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ape.Filter user manual

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1. 362 232 55 1 6 8 1476 395 55 1 6 7 922 746 55 1 6 6 576 716 55 1 6 5 360 488 55 1 6 4 225 280 55 1 6 3 140 800 55 1 6 2 88 000 55 1 6 1 55 000 55 1 6 You can obtain the same result using the next recursive formula fi f i 1 iterval_ratio where fi is the i th frequency i e f2 f1 1 6 55 00 1 6 88 00 e Harmonic COMPILER Harmonic COMPILER creates a bands grid according to the natural series of integer numbers 1 2 3 etc starting from a fundamental frequency There are two different ways to do it e base frequency in Hz fundamental frequency for the harmonic spectrum INHARM factor for spectral expansion compression You can stretched or compressed harmonic spectra putting the INHARM parameter at a non zero value In addition you can create stretched or compressed harmonic spectra putting the inharm parameter at a non zero value Stretched spectrum is obtained from values greater than zero while for compressed spectrum you have to select values less than zero F Fo n 1 exp The stretched or compressed spectra are created using the following formula where Fn n th frequency Fo fundamental frequency n index of Fn frequency exp factor of expansion compression inharm The inharm factor can vary from 0 2 0 2 When inharm 0 the generated frequency grid is pure harmonic e Scalar COMPILER Scalar COMPILER method distributes linearly all bands from freg min to freq max r
2. andwidth at 3dB from center freq a Bandstop center frequency gain and Q determining bandwidth at 3dB from center freq B E a Peak Notch center frequency gain and Q determining bandwidth at 3dB 3dB from center freq Low Shelf center frequency shelf gain and S determining transition width at 3dB 3dB from c freq High Shelf center frequency shelf gain and S determining transition width at 3dB 3dB from c freq Allpass center frequency gain and Q roughly determining 90 degree transition in phase on either side of center frequency roll a 10 dB the lowpass highpass bandpass and bandstop filters take just cutoff center freq and Q The filter gain parameter will always be set to 1 if the filter is not gain enabled for all filters shift click constrains gain cmd click sets gain to unit gain 1 0 the peaknotch and shelving filters take cutoff center freq gain and Q S Q damping coefficient S slope coefficient displaying phase response Tab numbers in the example is selected 7 select a detail filter band e cutoff gain N B filters band could be not progressive in frequency therefore you could have for instance the second cut off frequency upper of the follows bands and vice versa Three sliders below sets respectively cutoff center freq in Hz of the currently selected filter e gain in db of the currently selected filter Q resona
3. ange es e minimum frequency for the scalar spectrum e maximum frequency for the scalar spectrum Alternatively you can impose the gain top slider 19 dB in the example and Q below at all filter bands in the spectrum Also randomize all values 10 EQ 38 dB An hidden function in the equalizer multi slider is Equalizer gain 0 56 dB Mouse scrolling up down on the value you can change filters gains This value must be considered bi polar so in the above example 38 dB it means 38 38 equalizer range Snapshots e store recall A snapshot is a photo of graphical interface GUI parameters in the current state Each snapshots module can store up to 24 snapshots You can make transitions between two or more snapshots When you find an Interesting sound you can store new snapshot just shift click on a button in the snap pad thus the preset button will light orange You can recall a snapshot by simply clicking on the snap pad buttons all the widgets subscribed see later will be restored at current snapshot value immediately Alternatively a snap can be stored from pop up menu a message box will ask you to digit a snap number in see below for more explanations e micro pad The micro pad s goal is to obtain intermediate values between four snapshots interpolation You can configure four nodes snapshots locked lowest in the image nodes the mouse can only edit the nodes position and size
4. ape Filter Ableton Live Fx device user manual v 1 0 www densitygs com apeSoft 2011 MAXFORLIVE ableton Sey l 2 Starting system requirement installation Filter Equalizer general e compilers e cutoff gain e Fibonacci COMPILER e Geometric COMPILER e Harmonic COMPILER e Scalar COMPILER Snapshots e store recall e micro pad e transitions e snapshots manage e interpolation mode e transition curve Overview e save load device e info window e float window e history e acknowledgments 3 Starting e system requirement Macintosh ape Filter requires a Mac PPC or Intel machine running OS X 10 4 or later and 1 GB RAM Windows ape Filter requires a Windows XP Vista 7 machine and 1 GB RAM ape Filter requires Live 8 2 2 and Max For Live Details about Max For Live can be found at Ableton com This bundle will only work in Ableton Live not in the MaxMSP application Max for Live puts the power and potential of Max MSP inside Live Create all the instruments effects and extensions you ve ever wanted Go beyond the common and predictable and transcend the limits that conventional tools impose Build completely unique synths and effects create algorithmic composition tools or fuse Live and controller hardware into radical new music machines Join a society of makers and share ingenuity Max for Live was co developed by Ableton Live and Cycling 74 installation Macintosh Win
5. ave a notch at the Nyquist frequency and thus imitate the response of a analog filter The imitation analog filters are slightly more computationally expensive so this option disabled by default gain toggles the gain parameter for the lowpass highpass bandpass and bandstop filters The traditional definitions of these filters have a fixed gain of 1 0 but by gain enabling them their amplitude response can be scaled without the additional use of a signal multiply at the filters output log freq log amp and ph spect sets editor view range log linear frequency display log linear amplitude display e view phase spectrum pi to pi show parameters when mousing over the editor resets all filters gains BR 7710 30 select a region of the spectrum for frequency zoom Shift clicking to extends the range Command clicking Mac or Control double clicking Win and dragging shifts the current range up or down Option clicking Mac or Alt clicking Win and dragging up or down expands or shrinks the range Command double clicking Mac or Control double clicking Win selects the entire range Display filter coefficients only cor Lowpass cutoff frequency gain and Q Highpass cutoff frequency gain and Q JE E EE E Bandpass center frequency gain and Q determining bandwidth at 3dB from center freq Resonant bandpass with constant skitrtwidth center frequency gain and Q determining b
6. bers converges on the Golden Mean Ratio approximately 1 1 618 as its limit The computer music composition Stria by John Chowning was based in almost every structural aspect on this ratio Fibonacci compiler uses this sequence of numbers for compile the grid pitches You can set two parameters for the purpose to create a certain Fibonacci pitch grid You have to specify 1 The Nyquist frequency named after the Swedish American engineer Harry Nyquist or the Nyquist Shannon sampling theorem is half the sampling frequency of a discrete signal processing system It is sometimes known as the folding frequency of a sampling system The sampling theorem shows that aliasing can be avoided if the Nyquist frequency is greater than the bandwidth or maximum component frequency of the signal being sampled Wikipedia 2 Leonardo Pisano Bigollo c 1170 c 1250 also known as Leonardo of Pisa Leonardo Pisano Leonardo Bonacci Leonardo Fibonacci or most commonly simply Fibonacci was an Italian mathematician considered by some the most talented western mathematician of the Middle Ages Wikipedia 3 John M Chowning is known for having discovered the FM synthesis algorithm in 1967 In FM frequency modulation synthesis both the carrier frequency and the modulation frequency are within the audio band In essence the amplitude and frequency of one waveform modulates the frequency of another waveform producing a resultant waveform that can be perio
7. cur 1 you can enable or disable auto recall if enabled selecting a snap from the menu it will start the transition if disabled it will be only selected 12 reinit parameters resets subscribed parameters at default value only for the current streams or main save to disk current snapshots bank load from disk current snapshots bank client window open clients list to subscribe unsubscribe setting interpolation type and more storage window open storage window displays any stored presets open HV_PADS Hyper Vectorial Pads see HV_Pad open sequencer improviser unit see Snapshots Sequencer tab1 tab2 tab3 and tab4 open transition curve window see transition curve below speedlimit set update time limit for transitions and micropad nodes interpolations N B all the presets are saved in the Ableton Live project or device however you can manage individually save load snap bank for example useful for exchanging presets between the streams interpolation mode The two windows client and storage are both non interactive ANA Client Objects ape Filter Name Priority Interp Data ape Filters cascade 0 linear gt 405100 99 999992 1 30 000006 1 5 100 1325 1 30 000006 25 1002550 1 ape EQ 0 linear 0 0 0 0 client window accessible from the client window menu item see above shows the current subscriptions widgets list ever header name priority interpolation and data belong
8. dic or non periodic depending upon the ratio of the two frequencies Wikipedia 20 C C Ge DES base frequency in Hz i e fundamental frequency left slider 35 1 Hz for the Fibonacci spectrum first member of the series value of 2 in the example To complete the operation and generate the pitch grid you must press the compile button When auto is enabled changing value of one of parameters for each compilers methods grid pitch it will be automatically compiled e Geometric COMPILER Geometric COMPILER create geometric pitch space based on a ratio number For example if you specify 1 5 i e 3 2 you will create a pitch space grid based on the natural interval of 5th Since you can choose any relationship you can virtually have infinite pitch spaces Interval ratio defines a coefficient of multiplication this serves to build up the whole pitch grid while the fundamental frequency is the starting point from which grid take origin base frequency in Hz fundamental frequency for the geometric spectrum ratio for the geometric spectrum i e 1 6 also represented by the ratio 8 5 In the below example a geometric pitch grid is based on the interval 1 6 and fundamental frequency 55 Hz The grid will consist of a total of 13 bands starting from 55 Hz up to 15 481 1279 Hz as sketched in the next table of_FREQS Hz operation 13 15539 276 55 1 6 12 9675 702 551 6 11 6047 313 551 6 10 3779 571 55 1 6 9 2
9. dows e Download ape Filter zip e Double click to unzip e Drop to copy folder in Ableton Live Library Live Devices Instrument and or Fx 4 Filter Equalizer general ape Filter is an Ableton Live M4L device ape Filter can store and manage transitions micro pad nodes etc see snapshots for more details ape Filter 60e C oa met Cerere JE 70 8 ape Filter www densitygs com Alessandro Petrolati apeSoft 2011 You can dynamically employ up to 24 two pole filters i e biquad often referred to as second order sections In the Graphical filter editor you can set higher level parameters such as frequency amplitude and resonance Q or S through the mouse or selecting a filter band and using fine sliders 6 stream 1_filters biquad Tools bar is divided in three rows Follow explanation top down and left right on off switch to enable bypass filter peacknotch default menu sets a filter kind 0 display only 1 lowpass 2 highpass 3 bandpass 4 bandstop 5 peaknotch 6 lowshelf 7 highshelf 8 resonant see filter table at the end of this section 5 numbox showing 4 in the above image sets the number of cascaded biquad filters displayed in the filtergraph the range is between 1 and 24 default 4 analog toggles the analog filter prototype parameter for the bandpass and peaknotch filters Unlike the standard digital versions these imitation analog filters do not h
10. ing to subscribed widgets A few types of interpolation can be changed off no interpolation e linear Linear interpolation Presets recalled will be interpolated using a standard linear algorithm threshold Threshold Takes optional argument float which sets the threshold Presets recalled will recall data from the first preset specified when the fade amount is below the threshold and will recall data from the second preset specified when the fade amount is greater than or equal to the threshold e g threshold fade lt thresh value a fade gt thresh value b inverted threshold Inverse threshold Takes optional argument float which sets the threshold Presets recalled will recall data from the first preset specified when the fade amount is greater than or equal to the threshold and will recall data from the second preset specified when the fade amount is less than the threshold e g inverted thresh fade lt thresh value b fade gt thresh value a exponential curve Power curve Takes an additional argument float which sets the exponent to which the fade amount will be raised Presets recalled will recall data between the two specified presets along the curve described Power curves can be used to create faster or slower attacks and decays for the fade envelope table Table specified curve Takes optional additional argument see transition curve below which specifies the name of a table to use f
11. nce or S slope used for the shelving filters of the currently selected filter 1 filters biquad you can also set the cut off frequency opening the musical keyboard and clicking on the keys Keyboard control also the gain of selected band filter mouse clicking in different heights on each key If shortcuts is enabled pressing escape key you will hide keyboard If toggled the most right keyboard most right key switch the keyboard will be always in foreground disabled auto exit when click on a key e compilers Inspired to Eugenio Giordani Stria Multilevel Interactive Sound Synthesizer The lowest sections is considered improvisation unit You can experiment settings filters band with one of the COMPILERS red frame manage randomly or together gains Q purple frame and make manually an equalization for each band blu frame dry wet Son ss rane rand dry weet balance between filtered un filtered signal you can also set dry wet fade time in ms 25 ms default They increase complexity and coherence between the filters band space You can create your pitch space by the way of four different criteria of construction by determining a pitches grid The frequency range is depending by fundamental frequency and numbers of band filters When a frequency exceed Nyquist frequency i e SR 2 the filter band generation is inhibited e Fibonacci COMPILER Fibonacci COMPILER the ratio of two consecutive Fibonacci num
12. or curve lookup tab1 tab2 tab3 or tab4 Presets recalled will recall data between the two specified presets along the curve described in the table Tables are assumed to contain values between 0 and 100 representing the new fade amount 100 If the lookup fade amount does not fall exactly onto a table specified value linear interpolation is used to determine the new fade amount In Density m4l you have four draw functions to draw your curve see transition curve below 1 gsnaps will restore all Density m4l subscribed parameters at default values granular parameters and setup parameters 13 AMP Storage Slots ape Filter Name 1 2 6 12 14 ape Filters cascade 1000 4051 4051 4051 4051 ape EQ 0 0 0 0 0 0 0 0 O 0 0 0 00 0 0 0 0 0 0 The storage window displays any stored presets The active recalled preset is displayed in highlight green If any client value is changed are displayed in italics Eventually both of these windows will be configurable and editable so that they can provide display and editing control for clients and storage sets N B When you open storage window a message box will be displayed h f snapshots all GUI widgets must be stopped transitions HV_pads etc Do you want continue this is to avoid an overhead of CPU when displaying storage window transition curve All transition by defaults are linearly interpolation You can superimpose your interpola
13. pshot renumbers stored snaps seeeeeeo into consecutive beginning with 1 ape Filter recall pause resume transition toward selected snap stop transition reinit parameters clients at default value Alessandro Petrolati save snapshots bank to disk load poto mn www densitygs com float window Have you noticed ape Filter window have a top right tiny orange button toggle it to set float no float window When toggle float current window will be always in front This feature is saved on project like general setup e history ver 1 0 0 april 2011 new ape Filter independent device FILTER now is an indipendent device FILTER improved and refined FILTER enabled for transitions 16 e acknowledgments would like to thanks Eugenio Giordani and Nicola Casetta a special thanks to Renato Alberti Felix Petrescu and Pasquale Ascione ape Filter was created with MaxForLive http www cycling 4 com The Author Alessandro Petrolati apesoft All rights reserved 2010 2011 ape kagi com www densitygs com www alessandro petrolati it
14. tion curve at one or more subscribed widgets Four draw functions are available from toolbar or menu bar Open one of them and draw your curve This feature is saved on Density m4l device Draw a shape on the function window ANA tabl In order to maps one or more subscribed clients with one of four functions draw you need open clients window and select table from pop up menu over interp header Now you must digit the table name that can be tab1 tab2 tab3 or tab4 Name Priority Interp Data ape Filters cascade 0 table tab1 10000 in the above example we employ e tab1 like transition curve for ape Filters cascade and ape EQ See client window above for more explanations 15 Overview e save load device Just like an Ableton devices ape Filter device can be saved by clicking on the save button and digit a name You can load ape Filter Fx by dragging over an Ableton Live channel See Ableton Live user manual info window To avoid annoying hint messages when you are working on the GUI ape Filter sends all hints in the Ableton Live Info window In the clue window you can read some informations about ape Filter GUI widget when the mouse is over it See Ableton Live user manual umenu umenu store new snapshot using the next a A empty preset slot store snap number following optionally you can digit a m snapshot name in double quotes eeeeeeeen 0 clear selected snapshot clear all seeeeeeo sna
15. unlocked slider the mouse can only changes the slider location refresh highest in the image enable disable auto recall snapshots see below for more details e transitions 11 Transitions are a linear interpolation between two snapshots with one time duration You can start a transition only from the menu recall item Is important to understand that the transition occurs from the current parameters positions toward the selected snapshot actual Filter bands positions gt gt gt toward selected snapshot time duration You can change the transition time 2 00 sec in above image The transitions can occur simultaneously on the streams and or on the main If auto recall see above is enabled selecting from the menu a snapshot slot to start the transition e snapshots manage store new store snap number clear clear all renumber recall pause resume stop reinit parameters save to disk load from disk client window storage window tabi tab2 tab3 store new snapshot using the next empty preset slot store snap number enter a number or a list separated by space to store clear current selected snapshot clear all snapshot a message box will ask you to confirm renumber sort all snapshots stored into consecutive beginning with 1 recall start a transition toward current snap number pause resume current transition if occur stop transition cancel current transition if oc

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