The M-Tron is modelled on the legendary British instrument, the Mellotron. This instrument was manufactured in various guises between 1963 and 1986 and was used by artists including The Beatles, The Moody Blues' Mike Pinder, Black Sabbath, Rick Wakeman and Genesis' Tony Banks. The instrument faded into semi-obscurity during the late 80's and early 90's but due to its ability to impart instant nostalgia, more recently it's been used by Oasis, Radiohead and countless others.
The principle of the Mellotron was simple - take a 36 note keyboard and have a piece of tape for each note containing real recordings of real players and in that way it was arguably the World's first sampler. These recordings varied from single notes (as in the Flutes sound used on Strawberry Fields Forever) or entire musical motifs (as in The Beatles Bungalow Bill). Each sound lasted a maximum of eight seconds which forced the player to adopt a different technique and allowed the music to truly breathe.
However, while the sound was magnificent, because of the huge array of moving parts, the instrument's reliability was questionable and the price was expensive. Purchasing an original M400 now would cost several thousand dollars and would probably only come with the default three sounds, Cello, Flutes and Violins. The M-Tron, on the other hand, costs a fraction of this price and now comes with over 2Gb of sounds that don't require the instrument to be dismantled in order to hear them.
The M-Tron captures all of the character of the original instrument by sampling every note of every key and adhering to the eight second limit of each note. This way, not only do the resultant sounds ebb and flow, all the inherent and nostalgic magic of each sound is closely observed - minor imperfections and all.
Awards
Since its release the M-Tron has won numerous awards including Future Music Platinum Award, Computer Music Smart Buy Award and Mac Format Must-Have Award and is now available as a boxed product from M-Audio or GForce.
Features:
* Over 2GB of sounds (over 100 tapebanks) recorded from real Mellotron, Birotron, Novatron & Chamberlin instruments including strings, choirs, flutes, accordions, mandolins, oboes, brass, vibes, rhythms and many more ultra-rare sounds.
* Volume, tone, and pitch controls
* Attack and release envelope
Suggested Listening
The Beatles - Strawberry Fields Forever (Flutes)
Led Zeppelin - Stairway to Heaven (Flutes)
The Moody Blues - Nights In White Satin (Violins)
Genesis - Dance on a Volcano
Los Endos & Entangled (Choirs) & Watcher Of the Skies (Brass and Bass Accordion Mix)
Yes - Heart of the Sunrise & And You And I (Strings)
Radiohead - Exit Music (Choirs)
Putte - Jubel Buben (Flutes, Harp Glissandos & Cello)
King Crimson - In the Court of the Crimson King & Starless (Three Violins)
Selected Quotes
Paul White - Sound On Sound Magazine : "If you're into retro sounds the M-Tron is a must"
Mac Format : "Classic Mellotron sounds from the sixties and seventies without the price, bulk or technical problems."
Rick Wakeman : "Totally authentic. I use it in the studio and live."
Robin Rimbaud (Scanner) : "I just wrote a new National Anthem for Europe that was presented in Brussels at the weekend and I used M-Tron to create the voices on the main theme - very regal and worked fantastically. Thanks again to you for giving me this great software."
Putte : "The fact that I prefer to listen to organic music instead of electronic music brings me close to the M-Tron. I love its sound and wonderful warmth."
Formats :
Stand-alone or host use-VST 2.0
(PC/MasOS9/MacOSX), AU (MacOSX)
System Requirements :
Windows
Intel Pentium II 300MHz with 96MB RAM
Windows 98, 98SE, Me, 2000, and XP
VST 2.0 or RTAS compatible host application for plug-in operation
Macintosh
PowerMac G3 with 128MB RAM
OS X
VST 2.0, RTAS, or Audio Unit compatible host application for plug-in operation
? Hybrid Vector & Wavesequencer Sampler for VST Hosts ?
? ?
? Wusikstation is one of the most popular ?
? Sampler/Rompler out there. Since it started, if got ?
? more and more features asked by users, a new ?
? interface, easier way to load presets and much more. ?
? Its been rated by users the number one choice when ?
? creating songs
Stereoizer adds 'stereo spread' to a mono signal by splitting the frequencies into bands and then shifting them differently in the R and L channels. Great for adding depth to percussion loops and livening up synth pads, etc.
Features:
* Width and Resolution control for adding fully mono-compatible stereo enhancement.
* Phase Invert.
* Phase Shift.
* Expanding Interface.
* Linear Width Control.
* Mono Compatible stereo Dynamics Control.
* Preset and User patches.
* 24 voice polyphonic (CPU dependent)
* 9 drawbars
* Adjustable Percussion
* Adjustable KeyClick
* Adjustable motor noise
* Spread (play 5 detuned organs simultaneously)
* Distortion that ranges from warm to dirty
* Rotary speaker cabinet simulation
* Fully recognized Velocity
* Full integration in VST environment: Sample accurate timing, full automation and settings are saved with your song
Give your digital tracks a heavenly warmth. Or something a whole lot hotter.
Whether you want to burnish your tracks to an angelic warmth or heat things up with some rich, devilish distortion, Antares's Tube plug-in gives you the tools to create a wide range of modeled analog tube effects.
Based on technology from our multi-award-winning Microphone Modeler plug-in, Tube is incredibly easy to use and so DSP efficient that even a modest native system will support dozens of tracks' worth. (In our informal tests, a Mac PowerBook G4 400MHz was able to provide well over 40 instances of Tube.) And with an astonishingly low price, you won't have to sell your soul to afford it.
blue tube Choose our heavenly blue tube to add the warmth of a classic tube preamp to vocals, acoustic guitar, horns, strings, synth pads, in fact pretty much anything. The Drive control lets you select the degree of dynamic saturation. Or engage OmniTube to add subtle body to the entire signal.
orange tube Give in to the temptation of our fiery orange tube and ignite your tracks with the deep, warm distortion of an overdriven tube amplifier. Use it in moderation to impart a subtle (or not-so-subtle) effect to almost any track. Or crank it up on electric guitar, bass, synths, even voice.
In addition to the tube models, Tube includes a unique OmniTube function. Without OmniTube engaged, Tube functions exactly like a tube preamp, i.e., only the regions of the input signal that exceed the clipping level (typically transients) are affected and all other regions are passed with no change. With OmniTube on, a compressor is inserted into the signal path before the tube model. This compressor is set to compress the signal and then apply sufficient makeup gain to ensure that Tube's Drive control can drive the entire signal above the clipping level. After the tube effect is applied to the entire signal, an inverse gain function restores the signal's original dynamics.
Tube is available now for RTAS (Mac and PC), VST (Mac and PC), MAS, and DirectX. Check it out at your local Antares dealer or download a fully functional demo. Then prepare to make that age-old choice: Good? Or even Better?
FEATURES
* Models the effect of a high-quality analog tube preamp
* Two tube models for a variety of sonic effects
* OmniTube allows applying the tube effect to the entire signal
* Superb sound quality
* Really easy to use
* Extremely DSP efficient - use Tube simultaneously on dozens of tracks
PC RTAS, VST and DirectX
* A PC running Win 98, ME, NT, 2000, or XP (as required by your host program).
* A host program that supports the appropriate plug-in architecture.
Note: Digidesign Pro Tools does not support the Celeron processor. Therefore Tube is not guaranteed to function correctly in Pro Tools on a Celeron-based PC.
Wie eingangs erwähnt, ist daOrgan die ideale Orgel für jeden der ein anschlagsdynamisches Keyboard idealerweise mit Fußpedal besitzt. Es kann auch mit einem Expressionspedal und abgeschalteter Anschlagsdynamik gespielt werden, unterstützt jedoch lediglich ein Keyboard mit bis zu 88 Tasten. Auch wenn daOrgan für meine Ohren klanglich mit B4 und der evb3 mithalten kann, ist das Linplug Instrument für Musiker die nach einer authentisch zu spielenden Orgel suchen weniger geeignet. Die Simulation der rotierenden Leslie Lautsprecher kann in daOrgan sogar in Punkto Beschleunigung Synchron anhand des Taktes genau festgelegt werden. Ein interessantes Feature, das sogar "bei den Großen" vermisst werden muss: daOrgan gestattet die Geschwindigkeitsänderungen der rotierenden Lautsprechern von langsam auf schnell in 1,2,4,8 oder 16 vollen Taktlängen gleichmäßig und synchron zum Songtempo festzulgen. Wem das zu lange dauert der kann natürlich sofort umspringen oder 1/2, 1/4, 1/8, 1/16 sowie 1/32 Taktlänge als Beschleunigungszeitraum verwenden.
Auch die übrigen Einstellungen können sich durchaus sehen lassen - neben 9 Zugriegeln finden sich Regler für Übersteuerung, Anschlagsgeräusch (Click), Vibrato sowie Percussion und sogar dem Motor Brummen der B3. Interessant ist, dass daOrgan bei halben Reglerwert das verhalten der B3 am authentischten simuliert und daher bei vollem Anschlag die Klangeigenschaften von Motorbrummen, Übersteuerung und Rotorverhalten sozusagen mit doppelter Leistung der B3 arbeiten.
Fazit
daOrgan lässt abgesehen von der fehlenden Unterstützung für ein zweites Keyboard und der halbherzigen Berücksichtigung eines Expressionpedals (über Midi Learn an Lautstärke Regler koppelbar) kaum Wünsche übrig. Die Automatisationsunterstützung ist für Cubase als auch Logic vorbildlich realisiert. Wer die Anschlagsdynamik (Schalter Vel) aktiviert, erhält einen durchaus glaubhaften Orgelsound, der sich wie ein Klavier verhält und bedenkenlos mit dem Fußpedal gespielt werden kann.
The Secret of the Mastering Engineer - Bob Katz TC Electronics
Hier gibt es den "Heiligen Gral" der Producer. Viele nützliche Informationen rund um's Thema "Mastering". Ist vielleicht für den ein oder anderen hier interessant.
So, für heute ist schluss, jetzt muss erst wieder der Uploader qualmen. Morgen abend gibt's dann die Stormdrum Kontakt/Intakt DVD für euch.
@Fritzz
Ist doch wurscht, wer's zuerst hat, hauptsache die Orgel landet am Ende hier
Der Keygen fürn M-Tron hat hier funktioniert. zur Not nimmste die FULL Version, die habe ich hie zur Zeit laufen. Da ist nur die DLL zusammen mit dem ganzen Sack von Presets drin. Musst den Krempel nur direkt ins VST-Verzeichnis entpacken.
* New synthesis approach by "PPG legend" Wolfgang Palm.
* Real-time swapping between components of 33 sound sources.
* 97 special pre-analyzed sound sources included.
* 3 ADSRs, 3 synchronizable LFOs, global pitch LFO.
* Modulation section with Stereo Delay and Flanger effects.
* Up to 64 voices, 300 presets included.
* All parameters accessible via MIDI controller.
Stormdrum verzögert sich wahrscheinlich, da der Rapiduploader Scheisse baut, oder aber Rapidshare selbst manche Files fälschlicher Weise schon nach dem Upload als "blacklisted" registriert..?! Naja, "Kommando Rückwärts"... Und nomma... *kotz*
yeah, danke für das ebook !
haste da noch mehr sachen? (mix,mastering,aufnahme...)
vllt. auch auf deutsch ?
Ich habe hier mal noch eine kleine exemplarische Anleitung zum Mastern:
Exemplarischer Arbeitsablauf eines Masterings
Einleitung
Im Folgenden wird exemplarisch der Ablauf eines Masterings beschrieben. Diese Arbeitsanweisungen sind jedoch nur als Vorschlag und nicht als absolute Anleitung anzusehen. Einige Tipps können sicherlich in vielen Fällen helfen oder sind wie beschrieben durchzuführen (technische Bearbeitung wie Normalisieren, DC-Offset etc). Andere hingegen sind nur bedingt allgemein anwendbar, da es verschiedene Möglichkeiten der Bearbeitung (Routing von EQ und Kompressor etc) gibt, je nach Material und Vorliebe. Letztendlich liegt es im Ermessen des Mastering-Engineers, den richtigen Weg zu finden.
Alle Arbeitsschritte werden an Hand einer kurzen Demo-Sequenz erläuter.
Um die Ergebnisse möglicht gut zu hören sollten die Audiobeispiele über Studiomonitore oder Kopfhörer und nicht PC Speaker abgehört werden!
Unser Beispielsong basiert auf einem E-Piano Schema mit einem Drumloop, einigen Percussions, Arpeggiator Lines, einem moogartigen Bass und einer Funk-Gitarre. Trotz dessen es eine ?Rechnerproduktion? ist bei der lediglich die Gitarre eingespielt wurde, hat der Song noch eine recht hohe Dynamik und geringe Lautheit.
Arbeitsschritte
1 die Audiodatei sollte als Audiodatei auf dem Rechner vorliegen. Zunächst diese kopieren und die folgenden Arbeitsschritte mit der Kopie durchführen
2 im Bild zu sehen, das "rohe" Audiomaterial vor jeglicher Bearbeitung
3 die kopierte Audiodatei wenn nötig in die Bitrate konvertieren mit der das benutzte Programm intern arbeitet (z.B. bei Wavelab 32 Bit). Dies ist vor allem wichtig bei Fades und filigraner Musik wie Jazz, Klassik etc.) Noch kein Fades machen!
4 DC-Offset entfernen. Die sogenannten Gleichspannungsanteile sind heute dank guter Wandlerqualität selten geworden. Trotzdem sollte dieser Schritt nicht fehlen
5 Normalisieren: die höchste Pegelspitze (Peak) auf ca. ?0.3db bis ?0.4db normalisieren, nicht höher, da sonst im restlichen Mastering-Prozess nicht genügend Headroom vorhanden ist
6 Denoising: entweder mit Noise Print oder mit Noise Reduction Software, sofern nötig
7 Declicking: mit Software oder in schwierigen Fällen mit dem Stift Klicks und Knackser entfernen. Braucht meistens auch nur angewendete zu werden, wenn von älteren Schallplatten Material überspielt (Tonrestauration) wurde oder Drops Outs z.B. von DATs vorhanden sind.
8 Limiting: es ist durchaus sinnvoll, schon vor der weiteren Dynamikbearbeitung einen Limiter einzusetzen, um musikalisch nicht relevante Peaks zu limitieren und so schon im Vorfeld 3-4 dB mehr an Pegel zu erreichen. Dabei den Threshold so einstellen, dass wirklich nur die höchsten Pegelspitzen bearbeitet werden
9 EQing: Equalizer wenn möglich so einsetzen, dass Frequenzen abgesenkt werden. Anhebungen nur in Notfällen und sehr sparsam durchführen! Im Falle unseres Beispielsongs wurde ein Lo Cut bei 30 Hz gesetzte (darunter spielen sich eh kein relevanten Informationen mehr ab), der Bassbereich bei 67 Hz schmalbandig angehoben (+4dB) und bei 130 Hz etwas abgesenkt. So konnte der Bassbereich etwas ?entmulmt? werden, bedingt durch Überlagerung und daraus entstehender Resonanz von Bass Drum und Bass. Mit einem Hi Shelf bei 10 kHz wurde der Höhenbereich etwas aufgefrischt 8+4dB), ein Exciter oder Enhancer ist kaum notwendig
nachtrag von mir (den Text halt ich für Sinnvoll aber im Bereich Techno sollte man ein wenig herumexperimentieren, vor allen Dingen im Bassdrum Frequenzbereich eurer BD(s))
10 A/B Vergleich: eine Referenzproduktion auf den gleichen Lautsprechersystemen zur klanglichen Orientierung probehören
11 Dynamics anwenden: Kompressor oder ähnliche Dynamiceffekte anwenden; hierfür wurde ein Multibandkompressor verwendet um alle Frequenzbereiche leicht zu komprimieren
11 A/B Vergleich Probehören: den Track auf anderen Lautsprecher probehören
12 Psychoakustische Effekte anwenden: Exiter, Enhancer, Stereobasisverbreiterung oder ähnliche Klangverbesserer anwenden; da der Einsatz gerade von stereobasisverbreiternden Effekten immer eine heikle Angelegenheit ist und der Song schon aufgrund seines Mixes relativ ?breit? klingt wurde auf den Einsatz von solchen Effekten verzichtet. Der Obertonbereich klingt auch recht präsent, ein Exciter ist daher auch nicht unbedingt notwendig.
13 Limitieren: mit dem Limiter die Dynamik begrenzen. Die Reihenfolge von Limiter, EQ, Kompressor ist Geschmackssache und kann unterschiedliche Ergebnisse bringen. Letztendlich sollte am Schluss der Signalkette jedoch ein Limiter sein um einem Übersteuern vorzubeugen! In unserem Fall wurde ein Adaptive Limiter (Loudness Maximizer) eingesetzt, was bei Produktionen mit überwiegend akustischen Instrumenten und entsprechender Dynamik jedoch nicht immer gut klingen muss. Hier sollte man ein klein bisschen Leben erhalten! In unserem Fall bewirkte der Einsatz des Loudness Maximizers, dass Snare und Hi Hat plötzlich recht laut und
scharf klangen. Ein kurzes Nachregeln des EQs bei 3,5 kHz (-3,5dB) brachte Besserung
14 Normalisieren: der fertige Song sollte nochmals normalisiert werden, jedoch auch nur-0,4- 0,2 dB insofern das nach der Dynamikbearbeitung noch nötig ist
15 Sample Rate/ Bit Rate konvertieren: Em Ende der der Bearbeitungskette steht die Anpassung an das verwendete Master. Meist wird auf eine CD-R gebrannt, dann sollte der fertige Track in 44,1 Kbps/ 16Bit konvertiert werden. Ist ein hochwertiger Dithering-Algorithmus (wie z.B. Apogee UV22) verfügbar, diesen unbeding einschalten.
16 Song abspeichern: eine sinnvolle Benennung des Tracks ist vorteilhaft um den Überblick zu behalten
17 A/B-Vergleich mit dem Ausgangsmaterial: dazu die gewonnene Lautheit wieder reduzieren (Leveler am Ende der Bearbeitungskette), damit beide Versionen mit gleicher Lautstärke bewertet werden können, um evtl. Klangveränderungen besser wahrnehmen zu können. Diesen A/B-Vergleich auch nach jedem klang- bzw. dynamikbearbeitendem Schritt (ab 5.) wiederholen. Im folgenden Audiobeispiel wurde das Ausgangsmaterial mit dem Endergebnis zusammengeschnitten. Achtung, es wird nach ca 8 Sekunden recht laut!
Technical specifications
* A VST acoustic drum brain that can jam with you and your music like a human drummer would. Recognizes what and how you play and interacts with you to give you very realistic drum tracks. Steroids for your creativity!
* Built-in high-quality multi-velocity drum sample player with controllable ambience and bounce samples
* Integrated VST sub-host allows usage of any VSTi drum synthesizer, such as BFD?, NI Battery? and DFH Superior?. Combine the sounds of your favorite drum plugin with the jam power and realism of the jamstix brain!
* Extensive, unparalleled jamming capability. Analyzes MIDI input as well as audio input (with limitations) in real-time and develops rhythms and adjusts parameters instantaneously, giving you the feel of playing with a human drummer
* Limb-Priority-Control mechanism ensures that the output is at all times playable by a human drummer. You can layer rhythms and fills and accents as you wish and never have to worry about whether the result sounds realistic.
* Built-in arranger allows easy control of rhythms,intros,fills and endings in sync with your sequencer
* Easy-to-use but powerful rhythm controls allow you to modify or create rhythms. Comes with a library of hundreds of rhythms ready to use
* Choose between free jam mode which creates new rhythms from scratch in real-time or key word jam, which is based on preset rhythms selected through key words
* Incredibly realistic drumming with just a few mouse clicks thanks to features such as velocity variance, auto-cymbals, random accents, probable notes and automatic switch rules, such as ride over hihat on high velocity, sidestick over snare on low velocity, no-ghosting on low velocity etc.
* Extensive pitch and velocity mapping allows the adaptation of any key layout
* Over 20 MIDI-automation parameters allow control of various aspects of Jamstix by the host sequencer
* Four separate outputs with compression. Jamstix supports an additional twelve outputs as pass-through outputs to sub-hosted VSTi that require it.
* MIDI output to host sequencer as well as export to MIDI file for maximum flexibility
Specifications:
* VSTi 2.0 compliant virtual instrument with 4 to 16 separate outputs and MIDI-to-host output
* Sub-Host for a single VSTi (compatibility to specific products is not guaranteed. Use the demo to test prior to purchase) with full recall capability
* Internal sample playback engine with 370MB of drum samples including bounce samples and controllable per-sample ambience
* "Alive" rhythm library with 200+ rhythms and 100+ accents, fills, intros and endings
* Full-featured rhythm editor
* 1000-bar arranger in sync with host sequencer to control rhythm changes, intros, fills and ending
* Flexible Input-Output MIDI mapping with velocity translation to match any VSTi key layout and character
* Metronome with voice count and stick sound
* Supported sample rates: 44.1 - 96 kHz
* Copy protection: License Key
Requirements:
* Windows XP or 2k (Windows ME and 98SE are not supported but should work)
* P3 or Athlon 500MHz CPU (2GHz+ recommended)
* 512MB RAM
* 500MB free hard drive space
* VSTi 2.0 compliant host
Volcano is a versatile filter plugin with many modulation options. It has two independent filters with low-pass, high-pass, and band-pass responses, and 12/24/48 dB/oct slopes. You can choose between five different filter characteristics, from a smooth sounding filter with moderate overdrive to raw self-oscillating over-the-top madness.
With the panning setting on both filters, you can adjust the filter balance on the left and right audio channels, creating stereo filtering effects.
The frequency, peak, and panning settings can be modulated in any way imaginable with the two LFOs and the triggered envelope generator. The LFOs have triangle and square wave forms and can be synchronized to the host tempo.
Other features:
* Interactive filter display that shows both filter curves and lets you drag them around.
* MIDI learn.
* Smart Parameter Interpolation.
* Platform-independent presets.
* Sample accuracy and high sample rates.
CLAS/NR - Compressive Loudness Audio Shaping with Noise Reduction
Our CLAS plugin has been a huge smash with recording engineers and artists alike. But one of the problems with this kind of processing, when used at high relative treble enhancment levels as an exciter, is an increase in high-frequency noise and hiss. This is particularly troublesome for live sessions and old recordings.
To overcome this problem when recording, we have implemented a variant of CLAS that incorporates a sliding-filter Noise Reduction unit ahead of the CLAS processing. Analog implementations of sliding-filter noise reduction have been used in the past to implement Dolby-B and Dolby-C noise reduction encoding. We are doing it here, for the first time, in digital form.
In practice, one simply adjusts the Threshold for noise reduction until, under ambient noise levels, one sees the Noise Reduction meter fully closed down. The two red-bars on the left extend all the way to the bottom. At thresholds just below that point, you will see the noise-reduction bars dancing up and down.
If Threshold is set too far below your actual noise levels, then of course the noise-reduction bars retract all the way up, indicating that it thinks signal is present. The same happens with any noise gate technology.
Conversely, with this kind of noise reduction scheme, setting the threshold too far above the actual noise level will produce "noise modulation" on actual signals - a kind of stuttering sound. That typically happens when your threshold is 20 dB or more above the actual noise level, and it occurs because the sliding-filters are biting into the top of your sound spectrum in an effort to eat that much "noise".
But when adjusted properly, and this is very easy to do with our Noise Reduction bar graph, the sliding-filter noise reduction technique works very well indeed. With Noise Reduction ahead of CLAS processing, you can safely boost the highs without fear of undue noise degradation. As soon as the signal falls away, the sliding-filter slams shut, keeping noise at bay.
As with all of our other VST plug-in products you can try it out for 30 days.
CLAS-NR uses upsampling ahead of the filtering for sample rates below 80 kHz, using a slightly different technique than our PLParEQ, but for the same kind of quality improvements. Filtered results are downsampled again to your system sample rate. The upsampling used is much lighter weight than PLParEQ requiring much lower CPU loading. (Note: the free CLAS plugin does not perform this high SR filtering.)
CLAS-NR uses 64-bit computation throughout, and it is 64-bit ready for Sonar 5. For 24/32 bit systems, the 64-bit results are dithered back to your host wordsize using our sophisticated Gaussian PDF dither.
CLAS-NR performs equally well in stereo or on mono tracks, using autosensing of the host connections. (Some hosts do not advertise Mono inserts properly).
CLAS-NR can accommodate all sample rates.
As with all of our other products, a license entitles you to two installations and lifetime support and access to future updates and product improvements. The cost is $75 USD.
Version 1.53 retunes the integrator gains for best compromise between speed of noise gate action and noise modulation artifacts. The current tuning has these artifacts at well below -100 dBFS when fed with a pure sine tone at -6 dBFS 30 Hz (very loud!).
Version 1.53 also adds a second sliding filter, making a crossed pair, to remove noise at both extreme ends of the sonic spectrum. The high filter is most useful for removing hiss. The low filter is useful for removing rumble, such as wind noise in live mic recordings made outdoors.
Of course, use of noise reduction damages sounds and it should only be used when the results are better than not using it. You can't take out noise without also removing some desirable signal. Most clean recordings should not use noise removal ahead of CLAS. But live recordings can often benefit greatly. For best results set the noise removal threshold as low as you can to achieve the desired improvements.
vocSteady is an "intelligent leveler for vocals", intended to free musicians from the tedious task of automating vocals in the mix.
vocSteady's design causes it to behave less like a normal compressor, and more like an automated fader: it follows the vocalist's dynamics, and attempts to minimize volume differences between phrases, words and even syllables. Lyrics will be easier to understand, occasional bursts (or drops) in energy will be leveled out, and the whole track will feel more "solid".
DiVerSe claims that with vocSteady you'll get a stable and comprehensive vocal track even if your vocalist whispers in the verses and screams in the chorus. vocSteady can be used in situations that need immediate leveling of vocals: Broadcast microphones; Jingle vocalists; Mixes that should be ready by "yesterday"; etc.
Key features:
* An intelligent compression method, optimized for vocals.
* A unique algorithm avoids over-compression.
* One slider operation: no "threshold", "ratio" etc.
* Gate option to get rid of background noises.
* "Dual Mode" for handling extremely dynamic vocals.
* A low-cut filter option, to diminish "poofs" and hums.
* A large and intuitive Gain-Reduction meter with clip indicator.
* Trimmers for release time and output volume.
* Rhino features a huge list of built-in waveforms.
* Up 6 oscillators per voice for a huge sound!
Additive Generator
* The built-in additive synthesizer can be used to define new waveforms.
Ultra-slick graphic envelopes
* Graphical multipoint envelopes with curvature controls.
* Envelopes can be synchronized to tempo.
* Each oscillator has his own level, pitch, and waveshaping amount.
Waveshaping section
* Each oscillator can have his own waveshaper.
* Waveshaper can be modulated by an envelope, keyboard tracking, velocity or aftertouch.
Filter section
* Dual multimode filter, with -12/-24/-36 dB/oct. slopes.
* Use the amazing Rhino envelopes to control the filter and create rhythmic patterns.
Keyboard tracking controls
* Graphical curves for keyboard and velocity tracking.
* Control Envelopes levels and times, oscillators pitch, and many over parameters.
Routing Matrix
* A comprehensive routing matrix, visible at all times, defines the FM settings for the 6 oscillators.
* 2 Ring modulators enhance the power of the FM synthesis engine.
* Oscillators and Ring Modulators can be freely routed through the 2 multimode filters.
Step sequencer
* The built-in 16 steps sequencer allows to generate hypnotic patterns and unique effects.
* The sequencer is synced to the host tempo.
* Separate control over note, length and velocity.
* Random sequencer mode, to emulate the wonderful features of the Jupiter series arpeggiators.
Multi-effects section
* Many effects included (chorus, flanger, phaser, delays, reverbs, ...)
* Up to 2 effects per patch.
* Parallel and serial modes.
Other features
* Rhino includes Big Tick's new, unique user-defined controls system.
* Assignable midi controls.
* Microtuning (using Scala .tun files).
* Includes 64 presets.
* The demonstration version has save and automation disabled.
Phat Pro is an enhanced version of the Phat plugin: a subharmonic low frequency synthesizer, which gives more flexible controls on low frequency content.
It features 6th order elliptic low pass filters, three bass processing algorithms, hard limiter and post effect waveshaper.
Ideal for Dolby Digital / DTS low frequency (LFE) channel control.
DelayPack is a Package with very usefull DelayPlugins for Mac OS X and Windows.
DelayPack is made with Pluggo-MAX/MSP from cycling74. You need to install the Pluggo-Runtime (version 3.5.2 or higher, for UB version Pluggo 3.6.x) to use DelayPack. If you have further questions regarding DelayPack please contact delaypack@codeoperator.com .
DelayPack for OS X and Windows 1.3.0
PlugIns included:
* COSimpleDelay - a delay up to 5s, with Feedback, Feedback-HP/LP and HP/LP
* COSpektralDelay5 - a 5 band spectral delay with feedback
* COModulationDelay - a delay with modulation (flanger/chorus/ringmodulation)
* CODuckingDelay - a ducking delay
* COPingPongDelay - 5 delay-lines with individual pan
* COSpecialDelay - Delay with count and a modulation matrix
* COShapingDelay - Modulate parameter via LFO, Comb-Filter, Waveshaper, Phaser
* COTapeDelay - a Tape delay emulation with waveshaper, 3 tapeheads...
* CODirtyDelay - Delay with LoFi and modulation...
DelayPack history:
changes in Version 1.3.0:
* fixed a rounding error with note-sync
* fixed a problem with toLeft or toRight parameter
* a new and smaller GUI for all plug-ins
* CODuckingDelay:
o use external sources for ducking (via CODPBusSend)
* COModulationDelay:
o add a phaser as modulation modul
* COPingPongDelay:
o fixed a bug with pan
* COSpektralDelay:
o fixed a bug with overlapping frequencies
* COSpecialDelay:
o fixed a bug in the ring-modulator
o fixed a bug with the chorus/flanger level parameter
* COShapingDelay:
o double sin-button in LFO2
* a new delay called COTapeDelay
* a new delay called CODirtyDelay
changes in Version 1.2.1:
* fully compatible with Logic AU Validation
* fixed a bug with Live 5
changes in Version 1.2:
* bugfixes
* improved Flanger/Chorus effect in COModulationDelay and COSpecialDelay
* new parameter Predelay for:
o COSimpleDelay
o CODuckingDelay
o COModulationDelay
o COSpecialDelay
* new delay COShapingDelay
changes in Version 1.1:
* bugfixes
* improvements in sync to host
* you can set the tempo of every plugin by host or manually
* COPingPongDelay allows note sync
* COSpektralDelay allows note sync
* a new delay called COSpecialDelay
M51galaxy features custom SynthEdit modules and hybrid synthesis along with rhythmic algorithmic generators, and can be used for anything from ambient-space to high techno-trance.
Features:
* 2 completely independent synths each with:
o PDO (Phase Distortion Oscillator) with 2*8 waveforms.
o SUB (VA Oscillator) with 7 waveforms and a 12dB/oct SV Filter.
o FM capability for metallic sounds.
o Ensemble, Detune, Mono mode, Portamento, Vibrato.
o Two 8 Stage Graphic Envelopes (AMP and MOD) with save and load file functions.
o LFO (BPM synced) with an unique Arpeggiator.
o Pulsar (BPM synced, creates the algorithmic element).
o Easy to operate 3 * 5 Modulation Matrix.
* FX:
o Stereo Chorus/Flanger.
o Stereo BPM synced Cross-Delay.
o Auto-panner.
o 2 Warps with distortion (no/soft/hard) and resonant lowpass filter. Recordable gestures and load and save file functions.
* Global:
o Keyboard split.
o Virtual keyboard for audition.
o 16 step trance gate.
o Multi-timbrality: 2.
o Polyphony: 6 voices (each synth).
Ratshack Electronic Reverb is a detailed model of the Realistic Electronic Reverb, which was actually an analog delay. Audio Damage have done their best to exactly replicate the sound and quirks of the original including modeling the distortion that occurs by running a line signal in to the mic inputs.
FX Lord is a powerful multi-FX arsenal of sixteen diverse FX types, including classics like Phaser, Flanger, Reverb, Delay (including reverse delay), new school effects such as Psycho-Modulation, Glitch, Bit-Crushing, and even a Granulator, plus a complete suite of distortion FX (including stereophonic waveshaping), feedback FX such as a Comb Filter (with self-signal bleed), and even a cavernous Reverb unit.
Each FX unit has its own wet/dry settings to achieve a perfect balance, or you can turn off the FX units you're not using to save CPU usage. To make things even more interesting NOVUZEIT added the ModTrix (a Modulation Matrix).
Features:
* 16 FX Types.
* Modulation Matrix with 4 modulators using 52 parameters.
* Full Volume / Bypass controls individually and globally.
* Global clipping offsets.
* Supports SEND VST MIDI EVENT and RECEIVE VST MIDI EVENT.
* FX Engine is fully stereo, can convert to mono on the fly (toggle-able).
* 32-bit precision at 44Khz+.
PLParEQ - Our 10-Band Phase Linear Parametric Equalizer
PLParEQ is a 10-band Phase-Linear Parametric Equalizer of the highest quality. Each filter may assume any of many different filter characteristics, and operate in either traditional phase-warping mode, or our phase-linear mode. It uses the same internal DSP core as all of our other high-end products. Audio streams are treated in either stereo or mono. Individual filters can be applied to either or both stereo channels, middle only, or side only.
Phase Linear Operation is achieved by processing your sound in both the forward-time and reverse-time directions through classic filters - all in realtime. This completely removes the phase warping caused by IIR filtering, and applies their roll-off twice. So each filter type becomes two: one for classic IIR filtering, and the other for Phase-Linear operation.
Below is a screenshot from a mastering session by one of our Norwegian Mastering Engineer clients.
(click image for full-size preview)
This PC VST plugin can operate in traditional 32-bit mode (24-bit audio) or as a 64-bit plugin for Cakewalk's new Sonar 5. All internal processing is carried out in double-precision 64-bit floating point, regardless of external host mode.
For audio streams at sample rates below 80 kHz the DSP engine internally upsamples with high-quality Sinc interpolation, applies its filtering, and then downsamples back to your system sample rate.
You can run all of the filters at native sample rates higher than 80 kHz, and forego the internal upsampling conversions. Upper limits on the sample rate (> 96 kHz) are dicted primarily by your computer's capacity, and your need for high quality at the very lowest frequencies (below 100 Hz).
Individual filters can be operated as either traditional phase-warping, or phase-linear. Our algorithms employ blocked processing for phase-linear operation, and produce phase linearity by sending the signal through each filter twice - once in the forward time direction, and then again in the time reversed direction, thereby unwinding the phase back to zero.
The blocks are reassembled using very high-quality windowing and 8-fold temporal overlap. IMD resulting from phase-linear operation has been measured at below -150 dB from peak signal levels. Corner frequencies for filters can be adjusted from 10 Hz to 30 kHz at all sample rates.
Computation proceeds in double-precision floating point throughout the entire DSP core. At the last stage of conversion back to 24-bit audio, we dither with a carefully crafted white-Gaussian dither from our internal 64-bit samples to the 24-bit mantissas utilized thorughout by VST hosts. Our noise floor is typically measured at around -180 dB/Root(Hz). The Gaussian dither distribution is provably more preserving than commonly used triangular distributions.
Our tests indicate that PLParEQ requires about 5-6% of our computer speed capacity, with each additional filter enabled requiring an additional 0.5-1% at the highest quality levels. These tests were performed on a 3 GHz Pentium-4 computer with HyperThreading enabled. Different VST hosts will show varying requirements. Our tests were performed with Mackie's Tracktion 2 as the VST host. Performance on a more modern Pentium IV are likely to show improvements upon these results.
Get the Refined Audiometrics Advantage - The Smoothness of Analog in a Superb Digital Implementation
* Highest Quality Parametric Equalization
* Selectable Phase-Linear or Traditional Filtering
* 10 Bands + 22 Classic Filter Types
* Selectable Stereo, Left, Right, Left+Right-, Middle, Side, or Middle+Side- Filtering
* Output Stereo, Left, Right, Middle, or Side
* Frequencies from 10 Hz to 30 kHz
* Gains from -20 dB to +20 dB
* Q's from 0.20 to 20
* Choice of K-20, K-18, K-17, K-14, or K-12 RMS-Wide Metering
* 7 Different Quality Levels
* Group Enable/Disable for Instant A/B
* Filter Table displays settings in effect for all filters
* Live Graph permits mouse dragging of Frequency, Gain, and Q
* Global Attenuation/Gain control for -20 dB to +20 dB
* Low Rate Audio is Upsampled by 2x, Filtered, and then Downsampled again
* Uses the Highest Quality Data Windowing
* Sinc Interpolation on Upsampling
* Performs 8-fold Temporal Overlap Processing
* Sample Rates from 44.1 kHz to 192 kHz
* Internal 64-bit Processing Throughout
* Samples are Dithered back to 32-bits on Output
* Uses Sophisticated Gaussian PDF Dither
* All Parameters are Fully Automatable
* Measured IMD and Noise Floor Below -150 dBFS
* 64-Bit Capable for Sonar 5
* PC/Windows VST Plugin
Available Filter Types
* 2-Pole Resonant LowPass, HighPass, BandPass, Band Reject, and AllPass
* 1-Pole LowPass and HighPass
* 4-Pole Resonant LowPass and HighPass
* Classic Low and High Shelving
* Type 1, 2, and 3 Oxford-style Peaking Filters
* 3 dB/octave LowPass and HighPass
* 1/F, A-Weighted, B-Weighted, and C-Weighted Filters
* 6-Pole Notch Filter
Version 2.21 is now available, offering better performance, and with a unified build for Intel and AMD platforms.
Licenses may be purchased for the sum of $1,000.00 USD, and entitle you to two installations and lifetime support with free upgrades to the code and documentation. We will be continuing to add additional filtering capabilities to this plugin, making this one of your best values in professional audio mixing and mastering tools.
Physical modelled virtual E-Guitar + AMP + Cabinet + FX
* Emulates 3 different pick-up types with 9 string variants.
* Guitars material and size can be changed (good for acoustic sounds).
* Built in AMP/Cabinet simulation (6 Amp models & 6 Cabinets)
* 16 effects "pedals" (eight pre-amp, eight post-amp) including "Wah Wah", Tremolo, Phaser , Chorus, etc.
* Effect-pedals can be moved around to change the signal-processing order (Drag 'n' Drop).
* Up and down strumming is supported (up, down, alternate and velocity).
* Playing-aids for instant gratification.
* 64 presets.
* The effects section is also included as a separate effect plugin.
SCAMP is a resonant low-/high-pass filter plug-in. It features a separate envelope follower for each of 6 channels, a host-syncable LFO with multiple waveforms, as well as Admiral Quality's special Gliss-quanti feature for creating staircased LFO waveforms from the basic waveshapes. Also supported is full automation and MIDI CC control of all parameters.
SCAMP closely models the actual circuit paths of real world analog circuits, which makes it a processor-intensive effect; expect SCAMP to require processing power similar to the level of a powerful soft-synth.
Glitch is a highly-adjustable, semi-automated, real-time audio manipulation system which allows you to alter your music in a variety of different ways ranging from quite subtle to extremely bizarre. At its heart is a sequencer which synchronizes to the host tempo and slices the incoming audio into user-defined patterns, applying a random selection of DSP effects to each slice.
Features:
* Realtime, zero-latency VST plug-in effect.
* 32-bit internal precision.
* 9 fully adjustable DSP effect modules.
* Resonant filter on every effect with low pass, high pass, band pass and band stop modes.
* Stereo panning, dry/wet mix and volume controls for every effect.
* 64-step effects sequencer with 16 custom pattern banks.
* Almost every parameter can be automated.
* Most main parameters can be controlled via MIDI.
Linear-Darstellung
