineering degree as a foundation for his comprehensive study of the physical processes that occur when recording to analog tape through a tape machine. What began with science led to Fabrice’s discovery of the seemingly magical properties of tape and inspired him to create his most advanced, dynamic algorithms yet. Two industry-standard machines were precisely modeled. What followed was a year of critical listening and algorithm tweaking, with some help from the expert ears of Slate Digital CEO, Steven Slate.The result is the Virtual Tape Machines plugin. With models of both a 2 inch 16 track from NRG Recording, and a ½ inch 2 track machine from Howie Weinberg Mastering, the VTM will give your mixes an absolutely authentic analog vibe that will take your mixes to the next level. With added dimension, fatness, depth, and warmth, you’ll wonder how you ever mixed without it.Specifications:Mac Minimum RequirementsDual Core Intel Processor, At Least 4 GB of RAMMac OS X 10.6 or laterAU, RTAS, VSTPC Minimum RequirementsDual Core Intel or AMD Processor, At Least 4 GB RAMWindows 7 (32/64 Bit)RTAS, VST

SLATE DIGITAL's TRIGGER 2.0 is the next generation drum replacement VST/RTAS/AU/AAX plugin. TRIGGER has a phase accurate multi layered triggering engine. This means that in one instance of the plugin, you can seamlessly trigger 8 stereo samples simultaneously such as a close mic sample, stereo overhead sample, and a stereo room mic samples. Or, mix many direct mic samples to develop your own custom unique sounds. This multi channel triggering functionality allows the user to recreate the sound of natural drums with real multitracked samples. Each sample layer has parameters for customization such as velocity and dynamic control, attack, sustain, release and independent levels. Other features include 2 detection modes, MIDI in/out, automation of detecting parameters, up to 127 different articulations per instrument, up to 127 velocity layers per instrument and up to 127 alternation hits per velocity layer, and a unique "Leakage Suppression" function. Trigger Platinum also contains a transient triggered gate that is the ultimate tool for cleaning up drum tracks with excess leakage.When using TRIGGER, you will hear tight, phase accurate drum triggering without unwanted mistriggers, flamming, or phasing. Kick samples line up tight with the original kick drums, snare drum transients align perfectly, and tom fills come out exactly like the original tracks. Trigger has the unique functionality to be set in a way that in can automatically be instructed to NOT trigger certain leakages on a drum track using its Leakage Suppression mode. Using the Leakage Suppression technology you can trigger a drum track with 100% accuracy even if this drum track has many loud bleeds from other drums.Trigger comes with three latency modes, and can work with live triggering using a 4ms LIVE MODE. TRIGGER comes stock with a library of the industry standard STEVEN SLATE DRUMS drum samples. Elite mixers such as Chris Lord Alge, Mike Shipley, Dave Way, Jeff Juliano, Jay Baumgardner, Ben Grosse, David Kahn, and many more top names have relied on STEVEN SLATE DRUMS to help enhance the drum sounds in their mixes for years. All the samples in STEVEN SLATE DRUMS are recorded to 2 inch tape and processed with the finest in vintage and modern analog gear. STEVEN SLATE DRUMS include three sample layers including stereo room mic samples from the famous NRG RECORDING in North Hollywood, California.TRIGGER can also load expansion libraries such as Steven Slate Deluxe, Chris Lord Alge, and Terry Date. TRIGGER is the ultimate way to take your drum sounds to the next level without fuss or hassle. With its phase accurate engine, extreme detail, natural response, multiple drum layers, and professional STEVEN SLATE DRUMS sample library, you'll never have to worry about drums again.Instrument LibraryTRIGGER Platinum instrument library contains more than 2.3 GB of high-quality 24 bit samples compressed using TRIGGER lossless sample compression (6 GB uncompressed). TRIGGER Platinum instrument library includes: 46 snares (including 2 high-detailed Deluxe snares with adjustable overheads and 12 alternation hits per each velocity), 33 kick drums (including 2 Deluxe kick drums), 8 sets of toms.TRIGGER EX instrument library contains 10 snares including 1 from the Deluxe set and the "Old Zep" snare, 9 kick drums (including the modeled "Old Zep" and "ToneDef"), and 3 sets of toms. All samples are high-quality 24 bit samples compressed using TRIGGER lossless sample compression. With Trigger's instrument maker, you can also create your own multi velocity sample sets!!DemoThe Trigger demo version has some limitations.In the full version, there is some additional features that you won't get in the demo version:Midi In/Out WAV and AIF loading Additional TCI instruments loading Saving settings with project Preset savingMAC Minimum Requirements Multicore Intel CPU, clock speed of 1.7 GHz or moreDisplay resolution of 1024 x 768 or above2 GB RAM or more recommended2.4 GB free disc spaceiLok dongle (1 or 2)PC Minimum Requirements Windows 7® (32/64 bit)Intel or AMD CPU with SSE2 support, 1.7 GHz or moreDisplay resolution of 1024 x 768 or above2 GB RAM or more recommended2.4 GB free disc spaceiLok dongle (1 or 2)

Virtual Buss Compressors comprises three unique sounding, 100% analog modeled dynamic processors that are ideal for program material. Every nuance of real analog compressors has been modeled; including the exact nonlinear characteristics of their transformers, tubes, VCAs, amplifiers, phase distortions, harmonic distortions, and timing.All of the compressors in the VBC bundle have a variable hi-pass filter AND a mix knob that can be used to control the ratio/percentage of the compressed signal to the dry signal. Each compressor can be used in series, and the series rearranged by dragging and dropping (see information on “The Rack”). Compressing a mix with each one doing a very little amount can sound amazing! And, using the compressors in series within the ‘Rack’ also makes it very easy to A/B/C the compressors on source material. Finally, VBC will come with stock presets made by some of the industry's top mixers.The FG-GREY starts off with a very precise model of the classic British "4000 series console compressor". I've used the hardware model for many years but until studying the unit in the modeling process, I never fully appreciated just how magnificent the piece is. It has a way of being able to tighten, glue, and 'gel' the components of the mix in ways that can go from fairly transparent in auto release mode, to very aggressive in manual mode.This compressor is based on the classic RED faced compressor that has been a favorite of mix legend Chris Lord Alge, who first showed it to me when I visited his studio years ago. I thought it was odd actually. Here was this compressor that nobody else seemed to really use, and it was on a setting at 1.5:1 and the attack looked very fast for a Mix-Buss compressor. This was puzzling because I usually associated a faster attack with a loss of transient punch, but if you've heard Chris's mixes you know that they are the exact opposite of that! So being the classic gearslut, I bought one, (I got the unit with both input and output transformers). The first thing I did was put it on my mix in the settings that Chris showed me... 1.5:1, attack around 10/11 o'clock, auto release, compressing just 2-3db tops. And.... WOW! One of the clearest, punchiest sounding rock mixes I'd ever heard come out of my own speakers. This unit was amazing.I've been lucky enough to hear some of the most classic tube based buss compressors such as the Fairchild 670 and Manley Vari Mu. There is some real magic that happens when your mix goes through a brilliantly designed piece of tube gear. The midrange thickens, the lows get tighter and rounder, and the top end opens up with a beautiful sparkle... and even better, those harsh upper mid overtones seems to get tamed. For the FG-MU, we put all of these wonderful qualities into the algorithm. First and foremost, to hear the FG-MU, you don't even have to do any compression. Just going through the processor without any gain reduction will inhibit a beautiful open sound due to the modeling of the tube circuit path. Even when I'm using the FG-GREY or FG-RED, I love using the VBC Rack just so I can have the audio pass through the FG-MU.While all three compressors are available as separate plug-ins in your DAW, you can also use them in our Virtual Buss Compressors Rack plug-in, in which they can be chained, reordered via drag and drop of the u-rack handles, and soloed independently.During the development of the VBC, I spoke to many top pros in the audio industry about mixbuss plug-in compressors. There seemed to be an overwhelming consensus that even though many of them were advertised as 'analog modeled', they still didn't have the classic sound of analog compressors. So Fabrice and I got to work. First we needed to examine the hardware compressors and the current crop of analog modeled plug-in compressors to see what was going on! And what we found was surprising. The classic analog compressors that we studied added some very complex nonlinear artifacts in their signal paths. Their sound was not simply based on their specific compression topology like timing and compression curves. Some of the analog compressors had modulations, dynamic harmonic distortions, dynamic phase distortion, saturation, crosstalk, and more. And just like in our Virtual Console Collection plug-in, these artifacts added a sonic imprint to the compression. Was this the magic sound that made the plug-in compressors sound different than the analog compressors? There was only one way to find out. We had to precisely model it. All of it!

Despite the digital revolution in the audio industry, many of today's top commercial albums are still mixed on analog consoles. Audio engineers rely on analog mixing to provide the nonlinear musical qualities that digital mixing does not produce. "When you mix through an analog desk you get this life and body to the sound that just doesn't happen when you mix inside the workstation. The separation and imaging from the analog summing is very apparent, especially when your track count gets high" remarks mixer Jay Baumgardner (Papa Roach, Evanescence).Over the past ten years, multiple audio manufacturers have produced simplified analog summing boxes that allow DAW users to get the benefits of analog mixing without having to use a full fledged analog desk. In 2001, Steven Slate made his mark in the analog summing world by commissioning Roll Music Systems to design and manufacture a custom analog summing box that was later named the FOLCROM. The FOLCROM continues to be one of the most popular analog summing boxes on the market, used by mixer Mike Shipley and other top names. Now in 2011, Slate once again stirs up the analog summing world, this time in the digital domain.The Slate Digital VIRTUAL CONSOLE COLLECTION brings the sound of 5 of the world's top analog consoles into your DAW workstation. "Slate Digital CTO Fabrice Gabriel and I studied these consoles inside and out. We meticulously modeled the entire circuit path so that we could recreate every subtle nuance that makes these consoles the legends that they are," says Slate.The VIRTUAL CONSOLE COLLECTION consists of two plugins, Virtual Channel and Virtual Mixbuss. Each plugin allows the user to choose from one of five modeled consoles. Virtual Channel is applied on individual mixing channels. Virtual Mixbuss goes on the first insert of the master fader. When using the Virtual Console Collection, your DAW instantly takes on the personality of a real analog mixing desk. The imaging and depth improves, instruments sit better in the frequency spectrum, and mixing becomes easier and more musical. You can even push the DAW faders up to find each mixer's "sweet spot."Features:Choose from 5 meticulously modeled analog consoles.Includes two plugins that faithfully reproduce the signal path of the consoles channel and mixbuss.Input Level control.Drive control.Sophisticated Grouping System.Global and Group Level Calibration Settings.VU Meter Ballistics.Console Output Trims.Real-time and Offline Oversampling SettingsSpecifications:MAC Minimum RequirementsDual Core Intel Processor, At Least 2 GB of RAMMac OS® X 10.6 or laterAU, RTAS, VSTPC Minimum RequirementsDual Core Intel or AMD Processor with SSE2 Support, At Least 2 GB RAMWindows 7® (32/64 Bit)RTAS, VST

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 VIRTUAL MIX RACK continues the Slate Digital tradition of providing the absolute most authentically modeled analog sound in a digital audio plugin. The ‘VMR’ includes four world class mix modules, including two classic equalizers and two versatile mix compressors, with more modules to be developed.The ‘VMR’ operates as a Virtual 500 series rack with hot-swappable modules, putting your whole processing chain in ONE WINDOW. This allows you customize your own channel strip, with signal flow following the modules, in any order you choose. Mix and match filters from different EQs. Chain an EQ before or after your compressor. Moving modules around is easy, fast, and creates no break in audio playback. Audition signal chains faster, easier, and in real time with the Slate ‘VMR’.The FG-N is a digital recreation of one of the most classic discrete Class A British equalizers from the 70’s. All aspects of the circuit are modeled, including the rich harmonics and saturations that naturally occur when the equalizer is in use. The original model only contained one mid band, so we thought it would be nice to double it so that it has two mid bands. This equalizer is lush, fat, and bold. It sounds great on everything from drums to vocals, and can even add the perfect saturation to synth and electronic tracks.The FG-S is a digital recreation of one of the most famous British console equalizers from the 80’s. This equalizer also has rich harmonics and interactive mid bands with a fat musical tone. This is the ultimate workhorse mix eq that can be used on any track. It is great for surgical cutting strokes, or wide bell and shelf strokes for making drums and guitars punch out of the speakers.The FG-116 is the most precise digital replication of the classic American FET limiter. No expense was pared to capture every nuance of this classic sounding compressor/limiter. From its trademark timing characteristics to the extremely musical sound of its transformer, the FG-116 will give engineers the musical and fat tone that has made the hardware so famous. The FG-116 sounds great on everything like lead vocals, drums, and bass guitars.The FG-401 started as a recreation of the classic british console channel compressor, but morphed into quite a lot more. First, we gave it variable attack and release with increased range which added more tonal options. But then, we added an optimal transformer inout, and not just any transformer, the famous transformer from the British Class A Console!This adds a beautiful warmth and sheen to the compressor. Last, we added an entirely unique second circuit path that provides and additional smooth and rich tone. The FG-401 is perhaps the most versatile mix compressor in the world that can sound good on any source you feed into it!

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)

The Oxford Plug-ins have earned a world-class reputation for their high-quality processing, precision control and intuitive graphical interfaces. The Sonnox Elite Collection contains seven powerful mixing plug-ins used by countless professionals all over the world.Don’t waste money or time collecting Oxford plug-ins one by one, save a massive 40% off the individual prices, by purchasing them as the Sonnox Elite Collection.Contains: Oxford EQ, Oxford Dynamics, Oxford Inflator, Oxford Transient Modulator, Oxford Reverb, Oxford Limiter and Oxford SuprEsser

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

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.

VERB Session is based on the same fine technology used in the acclaimed IRCAM VERB algorithmic room acoustics and reverberation processor, with a modular construction employing a recursive filtering reverb engine, reproducing and synthesizing the specific acoustical characteristics of any spatial sound environment, developed utilizing the results of over a decade of intense research by the Acoustic and Cognitive Spaces Team at the IRCAM R&D centre in Paris.Tailored for simplicity, with a fast paced workflow well-suited for situations where the perfect result has to be achieved within seconds, VERB Session presents the ultimate solution whether you are a seasoned session engineer or a demanding broadcast and post-production mixer.

Complete mastering solution by Flux, containing five plugins for dynamic range adjustment and tonal balance.Solera implements 4 different detectors and envelop generators in parallel and simultaneously.Compressor, de-compressor, expander and de-expander which can all be independently contolled. In addition, here are some of the key features that make our plug-in the best available for dynamic processing in the digital domain : An analog like metering. RMS values are more useful than dB-fs when setting a dynamic processor. The "Angel's Share" parameter takes into account the signal for intelligent relaxation of the compressor action. The "Hysteresis" parameter allows compressing and de-compressing, independently of the sound level and can be mixed with the standard compression scheme. The Auto Release features an unique algorithm that avoids typical pumping effect. A delay line can be inserted in the signal path to produce a null attack time. The A/B Compare and Morphing section enables ultra fast and precise operations. Dry Mix function to achieve the full control of the dynamic processing amount.Epure features for every section: Individual bypass. The filter type can be set to low cut, low shelving, peak, hi shelving, hi cut The gain be set from -24 dB to +24 dB The frequency range covers the whole bandwidth The Q factor can be adjusted from 0.1 to 1Epure also features five unique functions to make equalization even more easy: Copy A and Copy B duplicate the twenty settings of a slot to the other in one click X2 and /2 affect the gain value for every equalizer section Invert turns the positive gain values to negative ones and vice versaEpure includes internal Input /output routings. A channel group management allowing to set many channels or individual channel setting is also available from the new setup window. From this window, up to 4 MS encoders and up to 4 MS decoders can be engaged if required. Combined with our exclusive A /B morphing system, these functions enable a global and efficient control of the equalization curve in a second. Pure Limiter is supposed to be the very last stage of your audio processing chain. It uses the exclusive FLUX technologies to generate a release envelop producing no artefact on the processed sound.A transparent limiting dramatically increasing the average audio level can now be achieved without damaging the perceived audio quality.

A comprehensive toolbox with the elemental components needed to take control over the dynamic processing when recording, mixing or mastering. Pure Compressor produces a wide range of compressions from ultra clean subtle compressions to classic heavy pumping ones. It's up to your artistic choices not to the technology limitations.

 Pure Expander produces a wide range of expansion process from subtle expansions to hard noise-gate.
 Using Flux:: proprietary Angel's Share and Hysteresis algorithms Pure Expander allows you to remove unwanted noises or reverberation without adding a processed character to your sound. Pure Limiter is using exquisite proprietary Flux:: algorithms generating a release-envelope producing no artifacts on the processed sound achieving a transparent limiting. A dramatic increase of the average audio level is easily achieved without damaging the perceived audio quality of the material.

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.

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.