English Manual Applies to System 6000 firmware version 6.5.0 TC Icon version 7.5.
About this manual 1 MD-3 107 Algorithms overview 2 MD-4 115 Introduction 5 MDX 5.1 123 REVERBS 7 EQS 131 TC reverb algorithms: an introduction 9 EQ-5.1 133 Core-2 11 Massenburg (MDW) EQ 137 DVR-2 15 FORMAT CONVERSION 141 NonLin2 19 DMix 143 Reverb-2 23 Unwrap HD 151 Reverb-3 27 LIMITING 157 Reverb 8 31 Brickwall 2 159 VSS™ 3 45 LOUDNESS CORRECTION 165 VSS™ 4 49 ALC 6 167 VSS™ 5.1 Source 53 LOUDNESS RADAR METERS 177 VSS™ 6.
About this manual About this manual This manual will help you learn understanding and operating your TC product. This manual is available in print and as a PDF download from the TC Electronic website. The most current version is always from the TC Electronic website. To get the most from this manual, please read it from start to finish, or you may miss important information. To download the most current version of this manual, visit www.tcelectronic.
Algorithms overview This is an overview of the algorithms in the various System 6000 MKII-based platforms and bundles. Algorithms overview Reverb 6000 • Mastering 6000 Mastering 6000 • Algorithm name Algorithm type De-Esser De-Essing • Delay 1 Delay • Delay 2 Delay • Reflector Delay • Reflector 6 Delay • MD 3 Dynamics processing • • MD 4 Dynamics processing • • MD5.1 Dynamics processing • MDX 5.1 Dynamics processing • EQ 5.
Algorithms overview VP-8 & Toolbox5.1 • BackDrop Toolbox 5.1 (Bundled) Reverb 8 AM6 • • Core 2 Reverb VSS M4 Reverb • VSS Surround Reverb • VSS3 Reverb Reverb • • VSS4 HD Reverb • • VSS 5.1 Source Reverb • VSS 6.1 Reverb Reverb • Engage Surround prod. Tools Toolbox 5.1 Surround prod.
Algorithms overview 4 System 6000 MKII Algorithms – Firmware version 6.5.
Introduction Introduction System 6000 contains a wealth of algorithms, and the list of algorithms will continuously be extended. Our main focus is to offer the best possible quality – both as stereo and multi-channel versions. Bank Bank Name F1: Reverb A – (Music St.) F2: Reverb B – (Music St.) F3: Reverb (Music Sur.) F4: Halls of Fame F5: Reverb A – (Film) F6: Reverb B – (Film) F7: Reverb C – (Film) F8: Reverb (Film Sur.
Introduction Channel distribution in surround algorithms To best comply with the channel allocation used by most digital AES format equipment, the Input/ Output channels on TC Electronic surround algorithms are allocated as follows: 1 2 3 4 5 6 Left Right Center LFE Left Surround Right Surround These channel allocations comply with the following standards: ►► ►► ►► ITU Recommendation ITU-R BR.
Introduction Reverbs English Manual – Updated 2018-02-28
Introduction 8 System 6000 MKII Algorithms – Firmware version 6.5.
TC reverb algorithms: an introduction TC reverb algorithms: an introduction The TC Reverb Palette Generic Reverb Until 15 to 20 years ago, digital reverb was mostly used as a generic effect applied to many sources of a mix. Nowadays, where more Aux send and returns are at disposal, new approaches have emerged. Elements of the mix are being treated individually, adding room character, flavor and depth in more creative and complex ways.
TC reverb algorithms: an introduction Source Reverb Sampling Reverb When elements of a mix are picked up individually, a chance exists to define exactly how each of them is to be heard. There is no reason to apply one generic reverb to several single sources (unless they are supposed to present an identical position to the final listener, or when you have run out of Aux sends).
Core-2 Main Core-2 The TC Electronic REV CORE algorithms are particularly good for small room simulations. Due to the high density structure the relatively short Reverb diffuse fields occurring in small rooms can be convincingly reproduced. Though the VSS™ algorithms are dedicated for Film and Post production the Rev Core algorithm, known from the TC M5000 are also highly usable for purposes as such. You will also find that it is a good choice on percussive material, as it is very smooth an non-coloring.
Core-2 In Level Range: Off to 0.0 dB Sets the level of the Input to the Reverb in 0.5 dB steps. generated totally independently. The feature is especially applicable for the film industry and post production suites. Reverb Out Level Range: Off to 0.0 dB Sets the Output level of the Reverb in 0.5 dB steps. Rev Core-2 algorithm – reverb page Lo Decay Range: 0.01 to 2.5 times Relative Decay time multiplier for low frequencies.
Core-2 Diffuse Range: -50 to 50 This parameter gives you more or less Diffusion than the algorithm designer intended for the given Decay time. For optimum performance the diffusion is automatically adjusted behind the scenes whenever you change Decay times. This parameter gives you the added control to vary the Diffusion around this automatic setting.
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DVR-2 DVR-2 Introduction DVR-2 offers Reverb and System 6000 users a pristine Generic Reverb with true vintage flavor. Generic Reverb is complementary to Source Reverb, and both types are at disposal in the 6000. You can read more about Generic Reverb elsewhere in this manual, but the term is used to describe a flattering sustain effect, which can be added to many sources of a mix. It produces little character but also does no harm, because the effect is blurred or washed out.
DVR-2 Reverb Decay Range: 0.2 to 4.5 s Adjusts the Master Decay time. xLo Range: 0.5 to 2.0 Decay multiplier for low frequencies. For a x1.0 setting, low frequency decay will equal the Decay setting. xHi Range: 0.5 to Max. Decay multiplier for hi frequencies. For a x1.0 setting, high frequency decay will equal the Decay setting. Pre Delay Range: 0, 20, 40, 60 ms Pre delay is the amount of time from an input is received until reverb starts building up at the output.
DVR-2 Input Trans Emulates the sound of Input transformers typically used in vintage Reverbs. Use it to create a warm vintage like sound. Especially good for short Decay times. Trim Lo Freq A subtle damping of selected frequencies in the lower end. Modulation Modulation Sets the Depth of the modulation. Normal set to 100 %. Increase if you like a more liquid, chorus like sound. Hi Cut Frequency This is DVR-2´s HiCut and can be used to limit or extend the overall frequency spectrum.
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NonLin2 NonLin2 algorithm – main page NonLin2 Introduction NonLin2 is an Effect Reverb with controllable Envelope, Attack, Hold and Release. It is capable of generating compact Vocal Ambience, dramatic eighties drum and percussion sounds, Reverse Reverb or completely new and twisted effects. NonLin2 also does classic Gated Reverb, but because it doesn’t need to be triggered, it can be used on all sorts of program material.
NonLin2 and unnatural effect is obtained at short Release settings. Max. range for this parameter depends on the Attack and Hold settings. Reverb Style Selects the basic Reverb Style subjected to the Envelope and Twist modifications. The Style parameter should be seen as an algorithm selection inside the algorithm. Width Range: 0 to 100 % Adjusts the Output Stereo Width. 0 % denotes mono, while 100 % is max width.
NonLin2 Wet Level Range: Off to 0 dB Adjusts the wet Output level. Lo Cut Range: 20 Hz to 20 kHz Lo Cut on the Reverb Input. Hi Cut Range: 20 Hz to 20 kHz Hi cut on the Reverb Input. Lo and Hi Cut can help keeping heavily Twisted processing better under control.
NonLin2 22 System 6000 MKII Algorithms – Firmware version 6.5.
Reverb-2 Reverb-2 The Reverb 2 algorithm initially created for the TC M5000 compliments most types of source material, however experience has shown that the Reverb 2 algorithm is especially good on percussive instruments, as it has a very well-defined precise buildup. The Reverb 2 algorithm is a Stereo In/Stereo Out Reverb.
Reverb-2 Out Level Range: -100 dB to 0 dB Sets the Output level of the. tifacts that would otherwise have dominated a room of this size. Reverb-2 algorithm – reverb page Size Range: 0.040 to 4.000 s Scales the dimensions of the simulated space depending on the SHAPE chosen. The specific room being simulated is scaled 1:1 at SIZE =1.00. This can then be scaled up or down.
Reverb-2 Hi Decay Range: 0.01 to 2.00 times Multiplier for the high frequencies. If Hi Decay e.g. is set to 0.5, the Hi Decay time is half that of the nominal Decay setting. This parameter responds according to the Hi xOver setting. Lo Xover Range: 20 Hz to flat Sets the crossover frequency for the Decay xLo time multiplier in 1/3-octave steps. Hi Xover Range: 20 Hz to flat Sets the crossover frequency for the Decay x Hi multiplier in 1/3-octave steps.
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Reverb-3 Main Reverb-3 This is a description of the parameters specific to the Reverb-3 algorithm. It is capable of making an exceptionally clear Reverb sound using a very dense and natural sounding Reverb Tail. Decay time can be controlled in four individually adjustable frequency bands. Using Diffuse and the Distance controls, sounds can be made in which practically no Early Reflections are heard.
Reverb-3 Hi Cut Freq Range: 500 Hz to flat Hi Cut filter, shelving type. Provides an overall Reverb high frequency roll-off (6 dB per octave) that is suitable for making the space sound warmer. Hi Cut Att Range: -40 to 0.0 dB The attenuation control sets the high frequency roll determined by Hi Cut Freq. Levels In Level Range: Off to 0.0 dB Sets the level of the Input to the Reverb in 0.5 dB steps. Out Level Range: Off to 0.0 dB Sets the Output level of the Reverb in 0.5 dB steps.
Reverb-3 Depth Range: 0 to 100 % Controls the amount of delay path modulation or “wander” in the Reverb. The control interacts with the MODRATE, so with either control set at a high setting you will start to hear pitch modulation. The amount of either parameter that you can add depends on the type of material to which you are adding Reverb. Percussive types of sounds can be much more modulated than for example violin or an opera vocal.
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Reverb 8 Reverb 8 Introduction Reverb 8 is a new reverb algorithm for System 6000 MKII. It has been developed for arbitrary channel counts from 8 and up. Reverb 8 Presets Once TC Icon software and the System 6000 MKII firmware have been updated, you will find ten Reverb 8 Presets in Engine Bank F8 / Dec. 9 – see Fig 1.
Reverb 8 ►► ►► ►► ►► ►► ►► ►► ►► 9: Front L, FL 10: Front R, FR 11: Front Center, FC 12: Back Center, BC 13: Back L, BL 14: Back R, BR 15: Side L, SiL 16: Side R, SiR Running more than one instance of Reverb 8 Each System 6000 MKII frame can run up to two instances of Reverb 8 – which also occupies all the 16 output channels it offers as seen in Fig 3. If more channels are required, you will need to use more System 6000 MKII units.
Reverb 8 Reverb 8 essentials Reverb 8 is supposed to cover an endless variety of formats and channel counts. This is why – unlike VSS algorithms – it does not output early reflection patterns. Accordingly, speaker placement is far less critical with Reverb 8 than with its VSS cousins. What Reverb 8 does have in common with VSS™ 4 HD and VSS™ 5.1, however, is uncolored, smooth reverb rendering without the need for modulation.
Reverb 8 Using Reverb 8 Using the Grid Load one of the Reverb 8 presets into an Engine and go to its Main page (see Fig 7). The Grid page of Reverb 8 provides a new approach to multichannel editing. The four parameters shown below the Grid can quickly be varied across the algorithm’s channels by offsetting a dot from its center position. ►► ►► ►► ►► ►► ►► ►► Fig 7. Main Page with typical fader assignments The most important pages for quick edits are Main and Grid.
Reverb 8 Grid setup – Res. parameters Grid setup – Focus parameters What happens when the dot in the Grid is moved off-center is defined on the Setup page of the Reverb 8 algorithm – see “Reverb 8 algorithm – Setup page” (page 39). The Focus parameters for the Decay, Lo Decay, Hi Decay and Hi Cut parameters on the Setup page determine how strongly neighboring channels are affected when the dot is moved toward one channel. With the Decay Res.
Reverb 8 Working with multiple Reverb 8 instances Reverb 8 algorithm – Main page With more than one Engine running Reverb 8, it may be useful to access more instances at a time. Besides from opening multiple Icons, remember there is the “E1-4” page – see Fig 10. Fig 11. Reverb 8 algorithm – Main page Name Use the Name field to edit the name of the currently selected Preset. Fig 10. Two instances of Reverb 8 shown on the E1-4 page.
Reverb 8 Mute Group 1 & Mute Group 2 button Use the Mute Group 1 & Mute Group 2 buttons to temporarily mute all Reverb Channels assigned to Groups 1 and 2. Reverb 8 algorithm – Grid page Pre Delay section Pre Delay Group 1 & Pre Delay Group 2 Range: 0 to 150 ms Use the Pre Delay Group 1 & Pre Delay Group 2 parameters to apply a pre-delay to the Reverb Channels assigned to Groups 1 and 2.
Reverb 8 Lo Decay Range: 0.5 to 2.0 Use the Lo Decay parameter to define the decay time for the lower frequencies of the reverb. This parameter acts as a multiplier to the Lo Mult. and Lo Mid Mult. parameters in the Reverb Color section on the Rev page. Hi Decay Range: 0.5 to 2.0 Use the Hi Decay parameter to define the decay time for the higher frequencies of the reverb. This parameter acts as a multiplier to the Hi Mid Mult. and Hi Mult. parameters in the Reverb Color section on the Rev page.
Reverb 8 Reverb 8 algorithm – Setup page Lo Decay Focus Settings: Narrow / Default / Wide / LR Only / FB Only Use the Lo Decay Focus parameter to specify to what degree moving the green dot in the Grid affects the Lo Decay times for neighboring channels. See “Grid setup – Focus parameters” (page 35). Fig 13. Reverb 8 algorithm – Setup page Hi Decay Res. Settings: -75 / -50 / -25 / 0 / 25 / 50 / 100 % Use the Hi Decay Res.
Reverb 8 Group 1 section Reverb 8 algorithm – Rev page Use the Group 1 section of the Setup page to specify the Reverb channels that should belong to Reverb Channel Group 1 by clicking the respective buttons. All Reverb channels that are not assigned to Reverb Channel Group 1 automatically become part of Reverb Channel Group 2. Use the two Reverb Channel Groups to control complex Reverbs more effectively. Fig 14. Reverb 8 algorithm – Rev page set to NHK 22.
Reverb 8 NHK channel assignments The NHK 22.2 settings have been tuned for optimized performance with these channel assignments: ►► ►► ►► ►► ►► ►► ►► ►► ►► Engine 1: Set to 22.2 Top Plane. This will drive all Top speakers except for TpC (“voice of God”): 1: TpFL, 2: TpFC, 3: TpFR 4: TpSiL, 5: TpSiR 6: TpBL, 7: TpBC, 8: TpBR Engine 2: Set to 22.2 Mid Plane.
Reverb 8 Rev page – Reverb Color section Lo Damp Range: -18 to 0 dB Reverb 8 is equipped with an adjustable low-cut filter that allows you to remove low frequencies from the Reverb. Use the Lo Damp parameter to set the amount of cut. Lo Freq Range: 20 to 200 Hz Use the Lo Freq parameter to set the filter frequency for Reverb 8’s low-cut filter . Hi Soften Range: -50 to 50 Hi Soften is a special filter used to “soften” the high frequencies of Reverb 8.
Reverb 8 Reverb 8 algorithm – Trim page Fig 15. Reverb 8 algorithm – Trim page Level trims and Delay offsets per output are available on the Trim page. All parameters are additive to Level and Pre Delay settings on the Main page. Delay Trim section Delay Ch. 1 / 2 / 3 / 4 / 5 / 6 / 7 / 8 Range: 0 to 120 ms Use the Delay Ch. 1 to Delay Ch. 8 parameters in the Delay Trim section of the Trim page to delay each Reverb Channel by up to 120 milliseconds.
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VSS™ 3 Rev Delay Range: 0 to 200 ms A delay to the diffuse field part of the Reverb. VSS™ 3 The VSS™3 Reverb algorithm incorporates dedicated Early Reflection types for motion picture use, e.g. Car, Bathroom and Conference Rooms. Adds additional time between the Early Reflections and the onset of the “diffuse field” of the Reverb.
VSS™ 3 Out Level Range: -100 dB to 0 dB The overall Output level of the Reverb. This is mostly used when the algorithms used in Serial with other Engines. Early Size Range/Type: Small, Medium or Large Changes the size of the Early Type parameter. Some of the Early Types are one size. VSS™ 3 algorithm – early page Early Lo Cut Range: Off to 400 Hz Sets the Lo Cut frequency for the Early Reflections. Early Balance Range: -100 dB R, Center, -100 dB L The left/right balance of the Early Reflections.
VSS™ 3 middle, Stereo is the normal stereo image width and Wide covers the outside of the stereo image. The RevTypes: Fast Wd and Alive Wd only have one width (extremely wide). Lo Cut Range: 20 Hz to 200 Hz Adjustable filter that allows you to remove low frequencies from the Reverb. The Lo Cut frequency can be perceived as a Threshold frequency. The cut will be performed according to the Lo Damp parameter. Lo Damp Range: -18 dB to 0 dB Sets the amount of cut in dBs. (See Lo Cut description above).
VSS™ 3 Hi Decay Range: 0.01 to 2.5 Multiplier for the frequencies above the Hi Xover frequency. Example: If the main Decay parameter is set to 2.0 sec and the Hi Decay parameter is set to 1.5, frequencies above the Hi-Xover will decay for 3.0 sec. Conversely if this parameter is set to 0.5 the Decay time above the Hi Xover point will be 1 sec. VSS™ 3 algorithm – modulation page SpcMod Rate Range: -100, default, +100 Allows you to offset the speed of the LFO from the factory default assigned to each type.
VSS™ 4 VSS™ 4 algorithm – main page VSS™ 4 Introduction VSS™ 4 is a True Stereo Reverb – or two Source Input to Stereo Output reverb – and radically departs from being an additive sustain diffuse field added to a two channel signal. Based on source related Reflections from multiple angles, the precision of VSS™ 4 is comparable to real-world mono or stereo sources positioned in an authentic or virtual space. We dare say: “Stereo room simulation finally has come off age.
VSS™ 4 Reverb Type Range: Normal or Colored When a Location Type is selected a Reverb type is automatically set. With the Reverb Type parameter it is possible to select between the most natural sounding Diffuse field (Normal) and a more colored “vintage” Diffuse field setting. Reverb Diffuse Range: -25 to 25 This parameter gives you more or less diffusion than the algorithm designer intended for the given Decay time.
VSS™ 4 VSS™ 4 algorithm – setup page VSS™ 4 algorithm – color page VSS™ 4 algorithm – setup page VSS™ 4 algorithm – color page Location Type Reverb Location Type Select between different Locations. Both the Early Reflection- and Diffuse Field-types are changed when switching Location to give the optimal starting point for further adjustments. Lo Cut Range: 20 Hz to 200 Hz Determines the Lo Cut frequency. Attenuation amount is controlled via the Lo Damp parameter.
VSS™ 4 Example: When a Decay multiplier is set to 1.25 and the Master Decay is 2.0 seconds the resulting decay time will be 2.5 seconds. VSS™ 4 algorithm – gloss page LoMid Decay Decay multiplier in relation to the Master Decay, for the frequencies above the Lo Xover and below the Mid Xover settings. Example: When a Decay multiplier is set to 1.25 and the Master Decay is 2.0 seconds the resulting decay time will be 2.5 seconds.
VSS™ 5.1 Source VSS™ 5.1 Source Choosing between VSS™ 5.1 Source and VSS™ 5.1 reverb ►► ►► ►► If you wish to add reverb to a complete 5.1 mix or 5.1 stem, use the “VSS™ 5.1 Reverb” algorithm. If you need free dynamic movement of sources utilizing external joysticks or SpacePan 5.1, use the “VSS™ 5.1 Reverb” algorithm. If you wish to position single or composite sources with max localization and sweet spot enhancement, use the “VSS™ 5.1 Source” algorithm.
VSS™ 5.1 Source Lo Color Range: ±50 Adjusts the low frequency spectrum for the complete Output. This can be used for altering the overall color of the Reverb. Hi Color Range: ±50 Adjusts the high frequency spectrum for the complete This parameter is only available when KillDry is set to Off. Out level Range: -100 to 0 dB Adjusts the total Output level for the (up to five) Output channels. VSS™ 5.1 Source algorithm – setup page Output. Relatively adjusts the four Hi Colors for the four sources.
VSS™ 5.1 Source Center Channel Range: Off/On/Phantom On This parameter enables on/off setting of the Center speaker in the two multi-channel setups. ►► The Phantom setting is a patent pending feature that makes the Center speaker integrate with the Left/Right front speakers more properly than a standard configuration. Location Type Range: Different Locations. Changes the Location-type, meaning both the Early Reflections and Reverb settings.
VSS™ 5.1 Source Source 1 to 4 Early Reflections Lo Color Range: ±50 Adjusts the low frequency spectrum for the selected Source. In Level Range: -100 to 0 dB Adjusts the overall Input level from the source. VSS™ 5.1 Source algorithm – reverb page Hi Color Range: ±50 Adjusts the high frequency spectrum for the selected Source. Early Start Range: 0 to 100 % Sets the starting point of the initial taps in the Early Reflection pattern (The shortest taps).
VSS™ 5.1 Source Will not be available when Output Mode is set to Stereo. Mid Color Range: ±50 Additional Color adjustment of the Surround channels. Adjusts the spectral balance in the mid-end frequencies of the Reverb. Will not be available when Output Mode is set to Stereo. Hi Color Range: ±50 Additional Color adjustment of the Surround channels. Adjusts the spectral balance in the hiend frequencies of the Reverb. Will not be available when Output Mode is set to Stereo. Center Decay Range: 0.10 to 2.
VSS™ 5.1 Source Reverb Modulation Modulation Type Select between different types of modulation on the five Reverb diffuse fields. Modulation Depth Range: 0 to 200 % Adjusts the Depth of the selected modulation. Modulation Rate Range: ±50 Adjusts the Rate of the selected modulation. 58 System 6000 MKII Algorithms – Firmware version 6.5.
VSS™ 6.1 Reverb VSS™ 6.1 Reverb Choose your reverb wisely! Make sure to read the Reverb Intro chapter of this manual to learn about the differences between Generic and Source Reverb types. VSS™ 6.1 is a Generic type. It takes in a complete multichannel mix or stem and adds complex Early Reflections as well as uncorrelated diffused response to the signal. VSS™ 6.1 is the first professional reverb to acknowledge the advantages of 6.
VSS™ 6.1 Reverb Surr. Hi Cut Range: 20 Hz to 20 kHz Hi Cut parameter for the Reverb Diffuse fields in the Surround channels. Note that this parameter can be linked to Front Hi Cut on the Decay page. Master Early Each Location Type (see Setup page) has a predefined colorization. The Hi and Lo Color alter the default color of the selected Location. Lo Color Range: -50 to 50 Master Lo Color adjustment of the Early Reflections. Hi Color Range: -50 to 50 Master Hi Color adjustment of the Early Reflections.
VSS™ 6.1 Reverb Location Location Type Select between different Locations. Both the Early Reflection and Diffuse Field characteristics are changed when switching Location to give the optimal starting point for further adjustments. ting for a send/return configuration, and is the default on Factory presets. VSS™ 6.1 Reverb algorithm – color page Variation For some of the Locations it is also possible to select between different variations of the same Location.
VSS™ 6.1 Reverb Hi Cut Range: 20 Hz to 20 kHz Determines the Hi Cut frequency. VSS™ 6.1 Reverb algorithm – decay page Hi Soften Range: -50 to +50 Hi Soften is a special filter used to soften the Reverb response. This is not a Hi Cut, but a complex and dynamic set of filters for a particular purpose. Hi Soften is automatically scaled with Hi Cut and Hi Decay. Decay Crossover Lo Decay Range: 0.01 to 2.5 Decay multiplier in relation to the Master Decay, for frequencies below the Lo Xover setting. VSS™ 6.
VSS™ 6.1 Reverb Channel Decay The following parameters are Decay multipliers for the diffuse responses. The multipliers relate to the Master Decay time. Example: If the Decay time is set at 2 seconds and a multiplier is set at 0.5, the actual Decay time of the selected channel is 1 second. Left Range: 0.1 to 2.0 Decay multiplier for the Left channel. Center Range: 0.1 to 2.0 Decay multiplier for the Center channel. Right Range: 0.1 to 2.0 Decay multiplier for the Right channel. Left Surround Range: 0.
VSS™ 6.1 Reverb Early Start Range: 0 to 100 % Adjusting the Start time is an efficient way of getting rid of the first reflections that normally color the source the most. By adjusting the Start time, the first reflections are discarded but the timing of the later reflections remain unchanged. Therefore this adjustment is typically more acoustically precise and useful than a normal Predelay control.
VSS™ M4 VSS™ M4 algorithm – main page VSS™ M4 Introduction Multiple mono reverbs may often be more useful than pre-configured structures for stereo, 5.1, 6.1 etc. The M4 algorithm offers 4 discrete mono reverbs with 4 in and 4 out in one Engine.
VSS™ M4 VSS™ M4 algorithm – levels page VSS™ M4 algorithm – reverb pages 1 to 4 VSS™ M4 algorithm – levels page VSS™ M4 algorithm – reverb pages 1 to 4 In Level (Reverb 1 to 4) Range: Off to 0 dB Individual input level of the four reverbs. If the input signal is close to full scale and/or long decay times are used, it may be necessary to attenuate the input to avoid overload. Reverb Level (Reverb 1 to 4) Range: Off to 0 dB Individual reverb (wet) level of the four reverbs.
VSS™ M4 Reverb Lo/HiCut Lo Cut Range: 20 Hz to 200 Hz Determines the Lo Cut frequency. Attenuation amount is controlled via the Lo Damp parameter. Lo Damp Range: 0 to -18 dB Attenuation amount of frequencies below the Lo Cut setting. Hi Cut Range: 20 Hz to 20 kHz Determines the Hi Cut frequency. Hi Soften Range: -50 to +50 Hi Soften is a complex filter used to shape the high frequency spectrum of the Reverb diffused field. Hi Soften is scaled with Hi Cut and Hi Decay.
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VSS™ SR VSS™ SR algorithm – main page VSS™ SR The VSS™ SR (Surround) algorithm is a unique room simulator with new facilities for 4:2:4 surround production. The diffused field of the simulation is turned into a Front/Rear composition with separate Decay, Level and Predelay parameters for front and rear. The composite Output of the simulator is compatible with mono, stereo and surround reproduction.
VSS™ SR Front Level Range: Off to 0 dB Changes the level of the Front/Center information in the signal. Early Type Range: Several types Pick the type that best compliments your material or best represents the effect you are going for. Rear Level Range: Off to 0 dB Changes the level of the Rear/Surround information in the signal. Early Size Range: Small, Medium, Large Changes the size of the Early Type parameter. Some of the Early Types are only one size.
VSS™ SR VSS™ SR algorithm – reverb page Hi Soften Range: -50 to +50 Hi Soften is a special filter used to “soften” the high frequencies of the Reverb. This is not a simple Hi Cut filter but a complex set of filters working together to remove those frequencies that make a Reverb sound “brittle” or harsh. Hi Soften is scaled/linked to the Hi Cut and Hi Decay parameters. Rear Level Range: -10 to 0 dB Changes the level of the Rear/Surround information in the signal.
VSS™ SR Mid Xover Range: 200 Hz to 2 kHz Sets the frequency at which the transition from the low-mid to the mid frequencies takes place. Hi Mid Decay Range: 0.01 to 2.5 The Ratio control multiplier for the Hi-mid frequencies. Hi Xover Range: 500 Hz to 20 kHz Sets the frequency at which the transition from the mid frequencies to the high frequencies takes place. Hi Decay Range: 0.01 to 2.5 Multiplier for the frequencies above the Hi Xover frequency. Example: If the main Decay parameter is set to 2.
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VSS™ SR 74 System 6000 MKII Algorithms – Firmware version 6.5.
Engage Engage The inputs and outputs of this algorithm are distributed as follows: Engage algorithm – introduction Input L R C The Engage algorithm deals with three-dimensional sound reproduction in headphones. Input 5.0 mixes purposed for cinema, DVD or audio only (e.g. DTS and DVD-audio), and achieve an engaging surround reproduction using ordinary headphones. The five Input channels are positioned as a “L, C, R, LS, RS” (5.
Engage Input Trim Left Range: Off to 0 dB Level control for the left channel. Center Range: Off to 0 dB Level control for the left channel. Right Range: Off to 0 dB Level control for the left channel. Limiter Brickwall type Limiter. LED on each Output meter indicates when limiter is active. Threshold Range: -12 to 0 dB Sets the Threshold point of the Limiter. Threshold is relative to 0 dBFS. Release Range: 10 to 1000 ms Release time for the Limiter.
Toolbox 5.1 Inputs Toolbox 5.1 Toolbox 5.1 is designed to support surround production. The algorithm offers different Level, Test and Bass management as well as Down Mix options. It is typically used in combination with the MD5.1 and VP-5.1 algorithms. Toolbox 5.1 is offered as an integrated part of the MD-5.1 and VP-8 algorithms License packages. Mute Range: Muted/Unmuted Sets the Mute-status on the Input of of each of the 6 channels. The default setting is Unmuted.
Toolbox 5.1 Note that the Bass management is placed before this format conversion in the signal chain. Use the distribute part of the Bass-Management to convert from 5.1 to 5.0 mix. Output format: 5.1 Thru The Limiter is inactive. Output Format Range: 5.1 (=Off or Thru), LCRS, Stereo or Mono Selects the Output format in which your five main channels Input material will be mixed down to. Output format: Stereo The Limiter operates as a Stereo Limiter on Left and Right front channels.
Toolbox 5.1 channels. The Bass Management is placed just before the Output Format conversion. Main Channels Lo Cut Range: 10 to 200 Hz Sets the frequency for the Lo Cut filter, on the five main Output channels (LFr, RFr, Cen, LSr, RSr) Order Range: Off, 2nd, 4th order Sets the slope of the Main channels Lo Cut filter. Main Channels To LFE – Extract mode In this mode the Level controls are used to extract signal from the Main Channels and feed them to the LFE channel. Use this mode when converting a 5.
Toolbox 5.1 Ref Level 1, 2 and 3 Range -100 to 0 dB For convenient switching between 3 different Reference Levels. To adjust level press the value field when Ref. Level is activated and use Fader 6. To select Reference Level 1 to 3 press the oval keys next to the value fields. Toolbox 5.1 algorithm – trim page Calibration page Test signal generator (Oscillator) Toolbox 5.1 integrates a comprehensive testsignal generator meant for aligning the monitor system.
Toolbox 5.
Toolbox 5.1 82 System 6000 MKII Algorithms – Firmware version 6.5.
De-Esser De-Esser algorithm – main page De-Esser The De-Ess algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R Thru Thru Thru Thru Thru Thru E1 E2 E3 E4 Output L R Thru Thru Thru Thru Thru Thru De-Esser algorithm – main page Threshold When the input level exceeds the Threshold, the De-Esser will be in operation.
De-Esser Side Chain Pressing Sidechain enables to monitor the part of the signal that is defined by the Frequency and Curve parameters. This is the part of the signal that the De-Esser will “compress” when in operation. Adjusting the Frequency and Curve parameters when monitoring the sidechain, makes it a lot easier to hear where the sibilant problems occur in the signal. De-Esser algorithm – setup page Link The Link parameter switches the De-Esser between Mono and Stereo operation.
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De-Esser 86 System 6000 MKII Algorithms – Firmware version 6.5.
Delay-1 Right Delay Range: 0 to 2500 ms Sets the Delay time for the right side. Delay-1 The Delay-1 algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R Output L R E1 E2 E3 E4 Delay-1 algorithm – introduction This algorithm deriving from the M5000 is a straight, 24 bit true stereo delay offering up to 2.5 seconds in each channel with basic feedback control.
Delay-1 88 System 6000 MKII Algorithms – Firmware version 6.5.
Delay-2 Delay Line 2 Delay-2 Delay 2 Range: 1 to 2500 ms Sets the Delay time for the right side. The Delay-2 algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R Output L R E1 E2 E3 E4 Delay-2 algorithm – introduction This M5000 algorithm is a 24 bit true stereo effect offering Delays of up to 2.
Delay-2 Delay-2 algorithm – FB Modulation page Feedback-1 Range: -100 to 100 % The percent of positive phase and negative phase feedback for channel 1 feedback path includes Lo Cut and Hi Cut filters. Feedback- 2 Range: -100 to 100 % The percent of positive phase and negative phase feedback for channel 2 feedback path includes Lo Cut and Hi Cut filters. XFB 1>2 Range: -100 to 100 % The percent of crossfeed from channel 1 Output to ch. 2 Input.
Reflector 6 & Reflector LCR pable of crossing the border between Ambience, Early Reflections, Reverb and Delay-effects.
Reflector 6 & Reflector LCR Delay Multiplier This parameter multiplies the value of the 24 individual reflections (“taps”). Output Format – Reflector 6 Select Output format between: ‘5.0’; ‘5.0 minus Center’; ‘6.0’; and ‘6.0 minus Center’. When set to ‘x 1.00’ this means: “what you see is what you get” when monitoring the delay settings. For ITU775 reproduction (home theatre) music and post production, normally the 5.0 with or without center is meant to be used.
Reflector 6 & Reflector LCR The panner response depends on selected output format. The graphics area above the buttons indicates the taps Delay, Level, Shade and Phase. Panposition and Feedback is not displayed. X-Position Adjusts the X-axis position for the dry-signal. There are six parameters per tap (reflection): Reflector 6 algorithm – dry page Delay Range: 0 to 9000 ms The screen shows the relative distance between reflections.
Reflector 6 & Reflector LCR Feedback Feedback parameter from 0 to 100 %. The faders for the selected tap are also active when displaying the hyper view page. Note 1: Reflector algorithms – Taps – hyper page When a * symbol is next to the parameter value, this indicates that the Feedback Master is different from 0, meaning the resulting Feedback value is different from the individual setting. Note 2: When ( ) symbols are around the parameter, this indicates that the parameter is out of range.
Reflector 6 & Reflector LCR Reflector algorithms – Taps – shade page Reflector algorithms – Taps – shade page Set the Shade response curve (equalization) on this page. Three bands (Lo-shelf, mid-parametric and Hi-shelf) are available to form the desired response curve for the 24 individual Shade parameters. Look at the Shade curve as a “max-range” for the color adjustment to the taps (reflections). When a Shade parameter is set to 100 %, the resulting EQ curve is the same as seen on the Shade EQ page.
Reflector 6 & Reflector LCR 96 System 6000 MKII Algorithms – Firmware version 6.5.
Reflector 6 & Reflector LCR Dynamics processing English Manual – Updated 2018-02-28 97
Reflector 6 & Reflector LCR 98 System 6000 MKII Algorithms – Firmware version 6.5.
MD 5.1 MD 5.1 The inputs and outputs of this algorithm are distributed as follows: MD 5.1 algorithm – introduction MD 5.1 is a multi-channel multi-band expander/ compressor algorithm, with limiters and extensive possibilities to assign channels to multiple Sidechains. Based on the MD-2 and Finalizer heritage, fourband dynamics are here available for 5.1 productions. With the MD 5.1 it is possible to integrate dynamics processing in a 5.
MD 5.1 Hi Xover Range: Off to 16 kHz Sets the Cross-over frequency between the Midand the Hi- Expander and Compressor bands for the five main channels (LFr, RFr, Cnt, LSr, RSr). Threshold setting at -4 dB, will cause the Compressor to start operating at -22 dBFS. MD 5.1 algorithm – side chain control page The two cross-over points are not allowed to cross each other. Therefore the parameter range can be less than going down to Off, if the Lo Xover parameter is set above the Off position.
MD 5.1 Setting a channel to unprocessed will preserve the processing delay through the algorithm, keeping the channel time-aligned to the other (processed) channels. The Sum settings will add the Input to the sidechain, whereas the Max settings only will contribute to the sidechain if the level exceeds the other Input channel levels. MD 5.
MD 5.1 Attack Range: 0.3 to 100 ms Sets the time it takes for the Expander to reach the attenuation specified by the Ratio parameter when the signal drops below the Threshold point. MD 5.1 algorithm – expander – All LMH page Meter Zoom Press Zoom to decrease meter range and have a more accurate metering. Bypass Exp. Press to bypass the Expander section of the MD 5.1 algorithm. MD 5.1 algorithm – expander – All LFE page MD 5.
MD 5.1 Example: If the Reference Level is set to -18 dBFS, a Threshold setting of -4 dB, will cause the compressor to start operating at -22 dBFS. Pressing any parameter will assign this to Fader 6. All – parameters Gain Range: Off, -18 dB to 12 dB in 0.5 dB steps. Adjusts the gain after the Compressor. These parameters are equivalent to the “All” – Threshold, Range, Ratio, Attack and Release parameters.
MD 5.1 Bypass Limiter Press to Bypass the Limiter section of the 5.1 algorithm. MD 5.1 algorithm – soft clip page Threshold Range: -12 dB to Off -6 to 0 dB in 0.1 dB increments -12 to -6 in 0.5 dB increments Brickwall limiter for the five multiband channels. Threshold is always relative to 0 dBFS. LED on each Output meter indicates when Limiter is active. Release Range: 0.01 to 1.00 seconds Release time for the Limiter. Ceiling Range: -0.
MD 5.1 Release Range: 0.01 to 1.00 seconds Release time for the Limiter. Ceiling Range: 0 to -0.10 dB in 0.01 dB steps. Fine-tuning parameter setting the Ceiling for the Limiter. The Ceiling parameter prevents the Output signal from exceeding the adjusted Limiter Threshold. It can be used to “hide” overloads to downstream equipment, but it does not remove the distortion associated with an over. MD 5.
MD 5.1 106 System 6000 MKII Algorithms – Firmware version 6.5.
MD-3 MD-3 High resolution, multiband dynamics processing has been a trademark of TC Electronic for more than a decade. Splitting audio into frequency bands before expanding or compressing, tremendously helps fight breathing and spectral intermodulation artifacts. Our algorithms are designed with the headroom to do so without introducing split- and recombination filter anomalies, and can be used for level optimization, mix assistance or discreet spectral balancing as required.
MD-3 Please note that when coming from e.g. Dual Mono mode EQ settings from the Left channel will be copied to the right channel. Dual Mono In Dual Mono mode the Expander/Compressor section of the algorithm uses separate SideChains for the Left and Right Channel and they can be operated individually. MS Linked In the MS Linked mode the MS Encoder is activated and the Expander/Compressor section of the algorithm uses one common SideChain for both Left and Right channels. EQ is unlinked.
MD-3 Output Mode Range: Left/Right, MS Left/Right: Straight Output without MS Decode. MS: MS decode before the limiter section. DC block Range: Off, On The DC block is a Lo Cut filter used to remove potential DC noise at 2 Hz. Expander/Compressor Setup Lo Xover Range: Off, 25, 32, 40, 50, 63, 80 Hz … 16 kHz Sets the Cross-over frequency between the Lo and Mid Expander/Compressor bands for the two channels. Nominal Delay Range: 0 to 15 ms 0 to 2 ms in 0.1 ms steps 2 to 15 ms in 0.
MD-3 Basic operation ►► ►► ►► Notch Filter – Narrow Type Press keys Lo, Mid1, Mid2 and Hi to activate/ deactivate the EQ bands. Select Freq, Gain, Type or Lo/Hi to access all four parameters on individual bands. Press Bypass EQ to bypass all four bands. Type Selector Press Type and use faders 1 to 4 to select filter types. For Lo and Hi filters select between filter types: Parametric, Notch, Shelve and Cut. ►► For Mid 1 and Mid 2 filters select between filter types: Parametric and Notch.
MD-3 Range – Hi band: 20 Hz to 40 kHz MD-3 algorithm – normalizer page Gain Press Gain and use Faders 1 to 4 to adjust gain for each of the four EQ bands.
MD-3 Pressing Threshold, Range, Ratio, Attack and Release keys will immediately assign Lo, Mid, Hi and Master values for these parameters to Faders 1 to 4. Press the L/R key to see all parameters at the same time. On the L/R page any parameter can be assigned to Fader 6 by pressing the individual parameter values. can be assigned to Fader 6 simply by pressing the parameter value. MD-3 algorithm – compressor page Threshold parameters Range: -50 dB to 0 dB (in 0.
MD-3 Example: With a Ratio setting of 2:1 the compressor will reduce every 2 dB above the Threshold point to only 1 dB. Attack Range: 0.3 to 100 ms The Attack time is the time the Compressor uses to reach the gain reduction specified by the Ratio parameter. Example: If the Input signal increases by 4 dB above the set Threshold with a Ratio set to 2:1 and the Attack time set to 20 ms, the Compressor will use 20 ms to reach a Gain reduction of 2 dB. Release Range: 20 ms to 7 s The fallback time.
MD-3 Output Fader Output Fader Range: Off to 0 dB Off to -40 dB: in 3 dB steps, -40 to 0 dB in 0.5 dB steps Output fader for both Outputs. Balance Range: -6 dBL to -6 dBR Changes the Output balance between the Left and Right channel. COMPARE button Press to activate compare function. Compare Level Range: -20 to 0 dB Due to the difference in level, “in-circuit” and “out-of-circuit” comparisons are often difficult to make using the BYPASS key. Use the Compare Level parameter to compensate. Bypass Lim.
MD-4 MD-4 algorithm – main page MD-4 MD-4 algorithm – introduction MD-4 is a multiband processor featuring newly developed DXP processing that lifts up low level detail rather than squashing the peaks. DXP processing is ideal for classical music, acoustic music, voice, film to broadcast transfers – and other situations where low level subtleties tend to get lost.
MD-4 convenient place to adjust more or less dynamics processing and Loudness. Trim offers the same “Drive” approach to level and Threshold used in classic analog Limiters. Unlike analog designs, MD4’s 48 bit internal resolution prevents resolution and signal/noise ratio from being sacrificed, because the processing dynamic range is larger than even the highest resolution Input would be.
MD-4 When mastering with DXP mode, start with a Reference Level around -8 dBFS. This will ensure that loud parts are not subjected to excessive gain and therefore limiting. When all other parameters have been adjusted to complement the material, try altering the Reference Level up and down a couple of dB to set the optimum average/peak ratio of that particular piece of material. 5 band Mode This is a powerful control that changes between three different ways of using MD4.
MD-4 (wide=S) such as ambience, or to treat fully correlated components (center=M) in a certain way. Preset examples can be found in Engine Factory Bank 10, decade 5. DXP processing doesn’t add anything to the signal that isn’t there already. It merely magnifies the details or spectral components which previously may have been masked. Like Normal and Parallel mode, resolution of DXP mode is 48 bit fixed point to maintain a processing margin over the source material.
MD-4 When set to 0.0 ms, the delay through MD4 caused by additional look ahead structures and up-sampling, amounts to 38 samples at 44.1 kHz (0.86 ms), 40 samples at complex mix. Because equalization is performed at 48 bit resolution, MD4 has the headroom to boost even full scale signals, regardless if processing is done in the L/R or M/S domain. 48 kHz (0.83 ms), 93 samples at 88.2 kHz (1.05 ms) and 98 samples at 96 kHz (1.02 ms).
MD-4 MD-4 algorithm – 5 band page will be best suited in combination with a Look Ahead Delay of at least 1 ms. Steer parameter (DXP mode) DXP processing enables boost of only low level material without affecting material that is already loud enough. This gives less audibility and less transient distortion. The boost is applied to levels below the Reference level, and reaches its max at Threshold. The more Steer, the more audio is steered towards Ref Level.
MD-4 Threshold L – Threshold R Range: -12 to 0 dBFS Output Limiter Thresholds. Consider linking L and R when used on stereo signals. Soft Clip threshold is relative to this setting. Profile Various profiles can be selected to best suit your material to process: Voice, Loud, Universal, Soft and Dynamic.
MD-4 122 System 6000 MKII Algorithms – Firmware version 6.5.
MDX 5.1 MDX 5.1 MDX 5.1 algorithm – introduction MDX 5.1 is a high resolution dynamics processor for multichannel signals. It presents a sophisticated new angle to dynamic range control, and is an alternative to traditional compression and limiting techniques. MDX 5.1 is capable of bringing up low level detail, rather than boosting everything, and then having to limit the transients afterwards, see Fig 1.
MDX 5.1 for instance, different settings for the Center or Surround channels, where speech intelligibility or low level ambience tend to get lost. Like when a feature film is re-purposed for broadcast or DVD under domestic listening conditions. Note, that the lower the DXP Threshold, or the higher a Steer setting, the more low level boost is applied. The low level boost can be different in different channels, and even in different frequency bands. If it is required to process more audio channels than 5.
MDX 5.1 ference between the two, for instance 15 dB or more. In this example, low level boost and spectral shaping is added to the Center channel. If the delivery resolution is low, the difference should also start smaller. For heavily data reduced multichannel broadcast using Dolby AC3, best results are typically obtained with a 6 to 10 dB difference between Ref Level and the Limit Threshold.
MDX 5.1 Input L R C LFE SL SR Xt E1 E2 E3 E4 Output L R C LFE SL SR MDX 5.1 algorithm – main page -4 dB, will cause the Compressor to start operating at -22 dBFS. DXP Defeat Level Range: Off to -3 dB MDX 5.1 may remove low level gain below the threshold set with this parameter to avoid having irrelevant sources (e.g. background noise) become audible. Low level gain is not revoked if the DXP Defeat Level parameter is set to Off.
MDX 5.1 can be less than 16 kHz if the Hi Xover parameter is set below 16 kHz. The illustration above reflects the processing parameter set to MDX 5.1 in Normal mode. Hi Xover Range: Off to 16 kHz Sets the Cross-over frequency between the Midand the Hi- Expander and Compressor bands for the five main channels (LFr, RFr, Cnt, LSr, RSr). Basic operation MDX 5.1 algorithm – link control page At the Control page it is possible to decide which Sidechains should control which channels.
MDX 5.1 Normal Range: Off / On When this parameter is set to “On” the Input channels selected to be controlled by the respective sidechain will also input to the sidechain. servation and resolution, and do not compromise audio in any way. MDX 5.1 algorithm – DXP all page SC1 Add1, SC 2 Add2, SC 3 Add3 Range: Off, LFr Max, RFr Max, Cnt Max, LSr Max, RSr Max, Extern Max, LFr Sum, RFr Sum, Cnt Sum, LSr Sum, RSr Sum, Extern Sum.
MDX 5.1 Soft Clip Full Range SoftClip Range: 6 dB to +3 dB to Off Softclipper Threshold setting after the Compressor for the five multiband channels. Threshold is always relative to 0 dBFS (Not the Reference Level). Ceiling Range: -0.10 dB to 0 dB Fine-tuning parameter setting the Ceiling for the Limiter. MDX 5.1 algorithm – LFE limit page LFE SoftClip Range: 6 dB to +3 dB to Off Softclipper Threshold setting for the LFE channel only. MDX 5.1 algorithm – main limit page MDX 5.
MDX 5.1 MDX 5.1 algorithm – output page MDX 5.1 algorithm – output page Trim Levels Output trims Range: 0 dB to -12 dB in 0.1 dB steps Level trim of the Output channels. Only the fader is placed after these trims. These parameters can be used to trim the levels of the monitoring system, but please note that it also affects the recorded material. Mute Allows muting of each Output-channel. Output Fader Range: Off to 0 dB Off to -40 dB: in 3 dB steps -40 to 0 dB: in 0.
MDX 5.
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EQ-5.1 ►► EQ-5.1 The EQ 5.1 algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R C LFE SL SR E1 E2 E3 E4 Output L R C LFE SL SR LFE channel Linked mode: Press Link Main Channels to link the 5 main channels. A “pop-up” display will ask whether you wish to copy settings for the Center channel to the Front and Surround channels.
EQ-5.1 EQ-5.1 algorithm – front / center / surround / LFE pages Shelving Filter Notch Filter – Narrow Type EQ-5.1 algorithm – front page Press Front/Center/Surr. or LFE (side fane) to access parameters for each of the channel groups. Basic operation ►► ►► ►► Press keys Lo, Mid1, Mid2 and Hi to activate/ deactivate the EQ bands. Select Freq, Gain, Type or Lo/Hi to access all four parameters on individual bands. Press Bypass EQ to bypass all four bands.
EQ-5.1 Cut Filter – Butterworth type Range for the Shelve filter: ►► Lo BW: 3 to 12 dB/oct ►► Hi BW: 3 to 12 dB/oct Range for the Cut filter: Lo BW: Bessel or Butterworth ►► Hi BW: Bessel or Butterworth ►► Bandwidth/Q – Key-Values: BW Q ►► 0.5 2.87 ►► 0.7 2.04 ►► 1.0 1.41 ►► Freq Press Freq and use Faders 1 to 4 to adjust frequencies for each of the four bands.
EQ-5.1 136 System 6000 MKII Algorithms – Firmware version 6.5.
Massenburg (MDW) EQ Massenburg (MDW) EQ advantages. All of them have processing capabilities worthy of handling any critical signal path.
Massenburg (MDW) EQ been established. Now a Relative link can be enabled, preserving the adjustment off-sets between channels, but not having to adjust both of them permanently. In the 6 channel version, up to three different Absolute links may be set. Before link channels can be changed, the Link Enable key has to be active. When a link is established, the settings of the lowest channel number in the group will be copied to the rest.
Massenburg (MDW) EQ 6-ch. version In the 6-ch. version a Channel select parameter is placed below the Output meters. This parameter is always present. Depending on assigned channels in the three Link groups (made on the Main page), the Channel select parameter indicates one or more of the six channel numbers when changing the parameter. E.g. if all six channels are assigned to Link group A, this parameter will not have any further steps to select between.
Massenburg (MDW) EQ 6 channel version The Gain parameters are located at the Level page. When Linking channels in groups, the Gain settings are not included. Parameter Link three groups, it can be operated as an individual channel. Copying of channel settings is performed instantly when linking – therefore we recommend that careful consideration is done when changing Link groups. We also recommend deactivating Enable Linking when not in use.
Massenburg (MDW) EQ Format conversion English Manual – Updated 2018-02-28 1
Massenburg (MDW) EQ 142 System 6000 MKII Algorithms – Firmware version 6.5.
DMix DMix DMix algorithm – main page DMix: Optimum Mobile Platform Delivery In just one engine, DMix can downmix, loudness process and true-peak limit any mono, stereo or 5.1 source. Input formats are dealt with automatically without the need for metadata, downmix takes place at overload-proof 48 bit resolution, loudness processing complies with ATSC or EBU standards, and transparent transcoding keeps the output perfectly conditioned for mobile TV, iPod or IPTV.
DMix true-peak limiting performed at 48 bit, fixed point precision. This enables the downmix gain to be set freely without worrying about overload or the loss of resolution. For extra emphasis on the Center channel, the gain may be run all the way up to 0.0 dB still without any risk of internal or output overload. Surround Gain Range: Off, -12.0 to 0.0 dB Downmix gain for the Surround inputs relative to L and R front. Default Surround gain would be between -3.0 and -6.
DMix the Loudness Control cannot attenuate the signal at all. Max Gain Range: 0 to +20 dB This is the maximum gain the Loudness Control is allowed to perform. If set to 0.0 dB, the Loudness Control cannot add gain to the signal at all. Freeze Level Range: -10 dB to -40 dB Sets the minimum level required before the Loudness Control will start adding more gain. It would typically be set to avoid boosting signals considered noise.
DMix the Output level is close to the Target Level, gain changes are relatively slow. Parametric Filter – Broad type The Average Rate offsets all time constants to be faster or slower. Values below 1 dB/Sec produces a gain change gating effect when the Output level is already in the target zone, while values above 4 dB/Sec will add density to sound. Slow Window Range: 0 to 20 dB The slow window is the area around the set Target Level. Within the slow window, the Loudness is only gently controlled.
DMix Cut Filter – Bessel type Type Press and use Faders 1 to 3 to set BW value for each of the 4 EQ bands. Range for the Notch filter: ►► ►► ►► Lo BW: 0.02 to 1 oct Mid BW: 0.02 to 1 oct Hi BW: 0.02 to 1 oct Range for the Parametric filter: ►► ►► ►► Lo BW: 0.1 to 4 oct Mid BW: 0.1 to 4 oct Hi BW: 0.
DMix Range: Xover ►► Xover ►► Xover ►► Xover ►► 1: Off to 1,6 kHz 2: Off to 4 kHz 3: 100 Hz to Of, 4: 250 Hz to Off Defeat Thresh Range: -3 to -30 dB This is a unique control which holds the gain from the multiband compressor below a certain threshold. No matter the spectral shaping applied from multiband system, below the Defeat Threshold, the frequency response is flat and gain is unity. Defeat Threshold is relative to Compressor Threshold, which is relative to Reference Level.
DMix levels, e.g. when preparing film or concerts for domestic or noisy environment listening. Try setting the Steer and/or Threshold parameters differently in the bands to hear the effect. High Steer values add more detail gain than low values, but remember that Threshold has to be negative to add detail gain at all. DXP Threshold relates to the Reference Level set on the Main page. To disable DXP detail gain at very low levels, use the Defeat Threshold and Defeat Ratio controls.
DMix bution, and not to process more than necessary for a certain broadcast platform. Don’t take pride in being the loudest station, but in being the best sounding and most consistent one. For broadcast stations early in the process of converting production to loudness based criteria, a relatively high Loudness adjustment Ratio may be initially needed, for instance 1:2, in order to avoid too much loudness fluctuation during transmission.
Unwrap HD Unwrap HD The UnWrap HD algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 2/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R E1 E2 E3 E4 Output L R C LFE SL SR Unwrap HD algorithm – introduction UnWrap HD: Professional Stereo to 5.1 Upconversion In a perfect world, all audio recorded over the last fifty years would be available on multi-track tape formulated to last for centuries.
Unwrap HD This way you can check all speakers one by one with pink noise first, and also collapse the 5.1 signal to stereo or mono to make sure the result is still hearable. Try loading some of the UnWrap HD presets. You can A/B the process by pressing Bypass on the UnWrap HD Engine, or collapse the signal to stereo again by selecting Stereo format on the Toolbox Engine, if it is inserted downstream as suggested. Basic conversion options Some conversion options are of a profound nature, e.g.
Unwrap HD Then choose between the four Contour Styles, and finally apply EQ to the center channel if desired. UnWrap HD’s 48 bit EQ can work wonders on most signals and be used to selectively suppress spectral ranges where the L/R width could otherwise get compromised, or to boost selected frequencies to strengthen the center anchor function. Surround page To control the surround channels, de-correlation, EQ and contour controls are provided. First set the Ref.
Unwrap HD LFE Processing LFE Hi Cut Frequency Range: 10 to 200 Hz Sets the Hi Cut frequency for the output from the LFE channel. ►► Press Bypass EQ to bypass the entire EQ. Bypass does not affect the selected Contour Style. Type Selector Press Type and use faders 1 to 4 to select filter types. For Lo and Hi filters select between filter types: Parametric, Notch, Shelve and Cut. For Mid 1 and Mid 2 filters select between filter types: Parametric and Notch.
Unwrap HD Range for the Shelve filter: Lo BW: 3 to 12 dB/oct ►► Hi BW: 3 to 12 dB/oct ►► Range for the Cut filter: Lo BW: Bessel or Butterworth ►► Hi BW: Bessel or Butterworth EQ The EQ for the Center channel features fourband parametric EQ with high- and low-pass filters switchable between Notch, Parametric, Shelving and Cut filters. ►► Unwrap HD algorithm – surround page Basic operation Select Freq, Gain or Type to access the same parameter for the four EQ bands.
Unwrap HD Range for the Notch filter: Lo BW: 0.02 to 1 oct ►► Mid1 BW: 0.02 to 1 oct ►► Mid2 BW: 0.02 to 1 oct ►► Hi BW: 0.02 to 1 oct ►► running 48 kHz, a 48 samples delay equals 1 ms, and at 96 kHz it equals 0.5 ms. Unwrap HD algorithm – output page Range for the Parametric filter: Lo BW: 0.1 to 4 oct ►► Mid1 BW: 0.1 to 4 oct ►► Mid2 BW: 0.1 to 4 oct ►► Hi BW: 0.
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Unwrap HD 158 System 6000 MKII Algorithms – Firmware version 6.5.
Brickwall 2 The BrickWall limiter is designed: 1) to investigate if your signal is contaminated with 0 dBFS+ peaks 2) to remove them.
Brickwall 2 ting, internal variables other than Release Time will be fixed. Absolute Bit Transparency Whenever the BrickWall limiter is not attenuating the signal, it is passed completely unaffected, transparent to the 24 th bit. Consequently, the BrickWall limiter may be used to remove 0 dBFS+ peaks on already mastered and finally dithered material. When the Limiter is bit transparent, a green indicator next to the Threshold control lights up. Input and Output gain controls have 0.
Brickwall 2 More information … About 0 dBFS+ signals, required headroom in the signal path, behavior of consumer CD players and production equipment can be found at the TC website, from which technical papers about related issues also can be downloaded. Brickwall 2 algorithm – main page Green Input Gain LED indicator Indicates bit transparency (read introduction for details). Link Press to Link L and R Gain controls and the sidechains.
Brickwall 2 subsequently low bit-rate data reduced. In such cases try around -2 dBFS, and listen through the encoder and decoder Softclip Range: Off, +18 dB, +12 dB, +6 dB, 0 dB, -6 dB, -12 dB When the post Gain level exceeds the threshold set, Soft Clip is activated. When Soft Clip is in operation, the red indications above the Gain Reduction meters light up. Soft Clip deliberately adds harmonic distortion, and may be balanced against the adaptive limiter for more apparent loudness, or as an effect.
Brickwall 2 Brickwall 2 algorithm – setup page Output Output Range: Off to 0 dB Output Level control. Brickwall 2 algorithm – setup page Timing Delay For alignment two Delay parameters is available for both left and right channels: The total delay-time for each channel is the sum of these two Delay settings. Delay Left and Right Adjust Delay time in course mode either displayed in milliseconds or Frames. Delay Fine Left and Right Additional Delay to the course parameter setting, adjusted in Samples.
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Brickwall 2 Loudness correction English Manual – Updated 2018-02-28 165
Brickwall 2 166 System 6000 MKII Algorithms – Firmware version 6.5.
ALC 6 ALC 6 ALC6 is an automatic loudness adjustment processor for mono, stereo, LtRt and 5.1 formats. It is a further refinement of the predecessor, ALC5.1, and available free of charge to users of DB4, DB8, DB4 MKII and DB8 MKII. ALC6 is primarily designed for broadcast ingest, linking and transmission, but it may be used as a safety belt in production and in live production as well. er broadcast platforms, such as analog TV, mobile TV and IPTV.
ALC 6 ALC6 – main page ALC6 – setup page Fig 1. TC Icon view of ALC6 Main page parameters. Fig 2: ALC6 algorithm – setup page Preset title The Main page of any algorithm in DB4 and DB8 displays the title of the current preset. Click on the Name field to edit a preset title, and Store the changes if you wish to keep them. Input Level Input gain applied to all 5.1 channels before loudness detection or processing is applied. The range of the Input Level parameter is -18 to +18 dB.
ALC 6 ALC6 – ALC page ALC6 uses a Target Level of -21 LFS for 5.1 programs, but a Target Level of -24 LFS for stereo. Fig 3: ALC6 algorithm – ALC page Target Level Target Level sets the Loudness Target aimed for by ALC6. The unit is shown as “LFS”, which denotes “LKFS” as well as “LUFS” (the two are identical). For normal broadcast, the value should typically be between -18 and -24 LFS. Note that the distance between this value and Limit Threshold on the Limit page is a quality-defining factor – i.e.
ALC 6 rides Corrections when the input level is higher than Target. Stridency Reduction As consumers, once we have adjusted the gain for a suitable sound pressure level, we are more tolerant of loudness dropping than of loudness going up – see “Loudness Descriptors to Characterize Wide Loudness-Range Material” from the 127th AES conference, 2009. This parameter determines how programs are dealt with (see Fig 4), and it is influenced by the Pre Process parameter.
ALC 6 ALC6 – Limit page The Limiter in ALC6 uses true-peak detection exceeding the oversampling requirements of BS.1770. LFE Trim Use LFE Trim to apply static gain control to the LFE channel after the ALC section, but before the output limiter. The range of the Trim parameter is -18 to +18 dB. Because DB4 and DB8 use 48 bit processing, a positive setting does not create overload, even if the signal is already at full scale.
ALC 6 threshold is exceeded on the LFE channel, LFE is limited independently. C, LR, LFE If the threshold is exceeded in the Center channel, only that channel is limited. If the threshold is exceeded in one of the other Main channels, all Main channels excluding Center are limited. If the threshold is exceeded on the LFE channel, only that channel is limited. Note Profile setting interaction. Profile Use Profile to optimize the adaptive true-peak limiter based on content, or based on the output platform.
ALC 6 ALC6 used for Transmission 1. Load a suitable ALC6 preset, then follow these directions in the given order. 2 a. If programs have already been normalized to Target level before arriving at the processor, set the Pre Process parameter to “Normalized”. 2 b. If many programs have not been normalized to Target level before arriving at the processor, set the Pre Process parameter to “Not Norm” or to “Universal”.
ALC 6 ALC6 used for Ingest 1. Load a suitable ALC6 preset, then follow these directions in the given order. 2. Use the LM6 radar meter in another engine to measure unprocessed parts of the program considered to be average foreground level. For a speech-based program, measure average speech. For a music-based program, measure mezzo-forte parts/movements. For a movie, anchors such as regular speech or intro/outro music can be used.
ALC 6 ALC6 used for Live Production Because of its low latency (0.8 ms, equivalent to the time it takes sound to travel 25 cm or 10 inches) and high resolution, ALC6 is also suitable for providing a loudness and true-peak level safety belt when live mixing, e.g. in OB trucks: You can monitor after the processor without worrying about mix-timing. 1. Load a suitable ALC6 preset, then follow these directions in the given order. 2. Use a radar meter, LM6, in another engine to measure the input of ALC6.
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LM6 LM6 LM6 represents a quantum leap away from simply measuring audio level to measuring perceived loudness. The old level method is responsible for unacceptable level jumps in television, for music CDs getting increasingly distorted, and for different audio formats and program genres becoming incompatible: Pristine music tracks from the past don’t coexist with new recordings, TV commercials don’t fit drama, classical music or film and broadcast doesn’t match.
LM6 tion with long-term descriptors from production onwards, is a transparent and well sounding alternative to our current peak level obsession. Not only for music, but also in production for broadcast or film. The engineer, who may not be an audio expert, should be able to identify and consciously work with loudness developments within the limits of a target distribution platform, and with predictable results when the program is transcoded to another platform.
LM6 conditions etc using normal DB4 and DB8 preset handling procedures. Radar page Current Loudness: Outer Ring The outer ring of the Radar page displays Momentary loudness. The 0 LU point (i.e. Target Loudness) is at 12 o’clock, and marked by the border between green and yellow, while the Low Level point is marked by the border between green and blue. The “0 LU Equals” and “Low Level Below” parameters are found on the Setup page.
LM6 rely on this parameter may take place in the consumer’s receiver. Therefore, its value should not be far off target, or the consumer results become highly unpredictable. LM6 algorithm – main page Program Loudness in LM6 is directly compatible with Dialnorm in AC3. Most broadcast stations work with a fixed dialnorm setting, for instance –23 LUFS. This would be the Program Loudness target level for any program.
LM6 This may be regarded as initial production guidelines: ►► ►► ►► HDTV and digital radio: Stay below LRA of 20 LU. SDTV: Stay below LRA of 12 LU. Mobile TV and car radio: Stay below LRA of 8 LU. Remember to use LRA the other way around too: If there is an ideal for a certain genre, check its LRA measure, and don’t try go below it. LRA should not be used for Limbo. Allow programs or music tracks the loudness range they need, but not more than they need.
LM6 Target Range: -36 LUFS to -6 LUFS The parameter specifies the loudness level to generally aim at. It affects a number of functions and displays in LM6, and must be set according to the standard you need to comply with. Current broadcast standards require Target to be in the range between -26 and -20 LUFS. For instance, EBU R128 calls for -23 LUFS while ATSC A/85 specifies -24 LUFS. The Target parameter affects these LM6 functions and displays: 1.
LM6 genres. BS.1770-2 enables the meter reliably to focus on foreground sound, and to transparently control loud commercials. ARIB (Japan) specifies BS.1770-2 in TR-B32. EBU (Europe) specifies BS.1770-2 in EBU R128 and in associated Tech Doc 3341. Target Level in these countries is -23 LUFS or -24 LUFS, measurement gating at -10 LU. United States: Page 11 of ATSC A/85 (May 25, 2011) references ITU-R BS.1770-1, even though BS.1770-2 was in effect at that time.
LM6 Stereo or 5.1 Integrity In this mode, Integrity is given when either Stereo or 5.1 Integrity are detected. This means that the LED is lit when neither valid Stereo nor 5.1 signals are detected. Off The Alert indicator is disabled. LM6 algorithm – stats page LM6 algorithm – stats page The Stats page gives an overview of essential descriptors. Note! The Reset button resets the meters and the log file.
LM6 built into the standard. The final BS.1770 standard included a multichannel annex with a revised weighting filter, R2 LB – now known as “K” weighting – and a channel weighting scheme. These two later additions have been less verified than the basic Leq(RLB) frequency weighting. ital peaks at 0 dBFS, is regarded a 0 dBFS tone, BS.1770 and LM6 output these results: The other aspect of BS.1770, the algorithm to measure true-peak, is built on solid ground.
LM6 Post Script Control of loudness is the only audio issue that has made It to the political agenda. Political regulation is currently being put into effect in Europe to prevent hearing damage and disturbances from PA systems, and to avoid annoying level jumps during commercial breaks in television. In Australia, something similar may happen.
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Matrix 88 Dissolve Timing Matrix 88 Features ►► ►► ►► ►► ►► ►► ►► 8 x 8 48 bit mixer algorithm Precision delay on all inputs and outputs 2 stereo AUX sends with pre/post 2 main mix outputs Smooth crossfading between channels Brickwall II Limiter Part of the routing matrix ►► @ Normal Sample Rate: 1 DSP Resource @ Double Sample Rate: 1 DSP Resource (= 4 instances possible) The inputs and outputs of this algorithm are distributed as follows: Input 1L 1R 2L 2R 3L 3R 4 5 Mute Duration sets the fade-out
Matrix 88 Ch1 / Ch2 / Ch3 Matrix 88 algorithm – mix page Inv Phase inverse for each input. Delay 1 to 3 Range: 0 to 333 ms / 0 to 14700 smp / 0 to 111 m / 0 to 373 feet Delay compensation for stereo channel pairs 1 to 3. Ideal for time alignment of input sources. See application examples. Balance 1 to 3 Left/right balance for stereo channels 1 to 3 The range is from hard left to hard right and is adjustable in 0.1 dB steps.
Matrix 88 input and output faders also operate in the 48 bit domain, thereby enabling loss-free cut and even boost. Matrix 88 is bit-transparent from all inputs to all outputs with or without delay enabled. Bit pattern transparency is indicated by green dots next to the faders on the Mix page. Note how Pan and Balance controls are bit transparent and unity gain at center position. Only one channel is affected when the control is moved away from center position.
Matrix 88 Main Outputs Delay Main A/B Range: 0 to 333 ms / 0 to 14700 smp / 0 to 111 m / 0 to 373 feet Use these parameters to set the delay compensation for each Master outputs. Master Fader Range: Off to 0 dB. Master level control for Main outputs A. Main B Fader Range: Off to 0 dB. Master level control for Main outputs B. 194 System 6000 MKII Algorithms – Firmware version 6.5.
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Chorus-1 Chorus Chorus-1 The Chorus algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R Output L R Speed Range: 0.1 to 10 Hz Controls the Rate of sweep in a range from 1 sweep every 10 seconds to 10 sweeps every second. Depth Range: 0 to 100 % Determines how wide a modulation (sweep) is produced.
Chorus-1 Feedback Hi Cut Range: 1 kHz to off Feedback Hi Cut enables you to remove high frequencies from the feedback loop. Delay Delay Range: 1 to 670 ms A Chorus is basically a Delay being modulated by an LFO. The Delay parameter controls the length of this Delay. A typical Chorus uses Delays of approx 10 ms. Hi Cut Freq Range: 100 Hz to flat Hi Cut filter enables you to make the Chorus sound more “warm”. This is a 6 dB per octave filter. Hi Cut Att Range: -40 to 0.0 dB Gain for Hi Cut filter.
Phaser-1 Phaser-1 algorithm – main page Phaser-1 The Phaser-1 algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: 1/4 DSP Resource The inputs and outputs of this algorithm are distributed as follows: Input L R Output L R E1 E2 E3 E4 Phaser-1 algorithm – introduction A Phaser in general is a group of comb filters that are swept back and forth by an LFO within a certain frequency range.
Phaser-1 Mix Range: 0 to 100 % Mixes between direct sound and phaser sound. 200 System 6000 MKII Algorithms – Firmware version 6.5.
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Backdrop BackDrop was specifically designed to remove noises found in today’s world (hums and buzzes, room rumble, air conditioning systems, tape hiss, etc.) and is not intended to be used for removing clicks and pops.
Backdrop signal. For certain stereo audio sequences, operating in M/S mode can give better results than when operating in Stereo mode. It should be noted that when you change the Processing Mode, BackDrop automatically recalculates the noise print information for the current operating mode. Backdrop algorithm – model page(s) Left and Right Input Trim The Left and Right Input Trim controls allow attenuation of the incoming signal level.
Backdrop The Max Reduction parameter tells BackDrop the maximum amount of noise reduction you are trying to achieve. Therefore, if Max Reduction is set to say 10 dB, then BackDrop will not try to lower the noise by more than 10 dB. Depending on how the Aggression parameter is set, BackDrop may reduce the noise by less than 10 dB, but not more. If Max Reduction is set to 0 dB, BackDrop will not remove any noise.
Backdrop MultiBand Model MultiType Model Backdrop algorithm – MultiBand model page Backdrop algorithm – MultiType model page The MultiBand model offers an enhanced set of controls compared to the Basic model. The MultiBand model provides three independent noise reduction engines split across three frequency bands. Therefore you can apply as much or as little noise reduction as is needed in each of the three frequency bands.
Backdrop is often dominated by lower frequency noise. Finally, the Hiss component consists of the broadband noise having a relatively flat spectrum. In the example shown in the figure the noise at higher frequencies is predominantly due to hiss. BackDrop allows you to selectively reduce each of these noise components. The MultiType model is a good choice when the noise is made up of two or more components or when you have noise (such as hum due to ground loops) with clear tonal components.
Backdrop print information based on the audio contained between the Trim Start and Trim End points of the captured audio buffer. It should be noted that when switching between Stereo and M/S processing mode, BackDrop automatically recalculates the noise print information for the current operating mode. BackDrop does this using the current Trim Start and Trim End settings. Play The Play button starts and stops playback of the captured audio buffer.
Backdrop put signal is relatively high, you will likely begin to hear strange low-level random chirping sounds. These low-level artifacts are usually referred to as “musical noise” or sometimes as “Marsmen”, or “space monkeys”. These artifacts are commonly found in other noise reduction products and can seriously limit the usability of the noise reduction system. Fortunately, BackDrop provides you with a means of eliminating these artifacts. Slowly increase the level of the Basilar Dispersion parameter.
Backdrop Max Reduction to 0 dB. Slowly increase Max Reduction and listen as the level of noise goes down. Use the Bypass button to turn the processing on and off. Adjust Max Reduction until you obtain the appropriate amount of noise reduction. With Aggression set to 100 %, you are probably applying more processing than necessary for the amount of noise reduction that you desire. Therefore, slowly lower the amount of Aggression until you here the level of the noise floor begin to increase.
Backdrop Backdrop algorithm – frequently asked questions Is it better to get as long a noise print as possible? The simple answer is yes, but the maximum length of the capture buffer is 1.5 seconds. So there is no advantage to having a noise print that is longer than 1.5 seconds. BackDrop doesn’t need more than 1.5 seconds in order to fully train itself on the noise in the signal. Is it important to get a very good loop of the noise? Again the answer is yes.
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SpacePan 5.1 Focus and Bleed parameters SpacePan 5.1 As these parameters highly affect each other, you should understand the functionality of both. Please read the following few lines. The SpacePan 5.
SpacePan 5.1 The parameters for each of the 8 sources can be adjusted as follows: With the Fader Link key disabled: Press S1 to S8 to locate the positions of each source. The selected Source is indicated with a yellow dot. Positions are selected by pressing the screen at the desired location. Scroll through the source parameters using the Fader Group keys.
SpacePan 5.
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VP-2 Stereo this processing can be conceived as a Dual Pitch Brain mode. VP-2 Stereo The VP-2 Stereo algorithm occupies: ►► ►► @ Normal Sample Rate: 1/4 DSP Resource @ Double Sample Rate: Not available The inputs and outputs of this algorithm are distributed as follows: Input Levels InLevel L/InLevel R Range: -100 to 0 dB Sets the Input level for left and right channel.
VP-2 Stereo Factor/Adjust Range: 1, 10 or 100 The Factor parameter is a multiplier factor for the Adjust handles. Use the Adjust handles to finetune the transpose value. VP-2 Stereo algorithm – effect page Pitch Splicing The Splicing parameters are used to optimize how the pitch brains compute the splice points. If very short Delay times are used, audio quality trade-offs must be expected. Max Delay – Left & Right Range: 0 to 100 % Sets the maximum delay used for pitch change purposes.
VP-2 Stereo Cross L to R Range: 0 to 100 % The percentage of processed signal from the algorithm’s left Output fed back to the right Input of the algorithm. Cross R to L Range: 0 to 100 % The percentage of processed signal from the algorithm’s right Output fed back to the left Input of the algorithm.
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VP-8 ►► VP-8 ►► ►► The VP-8 Stereo algorithm occupies: ►► ►► ►► ►► @ Normal Sample Rate: 2/4 DSP Resource @ Double Sample Rate: not available. ►► ►► ►► The inputs and outputs of this algorithm are distributed as follows: Input 1 2 3 4 5 6 7 8 E1 E2 E3 E4 Output 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 L R Center n/a SL SR LFE n/a For 5.1 transposing or Doppler shifting of music material, the recommended assignment is: ►► ►► ►► ►► ►► ►► ►► ►► 1 2 3 4 5 6 7 8 L R Center LFE SL SR n/a n/a For both 5.
VP-8 For non-music applications, you may prefer to keep the LFE channel free of any transposing. (Low frequency material requires long splice times and processing delay when transposed). VP-8 algorithm – main page Delay Units Range: ms, 24 frps, 25 frps, 30 frps, 30 Dframes This parameter changes the type of units the Additional Delay is viewed in. Pitch Mode Mode Range: Link, Split Enables linking of the Pitch and Pitch Splice parameters between the two channel groups.
VP-8 Factor/Adjust Range: 1, 10 or 100 The Factor parameter is a multiplier factor for the Adjust handles. Use the Adjust handles to finetune the transpose value. Pitch Splicing The Splicing parameters are used to optimize how the pitch brains compute the splice points. If very short Delay times are used, audio quality trade-offs must be expected. Delay – Ch. 1 to 4 and 5 to 8 Range: 0 to 100 % Sets the maximum delay used for pitch change purposes.
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