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1. 42 TONEBOOSTERS 2010 2015 Gain visualization Start End Algo Slope SCEQ Listen HQ mode Mid side Tonal Vce start Vce end Abs thrsld The de esser attenuation as a function of frequency is visualized in blue Sets the de essing start minimum frequency in Hz Sets the de essing end maximum frequency in Hz Sets the algorithm used for calculating the de esser attenuation function e A value of 0 results in a broad band de esser e g all frequencies will be attenuated by the same amount if excess sibilance is present e Avalue of 1 results in a single band de esser e g all frequencies between the start and end frequency will be attenuated by the same aount of excess sibilance is present e A value of 2 results in a matched filter de esser which targets specific frequencies only in between the start and end frequency range that are causing excess sibilance e Anyvalue in between 0 and 1 will give a response in between a broad band and single band de esser e Anyvalue in between 1 and 2 will give a response in between a single band and matched filter de esser Sets the filter slopes in dB per kHz Lower values will give a smoother frequency response in the gain function higher values will allow more surgical processing in the frequency domain Enables or disables a side chain equalizer If enabled three equalizer sections will appear that allow modification of the results shown by the s
2. e Command key left mouse click set the control at its default value e Shift key left mouse drag fine tuning of the control e Alt key left mouse click mouse move jump to the clicked position e Mouse wheel change the value up or down e Shift key Mouse wheel fine tuning of the control e Left or down key change value down e Uporright key change value up e Double left click if the control has a numeric entry manual data entry Controlling nodes Some plugins have nodes round dots in a two dimensional space or graph that can be dragged to change parameters These can be dragged with the mouse to change their position and the associated parameter values The following key combinations apply e Left mouse click activate node e Right mouse click de activate node e Drag with left mouse knob modify position in active mode e Drag with right mouse knob modify position in inactive mode e Drag with left mouse knob and alt key lock Y while moving node e Drag with left mouse knob and control command key lock X while moving e Shift key drag with left mouse knob fine tune position in active mode e Shift key drag with right mouse knob fine tune position in inactive mode e Mouse wheel change secondary parameter for example filter quality e Shift key mouse wheel fine tuning of secondary parameter VU meters VU meters will often support peak hold fu
3. e When using the single band algorithm algorithm 1 the slope parameter determines the steepness of the filter around the sibilance range Larger values will cause steeper filter responses e When using the matched filter algorithm algorithm 2 the slope parameter determines the specificity of the filter larger values will cause more surgical cuts in very specific frequency ranges while smaller values will result in a more global reduction of sibilance TONEBOOSTERS 2010 2015 47 10 6 3 Mid stereo side processing Excess sibilance may exist in certain regions of the spatial image only One example is a complex stereo track in which sibilant vocals sit in the middle of that mix In such cases the mid side parameter can help to mainly process the vocals in the complex mix while leaving other elements largely untouched e Setting the mid side parameter to 0 will apply de essing to left right mid side equally e Setting the mid side parameter to mid 100 will apply de essing to mid only e Setting the mid side parameter to side 410096 will apply de essing to side only e Values in between will apply partial de essing to mid stereo or side depending on the exact value 10 6 4 Attack and release For certain languages the onset of sibilant sounds is very important for language intelligibility If a pass through of such onsets is desirable and de essing should only start a short time after such onset increase the attack
4. zero latency binaural speaker and room simulation over headphones Isone is best used with high quality full range headphones having a flat frequency response It features e Zero latency processing allowing for studio and live operation e Support of all sampling rates from 22 to 192 kHz e Loudspeaker designer to model speaker frequency response e Customizable room volume distance early reflections diffusion e Customizable loudspeaker azimuth angle 0 to 45 degrees e Customizable HRTFs strength head size ear size The user interface ROOM DESIGNER A GP a a A v3 0 0 registered GUI section Control Purpose Speaker Loudspeaker Shows the frequency response of the current loudspeaker model both on axis setup response display thick line and 45 degrees off axis thin line designer Tweeter size Sets the characteristic size of the tweeter and consequently the directivity of the loudspeaker Speaker angle The azimuth angle of the loudspeakers Best set to 30 degrees Channel mode Allows to down mix or solo the input channels for example to verify mono menu compatibility of the audio mix at hand Speaker presets Selects a preset loudspeaker model menu TONEBOOSTERS 2010 2015 11 3 4 3 4 1 Out Output VU meter indicating the overall output signal level Click to reset the peak hold meter Clipping may occur for signal peak levels above 0 dB Reduce the SpkLev parameter to prevent clipping if neces
5. 7 4 7 4 1 Filter type Frequency Gain Quality Q Mode Amount Dynamics editor Nodes Comp Soft Attack Release A R Make up SC Input Generic settings DCF In gain Auto phase Out gain Setting up and using TB FIX Spectrum editor Select the filter type of the current section low pass high pass bell shape etc Sets the frequency of the current section Sets the gain in dB of the current section Sets the quality factor of the current section section A higher Q value means a narrow bandwidth or a higher resonance depending on the filter type Determines whether the section applies its processing in stereo left only right only mid only or side only channels Sets the amount of processing for the current section 0 means that the filter is not being applied 100 indicates full processing Each equalizer filter section has a dedicated compressor input output curve This curve determines the compressor gain for a given input level and the curve can be modified with 3 nodes e Left click a node to activate it e Right click a node to de activate it e Dragthe mouse to zoom into an area for microscopic editing e Click elsewhere not on a node in the editor to zoom out The detected input level will be shown as a highlighted area under the compressor input output curve Sets the dynamics compressor functionality on or off Enables or disables smooth soft curves rather than hard kn
6. adjustment This works best by setting the channel mode to Left or Right Listen closely to the test material Ask yourself the following questions e Where does the sound come from e Dolheara well defined image or is it spatially blurred or ambiguous Rotate the head size knob until the sound position is most defined and natural and is perceived at 30 degrees azimuth HTRF strength adjustment The cue strength knob modifies the strength of the HRTF elevation cues If this knob is set to O no elevation cues will be inserted in the audio and the HRTFs will have a flat frequency response Higher values will insert more stronger elevation cues Depending on your own preferences and the audio content the cue strength can be adjusted as desired Note If the HTRF strength is set to zero the ear size setting will not have any effect CSC Crosstalk Spectrum Compensation The CSC switch allows to enable or disable this compensation filter Due to cross feed the signals of the right channel will not only be fed into the right ear but also into the left ear and vice versa Therefore the total signal power and the loudness will in most cases increase as a result of this cross channel summation The cross feed signal has a low pass character because to account for the acoustic shadow effect of the head Consequently without compensation such cross feed will result in a stronger increase in signal energy at low frequencies than at high freque
7. and ITU R BS 1770 LKFS loudness scale e Inter sample ISP ITU R BS 1770 compliant true peak detection e Support of all sampling rates from 22 kHz upwards e Stereo and 5 1 surround modes e Includes a separate compact plugin for stereo content only several features are excluded e Virtually unlimited integration time e Loudness history up to a maximum of 2 hours with hover and zoom functionality e Ability to sync with play pause of the DAW host if supported by host e Based on the VST 2 4 specification to allow compatibility with virtually all host programs This plugin comes as a set of two plugins e TB EBULoudness is a 5 1 channel plugin assuming channel order Lf Rf C LFE Ls Rs It can process stereo and 5 1 content provided that the host program is capable of running 6 channel plugins e TB EBUCompact is a stereo plugin and cannot process multi channel content GUI section Control Purpose TONEBOOSTERS 2010 2015 17 Integrated Integrated analysis loudness Loudness range True peak Meter mode Channel configuration Realtime Mode analysis Analysis Start controls Sync Reset Integration time K weighted LI integrated loudness across the full integration time expressed in LU LUFS or LKFS The G10 and G70 indicators will illuminate when the relative and absolute gates are active respectively not for ITU R 1770 0 K weighted LRA loudness range across the full integration time
8. can estimate the effect of a room on the content he or she is working on Nevertheless the room simulation module in TB Isone can be switched on or off if desired 16 TONEBOOSTERS 2010 2015 TB EBULoudness Loudness and true peak meter compliant with EBU R128 ATSC A 85 and ITU R BS 1770 The EBU published its Loudness Recommendation EBU R128 It tells how broadcasters can measure and normalize audio using loudness meters TB EBULoudness and TB EBUCompact calculate k weighted momentary loudness LM short term loudness LS integrated loudness LI and loudness range LRA compliant with the EBU ATSC and ITU specifications Furthermore true peak levels dBTP are monitored as well Besides compliance to loudness requirements the TB EBU Loudness plugin is also very useful tool to align the perceived loudness of different audio tracks for example on an album Differences in loudness expressed as loudness units or LU can be directly translated into attenuation or gain expressed in dB to align the loudness of two or more tracks Furthermore the loudness range indicator can provide valuable information to verify the dynamic range of a track and the potential need for dynamic range compression or expansion e Loudness monitoring metering compliant with ITU R BS 1770 ATSC A 85 EBU R128 and EBU Tech report 3341 e Loudness range LRA support according EBU Tech report 3342 e EBU mode LUFS EBU 9 EBU 18 and EBU 27 loudness scales
9. folder Users UserName Library Audio Plug Ins Components Restart the host and make sure that the plugin list is being refreshed or that new plugins are being activated before they become visible Please consult the manual of the host program how to proceed Sometimes hosts cannot find Audio Unit plugins if these are placed in subfolders of the Components folder In case your host does not detect the plugins and you have used subfolders try to store the plugins directly in the Components folder and re start the host To install the trial evaluation plugins go through the following steps Download the free trial evaluation zip archive from the ToneBoosters com downloads page http www toneboosters com download Unzip the archive double click to retrieve the plugins those will have the file name extension vst Copy these vst files into the audio unit folder Users lt UserName gt Library Audio Plug Ins VST Restart the host and make sure that the plugin list is being refreshed or that new plugins are being activated before they become visible Please consult the manual of the host program how to proceed Sometimes hosts cannot find VST plugins if these are placed in subfolders of the VST plugin folder In case your host does not detect the plugins and you have used subfolders try to store the plugins directly in the VST folder and re start the host The trial evaluation versions of these plugins have pa
10. or 32 64 bit operating systems 8 1 10 3 Can I use the same key for my Windows machine and my Mac 8 1 10 4 My registration key file does not work in Windows XP sseeeeeeeeee 8 1 10 5 Ican tenter the registration key in the Windows registry sssss 8 1 10 6 My Mac cannot open the registration key file eeeeeeennnne 8 1 10 7 Icannot use your plugins with a VST to Audio Unit AU or RTAS wrapper 8 1 10 8 cannot use your plugins in Avid Pro Tools eeeeeenneennen nnn 8 1 11 Acknowledgements sess nnne en nennt nni ri nain nias is sss sans 8 1 12 Disclailfiels zat n ra E Er SEP eed RETE TERR EORR UE TCU MERE FE VL R ERRAT URS Le o EXE ER 8 2 User interface common controls esee e eee eee nennen eene nnne nnne 9 2 1 1 Controlling Knobs and sliders eese 9 2 1 2 Controlling nodes sto rete ote eI E Eo Ie esae em s eO aet eue tew dte 9 2 1 3 Music 9 2 1 4 Bypass PUNCT OBI ode eet tee Ee tee aep ATE R EY REED P Gaede ETUR rex eds 9 2 1 5 dg I cime 10 3 TB AroIp le 11 3 1 IMEFOGUCUIOM ter Es 11 3 2 gc rm 11 3 3 Theuser interface erret hh a rna Ee eDe IE AEE 11 3 4 Setting up and using TB ISOMEC cccccecceeeesceseeeseeseesesnececaaaa
11. or minimum phase response the latter having a lower delay than the first 32 TONEBOOSTERS 2010 2015 When the auto phase option is activated TB FIX activates a novel method to construct the phase response which aims at combining the best features of linear phase and minimum phase Depending on the input signal and the desired frequency response TB FIX will fully automatically modify its phase response to anything from linear phase to close to minimum phase to give the best possible sound quality 7 4 5 FIX vs FIX4 external side chain FIX comes as a set of two plugins e FIX the default plugin with stereo in stereo out plugin This version of the plugin does not support external side chains e FlX4 this version has 4 inputs and stereo out The 2 additional inputs can be used as external side chains by selecting Ext 3 4 in the SC input control of the dynamics editor Please consult the manual of your host program whether external side chains are supported and how to enable them TONEBOOSTERS 2010 2015 33 gt TB Dither World s first quantization and noise shaping plugin that allows the design of your own noise shaping curve as easy as working with an EQ TB Dither is a plugin designed to modify the bit depth of audio signals for example when authoring a CD or for archival purposes with minimum quality degradation Such process typically involves dithering quantization and noise shaping TB
12. parameter This will cause the de essing to kick in later and allow pass through of onsets of sibilant sounds Similarly the release parameter determines how fast the de esser recovers from attenuating sibilant sounds A longer value tends to give a smoother behavior but a too slow value may cause the de esser to recover too slowly before a non sibilant syllable starts For very fast talkers a shorter release time may therefore be beneficial 10 6 5 Side chain equalizer SC EQ If more precise control is required to determine what frequencies TB Sibalance is responding to one can use the side chain equalizer SC EQ to shape the spectrum before it is analyzed for sibilance levels The side chain equalizer has no effect on the signal at the output of the plugin it cannot be used as a signal equalizer It only shapes the signal used to analyze sibilance levels Enable the side chain equalizer by clicking on the SC EQ button Three different equalizer section handles will appear which combined determine an equalizer curve that is applied on the signal before analyzing sibilance levels Drag the handles around to change the sensitivity to detect sibilance at certain frequencies e Like with all TB plugins using handles left or right click on the handle enables or disables the section e Use the mouse wheel to change the Q or bandwidth of each section e The integrated spectrum analyzer will always show the modified spectrum e g which inc
13. will never end due to the hiss simulation To resolve this either 26 TONEBOOSTERS 2010 2015 e Disable Include Audio Tail in the bounce dialog or e Reduce the tape hiss level on the plugin or e Inserta noise gate after TB ReelBus and adjust it such that the gate will close and remove the tape hiss at the end of the track TONEBOOSTERS 2010 2015 27 7 1 1 7 1 2 7 1 3 TB FIX Dynamic equalizer blending flexible dynamics processing and equalization in one optimized plugin Equalizer section TB FIX Flex combines equalization and dynamics processing in one go It works just as most equalizers it has 6 filter sections with lots of controls to modify their effect on the spectrum More than 30 filter types are currently supported which include classic analog peaking and shelving filters and resonating low and high pass filters Besides these conventional filter types some not so common or entirely novel filters are available as well e Bell shape filters that have a flatter top than analog filters to give a more natural sound e Non resonating shelving filters to allow for steeper filter slopes e Gaussian filters because these filters have the shortest possible group delay e Gammatone filters because they closely mimic the frequency analysis of our hearing system e Linear and logarithmically spaced harmonic filters for creative effects e Brick wall highpass lowpass and bandpass filte
14. 015 e The white line shows the overall equalizer curve in real time In the upper left corner of the frequency editor there is a small drop down menu for quick initialization reset of the editor 7 4 2 Filter types Filter type Auto node link LSF no res Bell shape HSF no res Rectangle Gammatone Gauss Harmonic lin Harmonic log Analog bell Analog LPFx Brickwall LPF Pink 3dB oct Spectral balance Analog LSF Analog HSF Analog HPFx Brickwall HPF Analog BPF Brickwall BPF Broadband gain Purpose description The auto node link filter type automatically adjusts its filter response to construct a smooth filter characteristic through all nodes that are configured as auto node link This way you can create many different equalizer curves by just placing 2 or more filter nodes anywhere in the editor The constructed filter type will be shown by a highlighted area Non resonating low shelf filter LSF The steepness of the transition is determined by the Q factor but the filter will not resonate as analog shelving filters do Bell shaped filter with a peak that is more flat than analog filters for a more neutral sound Non resonating high shelf filter HSF The steepness ofthe transition is determined by the Q factor but the filter will not resonate as analog shelving filters do Rectangular filter shape to boost or attenuate a very specific frequency range Gammatone asymmetric fil
15. 2 LU Yes 20 1 LU 1 dB FS LU K 16 v2 d LU Yes 16 1 LU 1 dB FS LU K 14 v2 LU Yes 14 1 LU 1 dB FS LU K 12 v2 LU Yes 12 1 LU 1 dB FS et ting up and measuring loudnes Include the plugin in the last stage of the master bus as insert plugin Make sure that no other audio processing is performed subsequent to the loudness measurement The loudness measurement plugin does not modify the audio signal it only performs real time metering Specify the desired loudness measurement method Use the mode drop down menu to select one of the supported loudness measurement methods standards Reset the meters by clicking on the reset button Determine whether you want to stop and start measurement of integrated loudness via the host sync enabled or via the plugin sync disabled Play the audio with the meters activated Stop the host and or plugin when the measurement period is finished Read out the loudness and peak values of interest After the loudness of a program is measured the required corrective gain in dB for loudness compliance can be simply obtained by taking the target integrated loudness and subtracting the measured integrated loudness G dB Lltarget Ll measured For true peak compliance it is advised to use an ITU R BS 1770 compatible peak limiter with true peak detection functionality such as TB Barricade TONEBOOSTERS 2010 2015 19 5 1 5 2 5 3 TB Barricade M
16. 50 will gradually link instantaneous peaks across channels as well Multiband Barricade features a fully automatic multiband limiting algorithm Opposed to wide band envelope limiting this stage processes individual frequency components For many types of content a certain amount of multiband limiting will result in more transparent limiter behavior in situations of very high signal levels or extreme limiting Setting the control to O will switch off the multiband limiter The amount of multiband limiting is visualized in the limiter gain VU meters In most cases the signal attenuation as a result of multiband limiting will not exceed 6 8 dB to ensure that the timbre of the audio content is not changed significantly VU meters and scales TB Barricade features RMS and peak output meters Peak meters indicate instantaneous digital peak maximum amplitude RMS meters indicate the average signal power with an exponentially decaying time constant of 300 ms Four different output scales can be used e Digital peak A full scale digital signal corresponds to 0 dB on the meters e K12 A full scale digital signal corresponds to 12 dB on the meters This scale is typically used for broadcast applications e K14 A full scale digital signal corresponds to 14 dB on the meters This scale is also typically used for CD mastering e K20 A full scale digital signal corresponds to 20 dB on the meters This scale is typical for DVD author
17. Dither supports industry standard dithering noise types such as RPDF rectangular probability density function 1 LSB wide and TPDF triangular probability density function 2 LSBs wide A GPDF Gaussian probability density function is provided as well TB Dither s uniqueness lies in the flexibility to shape and minimize the audibility of noise inherently introduced by bit depth reduction Instead of providing a very limited set of a few fixed noise shaping curves TB Dither allows you to design the spectrum of the quantization noise using familiar tools such as low shelf high shelf and peaking filters just as any equalizer This provides an unprecedented ability to adjust quantization noise spectra according to the audio content and envisioned reproduction system s If you can work with an EQ you can work with TB Dither To get started no less than 7 different noise shaping curves are provided and can be recalled from a menu ranging from threshold in quiet curves inverse dB A weighting inverse ITU R 468 curves and several more According to double blind tests the only audible effect when converting high resolution audio to a sample rate of 44 1 kHz and 16 bits is the injected dithering and quantization noise TB Dither resolves this by decreasing the quantization noise level in the frequency range the human ear is most sensitive to and thereby increasing the dynamic range for those frequencies beyond the 16 bit limit When working
18. If you have installed a 32 bit version of Microsoft Windows the host program will be 32 bit and you will need to use the 32 bit version of the plugins e f you have installed a 64 bit version of Microsoft Windows but your host program is a 32 bit executable you will need the 32 bit versions even though the operating system is a 64 bits version of Windows e Only if you are running a 64 bit version of Microsoft Windows and the host program is a 64 bit executable you should use the 64 bit versions of the plugins During installation of a host program you will often have the possibility to choose between a 32 or 64 bit version of the program when installing it on a 64 bit version of Windows e 32 bit hosts on a 64 bit version of Windows will be installed in a subfolder of C Program Files x86 e 64 bit hosts on a 64 bit version of Windows will be installed in a subfolder of C Program Files e 32 bit hosts on a 32 bit version of Windows will be installed in a subfolder of C Program Files Installation of the free trial evaluation plugins To install the trial evaluation plugins go through the following steps e After determining whether you will be needing the 32 or 64 bit versions of the plugins download the free trial evaluation zip archive from the ToneBoosters com downloads page http www toneboosters com download TONEBOOSTERS 2010 2015 5 Unzip the archive to retrieve the plugins those will have the file name exte
19. Toneooosters Plugins installation and user manual March 2015 Table of Contents 1 Setting up the plugins for first use cesses eese e eee eee enne nnn 5 1 1 System requirements cies iera ern in ii nisse s 5 1 1 1 Host Progra aiaa saecasctpeanscbacscazeatesedanececesaatca seageanacesescanecan NEE E 5 1 1 2 Microsoft WIN OWS REM 5 1 1 3 Yet M 5 1 2 10 01810 g ATTE 5 1 3 sitio p e EE E A E AE E 5 1 4 VST plugins on Microsoft Windows 32 or 64 bits esses 5 1 4 1 Determine what version you need susesssesseeeee eene ns nnns 5 1 4 2 Installation of the free trial evaluation plugins cccccccesscecesseccssseeceesseeeeesaeees 5 1 5 Audio Unit AU plugins for OSX 32 or 64 bits operating system 6 1 6 VST plugins for OSX 32 or 64 bits operating system eeeeseeeee 6 1 7 Toal eripe ONS irean aAa EE EE 6 1 8 Upgrading from trial to full versions cccccccccccecsecceceecceseesseeeseceseceesaeaaecseeeeeeess 7 1 9 Installing plugin updates ssesseeeseseeeeeeeeennne nnne enn 7 1 10 Frequently asked questions ccccsssssssssecseeeeceeececeeeesseeseesseeecesssaasasaaecseseeeeess 8 1 10 1 Dolneed a serial number or hardware key eeeeeeeeeenennnn 8 1 10 2 Do I need separate keys for Windows or OSX
20. and when the payment is cleared you will receive an automated email that includes a download link for the zipped registration key files This link will be available for 3 weeks and you will have a maximum of 3 download attempts e Download and extract unzip the registration key file s TB_PluginName key and place it in the exact same folder as the corresponding demo evaluation version of the already installed VST or AU plugin s This means that on a Windows computer in one and the same directory you should see the following pair of files for each registered plugin TB PluginName v3 dll TB PluginName key Similarly on a Mac the finder window should display for VST plugins TB PluginName v3 vst TB PluginName key For Audio Units you should see TB PluginName v3 component TB PluginName key e Restart the host program The plugin should now display registered in the lower right corner of the GUI instead of demo Please make sure to make a backup copy of this registration key file if the registration key file is lost or damaged the plugin will automatically downgrade to a demo version Your computer s harddrive is NOT a good place for a backup Do not rename the key file The registration key file comes in a zip archive Just unzip the archive and copy the resulting key file into your plugin folder Renaming or modifying the file will cause the registration key file to become dysfunctio
21. astering grade transparent highly customizable peak limiter with integrated dithering and perceptual noise shaping Introduction TBB arricade is a stereo mastering grade peak limiter which supports control over the attack and release times look ahead time and includes a quantization dithering and perceptual noise shaping module to deliver high qual ity delivery signals with limited bit depths It is especially suitable to generate pristine final delivery signals for CD DVD online delivery broadcast or podcast applications Fea tures Fixed delay 1023 samples Adjustable input and output gains Adjustable look ahead attack and release times Inter sample ISP ITU R BS 1770 EBU R128 compliant true peak detection and limiting Supports both waveform and envelope limiting Highly transparent limiting even with very high input levels Peak hold VU meters with adjustable scales K12 K14 or K20 or digital peak Peak hold RMS meters Quantization dithering and perceptual noise shaping module Support of all sampling rates from 22 to 192 kHz Based on the VST 2 4 specification to allow compatibility with virtually all host programs The user interface GUI section Control Purpose Limiter gain Env Displays the limiter envelope reduction in dB including peak hold Click on the scale to reset the peak hold function Multiband Displays the signal amplitude reduction resulting from multi band limiting Clic
22. at TB Isone does not contain or employ measured characteristics of existing loudspeakers but instead relies on analytical theoretical models of loudspeaker cabinets including the size volume driver type resonance frequencies enclosure type and so on Flat Reference setup consisting of an essentially flat frequency response and speakers placed at and 30 degrees azimuth with directionality kicking in at around 3 kHz HiFi speaker Typical HiFi loudspeaker with a broad frequency response and a small boost at 60 Hz and 20 kHz _ Small monitor Typical small single driver stereo loudspeaker setup with a relatively narrow response and high directivity Monitor A A model that represent popular commercially available near field speakers Monitor B A model that represent popular commercially available near field speakers Monitor C A model that represent popular commercially available near field speakers Portable Typical frequency response of a portable stereo audio player with speakers placed closely together Laptop Very small loudspeaker simulation producing high frequencies only Flatpanel Simulation of a flatpanel TV watched from a distance Mono radio Single driver mono and band limited loudspeaker simulation as found in mono portable radios Too much Very wide loudspeaker setup 45 degrees azimuth with significant bass and treble boost Starting from scratch Build your own speakers TB Isone has a speaker setup de
23. d high quality modes If a bus signal or full track needs de essing great results can be obtained by de essing the mid channel only this is typically where the vocals are while leaving the side signals untouched Mid side mode of operation is available on TB Sibalance as well as a control to engage a high quality mode 10 2 5 Signal level dependencies It can be very desirable to reduce excess sibilance for relatively loud parts of a track while leaving less loud elements untouched The level threshold control of TB Sibalance influences how sibilant low level signals are Basically if the input signals approach the threshold set for level the measured sibilance will gradually be reduced and hence the amount of de essing will become more subtle or even absent 10 2 6 Processing of full mixes Ideally a sibilance tool is sufficiently flexible to also process full mixes for example to catch excess esses in a mix or simply to reduce the mix s harshness This is why you ll see controls to change the voiced frequency range analysis for mix processing these can be set to cover almost the complete audible frequency range TONEBOOSTERS 2010 2015 41 GUI section Control Sibilance Input output input output function function Threshold Ratio Knee Range Dry wet Analysis Attack Release Spectrum Spectrum analyzer analyzer Purpose The sibilance input output function sets the output sibilance as a funct
24. d to have the following two properties e The quantization noise due to bit depth reduction is not correlated with the audio signal and e The 2 moment power of the quantization noise due to bit depth reduction is not correlated with the audio signal In practice this means that the effect of quantization and dithering is a steady continuous low level white noise signal that is independent of the input audio signal RPDF dithering RPDF rectangular probability distribution function dithering noise has a lower noise level than TPDF and GPDF noises However the drawback of this particular dithering noise type is that although the quantization noise is not correlated with the audio signal its 2 moment still is In other words the quantization dithering noise level will fluctuate with the audio signal itself which can be an undesirable property Noise shaping The goal of dithering and noise shaping is to modify the spectral properties ofthe quantization noise introduced by bit depth reduction in such a way that it becomes less audible Although due to information theoretic constraints the total amount of noise cannot be reduced one can exchange a lower noise level in one frequency region for higher noise levels in other regions This trade off is provided by the noise shaping editor which basically works like an equalizer operating on the quantization noise only A reduction of quantization noise in a specific frequency range will au
25. de processing and high quality modes eese 41 10 2 5 Signal level dependencies ert ERES ERE EE TEES Rer 41 10 2 6 Processing OF full MIXES were certo et tete E Teen EVE E eae EEE eren Tut 41 10 3 serinterface 2 5 rr E Le ee op ee Eee eb ALEEA EATERS 42 10 4 Understanding excess sibilance sssini iasau nennen eene nnns 44 10 4 1 Voiced and sibilance frequency ranges ueeseessesseeeeeneene enne 44 10 42 Sibilance level eno rere iube Pede eee Crede terrre eee ee ec rr eee e OR read et 44 10 4 3 Absolute threshold tr eben cec Ne eet terrere ceu eae 44 10 4 4 Tonal and noise sensitivity liess nnne nennen nennen nina 45 10 4 5 Sibilance level sumimaty iiec eer YER RE EXPE TES Ier 45 10 5 Reducirg sIDI anice uere ertet a oe Ete hana ERE ERR eere EE ree Ud 46 10 5 1 Sibilance input output graph ccceecceesscecesseccsseeceesseeeccsaeccesseeeeeeseeeeesaeeeees 46 10 5 2 Set a maximum reduction in sibilance esee 46 10 6 Algorithm t ning 2 ori n b EUR RERO TERRE E E EER EREN 47 10 6 1 Broadband single band or matched filter eeeeeeeeenene 47 10 6 2 Filter slope i er ester te ate ive EE iue SI ord 47 10 6 3 Mid stereo side processing itri a REEE EE 48 10 64 Attack andirelease iere etri eee ree et a aep na Ret aa Ee deae erra a 48 10 6 5 Side chain equalizer SC EQ cccccccccccssssscceeceeeseseeeeccesessssseecessessesaeeeeseeses
26. djustment The changes in timbre and dynamics of TB ReelBus can be separated into a static spectrum part which is can be understood as an equalizer that imposes a certain tape frequency characteristic onto the audio signal The amount of this effect can be changed with the spectrum control Besides such static characteristic TB ReelBus also simulates the frequency and level dependent saturation and spectrum of tape The amount of this effect can be controlled by the saturation control Higher values will give a more pronounced effect Both the spectrum and saturation controls are offsets relative to the selected device model Wow and flutter Wow and flutter are the result of small variations in tape speed that cause changes in pitch frequency The wow and flutter simulation can be switched off by setting the controlto full left Full right gives twice the amount of wow and flutter of the selected device model Bias and overbias Tape recorders add a very high frequency bias signal to the incoming audio before recording the combined signal to tape The bias signal has a frequency that is typically around 100 kHz or higher and improves the response of the tape Low bias levels will give a brighter timbre as the high frequency response is more neutral at the expense of more harmonic distortion at lower frequencies Hence the bias control provides a trade off between high frequency response and low frequency distortion C
27. e compressed Determines the amount of linking between left right compression in left right mode or mid side in mid side mode A value of 100 results in full coupling of the compression in both channels a value of 0 gives fully independent compression operation in both channels The setting of this control will be stored independently for M S mode disabled and enabled Adjusts the side chain input level balance In left right mode this knob works like a left right pan knob on the side chain determining the left right balance adjustment going into the side chain level detector The same applies for mid side in the mide side mode The pan control will remember its setting independently for mid side and left right modes The display at the center of the GUI gives a graphical representation of the compression curve and the current input level The handle attached to the curve can be used to adjust a few basic compressor settings e Drag handle to change the threshold e Use the mousewheel to change the ratio e right button click on the handle to engage bypass Apply a left button click on the handle to disable bypass Click on the small downward triangle in the upper left corner of the display to change the display mode e The I O mode presents compressor output level as a function of input level e The Gain mode presents the compressor gain as a function of input level The lower half of the display shows a real t
28. eaeeeeeeeeeees 34 8 3 Dithering and information theOry ccccccscccccccccccceccecccssceseessseseeeeeecsscceaaeseeeeeeeees 34 8 4 FOACUIES isione arnie errre aea aa A E EAEE AE EE E bons euneeeees 35 8 5 Useri nterfaCe s eiecit rA E AEAT 35 8 6 Typical workflow for dithering and noise shaping ccceccecceeeeeeesseeeeesseessentenes 36 8 6 1 Determine the desired bit depth c cc ccccccscccccceececceccssceseesssesseeseesssaeaaesseeeeeeees 36 8 6 2 Insert TB Dither as the very last plugin in the processing chain 36 8 6 3 Choose the dithering and noise shaping settings sseeeeeeeeneere 36 8 6 4 34e 37 9 TB BusComppressSQFr oos co asc he eea cosine ideas ERR ERRRUA RAF SEES ES RAPSA PAR Rd S 38 9 1 ACO DUCE OM eee ck eaves Maeaed sues atsda ch eaves tues Sdaeneaeae 38 9 2 User interfdCe i ice neret ccd snes conieci ia supra ge vane E acu mua caved edu Rd aou 38 10 TB Sibalance 5 eerte E orsa ve eso vu vn cales ao secede E 41 10 1 ALO GU iO mt 41 10 2 FO QQUIES e Pn 41 10 2 1 De essing like a compressor lsesssssssssseseeen nennen enne nnn nnne 41 10 2 2 Algorithm TUsiOn ice aaa VL dud M e eva RENT Pane EUR ERR IIO 41 10 2 3 Tonal component sensitivity issesssssssseeseeeeeeene enne nennen enean 41 10 2 4 Mid si
29. eaeeeeess 48 10 7 Excess sibilance in signals other than VOCAIS ccccccscceseeseeeeeeeseesescneaaesaeeeeeeees 48 4 TONEBOOSTERS 2010 2015 1 1 1 1 1 2 1 1 3 1 4 1 1 4 2 Setting up the plugins for first use Host program The plugins require a host program that supports the VST 2 4 specification or a host program that supports Apple OSX Audio Units Microsoft Windows e Windows 7 SP1 or higher e Both 32 as well as 64 bit plugin hosts are natively supported e Intel processor Pentium 4 1 GHz or higher with SSE instruction support e AMD processor AMD Athlon64 1 GHz or higher with SSE instruction support Apple Mac e Mac OSX 10 5 Leopard or up e Mac OSX 10 10 Yosemite requires plugins version 3 0 9 or up e Intel based processor running at 1 GHz or higher If you have any problems installing or using any of these plugins please consult this manual before contacting us via our website at www toneboosters com TB VST plugins are not supplied with automatic installation programs We want you to maintain full control over what is stored and modified on your dedicated audio workstation This means that you will have to copy the plugin files into the correct folders manually Determine what version you need For Microsoft Windows the plugins are available as for 32 and 64 bit host programs You will first need to figure out whether your host that uses the plugins is a 32 or 64 bit program e
30. ecifications and may therefore not operate correctly 1 10 8 cannot use your plugins in Avid Pro Tools Unfortunately Pro Tools does not support open plugin standards such as VST or Audio Units e ToneBoosters would like to thank Nigel Khan and Yann for their support in compiling this manual e Audio Units implemented using Symbiosis from NuEdge Development e K xx scales in the EBU Loudness meters are based on ideas and suggestions from Roland L hlbach Compyfox Studios VST is a trademark of Steinberg Media Technologies GmbH 8 TONEBOOSTERS 2010 2015 2 1 1 2 1 2 2 1 3 2 1 4 User interface common controls Controlling Knobs and sliders The various knobs and sliders on the graphical user interfaces GUIs of the plugins can be controlled by left mouse clicks for switches or left mouse drags for rotary controls and sliders The following key combinations apply that modify the behavior of the GUI elements Windows e Control key left mouse click set the control at its default value Threshold e Shift key left mouse drag fine tuning of the control e Alt key left mouse click mouse move jump to the clicked position e Mouse wheel change the value up or down e Shift key Mouse wheel fine tuning of the control e Left or down key change value down e Upor right key change value up e Double left click if the control has a numeric entry manual data entry OSX
31. eeeees 26 Bounce tracks with TB ReeIBUS cccccccccccecceseeseseseeseeeceeececeeaeeeeesesecseseeeeeeees 26 Bouncing tracks in Apple Pro Logic eseeseeeeeeeneenenen nennen nnns 26 TO WU Sisi HON HESIODI ODSHOIUte 28 IntFOGUCctiOT usce beni eco caca emaxe nete me xu Fa Y POE Ter EXE Spa Yd wa d XV DEA Rl AM e n Eud ERE RA 28 Equalizer section idt o rte pe sa e Rope ever e dde reu s 28 Linear phase or minimum phase sesssssseeeeeneee nennen enhn nnn nnns nnns 28 Dynamics processing without limits eeeseceseseee eene 28 Putting itall together ie nmi rage n AASA 29 dere te 29 The user interfata rain de kk eara ap iUnd eaa C EX vr Rd Exe EU ra ad E Ea 29 Setting up and using TB FIX cccccccccecceecesceseseeeeeeseeeeecesaaaaaecaeceeeeeeeeeeeeeeeeeseeseees 30 TONEBOOSTERS 2010 2015 7 4 1 Spectrum editor tte eet e dcdit gau iss shisdesauues ena a EI ES doe deem uu eae A RS 30 7 4 2 Filter typesnitt Mies goat ccageena cu eR ML aueds sued E EEE Ep EAEE UE sa ue RR 31 7 4 3 COMPESS OMECITON 2 4 et 32 7 4 4 Auto phase optiOr ce a er Y ERE RON SEES ERES E SERE XE AERE deeedeetsecuas 32 7 4 5 FIX vs FIX4 external side chain eeeseeseeee eene 33 8 TB DICQe EID DTE ILIILLLLO LL LIC 34 8 1 WIMEO GUCCI OM et RR 34 8 2 Audibility of sample rate and bit depth reduction cccccccssessessssst
32. eeees 15 Loudspeaker SiMUIAation ccccssssssscsecceecaeeeeeeeeceeeecessessesssseseeeseeasausaaaaecaeeeeeeees 16 Room acoustic modeling cccceeeesesssccnecaeeceeeeececeecessceseesssesceseessaasaaaaesaeeeeeeees 16 TB EBULOUGnBSS ciini eoe eia oa eo pEcoP Roe sun tes ERR ERE QUSOR ER EON EDD T raw ES 17 IntrOdUctiOr nettes rnn da deed sv asas er oaa edu ceder CXv ase od Vara vix ner EO esa Eos rd a Tu Ruo 17 Id MPIIPSERREEL ELT II 17 TB EBULoudness and TB EBUCOompoact eeeseesseseeeenen nennen enne eene 17 The USEF Interface uci ec ai oov ara en epi rear DoD a aer e user P e gor ra ka ue D Ede 17 Loudness standards and target lOUdNESS ccccccccceceesceseeseeeeseeenecsensaeauecaeeeeeeees 18 Setting up and measuring loudness esses nennen nennen 19 IB BartiC3QUBsco sinit ietriec Up Ee ESSET ER NRIO RA NR INR ENRIS SCENE LCS PNACUSDNE INE FERAE 20 IntrOGQUCtiOT s iciciooe ie esu ooi ato tor d icona pacer eder tin A DONE Ya ahaa a sendass dida rada cadat 20 FeatUf 65 aid c ses waniowaiiceins o itaq idus usu Un eu de Ges Sv url o uu V o sa RSS E E deen 20 The user interface stes Pecesa voc co Dev ia dee ci peo Side rd a Ve veda Eva dera ave etae 20 Setting up and using TB Barricade ccccccceseesseeeeeeseecsecsenaeaaecaeeeeceeeeeeeeeeeseeseess 21 Input gain and output limit llseseseessseeeee eene enne nennen 21 Lookahead attack and release times eseess
33. een 1 and 3 milliseconds will generally suffice Part of the character and transparency of TB Barricade results from its intelligent algorithm that discriminates between instantaneous short peaks or overs and long term loudness increases that result in many consecutive overs TONEBOOSTERS 2010 2015 21 5 4 3 5 4 4 5 4 5 5 4 6 5 4 7 e Instantaneous sporadic overs are limited by fast reacting limiting action which is determined by the lookahead time e Long term loudness increases resulting in many or consecutive overs are limited by longer term loudness estimation The attack and release times of this loudness analysis are set by the attack and release controls e Along attack time will result in a slow reaction to loudness increases and will typically result in more loudness at the output of the limiter e A short release time will quickly recover the limiter from loud passages resulting in more loudness at the expense of a risk of breathing pumping artefacts Stereo link If the limiting gain is different for both audio channels the spatial image of the audio content may shift towards the center position To prevent distortion of the spatial image TB Barricade allows to link the limiter action between the channels e Stereo link values between 0 and 50 will gradually link loudness estimates between the channels but allow the limiter to still process instantaneous peaks in channels individually e Values above
34. ees Sets the attack time of the equalizer compressor section Sets the release time of the equalizer compressor section The value is ignored when A R Auto Release is enabled Enables or disables the Auto Release A R mode Sets the make up gain in dB of the dynamics editor Selects what signals are used for level detection side chain input The dynamics processor can detect stereo levels but also only operate on mid side left right or the side to mid ratio FLX4 has the additional option to use external input 3 4 for level detection external sidechain Enables or disables a DC reject filter If enabled frequency below 5 Hz will be removed from the plugin s output Sets the input gain in dB Enables or disables the Auto phase feature of the plugin Sets the output gain in dB The spectrum editor works very similar to any equalizer plugin It supports up to 6 filter sections with individual controls to set the filter type the filter gain its frequency and bandwidth quality Q factor Just activate left mouse click or de activate right mouse click a section and drag it to the frequency gain combination that is desired e You can freely re order the nodes e You can draw a rectangle in the editor to zoom in e Click anywhere but on a node to zoom out and or to bring up the generic settings display e The highlighted area displays the filter curve of the selected section 30 O TONEBOOSTERS 2010 2
35. epth of the host must be set to the exact same value as used in TB Dither e g export as 16 bit PCM if TB Dither was set to 16 bits TONEBOOSTERS 2010 2015 37 TB BusCompressor High quality transparent dynamics processor with adjustable knee and auto release functionality suitable for single tracks as well as complex mixes TB BusCompressor is a very transparent musical all round dynamics processor designed to be able to handle everything from single tracks to complex full mixes Even with ultra short attack and release settings harmonic distortion is extremely low often better than 150 dB re FS and CPU load is typically below 0 5 depending on hardware TB BusCompressor has the unique feature to set the compressor hold time in cycles rather than in seconds This dramatically reduces intermodulation distortion even with ultra fast attack and or release settings Expression of the hold time in cycles creates longer hold times at low frequencies at which one cycle has a long duration while still having a very fast response at high frequencies Another unique feature is to adjust the compressor sensitivity to noisy as opposed to harmonic signal components TB BusCompressor s advanced signal analysis toolset includes the separation of tonal harmonic and noisy percussive signals Therefore you can control the relative amount of these signal types that the compressor responds to For example in a certain situation you might
36. es of the head and ears pinnae The head size has the strongest influence on the binaural cues inter aural time and level differences Hence a mismatch in head size often results in a wrong azimuth but can also result in an ill defined sound source position or an unnatural sound percept The ear size has the strongest influence on the elevation cues peaks and throughs in the spectrum induced by reflections in the ear Hence a mismatch in the ear size often results in a lack of externalization or sources erroneously perceived from above TONEBOOSTERS 2010 2015 15 3 7 3 3 7 4 Loudspeaker simulation Besides HRTF adjustment TB Isone also allows to simulate a variety of virtual loudspeakers Instead of simulating specific loudspeaker models the approach taken in TB Isone is to simulate characteristic common attributes of loudspeakers instead of accurate simulation of specific models Room acoustic modeling Simulation of the acoustic environment is essential for a compelling simulation of loudspeaker listening over headphones Music or other audio content is almost never listened to in an anechoic environment and those who have experienced audio playback in such anechoic rooms know that this results in a very unpleasant listening experience Moreover your audience will listen to the content you work on in the car in the living room or any other echoic environment and hence it is crucial that the audio producer or engineer
37. eseeeenennnnnnn nnns 21 Stereo iNK RI IT D CITIES DUUM 22 Multiband uiii sus usu oae eit Eee amebganeesdodsabiaaeat pneu rdc Ea der ce qo pw add dr ad oes 22 Vly meters and scales 2 2 tasto sxadedetddeaya maaelacersiduaedteadasa va EROR EN Yo dd 22 nima RMS 3 M 22 kp Eiaa 22 DC reject titer ine i ete pars er ure eausa e ex uae Estee ens eu EE REEE 23 OUtPUt FESOIUTION EE 23 MB REGU BUS een 24 Introduction ecce en ada ova ec PE CEU i danced Vr CEU E Cv rcv 24 dri lk E 24 The user interface io cce civic eo uec veu eta cocer rcu e a T VR DEC Ee dva ER E 24 Setting up and using TB REEIBUS cccccccecceseesceeeeeeseesenneeaaecaeceeeeeceeeeeeeseeeseeeeees 25 Signal level dependencies rio en tet e er vv Y EEEL 25 MU tReters iecore eoe E Une vxo ed es ee ai ed ee USE SU ua Eo ee TRU ab ee E VPE SERES NU TY OS SOR UE UUY 25 Deviceimodels 5 t ibn edel cvi e Cueva cd ved er a a CO PRA CE CU Qe vM RR EE 25 Noise SOUNGE S votes te ibs Bick de e epa Waseda acs ERROR E VER ovn ae CV a C EY 26 Color adjustment e Rt i estet Gain edie hha EEEE 26 Wow and ifl tter eteci ceci oci tta a Cu EN va PR ako end VR C EN TR Ta ab RN 26 Bias and OVverblas i coitu dco ea coe ek Cu P TE Uva D TR E VR CEU Ee ev RR a 26 Si oic 26 Pre emphasis and post de emphasis cccccccecceccecceseeeseeseseeseeeeeeeesaaesaeee
38. evel works in the same way as conventional de essers without specific sensitivities to noise or tonal signals 10 4 5 Sibilance level summary In summary the sibilance level is determined as power spectrum levels within the sibilance frequency range relative to e the average level of noise like components in the voiced frequency range and e the absolute threshold level set by the user It should be noted that the examples and description in these sections are very schematic and exemplary of nature to outline the concept of sibilance the actual algorithms in TB Sibalance are much more sophisticated than visualized here TONEBOOSTERS 2010 2015 45 10 5 1 Sibilance input output graph The sibilance input output graph provides a wide range of controls to modify sibilance levels The input output graph shows some similarities with input output graphs shown on the ToneBoosters compressors However with TB Sibalance the input output graph shows the input sibilance level along the horizontal axis and the desired or output sibilance level along the vertical axis The units are in Decibels e The reduction in sibilance can be thought of as the difference between a dashed line that connects equal input and output sibilance and the actual input output curve shown by the solid line That amount is set by the ratio parameter A higher ratio will result in a stronger reduction of sibilance e The threshold determines the input sibilance le
39. expressed in LU LUFS or LKFS The numbers below of the loudness range indicate the 1096 and 9596 percentiles of the short term loudness distribution Maximum true peak dBTP observed since the last meter reset The number blow the true peak value will indicate the PLR peak to loudness ratio Sets the display and metering modes to one of e LU EBU R128 2014 e LUEBU 9 e LUEBU 18 e LUEBU 27 e LKFS ATSC A 85 2013 LUFS EBU R128 TB 3 LKFS ITU R BS 1770 0 LKFS ITU R BS 1770 3 LU K20 v2 20 LUFS LU K16 v2 16 LUFS LU K14 v2 14 LUFS LU K12 v2 12 LUFS LU K16 v2 d 16 LUFS Select 2 0 stereo or 5 1 surround metering configuration For 5 1 surround the channel order must be front left front right center LFE left surround right surround Selects the real time analysis mode e VU meters shows momentary and short term loudness VU meters as well as true peak meters for each audio channel e LS time Shows the history of observed short term loudness values Time indicates the range from most recent value backward In this mode the following interactions are enabled o Hover if one moves the mouse pointer over the plot the loudness value corresponding to the x coordinate of the mouse pointer is given o Select by left mouse click and dragging a selection of the curve can be made for a zoom detailed view of the data o Left mouse click without drag to zoom out completely Start continue t
40. fects VU meters Similar to real analog VU meters the VU meters of TB ReelBus do not represent digital peak values Instead they compute averaged signal levels with averaging time constants that are in line with those of analog VU meters The meters are calibrated to have a reading of O dB for a 1 kHz sinusoid with an RMS of 20 dB FS Device models TB ReelBus contains settings for several tape recording machines device models Some of these models are intended for high quality use while others have a more dramatic effect As these device models are all carefully TONEBOOSTERS 2010 2015 25 6 4 4 6 4 5 6 4 6 6 4 7 6 4 8 6 4 9 6 5 1 modeled according to real tape recorder units every device model has its own tape hiss asperity noise saturation spectrum circuitry clipping and wow amp flutter properties It is important to note that the controls on the user interface will always be offsets relative to the selected device model Noise sources TB ReelBus simulates both tape hiss which is an additive type of noise as well as asperity noise which is a modulating noise that depends on the input signal level Asperity noise is in part the result of inhomogeneities in the tape oxide coating presence of dust particles and other stochastic influences Both the tape hiss and asperity noise have been carefully modeled for each device model and will therefore change if a different device model is selected Color a
41. he integrated loudness and loudness range measurement Enable or disable the pausing of the integrated loudness and loudness range meters if the host DAW stops playback only for hosts that support this function Reset all meters Amount of time used for integrated loudness and loudness range measurement Not available in the EBUCompact meter 4 5 Loudness standards and target loudness Both EBU R128 for Europe and ATSC A 85 for USA are both based on loudness metering defined in ITU R BS 1770 The target loudness gating mechanism the loudness units and maximum allowed true peak levels are 18 TONEBOOSTERS 2010 2015 nevertheless different as indicated in the table below If true peaks or the integrated loudness value are outside the valid range the plugin will display the values in red instead of green Please note that the values below are taken from the 2011 2012 versions of the standards please consult the respective documents to verify that these values are still correct Mode Loudness unit Gating Target loudness Maximum true peak LUFS EBU R 128 2014 LUFS Yes 23 1 LUFS 1 dB FS LU EBU 9 LU Yes 0 1 LU 1 dB FS LU EBU 18 LU Yes 0 1 LU 1 dB FS LU EBU 27 LU Yes 0 1 LU 1 dB FS LKFS ATSC A 85 2013 LKFS Yes 24 2 LKFS 2 dB FS LKFS ITU R BS 1770 0 LKFS No 23 1 LKFS 1 dB FS LKFS ITU R BS 1770 3 LKFS Yes 23 1 LKFS 1 dB FS LUFS EBU R128 TB 3 LUFS Yes 23 1 LUFS 3 dB FS LU K 20 v
42. high frequencies The speaker angle represents the azimuth angle of the loudspeaker A value of 30 degrees indicates that the left and right loudspeakers are placed at degrees azimuth from the listener s point of view 14 TONEBOOSTERS 2010 2015 3 7 1 3 7 2 The channel mode menu allows to solo the left or right loudspeaker or to create a mono down mix that is subsequently reproduced by both virtual loudspeakers dual mono mode The preset menu contains a list of presets for the speaker setup designer that may be good starting points for tweaking Sound localization cues A virtual listening room is typically created by simulating the acoustical transfer from all loudspeakers to both ears These acoustical transfer properties are often referred to as Head Related Transfer Functions HRTFs Such HRTFs can be measured for each individual using specialized equipment The measured transfer functions can subsequently be used as filters to simulate a virtual sound source over headphones HRTFs can be decomposed into two aspects e Binaural cues defined by differences between the left ear HRTF and the right ear HRTF These cues comprise 1 inter aural time differences ITDs resulting from the difference in path length from a source to both ears and 2 inter aural level differences ILDs resulting from the acoustical shadow effects of the head The binaural cues predominantly determine the perceived azimuth left right of a so
43. ime histogram of the input levels This may provide guidance for threshold adjustment The height of the curve represents how often a certain input level was observed in the last 30 seconds approximately 40 O TONEBOOSTERS 2010 2015 10 TB Sibalance De essers can be an evil necessity Vocal recordings may be too sibilant requiring de essing or excess sibilance removal but most de essers come with very clearly audible drawbacks as well After de essing vocals may sound muffled the s may sound more like an f or even worse the operation of a de esser manifests itself as a clearly audible time varying filter TB Sibalance provides very powerful tools to reduce excess sibilance in a minimally invasive way In contrast to conventional de essers TB Sibalance uses so called matched filter technology to only process those frequencies that are causing excess sibilance while leaving all other frequency components untouched The result of TB Sibalance will therefore sound cleaner and more transparent than that obtained with other de essers 10 2 1 De essing like a compressor You may see a very familiar input output curve in the screenshot that looks like a compressor In this case the input output curve does not relate to level but to excess sibilance Sibilance is a property of audio that is largely independent of level signals sound sibilant if there is a relatively large amount of signal energy present in the sibilant ra
44. ing and classical music The aim of these various scales is to control the amount of headroom for peaks in the audio content with respect to the RMS or loudness level The proper use of these metering systems is beyond the scope of this manual The reader is referred to other resources The peak hold values indicated by the meters can be reset by clicking on the respective indicator AES17 RMS 3 Mathematically a sinusoidal signal has a peak value that is 3 01 dB higher than its power RMS For output metering on the other hand it can be convenient to align peak and RMS values for sinusoidal signals If this behaviour is intended the RMS 3 control should be activated This setting will increase the RMS readout by 3 01 dB and is recommended when interpreting RMS values of the various K scales ISP The True peak ISP switch determines whether Inter Sample Peaks ISP will be taken into account in the limiter if set to on Digital to Analog D A converters often employ up sampling and interpolation of audio signals During this process new audio samples are inserted in between current audio samples These samples may extend the full digital scale even if the original samples are all within the full digital scale 22 TONEBOOSTERS 2010 2015 5 4 8 5 4 9 When the True peak ISP switch is on the limiter will protect against such potential clipping problems The use of True peak ISP is only necessary if used as limiter o
45. ion of the input sibilance e The handle in the graph sets the threshold sibilance at which the de esser becomes active e Use the mouse wheel to set the ratio a higher ratio will result in a larger reduction of sibilance e The blue fills in the input output function graph will indicate the currently detected sibilance level Sets the sibilance level at which the de esser should become active in reducing sibilance Sets the ratio of the sibilance input output function A ratio of two indicates that a sibilance level that has a value of X above the threshold will be reduced to X 2 above the threshold In other words a larger ratio will result in a stronger reduction of sibilance Sets the size in dB of the transition around the threshold Increasing the knee value will create a softer knee around the threshold value Sets the maximum reduction of sibilance in dB The de esser will never apply a reduction larger than this value Sets the amount of dry wet mixing A value of 10096 indicates that only the wet signal processed signal is sent to the plugin s output Sets the analysis time for determining excess sibilance A larger value will cause the de esser to have a smoother but slower response Sets the time in milliseconds to react to increases in sibilance level Sets the time in milliseconds to react to decreases in sibilance level Shows the real time spectrum from which excess sibilance is detected in yellow
46. ircuit clip Some tape recorders cannot handle hot input signal levels accurately and tend to demonstrate analog clipping This clipping behavior is also carefully modeled for each device model A setting of 12h will result in the same analog clipping level threshold as the original device higher values will result in more clipping effect If no or very little effect of analog circuitry clipping is desired set this control to full left Pre emphasis and post de emphasis The timbre of the tape simulation can be modified by enabling a pre emphasis and a complimentary de emphasis The pre emphasis can increase decrease the level of the side signal relative to the mid signal Mid Side control or increase decrease the level of the high frequencies relative to the low frequencies Low High control This process is applied on the signals before tape simulation takes place The inverse process post de emphasis is automatically applied afterwards Increasing the input level of the high frequencies will result in more aggressive processing of these high frequencies and vice versa The same is true for the relative levels of mid and side TB ReelBus includes accurate simulation of tape hiss This can have consequences for bouncing tracks in some hosts Bouncing tracks in Apple Pro Logic The small amount of tape hiss generated by TB ReelBus can lead to very long or infinite bouncing behavior when the Include Audio Tail option is enabled the tail
47. is attenuated and all frequencies are treated equally e Algorithm 1 single band attenuation With this algorithm the frequencies within the sibilance range are attenuated by the same amount while frequencies outside the sibilance range are not attenuated e Algorithm 2 matched filter With this algorithm only specific frequencies within the sibilance range will be attenuated namely those that were responsible for the high sibilance level Usually this algorithm gives the most transparent results The difference in attenuation or negative gain for the single band and matched filter algorithms is shown below The single band algorithm attenuates the full sibilance range alike most conventional de essers The matched filter algorithm on the other hand applies a more surgical cut of frequencies that are most offensive in terms of sibilance level while leaving the remaining signals untouched Matched gain dB band Sibilance range Positive sibilance level Level dB 6000 8000 10000 Frequency Hz 0 4000 The algorithm selection control can be set to intermediate values as well For example a value of 1 5 will give A an attenuation behavior that is in between a single band and matched filter algorithm 10 6 2 Filter slope The steepness of the sibilance attenuation filter can be changed with the slope parameter e For broad band de essing algorithm 0 the slope parameter has no effect
48. is increased Analog high pass filter HPF of order x Higher orders will give a steeper cut off Resonance can be set with the quality Q factor Brick wall high pass filter HPF Removes everything below its frequency with a very steep slope The transition between pass and stop bands can be modified with the quality Q factor Analog band pass filter BPF with 6dB oct slopes Brickwall band pass filter BPF Will remove everything excerpt for a narrow frequency range around the frequency parameter Applies an overall gain independent of frequency When the compressor is activated a frequency curve will be shown that indicates the filter response of the compressor level detector side chain This allows you to precisely select which frequencies the compressor should respond to TONEBOOSTERS 2010 2015 31 7 4 3 Compressor editor 7 4 4 When one of the nodes in the frequency editor is selected a corresponding compressor editor will be activated This editor shows the compressor input output curve for the selected node in the frequency editor Similar to the spectrum editor dragging a rectangle with the left mouse button will zoom in a left mouse click anywhere in the editor but on a node will zoom out Nodes can be placed anywhere in the compressor editor The line between the nodes will indicate the compressor input output curve Some examples are given below Upward compression In this case compression i
49. it version of the associated plugin is stored 1 10 3 Can I use the same key for my Windows machine and my Mac Yes The registration key files are personal associated to you as an individual and hence you can use the same key file on multiple PCs even across platforms 1 10 4 My registration key file does not work in Windows XP Verify that the plugin and the registration key file are always in the same directory with the exact same name except for the extension which is key for the registration key file The access to the key file may be blocked for security reasons To unblock e Rightclick on the registration key file e Select properties e Check if an unblock button exists at the bottom of the properties sheet e Clickthe button to unblock access to the registration key file 1 10 5 I can t enter the registration key in the Windows registry There is no need to do so The registration key file must reside in the exact same directory as the plugin and there is no need to open the file include or import it in the Windows registry 1 10 6 My Mac cannot open the registration key file You do not have to open the file in fact you probably should not Just copy the file into the plugin folder and you should be all set 1 10 7 I cannot use your plugins with a VST to Audio Unit AU or RTAS wrapper We do not support the use of plugin wrappers Please note that 3 party wrappers may not be compatible with recent VST interface sp
50. ity in specific frequency ranges e Use the listen option to evaluate what signals are being removed or processed with the current settings 48 TONEBOOSTERS 2010 2015
51. k on the scale to reset the peak hold function Signal levels Input gain Gain applied to the input signal before limiting in dB Out ceiling Maximum output level of the limiter in dB 20 O TONEBOOSTERS 2010 2015 5 4 5 4 1 5 4 2 Limiter Attack Response time constant to loudness increases in seconds dynamics Release Response time constant to loudness decreases in seconds Lookahead Lookahead time of the limiter to respond to overs in seconds Stereo link Amount of linkage between the limiter operating on the left and right audio channels Higher stereo link levels will improve the stereo image at the potential expense of lower overall loudness Stereo link does not influence the waveform auto saturation operation Multiband Amount of multiband limiting Set to 0 to exclude multiband limiting Output Dithering Bit depth for final delivery output signals Set to off to exclude quantization resolution and dithering Noise shaping Amount of perceptual noise shaping applied to the quantization errors and dithering signals Higher values will result in lower quantization noise audibility Output level VU meters Peak with peak hold and RMS with peak hold display Click to reset peak hold values Meter type Select the meter scale peak K12 K14 or K20 Switches ISP Enable true peak ISP limiting for final delivery signals Monitor When enabled the limiter operation is applied to the input signal witho
52. lting signal will gradually not be classified as sibilant TONEBOOSTERS 2010 2015 43 10 4 1 Voiced and sibilance frequency ranges The goal of TB Sibalance is to reduce or remove excess sibilance or said differently sibilant sounds such as ess that are too loud are to be reduced in level It is important to realize that the phrase too loud or excess sibilance is defined within its context This context dependency is explained schematically below Let us start with showing a spectrum of an audio signal In the figure below you will see the power spectrum level of a sound as a function of frequency We can identify two frequency ranges that are not necessarily mutually exclusive they are allowed to overlap in frequency e Avoiced frequency range typically around 200 4000 Hz which is the frequency range in which voiced parts of speech such as a e i and alike are predominantly present and e Asibilance frequency range typically around 5000 11000 Hz which is the frequency range in which sibilant sounds such as s t and alike and excess sibilance often occurs Voiced range Sibilance range Level dB 0 4000 6000 8000 10000 Frequency Hz 10 4 2 Sibilance level Let us consider an example in which excess sibilance occurs The figure below one can clearly observe that the power spectrum level within the sibilance range is much higher than the average power spectrum level in
53. ludes the side chain equalizer curve Signals other than vocals may benefit from processing with TB Sibalance as well For example full mixes drum sounds or an instrument may exhibit some harshness caused by excess energy in the 4 8 kHz range Some aspects that might be good to remember when using TB Sibalance on signals other than vocals e As explained in the previous sections the sibilance level does not only depend on spectral power in the sibilance range but also on its context determined by the voiced level The frequency range for voiced level analysis can be modified For full mix processing for example it can be beneficial to extend the voiced range to a much wider set of frequencies for example 100 Hz 10 kHz In this way the full context is taken into account to determine sibilance context not just the 300 3000 Hz range e The tonal parameter may help in restricting the level of tonal components Setting it to higher values will cause an increasing sensitivity to harmonic signals such as instruments present in a mix On the other hand a low value is beneficial to process rhythm instruments while leaving harmonic instruments untouched e Mixing dry and wet by setting the dry wet parameter to a value less than 100 can help in creating a more neutral or transparent behavior e Additionally the range parameter may be of use to set a limit of how much attenuation is applied e Use the side chain equalizer to modify the sensitiv
54. n e Right click a node to de active the corresponding equalizer section Drop down Several noise shaping presets are provided via the drop down menu indicated in the menu upper left corner of the noise shaping editor Determine the desired bit depth For CDs the bit depth is 16 bits while for DVD audio bit depths of 24 bits are typically used The desired bit depth depends on the application at hand Insert TB Dither as the very last plugin in the processing chain Dithering and noise shaping must always be the very last step in the effects chain preferably even post master fader Dithering and noise shaping processes depend on the exact quantization levels that are used during the final export Any level adjustment filter or other effect being applied in between dithering and export will completely eliminate the positive effects of dithering and noise shaping This also implies that peak limiting must be applied prior to dithering and that any level normalization applied by the host must be disabled Choose the dithering and noise shaping settings TPDF and GPDF dithering For 24 bit audio the dynamic range provided by 24 bits is in many cases sufficient to simply use spectrally white noise for dithering without noise shaping That noise can be produced by the TPDF triangular probability distribution function noise or GPDF Gaussian probability distribution function dithering setting These two dither noise types are specifically crafte
55. n the compressor settings In many cases these automatic adjustments should reduce the need to use the manual make up level ALM has three levels e Off ALM is disabled e Green ALM is set to normal which aims at keeping the loudness constant when threshold ratio and dry wet controls are changed e Yellow ALM is set to boost which will usually give a boost in loudness Engages the upward compression mode TONEBOOSTERS 2010 2015 39 Display Mix Makeup M S Ch link Pan Compressor curve Display selector Histogram When this setting is enabled the quieter passages below the specified threshold will be boosted while leaving louder passages unchanged upward compression If disabled louder passages above the threshold will be reduced in level while leaving quieter passages unchanged downward compression The dry wet mix control allows New York style parallel compression inside the compressor itself A value of 75 indicates that the output consists of 75 of compressed signal and a remaining 25 of unmodified input signal Uniquely to TB compressors the effective input output curve is visualized accordingly Sets the makeup level of the compressor output Mid side mode TB BusCompressor can operate in left right or mid side mode In left right mode the left and right channels are compressed In mid side mode on the other hand the mid left right and side left right channels ar
56. nal Some FREE plugins also have an associated registration key file This registration key file is included in the evaluation download package and allows verification that the key registration system works on your computer Please do not delete these key files as it will downgrade these free plugins to demo evaluation versions Download the latest and greatest trial evaluation versions from the downloads page at www toneboosters com and overwrite your plugin files with the newer ones Do not delete or modify your registration key files The registration key files are the files that have the key extension Restart your host program and you are all set TONEBOOSTERS 2010 2015 7 1 10 1 Do I need a serial number or hardware key No Our trial evaluation plugins can be upgraded to fully functional versions with a separate registration key file The registration key file must be stored in the same folder directory as the corresponding VST plugin Please make sure to make a backup copy of this registration key file when you purchase it if the registration key file is lost or damaged the plugin will automatically downgrade to a trial evaluation version with parameter saving disabled 1 10 2 Do I need separate keys for Windows or OSX or 32 64 bit operating systems No you can use asingle registration key file on 32 and 64 bit versions of both Windows and OSX Justuse multiple copies of the same key file in all folders where a 32 or 64 b
57. ncies resulting in a perceptible change in overall timbre or spectral balance The cross feed spectrum compensation algorithm applies a correction filter to reduce this effect Jal nro Ci Vie Room Environment presets Nearfield Typical near field room setup with speakers at 0 75m from the listener and a T60 reverberation time of 0 3 seconds Midfield Similar as above but with a loudspeaker distance of 1 50m Farfield Similar as above but with a loudspeaker distance of 2 25m Even further Simulation of loudspeakers placed far away Small studio Typical simulation of a small relatively damped T60 0 4s studio room Large studio Typical simulation of a larger studio with a longer reverberation time T60 0 6s Very small studio Relatively dry studio with a low late reverb modal density Anechoic room Simulation of an environment without reflecting surfaces Untreated box Simulation of a almost square room with hard walls resulting in substantial standing waves and flutter echoes Echo box Simulation of a very large room and sound sources at a great distance with almost distinct echos TONEBOOSTERS 2010 2015 13 Very dry room Simulation of a room with only very subtle room acoustics and a short reverberation time T60 0 2s 3 5 2 Speaker presets UJ This menu provides a variety of loudspeaker models that can be selected The following table describes the characteristics of the various loudspeaker models Please note th
58. nctionality in which the most extreme value across time is indicated by a horizontal line with the peak value displayed numerically above this line e Click on the VU meter to reset the peak hold value if supported e Drag the VU meter scale to change its range only in a limited set of plugins Bypass function Most ToneBoosters plugins do not have a bypass switch since virtually all VST hosts have their own bypass functions it is assumed that bypass operation is managed by the host not by the plugin TONEBOOSTERS 2010 2015 9 2 1 5 Presets Most but not all ToneBoosters plugins come with a range of presets The plugins do not have any preset functionality themselves presets are managed by the host program Please consult your host program s manual on how to work with plugin presets 10 TONEBOOSTERS 2010 2015 TB Isone Binaural stereo loudspeaker setup and reproduction environment simulator for headphones Introduction With TB Isone a virtual stereo reproduction system and listening room can be experienced using high quality headphones Allowing for full control over loudspeaker cabinet type loudspeaker distance and room reverb the virtual listening room can be largely customized TB Isone can therefore be used to simulate a wide variety of loudspeakers and reproduction rooms during mixing mastering or to generate binaural recordings by post processing Features TB Isone is a plugin that allows real time
59. nge typically 4 11 kHz compared to the overall level Nevertheless TB Sibalance allows control of sibilance by means of a threshold a ratio a soft knee and a range parameter much like a compressor Of course a dry wet control is included as well 10 2 2 Algorithm fusion TB Sibalance has three algorithms 1 a broad band de esser 2 a band limited de esser and 3 a matched filter de esser The latter will create a filter dynamically that only reduces sibilant frequencies while leaving everything else untouched In contrast to many other de essers these algorithms can be fused on a continuous scale Do you want 60 of de essing using a band limited de esser and the remaining 40 using matched filter technology Just set the algorithm slider to 1 6 and it is all set Moving the algorithm slider from O to 1 toa value of two seamlessly fuses the broad band de esser the single band de esser and the matched filter de esser 10 2 3 Tonal component sensitivity In many practical situations vocal sibilance consists of noise like signals that one would like to suppress Tonal or voiced signals on the other hand are often better left untouched Conventional de essers cannot discriminate between noise like and harmonic signals they simply measure energies With TB Sibalance the relative contribution of tonal components to the measured sibilance can be adjusted so that the de esser works much more accurately 10 2 4 Mid side processing an
60. nsion dll In Windows this can be done by simply opening the zip archive double click Subsequently copy these dll files into the VST plugin folder of your host program Please consult your host s manual to determine where to store the plugins The following table might help to determine what folder s your host may be using Restart the host and make sure that the plugin list is being refreshed or that new plugins are being activated before they become visible Please consult the manual of the host program how to proceed Operating Host Plugin version Plugin folder s a host may be using system to download Windows 32 bit 32bit 32bit C Program Files Steinberg VSTPlugins Microsoft C Program Files Steinberg VST Windows x86 Gy WPmso giam Eules Vism Windows 64 bit 32bit 32bit C Program Files x86 NSteinbergNVSTPluginsN Microsoft C Program Files x86 Steinberg VST Windows x86 C Program Files x86 VST Windows 64 bit 64bit 64 bit C Program Files Steinberg VSTPlugins Microsoft C Program Files Steinberg VST Windows x64 CES Program Files VST To install the trial evaluation plugins go through the following steps Download the free trial evaluation zip archive from the ToneBoosters com downloads page http www toneboosters com download Unzip the archive double click to retrieve the plugins those will have the file name extension component Copy these component files into the audio unit
61. ose Rec level Adjust the simulated recording level The effect of this control is visualized through the VU meters A higher rec level will create higher internal input levels Device Device model Selects the reel to reel device model adjustment 24 TONEBOOSTERS 2010 2015 6 4 6 4 1 6 4 2 6 4 3 Pre post emphasis Noise adjustment Color adjustment Output gain W amp F Sets the amount of wow amp flutter Set to full left to disable wow and flutter simulation Overbias Increases the high frequency bias signal beyond its optimal operating point for the selected device model Circuit clip Increases the amount of electronic circuitry clipping Set to zero if no circuitry Enable emphasis Mid Side Low High Tape hiss Tape hiss 30 dB Asperity noise Asperity noise 30 dB Spectrum Saturation non linearities are desirable Enables disables pre and post emphasis This feature enables a pre emphasis applied to audio signals before recorded to tape and a complimentary inverse post de emphasis applied afterwards The pre emphasis can be configured in the spatial domain with the mid side control and the spectral domain low high Amount of pre emphasis in the spatial domain Negative values put more emphasis on the mid component positive values put more emphasis on the side component Amount of pre emphasis in the frequency domain Negative values put more emphasis on low frequencies p
62. ositive values put more emphasis on high frequencies Adjusts the amount of tape hiss relative to the tape hiss level of the selected device model Reduces the tape hiss by an additional 30 dB Adjusts the amount of asperity noise relative to the asperity noise level of the selected device model Reduces the asperity noise level by an additional 30 dB Adjusts the amount of spectral changes induced by the selected device model This can be compared to the EQ part of the device Set to 0 if no spectral changes are desired Adjusts the amount of tape saturation induced by the selected device model Set to O if no or very little saturation is desired Sets the output gain If the Auto switch is enabled the inverse of the Rec level control will be automatically included to compensate for level changes as a result of a non zero rec level setting Setting up and using TB ReelBus Signal level dependencies The input signal level can be adjusted with the large input gain control The operation of TB ReelBus is similar to real tape very much signal level and frequency dependent Higher signal levels will correspond to stronger tape saturation The amount of tape saturation can be controlled in two ways e Ahigher input level will result in stronger saturation and e The threshold at which saturation starts can be adjusted with the saturation control a higher value will result in stronger tape saturation ef
63. pectrum analyzer e Dragthe handles to change their frequency and gain values e Left click or right click the handles to activate de active an equalizer section e Use the mouse wheel to modify the Q factor bandwidth of the equalizer section If enabled the difference between original and processed signal will be produced at the output If no de essing takes place the output will therefore be silent Enables the HQ high quality mode This mode will run the algorithm at a higher sampling rate internally Sets the amount of de essing for mid and side e Avalue of 100 will apply de essing on mid only e Avalue of 0 will apply de essing in stereo e Avalue of 100 will apply de essing on side only Sets the contribution of tonal components in detecting excess sibilance e Avalue of 0 will set the de esser sensitivity to tonal signals to its minimum Excess sibilance will mainly be detected for noise like signals e A value of 100 will set the de esser sensitivity to tonal signals to its maximum Excess sibilance will be detected for both noise like as well as tonal harmonic signals Sets the start frequency for voiced signal detection The level within the voiced signal range determines if signals in the sibilant range are excess sibilance or not Sets the end frequency for voiced signal detection Sets the absolute threshold for excess sibilance If the spectrum analyzer indicates levels below this value the resu
64. perating on the master bus for generation of final output delivery signals The True peak ISP implementation of TB Barricade is compliant with ITU R BS 1770 DC reject filter To ensure a DC free output signals TB Barricade has a build in DC rejection filter with a fixed 3 dB cut off frequency of 1 Hz Output resolution If TB Barricade is used to deliver final delivery signals e g on a master bus with a limited bit depth for example 16 or 24 bits dithering and perceptual noise shaping module should be enabled Set the dithering resolution to the number of bits of the final output format 16 or 24 bits Quantization and dithering always results in the generation of quantization errors or quantization noise The audibility of this noise can be greatly reduced by the processes of perceptual noise shaping Noise shaping changes the spectrum of the quantization noise such that it becomes less audible The amount of perceptual noise shaping can be controlled with the noise shaping control A value of 0 indicates no noise shaping 10096 indicates maximum noise shaping Dithering and noise shaping should only be enabled if TB Barricade is the last processing plugin to render a final output signal TONEBOOSTERS 2010 2015 23 6 1 6 2 6 3 TB ReelBus Analog tape simulation plugin carefully modeled after legendary Japanese and Swiss reel to reel recorders Introduction TB ReelBus is an analog tape recording simulator that aim
65. rameter saving disabled This means that if in demo mode settings of the plugins will not be saved with your project Registration key files to upgrade to full versions are TONEBOOSTERS 2010 2015 available from www toneboosters com The full versions do not have a parameter saving limitation The status of the plugin is shown at the bottom right corner of the user interface e DEMO Plugin is running in trial evaluation mode with saving of parameters disabled e Registered Plugin is running in full registered mode with full functionality e FREE Plugin is supplied with a free registration key file Please make sure not to delete this key file because it will revert the fully functional plugin to trial evaluation mode with parameter saving disabled e NFR The plugin is registered as Not For Resale e EDU Plugin is running in educational institution mode with full functionality Existing installations of VST AU trial evaluation plugins can be upgraded to fully functional versions with a separate registration key file The registration key file can be acquired in the online shop at www toneboosters com Go through the following steps to acquire a registration key file e Visit the online shop to place an order for a registration key file for the plugin or bundle you are interested in The purchase process will be handled by Avangate an authorized ToneBoosters plugins reseller e After completion of the purchase process
66. rs e Analog resonating highpass and lowpass filters order between 1 and 16 e Analog bandpass filter e Spectral balance filter e and several more In the unlikely case that you want to create a filter shape that is different from any of the included ones TB FIX supports a so called auto node link mode In this mode the filter shapes will be automatically constructed such that their combined effect will give a smooth interpolated curve through all nodes that were configured as auto node link filter Each filter section has its own amount control to modify how much of that filter is actually applied to the audio signal Furthermore the filter can be applied in stereo left only right only mid only or side only channels Linear phase or minimum phase We have closely followed the sometimes intense debates on minimum phase and linear phase equalizers and the pros and cons of each of them We understand that you appreciate the surgical editing precision of linear phase and the snappy accurate time response without pre ringing of minimum phase Wouldn t it be great if these benefits could be combined in one algorithm without their cons TB FIX has the potential to solve the minimum vs linear phase debacle for once and for all by introducing auto phase The phase response of the equalizer is dynamically adjusted to give the best of both worlds continuously adapted to the input signal and the desired equalizer cu
67. rve Dynamics processing without limits Each filter section also has a dedicated dynamics processing compressor stage This means that a filter response can become more or less active depending on the input signal level The compressor input output curve is configured by dragging nodes in an input output graph This way the input output curve can be configured beyond what is possible with conventional compressors using threshold and ratio controls Expansion downward compression upward compression positive or negative ratios parallel compression New York style compression via the amount control are all easy to set up and can be configured for each filter section independently Of course you can set the attack and release times and an auto release AR function is supplied as well The level detector of each compressor section also has some innovative features Besides being able to respond to stereo level changes the level detector side chain can also operate on mid only side only left or right only and even on the side over mid ratio 28 TONEBOOSTERS 2010 2015 7 1 4 Putting it all together All user accessible and internal parameters are interpolated automatically with maximum precision using 4 times oversampling This gives ultra smooth zipper noise free behaviour for no compromise professional grade output quality And did we mention the integrated DC reject filter DCF e More than 100 parameters to shape the so
68. s applied to low input levels Low input levels are brought up in level while high input levels are not modified other than a static gain This type of compression is useful when soft parts of a signal need to be louder without modifying loud parts and transients Downward compression In this case signals with low levels are not modified while high input levels are decreased in level This method is especially useful to change the character of a signal for example to change the punchiness of a percussive sound Soft knee downward compression In this downward compression configuration the curve is smoothed instead of having a hard knee This mode of operation is often somewhat more transparent than hard knee compression Expansion In this configuration high input levels are further increased in level while low input levels are not modified This mode allows to increase the dynamic range of transients Negative ratio compression In this configuration a sound that decreases in input level will become louder at the output Compressor settings can be initialized efficiently by using one of the preset curves accessible via the small drop down menu in the upper left corner This menu also allows you to copy curves from one equalizer dynamics processor section to another Auto phase option The auto phase option activates a novel method to modify the phase response of an equalizer Most equalizers available today have a linear phase
69. s at accurate simulation of all properties related to tape including its frequency and level dependent saturation inter modulation effects bias dependencies tape hiss asperity noise wow and flutter and clipping of electronic circuitry It is especially suitable for bus processing including the master bus to subtly sweeten and enhance the sound TB ReelBus contains several tape recorder simulations device models which can be adjusted individually by offsetting their tape hiss asperity noise amount of spectrum and saturation processing and alike Features e Very low latency processing 4 samples compensated for by host as a result of analog design e Support of all sampling rates from 44 1 up to 192 kHz e Adjustable record level with auto level makeup option e Accurate simulation of existing reel to reel recorders with different tape speeds e Adjustable tape hiss and asperity noise levels e Adjustable tape spectrum and tape saturation e Adjustable wow and flutter strength e Option to amplify bias strength for overbiasing e Simulation of both tape saturation as well as analog circuitry clipping e Calibrated analog VU meters e Each and every processing element carefully modeled after analog circuits and filters e Based on the VST 2 4 specification to allow compatibility with virtually all host programs The user interface se DEVICE ADJUSTMENT PRE POST EMPHASIS US gt gt van tp SAP GUI section Control Purp
70. sary CSC Crosstalk Enables a filter to compensate for the low end bias of cross talk signals Spectrum Compensation 180 Inverts the phase of the output signals by 180 degrees Room Room designer Enables disables room acoustic simulation When disabled TB Isone will emulate designer on off switch an anechoic room Size Changes the simulated room size volume Early reflections Changes the early reflections level Diffusion Changes the amount of diffusion of sound reflected from walls SpkLev Changes the loudness of the speaker in the room and consequently the output signal level of the plugin T60 Changes the late reverb time of the room simulation Room presets Selects a preset environment room model menu HRTF HRTF strength Changes the strength effect size of the HRTF elevation cues designer Ear size Changes the HRTF ear size Head size Changes the HRTF head size Setting up and using TB Isone HRTF Adjustment 1 Recommended initial settings The calibration of the HRTFs to each user s ears can be a somewhat tedious process but fortunately is required only once if performed correctly Here are some recommended settings that will work in most cases e Relatively small loudspeaker distance about 2 meters e Room acoustic simulation disabled e CSC switch disabled e HRTF strength to relatively large values 90 or higher e Loudspeaker simulation disabled flat response A dedicated preset Calibrate me is incl
71. signer that allows you to set the most important properties of a virtual loudspeaker setup These include e The frequency response e The characteristic size of the smallest driver typically the tweeter e The loudspeaker setup angles e The speaker setup designer is shown below The two curves in the graph indicate the frequency response of the virtual loudspeaker both on axis yellow line and 45 degrees of axis grey line The curves represent the response expressed in dB as a function of frequency in Hz The response can be modified by dragging each of the 3 available nodes numbered 1 to 3 with the mouse The left mouse button will activate a certain node the right mouse button disables the node Nodes 1 and 2 determine the bandwidth of the loudspeaker the lowest and highest frequencies that are reproduced by the speaker Dragging these nodes above the 0 dB line will create a resonance at that particular frequency Node 3 can be used to generate an additional resonance or dip at any desired frequency The grey line 45 degrees off axis response will largely follow the on axis response The difference between these two responses is determined by the tweeter size A large tweeter will typically result in a more directive response at high frequencies and consequently the 45 degrees of axis response will decrease at high frequencies A more directive loudspeaker response will reduce the amount of wall reflections and reverberation at
72. sulting noise spectrum compared to applying no noise shaping at all In this example the quantization noise has been reduced by up to 20 dB for frequencies below approximately 10 kHz at the expense of an increase of noise in the frequency region humans are not so sensitive to Dithering and noise shaping should always be the last processing plugin to render a final output signal This means that TB Dither must be applied post fader or pre fader with a fader gain of exactly O dB Any level normalization process applied subsequent to dithering and noise shaping will eliminate the effect of dithering and noise shaping and should therefore always be disabled 34 TONEBOOSTERS 2010 2015 e Zero latency processing e Support of all sampling rates from 44 1 to 192 kHz e Supports industry standard RPDF and TPDF dithering noise e Unrivalled flexibility to shape dithering and quantization noise e Supports any bit depth between 8 and 24 bits e Dedicated modes to listen to quantization noise only e Advanced noise shaping overload protection algorithm GUI section Control General Resolution Quantizer mode Noise shaping Output mode Noise Section shaping editor Type Frequency Gain Purpose Sets the number of bits that audio is quantized to Can be any integer value between 8 and 24 bits Sets the type of quantization e Rounding no dithering no noise shaping e RPDF dither industry standard 1 LSB wide RPDF di
73. ter This filter is often used in perceptual models to mimic the behavior of our hearing system Now you can use its characteristic as an equalizer curve Gaussian filter shape Gaussian filters have the shortest possible group delay of all filter types Combination of 8 linearly spaced peaking filters The decay of the individual harmonics can be set with the Q factor control This filter can give some very artistically interesting effects especially if the frequency is changed over time Combination of 8 logarithmically spaced peaking filters Typical analog peaking filter Analog low pass filter LPF of order x Higher orders will give a steeper cut off Resonance can be set with the quality Q factor Brick wall low pass filter LPF Removes everything above its frequency with a very steep slope The transition between pass and stop bands can be modified with the quality Q factor Applies a 3dB oct filter above the frequency Useful to generate pink noise from white noise This filter is symmetric along its frequency when a certain attenuation is applied below the frequency the same gain is applied above its frequency Very useful to change the spectral balance Analog low shelving filter LSF This shelving filter will show the typical analog resonance when the quality factor Q is increased Analog high shelving filter HSF This shelving filter will show the typical analog resonance when the quality factor Q
74. the voiced range Schematically we could therefore consider the power in the sibilance range above the voiced level as excess sibilance as it clearly stands out in the context of the overall power spectrum and with respect to the voiced level Said differently the sibilance level is positive If on the other hand the power spectrum level in the sibilance range would be below the voiced level the sibilance level is negative Voiced range Sibilance range Positive sibilance level Level dB Voiced level 0 4000 6000 8000 10000 Frequency Hz 10 4 3 Absolute threshold As we have seen in the previous section a positive sibilance level manifests itself as power spectrum levels within the sibilance frequency range that stick out in their context In practice however this method of determining the sibilance level is not sufficient For example it can happen that very soft sounds such as background signals or whispered voices show a power spectrum in which the sibilance frequency region sticks 44 TONEBOOSTERS 2010 2015 out while one does not want to process it because the signals are very soft in level and are therefore not being perceived as having excess sibilance Such level dependencies can be accomplished by means of the absolute threshold parameter as shown below Voiced range Sibilance range Level dB Absolute threshold Negative sibilance level 0 4000 6000 8000 10000 Freq
75. ther no noise shaping e TPDF dither industry standard 2 LSB wide TPDF dither no noise shaping e GPDF dither Gaussian noise dither no noise shaping e Noise shaping dither user controlled noise shaping Sets the amount of noise shaping between 0 and 100 relative to the curve specified in the noise shaping editor Sets the output mode e Normal output is input with dither and noise shaping applied e Muted input output with silent input signal e Output input difference between input and output to audition the effect of quantization dithering and noise shaping Identify the noise shaping equalizer section 1 to 4 The noise shaping editor is only active if the quantizer mode is set to Noise shaping Set the noise shaping equalizer section type e LSF low shelving filter e PEQ parametric equalizer e HSF high shelving filter Sets the center frequency of the active noise shaping equalizer section Sets the gain of the active noise shaping equalizer section TONEBOOSTERS 2010 2015 35 8 6 1 8 6 2 8 6 3 Q factor Sets the Q factor or inverse of the bandwidth of the active noise shaping equalizer section Nodes Nodes can be dragged in the noise spectrum graph to modify the spectrum of the quantization noise The white line will always indicate the overall noise shaping curve subject to information theoretic limitations e Left click a node to active a noise shaping equalizer sectio
76. tomatically result in an increase for neighboring frequencies and vice versa Carefully crafted noise shaping can result in 16 bit audio that has the same noise audibility as 18 bit audio when using TPDF white noise and can give quite good quality for bit depths as low as 8 bits sample 36 TONEBOOSTERS 2010 2015 8 6 4 The most common technique is to increase the quantization noise level above 16 kHz to allow a lower level in the 4 8 kHz range However you are encouraged to experiment with different noise shaping characteristics Noise shaping is more effective at higher sampling rates 44 1 kHz is the minimum sampling rate for noise shaping to work properly but 48 kHz or 96 kHz will make the process much more effective There is no benefit from using multiple dithering noise shaping algorithms on the same audio signal in fact it is better to avoid this from happening If TB Dither is used make sure that all other processes do not apply dithering and or noise shaping either in a plugin such as a peak limiter or during export by the plugin host You can audition the effect of noise shaping very easily by temporarily making the following adjustments e Set the bit depth to a very low number such as 8 bits sample e Set the output mode to Output input so you can listen to the effect of quantization without the input audio Export If all noise shaping parameters are tuned correctly export the audio signal The export bit d
77. uded to support this process 2 Selection of suitable audio material The next step is to select suitable audio content to use during the adjustment process It is recommended to use material that e You are very familiar with e Has a broad frequency spectrum Suitable material comprises voice recordings snare drums etc Use mono content or stereo content with very limited stereo depth and little or no reverb Do not use band limited signals such as sinusoids or instruments covering a narrow frequency range and alike the human hearing system cannot localize such signals accurately The best situation for the HRTF calibration is when you sit in front of an actual loudspeaker setup with loudspeakers at the correct positions 30 and 30 degrees azimuth 0 degrees elevation 12 O TONEBOOSTERS 2010 2015 3 4 2 3 4 3 3 4 4 3 4 5 Ear size adjustment This is best performed by setting the channel mode to Dual Mono Listen closely to the test material Ask yourself the following questions e Where does the sound come from e Does it come from above or more from the front e Does it sound natural or do perceive unnatural timbres or frequency notches Rotate the ear size knob until the sound is perceived most natural and coming most likely from the front Wrong settings usually result in a sound perceived from above Some people report that the adjustment process works best with their eyes closed Head size
78. ueaaeceeeeeseeeeeeeeeesseeseess 12 3 4 1 HRTF ACjUStMe nt ccccecceeesseseesseeeaaaeaaecaeeeeeeeeeeceeeessesscesseeeeceeasaasaeaaecaeeeeeeess 12 3 4 2 Ear size adjustment eececceeseesseessenseaaecaecaeceeceeeeeeeecesseussesseeeeceessausaaaaesaeeeeeaees 13 3 4 3 Head size adjustment sisse nennen nennen nennen nini n nini s asas nass 13 3 4 4 HTRF strength adjustmrient iiiter aE ance e MERE Na Tea TER Ud 13 3 4 5 CSC Crosstalk Spectrum Compensation esses 13 3 5 Model presets uscite i anr rene tun pe a ye SERE R NE CEN e RES e PARE Ue 13 3 5 1 Room Environment presets cccsccccesssccesssceeccsceceessseceseseeeceaeecessaeeceeeeeeeecaaaees 13 3 5 2 Speaker presets riti E UNSERE INNER E e E AEEA 14 3 6 Starting from scratch Build your own speakers eeeeennn 14 3 7 Virtual listening set p ecd certet naaa tea ge ev e ERE E e eva EXE ATE ea E FOR EXE Qe 15 3 7 1 Sound localization CUBS sicui aeree Leak eacvasaastsstecedsaasiveoccsasstatitontessusaizinne 15 2 TONEBOOSTERS 2010 2015 3 7 2 3 7 3 3 7 4 4 1 4 2 4 3 4 4 4 5 4 6 5 1 5 2 5 3 5 4 5 4 1 5 4 2 5 4 3 5 4 4 5 4 5 5 4 6 5 4 7 5 4 8 5 4 9 6 1 6 2 6 3 6 4 6 4 1 6 4 2 6 4 3 6 4 4 6 4 5 6 4 6 6 4 7 6 4 8 6 4 9 6 5 6 5 1 7 7 1 7 1 1 7 1 2 7 1 3 7 1 4 7 2 7 3 7 4 Parametric HRTF technology ccccssssssssssccecceeceececeecesscesessseeeseeseeeasaeaaecseeee
79. uency Hz In this example the power spectrum level in the sibilance range sticks out with respect to the voiced level but is below the absolute threshold level set by the user As a result the sibilance level is negative In these exemplary figures the absolute threshold is visualized as a hard decision while in TB Sibalance the absolute threshold function results in a gradual decrease in detected excess sibilance when the power spectrum is near the absolute threshold parameter setting 10 4 4 Tonal and noise sensitivity Excess sibilance is often very noise like or said differently the power spectrum does usually not show strong harmonics Conventional de essers can unfortunately not discriminate between harmonic and noise like power within the sibilance frequency range and therefore have a tendency to respond to tonal signals as well e g by determining tonal signals as being excess sibilant Such dependency may not always be desirable and therefore TB Sibalance includes sophisticated algorithms to separate harmonic and noise like signals e Noise like power within the sibilance frequency range is always contributing to the estimation of the sibilance level e Harmonic or tonal signals within the sibilance frequency range have a user configurable contribution to the sibilance level between 0 and 100 If set to 096 harmonic or tonal signals are suppressed in the detection algorithm If set to 10096 on the other hand the sibilance l
80. und in a clean and simple interface e 6 filter sections with many controls to modify their behavior e More than 30 filter types including a unique auto node link filter type e 3 node dynamics processing editor for each filter section e Manual and auto release AR option e Integrated output spectrum analyzer with zoom functionality e Unique innovative auto phase filter mode for high resolution transient response e Based on the VST 2 4 specification to allow compatibility with virtually all host programs TB FIX v3 0 0 registered GUI section Control Purpose Spectrum editor Nodes sections Each node represents one equalizer dynamics processor section There are 6 of these sections available Each node can be placed anywhere in the grid The x coordinate determines the center frequency of the equalizer dynamics processing section the height determines the gain e Left click a node to activate a section and to activate its dynamics editor e Right click a node to de activate it e Click elsewhere in the editor to zoom out e Drag the mouse to zoom into an area for microscopic editing When a section is selected the filter response corresponding to that section is highlighted with a color depending on the section index The white line indicates the overall filter curve for all active filter sections simultaneously Sect Selects the equalizer dynamics section off or 1 to 6 TONEBOOSTERS 2010 2015 29
81. und source and hence result from acoustical cross talk between both ear signals e Monaural cues resulting from reflections in the pinnae shoulder and from the torso of the human body These reflections result in specific peaks and troughs in the signal spectrum that depend on the elevation of the sound source When room reflections are present the pair of transfer functions including the wall reflections for a certain sound source position to both ears is referred to as binaural room transfer function BRTF This is shown in the figure below HRTFs for the left loudspeaker are indicated by orange lines A single wall reflection is indicated by the line line The speaker angle is between the red lines Parametric HRTF technology Although the use of HRTFs has been shown to be very effective in numerous scientific publications it also has well documented shortcomings For example HRTFs vary from person to person as a result of differences in the head size ear size ear shape and so on Application of the wrong HRTFs results in significantly degraded sound source localization It is therefore very important to match HRTFs to each individual listener for a convincing and accurate effect Isone Pro the precursor of TB Isone was the first VST plugin ever that provides such pseudo personalized HRTFs Now TB Isone provides the means to adjust the HRTFs for each individual listener by compensating for differences in the anthropometric properti
82. ut incorporation of the input and output gains This allows to listen to the limiter operation without impacting loudness AES17 3dB When enabled the RMS readout is increased by 3 01 dB to align peak and RMS Meter reset levels of sinusoidal signals Reset all peak hold values of the GUI VU meters Setting up and using TB Barricade Input gain and output limit TB Barricade limits the maximum amplitude of its input signals The amount of limiting is determined by e the input signal level e the input gain control and e theoutput ceiling control Input signals are first attenuated or amplified by an amount determined by the input gain control Subsequently the maximum amplitude is limited to the value of the out ceiling control Values above the value of the output limit control are referred to as overs The amount of gain or attenuation applied by the limiter is indicated by the limiter gain meters for the left and right channels individually The effect of the limiter gain can be evaluated without actual incorporation of the input gain by activating the mon monitor switch Lookahead attack and release times Limiters need a certain amount of lookahead to allow for a smooth gain curve If an over is detected the limiter will already gently start attenuating the signal a few milliseconds in advance of the over This lookahead prevents distortion and intermodulation artefacts Depending on the audio content values betw
83. vel at which reduction of sibilance starts to take effect In other words the threshold value allows you to determine what sibilance level is excess sibilance above the threshold and what is not considered excess below the threshold e The currently detected sibilance level is shown by the filled polygon e A knee parameter determines the range of the transition from no sibilance reduction below the threshold to de essing above the threshold A larger knee value results in a softer knee around the threshold 30 Pi 7 co 7 d 4 Z G Reduction Pd z 0 a Current sibilance level 2 3 O Threshold 30 30 0 30 Input sibilance dB 10 5 2 Set a maximum reduction in sibilance The range parameter sets the maximum change reduction in sibilance that is allowed For example if the range parameter is set to 10 dB the maximum difference between input and output sibilance will be exactly 10 dB 30 Output sibilance dB 30 30 0 30 Input sibilance dB 46 TONEBOOSTERS 2010 2015 10 6 1 Broadband single band or matched filter Now that we have defined sibilance levels and have determined the amount of sibilance reduction we would like to apply through the input output curve we are ready to set the method or algorithm for applying this reduction TB Sibalance supports 3 algorithms which can be blended seamlessly e Algorithm 0 broad band attenuation With this algorithm the signal
84. w frequency content Turning this knob will boost or reduce the low frequencies in the side chain only adjusting the compressor sensitivity The pump control changes the behavior of the compressor With a setting of 0 the compressor will typically work in a very transparent manner with minimum amount of pumping For electronic music however pumping might be a desirable effect Increasing the value of this control will result in a stronger pumping behavior especially if the LF Gain is set to positive values Sets the amount of compression A ratio of 4 indicates that 4dB above threshold will be reduced to 1dB above threshold Sets the maximum gain attenuation that can be applied If the range is set to 20 dB for example the compressor gain or attenuation is limited to 20 to 20 dB Sets the soft knee for a smoother compression behavior near the threshold point A soft knee applies the ratio exponentially as the signal approaches the threshold point With the right setting it gives a more transparent sound For instance using a 6dB knee and a 12dB threshold subtle compression will begin at 18dB 6dB below the threshold and will gradually become stronger until the maximum compression is obtained at 6dB Sets the input level below or above which the compressor becomes active Assisted Level Makeup ALM provides support in levelling compressor output with input by adjusting the compressor input output curve depending o
85. want to compress harmonic instruments present in a mix more than the noisy snare drums The noise control of TB BusCompressor changes the amount of noisy components that the compressor is responding to A second application for this feature is the compression of vocals By changing the sensitivity to noisy components fricatives and sibilants will relatively be more compressed reducing the need for additional de essing Knee Transient Attack Hold Ratio Range Q Auto 91 HQ e Q Release GUI section Control Purpose Attack Attack time Sets the time to respond to increases in input level A short attack time will not let strong transients through a longer attack time will cause a slower decrease in gain when the input level increases Transient Sets the amount of additional compression applied to transients A higher value will result in stronger compression of transients Only transients with a level above the threshold will be affected 38 TONEBOOSTERS 2010 2015 Release Ratio Threshold Hold High Quality HQ Release time Adaptive release Hysteresis Auto Noise LF Gain Pump Ratio Range Knee Threshold ALM Upward Sets the hold time in cycles When set to 1 at most 1 cycle or less is used to hold the gain A typical setting of 2 cycles should sound great on many sources and prevents intermodulation distortion Enables the high qualit
86. with TB Dither there are a couple of things one should know about quantization and the resulting quantization noise Bit depth reduction will always introduce errors If appropriate dithering is applied these errors will manifest themselves as spectrally flat or white noise With additional noise shaping techniques we can change the spectrum of that quantization noise to make it less audible A very common approach is to reduce the noise level in the frequency region the human hearing system is most sensitive to 1 kHz to 10 kHz approximately Such noise shaping is however subject to information theoretic limitations the most important one being that the total amount of noise cannot be reduced only be increased In other words decreasing the noise level between 1 and 10 kHz will always result in an increase of noise below 1 kHz and or above 10 kHz Fortunately as end user of TB Dither you do not have to worry about information theoretic principles The way it works is as follows Use the four noise shaping sections to construct a desired noise spectrum in the same way you operate an equalizer TB Dither will in real time automatically find the best possible realization of that curve by shifting the curve up and down such that the overall quantization noise level is the lowest possible noise level for that specified curve An example is given in this screenshot The colored areas indicate the targeted spectrum while the white curve shows the re
87. y mode Engage the High Quality HQ mode to increase the oversampling factor of TB BusCompressor for sub sample accuracy Rest assured that even with the HQ mode disabled oversampling will still occur in TB BusCompressor but enabling the HQ mode will shift the oversampling parameters to the next gear for even more accurate timing Sets the minimum time to respond to decreases in input level Adaptive release increases the release time if the signal is not quickly dropping in level ensuring that the gain riding behavior of the compressor more closely matches the signal envelope Hysteresis makes the release time history dependent If signals in the past were of relatively low level and the compressor is merely reacting to a short transient its release will be short to quickly recover from the short transient If on the other hand the signal was consistently loud previously the compressor will react with a slower release Clicking the Auto button will engage the automatic content dependent release mode The release time hysteresis and adaptive release controls will become inactive if the auto release mode is enabled Sets the relative sensitivity to noisy signal components in the input as opposed to harmonic components A higher value will cause the compressor to react relatively stronger to sibilants percussion instruments noise like signals snare drums and alike Low Frequency gain sets the relative sensitivity to lo

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