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DSC28 User's Manual english v2

1. 0 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 7 Equalised system 3 examples of total delay response dB i LL 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 2 Target functions system weighted and band passes ms SUB LOW HIGH 5 MID o 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 4 Controller intrinsic response run time f AN VET i HAIN MAN 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 6 Equalised system total frequency response single frequency responses In a minimum phase system the run time increases strongly to low frequencies which is also the case in analog controllers If a correction of the run time to linear phase behaviour is required the delays of the higher frequencies have to be increased to that of the longest run time at the lowest frequency The result is a total signal delay and for live situations unacceptable A compromise correction allows linear phase behaviour to the lower frequencies which converts to minimum phase behaviour below a cut off frequency of 100Hz Chapter 14 2 Chapter 15 Limiter All components of a signal pr
2. 36 dB octave a minimum phase response up to 120 Hz and a linear phase response gt 120Hz a controller delay of 30ms has to be considered intrin sic controller delay of approximately 7ms filter delay of approximately 23ms Due to its configuration with FIR filter technology the use of the GAE DSC28 as a user programmable cross over allowing subsequent simulation of analog IIR filters is not intended Equaliser 11 Chap 8 Situated prior to the frequency crossover the user has 14 bands of full parametric EQ for each channel at his disposal These are of the analog IIR filter type adjustable at the user interface The following filter types can be selected Bell Low Shelving High Shelving Low Pass and High Pass Summary The DSC28 is a non compromising system controller concept operated via a standardised user interface allowing a choice of presets and the control of user parameters These presets are pre defined and made available as relevant operation parameters by the loudspeaker system manufacturer As such the DSC28 is ideally suited for OEM application Designed for use with selected GAE applications the DSC28 can on request be programmed to provide the necessary parameters for the driving of all types and makes of loudspeaker systems The parametric equaliser completely configurable by the user substitutes the need for an additional 19 device and facilitates the uncomplicated minor modifications necessary when adjusting
3. 96dBu 21 Bit 96dB Together with the function output scaling and the parallel switching of a stereo DA converter per path the matching to the amplifying factor of the connected power amplifiers provide to date January 2001 an unsurpassed excellent dynamic value and moreover a constant high resolution of the digital signal flow at the DA converter inputs of the controller Due to the step adjustment of the output range parameter in three 6dB steps a mis adjustment is still possible but only to a maximum of 6dB Exchanging an amplifier When exchanging an amplifier as well as ensuring identical power specifications it is important that amplification factors are also equivalent otherwise an adjustment of the two gain factors output scaling and output range is inevitable The value for output scaling is part of the coefficient set of the XEQ filters and as such is an integral component of the preset The output range value is set by a four position jumper within the analog area of the output circuitry Further more it must be considered that the limiter threshold values also components of the preset have to be in an exact defined relationship to both output values and that by this balance the output bargraph chains represent the exact driving state of the whole PA system If this balance is disturbed by the replacement of one or more ampli fiers with different power specifications and or amplifying factors in several paths of the PA system a
4. and output can additionally be used as a digital insert The device to be linked should be capable of 24 Bit processing otherwise a lamentable digression of the Dual Range Converter s prime values will be the result Parametric Equaliser PEQ The following 2 X 14 band parametric equaliser requires an additional Motorola DSP56009 81 In order to compensate boosting filters the function EQ Gain is integrated to reduce the input signal level An additional limiter step at the PEQ s output prevents digital overflow and a noise shaper attenuates quantization faults of this full 48 bit operating EQ when re quantisising to 24 bit Down sampling Of the following four signal paths only the designated high frequency path requires the maximum bandwidth for data processing allowing the sampling rate of the remaining three paths to be reduced by down sampling by the factors 4 MID and 16 SUB and LOW Keeping the number of filter coefficients constant the reduction of the sampling rate enables the increase of the length of the filter by the same factor and at the same time dramatically reduces the amount of computing power necessary Down sampling is carried out by means of 2 FIR low pass filters with 64 taps After low pass filtering the signal with the reduced sampling rate can be subjected to further processing For this application the filter Chapter 11 1 characteristic has been contrived for maximum rejection band damping gt 120dB so as to
5. digital device even if within the examined audio frequency range no signal modulation is performed i e the signal is simply AD converted passes through the DSP and is finally DA converted The minimum run time in the chain of AD converter no dual range principle DSP chip and DA converter required by all digital controllers is approx 1 5ms Of this time 98 5 is caused by the work of the two converters Further digital functions within the DSC28 e g down and over sampler the realisation of the dual range principle and the limiters require additional run times Efforts to reach better signal quality within the digital domain have to be offset against the ensuing delay times The intention is to optimise theses times in order to keep them as short as possible The run time of a digital controller does not necessarily need to have a constant value and the sum of the run times in parts of the DSC28 s signal flow is frequency dependent Subsequently the delay causing modules of the DSC28 and their basic run time are divided into 3 interdependent groups 0 01 0 1 1 10 kHz Fig 13 2 Basic run time behaviour Curve 1 Linear phase All passing frequencies sustain the same delay time A digital delay device shows such a behaviour Curve 2 Minimum phase The run time is dependent on frequency it runs towards zero above the frequency th
6. loop Several controllers can be linked in a chain or a loop via the MIDI in and outputs to enable e g the adjust ment of all the connected devices from a single device Within such a chain the first device may also be a PC Perform the necessary adjustment within the menu SYS MIDI Settings The list below is a summary of the MIDI Settings LI Chap 9 1 amp SYS 1 MIDI Settings 1 Channel 1 ON 16 Assign the controller with ist transmit receive channel basic channels 1 16 with which data can sent and received Enter Confirm your selection with ENTER 2 Out OutOnly OM Out Thr OM Loop OutOnly is a pure MIDI output Out Thr is a MIDI output with an additional Soft Thru function which passes the signals received at the MIDI input to the MIDI output Loop is a MIDI output which passes on all incoming data on channels other than the devices addressed channel This mode links several DSC28 controllers in a closed MIDI loop 3 Baudrate MIDI 31250 O RS 232 9600 Use RS 232 and the supplied cable to remote control the first device with the help of a PC via one of it s serial interfaces COM Note that the RS 232 connection cannot bridge long distances Use MIDI to link several DSC28 controllers Enter Confirm your selection with ENTER 4 ParaChg Off OM On If this value is set to On the device sends all performed parameter adjustments out As such several device Slaves can be commonly operated by on
7. write memory each hold the operating systems and parameters of the controller Because the data safety of an ROM is essentially higher than that of a RAM this two memory system ensures that even in the case of a data loss in the RAM the full functionality of the controller can quickly be restored A maximum of two presets as well as the first parameters are always available from the ROM The lithium backup battery for the RAM has a manufacturer guaranteed minimum life expectancy of 10 years and as well as this is over dimensioned in capacity The following diagram lists the memory structure First Parameters The first parameters copied from the EPROM uC Program converted to user parameters 8 12 presets can be down loaded DSP Program via the RS232 interface Initialise In case of a data loss in the RAM parameters adjusted by the user as well as possible down loaded presets are lost Defect parameters are overwritten with the first parameters in the initialisation proc ess lost presets have to be re loaded Should the scheduled self initialisation of the operating system not be performed indicated by an error message and the command Press Enter one of the following initialisation possibilities is provided If the initialisation is not successful the reason is a different or further failure In this case the device should be returned to the manufacturer for maintenance 1 Follow the directions in Chap 9 Menus under SHIFT SYS
8. 20Q maximum output level 18dBu into 375Q noisefloor lt 96dBu dynamic range gt 114dB all measures linear rated 22Hz 22kHz totally THD N lt 0 005 at operating limit output range steps 18 12 and 6dBu maximum output level per output a 24 Bit Stereo Delta Sigma converter in parallel mode input 24 Bit AES EBU with without pre emphasis sample rate converter 32kHz 96kHz output 24 Bit AES EBU sample rate 44 1kHz also useable as digital insert AD gt DigOut DigIn gt Controller input RS 232 and MIDI output MIDI baud rate 9600 and 31250 incl down over sampling AD DA converter and limiter 5 7ms path dependant FIR filter group delay dependant on chosen presets IIR filter quasi analog behaviour peak limiter with 1 5ms pre view and controlled overshoot to fully utilise the PA impulse reserves 48 Bit signal processing precision adjustment and extremely low distortion rms limiter with voice coil and magnet temperature modelling foil keyboard with eight tip keys incremental dial with additional ENTER function 2 lines 24 character vacuum fluorescent display blue 2 X input 30 24dBu 2 X clip gt 28dBu each 10 X LED green each 1 X LED red 8 X output 30 OdBFS 0 12dBGR 8 X mute each 7 X LED green 3 X red each 1 X LED rot universal input 85 265V lt 30VA over voltage protection safety fuse M1A M medium switched power supply is self protected 19 1U 260m
9. 28dBu an adjustment setting of about 30dB for 4dBu systems is normal SET 0 Input Gain Balance Input Gain Further menus SET 1 5 A SET 6 e At this stage you may possibly want to adjust the bass level Change into the menu Output Gain for the output paths 6 times DOWN key and press the RIGHT key to choose the required path Sub and or Low SET 2 Output Gain Further menus V SET 3 5 A SET 1 0 6 e If your system is correctly connected EI Chap 16 you can now perform your first event Further system settings should only be carried out after reading the respective points of the User s Guide Chapter 3 1 Chapter 4 LED Display FE CLIP R 1 MUTE 2 CH A E3 MUTE 4 11 MUTE 2 CH B BIMUTE 24 STATUS 24 0 dBGR 0 0 dBGR 0 ANALOG 18 6 I 6 6 l 6 DIGITAL INPUT 12 12 12 12 12 LOCKED 6 3 3 3 3 44 1 kHz Fs 0 6 6 6 6 48 0 kHz 6 9 9 9 9 MIDI DATA 12 12 12 12 12 SAFE OS 18 18 18 18 18 OEM KEY 24 24 24 24 24 OWNER 30 30 dBFS 30 30 dBFS 30 USER INPUT dBu LEVEL CHANNEL A LEVEL CHANNEL B Fig 4 1 LED Display INPUT L R The actual input level to the analog inputs is displayed by the input bargraphs in dBu OdBu 0 775VRMS A corresponding audio signal drives the digital input to the full without considering the word width of the source and as such is incapable of being overdriven
10. Re loading of presets updating the operating system initialise rrrrrrrrrnnrrrrronnrrrrrrnnvrrrnrrnnnn Ch Activating the remote on off switching ersnsennseennnnnnennnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnennnnnnnnnnnn Ch Parameter control of several devices ccccceceeeeeeeeeeeeeeeeceaeeeeaaeceeeeesaeeesaaeeeeneeeseaeeesaeeseneeenaees Ch Remote Conobea AA E E E Ch TECnNnMGal datauususosdasp EE E EAE E Ea Ch Adresses bibliography MisCellanGOUS 0 0 0 cc ceeeeeeeceeeenneeeeeeneeeeeeaaeeeeeeaaeeeeeeaaeeeeeeaaeeeeeeaeeeeeenaaes Ch 18 19 20 21 22 Chapter 1 Reference to EC statement of conformity This document confirms that the product GAE Digital PA Master DSC28 bearing the CE label meets all requirements in the EMC directive 89 336 EEC laid down by the Member States Counsel for adjustment of legal requirements Further more the product complies to the rules and regulations of the electromagnetic compatibility of devices from 30 August 1995 This product bearing the CE label complies with the following harmonised or national standards EN 55022 11 14 EN 61000 3 2 EN 61000 3 3 EN61000 4 2 EN 61000 4 3 EN61000 4 4 EN61000 4 5 EN 61000 4 6 EN 61000 4 11 The authorised declaration and compatibility certification lies with the manufacturer and can be viewed on request Responsible as manufacturer is opal audio vertrieb GmbH EngerstraBe 47 D 33824 Werther Tel 05203 236 237 Fax 2
11. System adjustments SYS 4 Initialise Should the data defect disturb the operation of the device proceed as in 2 2 Remove the device from the mains supply press the ENTER key now reconnect the device to the mains and only release the ENTER key if the boot display is substituted by the Input Gain menu SET 0 This procedure does not last longer than 4 seconds Updating the operating system An update of the DSC28 operating software can be loaded to the Flash ROM Due to data operating safety of GAE systems this software includes at least one preset so that in the event of an eventual RAM data loss the continued operation of a connected GAE system can be maintained after initialising the device Due to the fact that each preset contains information about the users power amplifiers the complete operating system is dependant on the amplifier configuration Re loading of presets A further 8 12 presets can be loaded to the battery puffered RAM The exact number depends on the size of the blocks After a successful transmission the transfered presets are avail able in the Preset List and can be recalled via the menu SET 7 EHI Chap 7 3 Immediately after initiation and during the whole procedure of preset block transmission the message Appending Module is shown in the top line of the controller s VFD display If transmission fails check that the RS232 interface of the DSC28 is active LJ Chap 9 1 amp SYS 1 1 MIDI Settings Baud rate T
12. a loudspeaker system clearly sets it apart from other loudspeaker management devices available on the market By utilising the DSC28 elementary sys tem parameters e g of a mid high unit can be decisively influenced during early development stages The user accessible presets represent as such the outcome of intensive development and determine together with the loudspeaker components and the power amplifiers the tonal and power handling behaviour of the sound system as a combined single unit even in its different configurations This product philosophy was already successfully realised by GAE in their drive devices e g system control ler BF1 programmable via interchangeable preset cards System relevant functions are stored as recallable parameter sets and important user definable variables are freely available As such even in the digital age of signal processing the totally user programmable cross over as a central driving device for GAE loudspeaker systems remains a second choice commodity Function The GAE DSC28 combines the functions of cross over equaliser delay and limiter in a 1 rack space enclosure It has been conceived as a remote controllable RS232 MIDI 2 channel control unit for up to 4 way high power sound system applications with two analog inputs a digital in and output AES EBU which can also be used as a digital insert and four analog outputs 1 SUB 2 LOW 3 MID 4 HIGH per channel A 2x14 band fu
13. devices A VCA Voltage Controlled Amplifier controls the adjustment of the level reduction The necessary control voltage for the VCA is determined by comparing the default threshold with the rectified output voltage If the output signal exceeds the threshold the current increase at the comparator output is passed to the VCA control input by means of an RC circuit whose time constant defines the attack time T1 Should the threshold exceeding voltage no longer be present the amplification is released back to the output value depending on the release time constant T3 Before this the level reduction determined by the attack is held for the period of the hold time T2 The hold function is missing in most of the commonly used analog limiter systems Fig 15 1 Analog peak limiter Problematic is that the comparator s erratic output voltage smoothed only by a low pass of 1 order corresponds directly to the VCA control voltage The only weakly attenuated high frequency portions of the control signal are directly multiplied by the wanted signal and as such produce additional distortion which leads to an audible crackle when using critical source material with little high frequency content Low pass filters of a higher order would cause too long a run time delay and therefore a limiter response which is too Chapter 15 1 slow Even the low pass of only 1 order has the effect that directly after an amplitude jump the limiter is not able
14. however where possible their employment is discouraged A complete equaliser crossover network adjusted to the working range of the connected PA system has already been configured within the employed presets via XEQ Bell type filters are used to compensate for tonal discoloration caused by room architecture and acoustics Firstly we recommend however especially compliant to the bass range the balancing of the individual outputs by means of the output gain controls Attention should be made to a restrictive particularity which appears when using shelving filters When a high shelving filter adjusted with 12dB octave limit frequency of 5kHz and a Q factor of 0 7 is set the user interface accepts no gain setting higher than 5dB A higher value can only be obtained by reduction of the Q factor or by increasing the threshold frequency These restrictions are set by the device as the calcula tion of filter coefficients for a higher boost level would exceed the DSPs Digital Signal Processors value range and as such are not available XEQ 1 Chap 14 The influence of the XEQ filter on the gain structure is characterised by two features Firstly the differences of efficiency of a PA system s individual sound transducers which leads to a relative scaling of the filters while authoring the coefficients Secondly the highest appearing peak of the ampli Chapter 12 1 tude frequency response of a part filter path requires the further gener
15. of the HIGH path Only modules wholly or partly relevant to gain are considered The at 24 bit indicates reduction from 48 bit to 24 bit by means of a noise shaper with 1 Order Error Feedback Dual range AD The introduction of a dual range AD converter enables the input dynamic to be increa sed by approximately 17dB The basic idea of the dual range AD is that two separate AD converters re produce the input signal simultaneously with different pre amplification In the case of an overloading of the amplified channel a subsequent intelligent switch within the digital range switches to the less sensitive channel To guarantee a high signal processing quality the characteristic differences of the two identical but not ideal converter modules against the set point deviating results of the analog amplifiers with diffe rent amplification factors as well as the DC offsets and general long term stability have to be taken into consideration This takes place during an additional adjustment of these parameters Introduced to the analog input adjustment of the converters is a pre emphasis filter causing a treble boost with a standardi sed frequency response denoted by the two time constants 50us and 15us This pre distortion is reversed by a de emphasis filter during the digital signal processing This treble boost affects the converters set noise which was not increased by the pre emphasis filter Comparing the noise level of a white noise si
16. prevent disturbances due to aliasing effects Delays A master delay for the adjustment of the entire system delay is situated prior to the crossover network whilst a further delay stage is situated in each of the output paths providing time align compensation between the individual speaker components and forming part of the correcting filter network The basic delay of the DSC28 is approx 5ms effected by A D D A conversion down over sampling and the pre viewing limiter concept All further delays originate from the signal processing within filters and depend particularly on the run time behaviour of the loudspeaker system filter slopes and frequency limits The controller s entire basic delay time for the loaded preset is displayed in the Master Delay menu Crossover network and equaliser XEQ Next on the signal path are the actual crossover networks and equalisers for the individual paths SUB LOW MID HIGH A processing power of 80 Mips made available to the controller s XEQ filters by two Motorola DSP56009 81 processors for both channels is intelligently distributed between the individual paths Considering the respective sampling rate the resulting values are listed in the table below no sampling filter Filter frequency limit approx resolution Factor taps Taps eff kHz kHz Hz Path Down Length of Length of Sample Frequency Frequency name Over sampler After XEQ filtering the sampling rate can be returned
17. to the loudspeakers the amplifier supply current inevitably sinks The amplifier design determines how much and at which rate the supply voltage drops when full output power is suddenly demanded Here a compromise between highest available impulse power and continuous output stability is necessary Considering the signal statistics of common program material which an amplifier has to deal with it seems illogical to construct the power supply as hard as possible i e with infinitely small internal resistance as this leads to an unnecessary high power loss to the amplifier Very helpful during the analysis of an amplifiers signal statistic is the so called crest factor Usually stated in dB the crest factor relates the peak power of a signal to its average performance A pure square wave modulation has therefore a crest factor of OdB whilst a sine wave signal amounts to 3dB With uncompressed wide band music and speech signals it lies in excess of 10dB After the signal separation to different frequency ranges and following strong compression by the output limiters in active multi path systems the smallest crest factors can be observed within the bass range The amplifiers for this frequency range must not only provide the highest output power but must also have the hardest power supply i e with minimum internal resistance compared to the amplifiers for the other paths Even with the roughest compression to unnatural signals of non musical natur
18. 0 Taps 64 Taps 64 Taps Limiter Limiter Output Gain High Scaling 4 HIGH Output Gain Mid Scaling 3 MID Bi Bi Md y FU Noise Yo 2x24 Bi Analog Shaper DA Output OQutput Gain Low Scaling 2 LOW 8 Bit 24 Bit Low pi Noise 2x24 Bi Analog Shaper DA Output Output Gain Sub Scaling 1 SUB Bi B Sub if Noise 7 f2x24 Bit Analog Shaper DA Output Analog input After passing through an analog pre emphasis filter the analog input leads to two different amplifier stages with different gain rates which drive both inputs of the 24 Bit Stereo AD converter Dual range converting In the digital domain of the Signal Processing SP block the DSP switches and adapts both AD channels to complete the dual range AD conversion reproducing the analog input signal in real 24 Bit resolution Following this the De emphasis filter is calculated and the digital signal is available for further processing In this way a dynamic range of 127dB is achieved which is much higher than that available from the newest generation of 24 Bit single converters Digital in output The 24 Bit AES EBU formatted digital output is permanently fed with the AD converted data stream As well as via the analog inputs the audio signals can be entered directly by means of an AES EBU digital input If requested by the digital signal flow s attendant files a De emphasis Filter can be introduced The combination of in
19. 2 necessary for the RS232 connection is not connected 5 pin DIN 1 RS232 Tx 2 GND 3 RS232 Rx 4 MIDI 5 MIDI Output conform to the MIDI convention standard 5 Pol DIN 1 free NC 2 GND 3 free NC 4 MIDI 5 MIDI This socket facilitates the remote on function by means of an ac or dc voltage of between approx 12 24V The remote on function has to be released before this function is available 14 Chap 19 6 Mains fuses cannot prevent an unexpected malfunction of electrical compo nents rather they should protect the user and its environment from damage For this reason never try to substitute the mains fuse by any other than the specified M1A type 1A medium slow behaviour Never try to repair or bypass a blown mains fuse Chapter 16 1 1 BALANCED IN Impedance 20 kOhms BALANCED SOURCE SHIELD l UNBALANCED SOURCE Pins 1 3 bridged at source SHIELD 3 BALANCED OUT IMPEDANCE lt 20 Ohms DSC28 OUTPUT BALANCED SHIELD 4 UNBALANCED OUT DSC28 OUTPUT UNBALANCED Pins 1 3 bridged at input ey E SHIELD Fig 16 1 Connection hints Eventually necesary for the cancellation of ground loops The active output of the DSC28 can be considered as a transformer output Only use 2 core shielded LF cable Chapter 16 2 Chapter 17 Re loading of presets initialise Memory structure A Flash ROM read memory only and a battery buffered RAM read
20. 38 The awarding of the CE label confirms the compliance with legal directives issued for the manufacture and marketing of electronic and electrical devices As such the CE label is not a seal of quality but rather proof that the device bearing the CE label is conform with the electromagnetic compatibility standards laid down in the above named testing regulations Liability and guarantee conditions Declaration of liability opal audio vertrieb GmbH accepts no liability for damage to loudspeakers amplifi ers or other devices that become damaged through the use of the DSC28 This applies to the regular as well as the improper or negligent start up and or installation of the DSC28 Also opal audio vertrieb GmbH accepts no claims in tort even from third parties based on speculations of alleged restricted or absence of function of the DSC28 e g cancellation of events Product guarantee Beyond the framework of the legal requirement opal audio vertrieb GmbH guarantees the DSC28 to be free from defects in material and workmanship for a period of 24 months after date of pur chase As valid evidence for the beginning of the period of guarantee is the date of an official GAE distributor s issued invoice As manufacturer opal audio vertrieb GmbH will replace faulty parts and restore defect modules within the period of guarantee if the defect has appeared under normal operating conditions The evaluation of a guarantee claim is acknowledged after
21. Analog for an analog source Digital for a digital source that transmits with 32kHz 96kHz Sample Frequency Ensure that the clock pulse generator of the DSC28 is operating correctly Status LED Locked must light Enter Confirm your selection by pressing the ENTER key The digital in and output of the device operate to the AES EBU protocol The digital output always supplies the digital form of the input signals with 44 1kHz Sample Rate The digital input is equipped with a sample rate converter and understands sample rates between 32kHz 96kHz SET 10 Limiter Release Er Limiter R Further menus 4 SET 11 A SET 9 7 Limiter Release 10dB s O 60dB s OM 250dB s 1dB s steps The value of 60dB s as the standard value for the limiter release behaviour results from practical experience Decrease increase this value for longer shorter Release times Attack and Hold times are constant EI Chap 15 SET 11 Delay Link uaa Mid Hi Further menus A SET 10 7 Off O Mid Hi ON Low Mid Hi For the operation with the GAE Director System the Delay Link should always be set at Mid Hi This setting ensures that in the Men Output Delay L SET 7 the 2 way active Director Top can only be commonly altered Chapter 7 4 Kapitel 8 Menus under EQ IIR Equaliser To enable the tuning of your system to differing acoustical situations the DSC28 is equipped with a full parametric equaliser Situated prior to the cross ove
22. Cursor Position F 1 6 B2 7 pP3 8 a9 5 10 Chap 6 1 amp 2 page menus 2 5 and 7 10 Phase Invert Sub Low Mid Hi No OM Yes With this controll you can easily invert the phase of an individual output path This can be of assistance in achieving optimal bass reproduction when working with bass systems of different designs and positioning to one another e g Low System Bass Sub zus tzliches Bassystem SET 5 Output Delay Paths Beim FET Hi Further menus A SET 4 0 6 LR 0 0 0 0 0 0 0 0 Cursor Position F 1 6 B2 7 pP3 8 ff 4 9 E 5 10 the display are rounded up down to one decimal Through the use of the output delays run time corrections can be made especially in the bass region when Mid high packs and Bass systems are operated at some distance to each other e g Mid highs with some Bass support flown and an additional centre bass cluster under the centre of the stage Do not attempt to use the output delays for time allignment purposes of loud speaker components in a GAE Mid High system The time allignment correction between the single compo nents within a system is an integral part of the preset and as such already adjusted to its optimum For this reason the parameters of the menu Delay Link L SET 11 should also not be altered Failure to observe this point will result in impairment to the sound character and the dispersion properties of the Mid High sy stem Menus above SET Menu 0 SET 6 0 Setup Further me
23. For this reason the clip displays are only present on the analog inputs The threshold value lies below 1kHz at 2 28dBu 28dBu 19 47Vrwms 27 53V peak and above 10kHz due to pre emphasis at 18dBu LEVEL CHANNEL A B The output bargraphs show green the digital output level before conversion in dBFS FS Full Scale as well as red the limiting action in dBGR GR Gain Reduction The 0dB position has the dual task of signifying full digital level OdBFS and the advent of limiting gt OdBGR a higher digital as well as analog output level is not possible The MUTE displays indicate inactive outputs Due to the level matching of the analog output stages to the amplification value of the connected amplifiers the exact maximum power pre determined by the loudspeaker components can be supplied to the loud speaker system Because of this the output bargraphs are an accurate visual check of the level at which the connected PA system is being driven Attention The power amplifiers should meet or exceed the power requirements of the connected compo nents If the power amplifier is matched without headroom reserve there is a risk that during critical mains voltage fluctuations the necessary DSC28 limiters do not come into effect before the amplifier is overdriven This is the normal situation for amplifiers with insuficient power In this case it is unfortunate that the excel lent tonal qualities of the DSC28 peak limiter compared to
24. LUD E GAE Digital PA Master DSC28 User s Guide SAE User s Guide Digital PA Master DSC28 from Serial No E04 0110 Table of Contents CE conformity liability and guarantee conditions contents of packaging een Ch imponan not esrvuda keen ae LAS Re Rene Ch Initial Start Up sisi aati A eee al A hehe aie A aed ede ts Ch LED display naar ape ger ae Abe Meade Ch Increment dial function keys and VF display arrvrnonnvnrorvrnnnrrnnnnvnnrnvnnnrrrvnnnrnnnnnennrnrrennnrnnennennn Ch Boot display first parameters and menus 40urs40usnnennnnnnennnnnnnnnnnnnnnnnnnnnnnnnsnnnnnnnnnnnnn nn Ch Menus under SET speaker setup uursnsussseennsnnnnnsnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn Ch Menus under EQ IIR EqualiS r cccccccccccccssececssecseseecseeecaeecsseeecaeescsaeeseasecseeecesaeeesessesseeeeesaes Ch Menus under SYS system adjustments rsnnsessssennnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nenn Ch Unit deseription 23 nee Genet Ch Block diagram and component description 44444444nHn ne nnnnnennennnnennennnnnnnnnnnnennnnnnnnnnnn nenn Ch Gain architecture and dynamics ssersnsunnnnennnnnnennnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnann Ch Runtime bEhavIburi i lasta tidene ae ga aent kirke kadett Ch Pr setS unne Ch Emile E E E ane AE nine Ch FlInts Om SONNSEUON Lave Adeang eee jan dd eee Ch
25. Master Delay Further menus SET 2 5 A SET 0 6 me Min Delay ON 1000ms 340m resolution 0 363ms The minimum value corresponds to the intrinsic run time of the controller This delay time is the sum of the different signal processing times required by such system components as the AD and DA converters the filters allocated to the converters down and over sampling filter as well as the pre viewing limiter Added to these are the loudspeaker system dependent delays derived from the system egalisation filter slopes adjusted amplitude and phase frequency response as well as time alignment requirements EJ Chap 13 amp 14 Delay times greater than the preset dependant Min Delay value can be set with the increment dial SET 2 Output Gain Wege Further menus Y SET 3 5 A SET 1 0 6 Cursor Position Bis E27 Esse Bas 5 10 is possible if necessary to balance the individual loudspeaker paths Situated in front of the limiters the Out put Gain controls are not suitable for level adjustment of a power amplifier with a different amplification fac tor than the originally assigned amplifier because the limiter threshold is not affected by this adjustment SET 3 Output Mute Further menus 4 SET 4 5 A SET 2 0 6 Cursor Position 1 6 B 2 7 Ds Bars 5 10 indicate the mute status of each output Chapter 7 1 SET 4 Output Phase Invert Bn Sub Low Mid Hi Further menus 4 SET 5 A SET 3 0 6 LR No No No No
26. R 2 Out OutOnly OM Out Thr OM Loop OutOnly is a pure MIDI output Out Thr is a MIDI output with an additional Soft Thru function which passes the signals received at the MIDI input to the MIDI output Loop is a MIDI output which passes on all incoming data on channels other than the devices addressed channel This mode links several DSC28 controllers in a closed MIDI loop H Chap 19 SYS 1 1 Midi Settings Further menus V SYS 2 M SYS 0 SYS 1 0 gt SYS 1 2 Cursor position 3 Baudrate MIDI 31250 i O RS 232 9600 Use RS 232 and the supplied cable to remote control the device with the help of a PC via one of it s serial interfaces COM or to re load presets or to update the operating system Use MIDI to link several DSC28 controllers by means of a standard MIDI cable EI Chap 19 Enter Confirm your selection with ENTER SYS 1 2 Midi TX Filters Further menus 4 SYS 2 A SYS 0 SYS 1 1 Cursor position Chapter 9 1 B 4 ParaChg Off OM On If this value is set on On the device sends all performed parameter adjustments out As such several device Slaves can be commonly operated by one device Master Enter Confirm your selection with ENTER 5 TX Chn 1 ONN 16 OM OCM The transmitting channel for parameter adjustments can be selected by this parameter As well as channels 1 16 an Omni Channel Mode OCM can be selected Parameter adjustments via this channel are accepted by the conne
27. RAM Further sets can be added via the RS 232 interface The user can choose between these parameter sets in the preset menu Example The following seven illustrations convey an impression of the results of system equalisation using the 3 way GAE DIRECTOR system The chosen target function is merely a superlative possibility of this loudspeaker system The presets of the DSC28 devices that are shipped with GAE DIRECTOR systems include a minimum phase system equalisation to a linear target function up to approximately 1kHz Above 1kHz the delay time behaviour is linear phase The Minimum phase portions which remain only very slight down to 200Hz can be seen in Figure 14 7 Chapter 14 1 dB 1X11 5 110 TI mt 80 70 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 1 Loudspeaker system single measurements of the components sensitivity 1W 1m dg HIGH o MID SUB LOW 10 20 30 40 0 05 02 05 1 2 5 10 kHz Figure 14 3 Controller intrinsic response amplitude dB o RSA 10 4 be VY 20 30 ae 0 05 0 2 0 5 1 2 5 10 kHz Figure 14 5 Controller outputs level ms
28. al scaling of all filters to the thresh old of the digital system Digital Full Scale Limiter Chap 15 An overshoot of the maximum rating of the output signal is detected by the peak limiters which reduce the amplification over a pre determined time with optimally adjusted time constants Over shooting peaks are automatically reduced to the exact value determined by the threshold level This procedure enables the exact adjustment of the limiter thresholds to the absolute maximum rating of the amplifiers and the connected loudspeakers This requires however an exact adjustment of all subsequent modules to these values The reference values for the limiters are the performance specifications of the loudspeaker components and the power amplifiers The RMS limiter protects the loudspeaker components in a similar way referring however to the absolute maximum thermal rating DA server and converter The DA converter transforms the digital signal flow with as little dynamic loss as possible into an equivalent analog signal Unfortunately the DA converters are the weakest ele ments of the signal chain and therefore determine the output noise level For this reason they are inte grated into a digital analog gain structure The function output scaling provides an optimal digital driving of the converters Digital Full Scale It is included within the generation of the XEQ filter coefficients The reference figures for the output scaling are the amp
29. alen Controllers f r Lautsprecher J Kleber Diplomarbeit am ITA RWTH Aachen 1996 Programmierung eines digitalen Equalisers auf der Basis von FIR Filtern Norbert Blissenbach Studienarbeit am ITA RWTH Aachen 1996 Verbesserung der Wiedergabequalit t von Laut sprechern mit Hilfe von Digitalfiltern D Leckschat Dissertation an der RWTH Aachen 1992 Untersuchung zur digitalen Lautsprecherentzer rung mit Hilfe binauraler Me technik P Niggemann Diplomarbeit am ITA RWTH Aachen 1997 Entwicklung und Erprobung einer digitalen pa rametrischen Filterbank R Thaden Diplomarbeit am ITA RWTH Aachen 1997 Digitale Signalverarbeitung f r Lautsprecher S M ller Dissertation am ITA RWTH Aachen 1999 Consulting MF Me system Audio Acoustics Consulting Dr Ing Anselm Goertz In der Linen 21 D 52134 Herzogenrath Fon 02407 565111 Fax 02407 565112 Mobil 0171 9338402 Email Anselm Goertz t online de SAE is a product of opal audio vertrieb GmbH Engerstra e 47 D 33824 Werther Fon 49 0 5203 236 7 Fax 238 Internet http www gae de Email info gae de Chapter 22 1
30. alue of the underlined first parameter becomes the basis for further parameter adjustments Menu areas The three menu areas System settings SYS Speaker set up SET and IIR Equaliser EQ can be selected by the pertinent function keys SYS SET and EQ Approx 4 seconds after power up the menu area SET is automatically activated with SET 0 1 Input Gain Multi page menus are located under SYS and EQ and are explained later in this guide 2 page menus are located under menus SET and EQ and have a common function which can be illus trated by using the mute menu as an example a link function for the simultaneous operation of both identical channels of the controller Channel Left and Channel II Right is adjustable at the most left hand cur sor position 1 LR denotes linked operation mode L gt denotes the adjustment is only valid for Channel I The right arrow indicates that beyond the far right cursor position 5 the second page of the menu 6 10 can subsequently be accessed R denotes the adjustment is only valid for channel Il Here the left arrow indicates that beyond the far left cursor position 6 the first menu page 5 1 can be accessed Mute Sub Low Mid Hi R No No No No E B2 E E ic a7 Bs io Bio Menu page 1 Cursor position L gt 1 The link Menu page 2 Cursor position R J 6 The link function LR can be selected by turning the incre function LR can be selected by turning the incre men
31. an optimally cor rected loudspeaker system to room acoustics The GAE DSC28 concept not only allows for better efficiency in design and realisation of future system de velopments in loudspeaker technology Even today the employment of the DSC28 with its FIR filter technol ogy offers astounding advantages in the driving of existing systems The audio quality of the signal processing determined mainly by the converters but also influenced by the analog circuitry in the signal path is engineered to a maximum with regards to present technical possibili ties and this at an acceptable price performance ratio The GAE DIGITAL PA MASTER DSC28 offers state of the art solutions not only to future sound reinforce ment applications Chapter 10 2 Chapter 11 Block diagram and component description Fig 11 1 The schematic diagram shows the complete signal flow of one channel of the DSC28 Input Gain High Gain 48 Bit 24Bit Main High XEQ High Peak Thermo High Delay Delay lt 300 Taps Limiter Limiter Input Gain Mid Gai in 48 Bit 24 Bit Digital 14XPEQ DS 1 4 Mid XEQ Mid OS 4x Peak Thermo Mid Output Option 64 Taps Delay lt 480 Taps 64 Taps Limiter Limiter Input Gain Low Gai in 48 Bit 24 Bit Digital DS 1 4 Low XEQ Low OS 4x OS 4x Peak Thermo Low Input 64 Taps Delay lt 650 Taps 64 Taps 64 Taps Limiter Limiter Input Gain Sub G ain 48 Bit 24 Bit Sub XEQ Sub OS 4x OS 4x Peak Thermo Sub Delay lt 70
32. components and are part of the correcting filter network The intrinsic delay of the DSC28 depends on the path and is between 5 and 7ms and determined by AD DA conversion down oversampling and the pre Chapter 10 1 viewing limiter concept All further delays occur during the signal manipulation of filters and are especially dependent on the run time behaviour of the loudspeaker system the cross over slopes and filter frequency thresholds Filter The DSC28 calculates crossovers and speaker equalisation as FIR filters finite impulse re sponse This type of filter requires more computing power in comparison to digital devices processing with IIR filters which simply simulate analog filters within the digital domain For this reason the computing power has been optimised by the use of down oversampling modules in each path frequency bandwidth with the exception of the HIGH path This results in band width limitations of the SUB LOW and MID paths FIR technology allows the realisation of equalising filters with linear phase behaviour which has the advantage of equal delay times for every frequency range of a complex signal Unfortunately linear phase response down to the lowest frequency levels lead to extended signal run times unacceptable for live situations This can be compensated by attenuation to minimum phase analog response below a specific frequency For the filtering of an active system with moderate crossover slopes lt
33. cted devices independent of thier channel address Enter Confirm your selection with ENTER SYS 2 AES Stat Samplerate SYS AES Stat Samplerate Further menus A SYS 1 O S hu Bes Shows the measured internal sample rate Men s unter SHIFT SYS Reboot Initialise Further menus 4 SYS 2 A SYS 5 10 2 E E 2 B 1 Reboot This command resets all DSPs The last loaded program and data set is re loaded and exe cuted The outputs of the device are muted during this operation Enter Execute function Bi 2 Initialise This command calls up the menu SYS 4 Enter Execute function SYS 4 Initialise System Further menus none Cursor position none F1 Exit F1 is the SYS Key The devicewill not be initialised F2 CIrRAM F2 is the SET Key This command returns the device to it s defined initial setting at the time of shipping All values return to the so called First Parameters values to which this guide refers when explain ing the menus All adjustments performed by the user are lost The device is ready for the initial start up Attention Even presets re loaded over the serial interface are lost and have to be re loaded after this action GQ Chap 17 Never apply this command without being aware of the consequences Chapter 9 2 Chapter 10 Unit Description The GAE DIGITAL PA MASTER DSC28 is a state of the art digital System Controller Its role as an inte gral system component even at the development stage of
34. e DSP s Digital Signal Processors processing range and as such can not be produced EQ 0 EQ Gain On Off EQ Volume EQ Further menus Y EQ 1 14 3 dB On Cursorposition Bi B2 1 Gain 24dB O 3dB OM 0dB 1dB steps Should the activated EQ bands be solely used for fre quency reduction then a value of OdB is recommended Should however frequency boosts also be applied then a gain reduction in accordance with the highest boost value is recommended B 2 On Off Off OM On EQ on off At EQ off all filters are non effective the adjusted EQ Gain however remains effective EQ 1 EQ 1 Further menus V EQ 2 14 A EQ 0 Cursor Position Bi 6 927 B3 8 H 4 9 5 10 1 Kanal I Link L gt O LR 6 Kanal Il Link R O LR To cancel the channel link function H Chap 12 1 2 page menus 2 7 Type HP12 HP 6 LP12 LP 6 O Peak OM LS 6 LS12 HS 6 HS12 Adjustments to an EQ only become affective after depressing the increment dial ENTER 3 8 Q 0 1 O 2 0 OM 6355 steps 0 1 Min 3 0 0 2 3 0 6 0 1 6 0 10 0 2 10 0 50 0 5 50 0 200 10 200 1000 20 1000 Max For a general step increment of 0 1 additionally press the SHIFT key during the Q selection 4 9 Hz 1 0Hz O 30Hz OM 20 0kHz steps 1Hz 1 0 100 0Hz 2Hz 100 0 150 0Hz 5Hz 150 0 300 0Hz 10Hz 300 0 600 0Hz 20Hz 600 0 1 00kHz 50Hz 1 00 5 00kHz 100Hz 5 00 20 0kHz For a general step increme
35. e device Master Enter Confirm your selection with ENTER B TX Chn 1 ONN 16 OM OCM The transmitting channel for parameter adjustments can be selected by this parameter As well as channels 1 16 an Omni Channel Mode OCM can be selected Parameter ad justments via this channel are accepted by the connected devices independent of their channel address Enter Confirm your selection with ENTER RS 232 Kabel The supplied RS 232 connection cabel is wired as follows Pin No SUB D9 DIN5 1 RS 232 Tx 9 2 Pin No SUB D9 5 DIN5 2 RS 232 GND Pin No SUB D9 3 DIN5 3 RS 232 Rx MIDI cable MIDI DIN5 pin standard cable has following wiring Pin No 1 not connected 2 GND 3 not connected 4 MIDI 5 MIDI Due to it s twofold occupation the RS 232 MIDI Input socket does not conform to the standard as pin 2 RS 232 GND necessary for the RS 232 connection is not connected Chapter 19 1 Chapter 20 Remote control A PC remote control software is not available Chapter 20 1 Analog inputs Analog outputs Remote control Intrinsic run time Chapter 21 Technical Specifications electronically balanced input impedance 20kQ maximum input level 28dBu lt 1kHz 18dBu 20kHz noisefloor 100dBu dynamic range 127dB all measurements linear rated 22Hz 22kHz per channel a Dual Range 24 Bit Delta Sigma AD Converter with pre emphasis 50 15us sample rate 44 1kHz electronically balanced output impedance
36. e does the crest factor hardly ever sink below 6dB within the bass channel This is only a difference of 3dB when compared to a pure sine wave signal Therefore the supply voltage should be able to collapse by this amount without causing the amplifier any difficulty in terms of permanent power capacity at defined peak power and with standard Chapter 15 2 source material Most power amplifier manufacturers design their power supplies to be stable in terms of permanent load capacity has a sign of quality Contrary to this however are the concepts providing extreme relationships between impulse and permanent load capacity which is without doubt efficient for use within the Mid HF range If the threshold of the peak limiter coincides with the amplifiers permanent load capacity short impulses will be limited to this capacity even though the amplifier might easily be able to reproduce them without compression Clip distortion resulting from the short term requirement of the maximal available impulse power by signal peaks are either not discernible or e g when hitting the bass drum even lead to the desired kick sound Consequently according to the hardness of the power supply significant power reserves remain unused At a peak permanent load capacity ratio DHR Dynamic headroom of 3dB half of the amplifiers impulse power lies neglected For this reason a further duty of the pre viewing peak limiter is the increase of the limiter thres
37. e output levels of all paths after the limit ers As such it is possible to reduce the possible maximum output level of the PA system and so rightly earns the title of Safe control The adjustment range is 0 24dB Dynamics The following noise and dynamic values are stated unweighted the measurement band width is 22Hz 22kHz The high input dynamic of 130dB at an input level of 28dBu is realised using dual range conversion at the input as well as the implementation of analog pre emphasis digital de emphasis filters Input dynamic is defined as the ratio between the driving limit of the non amplified AD channel and the noise level of the amplified channel The DA converters dominate the transient dynamics of the device with their low signal to noise ratio of 114dB whereby a Dynamic overflow of 130dB 114dB 16dB is generated which is added to the limiter function as additionally usable headroom The digital signal processing is adjusted to these values in such a way as to always ensure that it operates with sufficient accuracy 48 bit instead of 24 bit accuracy Implementation of the Noise Shaper with 1 order error feed back during reduction from 48 bit to 24 bit As such the digital re quantization noise is always lower than the noise level of the converters and other analog modules The great advantage of noise shaper imple mentation is the reduction of distortion appearing when amplifying low signal levels This improves the controlle
38. es for the actual task of system equalising Alignment Run time compensation of loudspeaker components L Ph XEQ Band passes and system equaliser L Ph L M Ph M Ph Limiter Pre viewing limiter concept L Ph 3 Additional run times The run time behaviour of modules that can be influenced by the user interface PEQ Equaliser dependant on configuration M Ph Master Delay Additional run time for all paths L Ph Path Delay Additional run time for single paths L Ph Basic run time In summary it can be stated that a basic run time is present in each of the controller s 4 paths Because of the different down over sampling values within the individual paths a deviating linear phase behaviour with minimum phase segments results As well as the amplitude frequency response the run time portion of the controller s intrinsic behaviour has to be considered when creating the system s equalising network The total basic run time of the DSC28 is strongly dependant on the type of system equalisation employed and as such dependant on the loudspeaker system which is to be connected The linear phase segment for an activated preset is represented within the menu Master Delay The additional group 3 run times generated by the user are not added to the total basic run time ms 15 10 1 2 4 5 3 0 0 01 0 1 1 10 kHz Fig 13 3 Intrin
39. gnal before and after the de emphasis filter a difference of 5 5dB is seen which corresponds to the dyna mic gain independent of using the dual range principle won with this procedure The increase to the high frequencies results in a reduction of the range of high level adjustment Max 28dBu 10dB 20kHz ho wever this represents no problem to musical signals within the normal spectral range PEQ Additionally available to the user is a parametric equaliser with 14 bands per channel situated prior to the crossover network These are of the analog IIR filter type and are adjustable via the user interface The following filter types can be selected Bell Low Shelving 6 and 12dB octave High Shelving 6 and 12dB octave Low Pass 6 and 12dB octave High Pass 6 and 12dB octave The preceding EQ gain function is designed to attenuate the input signal to compensate a possible boosting by the filters The signal leaves the gain stage with 48 bit accuracy The entire PEQ calculation including the filter coefficients is processed at this accuracy A limiter is introduced to the output of the PEQ to prevent possi ble overflow With the help of a noise shaper with Error Feedback attenuating any quantization faults dis tortions and noise of this module the signal is reduced back to 24 bit accuracy High and low pass filter types should only be used for additional bandwidth limitation on the lowest and highest end of the total transmission range
40. h linear phase behaviour which has the benefit of equal delay to all frequency segments of a signal Inconvenient is that linear phase behaviour down to the lowest frequencies of the transmission range causes extended run time which is unacceptable in live performance situations This can be compensated by attenuation to minimum phase analog response below a specific frequency For the filtering of an active system with moderate crossover slopes lt 36dB octave minimum phase response up to 120Hz and linear phase response gt 120Hz a controller delay of 30ms has to be considered intrinsic controller delay of approximately 7ms filter delay of approximately 23ms Parameter set construction In combination with the measurement system MF from the ITA Institut f r technische Akustik at the RWTH Rheinisch Westfalische Technische Hochschule Aachen which includes software for parameter set construction the DSC28 serves the loudspeaker system developer as a universal tool for creating XEQ filter and protection functions As such it is a direct constituent in the development of a loudspeaker system The completed parameter sets of the XEQ filters can be written into the Flash RAM of the controller by means of the RS 232 or MIDI interface The result can immediately be evaluated by measurement and hearing assessments On completion of system development several parameter sets e g for different stacking variations can be written into the Flash
41. he supplied Sub D9 DIN 5 pin cable establishes the RS232 connection to the serial interface COM of a PC For GAE Systeme the necessary files in the form of a self extracting ZIP file can be requested from the manufacturer Chapter 17 1 Chapter 18 Activating the remote on off switching The socket REMOTE ON facilitates the remote on off switching of the device by means of an auxiliary voltage ac or dc voltage of between approx 12 24V To activate this function proceed as follows e First remove the controller from the mains supply then open the cover e To avoid damage caused by electrostatic charge touch the device s casing and at the same time an earthed object such as the earth contact of a Schuko socket outlet the metal case of another still con nected device or a central heating radiator e Please refer to figure 18 1 Open the marked contact e Close the cover and only then reconnect the device to the mains supply Examine the function of the remote on off switching Turning on the auxiliary voltage switches on the controller ES a ml e010 11 REMOTE se C Fig 18 1 Contact for remote on off switching Chapter 18 1 Chapter 19 Parameter control for several devices L 1 DSC28 2 DSC28 3 DSC28 4 DSC28 5 DSC28 f 16 DSC28 1 DSC28 2 DSC28 3 DSC28 4 DSC28 5 DSC28 F 16 DSC28 Fig 19 1 Driving chain
42. hold value with a time constant feedback according to the ability of the amplifier Surge Unfortunately most amplifier specification sheets have no information regarding power supply behaviour and as such must be estimated for limiter programming purposes Chapter 15 3 Chapter 16 Connection hints Due to it s self explanatory nature an illustration of the device s rear panel has been dispensed with The connection ports and respective important notices are explained below ANALOG INPUTS ANALOG OUTPUTS AES EBU INPUT AES EBU OUTPUT MIDI RS232 INPUT MIDI OUTPUT REMOTE ON FUSE T1A Electronically balanced EHI Chap 11 1 source loading 20kQ CHANNEL A LEFT and CHANNEL B RIGHT 3 pin female XLR 1 Shield 2 Phase 3 Phase Electronically balanced 21 Chap 11 3 internal resistance 20Q maximum load capacity gt 375Q 18dBu maximum output level The DSC28 outputs can be considered connection wise as transformer outputs CHANNEL A LEFT and CHANNEL B RIGHT 4 paths 1 SUB 2 LOW 3 MID 4 HIGH 3 pin male XLR 1 Shield 2 Phase 3 Phase Transformer balanced source loading 1109 3 pin female XLR 1 Shield 2 Phase 3 Phase Transformer balanced internal resistance 1100 3 Pol Male XLR 1 Shield 2 Phase 3 Phase Consolidated input due to the respective hardware conventions Due to the twofold occupation of this socket the MIDI Input does is not conform to the standard as pin
43. ion in amplitude and phase under consideration of sound pressure and power values of the loudspeaker components as well as the controller s intrinsic response The equaliser is calculated in respect of a target function e g linear from Freq X to Freq Y e a band pass filter structure based on the cross over function of the individual paths including the dispersion behaviour within the cross over range as well as the performance data of the loudspeaker components e delay compensation time alignment of the PA s loudspeaker components whose acoustical focal points lie on different vertical lines e limiter thresholds related to the performance specifications of the connected loudspeaker components and power amplifiers e output scaling factor which guarantees an optimal balance between the analog default level amplification of the DSC28 the amplification factor of the connected power amplifier the limiter thresholds and the highest possible digital resolution of the signal processing Output gain Before being passed to the DA converter the signal can if necessary be reduced in common level all four paths by the output gain within a range of 0 24dB This function operates like a volume regulator but can be better interpreted as a Safe regulator as it s location after the limiters enables the setting of the maximum possible output power of the PA system DA converter A State of the art 24 bit Stereo DA converter of the Delta Sig
44. iven with low level sig nals Peak limiter The DSC28 fully incorporates the improved possibilities made available to limiter circuitry by digital technology An in advance signal analysis of the output signal allows pre determined threshold ex cesses to be detected and the level to be decreased over a fixed time factor with a matched time constant Over shooting peaks are automatically reduced to the exact value determined by the threshold level This procedure enables the exact adjustment of the limiter thresholds to the absolute maximum rating of the am plifiers and the connected loudspeakers Strong transient impulses additionally benefit from a pre masking effect which allows an inaudible processing time before the occurrence of the impulse The subsequent hold time prevents level modulation in the directly ensuing passages The pre viewing signal analysis also allows a much better generation of the control signal without worsening the attack time of the limiters High frequency distortion of the wanted signal by the control signal are thus as good as eliminated RMS limiter The RMS limiters of the DSC28 protect the loudspeaker components by simulating the thermal time constants of the voice coils and magnet materials Delay A master delay for the adjustment of the signal delay time is situated prior to the filter network A further delay is inserted to each output path for time alignment adjustment of the individual loudspeaker
45. lifying factors of the power amplifiers and the analog output driver stages of the DSC28 Output Range Analog output The adjustment to different amplifier ratings is performed here in three output gain stages of maximum possible output levels of 18 12 and 6dBu The noise level caused by the DA converter as main noise source is correspondingly reduced Gain regulator The function of the gain regulators is resolved from their operating range and their position within the signal chain e EQ gain This function decreases the input signal to compensate filter boosting Adjustment range 0 240B e Input gain This function has a wide range level adjustment of 83 45dB and as such can be used as a volume control It is however intended to allow the matching of the driving source signal to the PA system connected to the controller outputs particularly when using the analog inputs Due to the high driving level of the analog inputs lt 28dBu an adjustment value of around 30dB for a 4dB system is considered normal e Path gain This function represented here by high gain allows any necessary slight level adjustments of the individual paths in relation to each other The adjustment range is 18 6dB Placed ahead of the limiters this function is not able to perform level adjustment for amplifiers with differing amplifica tion factors than those originally assigned e Output gain This function allows the parallel decrease of th
46. ll parametric EQ situated prior to the cross over network enables the comfortable system tuning to room acoustics Input dynamic Without the necessity for analog level matching the GAE DSC28 achieves with the help of dual rate conversion and a combination of analog pre emphasis digital de emphasis a dynamic range of 130 dB with 28 dBu maximum input level lt 1kHz Output dynamic The output level of each of the outputs of the DSC28 can be individually matched to the input sensitivities of the connected power amplifiers DSP controlled noise shaper and dither stages with 1 Order Error Feedback DA Server serving the DA converter of the R2R instead of the Delta Sigma type are used This results in an extraordinary dynamic range of gt 114 dB Noise dynamics and headroom The signal to noise ratio of the DSC28 is 96dBu unweighted 22Hz 22kHz at a maximum output level of 18dBu The lowering of the output level when adjusting to a power amplifier leads to a corresponding reduction of the noise level At the same time the best possible digital resolution is attained these procedures unusual in a digital device make the DSC28 even superior to an analog standard controller Important modules within the digital range operate with 48 bit accuracy Where a reduction from 48 bit to 24 bit occurs a noise shaper with 1 Order Error Feedback is always im plemented so as to guarantee minimum noise level and distortion especially when dr
47. llable switch on function GJ Chap 18 will necessitate the removal of the device cover Never forget to disconnect from the mains before opening the cover 6 The device including the mains cable and plug may not be altered or redressed The operation with an opened enclosure is not permitted 6 Always ensure the correct grounding of the device via the mains plug Never cover the grounding terminal of the plug by means of insulation material 6 Mains fuses cannot prevent an unexpected malfunction of electrical components rather they should protect the user and its environment from damage For this reason never try to substitute the mains fuse by any other than the specified M1A type 1A medium behaviour Never try to repair or bypass a blown mains fuse 6 The substitution of power amplifiers in a system driven by the DSC28 may only be carried out without further consideration when the performance data and amplification factors of the new amplifiers are to tally comparable to those of the original amplifiers If that is not the case possible losses to the tonal be haviour and to the safety of the speaker components in the system must be taken into account 6 High sound pressure levels can lead to irreparable injures to the human hearing In the region of the threshold of pain even physical impairment of the entire organism cannot be exempted Modern sound systems are designed for high sound reproduction levels and as such when improperly ha
48. m 10 25 depth 3 75kg net without packaging Chapter 21 1 Chapter 22 Addresses bibliography miscellaneous Entwicklung eines Signalprozessors zur Entzer rung von Lautsprechern A Goertz Diplomarbeit am ITA RWTH Aachen 1990 Entzerrung von Lautsprechern mit einem Signal Prozessor System in Echtzeit A Goertz D Leckschat Tagungsband DAGA 1990 Digitale Lautsprecherentzerrung A Goertz D Leckschat Tagungsband VDT 1990 Nichtlineare Entzerrung von Lautsprechern in Echtzeit mit einem Signalprozessor A Goertz W Klippel D Leckschat Tagungsband DAGA 1991 Verbesserung der Wiedergabequalit t von Laut sprechern mit Hilfe von Digitalfiltern D Leckschat Dissertation am ITA RWTH Aachen 1992 Vergleich verschiedener Verfahren zur digitalen Lautsprecherentzerrung A Goertz Tagungsband DAGA 1992 Aufbau und Inbetriebnahme einer digitalen Fre quenzweiche mit einem Signalprozessorsystem S M ller Diplomarbeit am ITA RWTH Aachen 1992 Einsatz digitaler Limiter in Beschallungsanlagen A Goertz S M ller Tagungsband DAGA 1995 Klangeigenschaften digital entzerrter Lautspre cher K H Pflaum Diplomarbeit am ITA RWTH Aachen 1995 Literature studies and dissertations referring to the subject Optimierung der Entzerrer bertragungsfunktion f r Lautsprechersysteme durch Ber cksichti gung psychoakustischer Effekte A Goertz K H Pflaum Tagungsband DAGA 1996 Aufbau eines erweiterten digit
49. ma type is incorporated into each path and switched in parallel The resulting dynamic range is gt 114dB A 2 order low pass provides the reconstruction of the digital data flow of over sampled converter output signals Analog output The output amplifiers of the DSC28 are electronically balanced and of low impedance Connection wise they can be regarded as a transformer output i e the output signal only flows between the two active output poles EJ Chap 16 At a maximum output level of 18dBu the output can be loaded with gt 375Q The result remarkable for a digital device is an output dynamic value of gt 114dB The matching of the different amplifier values is performed in three maximal possible output level steps 18 12 and 6dBu and receive fine adjustment during XEQ filter scaling As such the superior dynamic values and the particularly high resolution of the digital signal at the DA converter inputs are maintained even when connecting to high power amplification Chapter 11 3 Chapter 12 Gain architecture and dynamics EQ Gain Input Gain High Gain Output Gain High Scaling Filter Scaling Threshold Output Range Analog AD 14X NE OV Noise DA Analog 18dBu XE Limit i 12dB Mel 24 Bit 8X 24 8it Tl output dBu 28dBu gt Digital FS 24 Bit 48 Bit 24 Bit 48 Bit 24 Bit 48 Bit 24 Bit Digital FS gt Max Output Fig 12 1 Gain architecture Fig 6 1 is an extract of the block diagram and shows the signal flow using the example
50. ndled can cause injury to the human hearing organs Never expose anybody not even yourself to extreme high vol ume levels over a longer period of time All Rights Reserved This User s Guide or extracts thereof may not be reproduced or duplicated by elec tronical mechanical or chemical means for any form of commercial use without the written consent of the author Chapter 2 1 Chapter 3 Initial start up The Digital PA Master DSC28 is a system controller for the driving of pre specified loudspeaker systems For this reason the manufacturer has supplied this unit with presets consisting of frequency cross over s H Chap 14 system correction L Chap 14 and limiter GJ Chap 15 for one or more systems One of these presets is loaded after connection of the device to the power source so that provided that it has been prop erly connected the system is immediately ready for use EJ Chap 16 The controller is the link between numerous signal sources and editing devices and the power stage of the system amplifiers transducers One of its top priority duties is the protection of the speaker components as well as the avoidance of excessive distortion For this reason the manufacturer has exactly calibrated the DSC28 to match your power amplifiers GJ Chap 12 Chap 14 To start up proceed as follows e Be sure that the controller s configuration matches your power amplifier set e Secure the device in a rack with four bolt
51. nt of 0 5Hz additionally press the SHIFT key when selecting the filter frequency Chapter 8 1 l 5 10 dB 99 0db press the SHIFT key duri A higher step increment O 0 0dB OM 12 0dB 1dB steps For a general increment of 0 1dB additionally ng the level selection during the parameter selection can be achieved by the simultaneous turning and pressing of the increment dial In EQ menu 1 the oper parameter default settin EQ 2 EQ2 Further menus Y EQ 3 EQ 3 EQ 3 Further menus Y EQ 4 EQ 4 EQ 4 Further menus Y EQ 5 EQ 5 EQ 5 Further menus Y EQ 6 EQ 6 EQ 6 Further menus Y EQ 7 EQ 7 EQ 7 Further menus Y EQ 7 EQ 8 EQ 8 Further menus Y EQ 8 ation of EQ ihas been described EQs 2 14 are shown to document their first gs The operation of all EQs is identical EO dype Q Hz dB 14 A EQ 1 0 LR Peak De 60 u EOS 2s Type Q Hz dB 14 A EQ 2 0 LR Peak AE 30 EQ 4 Type Q Hz dB 14 A EQ 3 0 LR Peak 2 0 279 0 0 EO 5 TYDE Q Hz dB 14 EQ 4 0 LR Peak DSD ro EQ 6 Type Q kHz dB 14 A EQ 5 0 LR Peak Re 0 EQ 7 Type Q kHz dB 14 EQ 6 0 LR Peak AA AOO OR EQ 8 Type Q kHz dB 14 A EQ 7 0 LR Peak 2 4 00 r0 Chapter 8 2 EQ 9 EQ 9 Further menus EQ 9 14 A EQ 8 0 EQ 10 EQ10 Further menus V EQ 11 14 A EQ 9 0 EQ 11 EQ11 EQ11 Type Q Hz dB Further menus 4 EQ 12 14 A EQ 10 0 LR EST 07 40 00 EQ 12 EQ12 EQ12 Type Q kHz dB Further men
52. nus V EQ 0 5 gt SET 6 1 by selecting the required set up and confirming choice by pressing ENTER The new set up is only made available after the ENTER key has been pressed gt Store With the RIGHT key move to this menu point to store the momentary settings and a neme to a memory position M These keys have no function in this menu Chapter 7 2 SET 6 1 Save Setup as This function cannot be interrupted characters and stored to one of 15 memory positions 000 014 M Space A Character A M Character a With the UP key the three indicated start positions for character selection with the increment dial are reached Pressing the DOWN key steps back through the list SHIFT gt This combination inserts a space at the cursor position gt Character 1 gt Character 2 gt gt Character 18 With the RIGHT key all 18 character positions are reached so as to adjust them by means of the increment dial Depressing the LEFT key enables the return to the previous positions The active position is underlined with a cursor ENTER Confirm the selected name by pressing the ENTER key The third page of the menu is now acti vated in which you can store the adjusted EQ Set up to a memory location Warning This function cannot be interupted The menu can only be exited by pressing the ENTER key SET 6 2 Save Setup to This function cannot be interrupted to be stored The name shown in the display represen
53. ocessing chain have a limited dynamic range which is restricted by noise at the lower end and by the operating limit at the higher end Analog signal processing components can be cost effectively dimensioned with sufficient headroom to make sure that no clip distortions can occur In contrast such an over dimensioning of a PA system power amplifiers loudspeakers is hardly economic and as such the operation of a PA system within it s critical range is the normal situation An exceeding of the limits can not only lead to high distortions but even to the destruction of the components Consequently the surveillance and control of the signal level belongs to the elementary functions of a signal controller Only by this means can a PA system be reliably operated within the admissible load range The module described as limiter is the last component before the power amplifiers within the signal processing chain It is able to effectively intercede particularly in active multi path systems due to its assigned magnitudes for each path being individually matched to the driven speaker component The necessary level attenuation carried out by the limiter is based on two different types of loudspeaker overloading Peak limiter The peak limiter prevents mechanical overload caused by too high acceleration forces resulting in excessive material stress the disruption of membrane and cones in tweeters and cone drivers the destruction of the voice coil caused b
54. oefficients EJ Chap 14 Limiter system Two limiter functions per path follow A pre viewing operation mode involving a short delay of 1 5ms has been realised for the peak limiter whereas a thermo limiter protects the loudspeakers against thermal overstrain During the creation of the XEQ filter coefficients each transducer is matched exactly to the performance specifications of the connected power amplifier The protection circuitry within the controller intervenes according to this data LI Chap 15 Multi path The so called multi path variation of the controller is not shown in the block diagram Two respectively max three paths of the controller can be summed post limiters and made available at a common output This allows the opportunity to create correcting filters even for loudspeaker systems with passive cross overs The XEQs and the frequency range limiter of the single passive drivers are individually configured Finally the single edited paths are summed together led to a further peak limiter level and made available at one of the outputs Possible multi path variations and their assigned outputs are as follows Chapter 11 2 LOW MID gt 3 MID MID HIGH gt 4 HIGH LOW MID HIGH gt 4 HIGH Presets A preset consists of a parameter set configured by the loudspeaker system manufacturer which is selected as a whole and activated for the system by means of the controller The DSC28 preset consists of e system correct
55. our inspection provided that the device has been returned freight and carriage paid and in the original packaging Excluded from guarantee are faults incurred by improper electrical or mechanical connection or as a result of transport or accident This guarantee is voided by any unauthorised repair attempts or by the removal or alteration of the device s serial number Contents of packaging The standard packaging of the DSC28 contains e 1 GAE Digital PA Master DSC28 with customer specific configuration e 11 5m connection cable Sub D9 DIN 5 pole for the connection to a serial PC interface COM e 1 replacement fuse M1A 1A medium slow behaviour e 2 lateral mounting elements e 1 front and 1 back mounting element e PE cover e User s Guide e single sheet documentation with serial number preset configuration output stages configuration etc Chapter 1 1 Chapter 2 Important notices J denotes Refer to further information in the chapter specified by the symbol denotes Please regard this warning as especially critical Before the initial start up of the device please read the following indications and warnings Read the User s Guide carefully It contains numerous pointers for the proper use of the device It cannot be excluded that this user s guide shows typographical failings or misprints it is however regu larly checked and proof corrections can be requested in the form of future updates Modifications which
56. p Access to the menu area EQ IIR equaliser Selects single menus within the menu areas Holding the button depressed enables quick step through Symbols UP key A DOWN key Moves the blinking Cursor within a single menu to an adjustable parameter also page change if a menu consists of more than one page X denotes cursor position X counted from left to right Holding the button depressed enables quick step through Symbols LEFT key RIGHT key gt Shift function key for all other tip keys and the increment dial Parameter selection by turning the dial Symbol OM Simultaneously pressing whilst turning the dial enables larger value steps For smaller value steps press the SHIFT key whilst turning the dial Confirm setting by pushing the dial PUSH ENTER Symbol A flashing point in the display top right warns that a parameter value has been changed and requires confirmation before becoming operational Infra red remote control activity An optional IR receiver is situated behind the small round window Attention This function is available as an optional extra A vacuum fluorescence display with two 24 character lines enables the visual control of all operation measures carried out at the user controls The display s blue coloration allows easy distinction of the characters even under adverse lighting conditions Cursor active menu e Is blinking requires confirmation by ENTER gt Fu
57. r network the user has access to 14 EQ bands per channel The EQ is of the IIR filter type and can simulate analoge filters adjustable at the user interface The following filter types can be selected Bell Peak Low Shelving 6 and 12dB oktave LS 6 and LS 12 High Shelving 6 and 12dB oktave HS 6 and HS 12 Low Pass 6 and 12dB oktave LP 6 and LP 12 High Pass 6 and 12dB oktave HP 6 and HP 12 Attention Only use the high and low pass types of EQ for additional band limitation on the lower and upper end of the total transfer range Or if possible do not use them at all Be aware that a system equalising cross over exactly adjusted to the loudspeaker system is already present within the preset Use the Bell type filters to compensate the tonal discoloration caused by room acoustics Especially within the bass range the possibilities of balancing the single loudspeaker paths through use of the Output Gain controls should first be applied EG Chap 7 2 Attention should be given to one restrictive particularity which appears when using shelving filters If for ex ample a 12dB octave high shelving filter with a cut off frequency of 5kHz and a Q factor of 0 7 is selected the user interface will accept no gain setting higher than 5dB A higher value is only obtainable by reducing the Q factor or raising the cut off frequency The device sets these limits due to the calculated filter coefficients required for a higher boost exceeding th
58. r s tonal behaviour Analog controllers offer dynamic ranges of around 120dB in relation to a usual maximum output level of 26dBu The resulting noise level is 94dB whereas the DSC28 is comparable value of 96dB is reached at an output level of 18dBu The reduction of the output level by adjustment to a connected amplifier leads to a corresponding decrease of the noise level Amplifier adaptation As already mentioned the adjustment of the DSC28 s output stages to match the amplifier to be connected is performed by the digital analog gain structure An example of mis adjustment shows the advantages of this procedure A power amplifier provides the requested power for Chapter 12 2 the connected loudspeaker component at an input voltage of OdB The necessary adjustment of the DSC28 s output level is not carried out and to adjust the controller s output scaling must be set at 18dB In the following table these values are compared to the values at correct adjustment The result is a worsening of the output noise level and the dynamic range by 18dB but also a worsening of the digital resolution by 3bit the value corresponding to 18dB Therefore the converter may only be driven with 21 bit to ensure that the power amplifier output is not overdriven Adjustment Output Output Output Noise level Noise level Resolution Output Scaling Level max Level eff DA Out digital dynamic 96dBu 114dBu 24 Bit 114dB 18dB 18dBu 96dBu
59. reshold an example of minimum phase objects are analog filters and speaker components The run time shown represents a Butterworth low or high pass 24dB octave 100Hz cut off frequency Curve 3 Combination The shown run time could belong to a digital controller which calculates the Butterworth filter by using the linear phase portion of curve 1 The run time frequency response group delay is generally favoured to the phase frequency response as the phase representation at wide frequency ranges and higher run times is not interpretable At a delay time of 5ms and a frequency of 1kHz the phase amounts to 1800 at 10kHz it amounts to 18000 The different run time behaviours are denoted by the abbreviations L Ph for linear phase M Ph for minimum phase and L M Ph for combination phase Chapter 13 1 1 Intrinsic run time behaviour The run time behaviour of the necessary modules for the function and operating principles of the controller which build the basic requirement for the use of further modules Analog input High and low pass pre emphasis anti alias M Ph AD converter Conversion decimation dual range principle L Ph De emphasis inverse to pre emphasis results L Ph Down sampler decrease of sampling rates M Ph Over sampler increase of sampling rates M Ph DA server Interpolation L Ph DA converter Conversion L Ph Analog output Reconstruction M Ph 2 Run time behaviour of the equaliser The run time behaviour of the important modul
60. rther menus right Further menus left gt Further menus right as well as left Chapter 5 1 Chapter 6 Boot display first parameters and menus Boot display ITA RWTH DIGITAL PA Further menus after approx 4 secs This display appears after power up The basic hard and software concept for the GAE DSC28 was devel oped within the framework of a research and development arrangement with the Institut fur Technische Akustik at the Rheinisch Westf lischen Technischen Hochschule Aachen ITA RWTH Aachen First parameters On shipping the device all positions in all menus of the system controller DSC28 are occupied by first parameters The device is as such instantly ready for operation This condition can be re called via the function Initialise 11 Chap 17 However to prevent misunderstanding the function Initial ise is not necessary for the initial power up EHI Chap 3 All menus shown within this guide are shown with these first parameter settings The relevant symbols are explained in the following two examples e Min O X OM Max denotes starting from the first parameter value X clockwise rotation of the incre ment dial increases values up to Max respectively counter clockwise rotation decreases values down to Min e L gt O LR denotes starting from the first parameter LR the parameter L can be adjusted by coun ter clockwise rotation of the increment dial After the first parameter adjustment the new v
61. s and connect the in and output ports according to the connec tor panel requirements L Chap 16 e Connect the controller with the mains voltage The unit operates safely within a range of 85V 265V The performance can even be conditionally maintained down to around 50V however the output paths will become muted at lt 75V Devices with a serial number up to 0030 recognise and store as to whether they are connected to the European net 230 240V or to the American net 110 115V and mute at volt ages of lt 155V or lt 75V respectively e After about 4 seconds the device is ready for operation and releases the protective mute to the 8 audio outputs Now you are automatically located at the menu point Input Gain and able to carry out the first important settings e During the initial start up make sure that you only operate your system at low nominal signal levels Double check the correct speaker link up of the particular paths of your system LJ User s Guide of the speaker manufacturer e Should you have not reached this first menu point read J Chap 17 Initialise e Match the power amplifiers to the source device by adjusting the Input Gain by means of the increment dial and parallel to this whilst observing the VU meters and the DSC28 s output limiter display move the master fader of your mixing console to the desired full operating level of your system Because of the high maximum dynamic range of the analog inputs of
62. s well as tonal and other disadvantages the safety of the loudspeaker components is endangered by the imple mentation of wrong limiters especially when the amplification factors are higher than the original ones If necessary a simultaneous adjustment to higher amplification levels can be carried out by the function out put gain provided that this corresponds to all paths at the same time An adjustment to amplifiers with lower amplification factors however is not possible and as such the proper guaranteed capacity of the loudspeaker system can not be fully realised 6 The replacement of power amplifiers to a PA system may only be performed without further con sideration when power specification and amplification factors are absolutely identical otherwise encasing losses in tonal responce and loudspeaker component safety have to be taken into ac count Chapter 12 3 Chapter 13 Run time behaviour Fig 13 1 Run time behaviour Fig 13 1 shows the signal flow as an extract from the block diagram using one of the three lower frequency paths as an example Only modules influencing the run time behaviour are considered Run time Every digital controller adds run time to the signal passing through the signal processing ICs causing an intrinsic delay which is inevitably higher than that caused by an analog device This applies also to the digital device which is simply simulating an analog device The signal is delayed when leaving the
63. serve the purpose of technical improvement of the device may be carried out without prior notification Keep the original packaging of your DSC28 so that in the case of returning the device for maintenance it can be shipped originally packed We reserve the right to replace non original packaging on returning the device to the owner In this case the packaging will be invoiced to the customer Always pay attention to the sufficient cooling of the device during operation This especially applies when installing in racks above other heat generating devices Never pull the mains plug by means of the mains cable Always pull the plug itself Be certain that the mains cable does not become crushed or damaged by sharp edges and never replace a damaged mains cable yourself During operation and storage always protect the device from dust moisture and direct sunlight Only clean the device with a dry linen cloth In the case of strong soiling this can be moistened with water and a small amount of household detergent Never use cleaning agents containing solvents to clean the device Use only high quality cable material to connect the device 6 Leave all repair and maintenance work to qualified technical personnel Any future guarantee claims will be invalidated by unauthorised manipulation 6 The opening of the device is not required for operation as there are no user adjustable components located within the casing Solely to release the remote contro
64. sic run time behaviour The basic run time of the four paths is represented by the values of the linear phase segment of the delays including the delay of the pre viewing limiters 1 5ms SUB LOW 7 2ms MID 5 4ms HIGH 5 6ms The minimum phase segment and the run time increase towards lower frequencies and are caused by the 1 order high pass filter within the analog input module Coupling capacitor Chapter 13 2 Chapter 14 Presets Presets A DSC28 preset consists of a set of parameters configured by the loudspeaker system manufacturer The preset can be selected and loaded in the controller which then runs the system To create a preset the following parameters have to be taken into account System equalisation The interpreted and if necessary edited acoustic pressure and acoustic power measurements of the speaker components the known intrinsic response of the controller as well as a preset target function for the desired resulting frequency response of the loudspeaker combination are the default parameters for the equalisation of the amplitude Differences in sensitivity of the loudspeaker components are considered Filter scaling Band pass structure The band pass structure is derived with the help of target band passes with attention to the crossover behaviour of the loudspeaker components to each other including their dispersion behaviour within the crossover range as well as their power specifications Run time behavio
65. t dial L gt OM LR ment dial R OM LR This 2 page menu function remains even during linked LR condition To reach page 2 from page 1 move the cursor with the right key beyond the far right cursor position 5 6 10 can now be accessed although it is not possible to distinguish the channels as both indicate LR Only when the link function is cancelled the distinction appears as L link function was cancelled at page 1 or as R link function was cancelled at page 2 Chapter 6 1 i B2 a3 a4 E ls a7 is io Bio Menu page 1 Cursor position LR 1 Menu page 2 Cursor position LR 6 The link function can be cancelled by The link function can be cancelled by turning the increment dial L gt O LR turning the increment dial SR O LR If under L and or R changes are made these will be written to the channel from which the link function LR is re activated Chapter 6 2 Chapter 7 Menus under SET Speaker Setup SET 0 Input Gain Balance KN Ba Further menus SET 1 5 A SET 6 De Cursor Position B 2 ment dial by adjusting the Input Gain For this whilst observing the VU meters and the DSC28 s output limit display bring the master fader of your mixing console to the desired position for full level operation of your system Because of the analog inputs high dynamic range limit of lt 28dBu an level of nearly 30dB for 4dBu systems is normal a level difference between the two channels SET 1 Master Delay
66. the signal limitation possibilities of the respective amplifier remain unused Should you however use inadequately dimensioned amplifiers it is important to pay attention to the use of amplifier integrated clip limiters for the sake of both your system components and your ears STATUS This LED bargraph displays information regarding internal settings of the DSC28 INPUT Here the activated input is indicated ANALOG DIGITAL DIGITAL INSERT LOCKED Fs Indicates the sample frequency 44 1kHz 48kHz The DSC28 operates with a sample frequency of 44 1kHz A sample rate of 48kHz can be factory configured for special purposes The DIGITAL INPUT is able to receive AES EBU signals with a sample rate between 32kHz and 96kHz These are then converted to 44 1kHz by means of a sample rate converter MIDI DATA Indicates data reception via the MIDI or RS 232 inputs and data transmission via the MIDI output KEY One of four available security levels for the prevention of unauthorised operation is indicated Attention Security level selection of is not yet supported by the software stand 01 2001 Chapter 4 1 Chapter 5 Increment dial Function keys and VF Displa AUDIO ENGINEERING Fig 5 1 control panel SYS SET SHIFT Increment dial VF Display Access to the menu area SYS system settings Access to the menu area SET speaker set up The menu SET 2 Input Gain amp Bal ance appears approx 4seconds after power u
67. tion is started when the input level exceeds the limiter threshold The threshold is an element in the digital analog gain structure of the amplifier adjustment and as such an integral part of a preset not adjustable by the user Depending on the degree of threshold excess the resulting action is the amplification reduction over a fixed time period with a time constant adjusted to the peak in dB s Should a further peak emerge during the attack operation the time constant is re calculated The subsequent hold phase of 20ms is re started should during this phase the input signal again rise to within 1dB of the limiter threshold Is neither limiter nor hold threshold reached the amplification is restored to it s original value dependant of the release time constant The release time constant in dB s is the only user adjustable limiter parameter Controlled overshoot Conventional power amplifiers differ mainly in circuitry details through the use of MOSFET or BIPOLAR transistors as well as by different approaches to power supplies Conventional power supplies are unstabilised transformer rectifier capacitor concepts Modern switched mode power supplies which are built up with small ferrite core transformers may be either stabilised or unstabilised whereby the introduction of PFC Power Factor Control sinusoidal mains current pull to this type of power supply ensures that they are always stabilised When loading a power supply with current flow
68. to it s original value by means of one respectively two 4x over samplers This process takes place in the reversed order to down sampling For the paths LOW and SUB two 4x over samplers are connected one after the other Analog to the down samplers the low pass filters are equipped with 64 taps and are optimised for maximum rejection band damping Intrinsic frequency response Ripples within the transmission range of the down and over sampling filters as well as the amplitude and phase response of the controller s analog and converter modules are taken into consideration and are inversely re compensated during the generation of the system dependant filter coefficients Input Gain The following Input Gain function has a wide level adjustment range of 83 45dB and can therefore be used as a volume control In the first instance this function enables the adaptation of the feed sources to the PA system connected to the controller s output ports Output gain Paths The next function is the gain function of the individual paths with a level adjustment range of 18 6dB With the OdB position as reference the individual levels of the actively driven loudspeaker groups particularly within the bass range can be balanced to each other according to personal preference The sensitivity of the individual loudspeakers varying within their SPL range have already been inversely taken into consideration by scaling during the calculation of the filter c
69. to react fast enough causing an inevitable short term excess overshoot of the default threshold Hold Gain reduction Release control Parameter calculation Figure 15 2 Pre viewing digital peak limiter Pre viewing peak limiter Digital signal processing also allows extended possibilities for the limiter concept The figure 15 2 shows the principle of the DSC28 s peak limiter Here a multiplication block corresponds to the VCA of an analog limiter The peak limiters of the DSC28 work with pre viewing signal analysis This allows an in advance registration of impending threshold excess values and the reduction of the level over a set period with optimally adjusted time constants In this process the peak is attenuated to the exact threshold level a controlled overshoot of the value is allowed This procedure allows the limiter threshold to be set at the exact value of the connected amplifiers loudspeakers Even sudden jumps in signal level no longer cause overloading For strong transient impulses the effect of pre masking can additionally be utilised to ensure that the control time before the impulse is not perceptible A subsequent hold time prevents level oscillations in the directly subsequent signal paths The pre viewing signal analysis is enabled by means of a 1 5ms delay within the signal path The control of the gain reduction defines in which operating phase the limiter is in attack hold or release phase An attack opera
70. ts the stored set up at this memory position which will be over written On leaving the factory all memory positions are occupied by Default Setup containing the original first parameter set V A gt These keys have no function in this menu Enter Confirm the number of the selected memory position with the ENTER Taste Enter also returns the display to the first page of this menu point Warning This function cannot be interupted The menu can only be exited by pressing the ENTER key Menus under SHIFT SET Speaker Setup SET 7 Preset DER Name 18 Zeichen Further menus 4 SET 8 11 Preset 1 ON Preset 2 ON Preset X As a system controller the DSC28 is configured with presets for one or several loudspeaker systems by the manufacturer A preset consists of frequency cross over GJ Chap 14 system equaliser L Chap 14 and limiter J Chap 15 The last selected preset is loaded auto matically after power up so that the system is immediately ready for operation provided that it is properly connected Q Chap 16 Before selecting any preset read the manual of the loudspeaker system carefully For systems comparable with the GAE Director and comprising Top Bass and additional Subbass 4 Way the memory of the DSC28 can hold up to approx 12 presets Enter Confirm your selection by pressing the ENTER key Chapter 7 3 SET 8 Input Select Further menus 4 SET 9 11 A SET 7 Analog ON Digital ON Dig Insert Select
71. ur The use of FIR filter technology allows the run time behaviour of the crossover to be separated from the amplitude frequency response A run time compensation time alignment between speaker components with acoustical centres on different vertical axis is included Limiter The limiter thresholds are selected respective of the power specifications of the connected loudspeaker components and power amplifiers Ei Chap 15 Output scaling Output scaling factors are an integral part of the parameter set These guarantee an optimum balance between the amplification of the analog output level of the DSC28 the amplification factor of the connected power amplifier the limiter thresholds and the highest possible digital resolution of the signal processing FIR filter technology The calculation of FIR filters Finite Impulse Response requires significantly more computer processing power when compared to digital devices operating with IIR filters Infinite Impulse response and as such merely simulating analog filters within the digital domain For this reason the available processing power is optimally matched to each of the paths frequency ranges except the HIGH path by down over sampling modules However this process requires that a bandwidth limitation of the useful frequency range of the SUB and LOW paths to approximately 1kHz and the MID path to approximately 4kHz be implemented FIR filtering allows the realisation of correction filters wit
72. us WV EQ 13 14 A EQ 11 0 LR HS12 07 5 00 0 EQ 13 EQ13 EQ13 Type Q Hz dB Further menus VW EQ 14 A EQ 12 0 LR ES12 07 40 0 EQ 14 EQ14 EQ14 Type Q kHz dB Further menus A EQ 13 0 LR HS12 07 5 00 0 Men s unter SHIFT EQ EQ 15 EQ Noise Shaper Select Te a P EQ Nois Shaper Select 1 Order Error Feedback Further menus keine None O 1 Order Error Feedback OM Lipsh 3 Taps Modified E OM Lipsh 5 Taps Improved E Re quantisation of the 48 Bit audio signal to 24 Bit for further proceedings Adjustments here are only audible at very low levels Chapter 8 3 Chapter 9 Menus under SYS System adjustments SYS 0 Display LEDs and VFD Sea Further menus V SYS 1 2 LEDs On VFD 100 Cursor position p 2 1 LEDs Off O On Here the LED display can be turned on or off The status LEDs remain on B 2 VFD 25 O 50 O 75 O 100 Select the desired brightness of the vacuum fluorescence display Midi Settings This menu runs over four pages and serves the adjustment of several parameters for the exchange of data via the serial or MIDI interfaces between several controllers or from a PC to one or seve ral controller s EQ Chap 19 SYS 1 0 Midi Settings S YS 5 Further menus V SYS 2 A SYS 0 Se oe gt SYS 1 1 Cursor position 12 1 Channel 1 ONN 16 Assign the controller address basic channels 1 16 at which data can sent and received Enter Confirm your selection with ENTE
73. y hitting the pole plate mainly cone loudspeakers or the breakage of the connecting wires of the voice coil Furthermore the task of suppression of disturbing distortions produced by the loudspeaker or by the clipping of the power amplifier is undertaken The basis for the loudspeakers admissible load is it s peak load capacity The characteristic variables of a peak limiter are the threshold at which the level reduction starts as well as three time constants the attack time at which the speed of the level reduction is defined the release time which defines the speed of releasing the gain reduction and the hold time which defines the period over which the level reduction is held after the threshold excess is ended RMS limiter RMS or thermo limiters prevent thermal overload particularly the burning of voice coils caused by constant too high a power supply The basis for the loudspeakers admissible load are the thermal permanent load capacity and two time constants a short one for the thermal capacity of the voice coil and a long one for the larger thermal capacity of the magnet material and the loudspeaker chassis The characteristic variables of an RMS limiter are the threshold at which level limitation starts and the emulation of the loudspeaker s thermal time constants Output gt Fullwave rectifier Analog peak limiter The block diagram above Fig 15 1 shows a peak limiter circuit found in most analog limiter

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