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Owner`s Manual Symetrix 602 Stereo Digital Processor

1. D 1 2 3 4 5 6 7 8 9 0 82 0 82 0 80 0 78 0 76 0 76 0 74 0 72 0 70 0 68 0 10 68 0 66 0 64 0 62 0 62 0 600 58 0 56 0 54 0 540 20 52 0 50 0 48 0 48 0 46 0 44 0 42 0 40 0 40 0 38 0 30 36 0 34 0 32 0 32 0 31 0 30 0 29 0 29 0 28 0 27 0 40 26 0 25 0 25 0 24 0 23 0 22 0 22 0 21 0 20 0 19 0 50 18 0 18 0 17 0 16 0 15 0 15 0 14 0 13 0 12 0 11 0 60 11 0 100 9 0 8 0 7 0 7 0 6 0 5 5 50 50 70 4 5 40 1 35 30 30 25 20 15 15 10 80 0 5 100 405 05 10 415 420 42 0 425 30 90 3 5 40 40 45 50 55 60 60 6 5 70 100 7 5 7 5 80 85 90 95 95 10 0 410 5 11 0 110 411 0 411 5 412 0 12 5 13 0 13 0 13 5 140 114 5 414 5 120 150 15 5 16 0 16 5 116 5 417 0 17 5 18 0 Table 28 Attn100 Table dB 0 1 2 3 4 5 6 7 8 9 0 100 100 98 0 96 0 94 0 94 0 92 0 90 0 88 0 86 0 10 86 0 84 0 82 0 80 0 80 0 78 0 76 0 74 0 72 0 72 0 20 70 0 68 0 66 0 66 0 64 0 62 0 60 0 58 0 58 0 56 0 30 54 0 52 0 50 0 50 0 49 0 48 0 47 0 47 0 46 0 45 0 40 44 0 43 0 43 0 42 0 41 0 40 0 40 0 39 0 38 0 37 0 50 36 0 36 0 35 0 34 0 33 0 33 0 32 0 31 0 30 0 29 0 60 2
2. MIDI Chan MIDI MIDI Input Into 1 2 Input into 1 2 input Into 1 2 Value Pan Atten Atten Value Pan Atten Atten Value Pan Atten Atten Buffer dB dB Buffer dB dB Buffer dB dB o 0 0 OFF 43 20 1 5 6 0 86 41 6 5 1 0 1 o 0 0 44 20 1 5 6 0 87 41 6 5 10 2 o 0 0 45 21 1 5 5 5 88 41 6 5 1 0 3 1 0 0 32 0 46 21 1 5 5 5 89 42 7 0 10 4 1 0 0 32 0 47 22 1 5 5 5 90 42 7 0 1 0 5 2 0 0 26 0 48 22 1 5 5 5 91 43 7 5 1 0 6 2 0 0 26 0 49 23 1 5 5 0 92 43 7 5 1 0 7 3 0 0 22 0 50 23 1 5 5 0 93 44 8 0 1 0 8 3 0 0 22 0 51 24 2 0 5 0 94 44 8 0 1 0 9 4 0 0 20 0 52 24 20 5 0 95 45 8 5 05 10 4 0 0 20 0 53 25 2 0 4 5 96 45 8 5 05 11 5 0 0 18 0 54 25 2 0 4 5 97 46 9 0 05 12 5 0 0 18 0 55 26 2 0 4 0 98 46 9 0 05 13 6 0 0 160 56 26 2 0 4 0 99 47 95 0 5 14 6 0 0 16 0 57 27 2 5 40 100 47 95 05 15 7 0 0 15 0 58 27 2 5 4 0 101 48 10 0 0 5 16 7 0 0 15 0 59 28 2 5 3 5 102 48 10 0 05 17 8 0 0 14 0 60 28 2 5
3. Attack Ed Buf 52 57 Threshold Release Ed Buf 51 56 Ed Buf 53 58 MIDI __ Ratio MIDI Knee Ed Buf 54 59 ed 65 Ed Buf 55 60 pro Ed Buf 66 compresor Level Level Determined Input 1 9 petermined Selector Selector She Filter w Switcher Witcher Lo Ti o Converter Conant Signal with Signal with Attack Maximum zx 1 Mr Input 2 MRY Selected ES ilter ened Release in Selected Knee Ratio Ed Buf 49 Ed Buf 48 MIDI d Ed Buf 67 Digital Controlled Expander Amplifiers Auto Release PeakRelease k Auto Release gt Output 1 Ed Buf 62 e rs E MIDI __ Si Noi 9 Threshold reso Delay ON Ed Buf 64 Output 2 Auto Relea se Hold WP Circuitry Digre Controlled ttenuators T d Sura i Rev G Figure 7 4 Dynamics block 7 3 4 De Ess and Noise Reduction Block NR system uses a common control chain for both channels Q The De ess system uses a common control chain for both channels MIDI edit buffer parameter numbers are shown in parenthesis level MAUDE D ESS Threshold ch V Ed Buf 1 Ed Buf 24 osre Switcher pner mipi Ed Buf 23 Ed Buf 25 HP Signal with Maximum Time Level Level LP Constant Selected Detector Q Filt idi Output 1 Release HP NR Absolute PN ind 32 ON OFF Threshol
4. 0 11 2 3 4 5 6 7 8 9 fo 6 6e je e je 6 e 6 je 10 6 6 6 6 6 6 6 6 6 6 20 6 6 6 6 6 6 6 6 6 6 30 6 6 12 12 12 12 12 12 12 12 40 12 12 12 12 12 12 12 12 12 12 50 12 12 12 12 12 12 12 12 12 12 60 12 12 12 12 18 18 18 18 18 18 70 18 18 18 18 18 18 18 18 18 18 80 18 18 18 18 18 18 18 18 18 18 90 18 18 18 18 18 18 24 24 24 24 100 24 24 24 24 24 24 24 24 24 24 110 24 24 24 24 24 24 24 24 24 24 120 24 24 24 24 24 24 24 24 Table 38 Makeup Gain Table Attn24 Tw h Ja 0 0 0 0 0 0 0 0 5 0 5 0 5 1 0 1 0 1 5 15 10 1 5 2 0 2 0 2 0 2 5 2 5 3 0 3 0 3 0 35 20 3 5 4 0 4 0 4 0 4 5 4 5 4 5 5 0 50 55 30 5 5 5 5 6 0 6 0 6 5 6 5 6 5 7 0 7 0 70 40 7 5 7 5 8 0 8 0 8 0 8 5 8 5 8 5 9 0 9 0 50 9 5 9 5 9 5 410 0 10 0 10 5 105 105 11 0 11 0 60 11 0 11 5 11 5 120 5120 120 125 125 13 0 13 0 70 413 0 13 5 13 5 13 5 140 140 145 145 145 15 0 80 415 0 15 5 15 5 15 5 160 160 160 165 16 5 17 0 90 17 0 5170 17 5 417 5 17 5 180 18 0 418 5 185 185 100 419 0 19 0 19 5 19 5 19 5 200 420 0 200 20 5 20 5 110 421 0 210
5. C 10 AGG PARAMETERS etate di Der te exe EI en da DR ut e ao e e ae C 10 ARM SENSE PARAMETERS C 11 LOG CONVERTER PARAMETERS tiic a te v d eed e ere Da de Ln T n totes C 11 OUTPUT TE OE CE OMO C 11 REALTIME MIDI BLOCK vis reti at eo A dai ee maven C 12 REALTIME MIDI 2 C 13 6 aie n a beet POR d C 13 ATINI FABLE DB s ri traut ooo danse EY E SERERE si EE ERE zie C 14 ATINS2 TABEE DB 5 intor rd Anette rade n emit C 14 ATINTOO TABLE DB 3 2 hera ts C 14 PARAMETRIC BANDWIDTH TABLE IN C 15 FREQUENCY TABLE HZ niea E deans Shanda NOYAU de C 15 OUTPUT LEVEL TABLE DB ii eet eec ep ee reinen tes C 15 EXPANDER RATIO 22i ri erae e t d rude C 16 COMPRESSOR RATIO GABLE cesset ches vocet tel cta vec vta Du Ua POR DECRE QU C 16 oic E ioi E C 16 ARM THRESHOLD DB is rue En tt edite ne ebd Heim t ee C 17 TIME 6 aaa C 17 COMPRESSER EXPANDER KNEE TABLE IN DB nana C 17 MAKEUP GAIN TABLE 24 nana C 18 SIDECHAIN LOOKAHEAD
6. tes 4 17 4 11 Restoring Factory 0 4 19 4 12 Disabling the Front Patel e 4 19 5 Rear Panel Overview LEE Ear rr ere 5 1 6 Fast First Time 1 6 1 6 1 6 EEE tA dc e dM Ond 6 1 6 2 Settings for Analog 6 2 6 3 Settings for Digital 6 4 7 Using 602 trn ari nae ears s ark cras n Yoon son ac oru qaa 7 1 7 1 7 2 Operational Details s cesso escoger eee qu ner 7 1 7 2 1 Stand alone Operation su Ave Ane ee eee t racio AA ae 7 2 7 3 Block Diagrams v so Zoe 7 4 7 3 1 Overall Block Diagram aoi eee vue 7 4 7 3 2 Sequence of Processing taza ccc ch TOR me Road I VSE 7 4 ZS A DYNAMICS BD IOGK EM ge c 7 4 7 3 4 De Ess and Noise Reduction 7 5 Tao Delay em MEL EE 7 6 7 4 System Interface a vaso aie ten sin qued ner 7 6 7 4 1 Using the 602 as a Channel Insert Device 7 6 7 4 2 Using the 602 in a Send Receive 7 6 7 4 3 Using the 602 as an A D 7 7 7 4 4 External Sample Rate 7 7 7 4 5 Input Outp r Glock
7. eaae eee eee ee 7 7 7 4 6 MIDI Programming etes A tnt Ed ete 7 8 7 4 7 Accessing Parameters via MIDI 7 8 7 4 8 Realtime D ER 7 8 FAD Program 7 8 7 4 10 Editing Parameters not Accessible from the Front Panel 7 9 7 5 Tips and Techniques for Using the 602 7 10 7 5 1 Recalling and Storing 7 10 ii 7 52 ES 7 10 7 5 3 Gain Setting ease 7 11 TIA ME DEC 7 11 7 5 5 Metering and the Dynamics Block 7 12 7 5 6 Dynamic Noise Reduction 7 12 oc edat Pc Adige 7 12 Z O GOmpressiOlT c meat i dan cornes 7 13 7 5 9 AGC 7 13 7 5 10 Downward Expander oce e etse dad alice decal again de eae 7 13 EON E eh E 7 14 T DM ECHO CLOG Sats 7 14 A52 Flangini os cei Questa 7 14 FDA TS EE 7 15 BAP Pll CAO DR 8 1 8 1 Broadcast Voice Processing 8 1 8 2 Voice over Processing tete idee etr de ka gc 8 1 8 3 Foley Processing 8 1 8 4 Digital Mastering n tne ena
8. GAIn Sets digital input gain over a 18 dB range before any digital processing modules The gain setting is saved on a per program basis InP Input selector Rotating the Wheel selects the input source as indicated by the Digital CH1 Stereo CH2 LEDs The input source may be the digital input AES EBU or S PDIF the line inputs or the mix of the line inputs The Digital In Sync LED flashes if the digital input signal is missing or defective Each input source may be routed to the inputs of the DSP as depicted by the display In stereo mode 1 2 gain controls for the two input channels may be split one channel per control or ganged two channels on one control by further rotating the Wheel after the LED display indicates 1 2 Display Shows Input routes to both outputs 2 Input 2 routes to both outputs 12 and input 2 are mixed and routed to both outputs Stereo The two inputs are separate Further rotation of the wheel gangs the two gain controls onto the Channel 1 control and only the Ch1 Stereo LED indicator illuminates Rev 1 1 11 15 94 4 14 ev 1 1 11 15 9 bAr1 bAr2 CLCI CLCE nP Prt 44 1 48 0 As you rotate the Wheel the 602 cycles through the line and digital inputs For each input the 602 cycles through the four routing options shown in the table For analog sources inputs 1 and 2 are the left and right channels respectively of the digital to analog
9. Transient signals which are shorter than the attack time of the device will not be affected by the gain reduction so it is important that the attack time be as short as possible TAD See also release time Bandpass Filter A bandpass filter is a filter which has a bandwidth Bandpass filters can be broad having a wide bandwidth or narrow having a narrow bandwidth They may be fixed in frequency and bandwidth or variable in either frequency and or bandwidth TAD Bandwidth The bandwidth of a bandpass filter is the upper cutoff frequency minus the lower cutoff frequency It is thus the extent in Hertz of the frequency range or band passed by the filter Bandwidth is literally a frequency span and is not necessarily connected to the specification of a filter For instance the human voice can be transmitted with good intelligibility if the frequency response of the transmission chain extends from about 100 Hz to about 3000 Hz Thus a 2900 Hz bandwidth is needed to transmit voice This is about what a standard telephone system attains The audio bandwidth however is generally considered to be about 20 kilohertz TAD Brickwall Filter Some lowpass filters have such a steep cutoff slope that the graph of the slope resembles a brick wall the slope of the sides being vertical is infinitely steep Brickwall filters are commonly used for anti aliasing and anti imaging TAD Chorus An electronic music effect that modifies the sound
10. 2 4 9 Peaking or Shelving The 602s equalizer can operate in either peaking or shelving mode The two terms refer to the overall shape of the equalizer s frequency response curve In Figure 2 3 you can see that the peaking equalizer s effect is concentrated at one frequency the center frequency with progressively less effect above or below the nus center frequency The shelving equalizer which acts more or less like the tone controls on a home stereo affects frequencies above or below its characteristic frequency depending on whether we re talking about a low frequency shelving equalizer or a high frequency shelving n equalizer AUDIO PRECISION AMPL dBu vs FREQ Hz Shelving EQ 30 00 Figure 2 3 Shelving and peak dip EQ curves At very narrow bandwidths small number peaking equalizers exhibit a phenomenon known as ringing This quite aptly describes the effect of the equalizer being sharply resonant at its center frequency which makes it almost oscillate In general use the shelving curves to create overall color changes to the entire signal and use the peaking curves to modify specific regions of the signal The peaking curves bring another variable into play bandwidth or Q as it is sometimes known The bandwidth parameter simply tells you how much of the region surrounding the center frequency will be affected
11. 21 0 21 5 215 920 22 0 22 0 225 225 120 22 5 23 0 23 0 23 5 23 5 23 5 24 0 24 0 Table 39 Sidechain Lookahead Time ms Table 0 1 2 3 4 5 6 7 8 9 0 0 00 0 02 0 04 0 06 __ 0 08 0 10 0 12 0 15 0 17 0 19 10 0 21 0 23 19257 027 1029 031 0 33 085 057 0 40 20 0 42 044 0 46 __ 0 48 0 50 0 52 0 54 0 56 0 58 0 60 30 0 62 0 65 10 67 0 69 0 71 0 73 0 75 1077 079 0 81 40 0 83 0 85 __ 0 87 10 90 10 92 0 94 0 96 0 98 1 00 1 02 50 1 04 1 06 1 08 1 10 1 12 1 15 1 17 1 19 1 21 1 23 60 1 25 1 27 1 29 1 31 1 33 1 35 1 37 1 40 1 42 1 44 70 1 46 1 48 1 50 1 52 1 54 1 56 1 58 1 60 1 62 1 65 80 1 67 1 69 1 71 1 73 1 75 1 77 1 79 1 81 1 83 1 85 90 1 87 1 90 1 92 1 94 1 96 1 98 __ 2 00 2 02 2 04 2 06 100 206 2 10 212 215 2 17 219 2 21 2 23 2 25 2 27 110 2 29 2 31 2 33 285 1287 240 242 2 44 _ 2 46 2 48 120 2 50 2 52 2 54 2 56 12 58 12 60 2 62 2 65 1 Table 40 Delay Time Table ms C 18 0 1 2 3 4 5 6 7 8 9 0 0 5 1 0 1 5 2 0 2 5 3 0 3 5 4 0 4 5 so 10 6 7 8 9 10 11 12 13 14 15 20 16 17 18 19 20 21 22 23 24 25 30 26 27 28 29 30 32 34 36 38 40 40 42 44 46 48 50 52 54 56 58 eo 50 62 64 66 68 70 72 74 76 78 so 60 82 84 86 88 90 92 94 96 98 100 70 105 110 115 120 125 130 135 140 145 150 80 155 160 165 170 175 180 1
12. 64 2 where v stored value 0 127 and offset 128 to 127 The value applied to the edit buffer is derived as follows e 0 where e new edit buffer value m modulation value v 7 stored offset stored scaling value from table Note in the 602 the scale factor is shown as SCAL on the display For the purposes of Realtime MIDI each edit buffer parameter has a range of 0 127 regardless of what the actual range of values is as specified in the tables at the end of this appendix Thus an on off type of parameter will be off if the result of the offset and scale operation ranges from O to 63 and on if the result is in the range of 64 127 Realtime Block 1 has two additional parameters that apply an upper CLPH clip hi and lower CLPL clip lo limit to the final parameter value You can use these parameters to keep realtime MIDI value within a useful range Realtime Block 2 has an additional modulation source bLC1 that is the output of Realtime Block 1 As each parameter change request arrives it immediately modifies the appropriate edit buffer location and inserts a DSP update request into a 128 byte FIFO queue The 602 processes the queue as time allows If a burst of requests fills the queue new requests are discarded until there is room in the queue If your MIDI controller spews data and overruns the 602 s queue the 602 may ignore the extra data If the stream of data ends before the 602 finishes processing
13. 7 0 6 5 6 5 6 0 5 5 5 0 4 5 4 0 4 0 3 5 90 3 0 2 5 2 0 1 5 1 5 1 0 0 5 0 0 0 5 0 5 100 1 0 1 5 2 0 2 5 3 0 3 0 3 5 4 0 4 5 5 0 110 5 5 5 5 6 0 6 5 7 0 7 5 8 0 8 0 8 5 9 0 120 9 5 10 0 11 0 11 0 12 0 13 0 14 0 15 0 C 15 Table 32 Expander Ratio Table 0 1 2 3 4 5 6 7 8 9 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 10 10 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 2 12 20 1 5 1 5 1 5 1 5 1 5 1 5 1 5 1 5 1 5 15 30 1 8 1 8 1 8 1 8 1 8 1 8 1 8 1 8 1 8 18 40 2 0 2 0 2 0 2 0 2 0 2 0 2 0 2 0 2 0 20 50 2 5 2 5 2 5 2 5 2 5 2 5 2 5 2 5 2 5 25 60 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 5 70 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 40 80 4 0 4 0 4 0 4 0 4 0 4 0 4 0 4 0 4 0 50 90 5 0 5 0 5 0 5 0 5 0 5 0 5 0 5 0 5 0 eo 100 6 0 6 0 6 0 6 0 6 0 6 0 6 0 6 0 6 0 70 110 70 7 0 7 0 7 0 7 0 7 0 7 0 7 0 7 0 8 0 120 80 80 80 80 80 80 80 Table 33 Compressor Ratio Table 0 1 2 3 4 5 6 7 8 9 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 2 10 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 5 1 5 20 1 5 1 5 1 5 1 5 1 5 1 5 1 8 1 8 1 8 1 8 30 1 8 1 8 1 8 1 8 1 8 2 0 2 0 2 0 2 0 2 0 40 2 0 2 0 2 0 2 5 2 5 2 5 2 5 2 5 2 5 2 5 50 2 5 2 5 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 0 60 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 4 0 70 4 0 4 0
14. MACH ENTER SETUP ENTER EDIT slider 1 control the event type page page page page slider 2 control the second parameter of the Real time page page page page page B 2 BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR Displays MACH 4 GMIDI SETUP SOURCE DEST SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slrl SOURCE DEST OUT SOURCE DEST DEFINE SYSEX BYTES 2 3 DEFINE SYSEX Data to Set Enter 15 10 see Real time MIDI Block 1 slrl sysex 1 FO 00 00 5E 01 00 1 47 slr2 SYSEX FO 00 00 5E 01 00 1 48 slr2 Comments set to machine 15 set to setup 10 via slider 1 or keypad now edit the setup source to slider 1 dest to SYSEX midi out 1 next page byte 1 can t change midi SYSEX mfrIDO mfrID1 next page mfrID2 device type unit channel edit buffer data set edit buffer 71 send slider setting use sliders 1 4 to set label to T Y MIDI block now do slider 2 use button 1 to change source use slider 2 to set dest to SYSEX next page SYSEX mfrIDO mfrID1 next page mfrID2 device type unit channel edit buffer data set next page edit buffer 72 send slider setting next page Rev 1 1 11 15 94 STEP MRCKey MRC Displays Data to Set Enter 38 Make slider 3 con
15. Tackling these terms in reverse order noise floor represents the signal level of the device s residual noise level Realistically this is somewhat lower than the lowest signal level that you d want to process unless you want the output to sound noisy Headroom is the difference between the average signal level and peak clipping Peak clipping occurs because the processor can t increase its output to follow the signal When this occurs the output signal simply flat tops over the period that it can t follow its input sort of like clipping the tip of the peak off with a hedge trimmer to level it off Audibly speaking clipping and using the hedge trimmer are about equivalent Dynamic range is the difference between the highest signal that may pass limited by peak clipping and the lowest signal that will pass limited by the noise floor In a digital processor a O dB signal may output a 120 dB noise floor but the smallest signal that may be represented by 18 bits is a 108 dB square wave because there is only one bit to toggle on and off Somewhere between these two points is the average level of your signal as set by the processor s level control Set the average level too high and peak clipping will smash your peaks flat set it too low and suddenly the noise floor becomes audible but you ve got lots of headroom Rev 1 1 11 15 94 2 1 The 602 allows you to set the signal levels in three different locations which allows you to make
16. 3 5 103 49 11 0 00 18 8 0 0 14 0 61 29 3 0 3 5 104 49 11 0 00 19 9 0 0 13 0 62 29 3 0 3 5 105 50 12 0 o o 20 9 0 0 13 0 63 30 3 0 3 0 106 50 12 0 00 21 10 0 0 12 0 64 30 3 0 3 0 107 50 12 0 22 10 0 0 12 0 65 30 3 0 3 0 108 51 13 0 o o 23 10 0 0 12 0 66 31 3 5 3 0 109 51 13 0 24 11 0 0 11 0 67 31 3 5 3 0 110 52 14 0 00 25 11 0 0 11 0 68 32 3 5 2 5 111 52 140 00 2 12 0 5 10 0 69 32 3 5 2 5 112 53 15 0 00 27 12 0 5 10 0 70 33 40 2 5 113 53 150 o o 28 13 0 5 9 5 71 33 40 2 5 114 54 16 0 00 2 13 0 5 9 5 72 34 40 2 0 115 54 16 0 0 0 30 14 0 5 9 0 73 34 40 2 0 16 55 18 0 00 14 0 5 9 0 74 35 4 5 2 0 117 55 18 0 00 32 15 0 5 8 5 75 35 4 5 2 0 118 56 20 0 0 0 33 15 0 5 8 5 76 36 5 0 2 0 119 56 20 0 00 34 16 1 0 8 0 77 36 5 0 2 0 120 57 22 0 00 35 16 1 0 8 0 78 37 5 0 1 5 121 57 22 0 00 36 17 1 0 7 5 79 37 5 0 1 5 122 58 26 0 o o 37 17 1 0 7 5 80 38 5 5 1 5 123 58 26 0 00 38 18 1 0 7 0 81 38 5 5 1 5 124 59 32 0 o o 39 18 10 7 0 82 39
17. 8 1 8 5 Musical erii uan Rute qtu 8 1 8 6 Sound Reinforcement 8 1 9 Troubleshooting Chart EA diana rna cusa 9 1 10 602 Stereo Digital Processor Limited Warranty 10 1 11 Repair ERE rn 11 1 T3 ROUTE NUTRI e eroe eer itu esto ees e oie pon 11 1 11 2 la Weartanty Repairs c st Aa ete peri iot ee AN pen io enc 11 1 11 3 Out of Warranty 11 1 12 Specification vin enka ni ea exin ri cw o ina a rue 12 1 A Editing Realtime Midi Settings esee A 1 A 1 Realtime MIDI Example iv ab A 2 Using the Lexicon MRC to Edit Realtime MIDI Settings B 1 C MIDI Implementation Notes nana C 1 AS AN ONG IV HTTP C 1 C 1 1 Control Change Br uide NAL C 1 C 1 1 1 Example SEC C 1 Ce Realtime MIB eu tine C 1 C 1 3 Sysex Implementation FO nn C 2 SySOX apart eed C 3 1 5 Recognized MIDI Commands nn C 4 C 1 6 Data Structure Per Prog elles tee rec SR REPE ER Erde C 8
18. C 1 7 MIDI Parameter Tables C 14 C 2 Hexadecimal Conversion Tables C 22 D Glossary and D 1 RM RE D 1 BE Z BIDNOGVAPNY oes 0 11 Architect s and Engineer s Specification 1 Disassembly Instructions nana 1 1 Top Cover Removal s F 1 F 2 Gircuit Board REMOVAL sce F 1 F 2 1 Analog Board Removal F 1 E 2 2 Digital Board REMOVAL suse iita ied t it d a F 2 F 2 3 Power Supply Board Removal F 2 E244 Front Panel Board Removal s ien Ec Ree RT eerta F 2 XLR Connector Removal F 2 Presets and Other Stuff G 1 G 1 602 Programme s Worksheet ror chat ty os tritt Preise G 2 G 2 Midi Implementation Chart nte etr ertet nie hn nean G 3 G 3 Presets and Building BlOCKS oe peti G 4 TABLE 1 TABLE 2 TABLE 3 TABLE 4 TABLE 5 TABLE 6 TABLE 7 TABLE 8 TABLE 9 TABLE 10 TABLE 11 TABLE 12 TABLE 13 TABLE 14 TABLE 15 TABLE 16 TABLE 17 TABLE 18 TABLE 19 TABLE 20 TABLE 21 TABLE 22 TABLE 23 TABLE 24 TABLE 25 TABLE 26 TABLE 27 TABLE 28 TABLE 29 TABLE 30 TABL
19. TAD A noise gate is an extreme example of a downward expander Group Delay The slope of the phase versus frequency curve of a frequency response function that is the rate of change of phase of the response as a function of frequency Group delay is a property of a device or a system A pure time delay equal at all frequencies gives a constant slope of phase versus frequency If in an audio component this slope is not constant but varies with frequency the component is said to produce group delay distortion This is equivalent to a time delay that varies with frequency For instance an anti aliasing filter will typically have a phase response curve which slopes sharply down at high frequencies which means that the high frequency components will be delayed longer in their passage through the filter The audible result is a loss of precision in musical transients they are spread out or smeared in time and a more diffuse stereo image results TAD Highpass Filter A highpass filter uniformly passes signals above a certain frequency called the cutoff frequency The cutoff frequency is where the filter response is 3 decibels below the nominal response The response rolloff in the stopband may be gradual or sharp The rumble filter found in many record player systems is a highpass filter TAD See also lowpass filter Impedance In an electric circuit containing direct current the magnitude of the current is determined by the voltage
20. You into mic Hello 602 Hello Hello Experiment with different delay times What does it sound like when the delay time is quite short say around 10 ms What does it sound like when the delay time is mid range say 40 to 80 ms Now experiment with different mix settings Listen in stereo and make the two delay times slightly different Now try making them radically different Try using duAL mode to sweep the different delay times You create repeating echoes by recirculating the output of the delay line back to its input On the 602 set the FEEDBACK to P 10 set the DELAY time to 330 ms Now speak into the mic You Hello 602 Hello Hello Hello Hello Hello Hello Higher feedback settings increase the number and duration of the echoes Be sure that you try varying the wet dry mix as well as the feedback and delay times 7 5 11 2 Flanging Its Audio history time The term flanging came about because the effect was originally created by using two three head tape recorders 30 years ago that was how we created delay inputs paralleled outputs mixed Then the engineer held his thumb on the reel flange of one machine to slow it down slightly which changed the time delay Varying the pressure on the reel flange changes the effect That s more or less what happened when The Small Faces made Itchykoo Park about 25 years ago Flanging is nothing more than comb filtering The modulation oscillator replaces the thumb on the reel
21. a good case for a downward expander Some microphones having a rise at the higher frequencies especially omni microphones benefit from some attenuation in this region Those microphones having underdamped diaphragms may ring at these frequencies causing an annoying sibilant distortion on speech On musical forms using hand percussion boosting this range frequently results in an astonishing and pleasing feeling of clarity 2 4 7 Conclusions When the article containing the above excerpts was written probably around 1963 stereo was just becoming a commercial reality you could still purchase mono and stereo versions of an LP and there were still more FM stations broadcasting in mono than stereo and as many mixers contained rotary mix pots as those that used slide pots The value of individual channel equalization was known but it was both technologically and financially prohibitive The article concludes thusly With the advent of stereo and three channel recording nearly three times the equipment with more elaboration seems indicated and expansion of console area in the horizontal plane offers the only direction in which to proceed But a single engineer has arms only so long How times have changed 2 4 8 Equalizing for Speech In broadcast equalizers are often used to create a sonic personality for the station s on air personalities In the past this has often meant using a single non programmable equalizer in the announce
22. lb 1c 1d 30 le if 20 21 22 23 24 25 26 27 40 28 29 2 2c 2d 2e 2f 30 31 50 32 33 34 35 36 37 38 39 3a 3b 60 3c 3f 40 41 42 43 44 45 70 46 47 48 49 4a 4b 4c 4d 4f go 50 51 52 53 54 55 56 57 58 59 90 5b 5c 5d Se Sf 60 61 62 63 400 64 65 66 67 68 69 6b 6c 6d 110 6 70 71 72 73 74 75 76 77 120 78 79 7a 7b 7c 7d 7e TF Rev 1 1 11 15 94 D Glossary and Bibliography Some terms used in this manual may not be familiar to you Their definitions are presented in the following glossary At the end of this chapter you will find a short bibliography which is a good starting point for further research Many of the glossary items and their definitions are taken with permission from The Audio Dictionary by Glenn White In the interests of brevity some of Glenn s definitions have been abridged and Glenn s extensive cross referencing has been removed Definitions taken from this book have the notation TAD appended to the definition A more complete bibliographic entry for this book may be found in the bibliography D 1 Glossary In this glossary words typeset as follows digital are cross references to other words in this glossary Glossary entries followed by TAD may be found in The Audio Dictionary Anal
23. 0 6dB 43 12 dB 85 18 dB 127 24 dB Table 18 Compression Parameters Offset Description Range Reference dec hex dec 50 32 Compressor Mode 0 out 64 AGC 127 compressor 51 33 Threshold See Attn100 Table 52 34 Attack Time See Tc Table 53 35 Release Time See Tc Table 54 36 Compression Ratio See Compression Ratio Table 55 37 Knee Control 0 6dB 43 12 dB 85 18 dB 127 24 dB 83 53 Makeup Gain 0 127 Shared with AGC 0 auto See Attn24 Table 1 0 dB makeup gain 127 24 dB makeup gain Table 19 Parameters Offset _ Description Range Reference hex dec 56 38 Threshold See Attn100 Table 57 39 Attack Time See Tc Table 58 Release Time See Tc Table 59 3B Compression Ratio See Compression Ratio Table 60 3C Knee Control 0 43 12 dB 85 18 dB 127 24 dB 83 53 Makeup Gain 0 127 Shared with compressor 0 auto See Attn24 Table 1 0 dB makeup gain 127 24 dB makeup C 10 Rev 1 1 11 15 94 Table 20 ARM Sense Parameters Offset _ Description Range Reference dec 61 3D Auto Release Threshold See Attn100 Table 62 3E ARM Peak Release Tc See Tc Table 63 3F ARM Integration Tc See Tc Table 64 40 ARM Threshold See ARM Threshold po cM eu Table 21 LOG Converter Param
24. 1 frequency attached to oscillator Filter 30Hz shelf 200Hz shelf 500Hz shelf Look ahead compressor with signal delayed Maximum look ahead compressor with signal delayed Minimum Low 1 0 Medium 5 High 3 Medium 50 Low 60 High 54 Medium 66 Low 74 Long P 8 P 14 P 20 P 70 120 Long P 8 P 14 P 20 P 70 96 Medium P 8 P 14 P 20 P 70 58 Fast P 8 P 14 P 20 P 70 31 Hard Knee Compressor Hard Knee Expander Medium Knee Compressor Hard Knee Expander Soft Knee Compressor Hard Knee Expander Hard Knee Compressor Medium Knee Expander Rev 1 1 11 15 94 RAM 122 123 124 125 126 127 128 ROM 250 251 252 253 254 255 256 Name Dynamics Preferences Dynamics Preferences Dynamics Preferences Dynamics Preferences Dynamics Preferences Dynamics Preferences Initialization Program Description Hard Knee Compressor Medium Knee Expander Medium Knee Compressor Medium Knee Expander Soft Knee Compressor Medium Knee Expander Hard Knee Compressor Soft Knee Expander Medium Knee Compressor Soft Knee Expander Soft Knee Compressor Soft Knee Expander Used to zero any user program 1 127 in the 601 for creating new programs or for starting over Note Program 256 works by overwriting the selected RAM location 1 127 with a set of rational settings Use this program to create a fresh starting point for a program of your own or for when one of your programming efforts turns into Godzilla
25. 11 kHz with a bandwidth of 05 to 3 octaves with a boost cut range of 15 dB to 50 dB The delay block shall provide two delays capable of up to 330 milliseconds of delay The delays shall be user and MIDI programmable The feedback path for delay recirculation shall be cross coupled between the two delays and the delay time shall be capable of accepting modulation either from an internal random number generator or from an internal sine or triangle wave source The delay time shall be independently adjustable for each delay and provision shall be made to allow adjusting the delay times simultaneously while maintaining an offset in the delay times The dynamics block shall provide the following functionality De ess Dynamic noise filter Compressor AGC Leveler and Downward Expander Within the dynamics block all sections are user and MIDI programmable and each dynamics function shall provide the following features De Ess High ratio compression driven by a high frequency selective sidechain The de esser shall provide a threshold control for user adjustment Dynamic Noise Filter Sliding high frequency rolloff controlled by the HF energy content of the input signal The DNF shall provide threshold and frequency controls Compressor Compression up to 10 1 ratio The compressor shall provide threshold ratio attack and release controls The compressor characteristic shall be changeable between a hard knee curve and a soft knee curve AG
26. 4 0 4 0 4 0 4 0 4 0 5 0 5 0 5 0 80 5 0 5 0 5 0 5 0 5 0 5 0 6 0 6 0 6 0 6 0 90 6 0 6 0 6 0 6 0 7 0 7 0 7 0 7 0 7 0 7 0 100 7 0 7 0 7 0 8 0 8 0 8 0 8 0 8 0 8 0 8 0 110 8 0 9 0 9 0 9 0 9 0 9 0 9 0 9 0 9 0 9 0 120 10 0 10 0 10 0 10 0 10 0 10 0 10 0 10 0 0 1 2 3 4 5 6 7 8 9 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 10 10 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 2 1 2 12 20 1 5 1 5 1 5 1 5 1 5 1 5 1 5 1 5 1 5 15 30 1 8 1 8 1 8 1 8 1 8 1 8 1 8 1 8 1 8 18 40 2 0 2 0 2 0 2 0 2 0 2 0 2 0 2 0 2 0 20 50 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 22 60 2 5 2 5 2 5 2 5 2 5 2 5 2 5 2 5 2 5 28 70 2 8 2 8 2 8 2 8 2 8 2 8 2 8 2 8 2 8 3 0 80 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 0 3 0 32 90 3 2 3 2 3 2 3 2 3 2 3 2 3 2 3 2 3 2 3 5 100 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 3 5 38 110 3 8 3 8 3 8 3 8 3 8 3 8 3 8 3 8 3 8 40 120 40 40 40 40 40 40 40 40 C 16 Rev 1 1 11 15 94 Table 35 ARM Threshold dB 0 1 2 3 4 5 6 7 8 9 0 OFF OFF OFF OFF 0 1 0 1 0 1 0 1 0 2 0 2 10 0 2 0 2 0 4 0 4 0 4 0 4 0 6 0 6 0 6 0 6 20 0 8 0 8 0 8 0 8 1 0 1 0 1 0 1 0 1 2 1 2 30 1 2 1 2 1 5 1 5 1 5 1 5 1 8 1 8 1 8 1 8 40 2 0 2 0 2 0 2 0 2 2 2 2 2 2 2 2 2 5 2 5 50 2 5 2 5 2 8
27. 5 7 8 ARM Signal Noise threshold The AGC uses a signal noise detector ARM or Auto Release Monitor to decide when to allow the compressor s gain reduction to recover to it s no signal value The detector uses the peak to average ratio of the signal to decide whether the signal is noise or not noise Raising the ARM signal noise threshold edit buffer 64 causes the detector to reject signals lacking much peak content Lowering the threshold makes the detector less picky eventually allowing noise to pass as signal 4 5 7 9 Log converter time constant The dynamics section s log converter converts the audio signal into a logarithmic representation of its signal level The log converter time constant is a simple time constant at the output of the log converter The time constant sets a minimum attack and release time for any signal Some smoothing is necessary to prevent the compressor or other dynamics processor from trying to follow the envelope of low frequency signals Edit buffer 66 controls the time constant 4 5 7 10 Lookahead delay time Overshoot is a problem with any compressor that is caused by the control signal arriving at the gain controlled element VCA in analog units after the leading edge of the audio signal A simple remedy for overshoot is to slightly delay the audio before it gets to the gain controlled Rev 1 1 11 15 94 4 11 element in essence giving the compressor time to think Edit buffer 67 controls the loo
28. 6 32 x 1 2 inch screws from the IEC power connector at the rear panel Disconnect the green chassis ground wire at the chassis by removing the nut securing it to the chassis stud 6 Slide the power supply board towards the front of the unit then lift it clear of the chassis m F 2 4 Front Panel Board Removal 1 Disconnect all of the ribbon wire connectors from the digital board 2 Loosen the setscrew on the Wheel and remove it 3 Rotate the two gain controls until you can see the two setscrews on the shaft coupler near the circuit board mounted potentiometer Loosen the screw located towards the front of the unit and slide the knob and shaft out of the chassis Repeat for the remaining gain control 4 Remove the five button head screws securing the front panel Remove the front panel 5 Remove the five 6 32 x 1 4 inch screws securing the sub front panel to the chassis 6 Lift the front panel board clear of the chassis F 3 XLR Connector Removal Important The XLR connectors must be disassembled prior to removing either the analog or digital boards from the chassis After disassembly the connector body remains riveted to the chassis and the connector body remains with the circuit board Unlock the XLR connectors by inserting a 2mm slot head screwdriver into the hole located between pins 1 and 2 of the XLR connector insert Twist the retaining lug CCW to unlock the connector body When separating the connector bo
29. Analog Outputs Digital Outputs Maximum input level Maximum output level Filter Block Shelving Characteristic Peak Dip Bandwidth Maximum boost cut Delay Block Effects Delay time Lowpass frequency Modulation Depth Dynamics Block Types Compression ratio max Expansion ratio max Attack time Release time DS De Ess NR dynamic noise reduction Output Processing Types Rev 1 1 11 15 94 XLR female 12 5 kilohms line level balanced bridging Two XLR female and RCA Female AES EBU or S PDIF Two 300 ohm source impedance balanced XLR male Two XLR male and RCA Female AES EBU or S PDIF 22 dBu 21 5 dBu Three band parametric equalizer 31 Hz to 21 11 kHz Baxandall approximation 0 05 to 3 octaves 18 dB boost 50 dB cut Echo generation with filtered feedback distance simulation flanging or chorusing 0 5 ms to 330 ms 600 18 kHz Random sine wave or triangle wave 0 100 De essing dynamic noise reduction downward expansion compression AGC leveling 10 1 1 8 100 microseconds to 10 000 ms 100 ms to 10 000 ms High ratio limiter driven by sibilance content Sliding low pass filter driven by high frequency energy content Level and pan 12 1 Performance Data Frequency Response Distortion THD Dynamic Range Sample Rates Converter Type Conversion method Parameter Storage Group Delay Input Headroom Display Analog Input Cl
30. Bandwidth and Q are inversely related that is a wide bandwidth large number corresponds to a low Q small number 2 5 De Essing De essing is the process of removing S sounds from speech or singing The technique was originally developed for motion picture dialogue recording when it was discovered that speech sounded more natural when the accentuation of sibilants s sounds was reduced By sensing and limiting certain frequencies the de esser is intended to provide more specific control over some of the higher frequency vocal sounds that tend to become overemphasized Most sibilant vocal sounds like s sh and t are very difficult to reproduce electronically because they contain a large percentage of very high frequency harmonics Since these sounds are so essential to the intelligibility of speech they can t be simply removed with equalization In fact to help maintain articulation many sound engineers routinely boost the higher frequencies of the vocal spectrum 3 kHz to 8 kHz and or use microphones with presence curves like the Neumann U 87 or AKG C 414 However certain individuals and even certain languages contain overemphasized sibilants and any sort of high frequency boost only exacerbates the problem 2 6 Rev 1 1 11 15 94 2 6 Noise Reduction Noise reduction is the process of removing the noise from a signal without hopefully affecting the signal itself There are two types of noise reduction single ended th
31. Chorus effect Big PA simulation Great for paging Mixed up delays Big voice sound Swing style voice sound Voice resonance Modulated effects Simulates radio call in show Delay 1 44ms Delay 2 62ms Delay 1 86ms Delay 2 104ms Front center slow cross fade Mid right slow cross fade Far center slow cross fade Mid left slow cross fade Building Blocks Used to create programs with specialized functions RAM ROM 80 208 81 209 82 210 83 211 84 212 85 213 86 214 87 215 88 216 89 217 90 218 91 219 92 220 93 221 94 222 95 223 96 224 97 225 98 226 99 227 RAM ROM 100 228 101 229 102 230 103 231 104 232 105 233 106 234 107 235 108 236 109 237 110 238 111 239 112 240 113 241 114 242 115 243 116 244 117 245 118 246 119 247 120 248 121 249 122 250 G 6 Name Panner Doppler effect Compressor Sidechain Compressor Sidechain Compressor Sidechain Compressor Signal Delay Compressor Signal Delay AGC ARM Sensitivity AGC ARM Sensitivity AGC ARM Sensitivity De esser Absolute Threshold De esser Absolute Threshold Signal Threshold AGC Signal Threshold Signal Threshold Global Time Constants Global Time Constants Global Time Constants Global Time Constants Dynamics Preferences Dynamics Preferences Dynamics Preferences Dynamics Preferences Description Output Levels attached to oscillator adjust rate control for faster or slower panning Output level and band
32. EE emer Peto ends 4 4 4 572 De Esser Block uie ek Sek eek IS Iis nd 4 5 4 5 3 Downward Expander Block ete edere ette teste docet tu periti ae 4 6 4 5 4 Compressor Parameter 4 7 AE BIGEK ent ra Pa Rabb ns ined uad 4 7 4 5 6 Dynamics Section Control 2 4 9 4 5 7 Additional Dynamics Parameters 4 9 4 5 7 1 Sidechain filter RE 4 10 4 5 7 2 Expander knee 4 11 4 5 7 3 Compressor knee control 4 11 4 5 7 4 AGG absolute threshold eet ie ne ett ens 4 11 4 5 7 5 AGC knee ettet prn nan notae nen 4 11 4 5 7 6 ARM peak release 4 11 4 5 7 7 ARM integration ret rrt mer treten 4 11 4 5 7 8 ARM Signal Noise 4 11 4 5 7 9 Log converter time 4 11 4 5 7 10 Lookahead delay 4 11 4 5 7 11 De ess absolute 4 12 25 Delay Group aces v 4 12 erne 4 14 4 8 System errors Pere Er O re 4 14 4 8 1 Global OWING RE 4 14 tenete naa D appetitu fect Perses Ala oe ko 4 15 4 9 Presets Group AU 4 16 Scenarios
33. TIME MS TABLE nnt C 18 DELAY TIME TABLE MS ui C 18 DELAY FEEDBACK TABLE NEGATIVE THEN POSITIVE FEEDBACK C 19 DELAY INE FIETER TAB E In va e UENIRE M cie C 19 REALTIME SCALING TABLE rh e e o x xo ld a d d C 19 DEFAULT PAN T AB EE en d E ERE RR C 20 NORMALIZED MIDI PAN INPUT TABLE C 21 HEX TO DEGIMAD tree eek an tabo ieu exer troie iud C 22 DEGIMAL TO FIEX usi ve rite re i mia ter ase i edi i p ae C 22 Alphabetical List of Tables AGC PARAMETERS AGC RATIO TABLE ARM SENSE PARAMETERS ARM THRESHOLD DB ATTN100 TABLE DB ATTN18 TABLE DB ATTN82 TABLE DB COMPRESSER EXPANDER KNEE TABLE IN DB COMPRESSION PARAMETERS COMPRESSOR RATIO TABLE DATA RESPONSE DE ESS PROCESSOR DECIMAL TO HEX DEFAULT PAN TABLE DELAY FEEDBACK TABLE NEGATIVE THEN POSITIVE FEEDBACK DB DELAY LINE FILTER TABLE DELAY PROCESSOR DELAY TIME TABLE MS EDIT BUFFER DATA REQUEST EDIT BUFFER DATA RESPONSE EDIT BUFFER DATA SET EXPANDER RATIO TABLE EXPANSION PARAMETERS FILTER 1 FILTER 2 FILTER 3 FREQUENCY TABLE Hz GLOBAL HEX TO DECIMAL IDENTIFY REQUEST IDENTIFY RESPONSE LOG CONVERTER PARAMETERS MAKEUP GAIN TABLE ATTN24 MISCELLANEOUS NOISE REDUCTION PROCESSOR NORMALIZED MIDI PAN INPUT TABLE OUTPUT OUTPUT LEVEL TABLE DB PARAMETER MA
34. Table 6 F1 Level 81 out See Attn82 Table 7 F1 Freq BW Rate of Change See Tc Table F1 Level Rate of Change See Tc Table Table 12 Filter 2 Offset Description dec hex po i 9 9 F2 Mode NOP always BP UE 10 A F2 Freq 11 B F2 BW 12 C F2 Level 13 D F2 Freq BW Rate of Change 14 E F2 Level Rate of Change Table 13 Filter 3 Offset _ Description Range Reference hex dec 15 64 shelving LEN E 16 10 F3 Freq See Frequency Table 17 11 BW See BW Table 18 12 F3 Level 81 out See Attn82 Table 19 13 Freg BW Rate of Change See Tc Table 20 14 F3 Level Rate of Change See Tc Table C 8 Rev 1 1 11 15 94 Table 14 De ess Processor Offset _ Description Range Reference hex dec 21 15 0 off 127 on 22 16 Absolute Threshold See Attn100 Table 23 17 Relative Threshold See Attn100 Table 24 18 Attack Time See Tc Table 25 19 Release Time See Tc Table Table 15 Noise Reduction Processor Offset Description Range Reference dec hex dec 26 lA Mode 0 off 127 on 27 1B Minimum Turnover Freq See Frequency Table 28 1C Reserved 29 1D Absolute Threshold See Attn100 Table 30 lE Relative Threshold See Attn100 Table
35. The ISP shall be capable of accepting and delivering digital input signals at either a 44 1 kHz or 48 0 kHz sample rate The ISP shall be capable of converting analog signals to digital form using either the 44 1 kHz or 48 0 kHz sample rates The ISP shall be capable of accepting digital input signals conforming to the AES EBU standard or to the S PDIF standard Two such digital inputs shall be provided The digital inputs shall utilize a 3 pin XLR female connector and an RCA connector The digital inputs shall conform to the AES EBU standard and S PDIF standard respectively The ISP shall be capable of delivering digital output signals conforming to the AES EBU or S PDIF standard Two such digital outputs shall be provided The digital outputs shall utilize a 3 pin XLR male connector or an RCA connector The digital outputs shall conform to the AES EBU standard and S PDIF standard respectively The analog inputs shall be active balanced bridging designs The line inputs shall be terminated in 3 pin XLR female connectors All analog input circuitry shall incorporate RFI filters The analog outputs shall be active balanced designs having equal source impedances and terminated with 3 pin XLR male connectors All XLR connectors used for analog input output shall conform to the AES IEC polarity standard The balanced inputs shall accommodate 22 dBu signals without distortion and the balanced outputs shall be capable of delivering 21 5 dBm into 600
36. across the circuit divided by the resistance of the circuit This is known as Ohm s law In a circuit containing alternating current the situation is more complex the resistance presented to the current is a function of frequency This resistance is called impedance and is also measured in ohms Impedance is the vector sum of resistance capacitive reactance and inductive reactance Alternating currents are affected by resistance the same way as direct currents and Ohm s law can be used for AC if the reactances are zero that is if there are no capacitors or inductors in the circuit TAD abridged D 6 Rev 1 1 11 15 94 Limiter A special type of compressor which prevents the signal from exceeding a certain preset level threshold no matter what the input signal level may be Limiters are sometimes used for special effects in popular recordings especially vocals A vocal with limiting will be essentially at the same level regardless of the effort put out by the singer from a soft voice to a shout The shouting will sound subjectively louder however because of the increased harmonic content of the sound The dynamic range of a singer at a close range to a microphone is far greater than that of any instrument or musical ensemble and when recording a vocal with an ensemble without limiting a great deal of gain riding must be done to maintain musical balance Limiters are sometimes used in front of power amplifiers in sound reinforc
37. action after exceeding the absolute threshold Changes the sidechain filter from Affects the signal level history of Shelving highpass filter to limit Initial log averaging time constant Sets threshold for onset of de ess EditBuffer Offset 0 49 55 56 60 62 63 64 65 66 67 22 4 5 7 1 Sidechain filter The dynamics section sidechain has a shelving highpass filter in its control chain that limits the response of the dynamics section to very low frequency sounds The frequency edit buffer 65 and mode shelving highpass or lowpass edit buffer 0 of this filter may be varied The default condition is shelving highpass In general raising the filter frequncy in highpass mode makes the dynamics section compressor AGC and downward expander less responsive to low frequency sounds This may be useful for preventing p pops from causing compression or opening the downard expander Rev 1 1 11 15 94 Lowering the filter frequency in lowpass mode makes the dynamics section less responsive to high frequency sounds This may be useful for preventing sibilance or high frequency noise hiss clicks etc from triggering the dynamics section 4 5 7 2 Expander knee control The point in the downward expander s gain curve immediately below threshold is known as the knee The width of the knee may be altered to make the transition to the expander s ultimate slope more or less gradual E
38. all LEDs Src Selects the source of the realtime MIDI control as follows Display _Description oFF off Cn MIDI control change packet needs 2nd parameter setting AF MIDI after touch Pb MIDI pitch bend dL1 Delay modulation oscillator 1 41 2 Delay modulation oscillator 2 LoG Dynamics section log signal level nr NR center frequency NR must be engaged CGr Instantaneous compressor gain reduction EGr Instantaneous expander gain reduction bLC1 Block 1 output only in Block 2 edit 2nd MIDI control type for MIDI control change packet SCAL Scaling factor to apply to source after adding offset value 4 oFt Offset to apply to source value CLPL Lowest value allowed from this block after all processing scaling and offset CLPH Highest value allowed from this block after all processing scaling and offset Press the LEAVE EDIT button to exit the realtime MIDI editing mode if the SrC or oFt parameters were modified then the menu item temporarily reverts to the previous parameter PAr or SCAL Changing the PAr parameter temporarily disables the parameter update for a second This helps avoid accidentally and worse invisibly overwriting another edit buffer entry while selecting a new one Be careful Mysterious things not necessarily wonderful can happen when the edit buffer values are arbitrarily or randomly rewritten To restore a program after clobbering the edit buffer reload the source program over the existing
39. and holding the SAVE button The program has been saved when the display reads donE You can also store a modified program by pressing and holding the save button at any time When the display reads donE the 602 reverts to whatever mode it was in when the save button was pressed 7 5 2 Metering The 602 has two LED bargraphs that serve as input and output meters In addition the right hand bargraph does double duty as a gain reduction meter whenever you are editing any of the dynamics group In gain reduction mode the meter indicates the change from unity gain for the current function and the LEDs read and move from right to left When operating as a level meter the LEDs read and move from left to right Each mode has its own scale markings as shown on the front panel Both bargraphs are calibrated as headroom meters This means that the scale of the meter is referenced to digital clipping full scale and in the case of the output bargraph digital clipping corresponds to clipping at the analog outputs Both meters are peak responding Therefore adjusting the output level for 6 dB of output headroom 900900000 sets the output level so that the highest peak signal level falls 6 dB below clipping or at 15 dBm peak at the balanced output Now it happens that 8 8 8 8 the peak to average ratio for most music falls somewhere between 10 and 20 dB which means that the peak level ends up being 10 to 20 dB higher
40. converter For digital sources inputs 1 and 2 are the left and right channels respectively of the digital input stream Temporarily shifts the INPUT HEADROOM display to monitor one or the other of the two input channels The measurement point is immediately prior to the digital gain trim Exiting this menu function restores the bargraph to its normal mode which displays the highest peak level of the two input signals Clock source for the internal ADC DAC and DSP If the display reads and if the input source is microphone or line then the clock source is the internal 44 1 kHz or 48 kHz sample rate oscillator If the display reads CLCE then the external AES EBU input is used for the clock reference This allows a master system clock to provide the sample rate reference for the 602 and precludes any problems with mismatch or drift between digital clocks Controls memory write protection When memory is not write protected the SAVE switch LED will either be on solid edit buffer not dirty or flashing edit buffer modified dirty When memory protection is enabled the SAVE switch LED never illuminates Selects 44 1kHz or 48kHz as the internal sampling rate for analog input signals This menu selection is only valid for internal clock sources CLCI 4 8 2 MIDI Switch The MIDI switch sets various MIDI parameters in the 602 In addition it accesses the realtime block editor allowing modification of either of the two realtime block
41. direction and velocity Thus turning the Wheel quickly causes the display to change very quickly and turning the Wheel slowly causes the display to change very slowly as if there were a gear reduction unit on the Wheel Select a function for editing by pressing its associated switch once The switch begins flashing Next select the parameter that you want to edit and press its switch The display indicates the current value Turning the Wheel changes the value If a function switch has several choices the display indicates the choice and the Wheel cycles through the options An example of this sort of function switch is the GLOBAL switch 4 1 1 Loading Programs A program is nothing more than a group of control settings To load recall a program press the LEAVE EDIT switch to return to the top most control level PRESETS SAVE COMPARE The display now indicates the current program number Rotate the Wheel until the desired program number appears flashing in the display Press the LOAD switch When the program has loaded the display says donE and the number stops flashing Note the 602 always loads a copy of the program stored in program memory into the edit buffer The contents of the edit buffer are lost only when another program has been loaded If the program stored in the edit buffer is dirty i e it has been modified the red SAVE switch flashes It is possible to intentionally overwrite the current program b
42. echo is desired in a recording a tape recorder is sometimes used to add a time delay tape delay the delay representing the time it takes the tape to move between the record and reproduce heads This is called tape echo and is appropriate usage A popular way to get the same effect is to use a digital time delay system digital delay where the time delay is variable TAD Equalizer An equalizer contrary to what its name implies alters or distorts the relative strength of certain frequency ranges of an audio signal In a sense it should probably be called an unequalizer However the first equalizers were used to make the energy at all frequencies equal or to achieve flat response in telephone lines and this is where the term originated Another early use of equalizers was in the sound motion picture industry where they were used to improve intelligibility in film sound tracks Later on equalizers were found useful for creating special sound effects in the early days of radio and movies where they are extensively used to this day All equalizers are made up of various circuits called filters which are frequency selective networks containing resistors capacitors C and inductors L Normally filters attenuate certain frequency ranges and do not boost them however some equalizers that boost the signal are called filters An equalizer can boost or attenuate a certain frequency band but in common usage equalize means to boost
43. film post 8 1 mastering 8 1 music 8 1 PA 8 1 send receive loop 7 6 ARM sense parameters Table 20 C 11 attn tables C 18 bAt flashing in display 9 1 battery dead 9 1 bibliography D 11 block diagram 6 1 7 4 CH nn 4 15 chorus 2 9 7 15 CLCI CLCE 4 15 clip LED 4 2 clock source selection 4 15 comparisons 4 2 4 16 compressor controls 4 7 parameters Table 18 C 10 settings 7 13 settings for voice 2 8 time constants 2 8 tutorial 2 7 connections channel insert 7 6 input and output 3 3 making 6 1 controls dynamics 4 9 input 4 2 line 4 3 mic 4 3 output 4 14 conventions used 1 3 critical 2 2 data dump 4 15 dead battery 9 1 de esser controls 4 5 parameters Table 14 C 9 tutorial 2 6 default pan table C 20 Index delay 7 14 chorus 7 15 delay filter table C 19 delay time table Table 40 C 18 feedback table C 19 flanging 7 14 making echoes 7 14 modes 4 13 modulation 2 9 7 14 7 15 parameters Table 16 C 9 settings 7 14 tutorial 2 9 diagram simplified block 6 1 differences digital amp analog 2 1 digital inputs 4 14 inputs and outputs 3 5 output 5 1 DIGITAL IN SYNC LED 4 3 disable front panel 4 19 display gain reduction 7 10 headroom 4 2 7 10 input level 7 10 output level 7 10 display reads Er nn 9 1 display abbreviations A 1 dL 1 dL 2 duAL 4 12 dnAl 4 15 dnEd 4 15 downward expander see also expander 2 7 downward expansion tutorial 2
44. greater distance the gain increases to keep the recorded level the same This type of machine is often used for radio interviews and usually the gain changes can be plainly heard as the background noise rises each time the speaker pauses for a few seconds only to suddenly fall the moment the next syllable is uttered TAD A more recent meaning for AGC is the combination of the device described previously and a signal sensing circuit that prevents the gain from changing when there is no valid signal present This prevents the rising and falling background noise heard when a simple compressor is used as an AGC The 602 uses this technique Analog to Digital Converter ADC In digital audio systems the audio signal analog must first be converted to digital form before it can be further processed This entails sampling the signal at very short successive time intervals and converting the height of each sample to a digital word which is simply a binary number indicating the amplitude of the waveform at that instant See also quantization The output of the A D converter is a series of digital words expressed in binary form Before the signal can be fed to an amplifier so it can be heard it must undergo digital to analog conversion This recovers a replica of the original audio signal from the digital words TAD Anti Aliasing Filter Before a signal is subjected to the process of A D conversion it must be passed thorough a low pass filte
45. impedance input and output 3 2 matching 3 1 initializing 4 19 InP 4 14 input ac 5 1 analog 5 1 digital 5 1 selector 4 14 installation requirements 7 1 leave edit 4 17 index ii LeaveEdit 4 1 LED clip 4 2 line 4 3 mic 4 3 level matching 3 1 loading programs 4 1 4 16 log converter parameters Table 21 C 11 makeup gain 4 7 4 8 matching impedance 3 1 level 3 1 rule 3 1 3 2 signal levels 3 2 Metering 7 10 7 12 MIDI channel number 4 15 connectors 5 1 default pan table C 7 input and outputs 3 5 Lexicon MRC B 1 parameters 4 15 possibilities 2 10 program storage 7 8 realtime 4 15 7 8 B 1 realtime block 1 Table 23 C 12 recognized commands C 4 summary 7 8 unit number 4 15 MIDI Mfr ID G 3 modulation waveform 4 13 n 90 4 13 noise floor 2 1 noise reduction controls 4 5 how to 7 12 parameters Table 15 C 9 tutorial 2 7 using expander for 7 13 Operational Details 7 1 output analog 5 1 digital 5 1 mic level 7 2 minimum load 3 2 parameters Table 22 C 11 P PAd 4 14 P 90 4 13 pan input table C 21 parameter adjustment wheel 4 1 rate of change 4 2 parameters AGC 4 8 compressor 4 7 de esser 4 5 dynamics section 4 9 equalizer 4 3 expander 4 6 noise reduction 4 5 phasing see flanging polarity 3 3 program data request 5 example using MIDI B 1 loading 4 1 saving 4 1 4 16 7 10 worksheet G G 2 G G 2 write protect 4 16 7 10 progra
46. its queue the 602 may miss the last message in the data stream A program change forcefully clears the queue 1 3 Sysex Implementation F0 All sysex messages use the universal system exclusive code format The MIDI sysex message uses the following format hex Send lt sysex gt lt mfrID gt lt unitID gt lt unitH gt lt command gt lt data gt lt EOX gt lt FO gt lt 00 gt lt 00 gt lt 5E gt lt 02 gt lt gt lt command gt lt data F7 Send a Edit Buffer Data Set message to the 602 and set the level for Filter 1 of the parametric equalizer block to 12 dB lt sysex gt lt mfrID gt lt unitID gt lt unitH gt lt command gt lt offset gt lt value gt lt EOX gt lt FO gt lt 00 gt lt 00 gt lt 5E gt lt 02 gt lt gt lt 1C gt lt 06 gt lt 70 gt lt F7 gt where is the midi sysex command 00 00 5E is Symetrix ID 02 identifies the 602 is the unit number 1C is the Edit Buffer Data Set command see Table 3 is the level parameter from the Filter 1 table Table 11 70 is the value for 12 dB from Table 27 see also Table 11 C 2 Rev 1 1 11 15 94 In the tables that follow Short data transfers are from or to the edit buffer only Block data transfers can access any of the stored program data including the edit buffer and system setup offsets are in decimal INPUT OUTPUT refer to the 602 REQUEST is a data request to the 602 RESPONSE is data from the 602 C 1 4 Sysex Echo Th
47. limiters noise reducers etc D 8 Rev 1 1 11 15 94 Release Time release time of a dynamics processor is the time required for the processor s gain to return to its nominal value after the controlled signal exceeds or doesn t exceed a preset threshold See also attack time compressor expander Reverberation The remainder of sound that exists in a room after the source of sound is stopped is called reverberation sometimes mistakenly called echo The time of reverberation is defined as the time it takes for the sound pressure level to decay to one millionth of its former value This is a 60 decibel reduction in level All rooms have some reverberation and an important subjective quality of a room is its reverberation time although other factors such as ratio of direct to reverberant sound are probably more important In a real room the sound heard by a listener is a mixture of direct sound from the source and reverberant sound from the room Reverberant sound is diffuse coming from random directions and the direct sound allows us to localize the source of they sound TAD abridged Ribbon Microphone Velocity Microphone A type of microphone which usually has a polar pattern shaped like a figure 8 The first velocity microphone was the ribbon microphone invented about 1931 by Harry F Olson of RCA research laboratories The ribbon microphone uses as an active element a small corrugated strip of very thin aluminum ribbon
48. mic s signal chain Considering the inverse rule of the knobs the more knobs you give them the easier it is for someone to get hopelessly screwed up the attitude of most station s PDs and engineers was to hide the equalizer somewhere preferably under lock and key The 602makes it easy for each personality to have their own individualized curve Granted if you give the jocks access to the unit someone will inevitably shoot themselves in the foot but at least everyone can have their own curve Some general thoughts on speech equalization 1 Try to use wider bandwidths Narrower bandwidths 1 2 octave and less less audible harder to hear and are generally only useful for remedial work Broader bandwidths are less obnoxious more pleasing sounding and easier to work with especially if you re boosting a range of frequencies 2 Try to avoid massive amounts of boost or cut If you re only trying to impart a flavor like sprinkling salt and pepper on a meal then 6 8 dB of boost or cut should be all that you need 3 Awide bandwidth cut is equivalent to a boost at the frequencies surrounding the cut 4 A quick way to figure out what s going on is to set the level of one band of the equalizer to full boost 18 dB then switch to the frequency control and vary the frequency of that band of the equalizer while listening to program material fed through the unit This usually makes quick work out of finding the region that you want t
49. of samples per second must be uniform and precisely controlled TAD abridged see also quantization Shelving Equalizer An equalizer whose frequency response curves rise or fall to a maximum value remaining at that value to the limits of audibility The bass and treble controls on most home stereo amplifiers are shelving equalizers Sibilance Vocal recordings especially if made with very close microphones are often characterized by excessive loudness of the voice sibilants and this effect is sometimes called sibilance The most difficult sibilants to reproduce accurately are the sounds s and sh TAD see also de esser Single D Microphone A single D microphone is a directional microphone having only one entrance for off axis sounds Single D microphones exhibit a property called proximity effect which is a boosting of low frequencies when the microphone is used close up to the sound source See also variable D microphone Slapback Slap Echo The single repetition of a signal at a fixed time delay to simulate an echo from a single reflecting surface as opposed to a multiple echo from a time delay where the delayed signal is repeatedly fed back into the delay input TAD Sysex AMIDI message command that stands for System Exclusive MIDI sysex messages are commonly used for controlling audio processors or other MIDI instruments The sysex message exists to allow programming controlling beyond that which is predefined in the MI
50. output LED display changes to read gain reduction Rev 1 1 11 15 94 4 5 THRESHOLD Switch ATTACK Switch RELEASE Switch FREQ Switch Sets the relative threshold for the start of de ess action The de esser measures the energy on each side highpass and lowpass of the sidechain filter Sibilant sounds above this threshold level are reduced in level The absolute threshold adjustable via the realtime editor sets a minimum level that the signal must exceed to receive de essing see Appendix A Sets the time required for the de esser to engage This means that the input signal must remain above the THRESHold setting for a time that is longer than the attack time The ATTACK time ranges from 0 1ms 100 microseconds to 10 000 ms the display reads 9999 but the time is really 10 000 milliseconds or 10 seconds Sets the time required for the de esser to recover once the sibilant sound has ceased The time displayed is the time required for full decay in response to a large above THRESHold change in the input signal The RELEASE time ranges from 100 ms 100 milliseconds to 10 000 ms the display reads 9999 but the time is really 10 000 milliseconds or 10 seconds Sets the transition frequency of the de esser s sidechain filter The frequency can be varied from 31 Hz to 21 112 Hz The default frequency is 5 kHz 4 5 3 Downward Expander Block The downward expander reduces its gain for any signal level below the THRESHol
51. r of the onset of filter activity Pressing on the THRESH switch again accesses the absolute threshold A which governs the transition between spectral content and signal level as the basis for the filter s action In general use lower resting frequencies to remove excess noise Higher resting frequencies result in a more subtle action To set the NR with signal applied set the resting FREQuency at 1 kHz Vary the THRESHold setting until you see activity on the right LED display Listening you should hear the noise reduction removing the noise and more than likely your signal Set the threshold at O zero Raise the filter FREQuency until you hear onset of the noise Lower the filter FREQuency until you hear the noise disappear Now lower the THRESHold setting until you find the magic compromise between the noise the music and the audibility of the filter working Higher THRESHold settings closer to zero make it more difficult to open up the dynamic filter and lower settings closer to 35 cause the filter to almost always run wide open Finally use the absolute THRESHold A to determine the signal level at which you want the filters action to become level dependent Usually this is at a fairly low level and it is probably more important to eliminate the noise even at the expense of the signal The useful range for this parameter runs from 80 dB to 50 dB 7 5 7 De Esser The de esser uses a limiter controlled by a mildly peake
52. resale repair replacement or use of any product will not exceed the price allocable to the product or any part thereof which gives rise to the claim In no event will Symetrix be liable for any incidental or consequential damages including but not limited to damage for loss of revenue cost of capital claims of customers for service interruptions or failure to supply and Rev 1 1 11 15 94 10 1 costs and expenses incurred in connection with labor overhead transportation installation or removal of products or substitute facilities or supply houses 10 2 Rev 1 1 11 15 94 11 Repair Information Should you decide to return your 602 to Symetrix for service please follow the following instructions 11 1 Return Authorization Symetrix will service any of its products for a period of five years from the date of manufacture However no goods will be accepted without a Return Authorization number Before sending anything to Symetrix call us for an RA number just ask we ll gladly give you one call 206 787 3222 weekdays 8am to 4 30 pm pacific time 11 2 In Warranty Repairs To get your unit repaired under the terms of the warranty l Call us for an RA number 2 Pack the unit in its original packaging materials 3 Include your name address etc and a brief statement of the problem Your daytime telephone number is very useful if we can t duplicate your problem 4 Put the RA number on the outside of the box 5 Ship the
53. section of the 602 uses two delay lines having separate inputs and separate outputs The outputs drive a lowpass filter that feeds the output mix and the feedback controls Each feedback signal mixes with the input signal at the delay line input of the opposite channel the feedback is cross coupled A signal flow diagram may be found in Figure 7 3 The delay times of the delays may be adjusted independently or ganged together The feedback factor lowpass filter frequency delay time rate of change level related rate of change and the wet dry mix are independently adjustable Finally the delay time of the two delays may be modulated with the rate waveform and depth parameters being adjustable The delay modulation source is either a sine wave generator triangle wave generator or a random number generator The RATE parameter sets either the sine triangle wave frequency or the random number generator s update rate The depth control limits the range of the delay time modulation Holding down the RATE button changes the delay modulation source of the previously mentioned parameters may be programmed via MIDI 7 5 11 1 Echo effects Creating an echo consists of delaying the input signal by some amount then adding the delayed signal back to itself This creates an echo having one repeat To create this type of sound set the MIX to 5096 set the DELAY to 330 ms for both channels set the FILTER to 18 kHz and finally set the FEEDBACK to O
54. source polarizing voltage When sound acts on the diaphragm the pressure variations cause it to move slightly in response to the sound waveform This causes the capacitance to vary in like manner and because the charge is fixed the voltage on the backplate will vary according to the laws governing the capacitor This voltage variation is the signal output of the microphone The condenser microphone has extremely high output impedance and must be placed very near a preamplifier to avoid loss of the signal It is possible by special treatment of the backplate and by combining several microphone elements to attain various directional patterns including bi directional figure 8 cardioid and super cardioid TAD See also phantom power dB Decibel Literally one tenth of a bel The bel is named after Alexander Graham Bell which is why the B in dB is capitalized and the number of bels is defined as the common logarithm of the ratio of two powers Thus two powers one of which is ten times the other will differ by 1 bel 10 watts are 1 bel higher in level than 1 watt A 360 horsepower car is 1 bel more powerful than a 36 horsepower motorcycle Any power ratio may be expressed in bels and it is important to note that only power ratios are allowed a bel is a pure number with no dimensions Decibel reference quantities Reference Remarks none Only useful in a relative sense i e Unit of Measurement 3 d
55. switches Pressing a switch transfers the display to that parameter s current value The parameter wheel allows you to change the value Finally the 602 allows you to compare your stored setting with the current edited setting without committing the edited settings to memory Of course all this processing power can be remotely controlled via MIDI The 602 s MIDI implementation includes simple program change as well as parameter editing All analog inputs and outputs are available via XLR connectors The AES EBU digital inputs and outputs use XLR connectors and the S PDIF digital inputs and outputs use RCA connectors The MIDI input and output connections use standard 5 pin female DIN connectors The 602 s unique set of digital tools can make voices instruments or sound effects jump out of any mix Its combination of factory presets and non volatile user program space guarantee predictable and repeatable effects from session to session performance to performance We recommend that you read this manual from cover to cover Somewhere between the confines of the two covers you should find the answers to most 98 of your questions both technical as well as musical If you re in a hurry like most of us or if you really don t believe that someone could write a decent owners manual that you can read and understand then do us both a favor and read the remainder of this section and Chapter 6 Fast First Time Setup Chapter 6 will help you g
56. tables show the parameters used by each different section of the Dynamics Section The first line of each table shows the function the second line shows the front panel designation on the 602 the third shows the parameter name and the fourth shows the parameter s range NR THRESHOLD ATTACK RELEASE FREQ RATIO threshold n a frequency 35 0 dB 1 0 21 11kHz DS DE ESS THRESHOLD ATTACK RELEASE FREQ RATIO threshold attack release frequency 35 0 dB 0 1 9999 ms 100 9999 31 21 11kHz EXPAND THRESHOLD ATTACK RELEASE FREQ RATIO threshold attack release expansion ratio 100 0 dB 0 1 9999 ms 100 9999ms_ 1 1 1 8 COMPRESS THRESHOLD ATTACK RELEASE FREQ RATIO threshold attack release compression ratio makeup gain RE 1 1 10 1 auto 24 dB THRESHOLD ATTACK RELEASE FREQ RATIO auto release attack release compression ratio threshold makeup gain bid 9999 ms 100 9999 ms 1 1 4 1 auto 24 dB 4 5 7 Additional Dynamics Parameters In addition to the previously mentioned controls there are several additional parameters affecting the dynamics processor Dynamic filtering compressor and AGC None of these controls are accessible directly from the front panel however they may be accessed via MIDI or by means of the rea
57. than the average which is what you read on a VU meter Thus the average level could be anywhere between 5 dBm to 5 dBm 9 to 1 VU at the balanced output depending on the source material 7 10 Rev 1 1 11 15 94 7 5 3 Gain Setting There are three places to adjust the gain of the 602 at the analog inputs before the DSP section and after the DSP section An understanding of this topic is essential to getting the most from your 602 A more basic discussion can be found under the heading Gain Setting in Chapter 2 First the analog input gains You make best use of the 602 s signal to noise ratio by ensuring that your analog input signals are adjusted to just barely fit within the input range of the A D converter Doing so ensures that the entire conversion range of the converter gets used ensuring maximum dynamic range through the digital portions of the unit Set the LINE gain controls so that the 6 dB input headroom LED illuminates on signal peaks The red CLIP LED should never illuminate Next the input digital gain gAln in the global parameter group Most of the time set the digital input gain to O dB If you are heavily equalizing at 2 or more overlapping frequencies on the equalizer you may also need to reduce the digital signal level slightly to accommodate the extra boost If you have a weak analog input signal and the analog input gain is already wide open then it is OK to add some digital input gain to bring the ov
58. the 602 Appendix B Appendix B tells how to use the Lexicon MRC with the 602 Appendix C Appendix C describes how to communicate with the 602 via MIDI This appendix also contains a description of the 602 s Midi implementation Appendix D Appendix D contains a glossary and a useful bibliography Appendix E Appendix E contains the Architects and Engineer s specifications Appendix F Appendix F contains disassembly instructions Appendix G Appendix G contains a listing of the preset programs and other miscellany 1 2 Operator Safety Summary The information in this summary is intended for persons who operate the equipment as well as repair personnel Specific warnings and cautions are found throughout this manual wherever they may apply they do not appear in this summary 1 2 Rev 1 1 11 15 94 The notational conventions used in this manual and on the equipment itself are described in the following paragraphs 1 2 1 Equipment Markings CAUTION RISK OF ELECTRIC SHOCK DO NOT OPEN RISQUE DE CHOC ELECTRIQUE ATTENTION NE PAS OUVRIR No user serviceable parts inside Refer servicing to qualified service personnel ne se trouve a l interieur aucune piece pourvant entre repar e l usager S adresser a un reparateur comp tent The lightning flash with arrowhead symbol The exclamation point within an eguilateral within an eguilateral triangle is intended to triangle is intended to alert the user of the alert the user of t
59. the best tradeoff between headroom and dynamic range The analog inputs each have gain controls to help you run these stages as hot as possible without clipping After conversion to digital form some signals may be too hot for any signal processing that results in an increasing signal level Thus the 602 has a digital gain control that allows you to raise or lower the level sent to the digital processors Finally there is an overall digital output gain control allowing you to restore the signal level to normal 2 4 Equalization Equalization is nothing more than selectively or not amplifying a signal based on frequency Since audio signals consist of combinations of fundamental signals and their harmonics changing the tonality or the spectral balance of a signal involves nothing more than altering the relationship of the fundamental to its harmonics and of the harmonics to themselves Each harmonic is responsible for one aspect of the audible character of a signal knowing these relationships allow you to quickly zero in on the correct frequency range of the signal and quickly apply boost or cut to enhance or correct what you are hearing The audio spectrum has several critical portions that are responsible for our perceptions of sounds that we hear Range Frequencies Musical Location Very Low Bass 16 64 Hz 1st and 2nd octaves 64 256 Hz 3rd and 4th octaves 256 2048 Hz 5th 6th and 7th octaves Lisping Quality 3000 Hz Between th
60. the two channels Digital signals at either 44 1 kHz or 48 kHz sample rates may be fed directly into the 602 for processing The processed signals are available at the outputs as AES EBU or S PDIF and stereo analog balanced line level The digital outputs and analog outputs operate simultaneously The 602 does not perform dithering or re dithering Regardless of the input source the 602 always treats its input signals as a stereo pair With the exception of the delay line the 602 always applies identical processing to both signals Digital signals may be up to 24 bits wide the 602 treats all digital signals as if they were 24 bit The equalizer is a digital implementation of a common three band parametric equalizer The usual complement of controls may be found and the outside bands may be converted into shelving equalizers All bands cover the entire frequency range The DS and Noise Reduction block have independent control chains The Noise Reduction system implements a variable frequency low pass filter controlled by on the relative high frequency content of the input signal The De esser is a broadband limiter having a sharply peaked filter in its sidechain The Dynamics block is a digital realization of an analog compressor AGC expander A common log converter provides a logarithmic representation of the amplitude of the input signal to the components of the Dynamics block Within the dynamics block the component having the greatest a
61. there is no input signal THRESHOLD Switch The NR uses two threshold settings one relative display reads r and the other absolute the display reads A You access the two threshold settings by pressing the THRESH switch when the NR has been selected The NR reacts to the ratio of the signal passing through the adaptive lowpass filter and the signal being rejected by the adaptive lowpass filter Higher less negative relative threshold settings require larger amounts of high frequency content to cause the filter to open up The absolute threshold setting determines the transition point below which the NR system ignores the high frequency content and relies strictly upon signal level information Typically the absolute threshold should be set to equal the noise floor of the program material the useful range for this parameter being from 80 to 50 dB 4 5 2 De Esser Block Like the NR system the de esser is a feedback control system The de esser uses a broadband limiter with a peaked highpass filter in its sidechain the frequency response is always flat regardless of the degree of de essing The attack release and frequency switches are functional and these parameters are also accessible via MIDI or the realtime editor DS Switch Toggles the de esser between active and out When editing the de esser switch LED flashes otherwise it reflects the state of the de esser in or out Whenever editing the de esser parameters the
62. unit to Symetrix freight prepaid Just do those five things and repairs made in warranty will cost you only the one way freight fee We ll pay the return freight If you choose to send us your product in some sort of flimsy non Symetrix packaging we ll have to charge you for proper shipping materials If you don t have the factory packaging materials then do yourself a favor by using an oversize carton wrap the unit in a plastic bag and surround it with bubble wrap Pack the box full of Styrofoam peanuts Use additional bubble wrap if you must ship more than one unit per carton Be sure there is enough clearance in the carton to protect the rack ears you wouldn believe how many units we see here with bent ears We won t return the unit in anything but original Symetrix packaging Of course if the problem turns out to be operator inflicted you ll have to pay for both parts and labor In any event if there are charges for the repair costs you will pay for return freight All charges will be COD unless you have made other arrangements prepaid Visa or Mastercard 11 3 Out of Warranty Repairs If the warranty period has passed you ll be billed for all necessary parts labor packaging materials and any applicable freight charges Remember you must call for an RA number before you send the unit to Symetrix Rev 1 1 11 15 94 11 1 Notes Rev 1 1 11 15 94 12 Specifications Input Output Analog Inputs Digital Inputs
63. used for MIDI output and input DIGITAL OUTPUT RCA connector and XLR male connector used for S PDIF and AES EBU respectively digital output Push push switch selects between protocols DIGITAL INPUT RCA connector and XLR female connector used for S PDIF and AES EBU respectively digital input Push push switch selects between protocols connectors transformer isolated Outputs XLR male balanced Analog audio output of the 602 Pin 2 is hot Inputs XLR female balanced line level analog inputs Pin 2 is hot Rev 1 1 11 15 94 5 1 5 2 Notes Rev 1 1 11 15 94 6 FastFirst Time Setup Follow these instructions to get your 602 up and running as quickly as possible The intent of this section is to get the 602 to pass signal If you need something clarified you ll find the answer elsewhere in this manual Figure 6 1 is a simplified block diagram of the 602 Take a moment now check the block diagram out and take note of the following points The diagram shows three different signals mono stereo and data All input signals treated as a stereo pair OPUS DIGITAL any processing applied ES EBU input SIGNAL AES EBU applies equally to both S PDIF NE a PROSE OUTPUTS S PDIF channels Line level signals are converted to digital and Anise applied equally to the CHa COMMAND pa LEFT left and
64. wiring refer to Figure 3 3 Ground loop check related system equipment grounding Are all system components on the same AC ground Check input signal Is it too hot or is it already distorted Is the HEADROOM display indicating clipping Check output loading Should be above 600 ohms Are the power amplifier s clipping Is something else clipping Check input digital gain and output digital gain settings Check input signal levels and level control setting The HEADROOM display should indicate signal up to but not including the CLIP led Check gain settings on downstream equipment The system gain structure should be such that the 602 operates at or near unity gain Is the input signal already noisy Is the unit plugged in and turned on Is the AC outlet OK Is the unit in BYPASS mode Power up error Try turning the unit off then on again Write the number down before you call us Memory backup battery death throes You have about two weeks to replace the battery before you lose your programs Contact the factory before trying to do this yourself Call us 9 1 9 2 Notes Rev 1 1 11 15 94 10 602 Stereo Digital Processor Limited Warranty This Symetrix product is designed and manufactured for use in professional and studio audio systems Symetrix Inc Symetrix warrants that this product manufactured by Symetrix when properly installed used and maintained in accordance with the instructions
65. 12 113 114 115 116 Set sl 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 Set sl 136 137 138 139 140 141 142 143 page page Displays DEFINE SYSEX BYTES 8 9 LABEL FOR slrl ider 2 to control delay 2 page page page page page page SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slr2 SOURCE DEST OUT Data to Set Enter 5E 01 00 1C 22 BYTE DLYI slr2 SYSEX FO 00 00 5E 01 00 1 23 DLY2 ider 3 to control feedback recirculation page page 144 Rev 1 1 1 SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 1 15 94 slr3 SYSEX 1 FO 00 00 5E Comments mfrID2 device type unit channel edit buffer data set edit buffer 34 send slider setting use sliders 1 4 to set label to D L Y I setup slider 2 use button 1 to set source to slr2 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 35 send slider setting use sliders 1 4 to set label to D L Y OH use button 1 to set source to slr3 use slider 2 to set DEST to SYSEX use s
66. 2 8 2 8 2 8 3 0 3 0 3 0 3 0 60 3 2 3 2 3 2 3 2 3 5 3 5 3 5 3 5 3 8 3 8 70 3 8 3 8 4 0 4 0 4 0 4 0 4 5 4 5 4 5 4 5 80 5 0 5 0 5 0 5 0 5 5 5 5 5 5 5 5 6 0 6 0 90 6 0 6 0 6 5 6 5 6 5 6 5 7 0 7 0 7 0 7 0 100 7 5 7 5 7 5 7 5 8 0 8 0 8 0 8 0 8 5 8 5 110 85 85 90 90 90 90 95 95 9 5 9 5 120 100 100 100 100 120 120 120 120 Table 36 Time Constant Table 0 1 2 3 4 5 6 7 8 9 0 100us 150us 200us 250us 300us 35008 40006 500us 600us 700us 10 800us 9005 1 0ms 1 5ms 2 0ms 2 5ms 3 0ms 3 5ms 4 0ms 4 5ms 20 5 0ms 5 5ms 6 0ms 7 0ms 8 0ms 9 0ms 10ms 12ms 14ms 16ms 30 18ms 20ms 22ms 24ms 26ms 28ms 30ms 32ms 34ms 36ms 40 38ms 40ms 45ms 50ms 55ms 60ms 65ms 70ms 75ms 80ms 50 85ms 90ms 95ms 100ms 110ms 120ms 130ms 140ms 150ms 160ms 60 170ms 180ms 190ms 200ms 210ms 220ms 230ms 240ms 250ms 260ms 70 270ms 280ms 290ms 300ms 320ms 340ms 360ms 380ms 400ms 420ms 80 440ms 460ms 480ms 500ms 520ms 540ms 560ms 580ms 600ms 620ms 90 640ms 660ms 680ms 700ms 725ms 750ms 775ms 800ms 850ms 900ms 100 925ms 950ms 975ms 1 0s 1 15 1 25 1 35 1 45 1 55 1 65 110 1 75 1 85 1 95 2 05 2 55 3 05 3 55 4 05 4 55 5 05 120 5 58 6 05 6 55 7 05 7 55 8 05 9 05 10 05 1 1 11 15 94
67. 3 microphone Warm EQ settings EQ set for plucked guitar Wide Chorus for 12 String Guitar Fat chorus for electric guitar Really fat chorus for E guitar Description EQ set to brighten signal stereo image widening Light compression De essing and noise reduction EQ set for voice Adjust Feedback control for more or less echoes Moving echoes Makes voices sound large Signal changes pitch and echoes Level and Frequency are modulated to the delay oscillator adjust rate for faster or slower rotation Output level modulated by the delay section oscillator ROC of Filters set to long fade once program is loaded Echoes change pitch Echoes create unison voice that follows signal Thin sound for telephone like response Resonant robot like voice Big scary sound Splattered echoes Repeated echoes Creates moving stereo image Simulates Stereo signal from mono source Deep resonant Flanger G 5 Name Small Semi live Room Presentation Room Gym P A System Backstage Interview Soft amp Dry Loud amp Wet Acoustic Chorus Public Address Airport PA Mondo Bizarro Voice of Doom Be Bop A Lula Stereo Robots Stereo Invaders Telephone voice Delay Chorus 1 Delay Chorus 2 Perspective 1 Perspective 2 Perspective 3 Perspective 4 Description Simulated room ambience Medium size room Large size room Medium size room with signal toward the front Loud signals cause longer echoes Delay Mix attached to Dynamics section threshold Light
68. 31 1 32 20 Table 16 Delay Processor Offset _ Description Range Reference hex dec 33 21 0 off 127 on 34 22 Delay Line 1 Delay Time 0 127 See DelayTime Table 0 500 us 127 330 ms 35 23 Delay Line 2 Delay Time 0 127 See DelayTime Table 0 5001 127 330ms 36 24 Delay Line Rate of Change 0 127 See Tc Table 37 25 Cross Recirculation O pos 0 dB See Delay Feedback Attenuation 64 off Table 127 neg 0 dB 38 26 Filter Frequency 0 127 see Delay Filter Table 39 27 Direct Delay Mix Percent 0 0 delay 100 direct See Normalized MIDI 127 10096 delay 096 direct Pan Input Table CH 1 pan tbl Direct CH 2 pan tbl Delay 40 28 Modulation Depth 0 127 41 29 Modulation Rate lt value x 1 Hz 1 Hz 42 2A Modulation Type 0 random 85 sine 127 triangle 43 2B Mix Rate of Change See Tc Table Rev 1 1 11 15 94 C 9 Table 17 Expansion Parameters Offset _ Description Range Reference hex dec 44 2C Mode 0 off 127 45 2D Threshold See Attn100 Table 46 2E Attack Time See Tc Table 47 2 Release Time See Tc Table 48 30 Expansion Ratio See Expansion Ratio Table 49 31 Knee Control
69. 31 33 36 36 38 41 44 44 47 10 50 54 54 58 60 67 67 72 77 82 20 82 88 95 100 100 109 120 125 125 134 30 144 154 154 165 177 177 189 203 218 218 40 233 250 268 268 287 308 330 330 354 379 50 406 406 435 467 500 500 536 574 616 616 60 660 707 758 758 812 871 871 933 1000 1072 70 1072 1149 1231 1320 1320 1414 1516 1625 1625 1741 80 1866 2000 2000 2144 2297 2462 2462 2639 2828 3031 90 3031 3249 3482 3732 3732 4000 4287 4287 4595 4925 100 5278 5278 5657 6063 6498 6498 6964 7464 8000 8000 110 8574 9190 9849 9849 10556 11314 12126 12126 12996 13929 120 14929 14929 16000 17148 18379 18379 19698 21112 Rev 1 1 11 15 94 0 1 2 3 4 5 6 7 8 9 o OFF OFF 900 88 0 840 820 820 800 780 76 0 10 74 0 72 0 720 700 68 0 66 0 640 62 0 62 0 60 0 20 58 0 560 540 520 520 50 0 490 48 0 470 46 0 30 46 0 45 0 44 0 43 0 42 0 42 0 41 0 40 0 39 0 38 0 40 37 0 370 36 0 350 34 0 33 0 32 0 32 0 31 0 30 0 50 29 0 280 270 27 0 2260 25 0 24 0 230 2220 220 60 21 0 200 19 0 18 0 3170 17 0 16 0 15 0 14 0 13 0 70 13 0 12 0 11 0 10 0 9 5 9 0 9 0 8 5 8 0 7 5 80
70. 5 5 1 5 125 59 E 0 0 40 19 1 0 6 5 83 39 5 5 15 126 60 OFF oo 41 19 10 6 5 84 40 6 0 1 5 127 60 OFF 0 0 42 20 1 5 6 0 85 40 6 0 15 Rev 1 1 11 15 94 C 21 C 2 Hexadecimal Conversion Tables Table 46 Hex to Decimal C 22 0 1 2 3 4 5 6 7 8 9 a b c d el f o ol 314 5 6 7 8 9 10 11 12 13 14 15 10 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 20 3 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 30 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 40 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 50 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 60 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 70 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 Table 47 Decimal to Hex 1011213456 7819 0 0 1 2 3 4 5 6 7 8 9 10 b C d e f 10 11 12 13 20 14 15 16 17 18 19 la
71. 500 4500 4500 4500 40 5000 5000 5000 5000 5500 5500 5500 5500 6000 6000 50 6000 6000 6500 6500 6500 6500 7000 7000 7000 7000 60 7500 7500 7500 7500 8000 8000 8000 8000 8500 8500 70 8500 8500 9000 9000 9000 9000 9500 9500 9500 9500 80 10000 10000 10000 10000 10500 10500 10500 10500 11000 11000 90 11000 11000 11500 11500 11500 11500 12000 12000 12000 12000 100 13000 13000 13000 13000 13500 13500 13500 13500 14000 14000 110 14000 14000 15000 15000 15000 15000 16000 16000 16000 16000 120 17000 17000 17000 17000 18000 18000 18000 18000 Table 43 Realtime Scaling Table 0 1 2 3 4 5 6 7 8 9 0 4 0 3 8 3 6 3 4 3 2 3 0 2 8 2 6 2 4 2 2 10 2 0 1 9 1 8 1 7 1 6 1 5 1 4 1 3 1 2 1 1 20 1 0 0 95 0 90 0 85 0 80 0 75 0 70 0 65 0 60 0 58 30 0 56 0 54 0 52 0 50 0 48 0 46 0 44 0 42 0 40 0 38 40 0 36 0 34 0 32 0 30 0 28 0 26 0 24 0 22 0 20 0 18 50 0 16 0 14 0 12 0 10 0 09 0 08 0 07 0 06 0 05 0 04 60 0 03 0 02 0 01 0 0 0 0 0 01 0 02 0 03 0 04 0 05 70 0 06 0 07 0 08 0 09 0 10 0 12 0 14 0 16 0 18 0 20 80 0 22 0 24 0 26 0 28 0 30 0 32 0 34 0 36 0 38 0 40 90 0 42 0 44 0 46 0 48 0 50 0 52 0 54 0 56 0 58 0 60 100 0 65 0 70 0 75 0 80 0 85 0 90 0 95 1 0 1 1 1 2 110 1 3 1 4 1 5 1
72. 6 1 7 1 8 1 9 2 0 2 2 2 4 120 2 6 2 8 3 0 3 2 3 4 3 6 3 8 4 0 Rev 1 1 11 15 94 C 19 Table 44 Default Pan Table Offset Channel Channel Offset Channel Channel into Pan 1 Atten 2 Atten into Pan 1 Atten 2 Atten Table dB dB Table dB dB Buf Buf 0 0 0 OFF 30 3 0 30 1 0 0 32 0 31 3 5 3 0 2 0 0 26 0 32 3 5 25 3 0 0 22 0 33 4 0 25 4 0 0 20 0 34 4 0 20 5 0 0 18 0 35 4 5 20 6 0 0 16 0 36 5 0 20 7 0 0 15 0 37 5 0 1 5 8 0 0 14 0 38 5 5 1 5 9 0 0 13 0 39 5 5 1 5 0 0 12 0 40 6 0 1 5 11 0 0 11 0 41 6 5 1 0 12 0 5 10 0 42 7 0 1 0 13 0 5 9 5 43 7 5 1 0 14 0 5 9 0 44 8 0 1 0 15 0 5 8 5 45 8 5 0 5 16 1 0 8 0 46 9 0 0 5 17 1 0 7 5 47 9 5 0 5 18 1 0 7 0 48 10 0 0 5 19 1 0 6 5 49 11 0 0 0 20 1 5 6 0 50 12 0 0 0 21 1 5 5 5 51 13 0 0 0 22 1 5 5 5 52 14 0 0 0 2 1 5 5 0 53 15 0 0 0 24 2 0 5 0 54 16 0 0 0 25 2 0 4 5 55 18 0 0 0 26 2 0 4 0 56 20 0 0 0 27 2 5 4 0 57 22 0 0 0 28 2 5 3 5 58 26 0 0 0 29 3 0 3 5 59 32 0 0 0 30 60 OF oo C 20 Rev 1 1 11 15 94 Table 45 Normalized MIDI Pan Input Table
73. 7 dynamic noise reduction 7 12 dynamic range 2 1 echo 7 14 edit buffer 4 1 AGC parameters Table 19 C 10 ARM sense paramters Table 20 C 11 compressor paramters Table 18 C 10 de esser Table 14 C 9 delay processor Table 16 C 9 expander parameters Table 17 C 10 filter 1 Table 11 C 8 filter 2 Table 12 C 8 filter 3 Table 13 C 8 global parameters Table 10 C 8 log converter paramters Table 21 C 11 miscellaneous parameters Table 25 C 13 noise reduction Table 15 C 9 output parameters Table 22 C 11 index i realtime block 1 Table 23 C 12 realtime block 2 Table 24 C 13 editing parameters not accessible from the front panel 7 9 equalization 7 11 adjectives used 2 6 boost vs cut 7 12 how to 7 12 peak shelf 2 6 tutorial 2 2 using 2 6 equalizer controls 4 3 parameter group 4 4 Er nn in display 9 1 expander controls 4 6 parameters Table 17 C 10 settings 7 13 external clock 4 15 7 7 figure analog hookup 6 2 connector wiring 3 4 delay block diagram 7 6 digital sources 6 4 DS and NR block diagram 7 6 dynamics block diagram 7 5 frequency ranges 2 5 line to mic pad 7 2 peak amp shelf EQ curves 2 6 sequence of processing 7 6 filter 1 Table 11 C 8 filter 2 Table 12 C 8 filter 3 Table 13 C 8 flanging 7 14 gain setting general 7 11 gain reduction display 4 4 global parameter offset table C 7 global parameters Table 10 C 8 headroom display 4 2 tutorial 2 1
74. 85 190 195 200 90 205 210 215 220 225 230 235 240 245 250 100 255 260 265 270 275 280 285 290 295 230 110 235 240 245 250 255 260 265 270 275 280 120 285 290 295 300 305 310 320 330 Rev 1 1 11 15 94 Table 41 Delay Feedback Table Negative then Positive feedback dB 0 1 2 3 4 5 6 7 8 9 0 0 0 0 0 0 1 0 2 0 3 0 4 0 5 1 0 1 5 1 5 10 2 0 2 5 3 0 3 5 4 0 4 5 5 0 5 5 5 5 6 0 20 6 5 7 0 7 5 8 0 8 5 9 0 9 0 9 5 10 0 11 0 30 12 0 13 0 14 0 15 0 16 0 16 0 17 0 18 0 19 0 20 0 40 22 0 24 0 26 0 26 0 28 0 30 0 32 0 34 0 36 0 38 0 50 40 0 42 0 42 0 44 0 46 0 48 0 50 0 55 0 60 0 65 0 60 65 0 70 0 80 0 90 0 100 90 0 80 0 70 0 65 0 65 0 70 60 0 55 0 50 0 48 0 46 0 44 0 42 0 42 0 40 0 38 0 80 36 0 34 0 32 0 30 0 28 0 26 0 26 0 24 0 22 0 20 0 90 19 0 18 0 17 0 16 0 16 0 15 0 14 0 13 0 12 0 11 0 100 10 0 9 5 9 0 9 0 8 5 8 0 7 5 7 0 6 5 6 0 110 5 5 5 5 5 0 4 5 4 0 3 5 3 0 2 5 2 0 1 5 120 1 5 1 0 0 5 0 4 0 3 0 2 0 1 0 0 0 2 3 4 5 6 7 8 9 0 600 _ 600 600 600 800 800 800 800 1000 1000 10 1000 1000 1500 1500 1500 1500 2000 2000 2000 2000 20 2500 2500 2500 2500 3000 3000 3000 3000 3500 3500 30 3500 3500 4000 4000 4000 4000 4
75. 9 0 28 0 27 0 26 0 25 0 25 0 24 0 23 5 23 0 23 0 70 22 5 22 0 21 5 21 0 21 0 20 5 20 0 19 5 19 5 19 0 80 18 5 18 0 17 5 17 5 17 0 16 5 16 0 16 0 15 5 15 0 90 14 5 14 0 14 0 13 5 13 0 12 5 12 0 12 0 11 5 11 0 100 10 5 10 5 10 0 9 5 9 0 8 5 8 5 8 0 7 5 7 0 110 7 0 6 5 6 0 5 5 5 0 5 0 4 5 4 0 3 5 3 5 120 3 0 2 5 2 0 1 5 1 5 1 0 0 5 0 0 C 14 Rev 1 1 11 15 94 Table 29 Parametric Bandwidth Table in octaves 0 1 2 3 4 5 6 7 8 9 o 0 050 0 050 0 050 0 050 0 055 0 055 0 055 0 060 0 060 0 060 10 0 065 0 065 0 065 0 070 0 070 0 070 0 075 0 075 0 075 0 075 20 0 080 0 080 0 080 0 085 0 085 0 085 0 090 0 090 0 090 0 095 30 0 095 0 095 0 10 0 10 0 10 0 10 0 20 0 20 0 20 o 30 40 0 30 0 30 0 40 0 40 0 40 0 50 0 50 0 50 0 60 0 60 50 0 60 0 60 0 70 0 70 0 70 0 80 0 80 0 80 0 90 0 90 60 0 90 10 10 10 1 1 1 1 1 1 1 1 1 2 12 70 12 13 1 1 1 4 1 4 1 4 1 5 1 5 15 80 16 16 16 1 6 1 7 1 7 1 7 1 8 1 8 18 90 19 19 1 9 2 0 2 0 2 0 2 1 2 1 2 1 21 100 22 2 2 22 2 3 2 3 2 3 2 4 2 4 2 4 25 110 2 5 25 26 26 26 26 2 7 2 7 2 7 28 120 28 28 29 29 2 9 30 3 0 3 0 Table 30 Frequency Table Hz 0 1 2 3 4 5 6 7 8 9 0 31
76. AES EBU external sample clock DACs from DSP O Under globals click to InP Set left input LEDs to digital O Click globals again to C or CLCI or CLCE Clock and signal source will be forced to external AES EBU Rev 1 1 11 15 94 4 17 This page is blank believe it or not Rev 1 1 11 15 94 4 18 ev 1 1 11 15 9 Note The next two sections are presented on their own page This makes it easy to remove the page should you want to prevent other readers from knowing how to initialize the 602 or bypass the front panel security features 4 11 Restoring Factory Presets Caution Do not reinitialize the 602 to the factory set values unless this is what you really want to do Reinitializing erases all user programs presets 1 to 128 and there is no way to recover your programs once you have done this You can reinitialize all programs to their factory set values by holding down the load switch while applying power to the 602 4 12 Disabling the Front Panel Some applications may require disabling the front panel Broadcasters using the 602 as their on the air mic processor may want to make the unit impervious to adjustment The 602 has three levels of security None This is what you normally get when you turn the 602 on Partial Disables everything except the Wheel and the load button Since no other buttons operate it is impossible to alter programs or to overwrite other programs Maximum Disables everything Nothing on t
77. AM Program 128 12900 12999 ROM Program 1 100 199 25600 25699 ROM Program 128 25700 25835 Global Parameters Global Parameter Description Value dec Offset dec 25700 MIDI Channel 0 15 127 0MNI 25701 MIDI Unit 0 126 127 all 25702 Current range 0 127 LSB 7 bits program number 0 255 25703 Current Pgm 0 1 MSB 2 bits of program number 25704 Reserved 25705 Signal Source AES input 1 toL amp R 1 AES input 2toL amp R 2 AES inputs summed toL amp R 3 AES inputs to L amp R stereo 4 Analog input 1 to L amp R 5 Analog input 2 toL amp R 6 Analog inputs summed to L amp R 7 Analog inputs to L amp independent gain controls stereo 8 Analog inputs to L amp R ganged gain controls stereo 25706 Signal Clock BITO 0 DSP gt DAC O ADC gt MCLK Configuration 1 AES gt DAC 1 AES gt MCLK BIT2 0 ADC gt INPUT 0 44 1 kHz sample rt 1 AES gt INPUT 1 48 0 kHz sample rt 25707 MIDI echo O no echo 1 echo 25708 Memory protect 0 no protection 1 protected 25709 Reserved 25710 Reserved 25711 Front panel lockout O enabled 85 partial 127 maximum 25712 Reserved 25713 Current sample rate 1 48khz 2 44 1khz read only 25714 5 fullleft pan see below 25774 5 center 25834 5 full right pan The pan tab
78. ATIO and the difference between the threshold setting and the actual signal level 4 5 4 Compressor Parameter Block The compressor reduces its gain for any signal level above the threshold setting The COMPRESSOR switch s LED indicates that the compressor is active When editing the compressor switch LED flashes otherwise it reflects the state of the compressor in or out Whenever editing the compressor or AGC parameters the output LED display changes to read gain reduction The compressor block and the AGC block are mutually exclusive you can only use one of them at a time There is no output gain control the 602 computes the correct amount of makeup gain based on the threshold and ratio settings although the auto makeup gain feature can be defeated and the amount of makeup gain can be set manually The shape of the knee of the gain reduction curve can be adjusted via MIDI or the realtime editor see Appendix ATTACK Switch Adjusts the ATTACK time milliseconds of the compressor time required for an above THRESHold signal to cause gain reduction RELEASE Switch Adjusts the release time time in milliseconds required for the gain to return to the below threshold value RATIO Switch Controls the compression gain ratio compression ratio The range is from 1 1 out to 10 1 A 10 1 ratio means that a 10 dB input change results in a 1 dB output change provided that the level of the entire change was above the threshold se
79. B hotter 1 mw 600 ohms 1 mw 600 ohms 0 775V RMS 0 775V open circuit Note open circuit small V dBV 1V open circuit Note capital V dBu 0 775V RMS open circuit Same as dBv Becoming more common because of confusion between v and V The bel had its origin in the bell Telephone Labs where workers needed a convenient way to express power losses in telephone lines as power ratios Because the bel is a power ratio of 10 and this is a rather large ratio it is convenient to divide it into tenths of bels or decibels abbr dB Ten dB is 1 bel thus the decibel is ten times the common log of the ratio of two powers The decibel was originally called the transmission unit or TU by the Bell Labs people TAD severely abridged The decibel is commonly used as a means of expressing audio signal levels In dynamic range processors like compressors and limiters their input to output relationship or compression ratio is a plot of the unit s input signal in dB to the unit s output signal also in dB Since the decibel represents a ratio of Rev 1 1 11 15 94 D 3 two quantities when discussing absolute signal levels it is important to know what reference quantity was used De Esser A de esser is a special type of compressor that operates only at high frequencies usually above 3 or 4 kHz It is used especially in the broadcast industry to reduce the effect of vocal sibilant sounds which are normally too strong when si
80. C Leveler AGC over a 70 dB range with adjustable gain platform and up to 4 1 ratio AGC shall provide auto release threshold ratio attack and release controls Downward Expander Downward expansion with up to 1 8 ratio Expander shall provide threshold ratio attack and release controls The output block shall provide level and panning for the output signal Both functions are user or MIDI programmable The level control shall operate in the digital domain over a 18 dB range The panpot shall also operate in the digital domain with a sine cosine characteristic law The ISP shall provide easy access to all user functions via a non hierarchical parameter selection and modification scheme There shall be a minimum of menus Every major parameter shall be accessible via a button press and subsequent adjustment of the parameter wheel Rev 1 1 11 15 94 E 1 The ISP shall provide a full MIDI implementation with the unit responding to the following messages Program Change Control Change Pitch Bend After Touch System Exclusive Sysex The MIDI implementation via MIDI Sysex Control Change and Program Change shall provide access to all major operating parameters of the ISP and real time editing capabilities shall be provided to allow real time parameter change during operation The ISP shall be capable of accepting line level signals ranging from 4 to 18 dBu The line input characteristics shall be 20 balanced bridging
81. DI specification Threshold A parameter commonly associated with dynamics processor and used to refer to a signal level at which processing begins or ends In a compressor the threshold level is that signal level where the change in level at the output no longer equals the change in level at the input Rev 1 1 11 15 94 D 9 Variable D Microphone A variable D microphone is a directional microphone having a multiplicity of entrances for off axis sounds Variable D microphones exhibit proximity effect although not to the degree that single D microphones do The term Variable D is a trademark of Electro Voice D 10 Rev 1 1 11 15 94 D 2 Bibliography For further research the following books may be useful The Audio Dictionary Second edition Glenn D White Copyright 1991 University of Washington Press Seattle Washington This revised edition contains extended definitions of many of the terms used in this manual In addition the appendices should provide many hours of enjoyable reading The book is available directly from the publisher 206 543 8870 or from Old Colony Sound 603 924 6371 Principles of Digital Audio Ken C Pohlmann Copyright 1989 Howard W Sams amp Company Indianapolis IN This is a good reference on the ins and outs of digital audio The author covers basics as well as advanced topics The book is available at major booksellers like Tower Books or from the publisher 317 298 5699 The following publicati
82. E 31 TABLE 32 TABLE 33 TABLE 34 TABLE 35 TABLE 36 TABLE 37 TABLE 38 TABLE 39 TABLE 40 TABLE 41 TABLE 42 TABLE 43 TABLE 44 TABLE 45 TABLE 46 TABLE 47 List of tables EDIT B FEER DATA REQUEST estet od e Bas ea P na ra a a de redu C 4 EDIT BUFFER DATA RESPONSE C 4 EDIT BUFFER DATA noi et e d d E a C 4 PROGRAM SETUP DATA nennen nnne nennen nnn nnne nennen nr enne nnne nnne nen C 5 DATA RESPONSE dette tv et a ve idum a e P vari dora e gan C 5 PROGRAM SETUP DATA WRITE C 6 IDENTIFY REQUEST e hr pte re et Pe aat la C 6 IDENTIFY kk nn ananasa C 6 PARAMETER MAP bul die ite ibn jena ke e ew trt a pe tr be d a edu C 7 gentes ie nine DUER Sova aa Ta INE VAS DIA RR se C 8 FIETER em 8 VETER ru MEC EDD Mene C 8 abide C 8 DE ESS PROGESSOR EE a od ree bx d iru C 9 NOISE REDUCTION 9 DELAY PROGESSO Riss la b nti test tei tea preter C 9 EXPANSION PARAMETERS tix el vd pa os ds ed vd a te p C 10 COMPRESSION PARAMETERS e nennen nnne nenne dudes
83. F PAN INPUT OUIPUT NUT DUNT MIC CLOCK ANALOG CLOCK GAIN OSC S DIG MAL INPUT INPUT SELECT in e LINE O CIRCUITRY S MIC MIC INPUT DUAL OUTPUT LEFT INPUT 2 9 PAD gt s DPI DE D A MUTE 6 CONVERTER RELAYS gt 9 Q DUAL Te LINE ER GAIN CONVERTER V UNE o o OUTPUT LINE DELAY INPUT gt tO RAM NE g RIGHT AA Ld REV E Figure 7 2 Overall block diagram 7 3 Rev 11 11 15 94 7 3 Block Diagrams On the preceding and following pages you can find the block diagrams for the de esser dynamic noise reduction dynamics processors delay processor and the entire 601 Please take a moment and take note of the following 7 3 1 Overall Block Diagram Refer to Figure 7 2 uud uu Two DSP chips handle all of the signal processing functions The AES EBU or S PDIF inputs and outputs may be re configured to connect between the DSP section and the D A converter The external digital inputs may also be used for an external clock reference Presets and global parameters are stored in battery backed up RAM Both audio channels are always processed together It is not possible to separate the two channels 7 3 2 Sequence of Processing Note the order of the different signal processors Figure 7 3 Sequence of Processin
84. H PIN 3 ZLOW TO BALANCED IN MALE XLR PIN 1 GROUND PIN 2 HIGH PIN 3 LOW FROM BALANCED OUT TIP MALETRSPLUG TIP HIGH RING LOW SLEEVE GROUND RING TO UNBALANCED IN FROM TRANSFORMER COUPLED OR FLOATING BALANCED OUTPUT BRI MALE TSPLUG B TP HIGH FROM UNBALANC ED OUT x e XI SLEEVE GROUND MALE TS PLUG LOW SUR cr SLEEVE G ROUND LOW TO BALANCED IN FROM BALANCED OUT TP _ w MALE TRSPLUG TIP HIGH TERMINAL STRIP RING LOW HIGH it SLEEVE G ROUND C LOW GROUND j SLEEVE TO BALANCED IN FROM UNBALANCED OUT TERMINAL STRIP HIGH F LOW TERMINAL STRIP xy GROUND m HIGH NEUSS 8 e x GROUND FROM NON TRANSFORMER ELEC TRONIC BALANCED OUTPUT TYPICALOF SYMETRIX PRODUCTS TO UNBALANC ED IN FEMALE XLR MALE TS PLUG PIN 1 2GROUND LOW NE H TIP 2 HIGH PIN 2 2 HIGH SLEEVE GROUND PIN 3 ZNOTUSED LOW REV B Figure 3 1 Input and output connector wiring These diagrams represent the majority of connectors used in modern audio equipment Locate the source connector in the left column and match it up with the destination connector in the right column Wire your cable according to the diagrams Rev 1 1 11 15 94 3 6 Digital I O Considerations The 602 has two similar but di
85. LED flashes this indicates the loss of digital data at the digital input if a mode requiring either digital data or digital clock has been selected 4 4 Parametric EQ Block The parametric EQ block encompasses a full function three band nm parametric equalizer All three bands provide reciprocal peak dip y e rm 5 equalization and bands 1 and 3 may be individually switched to je Ow shelving curves For flexibility each equalizer band covers the sies 5 entire frequency range The rate of change for level and frequency bandwidth and center frequency may be altered by holding down the LEVEL or FREQ switch until the display shows rt 4 4 4 EQ Band Select BAND 1 Band 1 is a bandpass peak dip or low frequency shelf The EQ range is 18 dB to 50 dB 31 Hz to 21 11 kHz 05 to 3 octaves bandwidth Q 29 to 0 4 Pressing this switch toggles the Band 1 equalizer between in and out BAND 2 Band 2 is a bandpass peak dip The EQ range is 18 dB to 50 dB 31 Hz to 21 11 kHz 05 to 3 octaves bandwidth Q 29 to 0 4 Pressing this switch toggles the Band 2 equalizer between in and out Rev 1 1 11 15 94 4 3 BAND 3 Band 3 is a bandpass peak dip or high frequency shelf The EQ range is 18 dB to 50 dB 31 Hz to 21 11 kHz 05 to 3 octaves bandwidth Q 29 to 0 4 Pressing this switch toggles the Band 3 equalizer between in and out 442 EQ Parameter Group The switches
86. M O POWER ANALOG TAPEDECK ANALOG MIXER ANALOG SOURCE Figure 6 2 Using the 602 to process a source MIDI OUTPUT INPUT BO O 602 DIGITAL PROCESSOR Symetrix en Oe 125W MAXIMUM DIGITAL sesen 5 AGIN ey N DIGITAL pod E SPDIF geje Ole A ANALOG TAPEDECK DESTINATION EI tit Figure 6 3 Using the 602 to process an entire mix POWER ANALOG SOURCE En 6 3 Rev 1 1 11 15 94 602 STEREO DIGITAL PROCESSOR MIDI DIGITAL TPUT bi ones 7 lt Symetrix OUTPUT INPUT INPUT THIS UNIT CONTAINS NO USER SERVICEABLE PARTS N SPDIF jese Iw 125W MAXIMUM 2 s POWER DIGITAL AUDIO DIGITAL AUDIO WORKSTATION SOURCE RECORDER O crx liii Rev B Figure 6 4 Using the 602 with a digital source and digital destination 6 3 Settings for Digital Sources For a digital source Figure 6 4 shows the connections required Set the controls and switches on the front panel as follows l After all rear panel input and output connections have been made apply power to the 602 and depress the Power switch When the display shows program number 1 proceed to the next step 2 Depress the Global switch once The display reads GAIn Rotate the Wheel to set the digital input gain to O 3 Depress t
87. MIX Switch Sets the MIX ratio 96 between the direct and the lowpass filtered delayed signal Holding down the Mix switch rt allows editing the rate of change of the mix gains This parameter s rate of change is shared with the FEEDBACK switch DELAY Switch dL 1 dL 2 duAL The two delay lines can be modified either individually or in tandem The display indicates the delay time in milliseconds and the DELAY switch toggles through the adjustment modes Adjustment in tandem duAL changes the delay time of both delays simultaneously This mode maintains any difference in delay time between the two delays For monophonic delays set both delay lines to the same delay time prior to entering duAL mode In duAL mode the display indicates the delay time of delay number one Note the delay time increments used change depending upon the current delay time At short increments the change increment is small 1 ms growing to 2 ms above 30ms growing to 5 ms above 100 ms Above 310 milliseconds the delay time increment is 10 ms Thus as you traverse the range of delay settings the increment between the two delays changes according to the increment used for that particular delay time You can see the actual setting of each delay by toggling the DELAY switch through dL 1 and dL 2 Refer also to the delay time table found in Appendix C Rev 1 1 11 15 94 4 12 ev 1 1 11 15 9 DELAY switch cont d FEEDBACK Switch FILTER Switch RATE S
88. P PARAMETRIC BANDWIDTH TABLE IN OCTAVES PROGRAM SETUP DATA REQUEST PROGRAM SETUP DATA WRITE REALTIME MIDI BLOCK 1 REALTIME MIDI BLOCK 2 REALTIME ScALING TABLE SIDECHAIN LOOKAHEAD TIME MS TABLE TIME CONSTANT TABLE vi TABLE 19 TABLE 34 TABLE 20 TABLE 35 TABLE 28 TABLE 26 TABLE 27 TABLE 37 TABLE 18 TABLE 33 TABLE 5 TABLE 14 TABLE 47 TABLE 44 TABLE 41 TABLE 42 TABLE 16 TABLE 40 TABLE 1 TABLE 2 TABLE 3 TABLE 32 TABLE 17 TABLE 11 TABLE 12 TABLE 13 TABLE 30 TABLE 10 TABLE 46 TABLE 7 TABLE 8 TABLE 21 TABLE 38 TABLE 25 TABLE 15 TABLE 45 TABLE 22 TABLE 31 TABLE 9 TABLE 29 TABLE 4 TABLE 6 TABLE 23 TABLE 24 TABLE 43 TABLE 39 TABLE 36 C 10 C 16 C 11 C 17 C 14 C 14 C 14 C 17 C 10 C 16 C 5 22 20 C 19 C 19 C 9 C 18 4 4 16 C 10 C 8 C 8 C 15 22 C 6 C 11 C 18 C 13 C 9 C 21 C 11 C 15 C 7 C 15 5 C 6 C 12 C 13 C 19 C 18 C 17 DWAMGPRXESNG SYSTEM PRESEIS PREK m em TUE NC Y RE CT JOLO 55099955556 rix 525252522220 1 Introduction The Symetrix 602 Stereo Digital Processor is a dual channel digital signal processor intended for use in a variety of recording broadcast live sound and post production applications Acting as a bridge between the analog and digital domains the 602 accepts stereo line l
89. ROM 602 Type Returns Short One parameter in Edit Buffer Off Value Range hex dec 0 lt gt FO 1 mfrID 0 2 lt mfdiD 1 gt 3 mfdID 2 5E 4 device type 2 5 lt unit channel gt O 7E al 16 command 1 7 lt parameter gt 0 7F 8 parameter value O 7F 9 lt EOX gt F7 Table 3 Edit Buffer Data Set DATA TO 602 Type Returns Short One parameter in Edit Buffer Off Value Range hex dec lt sysex gt 1 mfrID 0 gt 2 lt mfrID 1 gt 3 mfrID 2 gt 5E 4 device type 2 5 lt unit channel gt 0 7 7F al 16 command IC 7 lt parameter offset gt 0 7F 8 parameter value O 7F 9 lt gt F7 Rev 1 1 11 15 94 Table 4 Program Setup Data Request REQUEST TO 602 Type Requests Long Block of parameters by address See Parameter Map for address Range hex Off Value hex Range hex dec 0 lt gt FO 1 lt mfrID 0 0 2 lt mfrID 1 0 3 lt mfrID 2 5E 4 device type 2 5 unit channel O 7E 7F all 6 command 12 7 Offset to start of data top 2 bits 8 Offset middle 7 bits 9 Offset bottom 7 bits 10 number of bytes special case top 2 bits 0 0 0 transmit one edit buf
90. Symetrix Inc 14926 35th Avenue West Lynnwood Washington 98036 voice 206 787 3222 800 288 8855 fax 206 787 3211 602 Stereo Digital Processor Owner s Manual Manual Rev 1 1 11 15 94 Software Rev 2 03 Part number 530602 Subject to change at our whim and without notice Copyright c 1992 1994 Symetrix Inc All rights reserved Batteries not included Ground isn t ground Available at finer studios everywhere No part of this manual may be reproduced or transmitted in any form or by any means electronic or mechanical including photocopying recording or by any information storage and retrieval system without permission in writing from the publisher Production Information This document was written using Microsoft Word for Windows V2 0 and 6 0 The drawings and graphs in this manual were prepared using Corel Draw V2 0 Autocad V12 and Autoscript V5 then imported into Word for Windows via encapsulated PostScript files All page makeup occurred within Word for Windows Body text is set in Bookman 10pt and Section Heads are set in various sizes of Helvetica Bold Helvetica Narrow was used for Figure and Table captions This manual was printed directly from PostScript files generated by Word for Windows on a Xerox Docutech printer This unique device is actually a laser printer capable of 600 dpi resolution with a page throughput that rivals a high speed photocopier As a result every page is a first g
91. The preferred terminology for the actual process is boost cut rather than equalize attenuate In Britain preferred usage is lift dip Equalizers that can have peaks in their response curves such as parametric and graphic equalizers are characterized by the relative sharpness of the peaks The Q of a filter is a measure of this sharpness and is defined as the center frequency divided by the half power bandwidth For instance a one third octave filter centered at 1000 Hz will be 232 Hz wide at its half power points Its Q is thus 1000 232 or 4 31 Filters with Q values much higher than this tend to ring distorting transients and call attention to themselves when used in sound systems See also Q parametric equalizer shelving equalizer TAD abridged Expander A device for increasing the dynamic range and reducing the apparent noise of a signal A volume expander decreases the system gain as the signal level decreases making soft signals softer still This results in an apparent noise decrease because the relative level between the softest and loudest sounds is greater If the noise level is already low enough that the signal will mask it in the loud passages the expansion will put the low end of the dynamic range at a point where the ear has reduced sensitivity making the noise less audible TAD This definition for an expander is commonly used for downward expanders as well Filter filter is a type of equalizer which is designed to redu
92. ailor the response These are stored on a per program basis and can be edited through MIDI sysex or through the realtime MIDI functions on the front panel see Chapter 7 Regardless of the actual range of values required internally to the 602 the externally accessed range of values is mapped across the range of O to 127 decimal The parameter value can be scaled and offset to shift the value into a useful range The range of the scale factor is plus or minus 4 and the Realtime Scaling Table maps the O to 127 range used in the edit buffer to the stored scale factor value The Realtime Scaling Table may be found later in this chapter For instance to represent a scale factor multiplier value of 2 4 refer to the Realtime Scaling Table locate the value 2 4 within the table grid and read the step number from the row and Rev 1 1 11 15 94 C 1 column headings 119 in this example From MIDI you send an edit buffer data set command 1Ch with a parameter offset value of 4Ah and a parameter value of 77h 119d The offset value is added after multiplication by the scale factor and is stored with its own offset of 64 The stored offset is doubled before being applied to the realtime MIDI value The offset is stored in the 602 as an unsigned value having a range of 0 127 Each step in the stored value between and 127 represents an offset increment of 2 and the actual offset is derived from the stored offset as follows dec offset v
93. alities If you re a musician it means that the settings of the 602 can be changed from note to note measure to measure during the solo between songs get the picture The 602 s MIDI capability can also be used for dynamic parameter control realtime control via MIDI continuous controller and parameter editing making parameter changes via MIDI Both of these activities require an external MIDI controller or a MIDI equipped computer 2 13 Program Memory The 602 has 256 memory locations for program storage The first 128 locations are reserved for user memory the last 128 locations are reserved for the factory supplied programs You may recall any factory program or any other stored program edit it modify any of its parameters and store the result into one of the user memory locations Later these programs may be recalled via the front panel or via MIDI for reuse or further editing 2 10 Rev 1 1 11 15 94 3 Technical Tutorial This section discusses a multitude of things all related to getting signals in and out of the 602 3 1 Matching Levels vs Matching Impedances In any audio equipment application the question of matching inevitably comes up Without digging a hole any deeper than absolutely necessary we offer the following discussion to hopefully clarify your understanding of the subject Over the years we have all had impedance matching pounded into our heads This is important only for ancient audio systems po
94. ally these settings may be a good Neve 3 0 starting point if one of these equalizers is within your SSLG 14 28 frame of reference parametric equalizer offers perhaps the greatest SE ds flexibility of any type of equalizer however it can be more Figure 7 6 Bandwidth Specs for some difficult to arrive at a setting than with other equalizers A popular egualizers good strategy for setting any equalizer is to set the level control for maximum boost then vary the FREQUENCY and BANDWIDTH until you locate the portion of the spectrum that you wish to modify Then refine the setting of the LEVEL control for that band Next refine the setting of the BANDWIDTH control You may have to go back and forth 1 The source for these numbers is actual performance graphs published in the following article EQ Empirically Keith Andrews Studio Sound magazine December 1991 The API and Focusrite equalizers were measured at Symetrix Rev 1 1 11 15 94 7 11 between LEVEL and BANDWIDTH to find the magic setting Toggling the band switch between in and out can help too It is much easier to hear changes in amplitude level than it is to hear bandwidth changes It is also easier to hear the abundance of something rather than the absence of the same thing Even if you intend to apply cut negative level to a particular frequency it is still easier to find that frequency by boosting first tuning second and resetting the boost cut last according
95. an is that this is the minimum load impedance that they would like their gear to see In most cases seeing a output impedance figure of 10 000 10K ohms or higher from modern equipment that requires power batteries or AC is an instance of this type Rev 1 1 11 15 94 3 1 of rating If so then the input impedance of the succeeding input must be equal to or greater than the output impedance of the driving device Symetrix equipment inputs are designed to bridge be greater than 10 times the actual source impedance the output of whatever device drives the input Symetrix equipment outputs are designed to drive 600 ohm or higher loads 600 ohm loads are an archaic practice that won t go away You don t need to terminate the output with a 600 ohm resistor if you aren t driving a 600 ohm load If you don t understand the concept of termination you probably don t need to anyway The two facts that you need to derive from this discussion are l Match signal levels for best headroom and signal to noise ratio 2 For audio impedance matching is only needed for antique equipment and power amplifier outputs In all other cases ensure that your inputs bridge are in the range of 2 to 200 times the output source impedance your outputs 3 2 Signal Levels The 602 is designed around studio professional line levels 4 dBu or 1 23 volts The unit is quiet enough to operate at lower signal levels such as those found in semi pro or musical instrume
96. and must be destroyed Some of the preset programs modify parameters like the compressor knee that are accessed via the real time editor For this reason a program built on one of the factory programs other than program 256 may work differently than one built on program 256 It is a good idea to find a preset program that is close to what you want then modify it and save it Rev 1 1 11 15 94 G 8 Notes Rev 1 1 11 15 94
97. attn18 Table 26 C 14 attn82 Table 27 C 14 comp exp knee Table 37 C 17 compressor paramters Table 18 C 10 compressor ratio Table 33 C 16 data response Table 5 C 5 de esser Table 14 C 9 default pan Table 44 C 20 delay feedback Table 41 C 19 delay filter frequency Table 42 C 19 delay processor Table 16 C 9 delay time Table 40 C 18 edit buffer data request Table 1 C 4 edit buffer data response Table 2 C 4 edit buffer data set Table 3 C 4 equalizer bandwidth 7 12 expander parameters Table 17 C 10 expander ratio Table 32 C 16 filter 1 Table 11 C 8 filter 2 Table 12 C 8 filter 3 Table 13 C 8 index iii frequency Table 30 C 15 global parameter offset Table 9 C 7 global parameters Table 10 C 8 identify request Table 7 C 6 identify response Table 8 C 6 inaccessable parameters 7 9 input modes 1 1 input output clock 7 7 installation requirements 7 1 log converter paramters Table 21 C 11 makeup gain Table 38 C 18 memory protection 4 16 miscellaneous parameters Table 25 C 13 noise reduction Table 15 C 9 normalized pan input Table 45 C 21 output level Table 31 C 15 output parameters Table 22 C 11 parameter map Table 9 C 7 parametric bandwith Table 29 C 15 polarity 3 3 program setup data request Table 4 C 5 program setup data write Table 6 C 6 realtime block 1 Table 23 C 12 realtime block 2 Table 24 C 13 realtime scaling Table 43 C 19 recogniz
98. c noise reduction NR downard expander compressor and AGC Whenever one of the dynamics processors has been selected for edting the output LED display changes to a gain reduction display Note of the dynamics blocks use a threshold parameter Unlike analog processors that you may be familiar with each of the threshold settings in the 602 reference to digital clipping full scale rather than to some nominal signal level like O dBu This means that Rev 1 1 11 15 94 4 4 ev 1 1 11 15 9 you may not be able to directly translate threshold settings that you are familiar with from the analog world to the digital world 4 5 1 Dynamic Noise Reduction Block The dynamic noise reduction NR block uses a variable frequency low pass filter to perform single ended noise reduction The NR block is a feedback system the amount that the filter opens is self limiting and dependent on the high frequency content of the input signal and the THRESHold setting At higher THRESHold settings there will always be some high frequency loss NR Switch Toggles the NR between active and out When active the FREQ and THRESH switches are active When editing the NR switch LED flashes otherwise it reflects the state of the NR in or out Whenever NR is currently being edited the output headroom LED display changes to indicate gain reduction FREQ Switch Sets the resting frequency of the NR the 3 dB point of the dynamic lowpass filter when
99. c range of a sound is the ratio of the strongest or loudest part of the weakest or softest part it is measured in decibels A full orchestra may have a dynamic range of 90 dB meaning the softest passages are 90 dB less powerful than the loudest ones Dynamic range is a power ratio and has nothing to do with the absolute level of the sound An audio signal also has a dynamic range which is sometimes confused with signal to noise ratio Rarely is the dynamic range of an audio system as large as the dynamic range of an orchestra because of several factors The inherent noise of the recording medium determines the softest possible recorded sound and the maximum signal capacity of the system clipping level limits the loudest possible sound Many times an extremely wide dynamic range is not desirable e g in radio broadcasting for listening in cars and broadcasters frequently use compressors and limiters to reduce the dynamic range of the signals before they are transmitted This type of signal processing distorts the music in a more or less noticeable way symphonic music being most sensitive to it TAD Echo Commonly used incorrectly to mean reverberation echo technically is a discrete sound reflection arriving at least 50 milliseconds after the direct sound It also must be significantly above the level of the reverberation at that time Echo chambers are reverberation rooms which are carefully designed to be without echoes If an actual
100. ce the energy at a certain frequency or in a certain frequency band Filters always act as subtractive devices never adding anything to a signal at least they should not The most common types of filters are analog filters which operate on signals directly Rev 1 1 11 15 94 D 5 Digital filters operate on signals which have been digitized They are purely mathematical performing a series of arithmetic operations on the digital words In a sense digital filters are synthesized filters digital techniques being used to emulate or simulate analog filters Digital filters have the advantage of being drift free They always do their job in exactly the same way They can be designed for nearly any desired characteristics in the frequency domain and in phase response TAD abridged Flanging A special effect made popular in the 1960 s where a delayed version of a signal is mixed with the signal creating a swooshing sound Flanging was first done by recording a signal on two similar tape recorders playing them back simultaneously and mixing them together The record playback sequence on the tape recorders results in a small time delay of perhaps a tenth of a second Both output signals are delayed by the same amount if the tape recorders are similar and they add together in the mixer and the sound heard is essentially the same as the signal at the input to the tape recorders To achieve the effect of flanging one recorder is slowed down a littl
101. ceable parts inside the chassis Refer all service to qualified service personnel or to the factory 6 2 Settings for Analog Sources For an analog source Figure 6 2 and Figure 6 3 show the wiring required Set the controls and switches on the front panel as follows 1 After all rear panel input and output connections have been made apply power to the 602 and depress the Power switch When the display shows program number 1 proceed to the next step 2 Depress the Global switch once The display reads GAIn Rotate the Wheel to set the digital input gain this is not the gain applied to AES EBU or S PDIF sources to O 3 Depress the Global switch again The display reads InP Rotating the Wheel selects the input source and routing as indicated by Digital CH1 Stereo CH2 LEDs and the display The input possibilities are digital or analog line input Rotate the Wheel to select the digital or analog source and your desired routing For analog stereo operation rotate the Wheel until the CH1 STEREO and CH2 LEDs illuminate and the display indicates 1 2 4 Depress the Global switch again the display reads bArl 5 Depress the Global switch again The display reads CLCI Rotating the Wheel selects the clock source for the digital processors If set to CLCI the clock source is the internal 48 kHz or 44 1 kHz crystal oscillator If the display reads CLCE the clock source is the rear panel digital input Refer to Chapter 4 for additional
102. contained in the product s operator s manual will perform according to the specifications set forth in the operator s manual Symetrix expressly warrants that the product will be free from defects in material and workmanship for one 1 year Symetrix obligations under this warranty will be limited to repairing or replacing at Symetrix option the part or parts of the product which prove defective in material or workmanship within one 1 year from date of purchase provided that the Buyer gives Symetrix prompt notice of any defect or failure and satisfactory proof thereof Products may be returned by Buyer only after a Return Authorization number RA has been obtained from Symetrix and Buyer will prepay all freight charges to return any products to the Symetrix factory Symetrix reserves the right to inspect any products which may be the subject of any warranty claim before repair or replacement is carried out Symetrix may at its option require proof of the original date of purchase dated copy of original retail dealer s invoice Final determination of warranty coverage lies solely with Symetrix Products repaired under warranty will be returned freight prepaid via United Parcel Service by Symetrix to any location within the Continental United States Outside the Continental United States products will be returned freight collect The foregoing warranties are in lieu of all other warranties whether oral written express implied or statuto
103. d Ed Buf 31 Ed Buf 26 Laval Ed Buf 29 Detector High NR Rest ON Frequency Time Frequency 5 Constant Input 1 e Energy O onstan O pu Ed Buf 27 hM Digital Controlled NR Relative Attenuators Threshold 7 Filter Ed Buf 30 Input 2 LP Output2 Rev E Figure 7 5 Noise Reduction and De Ess block Rev 1 1 11 15 04 7 5 Delay 1 Ed Buf 34 Input 1 Mix Delay 1 Filter e lt Output 1 eru ANN Modulation gt nverter Type Wet Dry Mix Ed Buf 42 Mod Depth 4 noe _e Feedback e Feedback Polarity Filter O Ed Buf 33 Modulation Ed Buf 40 Ed Bur 3 Ed Buf 37 mis EUR Rate Ed But A3 Ed Buf 41 vo rau d 9 WN Input 2 Mix Delay 2 Filter Delay 2 Ed Buf 35 lt gt Output 2 Rev D Figure 7 6 Delay block 7 3 5 Delay Block There are two delay lines each independently adjustable The feedback recirculation signals for the delay lines are cross coupled The feedback setting is always the same for both delays For clarity the diagram shows the mix pots reversed When the mix parameter is 0 both outputs are their r
104. d energy is found up to the 2 5 kHz range Music editors and others engaged listening To music over long periods find that listening fatigue can be reduced by attenuating 5th and 7th octaves by about 5 dB 2 4 4 Lisping Quality The 3 kHz range delivers a generous stimulus to the ear At very loud levels the region of greatest ear sensitivity shifts downward from 5 kHz this is why many speakers have broad peaks in this region A characteristic of low level signals peaked at 3 kHz is a lisping quality and the total inability distinguish labial sounds such as m b and v In wide range lower level systems a peak in the 3 kHz region has a masking effect on important recognition sounds and on others which lie above 4 kHz Brilliance and clarity are lost and without attenuation of this region an unconscious strain with increasing fatigue is felt according to the amount of 3 kHz boost 2 4 5 Presence Range The usual band affecting clarity in male speech is 3000 to 6000 Hz In a woman s voice the fundamentals are roughly an octave higher than a man s and a woman s range of consonant clarity lies between 5000 and 8000 Hz the high end of this range approaches a region of hearing insensitivity in humans Furthermore the total range of a woman s voice is about half that of a mans stimulating fewer hearing nerves and for this reason is consequently still weaker upon reception Wide range sounas especially those of
105. d highpass filter in its sidechain In sibilant speech the dominant frequency component is the sibilance itself Reducing the overall gain during periods of sibilance reduces the level of the sibilant 7 12 Rev 1 1 11 15 94 In mastering applications the de esser can also be useful to reduce excessive high frequency content for instance repairing a mix when the cymbals have too much high frequency content and clutter the high end of the mix Set the de esser by adjusting the THRESH level until the sibilance is no longer objectionable The de esser and the noise reduction may be used simultaneously None of the other dynamics parameters are applicable to the de esser 7 5 8 Compression The compressor generally controls peak levels and maintains a high overall average signal level Used in this manner the compressor s action is generally inaudible Compressors can also be used creatively to make a source sound louder than it really is or to create a special effect For most level control applications moderate settings yield the best results We recommend a starting point of THRESH setting sufficient to cause about 6 to 8 dB of gain reduction on peaks using a RATIO setting of 4 1 Pick an ATTACK time that allows enough of the initial sound through to not lose crispness and a RELEASE time that allows the compressor to partially recover gain reduction display almost out between words For a highly compressed sound you know the used ca
106. d in Fast First Time Setup Elsewhere in this chapter you can find operational hints and suggested settings You can find additional discussion of many of these topics in Chapter 2 Basics 7 1 Installation The 602 may be installed free standing or rack mounted No special ventilation requirements are necessary Installation Requirements Mechanical One rack space 1 75 inches required 12 5 inches depth including connector allowance Rear chassis support recommended for road applications Electrical 105 125 V ac 60 Hz 20 watts 210 250V ac 50 Hz 20 watts export version Connectors XLR 3 female for inputs XLR 3 male for outputs Pin 2 of the XLR connectors is Hot RCA female connectors for S PDIF digital I O XLR 3 male and female connectors for AES EBU digital I O 7 2 Operational Details This section describes the details of operating the 602 Usage information can be found later in this chapter The 602 accepts stereo or mono analog line level input signals converts them to 18 bit digital form splits them into left and right signals processes them through two parallel DSP chains and then converts the signals back to the analog domain The processed signals are also simultaneously available at the AES EBU or S PDIF output connectors The 602 can also process the microphone input through one channel and the line input through the other channel The control signal processing is still shared between
107. d setting EXPANDER Switch ATTACK Switch 4 6 Toggles the Expander between in and out When editing the expander switch LED flashes otherwise it reflects the state of the expander in or out When active all of the dynamics parameter modification switches are active Whenever the expander is currently being edited the output LED display changes to read gain reduction Sets the time required for the expander to terminate expansion This means that the input signal must remain above the THRESHold setting for a time that is longer than the attack time The ATTACK time ranges from 0 1ms 100 microseconds to 10 000 ms the display reads 9999 but the time is really 10 000 milliseconds or 10 seconds Rev 1 1 11 15 94 RELEASE Switch Sets the time required for the expander s gain to decay once the input signal has fallen below threshold The time displayed is the time required for full decay in response to a large below THRESHold change in the input signal The RELEASE time ranges from 100 ms 100 milliseconds to 10 000 ms the display reads 9999 but the time is really 10 000 milliseconds or 10 seconds RATIO Switch Sets the expansion gain RATIO expansion ratio The range is from 1 0 out to 8 ratio of 1 8 or 1 dB input change to 8 dB output change THRESHOLD Switch Sets the THRESHold for start of expansion Signals below this level are reduced in level by an amount dependent on the setting of the expansion gain R
108. dit buffer 49 controls the downward expander s knee width 4 5 7 3 Compressor knee control The point in the compressor s gain curve immediately above threshold is known as the knee The width of the knee may be altered to make the transition to the compressor s ultimate slope more or less gradual Edit buffer 55 controls the compressor knee width 4 5 7 4 AGC absolute threshold The AGC normally acts as a compressor having its threshold level set very low This parameter edit buffer 56 controls just how low the actual threshold is Below this level there is no AGC action 4 5 7 5 AGC knee control The point in the AGC s gain curve immediately above threshold is known as the knee The width of the knee may be altered to make the transition to the AGC s ultimate slope more or less gradual Edit buffer 60 controls the AGC knee width 4 5 7 6 ARM peak release TC This parameter edit buffer 62 affects the recovery time of the auto release monitor ARM subsystem Normal settings are in the 1 to 3 second range and the default setting is 2 5 seconds Refer also to Section 4 5 7 7 4 5 7 7 ARM integration TC This parameter edit buffer 63 affects the signal level history of the ARM subsystem Shorter time constants require higher signal peak to average ratios to trigger the AGC hold function thereby releasing the gain reduction This time constant and the time constant used for offset 62 should be in the same range of 1 to 3 seconds 4
109. dy from the shell it may help to push on the connector body from the outside through the shell Rev 1 1 11 15 94 F 2 G Presets and Other Stuff This appendix contains material that defies categorization or inclusion elsewhere You ll find things like the Preset Programs list a Programmer s Worksheet and the MIDI implementation table Rev 1 1 11 15 94 Symetrix 602 Programmer s Worksheet Program Number Programmer Program Name Source O Dig CJ O CH2 Stereo CH1 Gain 2 Gain Comments Table revised 4 4 94 Filter Block 1 O Pk O Shif Filter Block 2 Peak only Filter Block 3 O Pk O Shif OIN OOUT FILTER DE ESS DYNAMICS Freq Hz Hz Level dB dB Abs Thr dB Thr dB Bandwidth Oct Oct NR OIN OOUT Release Freg BW Level ROC io Threshold Expander O IN ms ms as Compressor O IN ms ms as AGC O IN as SEt Param SEt Param DELAY Mod Rate OUTPUT Mix Mod Depth Output Level Delay 1 Level ROC Output Pan Delay 2 Delay ROC Feedback Output Osc Type Random Sine O Triangle Output ROC Block 1 Block 2 REALTIME Source Note ROC Rate Of Change Scale Factor Offset Para
110. e increasing its time delay This is done by pressing one s thumb against the flange of the tape recorder supply reel hence the name flanging When the time delay is different for the two combined signals there will be frequencies where the phase shift is 180 degrees and the signals will cancel causing deep dips or holes in the frequency response curve This is called the comb filter effect As the speed is varied the frequency of the dips is swept across the frequency range giving the swooshing sound Attaining the most desirable effect requires an educated thumb The best effect is obtained when the signal being flanged contains frequencies over a wide range TAD See also phasing Frequency Response Also known as magnitude response is the graph of the variation in output level of a device over frequency with a constant amplitude input signal Full Scale When audio signals are converted to their digital equivalents using an analog to digital converter the signal level that causes the output of the converter to reach its digital maximum is referred to as full scale Gate A circuit which performs like a switch allowing a signal either to pass or not is called a gate The position of the gate open or closed is controlled by an applied voltage which can come from a number of different places If the level of the signal determines the gate opening it is a noise gate closing when the signal level is so low that the noise would be audible
111. e 602 and double ended like Dolby noise reduction A double ended system such as the Dolby System eliminates noise contributed between its encode and decode processors By necessity this means that you must have access to the signal before it has noise added to it and afterwards For tape recorders and their ilk this is perfect Of course if you feed a Dolby noise reduced system a noisy signal it will simply hand it back to you without any added noise of course but with just as much noise as you gave it to begin with garbage in garbage out or GIGO A single ended noise reduction system works on whatever signal you hand it Single ended systems depend on noise masking by the signal That is when the signal is present it tends to mask the noise So when the signal is quiet or absent reduce the noise by reducing the high frequency response and when the signal is present remove the high frequency rolloff and pray that the signal masks the noise If you re handed a noisy signal then a single ended noise reduction processor is your best weapon against the noise If you combine this with some careful equalization you ll probably end up with a signal that is more listenable 2 7 Downward Expansion Expansion is the process of increasing a signal s dynamic range usually by increasing the signal s level by a precise amount for every dB over a magic signal level the threshold Unfortunately this requires infinite or at least n
112. e 7th and 8th octaves 4750 5000 Hz Between the 8th and 9th octaves 6500 16 kHz Part of the 9th through the 10th octave 2 4 1 Power and Fullness In the very low bass region lies the threshold of feeling where the lowest sounds like wind room effects and distant thunder are felt rather than heard In the upper half of the first octave of this range research has shown that the fundamentals of piano organ and even the harp reach well into this range Harvey Fletcher of Fletcher Munson fame charted the sensitivity of the ear for various parts of the spectrum at levels that are lower than those of reality Fletcher s compensation curves the well known Fletcher Munson curves show that for equal loudness in this range at lower recorded and reproduced levels shows requirements for tremendous boosts on the order of 10 to 30 AB Aside from the subjective effects of this range the ability to control unwanted sounds in this range is equally important to subdue stage rumble and outside traffic noise especially important where there are subways beneath buildings Overemphasis caused by close cardioid microphone placement can cause muadiness in the overall sound attenuating cutting the very low bass region can greatly improve overall clarity 2 4 2 Rhythm and Musical Foundation In bass region most of the low grave tones of the drum and piano can be found Here we can also find the fundamentals of the rhythm section as well as the
113. e amount of change at the upper end of the dynamic range with a compressor or limiter try using a downward expander to increase the amount of change at the lower end of the dynamic range 2 8 Compression Many times a signal s dynamic range must be modified to allow it to pass through a transmission channel without clipping or becoming noisy Most often audio engineers patch in a compressor to restrict the dynamic range of a signal 2 Dolby is a trademark of Dolby Laboratories San Francisco CA USA Rev 1 1 11 15 94 2 7 A compressor is a gain control device whose output is nearly constant in spite of variations in its input level A simple analogy is you re holding the volume control for a sound system and you re told to turn it down if the level gets louder than it is now and turn it down enough that the level is the same as it is now Figure 2 4 illustrates this concept graphically Compressors can also be used pau creatively that is to create an effect In 50 000 this case the rules such as they are go out of the window A large amount of compression applied to a voice over can 30 000 create the impression of excitement or intimacy or simply help make the signal very loud in a controlled manner 10 000 which might be useful in ensuring that the voice is always heard 40 000 20 000 For most voice applications pick a moderate ratio 4 1 to 8 1 and set the E
114. e display now reads bLC Select block 0 using the Wheel Press the MIDI button again Use the Edit Buffer parameter table to locate the offset of the desired parameter We want parameter 10 Press the MIDI button again The display reads SrC Since we want the log signal level to become the controller for the filter frequency select 6 as the modulation source using the Wheel Press the MIDI button twice The display should now read oFt While listening to the 602 s output adjust the offset value until you hear the equalizer filter begin to work You can then use the SCAL parameter to alter the range of the effect Note whenever one of the realtime blocks has been set up and attached to a parameter trying to edit that same parameter from the front panel results in the message rEAL whenever the Wheel is turned This is true only for realtime blocks one and two and not for the SEt function Accessing the parameter attached to the realtime block from the front panel results in the display temporarily showing the current value of the parameter This can be useful in determining useful limits for a parameter or to view the results of offset and scale operations Rev 1 1 11 15 94 2 ev 1 1 11 15 9 B Using the Lexicon MRC to Edit Realtime MIDI Settings Most of the 602 s internal parameters may be modified remotely using MIDI In addition many parameters may be modified dynamically while the 602 is passing signal as if there was a fron
115. ear infinite headroom A simple but entirely satisfactory solution is to reduce the signal s level for every dB below a magic signal level the threshold This is called downward expansion A similar and related device is the signal gate You can think of a signal gate as a special case of a downward expander or vice versa if you must Both devices reduce their output when their input signal falls below threshold The difference is the rate not speed at which they do it The 602 s downward expander output falls at an adjustable rate for every 1 dB below threshold of the input signal A gate s output falls by a nearly infinite amount for the slightest change below threshold of the input signal You can think of a gate as a downward expander taken to the extreme or you can think of a downward expander as a subtle example of a gate Gates are generally used to remove leakage unwanted signals from nearby sources from a signal Downward expanders are used to remove extraneous noise and to increase dynamic range by making the softer parts softer Compressors or limiters for the purposes of this discussion a limiter is simply a high ratio compressor are often used to reduce dynamic range by setting an upper limit on larger signals Sometimes when you re trying to fit a signal through a transmission channel it s better to put processing to work on the lower end of the dynamic range than on the upper end In other words instead of reducing th
116. ed midi commands C 4 sidechain lookahead Table 39 C 18 time constant Table 36 C 17 troubleshooting 9 1 threshold 4 4 tips and techniques 7 10 troubleshooting 9 1 U nnn 4 15 unbalanced loads driving 3 3 user interface summary 4 1 Using the 602 7 1 warranty 10 1 wheel parameter adjustment 4 1 index iv A Editing Realtime Midi Settings The 602 has the capability to modify its parameter settings in realtime either as a function of one of the MIDI continuous controllers or from an internal control source To access the realtime MIDI settings from the front panel press the MIDI button until the display reads rEAL A long press on the MIDI button then accesses the realtime block editor Successive presses of the MIDI button access each item on the realtime MIDI submenu Each submenu item allows editing one of the realtime MIDI parameters Each menu item is described as follows bLC Block select MIDI linkage 0 1 or SEt These linkages represent the two available realtime MIDI setups This menu item selects either the first or second MIDI linkage for editing or selects an arbitrary parameter offsets 0 70 within the edit buffer for setting PAr Selects the edit buffer parameter to be edited Use the edit buffer table to look up the offset of the desired parameter The right hand bargraph output headroom gain reduction shows the parameter s current actual edit buffer value on a linear scale of O no LEDs to 127
117. edit buffer or overwrite the edit buffer with ROM program 256 then write the edit buffer over the zombie program s memory location Rev 1 1 11 15 94 A 1 When selecting one of the realtime blocks you have the third option of selecting SEt This acceesses any parameter within the edit buffer displays its value and allows you to change that value SEt has the following menu items PAr Selects the edit buffer parameter to be edited Use the edit buffer table Appendix C to look up the offset of the desired parameter The right hand bargraph output headroom gain reduction shows the parameter s current actual edit buffer value on a linear scale of O no LEDs to 127 all LEDs oFt Offset from zero to apply to source value Press LEAVE EDIT to exit this mode A 1 Realtime MIDI Example Refer to the description of Realtime MIDI Block 1 which can be found in Appendix C We ll use modulation source six the dynamics section log signal level This control source represents the logarithm of the signal level presented to the dynamics section The dynamics section uses it to drive the compressor expander and sidechains We ll use it to control the filter frequency of band 2 of the parametric equalizer block Begin by accessing band 2 of the equalizer Set the LEVEL parameter to 15 dB and the BW parameter to 0 5 Access realtime edit mode by pressing the MIDI button until the display reads rEAL Pressing the MIDI button again th
118. ement systems or radio transmitters to prevent unexpected high level signals from causing overloading and large amounts of distortion TAD Lowpass Filter A filter which uniformly passes frequencies below a certain frequency called the cutoff frequency Usually this is defined as the frequency where the amplitude response of the filter is 3 decibels below its nominal value Many early tone controls were variable lowpass filters TAD MIDI The Musical Instrument Digital Interface MIDI is a standard communications interface for use between electronic music synthesizers of various manufacture TAD MIDI is also used for controlling other peripheral devices both musical and not Besides synthesizers audio signal processors like the 602 electronic drums and lighting controllers are now MIDI controlled Noise Gate See Gate Overload An overload is said to occur when the input signal level in an audio device is so large that it drives the device out of its linear range and into distortion or clipping Overload may be continuous or may occur only on short peaks in musical waveforms The latter condition is common with certain waveforms such as sharp percussive sounds which have a peak value much greater than their average value This peak clipping as it is called must be avoided for true high fidelity recording and reproduction although a small amount of it may be quite difficult to hear in practice TAD Oversampling In some digita
119. eneration image Ain t technology grand Table of Contents Ve B ea 1 1 NANO SEA tiu ed a 1 2 1 2 Operator Safety 1 2 1 2 1 Equipment Markings eoe d bee tni ner 1 3 EA qc UD 1 3 1 3 Other Safety Information ne oa estia ertet ae Ped 1 4 2 BS ASG S M 2 1 2 1 What Does the 602 DO ss conse sevo m Eye ded ehe qe 2 1 2 2 Digital and Analog Differences rte ent cae P pe aed 2 1 2 3 Gain Setting roc A E C RI ME A E 2 1 2 4 QUAN ZAM ON o n 2 2 24 1 Power and Fullness oer o EO HUE mE 2 2 2 4 2 Rhythm and Musical 2 2 2 4 3 Telephone 2 3 Quality ap m 2 3 2 4 5 Presence Range tie uS UT 2 3 2 4 6 E EE 2 3 2 4 7 Conclusions 2 4 2 4 8 Equalizing for Speech eoe ceca rec SN DUREE nere 2 4 24 9 Peaking or Shelving HO en 2 6 2 5 DE ESS uou 2 6 2 6 Noise Reduction 2 7 2 17 Downward EXPANSION isc sie 2 7 2 9 e uM LAM commie 2 7 up c o ME 2 8 oss i AAA AMA E A Abt AA AU 2 9 2 11 M
120. ent MIDI controller Step What to do 1 Press the MACH key on the MRC The display reads MACH Press 15 on the keypad or use slider 1 to set the machine number to 15 2 Press the ENTER key on the MRC 3 Press the SETUP key on the MRC The display reads SOURCE DEST Press 1 the keypad or use slider 1 to ensure that the current setup is 10 4 Press the ENTER key on the MRC 5 Press the EDIT key on the MRC 6 Adjust the slider under the SOURCE prompt on the display so that the source is slr1 7 Adjust the slider under the DEST prompt so that the DEST is SYSEX 8 Adjust the slider under the OUT prompt so that the OUT is 1 9 Press the PAGE key Note The length of this example may at first make the whole thing seem daunting or extremely complicated It is not A vast majority of the process is highly repetitive and once you have programmed two or three sliders the pattern should begin to emerge and you can begin working from memory Note Whenever one of the realtime blocks has been set up and attached to a parameter trying to edit that same parameter from the front panel results in the message rEAL whenever the Wheel is turned This is true only for realtime blocks one and two and not the SEt function under realtime MIDI edit Rev 1 1 11 15 94 B 1 14 15 16 17 18 20 21 Make 22 23 24 25
121. epth of the modulation applied delay time of the variable delay lines 100 is maximum 0 is off Pressing this switch toggles between out and the current modulation depth setting OUTPUT SYSTEM PRESETS 4 7 Output Group 15665 The output group encompasses the switches affecting the output ofthe cme T LEA MIDI LOAD EDIT 602 BYPASS Switch The BYPASS switch puts the 602 into bypass mode This is not a hard wire bypass the signal continues to flow through the A D and D A converters the DSP processing is simply disabled GAIN PAN Switch The GAIN PAN switch sets the output gain and left right panning of the 602 A long press on the GAIN PAN switch sets the rate of change of the level functions the display indicates rt The GAIN setting is saved on a per program basis L The display indicates the output gain setting L in dB or panning The pan display indicates the percentage towards the left or right Thus 50 indicates 5096 towards the left The GAIN and PAN parameters are saved on a per program basis The PAN setting is saved on a per program basis 4 8 System Group The system switches determine global state saved with unit not with programs operating parameters including MIDI All global parameters are stored in battery backed up memory thus they are retained even in the absence of AC power 4 8 1 Global Switch The GLOBAL switch is a multi mode switch with the following modes
122. eral instruments are given separate positions by using a panpot on each one when the final mix is made TAD Parametric Equalizer An equalizer allowing control of center frequency bandwidth and boost cut See also shelving equalizer peak dip equalizer Peak Dip Equalizer An equalizer capable of providing a bandpass peak or dip as differentiated from shelving in its frequency response Peak dip equalizers are available in many forms ranging from the program equalizers found on many mixing consoles to graphic or parametric equalizers Phantom Power Condenser microphones require a preamplifier to be close by due to the extremely high impedance of the microphone itself This preamplifier is in the housing of the microphone and it needs a power source Sometimes a battery is used but more often a multi wire cable brings the audio signal from the microphone and brings the power from an external power supply to the preamp This is a rather Rev 1 1 11 15 94 D 7 bulky and expensive arrangement To eliminate the multiconductor cable frequently a scheme called phantom powering is used whereby the preamp power is carried by the same two wires that carry the signal The key to its operation is the fact that the signal is alternating current and the power is direct current and they can be separated by the action of a transformer The voltage used for phantom powering is usually 48 volts but it can vary from about 12 to 52 volts Microphones wh
123. erall signal level up Note that doing so will cost you some noise performance Finally the output digital gain LEVEL PAN switch Set this parameter so that the 2 dB output headroom LED illuminates on signal peaks The red CLIP LED should never illuminate Neither CLIP LED monitors the signal levels within the DSP blocks you must use your ears If you fear clipping within the DSP blocks you can always reduce the input digital gain gAln slightly 7 5 4 Equalization The 602 s parametric equalizer has three overlapping bands Each band can operate as a peaking or notching equalizer and bands 1 and 3 may be converted to lowpass and highpass shelving curves Each band operates over a range of 31 to 21 11 kHz with a bandwidth range of 05 octave to 3 octaves The boost and cut range for each band is 18 50 dB Since each band covers the same frequency range it is possible to apply equalization at the same frequency in three places Doing so could conceivably increase the signal level by 54 dB at one frequency You may need to reduce the input digital gain to avoid distortion Electronic considerations aside one of the contributing BW max factors to an equalizer s sound is its bandwidth Figure 7 6 lists the bandwidths octaves for several possibly API 550 n a familiar analog equalizers as found on their respective Focusrite 06 18 mixing consoles While we make no promise that the 602 will sound identic
124. ere are several conditions under which sysex messages are echoed through the 602 to the MIDI OUT connector The message s manufacturer s ID or product identifier is for a different product The message s destination number does not match The message s command is not recognized The message s destination unit number does not match The message s destination unit number is the omni value of 7fh The 602 recognizes the message but is in unit ALL mode Oooo oo 0 O O O O C If the 602 recognizes the sysex message and the message was specifically addressed to the particular 602 then the message is absorbed and not echoed Rev 1 1 11 15 94 C 3 C 1 5 Recognized MIDI Commands The 602 recognizes the following MIDI sysex messages Command Description hex 11 Edit buffer data request 1 Edit buffer data response Edit buffer data set 12 Program Setup data request 12 Data response 1D Program Setup data write 13 Identity Request 3 Identity Response Table 1 Edit Buffer Data Reguest REGUEST TO 602 Type Reguests Short One parameter by number Off Value Range hex dec 0 lt sysex gt FO 1 mfrID 0 0 2 mfdID 1 0 3 lt mfdID 2 5E 4 device type 2 5 unit channel O 7E 7 6 command 11 7 parameter offset O 7F 8 lt gt F7 C 4 Table 2 Edit Buffer Data Response DATA F
125. espective input signals When the mix parameter is maximum both outputs are the lowpass filtered delay line outputs The MIDI edit buffer parameter numbers are shown in parenthesis 7 4 System Interface The 602 can be used in a variety of ways some of which may be obvious some of which may not be so obvious The next portion of this chapter describes some of the different ways to use the 602 7 4 1 Using the 602 as a Channel Insert Device The 602 can also be used as a channel insert device with your console Use one or both of the 602 s line inputs and one or both of the 602 s line outputs If you use both line outputs then you ll need a second channel at your console for the 602 s second line output 7 4 2 Using the 602 in a Send Receive Loop The 602 can also be used in a console s send receive effects loop Drive the 602 s line input s from the console s effects send and feed one or both of the 602 s line outputs to your consoles effects returns Ensure that the 602 s delay mix parameter is set at 10096 If you use both of the 602 s line outputs for stereo then you ll need a stereo effects return or a second mono effects return for the second line output 7 6 Rev 1 1 11 15 94 7 4 3 Using the 602 as an A D Converter You can use the 602 as an analog to digital converter simply by using the analog input s and the AES EBU or S PDIF output There are however several caveats l The clock accuracy speci
126. espresso beans from Starbucks for your trouble novelness of idea limited to our opinion decision of judges is final offer void where taxed or prohibited 8 1 Broadcast Voice Processing Use the 602 to create a unique sound for each of your on air personalities Give each announcer his or her own program number then create and store their sound If you have a way to send MIDI information to the 602 under control of a clock then the announce mic processing can change at shift change time Connect the 602 as an insert device after your console s mic preamp 8 2 Voice over Processing Create and store each of your favorite voice talent s settings in the 602 The next time that you work with them your starting point is a button press away 8 3 Foley Processing Use the 602 as in insert device in the console s signal path You can also use it to process field tapes during transfer to a workstation or other storage system 8 4 Digital Mastering Since the 602 can operate in digital in digital out mode use it when making production masters of digital material You can add compression make level changes EQ changes etc If you re not particularly enamored with the ADC in your digital recorder you can use the 602 as a converter if your digital recorder has digital S PDIF or AES EBU inputs 8 5 Musical Applications The particular combination of processors in the 602 make it ideal as an instrument processor especially for electronic ke
127. et Rev 1 1 11 15 94 1 1 connected tell you what the knobs do and send you on your way For MIDI information go directly to Appendix C which describes some of the things that you can do with the 602 using MIDI 1 1 Manual Sections This manual contains the following sections Chapter 1 Introduction introduces the 602 and this manual Describes important safety information Chapter 2 Basics lets you know what the 602 does and how it does it and some basic usage information Chapter 3 Technical Tutorial a basic and not so basic discussion of signal levels input and output impedances and connection polarity Chapter 4 Front Panel Overview gives a brief look at the controls and switches located on the front panel of the 602 Chapter 5 Rear Panel Overview gives a brief look at the rear panel of the 602 Chapter 6 Fast First Time Setup is a section written especially for people who just can t wait to get their hands on the knobs Chapter 7 Using the 602 describes the use of the 602 in detail Chapter 8 Applications describes some of the myriad uses for the 602 Chapter 9 Troubleshooting tells what to do if the 602 doesn t work Chapter 10 Limited Warranty describes the 602 s warranty Chapter 11 Repair Information tells how to get your 602 repaired Chapter 12 Specifications lists the technical specifications of the 602 s performance Appendix A Appendix A describes how to use the realtime MIDI features built into
128. eters Offset Description Range Reference dec hex dec 65 41 Control Chain Turnover 0 127 See Frequency Table Frequency See also offset O 66 42 Log Averaging Filter Tc See Tc Table 67 43 Sidechain Lookahead 0 0us See Sidechain 127 2 6 ms 48 kHz Lookahead Table linear scale Table 22 Output Offset _ Description Range Reference hex dec 68 44 Output Attenuation See Output Level Table 69 45 Output Pan 0 127 See Default Pan Table 0 ch 1 max 64 center 127 ch 2 max 70 46 Output Gain Rate of Change See Tc Table Rev 1 1 11 15 94 Table 23 Realtime MIDI Block 1 Offset Description Range Reference dec hex dec 71 47 Control Type 0 None Off parameter edit 1 Control Change 2 Aftertouch 3 Pitch bend msb 7 bit 4 Delay section modulation oscillator 1 5 Delay section modulation oscillator 2 6 Log signal level dynamics section 7 NR center freq 8 Instantaneous gain reduction value compressor 9 Instantaneous gain reduction value expander 48 3 Byte MIDI Message Second Parameter 0 127 49 Control Offset dec 0 127 64 0 0 128 127 127 74 4A Control Scaling 0 127 64 No Effect 0 4 127 4 Realtime Scaling Table NX 4B Parameter to Modify Offset dec Edit Buffer Offset Addre
129. evel analog signals converts them to 18 bit digital 44 1 kHz or 48 kHz sample rates performs 24 bit digital signal processing and sends them on their way via the digital and analog outputs The 602 uses two Motorola DSP 56001 digital signal processors DSP for an overall processing rate of 40 million instructions per second 40 MIPS The 602 has inputs and outputs accommodating all common analog and digital formats The following table lists all of the inputs and outputs Input Mode Output Mode Line x2 A Line x2 A AES EBU ID AES EBU D S PDIF ID S PDIF The stereo line inputs may be used in various combinations The input and output modes are separate you can use almost any combination of the analog and digital outputs simultaneously for example the AES EBU and S PDIF digital outputs cannot be used simultaneously While the 602 works great for voice singing or monologue dialogue enhancement its powerful digital engine works wonders on any signal Processing includes fully parametric EQ shelving EQ notch filtering dynamic noise filtering de essing delay first reflection stereo synthesis gating expansion compression and AGC automatic gain control Get the picture One aspect of many digital processors is the difficulty of use The 602 was designed to be easy to use yet powerful There are no menus to scroll through Each parameter is visible via the front panel push
130. ey MRC Displays Data to Set Enter 74 Slider 5 determines the parameter affected by the event selected by slider 1 MPLY 75 page SOURCE DEST OUT 76 SOURCE slr5 77 DEST SYSEX 78 OUT 1 79 DEFINE SYSEX BYTES 2 3 80 FO 81 00 82 00 83 page DEFINE SYSEX BYTES 4 7 84 5E 85 01 86 00 87 1C 88 page DEFINE SYSEX BYTES 8 9 89 AF 90 BYTE 91 page LABEL FOR slr2 92 PARM 93 page SOURCE DEST OUT 94 store 1 Use machine 15 setup 9 to control the delay 95 mach MACH 15 16 GMIDI Setup 1 96 enter TYPE 2NDP OFFS MPLY 97 setup GMIDI SETUP 9 1 DYNAM 1 98 enter TYPE 2NDP OFFS MPLY 99 edit SOURCE DEST OUT Set slider 1 to control delay 1 100 SOURCE slrl 101 DEST SYSEX 102 OUT 1 103 page DEFINE SYSEX BYTES 2 3 104 FO 105 00 106 00 107 page DEFINE SYSEX B 4 BYTES 4 7 use sliders 1 4 to set label to M P L Y setup slider 5 use button 1 to set source to slr5 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 79 send slider setting use sliders 1 4 to set label to PA R save your work use slider 1 to set mach to 15 use button 1 to set source to slrl use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 Rev 1 1 11 15 94 108 109 110 111 1
131. fer size worth 11 number of bytes special case 7F middle 7 bits transmit all 12 number bottom 7 bits 13 lt gt F7 Rev 1 1 11 15 94 Table 5 Data Response RESPONSE FROM 602 Type Requests Long Block of parameters by Offset dec Data ordering Edit Buffer 128 User Programs 128 ROM Programs Machine Setup Off Value Range hex dec 0 lt gt FO 1 mfrID 0 2 mfrID 1 gt 3 mfrID 2 gt 5E 4 lt device type gt 0 5 lt unit channel gt O 7E 16 command ID 7 lt Offset to start of data top 2 bits 8 Offset middle 7 bits 9 Offset bottom 7 bits 10 number of bytes top 2 bits 11 number middle 7 bits 12 number bottom 7 bits 13 first parameter O 7F 13 b lt gt F7 ytes C 5 Table 6 Program Setup Data Write Table 8 Identify Response DATA TO 602 RESPONSE FROM 602 Type Purpose Purpose Long Write a block of parameters by 602 response to Identify Request address 13h Data ordering Off Value Range hex Edit Buffer 128 User Programs 128 ROM dec Programs Machine Se
132. fferent digital input output formats AES EBU and S PDIF The AES EBU format uses XLR connectors is balanced and operates at 110 ohms line impedance The S PDIF Sony Phillips Digital InterFace format uses RCA connectors is unbalanced and operates at 75 ohms line impedance The digital input and output is transformer coupled for freedom from ground loops 3 7 MIDI I O Considerations The 602 has two MIDI connections MIDI in and MIDI out There is no MIDI thru connection Both connectors follow the standard defined by the IMA International MIDI Association The MIDI out jack echoes the MIDI input minus any sysex messages aimed at the 602 that owns the MIDI connectors All other messages are echoed There is a small throughput delay that may become significant if many 602s are wired in cascade series Rev 1 1 11 15 94 3 5 3 6 Notes Rev 1 1 11 15 94 4 Front Panel Overview DS O rom 00 cames RATION THRESWaB 655666566 6 2288 4 1 User Interface Summary The user interface of the 602 has been designed to be powerful yet intuitive Most switches have only one function and where a switch has several functions the display prompts for the parameter in question There are no hierarchical menus The parameter adjustment wheel Wheel modifies the parameter or function selected via the front panel switches The Wheel is sensitive to
133. fication stated in the AES EBU standard is quite stringent In applications requiring simultaneous digital sources like a digital mixer or digital multitrack recorder the sample rate clocks for every source should be phase locked synchronized to a common source This is described in section 7 3 5 2 The 602 has internal 44 1 kHz and 48 kHz clocks which should be adequate for most applications 7 4 4 External Sample Rate Clock The 602 can be synchronized to an external clock signal via the AES EBU or S PDIF digital inputs From the front panel select CLCE in the global block Apply the external clock signal to either of the digital inputs Avoid paralleling more than 2 inputs as loading of the clock source becomes a problem The sample rate clock should be a dedicated AES EBU signal source 7 4 5 Input Output Clock Summary The following table tabulates the various input output clock possibilities See also section 4 8 1 Input Input Digital Sample Clock Clock Notes Source LED IN SYNC Rate Ref Setting Setting Glows LED Analog Line off 44 1 kHz internal CLCI or 48 kHz Digital Digital steady either external CL 1 2 3 5 IN SYNC Em e Notes correspond to Notes column in previous table Connect the digital clock source to the digital input connector The source connected to the digital input supplies the sample rate clock The DIGITAL IN SYNC LED flashes if there is
134. flange On the 602 you create flanging by choosing a very short DELAY time 0 5 to 2 milliseconds set the MIX at 5096 set the modulation RATE at 1 set the modulation DEPTH at 100 You should hear a hollowness the jet plane sound that changes with time Increasing the amount of FEEDBACK makes the effect more pronounced Changing the FEEDBACK polarity phase shifts the comb frequencies Increasing the delay time Rate of change rt by holding down the DELAY button until the display reads rt smooths out the transitions and 7 14 Rev 1 1 11 15 94 makes the changes smoother You should definitely try making the two delay times slightly different Be sure to experiment with the different modulation sources 7 5 11 3 Chorus effects Chorusing is a variation on flanging The effect gives the impression of multiple sources On the 602 start with the DELAY time at about 10 milliseconds MIX at 5096 FEEDBACK around 80 modulation RATE around 20 modulation DEPTH at 100 and the delay time rate of change at its minimum setting Listen in stereo Experiments should include varying the delay time s altering the rate of change altering the wet dry mix and the modulation parameters Rev 1 1 11 15 94 7 15 Notes Rev 1 1 11 15 94 8 Applications Here are a few applications that the 602 lends itself to Do you have an unusual application for the 602 Send it to us and we ll consider sending you a can of slug chowder or some chocolate covered
135. foundation of all musical structure It was Leopold Stowkowski who said If had a thousand bass viols I could use them all This is not as extreme as it may sound A bass viol even though it is reinforced by its sounding board generally plays single notes and possesses little dynamic range In a large orchestra as many as 1 The majority of the material in section 2 4 is taken from Equalizing for Spectral Character Langevin Corporation 1966 Catalog 2 2 Rev 1 1 11 15 94 eight bass viols may be used A total of 1000 bass viols in this case would only give an additional 21 dB of level which is not an inordinate amount given a glance at Mr Fletcher s equal loudness curves Pay attention to this range because the overall musical balance of your program can be controlled by equalizing or attenuating the 100 Hz range 2 4 3 Telephone Quality The ear is reasonably sensitive the midrange frequencies and sound restricted to this range has a telephone like quality which is generally why telephone quality frequency response covers the 300 3 kHz range If you make the th octave 500 1024 Hz louder with respect to the other octaves the subjective result is a horn like quality If you emphasize the 7th octave 1000 2000 Hz the effect is one of tinniness The fundamental tones in most music lie equally above and below middle C 261 Hz from 128 to 512 Hz As most instruments are rich in the first overtones the majority of soun
136. g Input 1 gt Software gt Output 1 Software Software Poca ae Generated Generated Software SEES Noise Parametric Dynamics Generated Circuit Reduction Filter Processor Delay Circ uitry Circ uitry Circ uitry Circ uitry Input 2 B 7 3 3 Dynamics Block 7 4 Q The compressor and expander operate simultaneously The gain reduction value is determined by the processor having the greatest gain reduction output Q The auto release circuitry operates when the AGC Leveler is engaged Q The signal path delay compensates for the computational time needed to compute the gain reduction amount For extremely short attack times you may need to increase this parameter to allow the compressor to anticipate the input signal Q Both channels always receive the same gain reduction signal and the larger of the two input signals at any given instant becomes the source for any gain reduction computations The MIDI edit buffer parameter numbers are shown in parenthesis Rev 1 1 11 15 94
137. g MW Helping hand A hint to make your life a bit easier Bomb visual way of saying Caution 1 3 Other Safety Information Power Source This product is intended to operate from a power source that does not apply more than 250V rms between the power supply conductors or between either power supply conductor and ground A protective ground connection by way of the grounding conductor in the power cord is essential for safe operation Grounding The chassis of this product is grounded through the grounding conductor of the power cord To avoid electric shock plug the power cord into a properly wired receptacle before making any connections to the product A protective ground connection by way of the grounding conductor in the power cord is essential for safe operation Danger from Loss of If the protective ground connection is lost all accessible Ground conductive parts including knobs and controls that may appear to be insulated can render an electric shock Proper Power Cord Use only the power cord and connector specified for the product and your operating locale Use only a cord that is in good condition Proper Fuse The fuse is mounted internally and is not considered user serviceable The fuseholder accepts American sized fuses 1 4 in dia or European sized fuses b mm dia For 117V ac operation the correct value is 1 2A 250V ac fast blowing Bussman type AGC For 230V ac operation the correct value i
138. gain control the 602 computes the correct amount of makeup gain based on the ratio setting Whenever the AGC is currently being edited the output headroom LED display changes to read gain reduction The makeup gain may be set manually by pressing the RATIO switch until the display reads gAln Set the makeup gain using the Wheel ATTACK Switch Modifies the attack time of the AGC Leveler peak duration required to respond to a peak RELEASE Switch Modifies the release time constant of the AGC Remember that during no signal periods the AGC causes the release time to be infinite gain reduction release only occurs when there is no valid signal RATIO Switch Adjusts the compression ratio of the AGC Leveler between 1 1 out and 4 1 The compression ratio is the ratio of dB input change to dB output change The compressor s makup gain may be set manually by pressing the RATIO switch until the display reads gAln Set the makeup gain using the Wheel THRESHOLD Switch Sets the auto release threshold edit buffer offset 61 This is the level that a valid input signal must exceed to cause the AGC to readjust its gain to the new input signal To adjust the actual compressor threshold the compressor controlled by the auto release software use the realtime editor SEt procedure as described in Section 7 3 12 and edit parameter 56 dec Rev 1 1 11 15 94 4 8 ev 1 1 11 15 9 4 5 6 Dynamics Section Control Summary The following
139. gram number has been changed with the Wheel the LOAD switch LED and the preset number shown in the display flashes indicating that a new program is available for loading If the program number has not been changed but you still want to load the original program over the current edit buffer i e start over holding down the LOAD button forces a load operation In either case the display shows DONE when complete Rev 1 1 11 15 94 LEAVE EDIT Switch The LEAVE EDIT switch terminates any editing operation without disturbing or destroying the contents of the edit buffer You use this switch to return to the top most control mode program number shows in display 4 10 Setting Scenarios The following scenarios may help clarify setting up the 602 for various analog and or digital input signals Situation 1 Internal ADC internal sample clock DACs fed from DSP O Under globals click to InP Set left hand input LEDs to Ch1 Stereo and or Ch2 O Click globals again to or CLCI or CLCE Set to CLCI for internal master sample clock O Click globals again to 44 1 or 48 0 Select sample rate Situation 2 Internal ADC external sample clock DACs fed from DSP O Under globals click to InP Set left hand input LEDs to Ch1 Stereo and or Ch2 O Click globals again to or CLCE Set to CLCE for master sample clock from AES EBU reference O Connect external sample clock source to AES EBU input Situation 3 Input from
140. h flag head screws from the top cover Remove four 6 32 x 1 2 button head screws you ll need a 5 64 inch allen wrench Remove one 6 32 x 1 2 inch button head screw from the top middle of the front panel Lift the top cover free of the chassis mou s ec F 2 Circuit Board Removal There are five circuit boards inside the 602 Caution The circuitry within the 602 is static sensitive Use appropriate technigues to eliminate static electricity from your body and from the surrounding area If these techniques are not familiar to you you should refer servicing of your 602 to the factory 1 Ensure that the 602 is disconnected from the AC power source 2 Remove the top cover using the procedure described previously F 2 1 Analog Board Removal 1 Rotate the two gain controls until you can see the two setscrews on the shaft coupler near the circuit board mounted potentiometer Loosen the screw located towards the front of the unit and slide the knob and shaft out of the chassis Repeat for the remaining gain control 2 Remove the two 6 32 x 1 4 screws from the front edge of the circuit board 3 Disconnect the ribbon wire jumper between the analog and digital boards It is sufficient to remove disconnect only one end 4 Remove four 6 32 x 1 4 screws from the shield surrounding the analog board Remove the shield 5 Unlock the inserts within the four XLR connectors see procedure elsewhere in this section 6 Slide the analog board to
141. hanging loosely in a strong magnetic field The ribbon is moved by the action of air molecules which are set in motion by the sound wave the resonant frequency of the ribbon is very low below the audible range so the motion of the ribbon is mass controlled or is proportional to the velocity of the air particles For this reason it is called a velocity microphone TAD The motion of the ribbon within the magnetic field generates electricity which is the microphone s output signal S PDIF An acronym standing for Sony Phillips Digital Interface Format This term describes an interconnection standard method commonly used for consumer grade digital audio devices The reason Sony and Phillips are jointly named is because they are the two companies that developed the Compact Disc S PDIF signals carry two audio channels as well as status information The signal is unbalanced and RCA connectors are typically used for interconnection between devices The format of S PDIF signals is somewhat similar to the AES EBU format Sampling Rate In a digital audio system the audio signal must be fed into an analog to digital converter ADC to be changed into a series of numbers for further processing by the system The first step in this is sampling where the instantaneous signal amplitude is determined at very short intervals of time Sampling must be done very accurately to avoid adding distortion to the digitized signal The sampling rate which is the number
142. he Global switch again The display reads InP Rotating the Wheel selects the input source and routing as indicated by the DIGITAL CH1 STEREO CH2 LEDs and the display The input possibilities are digital or analog line input Rotate the Wheel to select the digital or analog source and your desired routing For analog stereo operation rotate the Wheel until the CH1 STEREO and CH2 LEDs illuminate and the display indicates 7 5 01027 Depress the Global switch again The display reads 1 5 Depress the Global switch again The display reads CL The clock sample rate reference is the external digital signal Refer to Chapter 4 for additional information 6 Depress the Global switch again The display reads nP not protected or Prt protected Select memory protection as required 7 Depress the level switch The display reads L 0 0 If not rotate the Wheel until the display reads L 0 0 This sets the output gain to O dB Set the input level by accessing the global gAln setting from the GLOBAL switch This parameter is the digital input gain Increase the gAln setting as required so that the 2 dB LED in the HEADROOM display illuminates Since the red CLIPPING LED is driven from the analog inputs it should never illuminate The 602 should now pass signal 6 4 Rev 1 1 11 15 94 7 Using the 602 This chapter is intended for more advanced users If you are a first time user we recommend that you start out by using the procedure foun
143. he front panel works To activate or deactivate any of the security features turn the 602 off press and hold one of the following buttons and turn the 602 on FILTER Enables no security Everything is accessible RATE Enables partial security DEPTH Enables maximum security With partial or maximum security enabled attempting to access any secured function results in a LoC indication on the display You can also defeat any security feature in effect by reinitializing the 602 Reinitializing erases ALL user programs presets 1 128 The security features can be activated via MIDI All 602 functions are always MIDI accessible regardless of the security level Rev 1 1 11 15 94 4 19 Notes Rev 1 1 11 15 94 4 20 5 Rear Panel Overview 602 STEREO DIGITAL PROCESSOR Symet IN THE d AES EBU BALANCED OUT R AG IN SAG IN MIDI MIDI DIGITAL DIGITAL PRESS PRESS PRESS OUTPUT INPUT OUTPUT weur Sym UNIT retri x e USER SERVICEABLE PARTS S PDIF EI RA EN a QO 125W MAXIMUM V s POWER Serial Number Do yourself a favor and write this number down somewhere safe and while you re at it please send us the completed warranty card AC Power Input IEC power connector Connect only to appropriate AC power source Refer to actual rear panel marking for correct AC source value POWER switch Push push switch turns the 602 on and off MIDI connectors 5 pin DIN connectors
144. he presence of uninsulated presence of important operating and dangerous voltage within the product s maintenance servicing instructions in the enclosure that may be of sufficient literature accompanying the appliance i e magnitude to constitute a risk of electric this manual shock to persons Caution To prevent electric shock do not use the polarized plug supplied with this appliance with any extension cord receptacle or other outlet unless the blades can be fully inserted to prevent blade exposure 1 2 2 Terms Several notational conventions are used in this manual Some paragraphs may use Note Caution or Warning as a heading These headings have the following meaning Convention Description Caution Identifies information that if not heeded may cause damage to the 602 or other eguipment in your system Note Identifies information that needs extra emphasis A Note generally supplies extra information to help you use the 602 better Warning Identifies information that if ignored may be hazardous to your health or that of others In addition certain typefaces and capitalization are used to identify certain words These situations are Convention Meaning CAPITALS Controls switches or other markings on the chassis Boldface Strong emphasis Helvetica Narrow Information appearing on the LED display Rev 1 1 11 15 94 1 3 Finally two symbols are used as visual hints They are Symbol Meanin
145. help remove microphone leakage from a signal Start by setting the expansion RATIO to 1 2 This means that the output falls 2 dB for every 1 dB of below threshold change in the input signal Next set the threshold so that the expander causes gain reduction right LED meter as the signal falls in level Higher expansion ratios will make the effect more obvious The ATTACK parameter determines the expander s response to a signal s duration shorter attack times allow the expander to respond to short duration sounds like clicking your tongue If the attack time is long enough the expander will ignore short duration sounds The RELEASE time parameter determines the length of time needed for the gain to drop once the input signal abruptly falls below threshold The RELEASE time and the expansion RATIO appear to interact somewhat This is not the case A 1 8 expansion ratio means that the output level Rev 1 1 11 15 94 7 13 will fall by 8 dB for every 1 dB of input change below threshold A release time of 1000 milliseconds says that it will take 1000 milliseconds for the output signal to decay from its initial value The RATIO parameter deals with the slope of the input vs output gain relationship independent of time and the RELEASE parameter deals with the rate of change in time units of the output signal when it transits the two points initial attenuation and ultimate attenuation as determined by the ratio setting 7 5 11 Delay The delay
146. hose parameters that are not directly accessible from the front panel Use the programs to start your own Voice Speech and Song RAM OI ele O1 S Oo 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 G 4 ROM 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 Name TLM 170 Male Maximum Intelligibility Talk Show Announcer TLM 170 Speech Delay TLM 170 Speech Long Delay Speech Chorus Clear Vox Metal Closet Speech Leveler AGC Leveler TLM 170 Female Vocal Flange Female Supremes Speech Leveling Female Aggressive bright narration Mellow narration FM Disc Jockey 1 FM Disc Jockey 2 FM Disc Jockey 3 Wide stereo image voice Vocal Handheld Mic Vocal Handheld Mic with Echo Vocal Flange Elvis Pop Vocal TV Commercial Announcer Stereo TV Commercial Announcer 50 s Country Vocal Rock blues Vocals Handheld Vox Female Handheld Vox Female Delay Background Vocals Thin Airy Vocals High definition Vocals Rock Echo Vocals Doubled Rock Vocals Tripled Rock Vocals Voice Thickener Warmer 1 Voice Thickener Warmer 2 Description Neumann TLM 170 optimized for male voice EQ set for voice range boost Small room ambience Ne
147. ich use the lower voltages have a regulator circuit to reduce the higher voltages so no harm is done when they are plugged into a 48 volt phantom power supply There is a DIN standard no 45 596 which specifies in detail the requirements for phantom power TAD Phantom powering is a compatible system suitably wired low impedance microphones may be plugged directly into a phantom powered input without regard to the presence or absence of phantom power The technique gets its name from the old telephone term Phantom Circuit which was a method for creating a second circuit on an existing pair of telephone wires Phasing phaser A phaser or phase shifter is a device which gives an effect similar to flanging but with less depth It works by shifting the phase of the signal and adding it back to the signal This causes partial cancellation at frequencies where the phase shift approaches 180 degrees Phasing is sometimes called skying in Britain TAD See also flanging P Pop p pop is the burst of air caused by uttering the letter P Spoken directly into a single d microphone this blast of air usually causes a loud audible popping sound A windscreen of some sort is a good cure for p popping as is an omnidirectional microphone A skilled announcer will soften the initial attack of the letter or turn their head slightly so as to avoid the microphone Proximity Effect Proximity effect is the increase in low frequency sens
148. in the parameter group modify the settings of the selected flashing parametric EQ section Each switch has the following action FREQ KHZ The Wheel modifies the center frequency of the selected equalizer from 31 Hz to 21 11 kHz Holding down the FREQ switch allows setting the time constant for the rate of change of the center frequency Shares time constant rt with WIDTH switch The filter frequencies are 1 10th octave ISO standard frequencies except for some special power line harmonic frequencies at the low end LEVEL DB The Wheel modifies the amount of boost or cut from 18 dB to 50 dB The display indicates the filter s boost cut setting in dB The display reads out when no filter contribution has been set same as LEVEL at 0 out Holding down the LEVEL switch allows setting the time constant for the rate of change the display reads rt This affects how quickly the LEVEL setting changes either due to MIDI command program change or rotation of the Wheel DYNAMICS PROCESSING KT 05 COMPRESS ATTACK mS RELEASE mS OK FREQ kHz 1 AGC RATIO X 1 THRESH dB F WIDTH OCT The Wheel modifies the bandwidth from 0 05 octaves very sharp or narrow to 3 octaves quite broad The rate of change value is shared with and accessed with the FREQ switch since both are frequency related parameters 4 5 Dynamics Processing Block The dynamics processing block encompasses the de esser dynami
149. information Select CLCI 6 Depress the Global switch again The display reads nP not protected or Prt protected Select memory protection as required 7 Depress the Global switch again The display reads 44 1 or 48 0 This represents the two sample rates only if you haven t selected CLCE in step 5 Rotate the Wheel to select the sample rate appropriate to your application 8 Depress the level switch The display reads L 0 0 If not rotate the Wheel until the display reads L 0 0 This sets the output gain to O dB 9 Set the input level by increasing the setting of the selected input level control until the green LEDs in the HEADROOM display illuminate Ideally the highest signal level should illuminate the 2 dB LED and the CLIPPING LED should never illuminate the CLIPPING LED operates at 1 dB below clipping 6 2 Rev 1 1 11 15 94 10 It is possible for the Clipping LED to illuminate even though the green LEDs in the Headroom display are not completely illuminated If this occurs decrease the setting of the appropriate gain trim control sufficiently to keep the Clipping LED from illuminating then access the global digital gAln setting from the GLOBAL switch Increase the digital gAln setting as required 11 The 602 should now pass signal AES EBU 602 STEREO DIGITAL PROCESSOR MANUFACTURED IN THE USA x DIGITAL DIGITAL OUTPUT INPUT OUTPUT 9 INPUT S NO S PDIF pad E O 125W MAXIMU
150. ings given as starting points for developing your own settings More general discussions of these topics may also be found in Chapter 2 of this manual 7 5 1 Recalling and Storing Settings Recall any program by pressing the LEAVE EDIT button then using the Wheel to select the new program then pressing the LOAD button The new program has loaded when the display reads donE The 602 always loads a copy of the program into the edit buffer regardless of whether you want to edit the program or not The program in the edit buffer is also the program that the 602s processor executes unless the COMPARE button has been pressed in which case the 602 executes the program out of the program RAM ROM When you modify the edit buffer by changing any parameter the SAVE switch flashes unless the program memory has been write protected either by location or via the global parameter block If the SAVE switch is flashing the edit buffer is dirty that is its contents have changed The modified program can be stored in program numbers 1 through 128 If you try to save a program to presets 129 through 256 which are always protected the display reads Prt indicating that the selected program number is protected read only Chose a program number between 1 through 128 for your program Store a modified program by pressing the LEAVE EDIT button then using the Wheel to select a program number for the modified program remember between 1 and 128 then pressing
151. ip Indicator Output Headroom Display Midi Implementation Access MIDI channel range Accessible parameters Connectors Data dump Manufacturer ID Physical Size hwd in amp cm Weight Ibs amp kg Electrical Power requirements 12 Hz 20 kHz 1 5 dB lt 01 1 kHz 1V RMS 104 dB This represents the difference between the largest and smallest signals that will pass through the 602 Measured using 8192 point FFT with Blackman Harris windowing function 44 1 kHz 48kHz Delta Sigma 18 bit linear 64X oversampling RAM with battery backup 1 4 3 98 ms 48 kHz 1 51 4 11 ms 44 1 kHz 9 LED bargraph Red LED indicates clipping at analog inputs 8 LED bargraph MIDI program change sysex aftertouch pitch bend bank select 1 128 omni mode Most front panel parameters plus internal constants midi in midi out current program or entire memory 00 00 5e 1 75 x19x7 in 4 44 x 48 26 x 17 78 cm 7 6 Ibs 3 5kg net 10 Ibs 4 6kg shipping 117V AC nominal 105 125V ac 50 60 Hz 20 watts 230V AC nominal 205 253V ac 50 Hz 20 watts In the interest of continuous product improvement Symetrix Inc reserves the right to alter change or modify these specifications without prior notice 12 2 Rev 1 1 11 15 94 abandon edit 4 17 AGC controls 4 8 parameters Table 19 C 10 settings 7 13 tutorial 2 8 applications 8 1 a d converter 7 7 broadcast 8 1 channel insert 7 6
152. istortion Frequency modulation distortion Examples of this are flutter and wow and doppler distortion caused by the motion of loudspeaker cones TAD Downward Expander See Expander Dynamic Filter A dynamic filter is a type of single pass noise reduction system that uses one or two filters whose cutoff frequencies are controlled by the level of the signal As the signal level falls during soft passages the high frequency response is reduced like turning down the treble tone control and when the signal level is high the full bandwidth is restored The effective operation of such a system depends on the fact that the noise will be masked by the signal during loud passages and this is true in many but by no means all cases A key element in the design of dynamic filters is the choice of time constants during the time that the bandwidth is changing If they are too fast distortion results and if too slow the noise will be heard to swish in and out as the signal level changes TAD D 4 Rev 1 1 11 15 94 Dynamic Microphone dynamic microphone consists of a diaphragm with a coil of wire attached to it such that sound pressure moving the diaphragm causes the coil to move in a magnetic field supplied by a permanent magnet Motion of the coil causes an electric current to be induced in it and this is the signal output of the microphone It is similar to a dynamic loudspeaker operating in reverse TAD Dynamic Range The dynami
153. it buffer 39 kills wet signal use sliders 1 4 to set label to KI L L setup switch 2 use button 1 to set source to swt2 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 218 5E 219 220 221 222 223 1C 224 page DEFINE SYSEX BYTES 8 9 225 2A 226 227 229 page LABEL FOR swt2 230 SINE 231 page Set switch 3 to set triangle modulation 232 page SOURCE DEST OUT 233 SOURCE slr5 234 DEST SYSEX 235 OUT 236 237 page DEFINE SYSEX BYTES 2 3 238 FO 239 240 241 242 243 page DEFINE SYSEX BYTES 4 7 244 5E 245 246 247 248 249 1C 250 page DEFINE SYSEX BYTES 8 9 251 2A 252 253 page LABEL FOR swt3 254 TRI 255 page Set switch 4 to turn on random modulation 256 page SOURCE DEST OUT 257 SOURCE swt4 258 DEST SYSEX 259 OUT 1 260 page DEFINE SYSEX BYTES 2 3 261 FO 262 00 263 00 8 Comments mfrID2 device type unit channel edit buffer data set edit buffer 42 228 sine use sliders 1 4 to set label to S I N setup slider 5 use button 1 to set source to slr5 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 42 7F triangle use sliders 1 4 to set label to T R I setup switch 4 use button 1 to set source to s
154. itivity of a microphone when the sound source is close to the microphone It is a characteristic of directional microphones and some are much worse than others Proximity effect is a shortcoming but sometimes it can be used to advantage If a directional microphone is placed close to a bass instrument the low tones will be enhanced which could be advantageous for some music A singer placed close to a directional microphone will sound much bassier and improvement in some voices I suppose Some of the early radio crooners and radio announcers used proximity effect to deepen and enrich their voices and many frequently still do TAD Q In reference to a resonant mechanical or electrical circuit or a capacitor Q stands for quality factor In the case of a resonant system Q is a measure of the sharpness of the resonant peak in the frequency response of the system and is inversely proportional to the damping in the system Equalizers that contain resonant circuits are rated by their Q value the higher the Q the higher and more well defined the peak in the response TAD Quantization representation of a continuous voltage span by a number of discrete values Quantization is inherent in any digital audio system and it adds quantization error noise and distortion to the signal The signal after quantization has a staircase shape rather than a continuous curve and the difference between this and the original signal is quantizati
155. kahead delay time 4 5 7 11 De ess absolute threshold The relative de esser threshold can be accessed via the front panel This threshold setting must be relative in order that it not be sensitive to the overall signal level at any given instant in time The de esser determines the relative amount of sibilant energy in the input signal which is then compared to the low frequency content of the input signal The high frequency content must exceed the absolute threshold level Once the signal exceeds the absolute threshold the sibilant energy content must then exceed the relative threshold setting Only then will de essing occur Edit buffer 22 controls the absolute threshold Edit buffer 23 controls the relative threshold and may be adjusted via the front panel the realtime editor or via MIDI 4 6 Delay Group The delay group encompasses a dual delay line with cross coupled feedback and delay time modulation The delay time of each delay can be set independently and the duAL mode changes the delay times of both delays simultaneously while maintaining any difference in the delay times You can also add modulation to the delay time s The modulation signal may be random a sine wave or a triangle wave The modulation signal depth amplitude and rate frequency are onim adjustable The delay group creates effects ranging from simple slapback through small room simulation chorusing and flanging DELAY MIX DELAY mS FEEDBACK
156. l audio components the sampling frequency 44 1 kHz for compact discs is raised to a multiple of that frequency For example if the sampling frequency were raised by a factor of 4 three artificial samples must be created in between each pair of original samples These samples are zero in level and they do not change the information content of the original samples Digital filtering is then used for interpolation of the zero samples to values intermediate between the true sampled values But because the sampling rate is now so much higher a very gentle anti aliasing filter can be used rather than the brickwall filter usually needed resulting in much less phase distortion See also sampling rate TAD Pan panpot Short for panoramic potentiometer which is two connected volume controls with a common knob so wired that as one is turned up the other is turned down If the stereo channels are controlled by a panpot the apparent position of the sound will move from left to right as the control is turned The balance control on most stereo amplifiers is actually a simple panpot Panpots are used in recording to place the apparent position of a sound such as a soloist or other instrument anywhere between the two loudspeakers Its operation relies on the ability of our ears to localize a sound by level differences heard by our two ears For a panpot to work properly it must follow an accurately prescribed attenuation curve In many recordings sev
157. lating the delay time This causes the delay time to vary according to the frequency and amplitude of the modulating signal If you choose a short delay time typically around 1 ms a low modulation frequency and roughly a 50 50 mix of direct signal and delayed signal you ll get flanging Adding feedback accentuates the effect If you alter the mix to favor the delayed signal and raise the modulation frequency you ll get pitch bending vibrato or chorusing Rev 1 1 11 15 94 2 9 2 12 MIDI If you aren t aware of MIDI well Several years ago a number of musical synthesizer manufacturers somehow agreed on a serial data protocol to exchange control information between synthesizers They called the result MIDI Musical Instrument Digital Interface The success of this standard is phenomenal not that it is perfect and the ability to control something via MIDI has been applied to everything from synthesizers to signal processors to lighting systems Nearly every parameter of the 602 may be controlled or modified via MIDI The 602 s MIDI implementation is described in Appendix C What does this mean for the 602 At a very basic level it means that you could have several setups stored in the 602 and change between them remotely If you re a broadcaster you could have a MIDI controller output program change commands based on a clock which would change the settings of the on air 602 to the personalized settings for each of your on air person
158. le locations 25714 25835 is loaded from the default pan table when the 602 is first initialized The table consists of 61 pairs of left right attenuation values These values may be edited should your application require something different The new values survive power on off cycles Rev 1 1 11 15 94 C 7 C 1 6 Data Structure Per Program All programs use the following data structure Each program parameter has a specified offset within the Edit Buffer By reading or writing these parameters you can query or set particular program parameters By dumping the entire range 0 99d you can look at the status of the entire edit buffer By reloading the same range you can superimpose your own values onto the same parameters You can also modifiy any value individually via MIDI You ll need an external MIDI program editor to perform this task For clarity each function is presented in its own table however the offsets shown refer to a contiguous block of memory in the unit Table 10 Global Offset Description Range dec Reference 0 Dynamics sidechain filter 0 highpass shelving See also offset 65 mode 127 lowpass 1 De esser sidechain filter 0 default 5 kHz See Frequency Table frequency 1 127 2 Input Gain See Attn18 Table Table 11 Filter 1 Offset dec Description Range dec Reference 3 F1 Mode 64 shelving E 4 F1 Freq See Frequency Table 5 F1Q See BW
159. lider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 B 5 145 146 147 148 149 150 151 152 Set sl 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 Set sl 171 172 173 174 175 176 177 178 179 180 B 6 page page page page page page page page page page Displays DEFINE SYSEX BYTES 8 9 LABEL FOR slr3 ider 4 to control the wet dry mix SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slr4 ider 5 to control rate of change SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 Data to Set Enter 01 00 1 25 slr4 SYSEX FO 00 00 5E 01 00 1 27 15 SYSEX FO 00 00 Comments device type unit channel edit buffer data set edit buffer 37 send slider setting use sliders 1 4 to set label to F B setup slider 4 use button 1 to set source to slr4 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 39 send slider setting use sliders 1 4 to set label to xX setup slider 5 use button 1 to se
160. log sources and analog loads To operate the 602 from unbalanced sources run a 2 conductor shielded cable that s two conductors plus the shield from the source to the 602 At the source connect the low minus side to the shield these connect to the source s ground connect the high plus side to the source s signal connection At the 602 the high plus wire connects to pin 2 the low minus wire connects to pin 3 and the shield always connects to pin 1 This is the preferred method as it makes best use of the 602 s balanced input even though the source is unbalanced The other alternative shown in Figure 3 1 converts the 602 s balanced input into an unbalanced input at the input connector This works but is more susceptible to hum and buzz than the preferred method There is no level difference between either method You can drive unbalanced loads with the 602 s outputs by using the XLR connector with pin 3 left open In an emergency the show must go on you can ground pin 3 but if you have the choice leave it open If you must ground pin 3 it is must be grounded at the 602 rather than at the other end of the cable The price regardless of whether or not pin 3 is grounded is 6 dB less output level This can be easily made up via the output gain controls If your system is wired with pin 3 hot pin 2 must float if you are driving an unbalanced load Rev 1 1 11 15 94 3 3 FROM BALANCED OUT FEMALE XLR PIN 1 lt GROUND PIN 2 lt HIG
161. loor Clip Realtime Block 1 Ceiling Clip Makeup Gain Program name Modified flag dec 0 127 0 127 0 127 127 0 0 127 O auto makeup gain 1 0 dB makeup gain Range 127 24 dB makeup gain Reference Block 1 only Minimum normalized edit buffer value Block 1 only Maximum normalized edit buffer value Shared between AGC amp Compressor Attn24 Table ASCII program name Forced to if program modified from front panel C 1 7 MIDI Parameter Tables Many of the parameters used in the 602 are extracted from tables When controlling the 602 via MIDI all values sent to the 602 via its MIDI port must be mapped from their real world values into a table based on 128 steps The following tables list various system parameters and their conversion values 10 20 30 40 50 60 70 80 90 100 110 120 Table 26 2 3 18 0 17 0 13 0 12 0 8 0 8 0 5 0 5 0 2 5 2 5 0 5 0 0 2 0 3 0 5 0 5 0 7 5 7 5 9 5 10 0 12 0 12 5 14 5 14 5 17 0 17 0 Attn18 Table dB 4 5 6 17 0 16 0 16 0 12 0 11 0 11 0 7 0 7 0 6 5 4 5 4 5 4 0 2 5 2 0 2 0 0 0 0 5 0 5 3 0 3 0 3 5 5 5 5 5 6 0 8 0 8 0 8 0 10 0 10 5 10 5 12 5 13 0 13 0 15 0 15 0 15 5 17 5 17 5 18 0 1 0 4 0 6 5 8 5 11 0 13 5 16 0 14 0 9 0 6 0 3 5 1 0 1 5 4 0 6 5 9 0 11 5 13 5 16 0
162. ltime editor refer to Appendix If you modify any of these parameters and get lost you can return to some semblance of reality by either loading program 256 and starting over or by loading any other program Many of the preset programs modify these parameters also this means that you can t simply duplicate a program by re entering its front panel parameters Rev 1 1 11 15 94 4 9 These parameters are Expander knee control Compressor knee control AGC Threshold AGC knee control ARM integration time constant ARM Signal Noise threshold frequency Log converter time constant Lookahead delay time De ess absolute threshold Dynamics sidechain filter mode ARM peak release time constant Dynamics control chain turnover Purpose highpass shelving to lowpass Sets number of dB required to reach ultimate expansion ratio Sets number of dB required to reach ultimate ratio Absolute threshold for AGC Sets number of dB required to reach ultimate ratio Determines the recovery time of the auto release monitor system the auto release monitor system Shorter times require higher peak to average ratios to release the AGC hold Sets peak average ratio for signal noise decision subsonic response of control chain sidechain for the dynamics sidechain Provides thinking time for dynamics sidechain which can prevent overshoot action The relative threshold affects the degree of de ess
163. meter Rev 1 1 11 15 94 Model Symetrix 602 MIDI Implementation Chart Doc Rev 1 1 11 15 94 Device Type 02 Midi Manufacturer ID 00 00 5E Function Transmitted Recognized Remarks Basic Default 1 16 1 16 Memorized Channel Channel 1 16 1 16 Default X X Memorized Mode Messages X X OMNI ON OFF Altered X X Note X X Number True Voice X X Velocity ON X X Note OFF X X After Key s X Realtime MIDI blocks Touch _Ch s Pitch Bender X Realtime MIDI blocks Control X 07 Volume Any using realtime MIDI 10 Pan blocks Change 32 Bank Select Program X Programi 1 128 Change True 0 127 System Exclusive O O System Song Pos X X Song Sel Common Tune X x x System Clock X X Real Time Commands X X Aux ON OFF X X Messages Al Notes OFF X X Active Sense Reset X X Notes Mode 1 OMNI POLY Mode 2 OMNI ON MONO O Yes Mode 3 OMNI OFF POLY Mode 4 OMNI OFF MONO X No Rev 1 1 11 15 94 G 3 3 Presets and Building Blocks The following table lists every factory supplied program in the 602 Note that those programs listed in the RAM column may be modified or overwritten by another possibly totally different program and those programs listed in the ROM programs may be modified but not saved except in unprotected RAM memory Programs 100 through 128 are building block programs These programs give you quick setups for certain parameters that involve realtime MIDI or t
164. mmer s worksheet G G 2 G G 2 rate of change 4 2 rEAL 4 15 realtime MIDI block 1 Table 23 C 12 realtime MIDI block 2 Table 24 C 13 Realtime MIDI source abbrev A 1 reinitializing 4 19 rt 4 2 safety information 1 4 safety summary 1 2 saving programs 2 10 4 1 security 4 19 setting AGC 7 13 analog sources 6 2 compressor 7 13 de esser 7 12 delay 7 14 digital sources 6 4 downward expander 7 13 equalizer 7 11 gain 2 1 MIDI channel 4 15 MIDI unit number 4 15 noise reduction 7 12 scenarios 4 17 store and recall 7 10 setup data request 5 signal levels 3 2 specifications 12 1 Stand alone Operation 7 2 storing programs 4 16 switch AGC 4 8 attack 4 8 AGC release 4 8 bypass 4 14 compressor attack 4 7 compressor release 4 7 compressor threshold 4 7 de esser attack 4 6 de esser release 4 6 delay 4 12 delay group 4 12 depth 4 13 DS threshold 4 5 DS De ess 4 5 EQ freq 4 4 EQ level 4 4 EQ width 4 4 expander 4 6 expander attack 4 6 expander ratio 4 6 expander release 4 6 expander threshold 4 7 feedback 4 13 global 4 14 level 4 14 mix 4 12 NR 4 5 NR freq 4 5 NR threshold 4 4 4 5 4 7 pan 4 14 power 5 1 preset 4 16 rate 4 13 system 4 14 system interface 7 6 table AGC parameters Table 19 C 10 AGC ratio Table 34 C 16 ARM sense paramters Table 20 C 11 ARM threshold Table 35 C 17 attn C 18 attn100 Table 28 C 14
165. mount of instantaneous gain reduction controls the gain of the digitally controlled attenuators DCA An adjustable delay before the DCA allows controlling the amount of overshoot occurring within the compressor Rev 1 1 11 15 94 7 1 The Delay block uses two delay lines one per channel with their recirculation paths cross coupled The feedback signal may be polarity inverted and the delay time may be controlled by an internal modulation oscillator or the front panel Many of the 602 s parameters have a rate of change parameter associated with them This parameter determines how quickly the 602 responds to a step change in the value of the parameter This parameter ranges from 100 microseconds to 10 seconds A 10 second rate of change setting makes the 602 change from the old value to the new value over a period of 10 seconds 7 2 1 Stand alone Operation A vast majority of users use the 602 as a stand alone device Here the 602 replaces their usual complement of signal processing and either feeds their tape machine or workstation directly in essence becoming a one input one output console If you are using the digital outputs of the 602 be sure to read sections 7 3 4 and 7 3 5 If you are using the analog outputs ensure that they are Dk plugged into a line level input 4 dBu nominal level If r PD BALANCEDIN you have to plug the 602 into a microphone input 40 PIN EH FER 150 3 dBu nominal le
166. ngers and announcers use very close up microphones When the high frequency energy exceeds a preset threshold the compressor starts to operate to reduce the high frequency response Low level high frequency sounds are not reduced TAD The 602 uses a variation of this technique The essential difference is that the threshold setting is relative to the ratio of sibilant to non sibilant sounds The compressor operates across the entire audio band i e all signals are reduced in the presence of a sibilant sound that exceeds the preset threshold Digital The application of digital computer based technology to the recording and reproduction of music is somewhat loosely called digital audio TAD Digital delay A digital device which provides an adjustable time delay Time delays are used in artificial reverberation systems for special echo effects in music recording and to provide delayed sound to certain loudspeakers in some sound reinforcement systems Before the advent of digital delays the only the effect could be achieved was by tape echo or by placing a loudspeaker at one of a long tube and a microphone at the other This gives a delay of about 1 millisecond per foot of length and it becomes bulky when long delays are needed Digital Signal Processing DSP The manipulation and modification of signals in the digital domain after having undergone analog to digital conversion A great many electronic music instruments use DSP as do certain tes
167. nk about the particular combination of processors in the 602 is in terms of a modern mixing console Today most mixing consoles have microphone and line inputs some sort of equalization effects sends and returns and occasionally on board dynamics processing For a typical voice over session you would probably have a compressor limiter and a digital delay patched in as outboard processors The 602 provides each of these processors wrapped into one tidy one rack space package 2 2 Digital and Analog Differences A large difference between the 602 and a mixing console is that the processing functions in the 602 are implemented totally within the digital domain whereas those within the console are most likely implemented in the analog domain Outwardly there is no difference between an analog and a digital processor A digital parametric equalizer has the same controls that you re familiar with in the analog world Granted the way that you access these controls may be different but how much difference is there in seeing 9 dB on an LED display or in reading it off of a knob against a scale on the front panel 2 3 Gain Setting Wire is probably the only component of a sound system where we don t need to take signal levels into account usually Any other active component of a sound system that passes signal has a finite dynamic range This means that our old friends dynamic range headroom and noise floor are present and must be accounted for
168. no digital source or if the digital data is faulty Connect the digital I O connectors to another digital I O processor The DIGITAL IN SYNC LED column shows the state of the DIGITAL IN SYNC LED on the front panel Jte ae Se Rev 1 1 11 15 94 7 7 7 4 6 MIDI Programming The 602 is MIDI programmable At one level you can simply send MIDI program change messages to load pre stored programs yours or the factory presets At another level you can manipulate program parameters via MIDI and at yet another level you can modify program parameters in realtime during operation The 602 responds to the following MIDI messages Program Change Control Change Sysex Pitch Bend Aftertouch SUP P bores You can theoretically operate up to 127 602s on a single MIDI bus Depending upon how they are programmed you can access them individually or as a group 7 4 7 Accessing Parameters via MIDI front panel parameters may be accessed via MIDI In addition all secondary hidden parameters may be accessed via MIDI A list of all accessible parameters may be found in Appendix A These parameters may be altered via a MIDI sysex message or by using the procedure found in Section 7 4 10 7 4 8 Realtime MIDI Many of the 602 s MIDI controllable parameters lend themselves to realtime control using a MIDI continuous controller Some of these parameters are 1 Output level and pan 2 Filter frequency and level 3 Delay mix An example
169. nt MI equipment 10 dBu or 300 millivolts 3 3 Impedances The 602 is designed to interface into almost any recording studio or sound reinforcement application This includes O 600 ohm systems where input and output impedances are matched O Unbalanced semi professional equipment applications O Modern bridging systems where inputs bridge and outputs are low source impedances voltage transmission systems The 602 s input impedance is 12 5 balanced and 9 4 unbalanced The inputs may be driven from any source balanced or unbalanced capable of delivering at least 10 dBu into the aforementioned impedances The 602 s output impedance is 300 ohms balanced 150 ohms unbalanced The output line driver delivers 21 5 dBm into 600 ohm balanced loads or 15 5 dBm into 600 ohm unbalanced loads 3 2 Rev 1 1 11 15 94 3 4 Polarity Convention The 602 uses the international standard polarity convention of pin 2 hot Therefore If your system uses balanced inputs and XLR Tip Ring Signal outputs and uses the 602 this way then the Sleeve polarity convention is unimportant If your 1 Ground system is both balanced and unbalanced then 2 High you must pay attention to this especially when 3 Low going in and coming out through different connector types like input on an XLR output on a phone jack 3 5 Input and Output Connections Figure 3 1 illustrates how to connect the 602 to balanced and unbalanced ana
170. o work on Now reduce the level setting to something tasteful It s sometimes difficult to translate what you are hearing into the numbers that make equalizers happy Seeing the frequencies associated with a voice or instrument can be helpful in deciding where equalization may be needed The chart shown in Figure 2 1 shows the relationships of many different instruments and a piano keyboard along with the frequencies involved 2 4 Rev 1 1 11 15 94 WIND Rev 1 1 11 15 94 BASS TUI KETTLE o 2 ui z te INSTRUMENTS STRING HUMAN VOICE UA Figure 2 1 Relationships of Musical instruments Piano and actual frequencies B Ov vO8ST 00 080vT 00 rv ScT OCSLTIT 00 87S0T 6 6 00 2061 00 0704 0072729 0977846 00 7 24 0978697 00 98Ty OT TS6E 0070446 00 9 TE 08 6 Z 007792 05 67 007 lt 602 05 5 6 0009 0078951 0696 OS 8TET OLYLTI 0S 970T LL L86 007088 66 8Z 97 869 9 699 88 00077 007266 57 67 9 6c 99 6Z 9 T9c v6 9vc 00 0ZZ 00 96T T9 7 T 8 9vT T8 0 T 000 1 00 86 18 59 7 19 00 00 67 99 0217 14 9 OLE L8 0 04 2 0902 S 8I 56 9T
171. of Realtime MIDI may be found in Appendix B 7 4 9 Program Storage The 602 provides non volatile storage for 128 user programs Program numbers 1 through 128 Program numbers greater than 128 are factory presets and are always protected You can edit any of the factory presets and store it in one of the user program numbers You can dump the contents of the 602 s program memory to the MIDI OUT connector on the rear panel Conversely you can also load the 602 s program memory via MIDI 7 8 Rev 1 1 11 15 94 7 4 10 Editing Parameters not Accessible from the Front Panel The front panel realtime update editing function can also be used to set the edit buffer value for parameters that are normally inaccessible from the front panel These parameters are Processor Offset dec Parameter name Reference Dynamics Processor 0 Sidechain Filter Mode 0 Hipass Shelving 127 Lowpass De ess Processor 22 Absolute Threshold See Attn100 Table Expansion 49 Expander Knee See Knee Table Parameters Compression 55 Compressor Knee See Knee Table Parameters Parameters 56 Absolute Threshold See Attn100 Table 60 AGC Curve Knee See Knee Table ARM Sense 62 ARM Peak Release See Tc Table Parameters Tc 63 ARM Integration Tc See Tc Table 64 ARM Threshold See ARM Threshold im kli i s o de 2 12 52 e LOG Converter 65 C
172. of a single instrument to simulate a large group of the same instruments for example a vocal chorus or a string section The subjective effect of a real chorus is caused by the fact that the many sound sources being mixed together all have slightly different frequencies and also do not have precisely steady frequencies The mixture because extremely complex as the relative phases of the signals cause partial cancellation and reinforcement over a broad frequency spectrum The synthetic chorus effect was first attained by subjecting the input sound to a series of very short time delays and mixing the delayed sounds The delays were then randomly varied or modulated to increase the uncertainty of the combined pitch This could be called the time domain chorus synthesis and can be quite expensive if enough delay times are used to ensure a satisfactory result A new type of chorus device operates in the frequency domain and is somewhat simpler and at the same time more convincing The signal is split into many frequency bands by a series of bandpass filters and each band is randomly varied in phase and amplitude after which they are recombined TAD Clipping If a signal waveform is passed through an amplifier or other device which cannot accommodate its maximum voltage or current requirements the waveform is sometimes said to be clipped because it looks like it has had its peaks clipped by a pair of scissors A clipped waveform contains a great deal
173. of harmonic distortion and sounds very rough and harsh Clipping is what typically happens when an audio amplifier output is overloaded or its input overdriven The clipping point of an amplifier is defined as the maximum sine wave signal level which when viewed on an oscilloscope shows no signs of flat topping to the trained observer TAD Comb Filter A comb filter is a filter which has a series of very deep notches or dips in its frequency response The spacing of the notches along the frequency axis is at multiples of the lowest frequency notch so they look evenly spaced along a graph plotted on a linear frequency scale On the more common logarithmic frequency scale the notches become closer together on the paper as frequency increases A comb filter is produced when a signal is time delayed and added to itself Frequencies where the time delay is one half the period and multiples of these frequencies are concealed when the signals are D 2 Rev 1 1 11 15 94 combined because they have opposite polarity If the signals are of equal strength the cancellation is perfect and the notches are infinitely deep TAD See also flanging phasing Compressor An audio device which reduces the dynamic range of a signal The compressor is the first part of a compander the combination of a compressor and expander The effect of the compressor is to make the loud parts of a signal softer and to make the very soft parts louder Compressors are frequen
174. og An audio signal is an electrical replica or analog of the waveform of the sound it represents The voltage of the signal varies up and down negatively and positively in electrical terminology the same way as the sound pressure varies up and down at the microphone As long as the signal is in this form i e is a voltage that varies directly with the sound pressure it is an analog and audio devices which use such signals are analog devices The majority of audio devices are analog in nature though digital devices are increasing in popularity An analog audio device need not be electrical the Edison mechanical phonograph was an analog device the groove depth being an analog of the sound pressure at the recording diaphragm TAD AES EBU A digital audio transmission system standardized by the Audio Engineering Society and the European Broadcast Union An AES EBU signal carries two audio channels as well as status information The AES EBU interface is balanced and uses XLR connectors There are subtle differences in the actual signal format from the S PDIF system An automatic gain control AGC circuit adjusts the gain of an audio device in inverse proportion to the signal level entering the device An example is a portable tape recorder which is designed for speech recording When the talker is close to the microphone the gain is reduced so as not to overload the tape As the level from the talker decreases for instance because of a
175. ogram audio That is they inhibited the gain reduction release if there was no audio present If audio was present then the compressor was free to release as much as it wanted but if there was no audio present the unit remained at the amount of gain reduction in force before the audio loss Both the Audimax and the Level Devil depended upon silence to control the gain release function In practice the silence detector can be fooled by a noisy input signal Since the AGC Leveler needs to work at very low threshold levels in order to accommodate a wide range of input levels ideally you want the AGC Leveler to function with signals ranging from near thermal noise to high line level an ordinary signal present detector would respond to hum or noise by mistaking it for a valid signal If you try using a simple compressor as there is no signal controlled gated release function Thus the overall gain is highest anytime that the signal falls below the compressor s threshold By itself this isn t disastrous perfectly workable with a noiseless input signal but the sudden change in noise level when a normal level signal presents itself is a dead giveaway that your compressor is lacking in the IQ department This is the BUT mentioned earlier The 602 s AGC function performs some analysis on the signal in order to make an informed decision about the signal s nature If the signal is determined to be noise or silence then the AGC s release f
176. ohm load The ISP shall be capable of operating by means of its own built in power supply connected to 117V nominal ac 105 130V 50 60 Hz 20 watts 230V nominal 207 253V ac 50 Hz where applicable The unit shall be a Symetrix Incorporated model 602 Stereo Digital Processor E 2 Rev 1 1 11 15 94 Disassembly Instructions Caution These servicing instructions are for use by qualified personnel only To avoid electric shock do not perform any servicing other than that contained in the operating instructions portion of this manual unless you are qualified to do so Refer all servicing to qualified service personnel Caution Parts of the 602 use surface mounted semiconductors Removing and replacing these parts requires special tooling and special techniques This is not a job for the faint of heart nor is it something that you should attempt for the first second or even third time Do not attempt this at home We strongly advise that you should refer all servicing to the factory Any damage to the 602 that in our sole opinion resulted from improper surface mount technique or improper tooling is not covered by the warranty Warning Lethal voltages are present inside the chassis Perform all service work with the unit disconnected from all AC power F 1 Top Cover Removal Ensure that the 602 is disconnected from the AC power source Remove three 6 32 x 1 2 inch screws from each side of the chassis Remove two 6 32 x X 1 4 inc
177. on error The amount of error will always be within one least significant bit LSB therefore the smaller the LSB the better In quantization of a sine wave whose frequency is a submultiple of the sampling frequency the error will have a definite pattern which repeats at a frequency of the signal Thus it will have a frequency content consisting of multiples of this frequency and it can be considered as harmonic distortion rather than noise For music however the signal is constantly changing and no such regularity exists The quantization error is then wideband noise and is called quantization noise Quantization noise is difficult to measure because it does not exist without a signal A sine test signal is not good because sometimes this results in distortion not noise If the sinewave frequency is chosen so it is not a submultiple of the sampling frequency the quantization errors will be more nearly randomized and will resemble random noise TAD Ratio Short for compression ratio or expansion ratio The term stands for the ratio of the change in the input signal of a device to the change in the devices output When graphed on linear scaled graph paper the result is the familiar compression ratio curve assuming the device is a compressor Although the term is most commonly used for compressors and expanders there is no reason why it cannot be used for any device that alters its gain in some signal level dependent manner i e de essers
178. ons are available from the Audio Engineering Society 60 E 42nd Street New York NY 10165 2520 212 661 8528 Digital Audio Collected Papers from the AES Premiere Conference Rye New York 1982 Present and Future of Digital Audio Tokyo Japan 1985 Music and Digital Technology Los Angeles CA 1987 The Journal of the Audio Engineering Society published monthly Rev 1 1 11 15 94 Notes Rev 1 1 11 15 94 E Architect s and Engineer s Specification The integrated signal processor ISP shall be a dual input dual output model accepting line level signals applying frequency response equalization delay based effects and signal dynamics processing to that signal and delivering the processed input signal to two outputs All signal processing equalization delay dynamics shall take place in the digital domain The ISP shall occupy one rack space 1U The equalizer block shall take the form of a user and MIDI programmable parametric equalizer capable of operating at three inflection points simultaneously Band 1 of the equalizer shall be switchable between a lowpass shelving characteristic or a peak dip characteristic Band 3 of the equalizer shall be switchable between a highpass shelving characteristic or a peak dip characteristic All three bands of the equalizer shall be capable of operating over the following frequency ranges and bandwidths 31 to 21
179. ontrol Chain Hipass See Frequency Table Parameters Freq 66 Log Averaging Filter See Tc Table Tc 67 Sidechain Lookahead 0 0 5 127 2 6 ms 48kHz See Sidechain Lookahead Time Table Modifying some of these parameters incorrectly can result in improperly operating modules but reloading the program will restore the original settings To edit one of these parameters use the realtime editing mode 1 Cycle through the MIDI switch until the display reads rEAL then hold down the switch The display reads bLC Block 2 Use the Wheel to select SEt Press the MIDI switch again The display reads PAr 3 Select the parameter to edit using the Wheel See Appendix C Realtime MIDI The OUTPUT HEADROOM display displays the 0 127 scaled level of the parameter Press the MIDI switch again The display reads oFt Offset 4 Use the Wheel to enter the desired value 5 Press LEAVE EDIT to exit the realtime editing mode 6 When using the realtime modulation modes the offset adjustment sets the new value within a resolution of 2 steps Shortcut 1 From any control level other than MIDI press and hold the MIDI button to access the realtime editor The display shows bLC Block when you are successful 2 Use steps 2 through 6 above to set the desired parameter Rev 1 1 11 15 94 7 9 75 Tips and Techniques for Using the 602 Following are some tips and techniques for using the 602 You should consider any sett
180. or AES EBU digital input This may be useful in situations using a single master clock source Designate the digital input as the clock source via the GLOBAL parameter switch parameter CLCE If you are using the analog outputs connect them to your console s balanced line inputs If you are driving an unbalanced input pin 3 of the XLR connector should float If your audio system uses pin 3 of the XLR connector as the hot connection then pin 2 of the XLR connector must float This is described in greater detail in Chapter 3 If you are using the digital inputs connect them to an appropriate digital source Set the push push switch to correspond to the input that you are using Rev 1 1 11 15 94 6 1 If you are using the digital outputs connect them to an appropriate digital input Set the push push switch to correspond to the output that you are using There is no need to observe polarity with regard to either of the AES EBU I O connectors The digital system is immune to polarity reversals on the signal wiring Connect the AC input to an AC power source of the proper voltage and frequency as marked on the rear of the unit Caution Failure to connect the 602 to the proper AC mains voltage may cause fire and or internal damage There are no user serviceable parts inside the chassis Refer all service to qualified service personnel or to the factory Warning Lethal voltages are present inside the chassis There are no user servi
181. ory location The SAVE switch flashes if the edit buffer has been modified is dirty and memory protection has not been enabled If the edit buffer is clean and memory protection disabled the LED illuminates steadily The LED is off when program memory is protected When memory protection is on trying to save the edit buffer displays the Prt protected message With memory protection off a long press saves the edit buffer using the currently displayed program number number seen when not in edit mode The display shows donE after the save operation has completed Remember that program numbers above 128 are reserved and always write protected The following table shows the effects of memory protection the edit buffer state and the SAVE switch Edit Buffer clean clean Memory SAVELED SAVE switch Display Protection reads enabled on no save Om El Dcum program saved COMPARE Switch LOAD Switch LOAD Switch cont d The COMPARE switch toggles the 602 s settings between those stored in the edit buffer and those stored in program memory This allows making quick a b comparisons between the original program and the current settings The display toggles between OLD and CURR to help you keep things straight The LOAD switch loads a copy of the program whose number currently shows in the display into the edit buffer for editing The display reads donE when the operation is complete If the pro
182. panel you must convert them via the parameter tables found at the end of Appendix C For instance on the MRC the feedback FB parameter varies from O to 127 If you set slider 3 on the MRC to minimum and then listen to the 602 s output the result should be intense echo with no signs of decay Why is this A glance at the Delay Feedback Table in Appendix C shows that sending the 602 a feedback value of O results in maximum negative reverse polarity feedback To set the feedback to O no feedback you must set slider 3 on the MRC so the MRC s display reads 64 Rev 1 1 11 15 94 C MIDI Implementation Notes This appendix describes the MIDI implementation of the 602 If you are a newcomer to MIDI you would do well to familiarize yourself with MIDI and its usage by reading one of the many introductory level books available at booksellers C 1 Overview There are two MIDI messages of importance to the 602 MIDI Control Change and MIDI Sysex The standard MIDI implementation table may be found in Appendix G The 602 responds to MIDI messages containing its unique device type and unit number as well as MIDI messages matching only its device type provided that Omni mode has been turned or MIDI Control Change messages affect volume panning bank select and omni mode MIDI Program Change commands change user programs in conjunction with the Bank Select command All other 602 program changes set get program data identify request occur
183. paulated ep RN rotae rre ttu 2 9 QUE e A REN NE 2 10 2 1a Program Memoriae iiit Neda etie teehee chee dae uestes 2 10 3 Technical THOM al i ccicivantncacuecsusdevencaancseaduadsdeadeocdsancuesdvencusdsverduascvencucssecs 3 1 3 1 Matching Levels vs Matching 3 1 3 2 Signal e o a i o Ave 3 2 3 3 I O Impedances 20 21 2350 3 2 3 4 Polarity Convention eet RC EE 3 3 3 5 Input and Output Connections enn 3 3 3 6 Digital I O Considerations naso ghe 3 5 3 7 MIDI I O Considerations nennen 3 5 4 Front Panel OvervieW rrr recon n nein sre e cei 4 1 4 1 User Interface Summary Side iot ea dete iit pe Pee eR as 4 1 4 1 1 Loading ve TOGA NAR EAR NDA 4 1 44 2 Saving PROOF REOS i ede irat Mee ted age irse ndun 4 1 4 1 3 Comparing Programs o eee Sean RB RIED aa 4 2 4 2 Rate of Change Parameter ce S ura be phan 4 2 4 3 Input Level Control Block 4 2 no Made eb 4 3 4 4 1 EQ Band 4 3 4 4 2 EQ Parameter pha Rd bei 4 4 4 5 Dynamics Processing 00 400 4 4 4 5 1 Dynamic Noise Reduction BIOCK e rrr
184. r salesman during the 3AM movie use a 10 1 ratio setting 10 dB or more of gain reduction and a fast release time fast enough to cause breathing 7 5 9 Automatic Gain Control is simply a smart compressor that knows when to allow its gain to change This simple concept allows using a compressor to track a varying audio signal while maintaining a more constant output level Note that the goal is to reduce the overall variation in signal level not to remove all variation completely You set the AGC much like you set the 602 s compressor The big difference is the THRESH setting which becomes the auto release threshold This determines the level at which the compressor allows its gain to rise You don t want the gain to rise trying to track a signal buried in noise right Set the THRESH so that the lowest desired signal causes fluctuation in the gain reduction meter 7 5 10 Downward Expander The downward expander reduces its gain for any signal level below the threshold setting Typically downward expanders are used to remove noise or unwanted signal from an audio signal by simply lowering the gain when the overall level falls below threshold Think about using the expander when you are faced with a noisy signal not necessarily hiss or when heavily compressing a voice and you want to remove some of the less desirable artifacts false teeth rattling lip smacking tongue noise etc You can also use the expander to
185. r to remove any components that are higher in frequency than one half the sampling frequency This is because it requires at least two samples per cycle to determine the existence and strength of a frequency component that is it would require at least one hundred samples per second to encode a tone of 50 Hz The A D process will create spurious signal called aliased components if this rule is not followed Rev 1 1 11 15 94 D 1 In order to affect the audible signal as little as possible an anti aliasing filter is designed to be very steep having an extremely rapid fall off above the upper frequency limit TAD abridged See also brickwall filter analog to digital converter digital to analog converter Anti Imaging Filter In a digital audio system in order to recover the signal from the digital words a D A converter is used The output of this is a stair step type of waveform which contains a great deal of high frequency energy called images To reconstruct a smooth replica of the original signal the stair step is passed through a steep low pass filter called an anti imaging filter It is similar or even identical to the anti aliasing filter found at the input of the A D converter but its purpose is quite different TAD See also brickwall filter analog to digital converter digital to analog converter Attack Time Attack time is the time it takes for a compressor or limiter to reduce its gain after a strong signal is applied to it
186. right digital CH2 DE PROCESSOR CONVERTER RIGHT inputs of the DSP chain DATA BUSS CONTROL INPUTS d STEREO SIGNALS Stereo signals applied mere 1 EE MONO SIGNALS via the digital inputs 1 remain stereo gt CONTROL DATA Rev B Figure 6 1 Simplified block diagram Mono signals applied via the analog inputs can emerge from one or both outputs if you use the delay with some difference in the delay times you can stereoize the output 6 1 Connections Connect your analog input source to the appropriate XLR connector The line inputs are intended for balanced or unbalanced line level inputs with signal levels between 10 and 4 dBu Connect the 602 s analog outputs to your console s line inputs using the XLR connectors The digital input output connectors are intended for sources or loads conforming to the S PDIF or AES EBU digital interface standards The 602 s digital input accepts any word length up to 24 bits The analog and digital inputs and outputs may be used in any combination i e analog in analog out analog in digital out digital in digital out digital in analog out The 602 operates at either 44 1 kHz or 48 0 kHz sample rates input and output are always the same rate If you are using the 602 s analog inputs with the digital outputs you can supply an external sample rate reference signal via the S PDIF
187. rt of the display reads the higest peak digital signal level of the two input channels left and right after the digital gain control at the input to the DSP section The display can temporarily indicate the signal level of the left or right DSP channels via the GLOBAL switch Rev 1 1 11 15 94 4 2 ev 1 1 11 15 9 The Clip LED responds only to overload at the output of the analog line input amplifiers To maximize the dynamic range set either of the two input gain controls Ch 1 Stereo or Ch 2 so that the green 2 dB LED illuminates The red CLIP LED should never illuminate DIGITAL IN SYNC The DIGITAL IN SYNC LED indicates the presence of digital signals at either the AES EBU or S PDIF digital inputs This LED also flashes to indicate error conditions including no signal present occurring with either of the digital inputs CH1 Stereo LED This LED indicates that the CH1 Stereo gain control is active CHI This rotary control determines the gain of the channel 1 line input circuit In stereo mode the ch1 and ch2 LEDs indicate whether the gain controls are separated or ganged CH2 LED This LED indicates that the CH2 control is active CH2 This rotary control determines the gain of the channel 2 line input circuit The three status LEDs indicate the status of their associated input as well as indicating error conditions For the analog inputs the status LEDs also indicate which gain control is active If the DIGITAL IN SYNC
188. ry Symetrix expressly disclaims any IMPLIED warranties including fitness for a particular purpose or merchantability Symetrix s warranty obligation and buyer s remedies hereunder are SOLELY and exclusively as stated herein This Symetrix product is designed and manufactured for use in professional and studio audio systems and is not intended for other usage With respect to products purchased by consumers for personal family or household use Symetrix expressly disclaims all implied warranties including but not limited to warranties of merchantability and fitness for a particular purpose This limited warranty with all terms conditions and disclaimers set forth herein shall extend to the original purchaser and anyone who purchases the product within the specified warranty period Warranty Registration must be completed and mailed to Symetrix within thirty 30 days of the date of purchase Symetrix does not authorize any third party including any dealer or sales representative to assume any liability or make any additional warranties or representation regarding this product information on behalf of Symetrix This limited warranty gives the buyer certain rights You may have additional rights provided by applicable law Limitation of Liability The total liability of Symetrix on any claim whether in contract tort including negligence or otherwise arising out of connected with or resulting from the manufacture sale delivery
189. s 1 4A 250V ac slow blowing Bussman type MDL Operating Location Do not operate this equipment under any of the following conditions explosive atmospheres in wet locations in inclement weather improper or unknown AC mains voltage or if improperly fused Stay Out of the Box To avoid personal injury or worse do not remove the product covers or panels Do not operate the product without the covers and panels properly installed User serviceable parts There are no user serviceable parts inside the 602 In case of failure refer all servicing to the factory The complexity of the DSP circuitry as well as the special assembly tools required make the feasibility of field service doubtful 1 4 Rev 1 1 11 15 94 2 Basics If the particular combination of processors in the 602 is strange or foreign to you then we suggest that you read and digest this section of the manual If you should find some of the terminology strange you ll find a glossary of terms at the end of the manual A very good dictionary style reference is also listed in the Bibliography 2 1 What Does the 602 Do The 602 is a unique combination of four digital signal processors in one box a versatile three band parametric equalizer a dynamic filter a dynamics processor and a digital delay All of these processors are implemented in the digital domain and the 602 can accept or output signals in either the analog or digital domains One way to thi
190. s or setting any parameter within the edit buffer A short press on the MIDI switch accesses the following parameters CH nn U nnn dnEd dnAI rEAL Rev 1 1 11 15 94 Sets the MIDI channel number where nn can be AL for omni mode or 1 16 for a specific MIDI channel number Stored as global parameter Sets the MIDI unit number Allows multiple 601s to share the same MIDI channel number for sysex type messages nnn can range from O 7e for specific unit numbers or AL to ignore the unit number in sysex messages Stored as global parameter Downloads the edit buffer Holding down the switch long press sends out the current state of the edit buffer as a sysex message After the complete buffer has been sent the display reads donE Downloads all stored programs and all ROM programs Holding down the switch long press sends out all programs After the dump is complete the display reads donE During the dump the decimal point walks to show progress Allows creating realtime MIDI setups as well as setting any edit buffer parameter from the front panel For a complete discussion refer to Chapter 7 and Appendix A of this manual A long press on the MIDI switch accesses the realtime block editor This is described in greater detail in Appendix A of this manual 4 9 Presets Group This group of switches handles memory and program related tasks SAVE Switch The SAVE switch saves the contents of the edit buffer to the selected mem
191. singing voices have fundamentals with harmonics in the 5 kHz region of good ear sensitivity Voices that are powerful or rich with harmonics at 5 kHz sound especially pleasing clear and full Male opera singers are particularly favored with 5 kHz sounds women less so In popular music this range shifts downward somewhat It follows that voices deficient in the 5 kHz range can be enhanced in listening value by a generous boost on the order of 5 to 8 dB at 5 kHz A secondary benefit of this boost is an apparent increase in level a rise at 5 kHz frequently gives an apparent increase of 3 dB to the overall signal Attenuating the 5 kHz range on instruments gives a transparent quality to the sound providing of course that the remainder of the signal is otherwise wide range Microphones having a dip in this region lack the punch or presence to which we Americans are accustomed 2 4 6 Brilliance Unvoiced consonants attributed tooth tongue and lip sounds are high in frequency and reach the 10 kHz range These frequencies account for some clarity and most brilliance even though they contain less than 2 of the total speech energy This also holds true for musical instruments especially percussion Boosting or cutting this range affects clarity and naturalness Rev 1 1 11 15 94 2 3 In speech the 9th and 10th octaves impart intimacy although too much emphasis can make secondary speech sounas lip smacking etc objectionable
192. ss 0 70 51 Realtime Block 1 Floor Clip 0 127 Minimum normalized edit buffer value C 12 52 Realtime Block 1 Ceiling Clip 0 127 0 127 Maximum normalized edit buffer value Rev 1 1 11 15 94 Table 24 Realtime MIDI Block 2 Offset Description Range Reference dec hex dec 76 4C Control 0 None Off default 1 Control Change 2 Aftertouch 3 Pitch bend msb 7 bit 4 Delay section modulation oscillator 1 5 Delay section modulation oscillator 2 6 Log signal level dynamics section 7 NR center freq 8 Instantaneous gain reduction value compressor 9 Instantaneous gain reduction value expander 10 Block 1 output 3 Byte MIDI Message Second Parameter 0 127 4E Control Offset dec 0 127 64 0 0 128 127 127 79 4F Control Scaling 0 127 64 No Effect 0 4 127 4 Realtime Scaling Table The Program Name offset 84 99 is not accessible from the front panel it is only accessible via MIDI You can use this with an external MIDI editor to give your 602 programs meaningful to Parameter to Modify Offset Edit Buffer Offset Address you names Offset dec 81 82 83 84 99 99 Rev 1 1 11 15 94 hex 51 52 53 54 63 63 Table 25 Miscellaneous Description Realtime Block 1 F
193. t equipment types such as the FFT analyzer Most DSP devices have a microprocessor inside them to do most of the work TAD The 602 uses two DSP 56001 digital signal processors and a 68HC11 microprocessor Digital to Analog Converter DAC The component within a digital audio device which converts binary digital words into an analog signal that can be amplified and sent to a loudspeaker etc The DAC is the last link in the digital chain just before the anti imaging filter TAD See also analog to digital converter Distortion Theoretically any addition or modification to a signal caused by any type of equipment could be called distortion but the term has come to be somewhat more restricted in its use Distortion may be conveniently grouped into six types 1 Nonlinear distortion manifested as harmonic distortion and intermodulation distortion Harmonic distortion is the production of harmonics of the original signal by the equipment Intermodulation distortion is the production of sum and difference products of the various frequency components that make up an audio signal 2 Frequency distortion the unequal amplification of different frequencies 3 Phase distortion an effect caused when phase shift in an audio device is not a linear function of frequency In other words different frequencies experience different time delays 4 Transient distortion including transient intermodulation distortion TIM Scale distortion or volume d
194. t panel knob s for the parameter s The Lexicon MRC MIDI Remote Controller can be used to edit the internal dynamic midi settings in the 602 by following these steps Although this chapter is devoted to the MRC and its use with the 602 there are other MIDI controllers available that can perform comparably This procedure was developed on an MRC having software revision 3 01 This procedure uses machine 15 MIDI port 1 setup 9 to control the delay block and setup 10 to control the real time MIDI block The following table lists the steps needed to program the to accomplish this We assume that you already own and are somewhat familiar with how it works The table is divided into steps and each step has four parts MRC Key MRC Display Data to Set Enter and Comments MRC Key represents a key on MRC that you must press MRC Display represents the display on MRC In some cases this represents a portion of the MRC display for instance a label for one of the sliders O Data to Set Enter represents data that you must enter into the MRC via the sliders buttons or keypad O Comments are just that comments The following 9 steps exactly parallel the first 9 steps of the table Refer also to the edit buffer parameter tables in Appendix C Read the notes presented after the procedure They explain some of the details behind the steps This should help if you re trying to translate the procedure to a differ
195. t source to slr5 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 5E mfrID2 Rev 1 1 11 15 94 181 182 183 184 185 186 187 188 189 page page page Displays DEFINE SYSEX BYTES 8 9 LABEL FOR slr5 Set switch 1 to set the mix to 0 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 page page page page page page SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slr5 Set switch 2 to turn on sine modulati 209 210 211 212 213 214 215 216 217 page page page Rev 1 1 1 SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX 1 15 94 BYTES 4 7 on Data to Set Enter 01 00 1 24 BYTE ROC swtl SYSEX FO 00 00 5E 01 00 1 27 KILL swt2 SYSEX FO 00 00 Comments device type unit channel edit buffer data set edit buffer 36 send slider setting use sliders 1 4 to set label to RO C setup switch 1 use button 1 to set source to swtl use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set ed
196. tar Fingered Guitar 12 String Guitar w Chorus E Guitar Chorus 1 E Guitar Chorus 2 Sweetening and Effects RAM ROM 40 168 41 169 42 170 43 171 44 172 45 173 46 174 47 175 48 176 49 177 50 178 51 179 52 180 53 181 54 182 55 183 56 184 57 185 58 186 59 187 60 188 61 189 62 190 63 191 64 192 65 193 66 194 67 195 68 196 69 197 70 198 71 199 72 200 73 201 74 202 75 203 76 204 77 205 78 206 79 207 1 1 11 15 94 Bad Audio Restoration Tape Transfer Mastering Tape Transfer Voice Basic Echo Random Pitch Echoes Big Troll Roon Troll Leslie Simulator Auto Pan Slow Slow Effects Fade Falling Pitch Trail Unison Singalong Telephone Simulator Computer Voice Darth Vader Munchkins 64 Funny Cars Stereo Randomizer Quasi Stereo Seventies Flanger Description EQ set for flute with light chorus effect signals above the threshold of the Dynamics section will modulate the chorus effect Fat bright sound for metal snare Deep round sound for kick drum Use line input adds chorus effect Bright setting for steel strings EQ set for Grand Piano Compressor set for containment slight delay for ambience Acoustic guitar program Chorus effect for acoustic piano Use line input EQ set for Bass w chorus effect Long echoes for electric guitar tight chorus effect for guitar Compressor set for bass slaps w chorus effect Guild G37 W Audio Technica 403
197. though 1 4 inch jacks are typically used for things like guitars which are high impedance and unbalanced this does not predispose them to only this usage After all 1 4 inch jacks are sometimes used for loudspeakers which are anything but high impedance Therefore the presence of 3 pin XLR connectors should not be construed to mean that the input or output is low impedance or high impedance The same applies to 1 4 inch jacks So what is really important Signal level and to a much lesser degree the impedance relation between an output signal source and the input that it connects to signal receiver Signal level is very important Mismatch causes either loss of headroom or loss of signal to noise ratio Thus microphone inputs should only see signals originating from a microphone a direct DI box or an output designated microphone level output Electrically this is in the range of approximately 70 to 20 dBm Line inputs should only see signals in the 10 to 24 dBm dBu range Guitars high impedance microphones and many electronic keyboards do not qualify as line level sources The impedance relation between outputs and inputs needs to be considered but only in the following way Always make sure that a device s input impedance is higher than the output source impedance of the device that drives it Some manufacturers state a relatively high impedance figure as the output impedance of their equipment What they really me
198. threshold low enough to achieve 6 dB or so of gain reduction Set the attack time to retain some of the edge of each word and set the release time fast enough to follow the speech For a heavily limited sound set the ratio Figure 2 4 Input Output curves of a compressor higher 10 1 or higher and use very fast release times For musical applications use low ratios 1 5 1 to 6 1 unless you want a deliberately squashed sound Set the threshold to achieve 4 to 6 dB of gain reduction This setting is useful for subtly controlling occasional peaks To prevent peak overload of a subsequent device use the highest ratio and set the threshold to achieve 2 to 4 dB gain reduction on the highest peaks If you re using the 602 to ride gain on mixed program consider using the AGC section 2 9 AGC The letters AGC stand for Automatic Gain Control An AGC can also be considered as a special case of a compressor having a relatively low ratio 1 1 gt 4 1 and a very low threshold level and a gated release time Thus any signal that exceeds the threshold causes some degree of gain reduction Additional gain applied after the compressor brings the signal level back up to line level Functionally an AGC works like an invisible and hopefully inaudible operator who monitors the audio level and imperceptibly raises or lowers the gain to maintain the audio level at some predetermined point In i
199. tly used in recording popular music and in radio broadcasting where very soft passages may be lost in the background noise of the listening environment For instance when music is playing on the radio in a car the car s noise level will easily mask the quieter musical passages The limiter acts something like a compressor but operates only at the top end of the dynamic range The subjective audibility of a compressor depends strongly on its time constants attack and release times and they are selected with care to minimize obvious pumping of the volume To restore the original dynamics to a compressed signal a volume expander can be used but great care must be taken that the time constants slopes and thresholds match those of the compressor TAD Condenser capacitor microphone One of the earliest types of microphones to be invented after Dr Lee DeForest invented the Audion amplifier in 1906 was the condenser microphone Thomas Edison is sometimes credited with its invention but this seems to be in doubt At any rate Wente of Bell Telephone Labs designed a condenser microphone in 1917 and introduced it commercially in 1931 The condenser microphone is a very simple mechanical system with almost no moving parts compared to other microphone types It is simply a thin stretched diaphragm held very close to a metal disc called a backplate This arrangement is an electrical capacitor and it is given an electric charge by an external voltage
200. to taste or need It s generally easier to apply boost to a sound for shaping and that s how many engineers start Many times however you may want to experiment with removing an offending sound as opposed to drowning it out with something else In a complex mix this may work better because it may require less overall EQ to remove the offending sound the end result will sound more natural 7 5 5 Metering and the Dynamics Block Each component of the dynamics block uses a concept called gain reduction Gain reduction is the degree to which the overall gain has been lowered in response to some signal condition When adjusting any of the dynamics block components dynamic noise reduction de esser expander compressor or AGC the right hand LED meter changes to a gain reduction meter Use the lower scale to translate the meter indication into numbers The meter reverts to displaying level whenever you leave any of the dynamics block 7 5 6 Dynamic Noise Reduction The dynamic noise reducer NR uses a sliding lowpass filter controlled by the relative level of the signal rejected by the filter This topology makes a filter that responds more to the content of the signal than its absolute level it is easier to adjust There are three front panel adjustable parameters FREQ and THRESH The FREQ parameter sets the resting frequency of the sliding filter and has a range of 1 kHz to 21 11 kHz The THRESH parameter sets the relative threshold
201. trol the offset applied to slider 1 s source 39 page 40 41 42 43 page 44 45 46 47 page 48 49 50 51 52 page 53 54 55 page 56 Make slider 4 control the multiplier value 57 page 59 60 61 page 62 64 65 page 66 67 69 70 page 71 72 73 page Rev 1 1 11 15 94 SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slr3 SOURCE DEST OUT SOURCE DEST OUT DEFINE SYSEX BYTES 2 3 DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR slr2 2NDP slr3 SYSEX 1 FO 00 00 5E 01 00 1C 49 OFFS slr4 SYSEX FO 00 00 5E 01 00 1C BYTE use sliders 1 4 to set label to 2 N D P setup for slider 3 use button 1 to set source to slr3 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 next page SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 73 send slider setting use sliders 1 4 to set label to O F FS setup for slider 4 use button 1 to set source to slr4 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 mfrID2 device type unit channel edit buffer data set edit buffer 74 send slider setting B 3 STEP MRCK
202. ts most simplistic form that s all there is to it But there is a BUT as you will see AGC amplifiers have been with us for many years In the broadcast world the old Gates Level Devil and CBS Labs Audimax are both examples of old circa 1960 products that performed this function The feature that sets these guys apart from common ordinary garden variety compressors is gated program controlled release 3 Fast attack times may become audible because of the time required to compute the amount of gain reduction analog compressors have the same problem but they usually limit the minimum attack time so you never have the problem You can use the 602 s compressor look ahead parameter to buy some time so that the initial overshoot of the signal is controlled by the compressor Fast release times cause problems of their own because changes in the gain reduction may occur during a single cycle of the waveform causing distortion Again analog compressors are not immune to this problem either 2 8 Rev 1 1 11 15 94 If you remember back to the early days of TV remember when someone at the network screwed up and let the program lapse the compressor at the local station would release and whoooosh up would come the noise floor until the guy at network woke up in which case it went suuuuuuuck and back into the program audio Both the Level Devil and Audimax fixed this problem by making the release time of the compressor a function of the pr
203. tting The compressor s makup gain may be set manually by pressing the RATIO switch until the display reads gAln Set the makeup gain using the Wheel THRESHOLD Switch Sets threshold for the start of compression Signals above this level are reduced in level by an amount dependent on the setting of the compression ratio and the difference between the threshold setting and the actual signal level Rev 1 1 11 15 94 4 7 4 5 5 AGC Block The AGC is a variation on a compressor that operates over a wide range of signal levels while trying to keep its output level constant The AGC switch LED indicates that the AGC is active When editing the AGC switch LED flashes otherwise it reflects the state of the AGC in or out Whenever editing the AGC parameters the output LED display changes to read gain reduction The AGC block and the compressor block are mutually exclusive you can only use one of them at a time There is no output gain control the 602 computes the correct amount of makeup gain based on the threshold and ratio settings although the auto makeup gain feature can be defeated and the amount of makeup gain can be set manually The shape of the knee of the gain reduction curve can be adjusted via MIDI or the realtime editor see Appendix A AGC Switch The AGC switch LED indicates that the AGC Leveler is active When editing the switch LED flashes otherwise it reflects the state of the AGC in or out There is no output
204. tup 0 lt gt FO Value Range hex 1 lt mfriD o gt Lo dec 2 lt mfrID 1 gt 0 0 lt sysex gt FO 3 a 2 5E 1 mfrID 0 lo 4 sdevice type gt 2 2 1 5 lt unit channel gt O 7E 7F all 3 mfrID 2 6 command 3 4 lt device type gt 7 MIDI channel gt 0 0 7F 5 unit channel 0 7 7F al omni 6 lt command gt ID 8 7 lt Offset to start of 9 data top 2 bits 10 M 8 Offset to start of 11 E data middle 7 bits 12 9 lt Offset bottom 7 13 R LER 10 number of bytes 15 top 2 bits 16 ih 11 number middle 7 17 6 bits gt 18 ig 12 number bottom 7 bits 19 2 13 data bytes 20 lt gt F7 data data lt gt F7 Purpose Request the identity of a 602 Value lt sysex gt lt mfrID 0 gt lt mfrID 1 gt lt mfrID 2 lt device type gt lt unit channel gt lt command gt lt gt C 6 Rev 1 1 11 15 94 The parameter map shows the location of various entities within the memory space of the 602 You can access these by using the program setup data write command 1Dh Table 9 Parameter Map Offset dec Description 0 99 Edit Buffer RAM Program 1 12800 12899 R
205. ture Per Program This data structure is a table of offsets each of which represents one parameter in the edit buffer Parameter 47h or 71d 47 hex equals 71 decimal is the Event Type under Real time MIDI Block 1 When the MRC sends the MIDI command represented by steps 10 through 20 the value sent at step 19 tells the 602 that the next data byte received gets stuffed into the edit buffer at offset 71 which is the Real time MIDI Event Type Step 20 This step has a value of BYTE which is the value represented by that slider s setting This is how you send a slider value to the 602 from the MRC Step 21 This step labels the slider so its function is a bit more obvious to humans Step 171 This step programs one of the switches buttons on the MRC The buttons are a little different than programming the sliders in that they only send one MIDI message per press and there is no way to create an ON OFF toggle on one button Instead you must program one button to send an OFF command and the other button to send an ON command Step 172 Notice that we set the source to switch 1 instead of slider 1 Step 185 Edit buffer offset 27h is the delay block mix control Sending a O value to in step 186 turns the delay off by making the level of the wet portion of the delay mix to zero When using the delay controller setup notice that the numbers shown in the MRC s display range from 0 127 To turn these numbers into 602 numbers that match the front
206. umann microphone with echoes short TLM 170 microphone with echoes long EQ optimized for speech and slight chorus effect Small room ambience with bright EQ settings Simulates small voice over booth AGC with low ARM sensitivity Program to simulate the Symetrix 421 AGC Leveler Neumann TLM 170 optimized for female speech Female EQ settings with light flanger effect Bright EQ settings with flutter echoes AGC program with EQ set for female speech AGC set for intelligibility Close intimate sound Morning show Drive time show Midnight program Voice image widened EV BK 1 Microphone with EQ set for live performance EV BK 1 Microphone with delay EQ set for Voice with Flange EQ set for the King with simulation of tape delay Set for live style performance with delay Institutional voice Widened image A touch of twang A bit of echo AKG C 535EB microphone with EQ set for Female voice C 535EB Microphone with delay settings Slight delay for width Harsh EQ for effect EQ set for vocal strength EQ set to cut through mix Delay set for doubling Stereo delay set at different intervals Adds stereo width More stereo width Rev 1 1 11 15 94 Instruments Mic and Line Inputs Name Flute W Chorus Pound Guitar Snare Kick Electric Piano Acoustic guitar piano Brass E Guitar Chet Chorus Piano Bass w chorus FX guitar Acoustic GTR w chorus wash Electric Bass slap Solo Acoustic Guitar Rhythm Acoustic Gui
207. unction is inhibited When the signal analyzer detects that the signal has returned the AGC is again allowed to release which causes the gain to rise or fall in response to the signal level 2 10 Delay One of the simplest things that you can do to an audio signal to dramatically change its character is to add in a delayed version of the signal By adjusting parameters such as delay time delay level and feedback you can create impressions of time space distance or reflection The delay in the 602 is a two channel delay with paralleled inputs and separated outputs The feedback paths between input and output are cross coupled This means that delay one s output feeds delay two s input and delay two s output feeds delay one s input For example you can get a 505 sound by adding 250 ms delay to a signal Adding feedback makes the delay effect linger since the feedback causes the echoes to repeat until they die out By shortening the delay and fiddling with the feedback you can simulate reflective rooms of various dimensions Making the delay times slightly different spreads the sound eliminating the point source effect If the delay times are long enough you ll hear the echoes bouncing back and forth between the speakers Note the dual delay used in the 602 is not sufficient to create any sort of realistic reverb Good sounding reverberation requires a multi tapped delay line 2 11 Modulated Delay Yet another wrinkle on delay is modu
208. vel then you ll need to pad attenuate the output of the 602 down to microphone level A simple U pad is sufficient A suitable design can be found in Figure 7 1 Although a far preferable connection would be to bypass your console or mixer s mic preamp this will Figure 7 1 A 44 dB U pad work Ensure that there is no phantom power present at the console s mic input terminals both sides of the mic input should read OV dc referenced to ground Note Padding attenuating the output of the 602 back to microphone level is a workable solution towards interfacing the 602 into a console or system having only microphone level inputs However workable the ultimate performance of the 602 will be limited by the performance of your system s existing microphone preamps If you can find a way to bypass the existing microphone preamps in your system do so It ll be worth the trouble 7 2 Rev 1 1 11 15 94 FRONTPANEL SWITC HES BATTERY AES EBU MIDI MIDI KUP AES EBU INPUT INPUT OUTPUT AND DISPLAYS OUTPUT SELECT i SELECT e 5 Ent PROGRAM IGR 21 AES EBU SUPERVISO RY INPUT CONTROLLER QE AES EBU AES EBU MIC ROPROCESSOR AES EBU INPUT OUTPUT 9 SELECT SELECT CIRCUITRY CIRC UITRY S PDI
209. via MIDI sysex messages An identification scheme allows a daisy chain of 602 s to share a MIDI bus Sending an identification request to the first unit in the chain causes all units to report their current MIDI channel and unit number along with the identifying string SYMETRIX 602 The responses are in the same order as the arrangement of units along the MIDI daisy chain In the following tables all numbers are written in the base decimal or hexadecimal listed at the head of each table Where necessary for clarity hexadecimal numbers are followed with h and decimal numbers are followed with d Type refers to the length of the request or response C 1 1 Control Change Bn Value dec The control change message commands all dec devices sharing a given MIDI channel to Volume 0 127 64dB input output a attn change one of the following parameters Typically only like devices share the same L R Pan MIDI channel xi Bank Select RAM 1 128 x ROM 129 C 1 1 1 Example m Command the 602 to set the output Omni Mode Off panning to center Omni Mode On 2 Send hex lt MIDI command gt lt nnnn gt lt data gt lt data gt Bn OA 40 where nis the MIDI channel number 0 F encoded in hex C 1 2 Realtime MIDI There are two Realtime MIDI setups available per program Each setup allows some predefined MIDI action to control any one parameter on the 602 An offset and scale factor multiplier t
210. wards the front of the unit then lift it clear of the chassis after the connector bodies clear the connector shells It may help to push on the connector bodies from the rear of the chassis Rev 1 1 11 15 94 F 1 F 2 2 Digital Board Removal Caution The circuitry within the 602 is static sensitive Use appropriate techniques to eliminate static electricity from your body and from the surrounding area If these techniques are not familiar to you you should refer servicing of your 602 to the factory 1 Disconnect the ribbon wire jumper between the analog and digital boards It is sufficient to remove disconnect only one end Disconnect the ribbon wire connectors connected to the Wheel and to the front panel circuit board Disconnect the power supply connector located at the front right of the unit Remove three 6 32 x 1 4 inch screws from the digital circuit board Unlock the XLR connector inserts using the procedure found elsewhere in this section Slide the digital board towards the front of the unit then lift it clear of the chassis after the connector bodies clear the connector shells It may help to push on the connector bodies from the rear of the chassis mee m 2 3 Power Supply Board Removal Disconnect the power supply connector located at the front right of the unit Remove four 6 32 x 1 4 inch screws from the power supply circuit board Remove two 6 32 x 1 4 inch screws from the heatsink attached to Remove two
211. wer amplifiers and RF Technically speaking the reason is power transfer which reaches a maximum when source and load are matched Modern audio systems are voltage transmission systems and source and load matching is not only unnecessary but undesirable as well Ancient audio systems operate at 600 ohms or some other impedance value and must be matched both at their inputs and at their outputs Generally speaking if you are dealing with equipment that uses vacuum tubes or was designed prior to 1970 you should be concerned about matching These units were designed when audio systems were based on maximum power transfer hence the need for input output matching O Power amplifiers are fussy because an abnormally low load impedance generally means visit to the amp hospital Thus it s important to know what the total impedance of the pile of speakers connected to the amplifier really is RF systems are matched because we really are concerned with maximum power transfer and with matching the impedance of the transmission line keeps nasty things from happening Video signals composite baseband or otherwise should be treated like RF Some folks seem to believe that balanced unbalanced lines and impedances are related or even worse that they are associated with a particular type of connector Not so Unbalanced signals are not necessarily high impedance and balanced signals lines are not necessarily low impedance Similarly al
212. witch DEPTH Switch Rev 1 1 11 15 94 A long press on the DELAY switch rt accesses the delay line rate of change with 0 1 being basically instantaneous and 9999 being very slow This adjustment along with the MODULATION DEPTH and RATE are used for chorus type effects or flanging n 90 oFF P 90 Sets the attenuation of the cross coupled feedback The range of control is from off to O dB attenuation The feedback signal may be in phase P nn or out of phase n nn where nn corresponds to the amount of attenuation applied to the feedback signal Thus P 10 corresponds to in phase positive feedback 10 dB down from unity gain Both channels are adjusted simultaneously The feedback polarity phase is especially important when creating flanging effects Pressing this switch repeatedly toggles between out and the current feedback setting The signal from the delay line drives a single pole 6 dB octave lowpass filter with a range of 600 to 18kHz The output of the lowpass filter then feeds the feedback and mix controls Sets the rate frequency for the delay time modulation generators DTMG The DTMG is either a random number generator whose value is updated rate times per second a sine wave of rate Hz or a triangle wave of rate Hz where rate is the value shown in the display Pressing and holding the RATE switch allows changing the DTMG from random rAnd to a sine wave SinE or to a triangle wave AnGL Sets the d
213. witch instantly returns you and the outputs to CUrr 4 2 Rate of Change Parameter In addition to the parameters visible on the front panel many of the 602 s parameters have a rate of change parameter rt associated with them The rate parameter affects how quickly the parameter changes from its current value to its new value either under direction of the front panel or MIDI In essence the rate parameter rt affects how fast the knob can turn the knob is a virtual knob that represents a parameter that can be adjusted using the Wheel or via MIDI You can see which parameters have an associated rate parameter by referring to the table in section C 3 1 2200 omases O su si PH 2 omo 6 COMPRESS ATTAGKInS RELEASEImS oues FEEDBACK BYPASS GLOBAL 602 2 clr STEREO o 6 PANDER rix MNS oy el ai 0 PROCESSOR nm momoe G O CH 1 STEREO 4 3 Input Level Control Block This switch and control block sets the operating conditions for the analog inputs of the 602 The controls and indicators operate as follows INPUT HEADROOM DB LED display indicates amount of headroom remaining at the output of the A D converters in the 602 The display ballistics are peak reading the display should be interpreted as the absolute amount of headroom remaining The numbered pa
214. wt4 use slider 2 to set DEST to SYSEX use slider 3 to set OUT to 1 SYSEX mfrIDO mfrID1 Rev 1 1 11 15 94 264 265 266 267 268 269 270 271 272 273 274 275 page page page page store Rev 1 1 1 Displays DEFINE SYSEX BYTES 4 7 DEFINE SYSEX BYTES 8 9 LABEL FOR swt4 Data to Set Enter 5E 01 00 1C 2A RAND 1 15 94 Comments mfrID2 device type unit channel edit buffer data set edit buffer 42 O random use sliders 1 4 to set label to N save your work you re done Notes If you take the time to key in this program into your MRC here s what you ll get Setup 9 Delay Block Setup 10 Real Time MIDI Controller Controller SW 1 sW 2 sW 3 SW 4 sW 1 SW 2 sW 3 SW 4 kill sine triangle random n a n a n a n a SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER 1 5 3 4 1 2 3 4 delay1 delay2 feedback mix Event 2nd parm offset scale type select SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER SLIDER 5 6 7 8 5 6 7 8 ROC filter speed depth parm n a n a n a Step 6 This tells the MRC that the slider that we want to program is slider 1 Steps 7 18 These steps are the same for every slider and button that we are programming in this example Step 19 This number 47h comes from Appendix C under the heading Data Struc
215. y holding down the LOAD switch regardless of its save status 4 1 2 Saving Programs To save a program press the LEAVE EDIT switch to return to the top most control level The red SAVE switch should be flashing if not then the program in the edit buffer has not been modified it doesn t need to be saved Turn the Wheel to select the desired save location memory locations 1 through 128 press and hold the SAVE switch until the display says donE Program numbers above 128 are read only The save operation displays if you try to save a program to one of these numbers Rev 1 1 11 15 94 4 1 If you press and hold the SAVE switch without pressing LEAVE EDIT first the 602 performs the W save operation using the current program number and returns you to where you were Caution If the LOAD switch is flashing this indicates that the current program number is different than the program number that the edit buffer was loaded from If you press and hold the SAVE switch you will overwrite the program number that is visible when you press LEAVE EDIT 4 1 3 Comparing Programs You can compare the program in the edit buffer with the unedited version of the program Pressing the COMPARE switch toggles the 602 between the edited and unedited versions of the program It is not possible to compare the edited program with any other program The display toggles between OLd and CUrrent to remind you what you re listening to Pressing any parameter s
216. yboards In the studio you could go as far as to use the AES EBU outputs as a direct digital output while listening to the analog outputs 8 6 Sound Reinforcement Applications One possible sound reinforcement application for the 602 is that of an ultimate channel insert processor Just think one channel insert patch and you have a parametric equalizer de esser de noiser compressor AGC and stereo delay at your fingertips Use just one use them all the important thing is that they re all there Another application simply uses the delay and possibly the EQ as an equalized stereo delay line The simplified user interface makes parameter changing fast and easy the programmability helps make changing modes easy Rev 1 1 11 15 94 8 1 8 2 Notes Rev 1 1 11 15 94 9 Troubleshooting Chart Symptom No output Hum or buzz in output Distortion Noise hiss No LED display No nothing Display reads Er nn nn is a two digit number Display flashes bAt at turn on Unit not plugged in but works anyway Rev 1 1 11 15 94 Probable Cause Check cables and connections Are inputs driven by outputs and outputs driving inputs Verify cables source and load by patching input and output connections together at the unit Check for AC power presence Check output by plugging headphones into analog output connector use an adapter Are the HEADROOM displays operating Check input and output connector

Owner`s Manual Symetrix 602 Stereo Digital Processor

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