TC Electronic SYSTEM 6000 MKII Algorithm Manual
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TC Electronic SYSTEM 6000 MKII Algorithm Manual

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  • Page 1 Algorithm manual English...

  • Page 3: Table Of Contents

    TAblE Of COnTEnTs PAGE HEAD Table Of Contents ............3 Introduction ..............4 Reverb Introduction ............5 VSS™3 ................7 VSS™SR (Surround) ..........10 Nonlin-2 ............... 19 DVR-2 ................21 VSS™ 5.1 - Source ............. 23 VSS™ 6.1 - Reverb ............. 27 Reverb - 2..............

  • Page 4: Introduction

    InTrODuCTIOn Channel Distribution In System 6000 contains a wealth of algorithms, and the list of algorithms will continuously be extended. Our main focus Surround Algorithms is to offer the best possible quality - both as stereo- and multi-channel versions. To best comply with the channel allocation used by most digital AES-format equipment the Input/Output channels on Mainframe 6000 holds surround algorithms are allocated as follows:...

  • Page 5: Reverb Introduction

    rEvErb InTrODuCTIOn The TC Reverb Palette all the difference in the world when it comes to obtaining depth, expression and natural imaging. Until 15-20 years ago, digital reverb was mostly used Source Reverbs are able to generate multiple, complex as a generic effect applied to many sources of a mix. early reflection patterns. For best results, if some Nowadays, where more aux send and returns are at reflections are already picked up by a microphone, they...
  • Page 6 rEvErb InTrODuCTIOn Reverb 6000 algorithm overview Algorithm Name Input Format Output Format Reverb Type Rev 3 Stereo Stereo Generic VSS 3 1 Source Stereo Source VSS 3 SR 1 Source LtRt Source NonLin 2 Mono Stereo Generic DVR 2 Mono Quad Generic VSS 4...

  • Page 7: Vss™3

    vss™3 VSS™3 Hi Cut Range: 20 to 20kHz The VSS™3 Reverb algorithm incorporates dedicated Early Rolls off the top end as it enters the Reverb. Reflection types for motion picture use, e.g. Car, Bathroom Used in conjunction with Hi Soften and Hi Decay to and Conference Rooms. “darken” a room. The VSS-3 algorithm occupies: Hi Decay @ Normal Sample Rate : 1/4 DSP Resource Sets the Decay time for the High-end frequencies of the @ Double Sample Rate...
  • Page 8 Early Reverb Early Type Reverb Several types are available - e.g. Bathroom, Car etc. Rev Type Pick the type that best compliments your material or best Range: Smooth, Natural, Alive, Fast, Fast Wd, Alive Wd represents the effect you are going for. Adjust this parameter with the Early Level turned all the way off and the Rev Level all the way up. Change the type Early Color to get a feel of what each one sounds like.
  • Page 9 Modulation Rev Balance Range: -100dB R, center, -100dB L The left/right balance of the Reverb. Allows you to offset the diffuse field from the normal center position. 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.

  • Page 10: Vss™Sr (Surround)

    vss™sr (surrOunD) VSS™SR (Surround) Rear Delay Range: 0 to 200ms The VSS™SR (Surround) algorithm is a unique room Changes the Reverb Feed Delay time of the Rear/Surround simulator with new facilities for 4:2:4surround production. information in the signal. The diffused field of the simulation is turned into a Front/ Rear composition with separate Decay, Level and Predelay Pre Delay parameters for front and rear.
  • Page 11 vss™sr (surrOunD) Early Reflections Reverb Early Type Rev Type Range: Several types Range: Smooth, Natural, Metal, Fast, Fast WD Pick the type that best compliments your material or best Adjust this parameter with the Early Lev turned all the way represents the effect you are going for. off and the Rev Lev all the way up. Change the type to get a feel of what each one sounds like. Early Size Rev Width Range: Small, Medium, Large Changes the size of the Early Type parameter.
  • Page 12 vss™sr (surrOunD) 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. Decay/Crossover Lo Decay Range: 0.01 to 2.5 Multiplier for the frequencies below the Lo Xover frequency.
  • Page 13 vss™-4 VSS-4 Master Decay Range: 0.1 to 20 sec VSS-4 is a True Stereo Reverb - or two Source Input to Adjusts the Master Reverb Decay time. Decay time can via Stereo Output reverb - and radically departs from being an the Color page be further adjusted using the multipliers for additive sustain diffuse field added to a 2-channel signal. the following bands: Lo, LoMid, HiMid and Hi.
  • Page 14 vss™-4 Setup Lo Color Range: -50 to 50 Lo Color adjustment of the Early Reflections Hi Color Range: -50 to 50 Hi Color adjustment of the Early Reflections 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 remains unchanged. Therefore this adjustment is typically more acoustically precise and useful than a normal Predelay control.
  • Page 15 vss™-4 Color HiMid Decay Decay multiplier in relation to the Master Decay, for the frequencies above the Mid Xover and below the Hi 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.
  • Page 16 vss™-M4 Introduction Main page 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. M4 reverbs can be used in multiple mono configurations, where the effect has to end up in predictable places, or combined to cover arbitrary multichannel configurations like 7, 10 or even 16 main channels with different settings, all within one Reverb/System 6000 frame.
  • Page 17 vss™-M4 Levels page Reverb pages 1-4 In Level (Reverb 1 to 4) Reverb Range: Off to 0dB Master Decay (Reverb 1 to 4) Individual input level of the four reverbs. If the input signal Range: 0.1 to 20s is close to full scale and/or long decay times are used, it Individual Decay parameters of the four reverbs.
  • Page 18 vss™-M4 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. A setting of “0” denotes a default setting. Decay Lo Decay Range: 0.01 to 2.5 Decay multiplier related to Master Decay, for frequencies below the Lo Xover setting.

  • Page 19: Nonlin-2

    nOnlIn-2 Introduction Main Page NonLin 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. Nonlin also does classic Gated Reverb, but because it doesn’t need to be triggered, it can be used on all sorts of program material. The eighties with a Twist Compared to effect Reverbs from the past, Nonlin features longer and more diffused response with a higher resolution, but classic, low density processing is also available.
  • Page 20 nOnlIn-2 Levels page Differences are more pronounced with low Diffuse settings and Twist Ratios. Diffuse Range: 0 - 100% Sets the amount of diffusion applied to the Reverb Style selected. For high Diffuse ratios, the Attack and Decay times of the reverb Envelope may be slightly affected. Twist Type Selects which type of funky effect treatment to be used on the reverb. Twist Type should be regarded as a spice capable of sparking imagination in the mix process. Try the effect on all sorts of different sources to add character or make them stand out. If spectral shaping using the Twist...

  • Page 21: Dvr-2

    Dvr-2 Introduction The DVR-2 Source algorithm occupies: DVR-2 offers Reverb and System 6000 users a pristine @ Normal Sample Rate : 1/4 DSP Resource Generic Reverb with true vintage flavor. Generic Reverb is @ Double Sample Rate : Currently not available complementary to Source Reverb, and both types are at disposal in the 6000.
  • Page 22 Dvr-2 Level Screen Reverb Mix Fr. Wet Level Range: Off - 0dB Adjusts the reverb level of the two front output channels. Fr. Dry Level Range: Off - 0dB Adjusts the dry signal level of the two front output channels. Sr. Wet Level Range: Off - 0dB Adjusts the reverb level of the two surround output channels. Sr. Wet Level Range: Off - 0dB Adjusts the dry signal level of the two surround output channels. Advanced Enable Levels...

  • Page 23: Vss™ 5.1 - Source

    vss™ 5.1 - sOurCE Main Page Choosing VSS-5.1 Source or VSS-5.1 Reverb • If you wish to add reverb to a complete 5.1 mix or 5.1 At the Main - Master page overall parameters working on stem, use the “VSS-5.1 Reverb” algorithm. the complete algorithm are available. Individual parameters • If you need free dynamic movement of sources utilizing for each of the four Early Reflection generators and five external joysticks or SpacePan 5.1, use the Reverb Diffuse fields are also available. They are placed “VSS-5.1 Reverb” algorithm. on the following sub-pages. • If you wish to position single or composite sources with max localization and sweet spot enhancement, use the “VSS-5.1 Source” algorithm.
  • Page 24 vss™ 5.1 - sOurCE Select Output mode between three speaker setups: • 5.1 Music (ITU775) with 5 identical speakers and Master Levels a +/-30/110 degree azimuth. ER Trim • 5.1 Cinema set-up with several surround speakers set-up Range: +/-12dB in double L configuration. Relatively adjusts the four Early Reflections Input level • Stereo (Output on Left and Right front-speaker ). parameters. Center Channel Reverb Trim Range: Off/On/Phantom On Range: +/-12dB This parameter enables on/off setting of the Center Relatively adjusts the four Reverb input levels from the four speaker in the two multi-channel setups.
  • Page 25 vss™ 5.1 - sOurCE Positions The parameter range limit will be less than 100% Listening Position if the Early Start parameter is set to anything other In some large locations it’s possible to change the listeners than 0%. position. In the smaller locations there is one listening position available. Reverb Position Enables to emphasize or change the Source position origin Source 1 to 4 positioning in the Reverb part of the algorithm. In the selected location it is possible to position the four Input sources individually.
  • Page 26 vss™ 5.1 - sOurCE Hi Color Center Decay Range: +/-50 Range: 0.10 to 2.00 Adjusts the spectral balance in the high end frequencies of Adjusts the Center speaker Reverb Decay time. the Reverb. All five diffuse fields respond to this parameter. RFr Decay Range: 0.10 to 2.00 Surround Channel Colors (Additive) Adjusts the right-front speaker Reverb Decay time. Lo Color LSr Decay Range: +/-50 Range: 0.10 to 2.00 Additional Color adjustment of the Surround channels.

  • Page 27: Vss™ 6.1 - Reverb

    vss™ 6.1 - rEvErb Main Choose Your Verb Make sure to read the Reverb Intro in this manual about distinguishing 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 6.1 room simulation to cover wide audiences like in a cinema, or make the most out of SACD 6.0 reproduction. The improvements over 5.1 must be heard to be believed, so make sure to enable 6.1 mode when a separate Surround channel is available.
  • Page 28 vss™ 6.1 - rEvErb Setup some reflections. Based upon perceptual experiments, we have found that reducing particular reflections is a method most useful for optimizing the room response to a given microphone placement. The 0% setting enables the full response, which should be preferred for very closely mic’ed sources, while 100% ER Decrease is useful for sources picked up at a distance.
  • Page 29 vss™ 6.1 - rEvErb Decay Crossover Dry Level Range: Off to 0dB Lo Decay Sets the amount of Dry signal that is passed through the Range: 0.01 to 2.5 algorithm. Off equals a kill-dry setting for a send/return Decay multiplier in relation to the Master Decay, for configuration, and is the default on Factory presets. frequencies below the Lo Xover setting. Example: When a Decay multiplier is set to 1.25 and the Color Master Decay is 2.0 seconds the resulting decay time will...
  • Page 30 vss™ 6.1 - rEvErb Gloss Reverb Diffuse Range: -50 to +50 Offsets the diffusion away from the default of a given Decay time. For optimum performance, the diffusion is automatically adjusted when Decay is changed. Reverb Size Range: 0 to 15 Adjusts the overall perceived size of the Diffuse Field part of the response. It affects the spread and correlation of the diffused response, not the decay time. The default value depends on the Location Type chosen.
  • Page 31 vss™ 6.1 - rEvErb the illustration the grey areas are the directions that are reproduced at full level. Front & Surround Level Range: Off to 0dB When the Front Restriction parameter is adjusted, the angle between the frontal and lateral zone is affected. When the Surround Restriction parameter is adjusted, the angle between the lateral and dorsal zone is affected. The level of reflections arriving from the frontal and dorsal zones are controlled by the level controls Front Level and Surround Level. If Front Level and Surround Level are set...

  • Page 32: Reverb - 2

    rEvErb - 2 Main The Reverb 2 algorithm initially created for the TC M5000 compliments most types of source material, however Decay experience has shown that the Reverb 2 algorithm is Range: 0.3 to 60.0s especially good on percussive instruments, as it has a very Reverb diffuse field Decay time. well-defined precise buildup.
  • Page 33 rEvErb - 2 Reverb Page Xfeed Range: On/off With this parameter switched off, the cross feeds in the Early Reflections will be killed. The ER xFeed switched off, simultaneously with the parameter Rev Width set to 100%, will create a true stereo Reverb. The effect from the left and the right channel will be generated totally independent. This is ideal for working with Dolby surround or for broadcasting in general where mono compatibility is important.

  • Page 34: Reverb - 3

    rEvErb - 3 Main This is a description of the parameters specific to the Reverb-3 algorithm. It is capable of making an exceptionally Decay clear Reverb sound using a very dense and natural Range: 0.3 to 30.0s sounding Reverb Tail. Decay time can be controlled in Reverberation Decay time. four individually adjustable frequency bands. Using Diffuse and the Distance controls, sounds can be made in which Pre Delay...
  • Page 35 rEvErb - 3 Reverb Page Modulation Rate Range: 1 to 200 The Modulation Rate varies the rate of modulation of the recirculating delay paths simulating the Reverb Tail. The control has no effect at a “0” setting. Adding modulation to the Reverb has the effect of smoothing out the frequency response of the Reverb, by effectively averaging out the room resonances.

  • Page 36: Rev Core - 2

    rEv COrE - 2 Main The TC Electronic REV CORE algorithms are particularly good for small room simulations. Due to the high density Decay structure the relatively short Reverb diffuse fields occurring Range: 0.3 to 60s in small rooms can be convincingly reproduced. Though Reverberation Decay time. the VSS algorithms are dedicated for Film and Post production the Rev Core algorithm, known from the TC Pre Delay...

  • Page 37: Rev Core

    rEv COrE - 2 Reverb Hi Xover Range: 20Hz to flat Sets the frequency at which the transition from the mid frequencies to the high frequencies takes place. Width Range: 0 to 100 % Sets the apparent stereo width of the reverberating part of the algorithm. At “0” the diffuse field will appear to be coming mainly from the center (mono compatible), whereas with Width set to “100” the L/R reverberators are independent.

  • Page 38: Md-3

    MD-3 The MD-3 algorithm occupies: High resolution, multiband dynamics processing has been a trademark of TC Electronic for more than a decade. @ Normal Sample Rate : 1/4 DSP Resource Splitting audio into frequency bands before expanding @ Double Sample Rate : 1/4 DSP Resource or compressing, tremendously helps fight breathing and Algorithm Inputs/Outputs are distributed as follows: spectral intermodulation artifacts.
  • Page 39 MD-3 MS Unlinked In the MS Unlinked mode the MS Encoder is activated and the Expander/Compressor section of the algorithm uses separate SideChains for the Left and Right Channels. EQ is unlinked. Output Mode Range: Left/Right, MS Left/Right : Straight Output without MS Decode. : MS decode before the section. DC block Range: Off, On The DC block is a Lo Cut filter used to remove potential DC-noise at 2Hz. Expander/Compressor Setup Lo Xover Range: Off, 25, 32, 40, 50, 63, 80Hz ..16kHz...
  • Page 40 MD-3 Reference Level the Expander/Compressor section of the algorithm uses one common SideChain for both Left and Right channels. Range: -24dBFS to 0dBFS in 0.5dB steps EQ is unlinked. This parameter sets the reference level in the algorithm. The reference level is the level at which the Compressor MS Unlinked Threshold parameters will start operating when set to 0dB. In the MS Unlinked mode the MS Encoder is activated and the Expander/Compressor section of the algorithm uses separate SideChains for the Left and Right Channels.
  • Page 41 MD-3 Cut Filter - Butterworth type Parametric Filter - Broad type Freq Shelving Filter Press Freq and use Faders 1 to 4 to adjust frequence for each of the four bands. Range - Lo band : 20Hz to 20kHz Range - Mid1 band : 20Hz to 20kHz Range - Mid2 band : 20Hz to 20kHz Range - Hi band : 20Hz to 40kHz Gain Press Gain and use Faders 1 - 4 to adjust gain for each of the four EQ bands. Range for the Parametric, Shelve and Cut type: Lo Gain : -12dB to +12dB Notch Filter - Narrow Type Mid1 Gain...
  • Page 42 MD-3 Expander Range for the Cut filter: Lo BW : Bessel or Butterworth Hi BW : Bessel or Butterworth Bandwidth/Q - Key-Values: 0.5 2.87 2.04 1.41 Normalizer Expander Page Pressing Threshold, Range, Ratio, Attack and Release keys will immediately assign Lo, Mid, Hi and Master values for these parameters to Faders 1-4. 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.
  • Page 43 MD-3 Compressor Ratio Range: Off to Infinity Specifies the Ratio of the performed compression. Example: With a Ratio setting of 2:1 the compressor will reduce every 2dB above the Threshold point to only 1dB. Attack Range: 0.3 to 100ms 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 4dB above the set Threshold with a Ratio set to 2:1 and the Attack time set to 20ms, the Compressor will use 20ms to reach a Gain reduction of 2dB.
  • Page 44 MD-3 Output page Output Fader Output Fader Range: Off to 0dB (<-40dB: in 3dB steps, >-40 in 0.5dB steps) Output fader for both Outputs. Balance Range: -6dBL to -6dBR Changes the Output balance between the Left and Right channel. COMPARE key Press to activate compare function. Compare Level Range: -20 to 0dB Due to the difference in level, “in-circuit” and “out-of-circuit”...

  • Page 45: Md-4

    MD-4 Introduction Input In Gain L - In Gain R Multiband processor featuring newly developed DXP Range: 0dB to Off processing that lifts up low level detail rather than Input Gain on Left and Right channels. squashing the peaks. DXP processing is ideal for classical music, acoustic music, voice, film to broadcast transfers Delay –...
  • Page 46 MD-4 Setup digital level, dBFS. Go to the Frame-I/O-Slot pages to check the analog settings. Examples: If Level In and Level Configuration Out are set at 18 dBu, a Reference Range: Stereo, Dual Mono, Stereo MS, Sum Mono, Left Level at -20 dBFS will be reached at an analog level of -2 Mono or Right Mono. dBu. If Level In and Level Out are set at 24 dBu, the same Reference Level setting will be reached at +4 dBu.
  • Page 47 MD-4 delay influenced “wet” part of the signal. Inside the If low level Steer gain brings up too much noise, a Defeat algorithm, before the limiter, the “dry” and “wet” signals are function is available to revert to unity gain a certain number summed. of dB below the DXP threshold. When Trim and Band gains are all 0.0 dB, gain through Example: MD-4 in parallel mode is unity, provided the signal is below the Threshold of compression.
  • Page 48 MD-4 EQ Page path. The optimum balance between Look Ahead and Attack may be best found as an iterative process, because Regardless of sample rate, the Eq in MD4 uses 48 bit, fs/2 less tight peak control in the 5 Band section can put more analog modeling. You can for instance specify a boost of 3 processing into the hands of the Limiter. If you use the gain dB at 28 kHz, even though the machine may be operating Defeat function of the 5 Band section, low level transient at a sample rate of 44.1 kHz. What this means is that the handling will be improved by allowing 1 ms of Look Ahead part of the 3 dB boost falling inside the current pass-band Delay.
  • Page 49 MD-4 5 Band Page Limiter Page 5 Band Compressor Fader Groups Brickwall Limiter The following parameter groups and X-Over settings are To exploit MD4’s 48 bit signal path, the output includes easily selected and assigned to the Faders pressing the a Brickwall 2 Limiter with upsampling and selectable, individual buttons. adaptive profiles. For more information about the Limiter, please consult the Brickwall 2 pages in this manual. • Threshold • Gain Soft Clip L - Soft Clip R • Steer (DXP mode) Range: -3, 0, +3, +6dB, Off • Attack When active, Soft Clip applies a saturation effect on signals...
  • Page 50 MD-4 and ambience. Link Limiter Press to link parameters and sidechains of the two channels. Limiter linking is independent of the Configuration parameter on the Setup page. If heavy limiting action is performed, the stereo image may be greatly affected if L and R channels are not linked. Fader Range: Off to 0 dB. Output control(s) for L/R or A and B channels.

  • Page 51: Mdx 5.1

    MDX 5.1 Introduction Applications MDX5.1 is a high resolution dynamics processor for MDX5.1 offers extensive inter-channel linking possibilities, multichannel signals. It presents a sophisticated new angle and is well suited for both music and film material. to dynamic range control, and is an alternative to traditional The ingenious topology of System 6000 allows for low compression and limiting techniques.

  • Page 52: Basic Operation

    MDX 5.1 Basic Operation Fig 3. Example of MDX5.1 Gain Meter.The meter shows max low The Ref Level parameter on the Main page sets the unity level gain and spectral response, gain point for all channels (unless gain offsets are applied), plus current gain and spectral see Fig 2. The Thresholds on the DXP pages are relative response.In the example, the Low...
  • Page 53 MDX 5.1 When it’s indicated to bring up dialog level and speech When making the final mastering adjustments, try intelligibility, you may end up with something like the level changing the Ref Level parameter up and down a diagram presented in Fig 4. few dB. This is an efficient way of trimming hundreds of parameters in MDX5.1 at the same time. Listen to the result, while deciding what is the optimum setting for that particular material. Fig 4.
  • Page 54 MDX 5.1 The MD - 5.1 algorithm occupies: DXP Defeat Level @ Normal Sample Rate : 1/4 DSP Resource Range: Off to -3dB @ Double Sample Rate : 2/4 DSP Resource MDX5.1 may remove low level gain below the threshold set with this parameter to avoid having irrelevant sources (e.g. Algorithm Inputs/Outputs are distributed as follows: background noise) become audible. Low level gain is not revoked if the DXP Defeat Level parameter is set to Off.
  • Page 55 MDX 5.1 Link - Control Side Chain Control Range - for the five main channels: • Unprocessed • Side Chain 1 • Side Chain 2 • Side Chain 3 Range - for the LFE channel: • Unprocessed • LFE Link - Feed Side Chain Control (this description is identical to the MD5.1 explanation) The Sidechain assignment possibilities in the MDX5.1 are very comprehensive. Carefully selecting which channels should be controlled by which Sidechains, is just as essential as dialing in the correct Threshold and Ratio values.
  • Page 56 MDX 5.1 DXP - SC1 pages. Bypass DXP Bypass function for the entire DXP processing section Limit - Soft Clip Sidechain Fader Groups 1/2/3 The DXP pages reveal separate controls for Sidechain 1-3 plus LFE. This allows for different settings for the Center SoftClip or Surround channels, where speech intelligibility or low level ambience tend to get lost, like when a feature film is Full Range SoftClip re-purposed for broadcast or DVD under domestic listening...
  • Page 57 MDX 5.1 Output Brickwall limiter for the five multiband channels. Threshold is always relative to 0dBFS. 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: -0.10dB to 0dB Fine-tuning parameter setting the Ceiling for the Limiter. Limit - LFE Limit Trim Levels Output trims Range: 0dB to -12dB in 0.1dB steps Level trim of the Output channels.
  • Page 58 MD 5.1 Introduction The MD - 5.1 algorithm occupies: @ Normal Sample Rate : 1/4 DSP Resource The MD-5.1 algorithm is a multi-channel multi-band @ Double Sample Rate : 2/4 DSP Resource Expander/Compressor, with Limiters and extensive possibilities to assign channels to Algorithm Inputs/Outputs are distributed as follows: multiple Sidechains.
  • Page 59 MD 5.1 Performance Settings link Inputs, for e.g. having the Center-channel contributing to the Sidechain Inputs of the two Front channels, to create Crest a more coherent sound from the front-channels. Range: Peak, 6dB, 10dB, 12dB, 14dB, 16dB, 20dB, 24dB, RMS The illustration above reflects the Processing parameter set Select compression method between RMS and PEAK. to MD-5.1 in Normal mode. The dB steps between RMS and Peak are the dBs needed for a peak-value to override RMS measurement. Basic operation At the Setup/Control page it is possible to decide which Nominal Delay Sidechains should control which channels.
  • Page 60 MD 5.1 Bypass Exp. the respective Sidechain Input. The extra Sidechain Input channels will not be processed by the sidechain. Press to bypass the Expander section of the MD 5.1 The Sum settings will add the Input to the sidechain, algorithm. whereas the Max settings only will contribute to the sidechain if the level exceeds the other Input channel All LFE page levels.
  • Page 61 MD 5.1 Compressor Compressor Comp. page All LFE Pressing any parameter will assign this to Fader 6. Pressing Threshold, Range, Ratio, Attack and Release keys will immediately assign Lo, Mid, Hi, All and All - parameters LFE values for these parameters to Faders 1-4. Be aware These parameters are equivalent to the “All” - Threshold, that the range of the All parameter is relative to the settings Range, Ratio, Attack and Release parameters. of the same parameters in the Expander section. LFE - parameters Threshold These parameters are equivalent to the “LFE” - Threshold, Range: -25dB to 20dB (in 0.5dB steps) Range, Ratio, Attack and Release parameters. Sets the Threshold level at which the Compressor starts to operate.
  • Page 62 MD 5.1 Limiter -12 to -6 in 0.5dB increments Brickwall limiter for the five multiband channels. Threshold The Limiter page is divided into three Sub-pages. One is always relative to 0dBFS. LED on each Output meter covering the Softclip section, one for the Full Range Limiter indicates when Limiter is active. and one for the LFE Limiter. Release Generic parameters in this algorithm: Range: 0.01 to 1.00 seconds Meter Zoom Release time for the Limiter. Press Meter Zoom to decrease meter range and have a Ceiling more accurate metering.
  • Page 63 MD 5.1 Output Trim Levels Output trims Range: 0dB to -12dB in 0.1dB 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.

  • Page 64: Brickwall 2

    brICkwAll 2 The Brickwall Limiter occupies: on or off using the Upsample key. @ Normal Sample Rate : 1/4 DSP Resource Level meters are always upsampled in order to identify @ Double Sample Rate : 1/4 DSP Resource intersample and 0dBFS+ peaks on Inputs as well as Outputs. The meter scale is extended to +3dBFS because Algorithm Inputs/Outputs are distributed as follows: excessive level normally falls between 0 and +3dBFS, and most consumer equipment already exhibits severe INPUT...
  • Page 65 brICkwAll 2 Soft Clip More information Soft Clip is a bandwidth limited distortion effect, not unlike About 0dBFS+ signals, required headroom in the analog tape saturation. Soft Clipping can add loudness to signalpath, behavior of consumer CD players and a signal at the price of distortion. The Soft Clip threshold production equipment can be found at the TC website, from should be adjusted so only the peak content of the signal is which technical papers about related issues also can be affected, not the average part.
  • Page 66 brICkwAll 2 Green Limiter LED indicator other reasons. Brickwall 2 extends the adaptive release Indicates bit transparency (read introduction for details). time control of Brickwall 1 to include numerous new internal Threshold variables, to make the algorithm respond intelligently for a Range: Off, 0dBFS to -25dBFS number of applications. Activate Adaptive (green) to take When the input level exceeds the Threshold, the signal advantage of the different Profiles. will be attenuated. Gain reduction meters at the bottom of the display display the max amount of reduction applied Profile to either of the channels. When Link is active, the same...
  • Page 67 brICkwAll 2 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. The Samples delay will vary in time depending on the sample-rate.

  • Page 68: Toolbox - 5.1

    TOOlbOX - 5.1 The 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. Signal Flow Input Output Level Trim Bass Format Level Trim Solo Manage- Conversion Ref. Levels Mute ment Limiting Solo/Mute Calibration Solo noise-tone When a Solo button is selected, the Inputs of all the five remaining channels will be set to Off, but they can be selected as additional solo channels.

  • Page 69: Bass Management

    TOOlbOX - 5.1 Output Format Range: 5.1 (=Off or Thru), LCRS, Stereo or Mono Release Selects the Output format in which your five main channels Range: 10 to 1000 ms Input material will be mixed down to. Sets the Release time for the selected Limiter. 90º Mono 90 degrees mono Insert. This option is placed just before Bass Page the two Limiters, meaning at LFr + RFr when Output format is set to Mono, and LSr + RSr channels when LCRS is...
  • Page 70 TOOlbOX - 5.1 Main Channels To LFE/ To prevent this from happening at any time the “Disable Bypass function is an excellent option. LFE To Main Channels The Disable Bypass function is stored at preset level. Depending on the selected Bass Management Mode, Mute Distribute or Extract, the Last section on the Bass page Overall mute function of all Outputs. will appear as: “Main Channels to LFE” or “LFE to Main Channels”.

  • Page 71: Output Level

    TOOlbOX - 5.1 Calibration Page Utilizing Thru Channels 7 and 8 Example 1 In this example we have the VSS™3 algorithm loaded in Engine 1 and we want to monitor the Output on both digital and analog Outputs. The Outputs of the VSS™3 is routed both to the AES/EBU 1 L/R and via the Toolbox Thru channels to the ADA A cards Left and Right Output. Test signal generator (Oscillator) Toolbox-5.1 integrates a comprehensive test-signal generator meant for aligning the monitor system.

  • Page 72: Spacepan 5.1

    sPACEPAn 5.1 The SpacePan 5.1 algorithm occupies: Bleed @ Normal Sample Rate : 1/4 DSP Resource Range: Off to 0dB @ Double Sample Rate : 1/4 DSP Resource Sets the minimum difference of level to channels enhanced via the Focus parameter. If set to “100%” the Focus Algorithm Inputs/Outputs are distributed as follows: parameter will have no effect on the signal. If set to “Off “...

  • Page 73: Engage

    EnGAGE Main Page Introduction 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” (5.0-mix) in the headphones, and delivers optimum surround results for in-flight entertainment, web, games, or headphone optimized tracks for MiniDisc and CD.
  • Page 74 EnGAGE Output Page Out Level Left Range: Off to 0dB Output level control of the Left channel. Right Range: Off to 0dB Output level control of the Right channel. Limiter Brickwall type Limiter. LED on each Output meter indicates when limiter is active. Threshold Range: -12dB to 0dB Sets the Threshold point of the Limiter.

  • Page 75: Unwrap Hd

    unwrAP HD The UnWrap HD algorithm occupies: UnWrap™ and UnWrap™ HD @ Normal Sample Rate : 1/4 DSP Resource The original algorithm was called UnWrap, while the latest @ Double Sample Rate : 2/4 DSP Resource version with more dialog and envelopment extraction functions is named UnWrap HD. Presets generated with Algorithm Inputs/Outputs are distributed as follows: the original UnWrap application can be read using UnWrap HD, and still sound precisely like when they were created.
  • Page 76 unwrAP HD Basic conversion options Main Page Some conversion options are of a profound nature, e.g. if Input trims are provided to carefully match the L/R balance. the process is thought of as purely additive, if the ability to If working from analog tape, adjust balance with a 1kHz collapse to mono or stereo again is important, or if an LFE calibration tone. If working from a digital master with stereo channel or Center channel should be derived at all.

  • Page 77: Input Trim

    unwrAP HD Main Note that styles “Rotate” and “Dry” disable the Focus parameter. Also note that the Rotate style deliberately plays phase tricks with the surround signal, so it’s probably best used for sound effects purposes. Tuning of the surround parameters is an iterative process, and should include the Delay settings as well. Presets The original UnWrap music upconversion presets can be found in Factory Bank F11-3. New UnWrap HD presets for Music and Sound Effects are located in Factory Bank F11-4. These presets show new degrees of foreground/background discrimination available in the HD version.
  • Page 78 unwrAP HD Center Contour Style Type Select between different styles as processing for the Press and use Faders 1-4 to set BW value for each of the Center channel output. 4 EQ bands. Parameter range: Off and a selection of styles. Range for the Notch filter: Center Contour Threshold Lo BW : 0.02oct to 1oct Sets the Threshold point for the Contour Style to be Mid1 BW : 0.02oct to 1oct...

  • Page 79: Output Delay

    unwrAP HD Decorrelate Tone Range for the Parametric filter: Range: +/- 40 steps. Lo BW : 0.1oct to 4oct Adjust the tone (color) of the decorrelated part of the sound Mid1 BW : 0.1oct to 4oct on the surround outputs. Mid2 BW : 0.1oct to 4oct Hi BW : 0.1oct to 4oct Range for the Shelve filter: The EQ for the Center channel features four-band Lo BW : 3dB/oct to 12dB/oct...

  • Page 80: Output Levels

    unwrAP HD Output Outputs Mute Range: Muted/Unmuted Sets the Mute-status on the Output for each of the 6 channels. Solo When a Solo button is selected, the Outputs of all the five remaining channels will be set to Off, but they can be selected as additional solo channels.

  • Page 81: De-Esser

    DE-EssEr The De-Ess algorithm occupies: uses to reach the gain reduction specified by the Ratio @ Normal Sample Rate : 1/4 DSP Resource parameter. @ Double Sample Rate : 1/4 DSP Resource Release Algorithm Inputs/Outputs are distributed as follows: Release sets the fall-back time of the DeEsser, after the signal drops below the Threshold.
  • Page 82 DE-EssEr Link The Link parameter switches the DeEsser between Mono and Stereo operation. When activated one common sidechain will be used internally (applying the same DeEssing rate to both channels), and the parameters on the Main page will be linked. When activated the DeEsser will operate internally in Mid and Side mode. An M/S encoder is placed on the Input and an M/S decoder is placed on the output. This way it is possible to input a L/R stereo signal, apply deessing to the mono and side information separately, and output a new L/R stereo signal.
  • Page 83 EQ - 5.1 Levels Page The EQ 5.1 algorithm occupies: @ Normal Sample Rate : 1/4 DSP Resource @ Double Sample Rate : 1/4 DSP Resource Algorithm Inputs/Outputs are distributed as follows: INPUT OUTPUT • • • • • • E1 - E4 •...
  • Page 84 EQ - 5.1 Type Selector Cut Filter - Bessel type • Press Type and use faders 1-4 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. Parametric Filter - Broad type Cut Filter - Butterworth type Shelving Filter Freq Press Freq and use Faders 1 to 4 to adjust frequence for...
  • Page 85 EQ - 5.1 Type Press and use Faders 1-4 to set BW value for each of the 4 EQ bands. Range for the Notch filter: Lo BW : 0.02oct to 1oct Mid1 BW : 0.02oct to 1oct Mid2 BW : 0.02oct to 1oct Hi BW : 0.02oct to 1oct Range for the Parametric filter: Lo BW : 0.1oct to 4oct Mid1 BW...

  • Page 86: Massenburg Eq

    MAssEnburG EQ The MDW Hi Res EQ 2 channel algorithm occupies: Sample Rate @ Normal Sample Rate : 1/4 DSP Resource If the System rate is set at 44.1 or 48kHz, the signal is @ Double Sample Rate : 1/4 DSP Resource up-sampled to double rate, processed and down-sampled to System rate. This approach is one way of mimicking Algorithm Inputs/Outputs are distributed as follows: more precisely the behavior of analog EQs plus reducing alias distortion.
  • Page 87 MAssEnburG EQ Operating the EQ 6-ch. version In the 6-ch. version a Channel select parameter is placed Most of the operation of the 2-ch. and 6-ch. versions below the Output meters. This parameter is always are identical. In the following it will be indicated which present.
  • Page 88 MAssEnburG EQ Copying of channel-settings are performed instantly when linking, therefore we recommend that careful consideration is done when changing Link groups. Also we recommend de-activating the Enable Linking when not in use. Input/Output Gains For all channels an Input gain and Output gain parameter is available. Parameter range: 0dB to -120dB (Off). 2-ch. version The Gain parameters are located at the Main page. When Link Absolute the Gain settings are included in the linking.

  • Page 89: Vp-2 Stereo

    vP-2 sTErEO Pitch The VP-2 Stereo algorithm occupies : @ Normal Sample Rate : 1/4 DSP Resource @ Double Sample Rate : Not available Algorithm Inputs/Outputs are distributed as follows: INPUT OUTPUT • • • • • • E1 - E4 •...
  • Page 90 vP-2 sTErEO Feedback Levels Pitch Splicing Illustration Feedback L Range: 0 to 100% Determines how much processed signal from the left channel that is fed back to the left Input channel of the algorithm. Feedback R Range: 0 to 100% Determines how much processed signal from the right channel that is fed back to the right Input channel of the algorithm. 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.

  • Page 91: Vp-8

    vP-8 The VP-8 Stereo algorithm occupies : For both 5.1 situations, this assignment may work better in @ Normal Sample Rate : 2/4 DSP Resource special cases: @ Double Sample Rate : Not available Algorithm Inputs/Outputs are distributed as follows: 3 SL 4 SR 5 Center INPUT OUTPUT 6 LFE • •...
  • Page 92 vP-8 Level Input Level Pitch Range: -100dB to 0dB Pitch 1-4 and Pitch 5-8 Sets the Input level for all eight channels. Range: +/- 200cent Dry Output Level Sets the transpose values respectively for the two channel Range: -100dB to 0dB groups. Press the parameter and use fader six, or use the Sets the Dry Output level for all eight channels.

  • Page 93: Chorus - 1

    CHOrus - 1 Chorus This algorithm that derives from the M5000 can produce high resolution Stereo Chorus and Flanging effects, or Speed Modulated Stereo Delays of up to 650 ms per channel, Range: 0.1Hz to 10Hz and offers a modulation oscillator with relative L/R phase Controls the Rate of sweep in a range from 1 sweep every control. For micro-delay sensitive algorithms like flanging, 10 seconds to 10 sweeps every second. note that predictable results are best obtained using a direct feed-through mixed inside the effect, rather than Depth...

  • Page 94: Phaser - 1

    PHAsEr - 1 Phaser This M5000 algorithm was originally designed as a digital version of the acclaimed TC XII stomp box phas- Speed er. As its predecessor, it offers a selectable number of Range: 0.1 to 10Hz swept filter sections (4, 8 or 12) to control the max Controls the Speed of the moving filters in the Phaser.

  • Page 95: Delay - 1

    DElAy - 1 Delay This algorithm deriving from the M5000 is straight, 24 bit true Stereo Delay of up to 2.5 sec in each channel Left Delay with basic feedback control. The System 6000 imple- Range: 0 to 2500ms mentation of Delay 1 can be used with sample rates of Sets the Delay time for the left side.

  • Page 96: Delay - 2

    DElAy - 2 This M5000 algorithm is a 24 bit true stereo effect offering Delays of up to 2.5 sec in each channel with two filtered feedback controls per channel (self feedback and x feedback). Delays may be modulated with control over relative L/R phase for Chorus or Tape Delay emulation.
  • Page 97 DElAy - 2 Levels Modulation Speed Range: 0 to 100% Range: 0.1Hz to 10Hz Sets the mix between dry and wet signal. Controls the rate of modulation sweeps in a range from 1 sweep every 10 seconds to 10 sweeps per second. In Level Depth Range: -100 to 0dB...

  • Page 98: Reflector 6 & Reflector Lcr

    rEflECTOr 6 & rEflECTOr lCr Introduction Main Page Reflector is a versatile Delay and Boundary effect for multichannel applications. Output configurations are optimized for LCR, Stereo or Mono production. Advanced features such as individual EQ and multiple layers of Diffusion, makes Reflector capable of crossing the border between Ambience, Early Reflections, Reverb and Delay-effects. Reflector is able to generate plain delays, realistic ambient locations, and with the long delay-times and diffusion available, it can also simulate the large outdoor locations sought after in film- and post production. Environments...
  • Page 99 rEflECTOr 6 & rEflECTOr lCr Dry - Reflector 6 Output Format - Reflector LCR Select Output format between: Center Mono, LR Mono, LR Stereo, LCR On this page the dry signal is positioned. If you work in a send/return loop without passing dry Output Format - Reflector 6 signal through Reflector (master Dry control turned Select Output format between: ‘5.0’; ‘5.0 minus Center’;...
  • Page 100 rEflECTOr 6 & rEflECTOr lCr Taps - Single Page Level Adjusts the Output level for the tap (reflection). Height of the tap indicates level. Pan - Reflector 6 360-degree panner for positioning of the tap. Pan - Reflector LCR +/-30 degree (from Left to Right) panner for positioning of the tap. Shade The tap color is a combination of the Shade response and Diffusion type.
  • Page 101 rEflECTOr 6 & rEflECTOr lCr Auto Press the Auto Fly button and make the picture rotate automatically (It is still possible to move the graphics even when Auto rotating). This function is not only meant to be a sales tool, but is actually useful to provide an overview of the effect :-) Tip: If you access the Hyper view page from the Single view page, it is possible to adjust the last selected tap...

  • Page 102: Backdrop

    bACkDrOP Main The BackDrop algorithm occupies: @ Normal Sample Rate : 1/4 DSP Resource @ Double Sample Rate : Not available Algorithm Inputs/Outputs are distributed as follows: INPUT OUTPUT • • • • • • E1 - E4 • • •...
  • Page 103 bACkDrOP Listen maximum amount of noise reduction you are trying to The Listen control allows you to select what is being achieve. Therefore, if Max Reduction is set to say 10dB, monitored at BackDrop’s output. By selecting the Removed then BackDrop will not try to lower the noise by more than button, you can listen to the noise that is being removed 10dB.
  • Page 104 bACkDrOP Link bands. The Link button links or gangs the controls so that they MultiType Model operate on both the left and right channels together. When not linked, separate controls are available for the Left and Right channels. When enabling Link mode, settings for the Right channel will be copied to the Left channel. A pop-up display will ask you to confirm this operation. It should be noted that the noise reduction processing remains inde-pendent for each channel even if the controls are linked.
  • Page 105 bACkDrOP you can selectively reduce each of these three components sources having a flat spectrum include tape hiss and while minimizing the processing applied to the signal you’re quantization noise. cleaning up. The way in which BackDrop decomposes the noise into The Max Reduction, Basilar Dispersion, Transient Recovery, and Link Controls function exactly as in the three components is best described in the frequency domain as illustrated in above figures.

  • Page 106: Backdrop - Tutorial

    bACkDrOP - TuTOrIAl BackDrop does this using the current Trim Start and Trim of the controls. To hear the unprocessed capture buffer, End settings. simply press Bypass. Play Go to the Model screen and select the Basic algorithm The Play button starts and stops playback of the captured using the Algorithm Model control. In order to get an audio buffer. The audio buffer plays repeatedly until Play is initial feel for BackDrop’s controls we will use some turned off.

  • Page 107: Advanced Use

    bACkDrOP - TuTOrIAl controls and to listen to the amount of noise reduction, reduction that you desire. Therefore, slowly lower the the occurrence of musical noise, and the smearing of amount of Aggression until you here the level of the transients (signal distortion).
  • Page 108 bACkDrOP - TuTOrIAl preserved as you switch, allowing you to compare your 5) Do I always have to use Play, Trim Start/End and best settings for each. Calculate to get a good loop? It is also a good idea to periodically use the Removed No. This is usually required when working with a tape button to hear what is being removed from the input signal. transport or when taking a print of a live room where it may If you hear components of the signal, then this may be an be hard to anticipate the material.

  • Page 109: 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 co-exist with new recordings, TV commercials don’t fit drama, classical music or film and broadcast doesn’t match. The most fundamental audio issue of all – control of loudness – every day makes millions of people adjust the volume control over and over again. LM6 is part of a universal and ITU standardized loudness control concept, whereby audio may easily and consistently be measured and controlled at various stages of production and distribution. LM6 works coherently together with other TC equipment, or with equipment of other brands adhering to the same global standard. Follow the guidelines given to allow audio produced for different purposes to be mixed, without low dynamic range material such as commercials or pop CD’s always emerging the loudest. - Loudness meter fully compliant with EBU R128 - Loudness meter fully compliant with ATSC A/85 - Loudness meter fully compliant with ITU-R BS.1770 - Loudness meter fully compliant with ITU-R BS.1770-2 - Radar meter showing Momentary and Short-term loudness - True-peak bargraph meters - Advanced Logging functionality...
  • Page 110 Introduction Since 1998, TC has performed listening tests and evaluation of loudness models; and therefore holds an extensive, Universal Database of loudness, based on ten thousands of assessments. The database covers all sorts of broadcast material, music, commercials, feature film and experimental sounds, and is verfied against other independent studies. Fig 1. Left: DRT for consumers under different listening situations Right: Peak level normalization means that material targeted low dynamic range platforms gets loud. The Universal Database is authoritative from an LM6 offers a standardized option: The visualization of academic as well as a practical point of view.
  • Page 111 Fig 2 Color coding and target loudness for selected broadcast platforms based on a consumer’s Dynamic Range Tolerance, DRT. The aim is to center dynamic range restriction around average loudness, in this case the –20 dB line, thereby automatically avoiding to wash out differences between foreground and background elements of a mix. Note how different the broadcast requirements are from those of Cinema.

  • Page 112: Basic Use

    Basic Use LM6 makes use of a unique way of visualizing short-term loudness, loudness history, and long-term statistical descriptors. It may be used with mono, stereo and 5.1 material for any type of program material. Press the Radar key to bring up the Radar page. This page will be used most of the time. The basic functionality of the Radar page is shown in Fig 3. Fig 3 - Radar page features of LM6 in System6000 The numbers associated with the outer ring may be Target Loudness is displayed at 12 o’clock of the outer referenced at either maximum loudness, or have a zero ring, and at the bold circle of the radar indicated also by the point set set at Target Level. Choose “LUFS” or “LU” at the transition from green to yellow.
  • Page 113 Long-term measurements Universal Descriptors and AC3 Metadata The “Dialnorm” parameter in AC3 metadata should indicate Universal descriptors may be used to make the average loudness of a program. Basic dynamic range program-duration measurements, or you may “spot-check” and level control that rely on this parameter may take place regular dialog or individual scenes as required. It is in the consumer’s receiver. Therefore, its value should not recommended not to measure programs of a shorter be far off target, or the consumer results become highly duration than appoximately 10 seconds, while the unpredictable.
  • Page 114 Main page program and stays the same downstream of production if no dynamics processing has been applied. You may even check the number out of a consumer’s set-top box to verify that distribution processing and Dolby DRC has been disabled. Like with Program Loudness and Loudness Max, the meter should be reset before measuring LRA. Prog. Loudn. Program Loudness returns one loudness number for an entire program, film or music track. Its unit is LUFS. Some vendors and countries use the unit “LKFS” or “LUFS”, but all three are the same: An absolute measure of loudness...
  • Page 115 LUFS/LU Note 1: Because the Sliding Loudness measurement is This setting is similar to the ‘LUFS’ setting, except that the completely un-gated, it may also be used to spot check Radar display uses an LU scale rather than an LUFS scale, sections of a program complying to “raw” ITU-R BS.1770 on the Icon. There is no difference between the LUFS and and the first revision of ATSC A/85. LU/LUFS settings, when the LM6 is used in stand-alone mode. Note 2: LM6 makes use of optimized statistics processing in order to display a sliding loudness value (a prognosis) as quickly as possible after a reset. In this setting, measurements of program loudness and sliding loudness are shown in units of LU, that is, Loudness Max in Loudness Units on a relative scale. The 0 LU is by Loudness Max displays the maximum loudness registered definition the target loudness level, such as -23.0 LUFS. So...
  • Page 116 Radar Speed United States Page 11 of ATSC A/85 (May 25, 2011) references ITU-R Radar Speed controls how long time each radar revolution BS.1770-1, even though BS.1770-2 was in effect at takes. Select from 1 minute to 24 hours. You may “zoom” that time. The same page also says that “All referenced between the settings, as long as the history isn’t reset. documents are subject to revision”. The wording is Pressing the Reset key resets the meter and descriptor ambiguous and it’s up to the reader to decide whether or history. not a relative gate (the difference between BS.1770-1 and BS.1770-2) is applied when measuring Program Loudness. Radar Resolution The “Leq(K)” setting in LM2 disables the relative gate, Radar Resolution sets the difference in loudness between...
  • Page 117 The Stats page gives an overview of essential descriptors. Note! The Reset button resets the meters and the log file.
  • Page 118 Level versus Loudness and correlate well with human test panels. It therefore seems justified to use Leq(RLB) as a baseline measure for When level normalization in audio distribution is based on a loudness, especially because room for improvement is also peak level measure, it favors low dynamic range signatures built into the standard. The final BS.1770 standard included as shown in Fig 1. This is what has happened to CD.
  • Page 119 Display based level jumps are currently creating havoc in digital LM6 may use either the measurement unit of LU (Loudness TV, and LM6 helps correct that situation. The LM6 Units) or LUFS (Loudness Units Full Scale). LU and LUFS Loudness Meter can be used to control level and improve are measurements in dB, reflecting the estimated gain sound, not only in Dolby AC3 based transmissions, but offset to arrive at a certain Reference Loudness (LU) or also on other broadcast platforms, such as analog TV, Maximum Loudness (LUFS) as defined in BS.1770. Since...

  • Page 120: Matrix 88

    MATrIX 88 Features Limiter • 8x8 48 bit mixer algorithm Threshold • Precision delay on all inputs and outputs Range: -12dB to Off - in 0.1 dB increments • 2 stereo AUX sends with pre/post Brickwall limiter for all channels. Threshold is always • 2 main mix outputs relative to 0dBFS. LED on each Output meter indicates • Smooth crossfading between channels when Limiter is active. • Brickwall II Limiter • Part of the routing matrix Profile Range: Dynamic, Soft, Universal, Loud, Voice Select one of the 5 standard profiles that fist your source Matrix 88 occupies: material best possible.

  • Page 121: Aux Master

    MATrIX 88 Mix page when the control is moved away from center position. This approach could be called “constant transparency” rather than, i.e. “constant power”. Because of bit-pattern transparency, you may pass and gain adjust even dithered master tracks without introducing dither again. If the limiter kicks in, dither will momentarily be upset, but only at a point in time where it makes no difference anyway, namely just after a peak has occurred. In addition to perfect digital summing, Matrix 88 is also the ultimate A/B/C stereo switch. Inputs may be aligned precisely in level and time, and switching between them may have well defined cross fades.

  • Page 122: Main Outputs

    MATrIX 88 Main Outputs Delay Main A/B Range: 0-333 ms / 0-14700 smp / 0-111 m / 0-373 feet 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.

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