Copyright: (c) Phobos Audio
Citation: (c) Phobos Audio


Trans - FM 2007 Digital manual
Trans-DIGI 2007

 Trans-DIGI 2007 is DSP based Audio Broadcast Processor

Some differencies between this manual and your actual firmware version can occur, because firmware is upgraded continually.

Version 5.05 improvements: ITU R 412 spectral mask and multiplex energy compatibility

(Communication parameters between this software and Trans-DIGI 2007 processor is: 1200,N,8,2)

© 2005 Phobos Engineering

This Help includes older wersion of Trans-DIGI manual. Some news (ITU R 412…..) are not included in this manual.

1	Basic information about equipment (Software ver. Classic)

1.1	Case

format 19", 1 unit, 320 mm depth

chassis: passive steel CD5B: surface is highly conductive

made from one piece of steel

cover: passive steel CD5B

cover is mounted by 12 screws

excellent electromagnetic compatibility

dural front panel, plastic vinyl surface

PCB with SMD

IEC power plug

1.2	Front panel

2 lines LCD display for control and adjustments

Power control LED display

Keyboard for adjustments

1.3	Rear  panel

IEC power plug

LED  indication of analog overload controlled from DSP

BNC output MPX (2x)

BNC output Pilot 19 kHz

port Remote RS 232 - Canon 9 Female for programming

XLR line output

XLR digital input AES/EBU (Fs 32, 44.1, 48 kHz)

XLR line output

AUX port for external reset (between pins 5,3) and day/night switch (pins 8,3)

Grounding screw M5

1.4	PCBs

Dual sided PCBs with SMD

DSP circuits Analog Devices SHARC

A/D a D/A converters Analog Devices

Reliable components

Excellent electromagnetic compatibility

Low noise analog circuits

sophisticated algorithms of signal processing

remote control via RS 232 port and Windows software

2	Important

Do not exposure appliance to rain or moisture. It prevents risk of fire or shock hazard.

Do not open equipment, high voltage inside. Refer service to qualified service personnel only.

Never install appliance in hot closed places or on direct sunlight. Free air circulation must be around rear and side panels.

Do not exposure appliance to rain or moisture. It prevents risk of fire or shock hazard.

2.1	Placing

Do not place appliance to places where is:

-extremely hot or cold (over 40 Celsius or less then 5 Celsius)

-dust

-high wet

-vibrations

-direct sunlight

If some trouble appears in run time, or some doubt about right operation appears, disconnect processor  from mains immediately and contact service organization or producer.

Appliance have not power switch.

Producer is not responsible for damages caused by using other way then following this manual.

3	Description

3.1	General features

·	DSP based digital modulation audio processor for FM

·	symmetrical XLR inputs, XLR AES/EBU digital input (Fs 32,44.1 and 48 kHz)

·	automatic digital deemphasis (50/15 us or J17)

·	built-in stereophonic encoder (digital FIR filtration, preemphasis 50us for European version), output for two transmitters and 19 kHz output of pilot signal

·	signal handling in 5 frequency bands, two independent levelers

·	RS 232 interface for adjustments of parameters, adjustment from front panel (keyboard, display)

·	software for 32 bit Windows (95,98,2000,ME,XP, NT) enables setting of all parameters and storage of presets

·	dural front panel with plastic vinyl surface

·	two years warranty

3.2	Basic characteristic

Phobos Audio Trans-DIGI 2007 DXC is DSP based modulation audio processor  for FM broadcasting. It works purely mathematically. Signal processing like filtering, compression, limiting etc. is realized by calculations on input samples. Processor works with floating point DSPs circuits. DSPs follows instructions stored in the program memory. Program memory can be upgraded and then structure of processing can be changed in future. All processing structure is then defined purely by software like in a computer programs. Processor works with two independent levelers and five band limiter.  It has build-in stereophonic encoder with output for two transmitters. Processor adjustments are made via RS 232 line from PC or by keyboard and display.

Parameters are stored in EEPROM non volatile memory.

Software for 32-bit Windows enable adjustments, storage of parameters into presets and files and uploading.

Parameters can be adjusted in wide range from slightly processed to highly competitive sound.

Circuit design and used algorithms interlocks clear processing without hearable distortion.

Output limiter and clipper controls peek deviation and enables to use full acceptable deviation (+/-75kHz).

Stereo encoder with FIR input filtering  generates output waveform via discrete D/A conversion. It enables very good spectral purity and stereo separation.

Procesor uses two microcontrollers and DSPs. First microcontroller is for communication with PC, keyboard, display and DSPs, second is for stereo encoder.

3.3	Principle of operation

You can see the block diagram of Trans-DIGI 2007 DXC in last section of this manual. Input section consists from desymmetrization block. A/D converters follows. A/D conversion works with 16 bit resolution and 32 kHz Fs. Sampling on 32 kHz is optimal solution for FM use. Because audio signals on FM are frequency limited to 16 kHz, 32 kHz is enough to carry all audio information transmitted via FM. Moreover, power of DSPs is optimally invested into sound algorithms.

Quality and complexity of sound algorithms makes sound and overshot protection on processor’s output. Instead of arguments with high sample rate we rather invest all DSPs power into quality of audio processing itself. Result of this is great pilot spectral protection and no overshot. Some our competitors use 48 or 96 kHz sample rate. It means, they need (in 96 kHz case) more then 3 times bigger power of DSPs. More because there is need of filtration of output audio signal to 16 kHz bandwidth. Some our competitors do not apply output filtration. Result is an overshot in stereo encoder. They need strong composite clipping to delete this over-shots. Pilot protection is very low in this case. Spectral purity in region of sub-carrier (57 kHz) is not good. It means RDS coverage can be reduced and stereo separation harmfully affected.

In our conception signal is spectrally limited to 16 kHz. We are using only safety clipper on composite output. Result is high pilot protection and high protection of sub-carrier region (57 kHz).

Input A/D converter itself works with over-sampling and input FIR filtration. It is insensitive for spectral components over 16 kHz on input.

AES-EBU digital input block contains SRC (sampling rate converter). It converts input sampling rates 32-48 kHz to output Fs 32 kHz on which DSPs works. Digital deemphasis (50/15 us and J17) is also incorporated.

DSPs runs program (processing algorithms). Internal calculations are made with 32 bit precision.

After DSPs signal flows into output D/A converter. D/A uses over-sampling and digital FIR (finite impulse response) filtering. This kind of filtration provides minimal overshot and high spectral purity over 16 kHz. It is big advantage, because following stereo encoder has no input filtering on 19 kHz. Without this filtering encoder generates no overshot on its output signal. Reproduction of D/A output is then near perfect. In the pilot region spectral purity of –75 dB is achieved (relatively to full output) or –55 dB (relatively to pilot signal level).

DSPs  processing parameters are send from main microcontroller. It stores data into non-volatile EEPROM memory, communicates with keyboard, display and PC. After every start of Trans-DIGI or change of processing parameters, data are send from microcontroller indo DSPs.

Following explanation is based on processing structure, in other words, what is signal doing inside DSPs. Explanation is based on block diagram (on the end of this manual). In contrast with analog appliance block diagram is not describing physical structure of machine but rather structure of audio processing software. All operations are made in numeric domain.

First block of signal processing is high-pass IIR filter with –3 dB point on 30 Hz. It eliminates sub-acoustic components and DC components of input signal. Spectrum is limited to 16 kHz (by input FIR filter of A/D converter) on other side.

Second block splits signal into two bands by filtering. Crossing point is set to 200 Hz. Low band (30 Hz – 200 Hz) is then processed in low band leveler. Amount of processing in this leveler is affected by Bass Density control. Output from low band leveler is connected to Optimizer spectral balance control and then into Optimizer (summing point).

Processing above 200 Hz is much more complex. Signal is processed in high-band (main) leveler first. Both levelers (bass and main) has three controls. Two of them are the same for both. It is Gate Level and Leveler Ranges. Gate level determines input signal level relatively to A/D clipping (or full scale dBFs). If input signal level falls under gate level, gain of both levelers is frozen. Gate level can be adjusted in range –10 to –30 dB (under full scale of input digital representation).

Second common parameter for both levelers is Leveler Ranges control. This parameter sets range in which levelers adjust gain (for both levelers). If we adjust –20 dB, levelers can increase gain maximally to 20 dB. If we adjust –6, levelers can increase gain maximally to 6 dB.

Third control is Bass+Master release. This control is independent for both levelers. Adjustment of release time affect speed of levelers gain riding. Bass leveler is slower then main. Reason is acceptable distortion, induced by leveler action.

Loud and weak section of program material has similar level after levelers.

After main leveler there is a Density control. Amount of density affects amount of processing in limiter structure. Between density control and limiters, preemphasis is applied to signal. Preemphasis (boost of high audio frequencies) is defined by a norm (50 or 75 us). Use of preemphasis enables increase signal to noise ratio on the output of receiver.

Signal splits into 4 bands now. In every band there is EQ control. It enables control of loading in every band separately. Four band limiter has adjustable release time for every band. Limiter is following by adjustable expander in highest band (3,8 – 15 kHz). Expander prevents noise modulation effects if processing noisy program material.

Optimizer block follows (Optimizer = summing point + master limiter + master clipper). Signals from bands are summed together first. Summing relations are adjustable. Signal then flows via Optimizer drive control. Optimizer drive control determines drive of output limiter. Output limiter cares about final output level setting. Master clipper follows after master limiter. Adjustable Master clipper removes all over-shots from signal.

Relation between limiting and clipping affects character of sound, it will be discussed onward in this manual.

Main expander is placed on final output of structure. If output level falls under adjusted level, main expander increase signal to noise ratio by expanding low level signal.

Final signal is filtered by FIR filter and then D/A converted. This conception provide for high pilot protection, high spectral purity above 16 kHz and no over-shots.

Following stage is internal stereo encoder. It is strongly recommended to use internal encoder, if it is possible. Internal encoder consists from switched discrete A/D converter. This converter runs on four times higher frequency then 38 kHz, on 152 kHz. This „over-sampling" enables to have output LP filter with high cutoff frequency. Such filter practically has not affect frequency characteristic and group delay in the band under 53 kHz. Good stereo separation is then certain for all frequencies in audio band.

Because input filters (FIR) have flat characteristic of group delay, output composite clipper is not required. We have installed only safety output clipper. This action leaves spectrum of composite signal without impurities. Pilot signal is protected and good separation and effective stereo coverage is available. Pilot signal is summed with output of LP filter in summing point. After summing point signal goes into output amplifiers. Output amplifiers can drive 600 ohm load. Both outputs can be adjusted separately. Levels are adjustable in wide range. Third output of encoder is output of pilot signal. Level of this signal is fixed. All outputs of stereo encoder are on BNC connectors.

The whole system is controlled by two microcontrollers. First is system microcontroller. This microcontroller controls main board, DSPs, display and keyboard and communicate with PC via RS 232. Second is stereo encoder microcontroller. It controls all switching and timing inside encoder.

Installation

3.4	Connection

Installation into audio rack is recommended. Around processor free air circulation must be possible.

Output from mixing desk (or output line buffer) connect into XLR inputs resp. AES/EBU input. Composite output on BNC  connect into input of FM exciter, composite link or RDS encoder. If RDS encoder needs for synchronization 19 kHz pilot signal, use 19 kHz output of processor.

If link to transmitter is only by L/R stereo line, use XLR line outputs.

Supplied RS 232 9 pin cable connect to RS 232 port and on other side to free port of PC.

Warning, important !!!!

If using analog inputs, it is necessary to adjust input sensitivity properly. After A/D conversion we cannot remove distortion caused by A/D overloading, or systematically low loading. It means, you have to take maximal care about input sensitivity adjustment. Input sensitivity switches are on rear panel, for each channel independent.

Every input sensitivity switches on rear panel switch to “UP” position.

First step is load processor from mixing desk by full level of program material. It means “faders up”, or maximal output level which really exists on the output of mixing desk during normal on-air operation.

Now, switch piecemeal input sensitivity switches (+3,+6,+9,+12 dB) to low position. In the same time observe LED OVER. If LED start blinks or lights, stop switching and return to previous position. This operation must be repeated for both (L and R) channels. If all is OK, configuration of input sensitivity switches will be the same for both channels. If no, you have to check mixing desk output, cables and (optional) preamplifiers in studio chain for symmetry of output levels.

It is recommended to check configuration of input sensitivity switches on some program materials and moderators.

If input signal has very high level, in this case can be impossible to reduce input sensitivity sufficiently. In other words, all input sensitivity switches are in UP position and LEDs blinks. In this case you have to reduce output level of signal source (mixing desk….). If you cannot reduce output level, you have to add attenuate circuit between signal source and processor. Schematic diagram of simple –10 dB attenuation circuit is on the fig.

3.5	Software installation

3.5.1	Windows 95/98/2000/ME/XP or NT (Note: screen shot is from older model Trans-FM 2005)

Insert CD PHOBOS AUDIO® software into CD drive of PC. Installer starts automatically (If no, start Install.exe from root of CD). Select language and follow instructions of installer. In Programs shortcut TFMdigi appears. (This program window is a main window of application in version 2.09, newest version can slightly vary from this window)

3.5.2	Setup

In the main window of application we can see processing structure and available controls. We have three types of controls (check boxes in menus, push buttons in icon line and sliders in main window). We can move the controls by a mouse (click for check boxes and push buttons, drag for sliders).

In menu „File“ we can save the file, upload the file, and exit the program.

In menu „Port Setup“ we can select communication port. If we have not RS 232 port installed on our computer (notebook), communication via USB is possible. For this case we need reduction cable USB-RS 232 and installed Windows driver for it. In this case select one of ports 3 or 4 (typically 3). Cable and driver can be ordered in some computer shop or from Phobos Audio.

In menu „Preset Sounds“ we can find some basic sound presets. We can use this presets for quick set-up or for further experiments.

In menu „Options“ we can find input select (Digital ON/OFF – Digital OFF r Analog) and activation of Spatial Enhancer.  Latest option is enabling of Day/Night switching over AUX connector on rear panel (AUX connector is installed after 1.9.2002).

In menu „Preemphasis“ we can select between 50 or 75 us preemphasis. In Europe we have to use 50 us, in North America 75 us.

Menu „Test Signal“ enables start test signal 1 kHz in left channel. Right channel will be muted. Signal have 85% of maximal output level in left output, or 88% of maximal output level on MPX output. This signal is useful for channel identification, deviation pre-adjustment and basic check o multiplex characteristics (because spectral components of 1 kHz , 19 kHz and 38 kHz are present on multiplex output). Test signal is generated digitally in DSPs.

Menu “Day/Night” enables save actual configuration into Day/Night internal preset and recalling of this presets. Second function is enable or disable control by a switch. Switch must be connected to AUX connector or rear panel of processor in this case.

Menu “User Presets” enables save and recall actual configuration into four internal User Presets.

Menu „Help“ inform us about starting of this help and gives basic information about program and communication parameters used (1200,N,8,2).

Button “Show/Export/Print/Edit Exact parameters” switch to table of parameters. We can see ecxact values of all parameters (0-255). Press Print for printer output. Press Export for export parameters into commented text file. We can edit parameters by clict into parameter value, press Del and type new value (0-255). Then press “Send Edited Parameters to Main Setting Panel“ button. New parameters will be sent to main application window.

Under Menu bar, there is Icon bar.

First icon with up-arrow enables (by mouse click) send parameters from PC to processor.

Second icon with down-arrow enables upload processors parameters to PC.

Push-buttons named 1-5, follows. It is preset buttons. If we press some of this buttons, appropriate file named preset1.DXC - preset5.DXC is uploaded from disk. We can create this preset files by File menu, if we save file with name preset1.DXC – preset5.DXC.

Button Show (Print)/Edit Exact parameters is right from icons. It skips into new window in which we can see exact values of all parameters. Print is also possible. If we edit parameters, we can send them back to main window by pressing Send edited parameters to main panel button.

We can see information line right from icons. Name of actual DXC file is indicated in this line. Actual.DXC is open after program starts. This file was saved in moment of program exit. It means, program starts in the same configuration (position of controls is not lost in program exit time).

 Review of controls

Review of controls follows. Every control is discussed for some program types.

3.5.3	Gate lev / -10...-30 dB /

Sets the gate level in dB relatively to full loading (0 dBFs). Setting for low levels (example –30dB) enables leveler action in wider range of input levels. If respecting of original dynamic relations is first goal of your processing, best is set this parameter to higher level (example –10 dB). Gate level is common for both levelers, main and bass. For „Hit, Pop or Rock“ format set –25 –30dB. For classical music set –10-15dB.

3.5.4	Lev. ranges / 6 – 20 dB /

Sets range of levelers action in dB. More action reduces original dynamic of program strongly. If low dynamic reduction is goal of your processing, set low range (example 6 dB). For big reduction of dynamic, set 20 dB. In case of „Hit, Pop or Rock“ format set 20dB. For classical music set 6-10dB. Action of levelers is affected by control Gate level too.

3.5.5	Bass + Master release / 0.5 - 10 s (bass), 0.25 - 6 s (master) /

Sets release time of levelers. Release time is time for 10 dB gain advance (if Gate is not activated).

Setting to low time results in better tracking of input signal level variations. We recommend to use this setting in case of "Hit, Pop or Rock".

Setting to mid and high release times is good for universal program and classical music.

Action of levelers is affected by  Lev. ranges and Gate lev too.

If we set bass release time to minimal time (0.5 s), it increases bass punch. It could be very positive for “Hit, Pop or Rock" formats. We can successfully combine minimal bass release time with medium (2–3 s) main release time for this formats.

We cannot recommend adjust highly different release times for main and bass leveler for universal and classical format

3.5.6	Bass density / 0 – 12 dB /

Density for bass affect amount of bass. This control enables drive bass leveler from 0 to +12dB.

For original sound character on bass we recommend set this control to 0 – 3 dB. For "hit pop or rock" format set 6 – 12 dB, for big bass punch (in combination with Bass release to 0,5 s).

3.5.7	Density / 0 – 12 dB /

Sets amount of loading of four-band limiter and then density of sound.

Density is amount of sound compression in four band limiter.

If density is high / 6 – 12 dB /, sound is highly limited in four band limiter. Dynamic of original program is strongly reduced. All is loud.

Spectral normalization takes place. Every program material after processing have similar audio spectrum. It is typical setting for loud “pop, hit or rock” format. For long term listening we recommend do not use density over 9 dB, to avoid listening fatigue. Use such setting if maximal loudness is only goal.

Middle density / 3 – 6 dB /, is good for standard pop or universal program. Spectral normalization is weaker, listening fatigue is not a problem. Such sound is “funny” and transparent. It can be successfully combined with setting of bass density to (example) 9 dB and Bass release to 0,5s. In case of pop format, it gives pleasant sound with good bass punch.

Low density / 0 – 3 dB /, is good for universal program or for classical music. Spectral normalization is practically out of game. Such sound is not loud, but is very natural.

3.5.8	Limiters EQ  / - 6...+ 6 dB /

Sets amount of loading of bands in four-band limiter. In other words, we can selectively affect “density” in entire band. Sound character can be changed by this way. For pop format we can apply +3dB in the band 3,8-15 kHz and +2dB in the band 1,6-3,8 kHz, for example. Some listeners prefers attenuation 1-3 dB in the band 200-600 Hz.

Recommended setting of all EQ controls is 0 dB for universal program. For classical program we can set 1,6-3,8 kHz to 1-2 dB and 3,8-15 kHz to –2 dB.

3.5.9	Release times / 3 – 30ms,x2,x4,x8 /

Sets release times for limiters in four-band limiter in ms. Release time affect dominantly character of limitation of entire band. If we set, for example, all bands to middle position, only 3,8-15 kHz band to minimum (3 ms), high frequencies will be subjectively boosted. This boost is based on increase of average , not peak level of high frequencies. Such way it then effective tool for affecting sound character without change of real spectral balance and without change of peaks.

We can set band 3.8-15 kHz for low release time (3-10ms) for „pop“ format. The same (in less amount) for 1.6-3.8 kHz band. Such setting guaranties open and funny mid and high frequencies.

We recommend set all this controls to mid position for universal and classical format.

3.5.10	Optimizer drive / 0 – 4 dB /

Optimizer is summing point + output limiter. This control sets amount of loading for limiter section of optimizer. If 0 dB is set, limiter is activated only very occasionally. Such setting produces lower loudness, but without risk of spectral gain intermodulation. Setting to +4 dB produces relatively high loudness, but wit some distortion and intermodulation risks. Setting of this control cohere closely with setting of Master clipper. We can say, that final sound set-up must be based on limiting or clipping, not both. If we than set Optimizer drive to more then 2 dB, we cannot set Master clipper to more then 1-2 dB and vice versa. Generally sound based on clipping is more naturally, but with more distortion risk. Many listeners prefer clipping–based sound.

3.5.11	Optimizer spectral balance / - 3.....+ 3 dB /

Sets summing balance in sum section of optimizer. This balance affect spectral balance of output signal.

We recommend to use this controls canny and avoid strongly boosted high frequencies (depends on Release time 3,8-15 kHz and Limiters EQ 3,8-15 kHz too). Boost of high frequencies of audio can cause reduction of effective coverage of transmitter, especially on car radios.

3.5.12	Expanders HF a Main / - 30....- 45 dB, resp. – 40....- 55 dB /

Sets expansion levels for HF or Main expanders respectively. Expanders action cause improvement of signal to noise ratio, especially when noisy program material is processed.

Setting for high levels (-20, resp. –30 dB) cause massive expanders action. For noisy material it is OK, but for good quality material it cause canceling of reverberation and low-level high frequencies. We recommend set both controls to region –40 -45dB (HF) resp. –50-55 dB (Main).

3.5.13	Master Clipper

Mater clipper is output clipping device. It prevents processor output from over-shots. Setting of this control depends on setting Optimizer drive control and was discussed in Optimizer drive section. We cannot set both controls (Optimizer drive and Master clipper) to high levels simultaneously. Sound must be based on limiting or clipping, not both. Clipping based sound is more natural, limiting based sound sounds more “processed” but has lower distortion.

Pots on rear panel enables to adjust output levels for line and composite outputs.

3.5.14	Setting via keyboard

All adjustments can be done without PC, via keyboard and display on front panel.

Setting via keyboard is easy and user friendly. We recommend to study processor structure (from block diagram in section 6 of this manual) prior to adjustments.

Display inform us about parameters we are adjusting. Basic display looks inform us about selected input (Digi means AES/EBU, Analog means line XLR) and Space status (Spatial Enhancer ON/OFF).

First display line has some additional functions. If digital input is activated and no digital signal is on input (no synchronization),  processor automatically switch to analog input. First line of display inform us: DIN Unlock>Analog In. If synchronization on digital input appears, processor automatically switch back to digital input and display returns to basic status. Analog input then can be use like reserve for (for example) digital AES/EBU line failure.

If analog input is activated, first line of display inform us about overloading. In every moment in which some (L or R) channel of input A/D converter is overloaded, display inform us: Overload !!! If such information appears, it is necessary readjust input sensitivity of processor !! Overloading of A/D converter cause not removable distortion and degrade output sound quality drastically.

If we want adjusting parameters, first is necessary to unlock keyboard. It can be done by pressing keys arrow up, arrow left, arrow down in less then 1 s each to other. Keyboard is then unlocked to 3 minutes from last touch. After 3 minutes, keyboard locks automatically.

By pressing ESC number of actual firmware version is displayed:

By pressing ENTER we enters menu for system (presets, spatial enhancer, input select, test tone…..):

By pressing right arrow (>) we enter to system functions, by pressing up arrow to menu parameters.

If we enter system functions (>), following appear on display:

By pressing ENTER enters menu factory presets. First preset is Natural sound. Display looks like:

By ENTER we activate Natural Sound preset, by arrows left/right listing in factory presets (9 presets, same like in PC program). By pressing ESC we enters basic menu. Other system function is User presets.

By pressing ENTER enters menu recalling of User presets. User presets are presets definable by user. Processor has four user presets plus Day and Night preset. First preset is User preset 1.

Display shows:

To recall User preset 1 press ENTER. By arrows left/right we can scroll over User presets 1-4. Press ESC for return to main menu.  Saving of User presets is located in Parameter Setup menu.

Next menu is Spatial enhancer menu:

Selection between ON/OFF by arrows left/right. By pressing ENTER save status into memory.

The same is Digital input ON/OFF menu. If Digital input is OFF, Analog input is ON.

Menu Test Tone is similar. Enables Start/Stop test tone 1000 Hz in left channel. Level of test tone is 85% of maximal level on line output and 88% of maximal level on MPX output.

Next menu is Day/Night switch:

This menu enables switching between day and night setting. Switching is made via pins 1 and 5 of AUX connector on rear panel. In this pins are connected together, night setting takes place. Day and night are presets in USER PRESET menu. AUX connector is installed only on processors made after 1.9.2002.

By pressing up arrow, we enters into parameters menu.

 Display is similar for all parameters. Two examples:

In first line is (on the left side) parameter name, on the right side is parameter value. This value is in the range of 0-255 for all parameters. In the second line are indicated units and range for given parameter. Between its is a bar, indicated value of parameter grafically. Because bar has only 12 segments, exact value is on the right side of first line.

Parameters adjustments is realised by left/right arrows. After adjustment press ENTER. In the moment on pressing ENTER, parameter is saved into memory and we can hear sound with new parameters.

For next parameter press up/down arrow. Following parameters are displayed sequentially.

Gate Level

Levelers Ranges

Bass Release

Master Release

Bass Density

Density

EQ 200-600 Hz

EQ 600-1,6 kHz

EQ 1,6-3,8 kHz

EQ 3,8-15 kHz

Rel 200-600 Hz

Rel 600-1,6 kHz

Rel 1,6-3,8 kHz

Rel 3,8-15 kHz

Optimizer bass

Opt 200-600 Hz

Opt 600-1,6 kHz

Opt 1,6-3,8 kHz

Opt 3,8-15 kHz

Optimizer Drive

M. Clip Drive

HF Expander

Main Expander

Save to user preset enables to us save new parameters into User preset 1-4. Select of preset number by left/right arrows. Save into selected preset by ENTER.

4	Installation of  processor into ON-AIR chain

Processor can be connected to transmitter by MPX output or by L/R line outputs. MPX connection is highly preferable.

Overshot free signal from internal stereo encoder is on this output. Main advantage is using internal FIR filtration and internal preemphasis. Processor itself must be installed close to transmitter or multiplex transmission line.

4.1.1	Adjustment in the case of MPX connection (highly preferable)

One MPX output connect to MPX input of FM exciter or multiplex line. Activate test tone of processor. Because level of this signal is 88%, we have to adjust deviation caused by this signal to 66 kHz. This adjustment can be done by MPX output level control on rear panel, or by sensitivity control of exciter. If we incorporate transmission line, set loading of this line to minus 1,1 dB under maximal level.

After this basic adjustment, test deviation with real program. Set deviation precisely to 75kHz peak on real program.

4.1.2	Adjustment in the case of L and R line outputs (not recommended)

Connect both output lines from processor to L and R inputs of transmission line or transmitter. Activate test tone of processor. Because level of this signal in 85% of maximal level on line outputs, we have to adjust deviation caused by this signal (and pilot from external stereo encoder) to 66 kHz. This adjustment can be done by L and R output level controls on rear panel, or by sensitivity control of line or external stereo encoder. If we incorporate transmission line, set loading of this line to –1,4 dB under maximal level.

After this basic adjustment, test deviation with real program. Set deviation precisely to 75 kHz peak on real program.

Warning !

Adjustment of processor / transmitter needs measuring equipment and some skill. It is strongly recommended to call some installation company for this work.

5	Technical parameters

Input impedance of XLR input:	> 10 kohm

XLR input sensitivity (adjustable-typically):	1.55 V

Input impedance of AES-EBU input:	110 ohm

Acceptable sampling rate on AES-EBU input:	32,44.1 a 48 kHz

Output impedance of XLR outputs:	60 ohm, minimal acceptable load 600 ohm

Output impedance of MPX BNC outputs:	75 ohm, minimum acceptable load 600 ohm,recommended 10 kohm

Output level on XLR outputs (adjustable-typically):	1.55 V

Output level on BNC MPX outputs (adjustable-typically):	8 VPP (280 mV pilot signal)

Pilot signal:	19 kHz +/- 1,5 Hz

Pilot spectral protection (relatively to –20 dB):	-55 dB

L and R line outputs – range of gain adjustment:	15 dB

MPX BNC composite outputs – range of gain adjustment:	+3/- 40 dB

Processing of signal: 	5 freq. bands

S/N ratio of entire processing:	91 dB (30 Hz-15 kHz)

Temperature conditions:	0-40 Celsius

Ext. reset on AUX port between pins	5,3

Day/night switch on AUX port between pins	8,3

Power source:	230 V, 50Hz  +/- 10%

We cannot specify more parameters, because this cannot be described by standard measurement procedures. Moreover processor is adjustable by software and has more then 16 millions of parameters combinations.

If appliance works not correctly, disconnect it from power source immediately and call authorised service center. Do not open cover, do not use serviceable parts inside. High and dangerous voltage inside ! Refer servicing to authorised service personnel only. Device is static sensitive inside ! Device is moisture-sensitive !

6	Trans-DIGI 2007 DIGITAL block diagram

7	Warranty

 Manufacturer gives to you warranty of 2 years. Warranty do not cover natural and other disasters and non-professional use of this appliance.

Producer: Phobos Engineering s.r.o.

   U Druzstva Zivot 4

  140 00 Prague 4

  Czech Republic

tel/fax:	(+420) 241413650, (+420) 241412938

website:www.phobosaudio.cz

e-mai     :phobosaudio@email.cz

Registration / DIČ:	CZ27159639