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Come Follow me on SoundCloud---Free Downloads [05 Sep 2011|02:24pm]

Come check out my latest remix and scroll to the the older tracks for some free downloads. Please check them out and follow me on soundcloud for future song releases and free tracks.
Here is my latest remix featuring Carley Roxann
Phoenix Taylor

Lucky Date Feat. Carley Roxann - Mr Right Now - Atom West Remix by Atom West

Catch up with me on http://soundcloud.com/atom-west or http://atomwest.tumblr.com
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2 new tracks from me [01 Sep 2011|05:31pm]
download links:

Once Upon in Chinatown

Dont Panic
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[dystopiaq020] Various - Dystopiaq 3 [07 Jun 2011|11:15am]


Artist: Various
Title: Dystopiaq 3
Release: dystopiaq020
Date: June 7, 2011
Genres: Electronic, Ambient, Synthpop, Broken-Beat, Microtonal, Trip-Hop, Breaks, IDM, Rock, Improvisation, Hip-Hop, Avant-Garde, Experimental

Download at Archive.org.

01. Azotic Compounds Laboratory – Eclipse (3:36)
02. Nick R 61 – Mulatto (4:27)
03. Fature – Mindstream (5:02)
04. Flaccid Redux – |<0tb (edit) (2:35)
05. catchers of mad wreck – 3 palms 1 finger (4:15)
06. Ambient Anonymous – Perpetual War (5:12)
07. Incentive – Solecism (6:18)
08. Zombie Ninja Schmoe – it’s more like a bad dream (0:38)
09. Tucker the Circuit – Mission Panther 236 (4:21)
10. O.B.B.B.M:\/ – Italotillian Schizochic Garmet District (7:48)
11. ShakySuperFly – Lazy, Thoughtless Music (2:01)
12. Scissor Shock – psychic existentialist (6:02)
13. je.vi – Escape Rober Steemp (2:27)

Produced and Compiled by Incentive
Artwork by je.vi

Creative Commons License
Dystopiaq 3 is licensed under a Creative Commons Attribution-NonCommercial-ShareAlike 3.0 Unported License

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[dystopiaq015] Mzai - Marionette [22 Feb 2011|12:30pm]


Artist: Mzai
Title: Marionette
Label: Dystopiaq
Release: dystopiaq015
Date: February 22, 2011
Genres: Trip-Hop, Glitch, IDM
Download: Archive.org, Dystopiaq

01 Terminal
02 Save from your heart
03 Scarlett
04 This day
05 Story about girl
06 Slow motion
07 Story about girl (Dub version)
08 I was made for you
09 Marionette
10 Miracles
11 Love

Creative Commons License
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[dystopiaq014] Chill'EM - Metamorphosis [18 Feb 2011|08:58pm]

Artist: Chill’EM
Title: Metamorphosis
Label: Dystopiaq
Release: dystopiaq014
Date: February 15, 2011
Genres: Psychedelic, Dub, Ethno
Download: Archive, Dystopiaq

01 bis’ cause (3:02)
02 liquid night dub (11:01)
03 illusion dreams machine (5:13)
04 2dub (8:27)
05 driven tools (3:08)

Total Runtime: 30:51
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[dystopiaq013] Incentive - Sonnenfinsternis [21 Dec 2010|08:11pm]


Artist: Incentive
Title: Sonnenfinsternis
Release: dystopiaq013
Date: December 21, 2010
Genres: IDM, Minimal, Technoid
Download: Archive, Dystopiaq

01 Tribal Vengeance
02 Sonnenfinsternis
03 Kandahar I
04 Kandahar II
05 Fat Man
06 The Complex
07 Pig Song
08 Mad Poison
09 Billy Joel
10 Kandahar III
11 Junk Tracks
12 Zero Logic

Creative Commons License
Sonnenfinsternis by Incentive is licensed under a Creative Commons Attribution-NonCommercial-ShareAlike 3.0 Unported License

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dystopiaq007 - Incentive - UMASA [25 May 2010|03:41pm]


Artist: Incentive
Title: UMASA
Release: dystopiaq007
Date: May 25, 2010
Genres: , ,
Download: Dystopiaq.com, Archive.org


Mastered by Robert Lamont
Additional Programming by je.vi (Track 2)
Creative Commons License
This work by Incentive is licensed under a Creative Commons Attribution-NonCommercial-ShareAlike 3.0 Unported License
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dystopiaq006 - concrete swords - Unbalanced EP [18 May 2010|12:15am]


Artist: Concrete Swords
Title: Unbalanced EP
Release: dystopiaq006
Date: May 18, 2010
Genres: Dubstep, Minimal, Techno
Download: Dystopiaq.com , Archive.org

01 Gobs End
02 Fifty Four
03 Unbalanced

Creative Commons License
Unbalanced EP by concrete swords is licensed under a Creative Commons Attribution-Noncommercial-Share Alike 3.0 United States License

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Remix Contest [07 Apr 2010|02:51am]

[ mood | cheerful ]

Your chance to have YOUR MUSIC HEARD BY MILLIONS of people around the world, and gain international recognition as a leading remix artist!


Lena Katina, the star of the Russian pop sensation, t.A.T.u, launched a remix contest on Kroogi.com. You can put your spin on several t.A.T.u. songs. The winning tracks will be featured on the group’s upcoming remix compilation release.


Track stems are available here: http://tatu.kroogi.com/content/show/653951


The deadline for the remix submissions is May 1, 2010. Hurry! ;)


Good Luck!

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91 Free VST Effects Plugins - PC [08 Jan 2010|10:01am]


Did we miss your favorite free plugin? Would you like to comment on any of the plugins in the list? Please leave a comment below.

1. BootEQ MKII

"The BootEQ mkII is an equalizer and pre-amp simulator combination which provides some nice and musical signal coloration effects."

2. Density MKII

"Density mkII was primarily designed to work in a typical stereo audio group mixing situation or for summing and to glue all things together in an unobtrusive way."

3. NastyLF

"NastyLF a offers push/pull style low-end EQ with an added saturation stage. EQ levels and frequency can be dialed in variably as well as saturation amounts and a rather modern or vintage curve behaviour."

4. NastyHF

"NastyHF offers a really nice and musical vintage style HF shelf with switchable narrow or rather wide curve behaviour."

5. Nasty Tabletop

"NASTYtableTop is an easy-to-use low-end signal saturator."

6. Rescue

"The Rescue plug-in is a gadget which offers envelope dependend signal amplification and saturation in a mid/side configuration. In ANALOG mode it offers some addtional signal alterations similar to hardware. It comes in two versions: Rescue and RescueAE (anniversary edition) – the latter one performs interanl oversampling and adds an output limiter function."

7. TesslaSE

"TesslaSE adds some gentle and pleasant saturation effects to the incoming audio."

8. TesslaPRO

"TesslaPro is a signal saturator with added transient control. It works entirely oversampled and provides smoothest saturation effects."

9. epicVerb

The “epicVerb” reverberation device aims at both: Tight small room and ambience effect simulation well suited to modern drum and vocal productions up to large “epic” halls as known from high quality outboard gear.

10. epicVerb vstpresets

C4 vstpresets with different dry/wet mixes – use this when using eV on the insert bus (files provided by susiwong)

11. RescueAE preset bank

The original Rescue factory bank for RescueAE (provided by nitex)

12. FerricTDS

"This device simulates some dynamic effects as can be obtained with some high quality tape gear."

13. Modern Compressor

Power On/Off button. Gate On/Off button. Gain Reduction meter include. Post-gate control (-96 to -30 dB) Gate time control. Threshold control (0 to 48 dB). Attack time control (1 ms to 50 ms). Release time control (60, 125, 250, 500ms, 1, 2, 4sec). Ratio switch (1.5:1, 2:1, 4:1, 8:1, 16:1, 32:1 and Tank mode). Side-Chain high pass filter control (off to 300 Hz). Mix control (0 to 100 %). Gain control (0 to 30 dB).

14. Modern Equalizer

EQ In/Out button. Phase flip In/Out button. High-pass filter control (off, 60, 120, 180, 240, 300, 360 Hz). Low-freq control (60, 80, 100, 150, 200 Hz). Low gain control (-18 to +18 dB). Low Peak/Shelf modes button. Med-freq control (500, 1200, 1800, 2400, 3200 Hz). Med gain control (-18 to +18 dB). Med bandwidth control. High-freq control (3000, 6000, 9000, 12000, 16000 Hz). High gain control (-18 to +18 dB). High Peak/Shelf modes button. Mix-Ratio control. Output control (-18 to +18 dB).

15. Modern Spacer

Power On/Off button. Stereo level meters include. Pre-Delay control (off to 2 secs). Time control (off to 10 secs). Room Depth control (15 to 3000 m). Room Width control (20 to 300 m). Mix control. LS, MF, HF, HS controls for Reverb EQ. Output control (-18 to +18 dB).

16. Modern Exciter

Power On/Off button. Stereo level meters include. Extreme Low, High controls. Balance control. Output control (-18 to +18 dB).

17. Modern De-Esser

Power On/Off button. Gain Reduction meter include. Threshold, Ratio, Width, Release controls. Freq control (3000 to 9999 Hz). Output level control (-15 to +15 dB).

18. Modern Limiter

Power On/Off button. Gain Reduction light include. Link/Split modes switch. Input control (0 to 20 dB). Release control (30 to 2000 ms). Limit range control (0 to 96 dB). Output control (0 to -20 dB).

19. Modern Lost Angel

Classic LA-2A® Compressor clone. Power On/Off switch. Gain Reduction meter include. Peak control (Threshold, 0 to -48 dB). Gain control (0 to 48 dB). Attack time (Variable, 0.1 to 10 ms). Release time (Variable, 0.05 to 3 secs). Comp/Limit modes switch.

20. Modern Seventh Sign

Classic 1176LN® Compressor clone. Power On/Off switch. GR/VU meter mode switch. Input control (Relative threshold). Threshold level (0 to -40 dB). Ratio switch (4:1, 8:1, 12:1, 20:1 and Crush). Attack time control (0.03 ms to 10 ms). Release time control (50 ms to 2 sec). Output control (0 to 40 dB).

21. Modern Fire Chainer

Classic Fairchild® 660 Compressor clone. Power On/Off button. Input Gain control (1 dB steps, 0 to 20 dB). Threshold control (Variable, 0 to 30 dB). Time constant selector. Attack time (0.2, 0.2, 0.4, 0.8, 0.4, 0.2 ms). Release time (0.3, 0.8, 2, 5 secs, auto 1, auto 2). Ratio (Variable, 1:1 to 20:1). DC Adjust control (comp to limit). Metering switch (VU, GR, Off). Output Gain control (0.5 dB steps, 0 to 20 dB).

22. Modern Black Dragon

Classic Pultec® EQP-1A Program Equalizer clone. EQ In/Out button. 16 KHz cut In/Out button. Low shelf freq selector (20, 30, 60, 100 CPS). Low shelf boost control (0 to +16 dB). Low shelf attenuate control (0 to -20 dB). High peak freq selector (3, 4, 5, 8, 10, 12, 16 KCS). High peak boost control (0 to +20 dB). High peak bandwidth control. High shelf attenuate control (0 to -18 dB). High shelf freq selector (5, 10, 20 KCS). Trim level control (-15 to +15 dB).

23. Modern Deathcore

Classic Distressor® EL-8 Compressor clone. Power On/Off button. Gain Reduction meter include. Input level meter include (-12 dB detect). Input control (Relative threshold). Threshold level (0 to -36 dB). Ratio switch (1, 2, 3, 4, 6, 10, 20:1 and Deathcore). Side-Chain Filter modes switch (HP, LP, BP). Pre-Coloring modes switch (HP, DS, HPDS). Attack time control (0.03 ms to 10 ms). Release time control (35 ms to 3 sec). Output control (0 to 36 dB)

24. Modern Premier

Mic preamp simulator. Signal In/Off button. Monitor On/Off button. Input control (-18 to +18 dB). Level control (Mix 0% to 200%). Transparent control. Filter control. Five preamp model selector. Harmonic control.

25. Modern Amplifier

Power On/Off switch. Gain Reduction meter include. Hidden OLED level meters include. Gain control (0 to 50 dB) Peak Reduction control (0 to -50 dB). Fast/Slow modes switch. Compress/Limit modes switch.

26. Modern Deep Purple

Signal In/Off switch. LCD display include. High-pass filter control (20 to 200 Hz). Low-pass filter control (22 KHz to 12 KHz). 3 Band Program EQ (LF, MF, HF). Gain, Freq, Q controls. Trim level control (-18 to 18 dB).

27. Modern Analoguer

Analog tape simulator. Power On/Off switch. Standard/Monitor modes switch. Hidden OLED level meters include. Input control (0 to +18 dB). Output control (-18 to 0 dB). Color control. Level control. Filter control. Ceiling control(+6 to -6 dB).

28. Modern Flash Verb

Process On/Off button. Send mode On/Off button. Input control (-6 to +6 dB). Level control (Relative reverb time). Reverb time (off to 10 secs). Room control. Room size (box to infinity). Pre-Delay time(off to 2 secs).

29. MCompressor

"MCompressor provides standard compression with volume maximization and has adjustable compression shape, which gives you power to set custom smoothing or even interesting sound effects."

30. MEqualizer

"MEqualizer is a 6-band equalizer with increadibly fast and easy controls."

31. MRingModulator

"MRingModulator performs standard ring-modulation effect using two oscillators."

32. MAnalyzer

MAnalyzer is a FFT based analyzer for audio frequency content displayed as 1/3 octave bars or FFT curve. Allows comparisons, magnitude normalization, ". "averaging...

33. MPhaser

"MPhaser is a traditional phasing effect plug-in."

34. MLimiter

"MLimiter performs smooth saturation in a standard tube-like way improving clarity of the resulting sound. It can also be used as a distortion module for guitars and other instruments."

35. MWaveShaper

"MWaveShaper is a traditional wave-shaping plug-in. Unlike other plug-ins it has an adjustable shape instead of a few predefined shapes."

36. MAutopan

"MAutopan is a traditional automatical panner with adjustable shape."

37. MVibrato

MVibrato is a traditional vibrato with adjustable shape.

38. MTremolo

MTremolo is a traditional tremolo with adjustable shape.

39. MStereoExpander

"MStereoExpander provides stereo field modification to improve or smaller clarity of differences between channels."

40. Dasample GlaceVerb

"GlaceVerb is Dasample's first attempt to implement its works on Residual Vector Modulation (RVM) in a VST plug-in. RVM is a proprietary algorithm developed to calculate the deformations, the vibrations and the acoustic response of surfaces and materials."


"T-SLEDGE is a multi-band compressor for mastering, equipped with peak limiter, level maximizer and 4 compressors/expanders/limiters. And it is equipped with 2 kinds of dividing filter IIR/FIR. So, it also operates as a dynamics EQ."

42. Classic Auto-Filter

"The Classic Auto-Filter is an analog modeled four-pole filter with resonance control. It can be used for creative effects like auto-wah-wah and filter sweeps, as well as a fixed filter for equalizing purposes. The filter is selectable between low-pass, high-pass, band-pass and band-reject (notch). The filter cut-off frequency can be modulated from the envelope follower and the LFO."

43. Classic Chorus

"Classic Chorus is a great sounding and easy to use classical Chorus VST plugin. Nice Plugin for adding Depth, Fatness and Stereo perspective to almost any Electrical Instruments or Vocal."

44. Classic Compressor

"Classic Compressor is a classic analog style VST Compressor Plugin with a lot of warmth and punch. Special designed to use on individual Instruments and Vocal, but also useable on your final mixes."

45. Classic Delay

"The Classic Delay emulates three of the most commonly used delays / echo sounds types: Tape, Analog, and Digital, but without the introduction of noise. A quality control is included where the sound of the echo machine can be adjusted to give the preferred amount of “imperfection”. The delay can be synchronised to the host, and the feedback can be reversed to give a ping-pong effect."

46. Classic EQ

"Classic EQ is a 7 Band Stereo Equaliser with a warm analog sound, well suited to make non-surgical tonal corrections on all instruments, vocals and final mixes. The passive and additive structure, together with unique “Warm” and “Saturation” algorithms, produces warm and pleasant sound, just like some of the most expensive vintage gear. The left and right channels can be adjusted individually or linked together."

47. Classic Flanger

"Take a trip back to the mid 80's with our Classic Flanger. It will give you that subtle or massive swirling effect that is used mainly for guitar, but can be used for many other instruments as well. Due to a tightly controlled feedback path, it is possible to boost the flanging effect to maximum without getting unwanted side effects."

48. Classic Master Limiter

"The Classic Master Limiter VST plugin is specially designed to boost the overall level of your final mixes, but is also highly useable on very dynamic instruments. With just one control on the front panel operation is as simple as it gets; just turn the Threshold down and hear how your mixes gets louder and louder. Very high compression ratios can be obtained without changing the balance of the mix."

49. Classic Phaser

"The Classic Phaser emulates analog phasers from the 70’s and 80’s. These vintage phasers got their characteristic sound from the analog phase shifting filters used, unlike the delay lines used in most modern digital phasers. To get a deeper and fatter phasing effect, more filter stages were cascaded. The Classic Phaser works just like an analog phaser, and with selectable filter stages from 4 to 12, it can resemble anything from phaser pedals to expensive studio phasers from that time period."

50. Classic Reverb

"Classic Reverb is a nice and smooth reverb that does a good job on almost any instrument. It can be adjusted to sound like most any acoustic space, ranging from a small shower to a large concert hall. With the "Hi-Damping" control, it is possible to change the sound from bright and sibilant to a more natural / warmer sound."

51. Endorphin

"dual-band stereo compressor"


"BLOCKFISH is a versatile compressor with countless ways to shape the incoming audio. Unlike many other digital compressors, BLOCKFISH has 'soul'. It likes to bite, but it can be tamed easily once you've learned how to use its potential."


SPITFISH is a dead-easy to use de-esser, aimed mainly at mono or stereo vocal tracks. Just like classic analog de-essers, this unit dynamically filters out harsh, annoying s-like sounds that would otherwise 'spit' in your face.


FLOORFISH is a simple expander / gate device, easy to set up and offers a broad variety of processing, ranging from slight expansion, backgound noise attenuation to extreme gating effects. What's different from similar units is the ability to scan the source and adjust the transistion curve accordingly.

55. dominion

"Signal modelling device that contains: envelope shaper, tube saturation emulator, exciter and at the end of audio chain is a compressor/ limiter for avoiding clipping."

56. Normalizer

"Normalizer – denormals eleminator. Denormalisation is an issue with some audio hosts that don’t switch off the plugin’s audio task when the song is stopped or a track contains silence (or with instruments: no key is pressed while a sound decays)."

57. AmpliTube® Duo LE

This guitar-based plugin is normally not free, but comes bundled in the free version of Sonoma RiffWorks T4 (for Mac and Windows), which you can download fro the link above. Studio Devil BVC (below) is also bundled with the software.

58. Studio Devil BVC (British Valve Custom)

"They offer several AWESOME plugins targeted for guitar usage, some commercial,but this one is free. British Valve Custom is based on the gain structure of popular British Valve (Tube) Rock amplifiers (like the infamous Marshall tube heads) with some subtle modifications to give it slightly more gain. This amplifier model has considerable bite and attack making it perfect for classic rock and hard rock sounds. It also works great with neck pickups for blues, rock, and even metal guitar solos."

59. Red Skull Distortion

"The Red Skull plug-in is an aggressive high-gain distortion pedal simulator with unbelievably chunky metal tones for playing Nu Metal, Grind Core, Hard Core, Trash Metal or any genre that requires crushing sound."

60. Fuzz-Stone Distortion

"Fuzz-Stone is the second plugin in Freetortion Series. The Fuzz-Stone is a ‘60’s styled guitar fuzz tone pedal which simulates the Legendary Stones’ Satisfaction guitar tone."

61. Fuzz-Stone[Ge] Distortion

"Fuzz-Stone[Ge] is the 3rd plugin in Freetortion Series. The Fuzz-Stone[Ge] is a great Fuzz style Germanium transistor distortion simulator."

62. California Sun

"The California Sun is a classic guitar combo amplifier simulator with two channels (Clean and Dirty). "

63. Whamdrive Distortion

"Whamdrive Distortion/Pitch Shifter – This looks exactly like Digitech Whammy pedal… and sounds and reacts quite similar. Whamdrive is a VST plug-in to shift the pitch of an incoming audio signal. It allows you to pitch an incoming audio stream in realtime with an external MIDI foot controller or with the GUI controls. It Offers a cabinet simulator, like Digitech pedals."

64. Tom Pong

"Tom Pong is a ping pong delay VST Plugin. Windows only. A ping pong delay is a delay which alternates from one speaker to another. If the balance is set to 0.5, a stereo signal will invert itself each time the delay buffer is played back. If the balance is set to either side of 0.5, the signal bounces back and forth between the speakers. Tom Pong can be synced to the host sequencer, or delay times can be set by sample count. Feedback and output level are also adjustable."

65. Wow & Flutter

"A VST plug-in which emulates the playback speed imperfections of the pre-digital era."

66. Echolive

"I built this plug-in for a live performance at the Bazillus Club in Zurich. It features a tap tempo function which makes it easy to create echoes that fit the timing of a live band. "

67. Dub Siren

"Emulates the sounds of those selfmade siren units used by Dub and Reggae Sound Systems like Jah Shaka, Aba Shanti and countless others."

68. Tape Delay

"On user request I made the tape delay from the DUB SIREN available as a separate plug-in."

69. Analogic Delay

"This plug-in mimicks the tape delay in Logic Audio. I created it on request from a friend who switched to another sequencer software after Logic's discontinuation on the PC platform."

70. Bionic Delay

"This enhanced version of the Analogic Delay gives you separate control over the delay times on left and right channel. suitable for ping pong delays and drifting delays."

71. Echomania

"Besides being a CPU hog this plug-in goes further than most (all?) real tape delay machines. It's capable of stereo delays, ping pong delays, multitap delays and complex delay textures."

72. ADT - Artificial Double Tracking

"The ADT technique was developed at Abbey Road Studios by engineers recording the Beatles in the 1960s. To free John Lennon from having to sing everything twice for real double tracking they came up with an artificial replacement: they sent the original signal to another tape machine and re-recorded it. Due to the physical distance between record and playback heads the new signal was delayed. The length of the delay depends on the tape speed (the slower the tape is running the longer it takes for the signal to travel from the record to the playback head). However, due to the machine's (small amounts of) Wow and Flutter the delay time was not fixed but varied slightly, giving an additional chorus-like effect."

73. Poor Plate - Stereo Plate Reverb

"Plate reverbs have been among the first devices for artificial reverberation. The first digital reverb units by EMT and Lexicon tried to model plate reverbs instead of real spaces. The Poor Plate is a Plug-In for all those people who can neither afford a real plate nor one of the classic digital units."

74. RIAA - Phono Equalization

"This Plug-In provides normal and inverse RIAA equalization. It thus can be used for record playback or vinyl cutting purposes. Another nice trick is to use it to simulate certain 'vinyl sound effects' by applying the inverse RIAA curve to your signal and after running it through a compressor converting it back to flat."

75. NoAmp! Free

"NoAmp! is based on a famous stomp box that made life easier for many guitar players and gained much popularity for its sound quality, versatility and ease of use."

76. Drumz Strip

"Simple Drum Player unit: twelve sample channels / outs with independent control over level, pan, mute and sample trigger."

77. X-Cita

"Analog style exciter for mastering purposes, inspired on the BBE Maximizer models. Its warm sound and simplicity of using makes this exciter very suitable to add some subtle brightness in the high frequencies, and also make the lows more dense. Apart from the mastering stage, you'll find that it can also make a good work on individual tracks."

78. Elottronix XL

"VST plugin which emulates the famous Robert Fripp's effect called "Frippertronics": two Revox B-77 making a continuous loop. This enhanced version adds many new features: 80 seconds maximum delay, Delay and LFO Pan now syncable to host, Biquad X filters, Tape noise generator..."

79. Sun Ra

"Ambient texture generator. It's based on a dual synthesis engine ( 1substractive oscillator + 2 wave players), completed with several randomization options and a battery of built-in effects. It also features a good set of 24 presets."

80. Baxxpander

"Vintage style saturation/bass expander unit, presented as VST effect. We think this plugin is specially good to add warmth and saturation to drums and basses, and with vocal tracks that need a little punch. It uses very little CPU, and includes 8 factory presets."

81. Cosmogirl II

"New version of the old Cosmodelia's synth of the same name. We have designed a fresh GUI, 64 new presets and automation of all parameters. It's a 2 oscillator analog style synth, specially geared to the creation of spatial FXs, ambient textures and experimental music."

82. Elottronix

"VST processor which emulates the famous Robert Fripp's effect called "Frippertronics": Two Revox B-77 making a continuous loop (2-8 seconds delay)."

83. Naive LPF

"Naive LPF is a resonant low/high-pass filter plug-in VST effect for Windows with a very convincing, classic sound. Naive LPF features a separate envelope follower for each of 6 channels and full automation, MIDI CC and even joystick control of all parameters."

84. DIG - Dual Integrated Gain

"New DIG 2.0 features matched only by high priced proprietary rivals. Never before have you been given this much flexibility in tone and gain control at a price impossible to beast. Its free!"

85. Cakewalk SFZ+

"Now free on the Cakewalk store."

86. Cakewalk Square I

Now free on the Cakewalk store.

87. Cakewalk AudioFX 1

Now free on the Cakewalk store.

88. Cakewalk AudioFX 2

Now free on the Cakewalk store.

89. Cakewalk AudioFX 3

Now free on the Cakewalk store.

90. Rough Rider Compressor

"Rough Rider is a modern compressor with a bit of "vintage" style bite and a uniquely warm sound. Perfect for adding compression effects to your drum buss, it also sounds great with synth bass, clean guitar, and backing vocals. Definitely not an all-purpose compressor, Rough Rider is at its best when used to add pump to rhythmic tracks. Of course, you can use it however you'd like. The Compressor Police aren't gonna come to your house and give you a citation. Slap it on a track and crank some knobs."

91. SPL Free Ranger

"The EQ Ranger plug-ins are based upon the EQ modules for the SPL RackPack series. They combine the amazing sound of passive fliters with extremely efficient operation. The Free Ranger is based upon the Full Ranger-EQ but is limited to four useful bands. Scroll to the bottom of the page to request a download.

hit me with your best riff

A Beginner Guide to Microphones [14 Sep 2009|11:51pm]


How Microphones Work

At their most basic, microphones are transducers. A transducer is an electrical device that converts energy from one form to another. In this case, the transducer is turning sound — acoustical energy — into an audio signal — electrical energy.

Most of you would know that sound is essentially fluctuations in air pressure. The component all microphones have in common is called the diaphragm. When sound waves hit the diaphragm, it vibrates, and the vibrations (which represent the fluctuations in air pressure) are turned into electrical energy (current). At the other end of the mic lead, that current is turned into the audio signal.

Granted, that’s a pretty basic explanation of how the microphone actually works, but as a musician, producer or engineer it’s all the science you really need to know about them.

Types of Microphone

There are plenty of types of microphones, all representing a different way of converting sound into signal. Each type of microphone has its own sound when compared to the other types, so knowing which is for which is just as important as knowing your mic’s frequency response. Microphone types usually get their names from their transducer type and directional patterns (for instance, you can get an omni condenser and a super-cardioid dynamic). Let’s look at the most common types of microphone you’ll be using.

Dynamic Microphone: Dynamic mics are one of the most common types of microphone. That’s because they’re cheaper, they can be used in both live and studio situations, and quite a bit hardier than most other microphones — for instance, put a condenser and a dynamic in front of the same number of spitty singers or tuba players and you’ll find the dynamic lasts longer due to its resistance to moisture.

Generally, dynamics don’t pick up as much detail as a condenser, so they’re not used in the studio as much as they are live, but they do come in handy on loud instruments such as electric guitar where condensers are only useful a few feet away. They’re also used by bands who want to get “that live sound” in the studio.

Condenser Microphone: Condenser microphones are incredibly popular, but not as common as the dynamic because they’re expensive, and they aren’t easy to use in a live situation unless they’ve been specifically designed for that — they generate feedback very easily. Condensers generally pick up a lot more detail than dynamic microphones and are better for quieter, subtler sounds. They also require 48V phantom power, where their dynamic cousins just need to be plugged into whatever’s available to receive the sound.

Condensers are great for picking up loud sounds without losing detail, though if you don’t have a wide breadth of condensers available to you, you may end up using a dynamic for those. Condensers are fragile and anything from air moisture to a bit of a bang can ruin them in no time.

You can get live condensers, such as the Rode M2. I’ve always preferred them as I find few dynamic microphones suit my voice.

Ribbon Microphone: Ribbon microphones are quite expensive, particularly fragile, and much less common than dynamic and condenser microphones. In the studio, they’re used frequently and I know one producer (misguided or not) who said he never used any mic except ribbon mics anymore. Like dynamic microphones, they don’t require 48V power, but unlike dynamic microphones, they can be damaged if that much power is fed into them, just to prove to you they really are fragile things (there are some ribbon mics being produced now that won’t be damaged if you forget to turn the phantom power off).

Ribbon microphones get their name from the the thin metal ribbon suspended in a magnetic field that picks up the vibration and turns it into a signal by magnetic induction. They’re good for a number of purposes including stereo recording and isolating an instrument in a noisy room (think drum kits).

There are other types of microphone, including the carbon and crystal microphones, but these are the three you need to know if you’re getting started in the world of recording.

Polar Patterns

The polar pattern of a microphone determines from which direction it picks up sound — or more accurately, how sensitive they are to sounds arriving from different angles, since if you sing into the back of a cardioid microphone you’re still going to hear something (even if it’s very quiet). Read on to find out about polar patterns, including what on earth a cardioid microphone is.

  • Omnidirectional: an omnidirectional microphone picks up sound equally from every angle. Omnidirectionals are great for recording choirs, a bunch of string players standing in a circle, and so on.
  • Cardioid: cardioids are the most popular polar pattern, and pick up mostly noise from a wide front area with about bit of sensitivity around the sides of the back, and almost no sensitivity at dead-center rear. They’re named cardioid because the polar pattern is heart-shaped when demonstrated in diagram format.
  • Hyper-cardioid: like a cardioid, but picks up a thinner area at the front and is less sensitive at the back.
  • Super-cardioid: has about as much as front sensitivity as the cardioid but even less rear sensitivity than the hyper-cardioid.
  • Bi-directional: these microphones pick up sound from the front and the back, but not so much the sides. Good for duets or other situations where you want to record two sound sources but exclude any others.
  • Shotgun: shotgun microphones are named so because you point them at a sound source and they won’t pick up anything but that sound source. Technically, that’s not quite true — they have some sensitivity on the sides and at the back — but it’s far less than any other microphone. They’re often used in field recording and on television, but they come in handy when you’re recording drum kits and the like where you want isolation.

Frequency Response

Just about every microphone has a frequency response chart in the manual. It looks like an EQ graph with squiggly lines showing how sensitive it is (or isn’t) to certain frequencies. A microphone that’s designed to record a kick drum will generally have slight boosts in the bass regions and cut off a bit of high-end. Be careful to check the frequency response of every mic you buy before you actually buy it. There’s no point recording a bass guitar with a microphone that has a high pass.


Most microphones use an XLR lead and plug in on the male end, though you can buy some crappy microphones that’ll plug in to a 1/4 inch jack. Steer clear of those — stick with XLR microphones or you’ll be climbing below the very worst of the recording quality microphones.

What’s important between the mic and the pre-amp is whether your lead is balanced or unbalanced. Unbalanced leads won’t do anything to stop noise. Balanced leads essentially run the signal down two wires, one with the phase flipped, so that at the other end you can combine the two signals and any noise that was introduced in the lead itself will disappear as it will be out of phase.

Electrical Current Level

The level of current a microphone generates determines gain at the other end of the lead. Mic level is a tiny amount of current, whereas line level and instrument level are quite loud as they are. When you plug a microphone into a rack in the studio or a mixer in a live situation, the first thing you need to do is amplify the signal. This is done when you are setting up your gain structure. The goal is to get all signals to line level or unity gain so they can be mixed relative to each other as easily as possible.

In a nutshell: don’t skip pre-amplification! It’s a common newbie mistake to go buy a microphone without buying some sort of pre-amp. Do your research and get something that makes your mic signal sound good — lively, loud and noise-free.

Protecting from Plosives and Wind

There are a few ways to stop plosives and pops from your vocalist or wind noise in general, but they all boil down to: put something between the microphone and the sound source (the wind is a sound source for the purposes of this tutorial). Sure, we could crap on about reducing noise by putting the singer on a funny angle or any number of tricks to get rid of plosives, but at the end of the day, some sort of mesh or fabric is going to have to go in front of the microphone. Don’t even try to record without something.

The best solution is a pop filter. You’ve no doubt seen these in use, even if only in a music video. It’s a circular fabric or metal mesh that sits in front of the microphone and clamps onto the stand. I prefer metal pop filters as I find the fabric can take some of the high-end frequencies off the top — it’s not always noticeable but I prefer to be able to take the sizzle out of a sound myself.

That said, a fabric pop filter is far better than the poor man’s solution I used years ago: sticking a sock over the microphone itself. Never had my sound clip due to a plosive, but it sure does sound muffled to my ears these days!

Buying a Microphone

So now you know how microphones work and what types of microphone exist. There are a few things to look at when you actually go and purchase a microphone. The first thing to know is what the microphone will be used for. Will it be for vocals? Guitar? Drums? Piccolo (if anyone even plays those anymore)? Or will it be an all-rounder? Be forewarned that you can’t get an all-rounder microphone — you can only get a mic that works on more sound sources than another mic.

The first factors to decide on are ones we’ve already discussed:

  • Type of mic — will you be recording live or in the studio? If you’re recording in the studio, do you want something that can tolerate the beating a loud, distorted electric guitar through a big stack will put it through? What about a metal screamer? The rule generally goes: dynamics for live situations, condensers for the studio, unless the sound is loud, in which case you go for the dynamic anyway. There are probably more exceptions to the rule than there are actual cases where you’d follow it, so do your research properly and don’t flame me if you buy a dynamic vocal mic for the stage that you hate.
  • Polar pattern — which polar pattern you want depends on so many factors. If you’re in a live situation, I would go for cardioids and mics that isolate. Maybe you want to pick up the room or multiple sound sources in a studio, in which case you’d go for something with a more open directional sensitivity.
  • Frequency response — I think that the flatter the microphone’s frequency response, the better, but perhaps you want a live mic that has a bit of extra kick in a sound source’s dominant range (this is what the SM58 does for vocalists). Keep in mind that you can’t change a microphone’s inherent response, but you can accentuate frequencies later with a bit of EQ, hence why I think flat response is the best response.

Other factors to consider:

  • Impedance — microphones are either high impedance or low impedance. I won’t go into the details of impedance, because frankly I sometimes have a hard enough time getting my head around it myself and it’s certainly not important for making great music — but generally speaking you want to get a microphone with lower impedance. High impedance microphones are cheaper and they’re fine if you’re not using a ridiculously long cable, but if you’re playing a stadium and want to run around with a twenty meter cable, it becomes more important to get a low-impedance mic and a low-impedance cable to reduce noise and interference.
  • Noise cancelling — some microphones have features to help control noise, such as suspending transducer components to isolate unwanted vibrations.

The number one factor to consider when purchasing a microphone is sound quality. Above all else, try the microphone on the sound you want to record with it and see how it compares to other microphones in your budget range.

Even with the same polar pattern, transducer type and frequency response, one microphone will sound better than the other. Tone is supposedly a matter of overtones and matching frequency responses should provide matching sounds, but microphones that are built better simply sound better. Don’t listen to anyone who tells you that two microphones will sound the same because they have the same specifications — it’s not true!

That’s all you need to know in order to know quite a bit about microphones. Next time we’ll look at microphone technique and placement.

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Understanding Synthesis in Propellerhead Reason 4 [29 Jan 2009|03:19pm]


As of version 4, Reason features three different synthesizers:

  • The Subtractor - An analogue synthesizer that features 99 voice polyphony based on subtractive synthesis with two oscillators.
  • The Malström - A graintable synthesizer that uses a complex synthesis method called granular resynthesis.
  • The Thor - A semi-modular, polyphonic synthesizer that features six oscillators and four filters, along with a step sequencer.

Let’s talk a bit more in depth about the type of synthesis involved in each of these.

  • Subtractive synthesis: The oscillators generate a waveform from scratch, and then the sound is run through various audio filters. The filters then use a cut-off frequency to subtract various harmonic content from the sound, hence the title “subtractive.”
  • Granular synthesis takes a sample of a piece of audio, and slices up the original waveform into tiny pieces, and then modifies them using the Attack/Decay/Sustain/Release (hereafter referred to as ADSR). It can also replay the pieces in a different order.
  • Graintable synthesis: This is an interesting one, as graintable synthesis is a cross between granular synthesis and wavetable synthesis. The basic principle behind graintable synthesis is a sample sound that is processed, resulting in a set of periodic waveforms that can be manipulated by the synthesizer. It combines the best of granular and wavetable synthesis functionality, so in a graintable synthesizer, functionality from either or both can be used.
  • Wavetable synthesis is essentially generating a waveform using an oscillator, similar to the Subtractor, and then modifying that waveform. The Malström gives us the best of both those worlds.

At the other end of this, the Thor synthesizer lets you basically do whatever you want. This is truly a beast, though it may not seem that way at first. Thor provides six open filter and oscillator spots, so you can load three different filters and three different oscillators at the same time.

This is one of the main reasons that the Thor synthesizer provides all new sounds and functionality in Reason 4. The release of this synthesizer was a big event in the scope of Reason 4 and synthesis development.

One last thing before we take a look at each individual synthesizer. I mentioned Attack/Decay/Sustain/Release (ADSR) above, but I didn’t explain it, so I will do that here. ADSR is a fundamental component of synthesis.

  • Attack: How quickly the sound reaches its full volume.
  • Decay: How quickly the sound drops back down to the sustain volume level.
  • Sustain: The constant volume that the sound reaches after decay.
  • Release: How quickly the sound fades to silence.

Now that we’ve got that out of the way, it’s time to look at each individual synthesizer in Reason.

The Subtractor

As mentioned before, the Subtractor is an analog subtractive synthesizer. It generates waveforms using oscillators, and then filters are applied to change the harmonic elements. Looking at the Subtractor, let’s start on the left. First, there are the usual controls, such as portamento (how the sound slides from one note to another), polyphony, and legato.

Below that, there is the modulation wheel and pitch bend, both of which are commonly used in automation.

To the right, there is an external modulation section, allowing the modulation wheel to be assigned to various parameters.

The Subtractor has two oscillators, each capable of generating 32 waveforms. Oscillator 1 is always on, and Oscillator 2 can be toggled on or off using the button next to it. The mix knob next to the two oscillators controls the mix between the two. When the knob is at the left, the only sound in the mix comes from Oscillator 1, and when it’s at hard right it plays only Oscillator 2. Anywhere between is a mix of the two.

You can toy with the Phase knob with each oscillator. This will generate another wave of the same form, and offset it by the phase you set. Think back to physics class, when you had two waves that collided, and they would either amplify each other, or they would subtract from each other. You can then change the mode by clicking the button, which will tell the waves how to interact (X multiplies, - subtracts, and O means no phase modulation).

There is a noise generator, along with FM synthesis, and a feature for ring modulation. Then, in the filter section, which is a combination of a multi-mode filter and a second linkable low-pass filter. There are also three envelope generators, to control modulation, filters, and amplitude.

Next, we have the LFO section, followed by the play parameters that make the sound’s velocity dependent.

Also, just a neat fact for our all our trivia gurus out there: try spinning the screws on the Subtractor (use your mouse like you are trying to turn a knob). The Subtractor is the only device in Reason with working screws. So if you take anything away from this tutorial, be sure to remember that! But seriously, onto the Malström we go!


The Malström

As I said before, Malström combines granular and wavetable synthesis. Some things here will seem familiar from the Subtractor, but there are other concepts here that are entirely new. As usual, let’s take a walk around the Malström. First off, we have the global settings area, same as with any instrument in Reason. This section contains the polyphony, portamento, legato, and modulation and pitch bend wheels.

Now, I’ve talked about Legato in passing, but haven’t really explained what it is. Legato essentially means that, if you change notes, there will be no silence. If you turn legato on, and play a note, and then play another note quickly, there won’t be a gap of silence. Instead, you’ll hear the second note as simply an extension of the first. Legato is commonly used in string instruments, vocals, and synthesizer leads.

Next up, we have Oscillators A and B, both of which can be toggled on or off using the buttons.
You can select a graintable by clicking the up or down arrow next to the name, or right clicking on it. The index slider is what controls playback, as in, which grain to play first. The motion knob then tells us how fast to play through the grains in the graintable.

There is a neat trick that you can use to hear all the grains in your graintable. Turn the motion knob to -64, all the way to the left, and then slide the index from side to side. Because the motion is set to -64, it will only play through the first grain. By changing the index, you’re changing which grain plays first, allowing you to hear each individual grain in the graintable.

Each oscillator also has an ADSR envelope, a volume control, and a shift knob (which chooses which harmonics to emphasize in the sound). There is also an octave knob, which selects which octave to play the sound in, and a semi and cent knob to help shift the oscillator.

There are two modulators in Malström that function as the LFOs, but also have a specific section for graintable synthesis. Mod A modulates the rate, index and shift functions. Mod B modulates the rate, motion, volume and filter. Using the switch, you can choose which oscillator (A, B, or both) the modulator affects.

What you might notice is that some knobs in Malström don’t seem to have any effect on the sound. This is because of the way that Malström routes the sounds. Similar to the Thor below, there is a path to how the sound travels through the components of the synthesizer.

If you follow the arrows and lines, you’ll notice that Oscillator A and Oscillator B are routed.
Oscillator A travels into the shaper, and then into Filter A. Oscillator B travels into Filter B.
These routes can be changed, using the various buttons to turn each path on or off. This may seem kind of confusing at first, but after a few seconds of toying with it, it becomes much clearer.

The filter knob in Modulator B won’t work on an initialized patch. You have to route Oscillator A through the Shaper (although you can turn the shaper off if you don’t want to hear the effect) and Filter A in order for it to have any effect.

There is one last feature of the Malström, which is the small output section. Obviously, the volume controls the volume output level, but the spread knob needs some explaining. The spread knob moves Oscillator A more to the left, and Oscillator B more to the right.

The Shaper deserves an extra mention, as it is possibly one of the most destructive forces in the Malström. The Shaper can take a mediocre initial Malström sound, and turn into into a beast. The Shaper wasn’t meant to be the defining piece of the Malström, but if you go overboard with it, it can completely dominate your sound. Sometimes this is useful, but often it is a little too destructive for most practical purposes, so keep it under control.


The Thor

The Thor is an absolute beast. It boasts three changeable oscillator slots, and three changeable filter slots. It also has a mixer for mixing the output from each of the oscillators. It comes with an arpeggiator and a step sequencer. If you think of Reason as a rack of synthesizers, then using Thor gives you a rack of synthesizers inside a rack of synthesizers. It’s truly incredible.


Go to Create > Thor Polysonic Synthesizer. Click the Show Programmer button. I’ll pause so you can pick your jaw up off the keyboard.

If you look at the image following this paragraph, things won’t be as ridiculous, as I’ve set all programmable options to Bypass. On our left, we have the three programmable oscillator slots. If you click the gray down arrow in the top left of each space, you can choose an oscillator for each space.

The options we have to choose from are:

  •  Analog
  • Wavetable
  • Phase Modulation
  • FM Pair
  • Multi-Oscillator
  • Noise

Each of these various oscillators brings different functionality and a different sound to the synthesizer. The analog oscillator is probably the most common, featuring the four major waveforms (saw, square, triangle, sine). There are more options available to us in the Thor than in the Subtractor (which also uses analog oscillators).

The wavetable oscillator brings 32 wavetable settings. Once a setting is selected, the position knob allows you to control which section of the wavetable is used. This is similar to the Malström (which combines wavetable and granular synthesis).

The phase modulation oscillator starts with a default wave that is then blended by up to two waveforms (First and Second). Once you choose a wave for First, you can adjust the phase modulation knob and hear the default sine wave slowly change into the wave of your choice.

The FM (Frequency Modulation) pair oscillator uses a modulator and a carrier to control sound. The carrier creates the main oscillator sound, while the modulator controls harmonics. If you leave the FM knob turned to 0, you will hear a sine wave, as the sine wave is the default for FM synthesis.

The multi-oscillator essentially emulates several oscillators all playing at the same time. However, there are controls to allow you to detune the sound, and also control the octave and toy with the way the sound plays back.

The noise oscillator is primarily used to add a grunge feel to sounds, as it generates various forms of noise that can be controlled using the mode knob. Each of the different settings produces a different type of noise.

What you then see is a bunch of arrows pointing all over the place. If any of you have ever taken a circuits class, think of this synthesizer as an electrical circuit (because it essentially is). Oscillators 1 and 2 both get routed into the mixer, and the balance between the two sounds is mixed and controlled using the rotary dial titled Balance 1-2.

The resulting output of that mix is then mixed with Oscillator 3 and controlled using the two sliders called 1+2 and 3. Next, the output from there is routed to either Filter 1 or Filter 2. The red buttons allow you to choose which signals actually get passed through the filters. There are various filter choices available to us:

  • Low Pass Ladder
  • State Variable
  • Comb Filter
  • Formant Filter

From there, the output heads on over to the next section, where an LFO and three envelopes are applied (or not applied if you deselect Gate Trig), and the resulting output is then passed to Filter 3. Once in Filter 3, the various other effects are applied, such as Delay, Chorus, LFO2 and whatever filter you select for Filter 3.

Hopefully that helps make a little more sense to you. I know not everything is covered, but if you start worrying about the details before the basics, it quickly gets overwhelming.


But wait, there’s more! There’s still an arpeggiator, a step sequencer, and a modulation matrix.
I’ll break down the basics of each of these now:

The modulation matrix allows you to choose a source, destination and amount in order to program the way you want the modulation of the synthesizer to happen. For the sake of not sounding like a signals and systems professor, I’m going to leave this relatively alone, as for the most part, this will be done for you by Reason. If you ever get curious, click the down arrow next to the source, and see what is available to you.

At the very bottom of the Thor is the step sequencer and arpeggiator. They are combined into one, so the Thor can generate patterns and/or notes. If you adjust the slider from OFF to Repeat and click Run, and begin to choose between the various settings, you can enable arpeggiation for a rhythmic feel.



Now, theory is all good and fun, but aren’t we really after the real use of these synthesizers? Of course we are, so now let’s take a look at the application. In the name of generalizing things, I’m going to try to break down the realm of synthesizer patches into several types:

  •  Leads
  • Pads
  • Basses
  • Glitch

In general (but not always), each of the synthesizers in Reason has a strength, a particular type of sound that it’s very good at. Obviously, you can use any synthesizer for almost any application, but these guidelines will help if you’re struggling to create a sound.

  • The Subtractor is widely used for basses, and very flat leads. The Subtractor is excellent at generating waveforms, which is about all you need for a bass, or for a straight sine wave lead.
  • The Malström is much more widely used for pads and for glitch effects. The filters and the way that the sounds are synthesized make it much easier for these sorts of things. Whereas in Subtractor turning knobs yields more subtle changes, changing a knob in Malström can completely destroy the sound (using the noise filter mode on the shaper is one such example).
  • The Thor is still relatively new, but it has proved its usefulness in all sorts of sounds, but it is less commonly used for glitch and bass than it is for leads and pads.

Now, remember, this is generalized. I’ve heard some wicked bass patches from the Malström, and I’ve heard some great pads from the Subtractor. But if you are having trouble creating the sound you want, make sure you’re using the correct synthesizer for the task.

Just for kicks, I’m going to show you a quick application of the Thor synthesizer. I toyed around with the Low Pass Ladder Filter in the Filter #3 slot, and changed the Drive to 63 and the Res to 77, and then saved it as Epic Poly_Whisper.

This gave the synthesizer a “whispery” quality to it, as it is now more resonant and less direct. You can find it in the Play Pack for this tutorial.

Now, play the melody I have shown below:


Right click on the Thor icon in the sequencer, and go to Parameter Automation, and choose Mod Wheel. Grab your pen tool, draw a new group, then double click the pen tool, and draw a line looking like this:


We now have a very trance-esque synthesizer line with automation on the modulation. This sort of technique is used very often in electronic music. I’m going to go ahead and add in a Subtractor and a Malström line.

I’m going to use “WarmPad” for the Subtractor and “SoundOnSound” for the Malström. Nothing all that exciting, but I want you to see a practical use of each.

The Subtractor:

The Malström:

All three:


We’ve covered some of the scientific explanations of synthesis, and we’ve looked at the various synthesizers in Reason. As I’m sure you already figured out, there’s a near endless amount of things that you can do with synthesizers, far more than I could cover in 10 tutorials.

However, this tutorial should have given you the foundation and fundamentals to be able to use Reason’s synthesizers to their potential. As always, I’ll be checking comments frequently and will try answer any questions and concerns you might have. Happy producing!

Download the Play Pack for this tutorial (9.3MB)


  • Reason 4 Source File
  • Example Audio
  • Synthesizer Patches
hit me with your best riff

Make your offers, please! :] [28 Oct 2008|01:57am]

If OT, mods please feel free to delete this post.

I'm selling my Roland JP-8080 and Edirol FA-101 on ebay, both in mint condition.

If anyone is interested, I'm also open to direct sales, and can take the listings down and offer an additional discount as the final ebay fees are quite a bit. :s

Thanks for looking!
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Online guides for Mixing, Mastering, and Leveling [18 Jun 2008|11:39pm]

Mixdowns for Beginners:

Mastering for Beginners:

Leveling and Metering:
1 melodious critic alleged that....| hit me with your best riff

MIDI & Digital Audio Music Sequencer Techniques [20 May 2008|06:32pm]

Music technology has finally gotten to the point where almost anything sonically imaginable can be achieved. Audio can be recorded, chopped into pieces, and then rearranged into something completely different - all with a few clicks of a computer mouse. Whole environments can be simulated so that while it sounds like your singing on stage in a huge concert hall, you're actually standing in the middle of your bedroom studio. Complete professional-quality songs can be recorded, produced, mixed, and mastered using a single standard computer system with MIDI/ digital audio sequencing software.

Of course, just because you can imagine something doesn't mean you know how to accomplish it. Today's digital audio sequencers pack so much power and contain feature upon feature, that it's easy to stay comfortable and just stick with the basics, especially when you're facing deadlines and don't have the time to experiment. 

We're going to take a look at some sequencing techniques that you can use in your day-to-day digital audio dealings. We'll cover a number of generic techniques that will work with any sequencing software you may have. I'm going to assume that you already know the ins-and-outs of MIDI and digital audio. Ready? Set? Let's go...

The real-time effects built into many sequencers aren't the only way to add excitement to your MIDI and digital audio tracks. With a few applications of the copy, paste, and pitch shift commands, you can easily simulate a number of different effects including delay, flange, and chorus. This can free up some of your computer's precious processing power for handling the more complex effects like compression and reverb.

Digital delay. This technique actually covers a number of different effects, and can be applied to both MIDI and audio tracks. Simply copy the MIDI or audio clip that you want to process, and paste it to a new track. Then slide the copied clip forward in time to create a delay. Depending on how much of a time offset you use, different delay- related effects can be achieved. Slide the clip between 5 and 10 milliseconds and you get a thickening effect. Slide the clip between 10 and 20 milliseconds and the thickening turns into doubling. Anything above that amount and you start creating discrete echoes.

If you want to create a delay that is synchronized in time with the music, just move the copied clip ahead by snapping it to a note value such as a sixteenth. To make things more realistic, make a few clip copies and slide each one a sixteenth note apart. Then lower the volume of each successive echo so that the effect fades over time. As an example, for three echoes, lower the volume of the first to 75%, the second to 50%, and the third to 25% of the original. You can even get a little crazy by sliding each copy to a different uneven time offset, and maybe give them different pan positions too. You can't get that kind of precise flexibility with a real-time effect.

Flange. This effect can be applied to both MIDI and digital audio tracks but they require slightly different procedures. As with delay, simply copy your clip to a new track. From here, you can easily produce a static flange effect by moving the copy forward in time from 1 to 5 milliseconds. While this creates an interesting sound, it doesn't have that authentic analog sweeping that we all love and cherish. To achieve that, we need to use a little pitch shifting.

For audio, instead of moving the clip copy, just shift its pitch anywhere from -12 to +12 cents. This will make the copy longer or shorter by a minute amount, causing it to drift out of sync with the original. Of course, depending on how long the clip is, eventually there will be so much of a delay that the flanging will turn into doubling. To get around that, just shift the pitch of small segments of the clip, alternating the direction of the shift with each segment. For example, shift the first two seconds of the clip by -9 cents, the next two seconds by +9 cents, and so on. This is also how you can control the rise and fall of the sweeping characteristic.

For MIDI, the procedure is similar except you need to apply pitch wheel controller changes to the clip copy. Just draw in some pitch controller changes over the course of the clip that vary by rising above and falling below the zero axis. I found it best to stay within the +600 to -600 bend range because any changes larger than that start sounding like detuning rather than flanging. To apply precise changes, many sequencers will allow you to insert a series of controllers that change evenly from a start value to an end value over a set amount of time. This will give you more precise sounding effects but you may prefer the more random fluctuations you can get from just drawing the controller changes in by hand. Either way, the effect is not quite as authentic as with audio. Many times it sounds more like chorusing than flanging.

Chorus. You create audio chorusing in almost the same way as flanging except you need to stretch the pitch, rather than shift it - meaning you need to change the pitch of the audio clip without changing its length. Most programs have this capability. To apply chorusing to an audio clip, simply copy the clip to a new track and stretch its pitch, plus or minus 9 to 12 cents. The more you stretch it, the deeper the effect. You can also use this technique to create a pseudo-stereo effect from mono parts by simply panning the original clip and clip copy to different locations. The more pan, the wider the stereo image.

If you really want to go wild, you can create something I like to call the Super-Combo Chorus Effect. This time, instead of creating just one clip copy, make 3 or 4 or more. Stretch the pitch of each copy using slightly different values from the rest. Now slide each copy in time using the delay techniques described earlier with slightly different values. For the final touch, evenly pan each copy throughout the stereo field. You can also give each copy a different volume level for more subtlety. When you're finished, hit play to experience one very cool chorus simulation.

When destructively applying plug-in effects to digital audio, many sequencers will let you either replace the original material or leave the original material intact and create a new track (or set of tracks in the case of stereo) for the new effected material. If you adjust the levels of the effect module so that the dry mix level is 0% and the wet mix level is 100%, your new track(s) will contain the effect output minus the original signal. Why would you want to do this? Because it gives you some very interesting possibilities.

For instance, you can now control your track effects in a variety of ways. Using volume controllers you can bring the effects track(s) in and out to add an artistic flair to the mix. You can also use panning to move the effected signal around the stereo field while the original signal either remains stationary or moves around as well but independently. Even more possibilities arise when you consider that you can add effects to the effects track(s). How about applying different EQ to the original and effected tracks? If your effected tracks contain reverb, how about applying another reverb effect on top of that for some very far out spatial displacement?

And if you happen to have multiple soundcards or multiple outputs, you can assign your effects track(s) to their own outputs. This way you can even use your outboard modules to add effects to the already effected track(s). Experimentation is the key here.

You've probably heard of the term "Groove Quantizing" or something similar, where a sequencer uses a pre-defined MIDI pattern that contains data in a specific rhythmic style and applies it to recorded MIDI data so that it conforms to the same rhythm. This feature is a great way to breathe new life into a tune or even give it a different sense of feel.

Well, in case you didn't know it, you can also use "Groove Quantizing" on audio tracks. You just need to do one small preparatory step. You need to break down the audio track (or audio clip) into small pieces or clips. Some sequencers will have an automatic feature for this. If yours doesn't, you can try exporting your track to an audio editing program like Sound Forge. Using Sound Forges' Auto Region and Extract Regions tools, you can easily break down a large audio file into smaller ones that contain a quantizable musical unit such as one beat of data.

When you import the small audio files back into your sequencer, each one will have its own start time, just like a MIDI event. This lets you easily quantize the playback of the audio segments, and you also don't get the weird anomalies that can arise from trying to stretch the audio. This technique works especially great with percussion clips but don't rule out instrumental or even vocal clips. Try turning a normal vocal track into a cool rhythmic scat. It's a lot of fun!

In addition to responding to the usual MIDI controller messages such as pan, volume, and pitch bend, most synthesizer modules have a number of "hidden" parameters that can be of great use in adding expression to your tunes. The way to get to these parameters is through the use of Non-Registered Parameter Numbers (NRPNs). As an example, let's take a look at the Roland Sound Canvas. This modest module responds to no less than 13 different NRPNs (see Table 1) that allow you to control things like the attack, decay, and release of a patch's envelope, as well as the cutoff frequency and resonance. As a matter of fact, by manipulating these parameters you can create brand new timbres based on the existing sound set, which is otherwise uneditable.

Table 1: NRPN's for the Roland Sound Canvas
Param. Control 98 Control 99 Control 6
Vibrato Rate 8 1 14-114
Vibrato Depth 9 1 14-114
Vibrato Delay 10 1 14-114
Cutoff Frequency 32 1 14-114
Resonance 33 1 14-114
Attack Time 99 1 14-114
Decay Time 100 1 14-114
Release Time 102 1 14-114
Drum Pitch Drum#0-127 24 0-127
Drum TVA Drum#0-127 26 0-127
Drum Pan Drum#0-127 28 0-127
Drum Reverb Drum#0-127 29 0-127
Drum Chorus Drum#0-127 30 0-127

For instance, let's say I wanted to change my basic piano track into something with more of a string feeling but keeping the piano timbre. I would open the event list for that track and insert nine new controller messages (see Table 2). The first three messages would set a new envelope attack, the next three would set the decay, and the last three would set the release. Controllers 98 (Non- Registered Parameter LSB) and 99 (Non-Registered Parameter MSB) are used to set the type of parameter to be changed (in this case envelope attack, decay, and release), and controller 6 (Data Entry MSB) is used to set the value of each selected parameter. All of the settings for each of these controllers depend on what MIDI device you're using them with. Have a look in your synthesizer manual and you might be surprised just how much control you have over that little black box.

Table 2: NRPN Example
Message 1 - Control: 98 Value: 99
Message 2 - Control: 99 Value: 1
Message 3 - Control: 6 Value: 70
Message 4 - Control: 98 Value: 100
Message 5 - Control: 99 Value: 1
Message 6 - Control: 6 Value: 100
Message 7 - Control: 98 Value: 101
Message 8 - Control: 99 Value: 1
Message 9 - Control: 6 Value: 70

A couple things to keep in mind when using NRPNs - send any patch changes down the wire beforehand otherwise you may end up with a different sound than what you were aiming for - although this could result in a pleasant surprise. The other thing to remember is that controller 6 is dynamic, meaning it changes the value of whatever was the last parameter that was set with controllers 98 and 99. In the above example, if later on in the track I sent another controller 6 message, this would change the envelope release since that was the last set parameter. In order to change either the attack or decay, I'd have to resend the appropriate values for controllers 98 and 99.

Another exciting aspect about NRPNs is that they can be sent in real-time, just like any other controller messages. What this means to you is that depending on your synthesizer module, you can now dynamically change the characteristics of patches or any other supported parameters and record those changes into your mix. In the case of the Roland Sound Canvas, you could easily add expressiveness to your sounds by varying their cutoff frequencies and resonance. In addition, you can dynamically vary the pan position, volume and pitch of the individual drums in a set. That could make for some very cool percussion parts.

Of course, typing in all these parameters would be a bit tedious. A better way would be to setup a special mixing template for just this sort of task. Most high-end sequencers provide the capability for you to create your own virtual mixing controller. In Steinberg Cubase the feature is called the MIDI Mixer. In Cakewalk Pro Audio and Sonar it's called StudioWare. Everyone has a different name for it but the feature is still the same. If you're lucky, your software will come with a pre-made template for your MIDI instrument. Cakewalk has a Roland GS template for use with a Sound Canvas but it's missing any drum parameter controls.

If I wanted to add individual drum panning to the existing Cakewalk Roland GS template, I would open a new copy of the template into my existing project. I'd then add a new Cluster widget to hold the new set of controls. Then I would add a slider object to represent the drum instrument number. From there, I'd set the necessary properties for the slider including Label (Drum Number), Alias (Drum_Number), Automate in Track (AutoTrack), Range (0-Min, 127-Max), Primary Action (Kind: Controller, Channel: Part, Number: 98, Value: Drum_Number), and Return Action (Kind: Controller, Channel: Part, Number: 99, Value: 28).

The Alias acts as sort of a program variable which is then used in the Value setting of the Primary Action property to tell Cakewalk to use the value of the on-screen slider (0-127 set in the Range property) as the MIDI controller value. In contrast, the Value setting of the Return Action is permanently at 28 because MIDI controller 99 is a constant value in this case. The Label is just an on-screen text name for the slider and the Automate in Track property just tells Cakewalk which track to record the automation data from this slider to during mixdown. In this case, AutoTrack automatically sets this property to whatever the currently selected track may be.

To complete the drum panning controls, I would also need to insert a Knob widget to represent the drum panning value. With this new control set I could now easily select any of the Sound Canvas drum instruments by moving the slider to the appropriate number and then pan the instrument using the knob - recording any changes to a designated track. Of course this whole procedure would be a bit different in another sequencing program but the concept is still the same. Once you learn how to create your own virtual mixer templates, the possibilities are endless.

When inspiration hits, you don't want to waste your time fiddling with sequencer setup parameters. You want to be able to start your software and get right to work. If you create a template file that contains everything set just the way you like it, you'll have a much better chance of getting that cool lick down before you forget it. Here's what you need to do to create the ultimate project template...

1. Use the File > New command to create a blank project. Then setup the track display with all of the instruments in your studio. Set the name, input, output, channel, bank, patch, volume, pan, etc. for each track.
2. Go through the rest of the program and set things up exactly the way you like them. That includes window positions, zoom, and even the last tool used. In Cakewalk Pro Audio and Sonar other parameters like timebase, file information and comments, tempo settings, meter and key settings, clock and synchronization information, and more can all be saved.
3. If your software has a built-in System Exclusive librarian, fill it up with all of your favorite Sysex macros, so that you have them readily available. This could be anything from a favorite bank of patches for one of your synthesizer modules or even basic messages like: Roland GS Reset (F0 41 10 42 12 40 00 7F 00 41 F7) and Yamaha XG Reset (F0 43 10 4C 00 00 7E 00 F7). If your software doesn't have a Sysex librarian feature, then just designate a special track for Sysex messages and save each macro as a separate sequence or clip. Be sure to keep that track muted so its data won't be transmitted during playback. When you need to use a certain Sysex macro, just drag-and-drop it to an open track.
4. Taking the Sysex track idea one step further, you can also create special tracks for your favorite MIDI controller macros. A MIDI volume track can hold different sets of volume changes such as fade in and fade out segments for a number of bars. A MIDI pan track can hold different sets of pan changes. Some interesting ones include a left to right sweep, and right to left sweep, and a center to right to left back to center sweep. In addition, you can create tracks for your favorite drum patterns or sequenced melodic patterns as well. And if your sequencer allows you to store MIDI and audio data together in the same file, you can set up tracks containing some of your favorite digital audio samples and loops too. This way you have all the tools you need right at your disposal for quick and easy drag-and-drop use.
5. The final step is to save your file with all of its settings so that when you open it again, everything will be perfect for your next session. If your sequencer supports it, you can also have the template file load automatically every time you load the software. For Steinberg Cubase on the Mac, name the file Autoload. For Steinberg Cubase on the PC, name the file DEF.ALL. And for Cakewalk Pro Audio and Sonar, save the project as a template file called NORMAL.TPL.

Today's digital audio sequencers provide you with a lot of power to accomplish your musical goals. Many times you can achieve the same results in a number of different ways. The tips we've covered here are mainly productivity boosters and techniques you might not have otherwise thought about.

One final word of advice is to go out and buy yourself a small binder to use as a studio journal. As you work on different projects, take a few moments to jot down any new techniques you may come up with in the process. It's also a good idea to reference what you read about in publications, so that you can easily go back and find the issue and page number where you initially saw it. Not only will this save you time on future projects, but eventually you'll have a nice collection of information to refer to.

While technology shouldn't govern the way you make your music, it can certainly help you create the songs of your dreams. The more you learn about how to utilize the tools at hand, the easier the process will be.

hit me with your best riff

Add More Boom to your Bass with Sound Forge [13 May 2008|10:43pm]

Recently someone asked me how they could get that really low bass sound that you hear these days in a lot of techno or pop music. You know, it's not quite a bass drum sound, but more like a low thump that really gets your speakers pumping. Well, believe it or not you can create your own sample of this type of sound very easily using Sound Forge. Here's how:

1) Select File>New, and choose 44,100 Hz, 16-bit, mono to create a new window for your sound.
2) Select Tools>Synthesis>Simple to open the Simple Synthesis dialog box.
3) Set the amplitude parameter to 0 dB.
4) Set the Waveform Shape parameter to Sine.
5) Set the Length to 0.100 seconds.
6) Set the Frequency to 50 Hz.
7) Click OK.

Hit the space bar on your computer keyboard to play the sound. Cool, isn't it? It's also very quick in duration, but that's what you need because you'll be using it in conjunction with your bass sound.

Now if you'd like to use the sound in your sampling hardware or software, just save it as a WAV file. Then import the WAV file as a new sample (this procedure varies depending on your sampling hardware or software) and you're ready to start triggering your new sound via MIDI.

If you don't have sampling hardware or software, you can also import the WAV file into your digital audio sequencing software. Let's use Cakewalk for this example...

1) Select the Track to which you want to import the WAV file.
2) Set the Now Time to the beginning of the project.
3) Select Insert>Wave File.
4) Choose the WAV file and click OK.

Now just use this Track to hold your new WAV sound sample, and simply copy and paste the sample into another Track so that it coincides with each note in your bass Track. Now whenever a bass note plays, it will get an extra boost from your boom sample. Just be careful not to play your music too loudly, because you're sure to get complaints from the neighbors!
hit me with your best riff

Propellerhead Reason Subtractor Synth - Using Envelopes [18 Apr 2008|04:19pm]

When discussing sound and sound synthesis, the word envelope is used to describe the shape of a sound over time. The Subtractor has three envelopes to play with: the Amp Envelope, the Filter Envelope, and the Modulation Envelope. You may be wondering what I mean by an envelope describing the shape of a sound over time. Imagine striking a large bell with a hammer. Think of the sound of this event happening in real time. At the beginning of the event is the initial strike. In envelope-speak, this is referred to as the attack. After the attack, the sound begins to fade out, or decay. This vocabulary, as you will read below, has been expanded to incorporate more detailed concepts, but you are probably starting to get a feel for where this is headed.

When we want to affect the volume (or amplitude) structure of the sound event over time, we will adjust the Amp Envelope. You may have been thinking in terms of volume or presence of sound as you considered the bell example. When we want to affect the tonal or frequency characteristics of a sound over time, we will be adjusting the Filter Envelope. And, of course, if we want to affect the manner in which modulation is applied to the signal over time, we will use the Modulation Envelope.

ADSR: Four Letters You Cannot Live Without
ADSR—these four letters stand for Attack, Decay, Sustain, and Release. Please say the words to yourself a few times if this concept is new to you. From here on out, you will see these four parameters again and again throughout Reason, and also on just about any analog or emulated analog synthesizer you ever use in the future. With these four parameters, you will sculpt quite a bit of your sound. If you take away nothing else from this book, I hope at least that this section will be clear in your mind for eternity. If you tell someone that you read this book, and that person asks you to show him how ADSR works in Reason, and you are unable to do so, then we will have both brought shame upon me, my ancestors, and all my future generations. No pressure, but with that in mind, please read on.

Amp Envelope
Perhaps the most familiar area in which to play with ADSR is the Amp Envelope (see Figure 1.13). An amplifier, in audio synthesis, makes a signal louder by increasing the amplitude of the signal. If you are looking at a sine wave on an oscilloscope, the amplitude will be the distance between the highest point and lowest point of one cycle of the sine wave. A shorter distance means a quieter signal, while a greater distance (a "taller") sine wave will result in a louder signal.

While the Subtractor's Level control affects the overall volume (or amplification) of your sound, the Amp Envelope controls the shape of how that amplification is applied over time. Will the sound start sharply when you strike a key, or will it fade in slowly? Will it end abruptly when you let go of the key, or fade out slowly? Is a short sound produced, like a drum hit, or is a long, sustained tone produced, like holding down a key on an organ?

Here is how ADSR relates to the Amp Envelope. You can refer to this when considering other ADSR envelopes as well:
* Attack — The amount of time it takes for the amplitude of a sound to climb from zero to its peak level.
* Decay — The amount of time it takes for the amplitude of a sound to fall from its peak level to the level determined by the Sustain parameter, assuming you keep holding down the key(s).
* Sustain — At the end of a note's decay, the Sustain value determines the level at which the amplitude rests as long as the note is being held.
* Release — The amount of time it takes for the level of a sound to drop to zero from whatever level it was when you let go of a note.

The following exercise should make this all quite clear.
1. Start with an empty rack. Create an instance of the reMix mixer by choosing Mixer 14:2 from the Create menu, and then create an instance of Subtractor. In the Subtractor's Patch Browser window, it should say Init Patch.
2. Turn the Freq (frequency) slider on Filter i all the way up. In doing this, you are actually deactivating Filter Why? Because the Filter Type control on Filter r is set for LP 12, which you know stands for 12-decibel Low Pass. It allows low frequencies to pass unaffected, while filtering out high frequencies (any frequencies above the frequency chosen with the Freq slider). So when you turn the Freq slider all the way up, there are no frequencies above that to be filtered. It's wide open.
3. Change the waveform on Oscillator i to a Square wave. This should be slightly less grating than the Sawtooth waveform. It's also fun if you know how to play any classic video game music, such as the Mario Brothers music.

Now that we can hear our naked square wave without any bells and whistles to confuse the issue, we are ready to explore ADSR in the Amp Envelope:
1. By default, the Attack setting on the Amp Envelope is all the way down at 1 (fastest/shortest attack possible). So when you hit a key, you hear sound instantaneously. Please move the Attack slider up to a value of about 69 (a little over halfway). Depress a key on your MIDI keyboard, and you will now hear that the note fades in slowly. Feel free to experiment with different values to get a feel for this.
2. Turn the Attack back down to zero, and also turn Sustain and Release down to zero.
3. Set the Decay to a value of about 5o. Not such a long note anymore! If you turn it down even lower (to any value between zero and 20), the note will be so short as to sound like a percussion instrument. I have mine set at 20 right now, and am amusing myself by playing "Popcorn," the whimsical, pioneering electronic dance mega-hit by Gershon Kingsley. This song was originally played on a Moog synthesizer, by the way. (Google this stuff if you don't know about it!)
4. With your Attack still at zero, and your Decay set at 20, turn up your Sustain level to 5o. Hold down a key. You will hear the sharp attack and the quick decay, but the decay will no longer fall down all the way to silence. It stops and holds at an intermediate level. Now turn up your Sustain all the way. When you hold down a key, the sound stays at its maximum level until you let go. Theoretically, the Decay setting would still allow the level to fall to the value designated by the Sustain slider, but since the Sustain value is set at maximum, there is no place to fall!
5. With Attack still at zero, Decay at 20, and Sustain at maximum, move the Release slider (which should still be at zero right now) up to a value of 6o. Now strike a key and let go. Notice how the note continues to fade out after you release it.
6. Move the Release slider up to maximum. Now hit a note or a chord, and hear it sustain forever (almost)! Before this drives you crazy, turn the Release slider back down. It would eventually fade out after a while, but it takes a long time.

Honestly, in the past I have suffered some confusion regarding this sustain, release, and decay thing. (Attack always seemed pretty straightforward to me.) You may be a quicker learner than I am, but if you could use some further clarification, please try the following exercise, maintaining the settings from the previous exercise.
1. Set Attack and Sustain to zero.
2. Set Decay to 6o and Release to 81.
3. Depress and hold a key until the sound fades completely away; then let go. Even though your Release is set at a value of 81, you should not hear anything after letting go of the key.
4. Strike and let go of a key as quickly as possible. (lust quickly tap it.) If you do it right, you should hear the effect of your Release setting. The note will continue to fade out even though you have let it go.

Steps 3 and 4 work the way they do because the Release setting controls the amount of time it takes for the level of a sound to drop to zero from whatever level it was when you let go of a note. When you tap and release the note, it hasn't had time for the Decay portion of the envelope to finish. The note has not dropped back down to zero before you let go. So after you let go, the level will fall at a rate determined by the Release value. You could also turn the Decay setting all the way down, and turn the Sustain all the way up, and still hear the effect of your Release setting.

Filter Envelope
Now, as we learned a little earlier, it is fun to sweep filter frequencies, and I will certainly attest to the fact that it is fun to twiddle knobs with one's bare hands in order to do so. However, it's also nice just to hit a chord and let something happen while you hold it. The Filter Envelope is the first of several features to be discussed that allows you to build some hands-free dynamic sonic action into your patches. It is an ADSR envelope, which also includes a Filter Envelope Amount knob and a Filter Envelope Invert button.

1. Start with an empty rack, then add reMix and Subtractor. In the Subtractor's Patch Browser window, it should say Init Patch. Please play a few keys on your MIDI keyboard. Notice the percussive attack that is present in the Init Patch.
2. In the Velocity section, turn down F.Env to zero (12 o'clock). You will see the little red light above it turn off. If you strike a few keys now, you will hear the little percussive attack that was there when striking the keys hard has disappeared because velocity (how hard you strike a key) is no longer affecting how much of the Filter Envelope is applied. The Filter Envelope Amt (amount) knob is already at zero, so the Filter Envelope is not being used at the moment.
3. In the Amp Envelope section, turn the Release slider up to 64.
4. Change the filter type of Filter i to LP 24 by clicking the Filter i Type button.
5. Turn the Filter 1 Frequency slider down to a value of 24. If you play some keys on your MIDI keyboard, you will not hear very much because most of the frequencies are being filtered out now.
6. Turn the Filter Envelope Amount knob all the way to the right (to its maximum value of 127). Now play some keys, and you will hear the sharp attack and short decay of the Filter Envelope affecting the frequency of Filter i.
7. Adjust the Filter Envelope so that Attack = 99, Decay = 99, Sustain = zero, and Release = 76.
8. Play and hold a chord on your MIDI keyboard. Now you can do a nice slow filter sweep while keeping both hands on your keyboard!

The next exercise demonstrates the function of the Filter Envelope Invert button. Clicking the Filter Envelope Invert button turns the whole Filter Envelope upside down. For instance, the Decay parameter would normally control how fast the frequency of Filter i is lowered. When you engage the Filter Envelope Invert button, this is reversed, and the frequency of Filter I will actually increase by the same amount at the rate set by the Decay parameter. To hear this in action, please do the following exercise.

1. Start with an empty rack, then add reMix and Subtractor. In the Subtractor's Patch Browser window, it should say lnit Patch. This exercise can get a hair loud, so turn down the Subtractor's Master Level slider to about 6o.
2. In the Velocity section, turn down F.Env to zero (12 o'clock). You will see the little red light above it turn off.
3. Set Filter i so that Freq=2o, Resonance=o, and Type=Notch.
4. Click on the Filter 2 Link button and the Filter 2 On/Off button (which will both light up red).
5. Set Filter 2 Frequency to 58 and Filter 2 Resonance to 89.
6. Set the Filter Envelope so that Attack=o, Decay=63, Sustain=o, and Release=o.
7. Set the Filter Envelope Amount (Ann) knob to 28.
8. Play a few notes, and you will hear the instant attack (Filter i frequency is at its highest point), followed by the decay (Filter i frequency falls at a rate determined by the Filter Envelope Decay slider).
9. Now click on the Filter Envelope Invert button. Play some more, and you will hear that the envelope has indeed been inverted. Now Filter 1 frequency starts at its low point and rises at a rate determined by the Filter Envelope Decay slider.

Mod Envelope
The Mod Envelope is another magical pair of invisible knob-twiddling hands that I have to tell you about. The Mod in Mod Envelope is (of course) short for modulation, and we know that to modulate in this context simply means to change or adjust a characteristic. So there is a change that is going to take place, such as changing the pitch of an oscillator or the frequency of Filter 2. The degree of this change will be determined by the Mod Envelope Amount (Amt) knob. How that change is implemented over time will be determined once again by an ADSR envelope.

The Mod Envelope looks almost exactly the same as the Filter Envelope. Like the Filter Envelope, it has sliders for Attack, Decay, Sustain, and Release. It also has an Amount knob and an Invert button. However, unlike the Filter Envelope, the Mod Envelope has a Destination button. While the Filter Envelope is hardwired to affect Filter i frequency, the Modulation Envelope can be assigned to affect Oscillator i pitch, Oscillator 2 pitch, the mix between Osc i and Osc 2, the amount of frequency modulation (FM), the phase offset of both Osc 1 and Osc2 (simultaneously), and Filter 2 Frequency. Let's give it a spin. You know the drill by now, but here it goes.

1. Start with an empty rack, then add reMix and Subtractor. In the Subtractor's Patch Browser window, it should say Init Patch.
2. Turn the Mod Envelope Amount (Amt) knob up to 12 o'clock. Play some notes on your MIDI keyboard, and you should hear an immediate effect. By default, the Mod Envelope Destination is set for Osc 1, so the Mod Envelope is affecting the pitch of Osc i. Since the Mod Envelope Attack is set for 0, the Mod Envelope affects the pitch of Osc I immediately to a degree determined by the Mod Envelope Amount knob. The short decay allows the pitch of Osc 1 to fall back to normal quickly, almost like an electronic drum sound.
3. Click the Osc 2 On/Off button so that it is lit red, then click the Mod Envelope Destination (Dest) button until the red light next to FM is lit. Play a few notes, and you will hear the Mod Envelope applied to the FM amount.
4. Click on Filter z's On/Off button so it is lit up red. Set the Filter 2 Resonance (Res) slider to 99.
5. Select Filter 2 frequency (Freq 2) as the destination for the Mod Envelope. Play your MIDI keyboard to hear the effect.

hit me with your best riff

Propellerhead Reason Tips - The Combinator [18 Apr 2008|04:16pm]

The Combinator is a beast that can hold entire songs, let alone a few devices. You can learn this device easily by opening up the Combinator patches. In this way, you can experience a good portion of its potential and how it can benefit your song. The basic idea behind the Combinator is to combine devices and their parameters. This means the Combinator can combine the sounds of two or more devices into one sound, or control two or more devices with one sequencer track or MIDI controller. Four knobs and four buttons on the Combinator panel can be assigned to many parameters simultaneously.

To combine devices simply means to put devices within the Combinator. You can either put new devices in the Combinator or drag previously added devices into the Combinator, thereby "combining" them. When the desired devices are combined, you can then assign a combination of any parameters on the combined devices to be controlled by the knobs and buttons on the Combinator device panel. Not only can the knobs be assigned, so can their range. For example, if you assign Rotary 1 on the Combinator to a volume slider for a SubTractor within that Combinator, you can set the range of modification for that volume slider from, say, 52 to 65, recalling that the normal range is 0 to 127. This means that when Rotary 1 is turned to 0, the volume on the SubTractor will be at 52. Similarly, when the Rotary 1 knob is turned to 127, the SubTractor knob will be at 65. Simultaneously, you can have the Rotary 1 knob control another knob on the SubTractor, or another parameter on another device that you have placed within that Combinator. The Combinator is controllable by a Matrix for pattern sequencing, and all the rotary knobs have CV inputs on the back for automation.

Despite its relatively simple operation, the Combinator is actually the most complicated device in Reason aside from the NN-XT. This is because you can create worlds of sound inside the Combinator that may run for extremely long screen lengths, comparable in size to many artists' racks for an entire Reason song. The Combinator keeps this potentially enormous sound creation organized by its collapsible device window and its parameter-controlling Programmer window. The Programmer has all the instruments named in order, as they are selectable for changing parameters and assigning knobs to the rotaries and buttons. Thanks to the rotaries, CV can now control every automatable parameter. You can also set both the key range and the velocity range for individual instruments, opening the doors for creating split patches and key-sensitive patches within the Programmer. As you can see, the Combinator really takes Reason to the next level.

Splitting Instruments in the Combinator
Creating a split instrument Combi is extremely easy. You simply split off key ranges for the sound-generating instruments within the Combinator's Programmer. A key range is the area on a keyboard in which a sound is played. For example, you can set a key range for a sound so it plays only on the bottom half of the keyboard, meaning that when the top half of the keyboard is played, no sound comes out. When you split off key ranges, it means that you set up one sound to play on part of the keyboard and the other sound to play on the remainder of the keyboard. Because this procedure as outlined in the Reason manual may be confusing to the average user, I have developed an alternative process:

1. Because this sample procedure uses two sound-generating devices, route at least two sound-generating devices by either a merger or a mixer into the From Devices input.
2. Open the Programmer, select the first device whose key range you want to manipulate, and click the Receive Notes check box to mark it. The check box must be marked in order for you to be able to manipulate the key range.
3. By default, the key range for a sound-generating device is at the device's maximum range, C -2 to G 8. To set the key range for the first device, leave the Lo setting at C 2, and change the Hi value to halfway up the keyboard, at E 3.
4. Change the key range of the second device to cover the rest of the keyboard, which would be F 3 to G 8.
5. Play your keyboard and notice the split. Now that you know how to use this method, you can utilize more than two devices to create a split; indeed, you can literally have one device per key!

Adding Mergers
In order to have two or more sound devices within the Combinator, you must use an audio merger. That way, the sound is actually combined and fed into the Combinator input. To accomplish this, you can use a Spider audio merger, a reMix mixer 14:2, or a line mixer 6:2. When Propellerhead Software created the line mixer 6:2, they designed it to be smaller so it would take up less space within the Combinator.

One reason it's important to use a mixer is to have a volume control for the overall internal Combinator mixdown other than the one in your main mixer or submixer. It's somewhat annoying that the Combinator itself doesn't have a master volume knob, but there is good reason for this: The Combinator has no pre-amp, nor does it have a processing engine. Instead, the Combinator lets the devices that it contains create and process the sound. If you want to program a volume knob to control all the devices simultaneously from one of the rotaries, that can be done in the Programmer. You can also program a rotary to act as a master panning device by using the Programmer's Modulation Routing section to assign each pan knob on the mixer within the Combinator to a single source. Don't forget that the Combinator has its own mixer channel. For the mixer to which the Combinator is cabled, create a sequencer track so that the mixer is easily accessible when "lost" in the rackmountthat is, set in the middle of several devices, making it hard to find.

Tuning Combis
Because there are so many Combis to choose from in the Factory Sound Bank (FSB), it can be really hard to decide what's right for your song. To make it easier, all the Combis can be easily changed to sound more like the sound you are looking for. These new sounds can be saved as different Combis after you change the Combi setup to your liking. The problem with changing the sounds is that the setup of the Combi can be very complicated, and thus hard to understand. Here are some guidelines to help you out:

* Combis are simply devices, primarily from Reason Version 2.5, that you can use to further control CV and group parameter modulations with the Programmer. To change the individual sounds, the standard rules for tuning synths and samplers apply. Look for any pitch or filter modulations that the Combi may be using on any sound-generating device. If you can't control any of the knobs that have a rotary as a source in the Programmer, it may be that a CV is controlling them.

* The Modulation Routing section of the Combinator can be fine-tuned; simply change the range numbers to your liking. For example, suppose you have Rotary 1 controlling a filter on a MalstrOm. Instead of having it oscillate from 120 to 127, you can change it to a more midrange modulation—say, 30 to 60. If you want a parameter modulation to go in reverse, simply switch the two range numbers.

* There are two programmable sources in the Modulation Routing section. This means that up to three parameters can be controlled with one knob. To enable the use of more controls than this, try using the mod wheel on the sound-generating device as one of the targets, and then program the mod wheel with its various associated parameters, such as the FM Mod Wheel Amount knob on the SubTractor.

* If there is a Matrix controlling a rotary or button though CV, you might want to see if the Unipolar/Bipolar switch is set to cover the correct range of modulation.

* Both the key range and the velocity range can be tuned in the Key Mapping section. If you choose a split Combi from the FSB and you want to switch the side of the keyboard on which an instrument is played with the other instrument, do it from the Key Mapping section. This goes for changing the velocity range of an instrument as well.

* Combis whose names contain the word "Run" include pattern devices. Remember the Run Pattern Devices button on the Controller panel? This button activates all pattern devices, thereby acting as a separate Play button. It responds to the Stop and Play buttons on the transport bar.

* Multi-instrument Combis usually use a line mixer. If there is a sound that is not agreeable, you can use this line mixer to solo out instruments within the Combinator that you want to change. Use the line mixer for send effects within your Combi as well.

Basically, the trick is to mentally separate each instrument and to tune each instrument within the Combi individually. Some devices might be doing things they weren't originally intended for. For example, the Stereo Imager can act as a frequency splitter instead of a widener. Remember that some devices are there simply to act as modulators rather than to generate the sounds. By studying the Combis, you can begin to make some sense of Reason's infinite capabilities.

Combining Old Songs
Now that you have Reason, you might as well utilize its features to upgrade all the songs that you composed in previous versions of Reason. One way to do this is to put all the devices, wiring and all, into a fresh Combinator. That way, you can enjoy Reason's new methods of control over your songs—namely, the rotary knobs and the Programmer. You cannot place a Combinator within another Combinator, but if you need to have certain devices within their own Combinator, you can mimic that procedure.

Before you combine your old songs, I suggest removing any master insert effects. This way you can create a separate Combinator after combining your song to use as a master insert Combinator. You should then fill your new Combinator with all the goodies associated with the MClass label. I also suggest running all original devices through two individual Combinators so there's less wiring involved, making it easier to keep track of what's going where. Some excellent Combi patches intended strictly for sound processing are excellent for the overall mix. If you not sure what to use on your song for mastering, just browse through the Combi effects patches.

Here is the exact procedure for combining old songs:
1. Open the old song and, while holding down the Shift key on your computer keyboard, click all associated devices.
2. Right-click the selection and choose Combine. Everything should now be within the new Combinator. Alternatively, you can drag the selection to the Combinator, as shown in Figure 1.11.
3. From this point on, whenever you create a new instrument within the Combinator, you must manually create a sequencer track for it.

Here are some added bonuses that come with this procedure:
* Because you now have a sequencer track dedicated to the one Combinator, you can play your keyboard and see what it sounds like to have all your sounds from your song played at once.
* You can save what's now in the Combinator, which is basically your whole song, as a Combi patch. You won't retain the sequencing except what's in your pattern devices, but it's perfect for giving someone the file so he or she can do your remix!
* The Programmer in the Combinator can now control any knob in your song.
I recommend this for all songs written in previous versions of Reason. All controls are now limitless!

Modulation Routing Tricks
Here are some tips and tricks for using the Modulation Routing section of the Programmer on the Combinator. This is the section where you can set the controls for any parameter within Reason, excluding some of the NN-XT controls.

* Use CV routing to modulate the rotaries. Cable the Mod A or Mod B CV output from the Malstriim or the Curve output from the Matrix to any of the rotary inputs on the Combinator. Assign the target of any device parameter and watch the knobs turn by themselves! If you use the MalstrOrn, be sure the Mods are not set to One Shot so the knob will continue to turn without playing the song. If you use the Matrix, make sure the pattern is enabled so that it modulates the parameter continuously.

* Use the buttons as on/off switches for complex modulations. For example, if you use the Malstram's Mod A as an LFO to control Rotary 1, assign the Mod A On/Off button to Button 1. Then make sure that the minimum amount is set to 0 and the maximum amount to 1. Having the button turned on will turn on Mod A. This minimum/ maximum amount can be switched to reverse the button control.

* Although you can have only three parameters on a device mapped to one control, you can still have that control mapped to other devices. You can also map a control to the mod wheel (if the device has it), adding yet more controls!

* Certain functions—usually switches on processing devices—show up as a numerical value of 0-2 on the Min/Max section. Pay close attention to these parameters, because they can be very valuable for certain functions. These functions can be reversed as well by switching the values.

* You never need to program a button to turn on pattern devices, such as the Matrix or the Redrum. This is included with the front Controller panel. By simply clicking this button, you can turn on all these types of devices. The same goes for bypassing effects; it just takes one button.

Studying the Combis can help you realize the potential of modulation routing. To record automation of the individual parameters for each sample within the NN-XT.

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Detroit Electronic Music Festival 2007 Mixsets/Live PAs [06 Mar 2008|06:47pm]

richie hawtin - live @ demf 07 (28-05-2007)

steve bug - live @ demf 07 (27-05-2007)

mathew jonson - livepa @ demf 07 (27-05-2007)

seth troxler - live @ demf 07 (27-05-2007)

guido schneider - livepa @ demf 07 (26-05-2007)

kevin saunderson - live @ demf 07 (28-05-2007)

scan 7 - livepa @ demf 07 (26-05-2007)

misstress barbara - live @ demf 07 (27-05-2007)

ryan crosson - livepa @ demf 07 (27-05-2007)

angel molina - live @ demf 07 (27-05-2007)

baby ford - live @ demf 07 (27-05-2007)

monolake - livepa @ demf 07 (27-05-2007)

pier bucci - livepa @ demf 07 (26-05-2007)

lee curtiss - livepa @ demf 07 (27-05-2007)

audion & ryan elliott - live @ demf 07 (27-05-2007)

luciano @ demf 07 (28-05-2007)

butane @ demf 07 (26-05-2007)

gui boratto - livepa @ demf 07 (27-05-2007)

christian smith - live @ demf 07 (27-05-2007)

gary martin - live @ demf 07 (26-05-2007)

claude von stroke - live @ demf 07 (26-05-2007)

kate simko - livepa @ demf 07 (27-05-2007)

losoul - livepa @ demf 07 (26-05-2007)

michael mayer - live @ demf 07 (27-05-2007)

octave one - livepa @ demf 07 (26-05-2007)

pole - livepa @ demf 07 (26-05-2007)

sassmouth - live @ demf 07 (26-05-2007)

stacey pullen - live @ demf 07 (28-05-2007)

abe duque & blake baxter - livepa @ demf 07 (27-05-2007)

charles webster - live @ demf 07 (26-05-2007)
hit me with your best riff

A rolling stone gathers no moss? [01 Mar 2008|05:41pm]

[ mood | flyeriffic ]

Be there or be a fucking aubergine.

Love youse.

hit me with your best riff

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